From fa8e30c1a8b467c0940b759f011b6add34019a6c Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Sun, 16 May 2021 13:34:04 +0100 Subject: [PATCH 01/47] Test commit of rtaudio This commit doesn't actually use RTAudio for anything other than enumerating devices and is a proof of concept for each supported platform. --- audiohandler.h | 12 + rtaudio/ChangeLog | 10 + rtaudio/LICENSE | 27 + rtaudio/README.md | 38 + rtaudio/RtAudio.cpp | 10911 +++++++++++++++++++++++++++++++++++++++ rtaudio/RtAudio.h | 1185 +++++ wfmain.cpp | 49 + wfmain.h | 2 + wfview.pro | 23 +- wfview.vcxproj | 4 +- wfview.vcxproj.filters | 8 + 11 files changed, 12266 insertions(+), 3 deletions(-) create mode 100644 rtaudio/ChangeLog create mode 100644 rtaudio/LICENSE create mode 100644 rtaudio/README.md create mode 100644 rtaudio/RtAudio.cpp create mode 100644 rtaudio/RtAudio.h diff --git a/audiohandler.h b/audiohandler.h index b8fc9425..e7145c72 100644 --- a/audiohandler.h +++ b/audiohandler.h @@ -1,22 +1,34 @@ #ifndef AUDIOHANDLER_H #define AUDIOHANDLER_H +#define USE_RTAUDIO +#define USE_QTAUDIO + #include #include #include #include #include + +#ifdef USE_QTAUDIO #include #include #include #include #include +#endif + +#ifdef USE_RTAUDIO +#include "rtaudio/RtAudio.h" +#endif + #include #include #include #include "resampler/speex_resampler.h" + #include //#define BUFFER_SIZE (32*1024) diff --git a/rtaudio/ChangeLog b/rtaudio/ChangeLog new file mode 100644 index 00000000..6282a761 --- /dev/null +++ b/rtaudio/ChangeLog @@ -0,0 +1,10 @@ +30.04.2018 + * Additions and fixes for realtime operation: (Tim E. Real of MusE) + - Added realtime operation to Pulse driver. + - Fixed ALSA realtime support. Attributes are once again all set + in probeDeviceOpen(). + - Fixed OSS realtime support. Same mods as done to ALSA driver. + OSS untested, but should work, it's the same code. + - A diagnostic message (streamed to cerr) in each of the callback + handlers informs the user if realtime is really running. + diff --git a/rtaudio/LICENSE b/rtaudio/LICENSE new file mode 100644 index 00000000..c55dbfa2 --- /dev/null +++ b/rtaudio/LICENSE @@ -0,0 +1,27 @@ + +RtAudio: a set of realtime audio i/o C++ classes +Copyright (c) 2001-2019 Gary P. Scavone + +Permission is hereby granted, free of charge, to any person +obtaining a copy of this software and associated documentation files +(the "Software"), to deal in the Software without restriction, +including without limitation the rights to use, copy, modify, merge, +publish, distribute, sublicense, and/or sell copies of the Software, +and to permit persons to whom the Software is furnished to do so, +subject to the following conditions: + +The above copyright notice and this permission notice shall be +included in all copies or substantial portions of the Software. + +Any person wishing to distribute modifications to the Software is +asked to send the modifications to the original developer so that +they can be incorporated into the canonical version. This is, +however, not a binding provision of this license. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, +EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF +MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. +IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR +ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF +CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. diff --git a/rtaudio/README.md b/rtaudio/README.md new file mode 100644 index 00000000..d58883b9 --- /dev/null +++ b/rtaudio/README.md @@ -0,0 +1,38 @@ +# RtAudio + +[![Build Status](https://travis-ci.org/thestk/rtaudio.svg?branch=master)](https://travis-ci.org/thestk/rtaudio) + +A set of C++ classes that provide a common API for realtime audio input/output across Linux (native ALSA, JACK, PulseAudio and OSS), Macintosh OS X (CoreAudio and JACK), and Windows (DirectSound, ASIO and WASAPI) operating systems. + +By Gary P. Scavone, 2001-2019 (and many other developers!) + +This distribution of RtAudio contains the following: + +- doc: RtAudio documentation (see doc/html/index.html) +- tests: example RtAudio programs +- include: header and source files necessary for ASIO, DS & OSS compilation +- tests/Windows: Visual C++ .net test program workspace and projects + +## Overview + +RtAudio is a set of C++ classes that provides a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, JACK, PulseAudio and OSS), Macintosh OS X and Windows (DirectSound, ASIO and WASAPI) operating systems. RtAudio significantly simplifies the process of interacting with computer audio hardware. It was designed with the following objectives: + + - object-oriented C++ design + - simple, common API across all supported platforms + - only one source and one header file for easy inclusion in programming projects + - allow simultaneous multi-api support + - support dynamic connection of devices + - provide extensive audio device parameter control + - allow audio device capability probing + - automatic internal conversion for data format, channel number compensation, (de)interleaving, and byte-swapping + +RtAudio incorporates the concept of audio streams, which represent audio output (playback) and/or input (recording). Available audio devices and their capabilities can be enumerated and then specified when opening a stream. Where applicable, multiple API support can be compiled and a particular API specified when creating an RtAudio instance. See the \ref apinotes section for information specific to each of the supported audio APIs. + +## Further Reading + +For complete documentation on RtAudio, see the doc directory of the distribution or surf to http://www.music.mcgill.ca/~gary/rtaudio/. + + +## Legal and ethical: + +The RtAudio license is similar to the MIT License. Please see [LICENSE](LICENSE). diff --git a/rtaudio/RtAudio.cpp b/rtaudio/RtAudio.cpp new file mode 100644 index 00000000..3e64b85f --- /dev/null +++ b/rtaudio/RtAudio.cpp @@ -0,0 +1,10911 @@ +/************************************************************************/ +/*! \class RtAudio + \brief Realtime audio i/o C++ classes. + + RtAudio provides a common API (Application Programming Interface) + for realtime audio input/output across Linux (native ALSA, Jack, + and OSS), Macintosh OS X (CoreAudio and Jack), and Windows + (DirectSound, ASIO and WASAPI) operating systems. + + RtAudio GitHub site: https://github.com/thestk/rtaudio + RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ + + RtAudio: realtime audio i/o C++ classes + Copyright (c) 2001-2019 Gary P. Scavone + + Permission is hereby granted, free of charge, to any person + obtaining a copy of this software and associated documentation files + (the "Software"), to deal in the Software without restriction, + including without limitation the rights to use, copy, modify, merge, + publish, distribute, sublicense, and/or sell copies of the Software, + and to permit persons to whom the Software is furnished to do so, + subject to the following conditions: + + The above copyright notice and this permission notice shall be + included in all copies or substantial portions of the Software. + + Any person wishing to distribute modifications to the Software is + asked to send the modifications to the original developer so that + they can be incorporated into the canonical version. This is, + however, not a binding provision of this license. + + THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR + ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF + CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION + WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. +*/ +/************************************************************************/ + +// RtAudio: Version 5.1.0 + +#include "RtAudio.h" +#include +#include +#include +#include +#include +#include + +// Static variable definitions. +const unsigned int RtApi::MAX_SAMPLE_RATES = 14; +const unsigned int RtApi::SAMPLE_RATES[] = { + 4000, 5512, 8000, 9600, 11025, 16000, 22050, + 32000, 44100, 48000, 88200, 96000, 176400, 192000 +}; + +#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) + #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) + #define MUTEX_DESTROY(A) DeleteCriticalSection(A) + #define MUTEX_LOCK(A) EnterCriticalSection(A) + #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) + + #include "tchar.h" + + template inline + std::string convertCharPointerToStdString(const T *text); + + template<> inline + std::string convertCharPointerToStdString(const char *text) + { + return std::string(text); + } + + template<> inline + std::string convertCharPointerToStdString(const wchar_t *text) + { + int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL); + std::string s( length-1, '\0' ); + WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL); + return s; + } + +#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) + // pthread API + #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) + #define MUTEX_DESTROY(A) pthread_mutex_destroy(A) + #define MUTEX_LOCK(A) pthread_mutex_lock(A) + #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A) +#else + #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions + #define MUTEX_DESTROY(A) abs(*A) // dummy definitions +#endif + +// *************************************************** // +// +// RtAudio definitions. +// +// *************************************************** // + +std::string RtAudio :: getVersion( void ) +{ + return RTAUDIO_VERSION; +} + +// Define API names and display names. +// Must be in same order as API enum. +extern "C" { +const char* rtaudio_api_names[][2] = { + { "unspecified" , "Unknown" }, + { "alsa" , "ALSA" }, + { "pulse" , "Pulse" }, + { "oss" , "OpenSoundSystem" }, + { "jack" , "Jack" }, + { "core" , "CoreAudio" }, + { "wasapi" , "WASAPI" }, + { "asio" , "ASIO" }, + { "ds" , "DirectSound" }, + { "dummy" , "Dummy" }, +}; +const unsigned int rtaudio_num_api_names = + sizeof(rtaudio_api_names)/sizeof(rtaudio_api_names[0]); + +// The order here will control the order of RtAudio's API search in +// the constructor. +extern "C" const RtAudio::Api rtaudio_compiled_apis[] = { +#if defined(__UNIX_JACK__) + RtAudio::UNIX_JACK, +#endif +#if defined(__LINUX_PULSE__) + RtAudio::LINUX_PULSE, +#endif +#if defined(__LINUX_ALSA__) + RtAudio::LINUX_ALSA, +#endif +#if defined(__LINUX_OSS__) + RtAudio::LINUX_OSS, +#endif +#if defined(__WINDOWS_ASIO__) + RtAudio::WINDOWS_ASIO, +#endif +#if defined(__WINDOWS_WASAPI__) + RtAudio::WINDOWS_WASAPI, +#endif +#if defined(__WINDOWS_DS__) + RtAudio::WINDOWS_DS, +#endif +#if defined(__MACOSX_CORE__) + RtAudio::MACOSX_CORE, +#endif +#if defined(__RTAUDIO_DUMMY__) + RtAudio::RTAUDIO_DUMMY, +#endif + RtAudio::UNSPECIFIED, +}; +extern "C" const unsigned int rtaudio_num_compiled_apis = + sizeof(rtaudio_compiled_apis)/sizeof(rtaudio_compiled_apis[0])-1; +} + +// This is a compile-time check that rtaudio_num_api_names == RtAudio::NUM_APIS. +// If the build breaks here, check that they match. +template class StaticAssert { private: StaticAssert() {} }; +template<> class StaticAssert{ public: StaticAssert() {} }; +class StaticAssertions { StaticAssertions() { + StaticAssert(); +}}; + +void RtAudio :: getCompiledApi( std::vector &apis ) +{ + apis = std::vector(rtaudio_compiled_apis, + rtaudio_compiled_apis + rtaudio_num_compiled_apis); +} + +std::string RtAudio :: getApiName( RtAudio::Api api ) +{ + if (api < 0 || api >= RtAudio::NUM_APIS) + return ""; + return rtaudio_api_names[api][0]; +} + +std::string RtAudio :: getApiDisplayName( RtAudio::Api api ) +{ + if (api < 0 || api >= RtAudio::NUM_APIS) + return "Unknown"; + return rtaudio_api_names[api][1]; +} + +RtAudio::Api RtAudio :: getCompiledApiByName( const std::string &name ) +{ + unsigned int i=0; + for (i = 0; i < rtaudio_num_compiled_apis; ++i) + if (name == rtaudio_api_names[rtaudio_compiled_apis[i]][0]) + return rtaudio_compiled_apis[i]; + return RtAudio::UNSPECIFIED; +} + +void RtAudio :: openRtApi( RtAudio::Api api ) +{ + if ( rtapi_ ) + delete rtapi_; + rtapi_ = 0; + +#if defined(__UNIX_JACK__) + if ( api == UNIX_JACK ) + rtapi_ = new RtApiJack(); +#endif +#if defined(__LINUX_ALSA__) + if ( api == LINUX_ALSA ) + rtapi_ = new RtApiAlsa(); +#endif +#if defined(__LINUX_PULSE__) + if ( api == LINUX_PULSE ) + rtapi_ = new RtApiPulse(); +#endif +#if defined(__LINUX_OSS__) + if ( api == LINUX_OSS ) + rtapi_ = new RtApiOss(); +#endif +#if defined(__WINDOWS_ASIO__) + if ( api == WINDOWS_ASIO ) + rtapi_ = new RtApiAsio(); +#endif +#if defined(__WINDOWS_WASAPI__) + if ( api == WINDOWS_WASAPI ) + rtapi_ = new RtApiWasapi(); +#endif +#if defined(__WINDOWS_DS__) + if ( api == WINDOWS_DS ) + rtapi_ = new RtApiDs(); +#endif +#if defined(__MACOSX_CORE__) + if ( api == MACOSX_CORE ) + rtapi_ = new RtApiCore(); +#endif +#if defined(__RTAUDIO_DUMMY__) + if ( api == RTAUDIO_DUMMY ) + rtapi_ = new RtApiDummy(); +#endif +} + +RtAudio :: RtAudio( RtAudio::Api api ) +{ + rtapi_ = 0; + + if ( api != UNSPECIFIED ) { + // Attempt to open the specified API. + openRtApi( api ); + if ( rtapi_ ) return; + + // No compiled support for specified API value. Issue a debug + // warning and continue as if no API was specified. + std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl; + } + + // Iterate through the compiled APIs and return as soon as we find + // one with at least one device or we reach the end of the list. + std::vector< RtAudio::Api > apis; + getCompiledApi( apis ); + for ( unsigned int i=0; igetDeviceCount() ) break; + } + + if ( rtapi_ ) return; + + // It should not be possible to get here because the preprocessor + // definition __RTAUDIO_DUMMY__ is automatically defined if no + // API-specific definitions are passed to the compiler. But just in + // case something weird happens, we'll thow an error. + std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n"; + throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) ); +} + +RtAudio :: ~RtAudio() +{ + if ( rtapi_ ) + delete rtapi_; +} + +void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, + RtAudioCallback callback, void *userData, + RtAudio::StreamOptions *options, + RtAudioErrorCallback errorCallback ) +{ + return rtapi_->openStream( outputParameters, inputParameters, format, + sampleRate, bufferFrames, callback, + userData, options, errorCallback ); +} + +// *************************************************** // +// +// Public RtApi definitions (see end of file for +// private or protected utility functions). +// +// *************************************************** // + +RtApi :: RtApi() +{ + stream_.state = STREAM_CLOSED; + stream_.mode = UNINITIALIZED; + stream_.apiHandle = 0; + stream_.userBuffer[0] = 0; + stream_.userBuffer[1] = 0; + MUTEX_INITIALIZE( &stream_.mutex ); + showWarnings_ = true; + firstErrorOccurred_ = false; +} + +RtApi :: ~RtApi() +{ + MUTEX_DESTROY( &stream_.mutex ); +} + +void RtApi :: openStream( RtAudio::StreamParameters *oParams, + RtAudio::StreamParameters *iParams, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, + RtAudioCallback callback, void *userData, + RtAudio::StreamOptions *options, + RtAudioErrorCallback errorCallback ) +{ + if ( stream_.state != STREAM_CLOSED ) { + errorText_ = "RtApi::openStream: a stream is already open!"; + error( RtAudioError::INVALID_USE ); + return; + } + + // Clear stream information potentially left from a previously open stream. + clearStreamInfo(); + + if ( oParams && oParams->nChannels < 1 ) { + errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one."; + error( RtAudioError::INVALID_USE ); + return; + } + + if ( iParams && iParams->nChannels < 1 ) { + errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one."; + error( RtAudioError::INVALID_USE ); + return; + } + + if ( oParams == NULL && iParams == NULL ) { + errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!"; + error( RtAudioError::INVALID_USE ); + return; + } + + if ( formatBytes(format) == 0 ) { + errorText_ = "RtApi::openStream: 'format' parameter value is undefined."; + error( RtAudioError::INVALID_USE ); + return; + } + + unsigned int nDevices = getDeviceCount(); + unsigned int oChannels = 0; + if ( oParams ) { + oChannels = oParams->nChannels; + if ( oParams->deviceId >= nDevices ) { + errorText_ = "RtApi::openStream: output device parameter value is invalid."; + error( RtAudioError::INVALID_USE ); + return; + } + } + + unsigned int iChannels = 0; + if ( iParams ) { + iChannels = iParams->nChannels; + if ( iParams->deviceId >= nDevices ) { + errorText_ = "RtApi::openStream: input device parameter value is invalid."; + error( RtAudioError::INVALID_USE ); + return; + } + } + + bool result; + + if ( oChannels > 0 ) { + + result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel, + sampleRate, format, bufferFrames, options ); + if ( result == false ) { + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + + if ( iChannels > 0 ) { + + result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel, + sampleRate, format, bufferFrames, options ); + if ( result == false ) { + if ( oChannels > 0 ) closeStream(); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + + stream_.callbackInfo.callback = (void *) callback; + stream_.callbackInfo.userData = userData; + stream_.callbackInfo.errorCallback = (void *) errorCallback; + + if ( options ) options->numberOfBuffers = stream_.nBuffers; + stream_.state = STREAM_STOPPED; +} + +unsigned int RtApi :: getDefaultInputDevice( void ) +{ + // Should be implemented in subclasses if possible. + return 0; +} + +unsigned int RtApi :: getDefaultOutputDevice( void ) +{ + // Should be implemented in subclasses if possible. + return 0; +} + +void RtApi :: closeStream( void ) +{ + // MUST be implemented in subclasses! + return; +} + +bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/, + unsigned int /*firstChannel*/, unsigned int /*sampleRate*/, + RtAudioFormat /*format*/, unsigned int * /*bufferSize*/, + RtAudio::StreamOptions * /*options*/ ) +{ + // MUST be implemented in subclasses! + return FAILURE; +} + +void RtApi :: tickStreamTime( void ) +{ + // Subclasses that do not provide their own implementation of + // getStreamTime should call this function once per buffer I/O to + // provide basic stream time support. + + stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate ); + +#if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); +#endif +} + +long RtApi :: getStreamLatency( void ) +{ + verifyStream(); + + long totalLatency = 0; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + totalLatency = stream_.latency[0]; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) + totalLatency += stream_.latency[1]; + + return totalLatency; +} + +double RtApi :: getStreamTime( void ) +{ + verifyStream(); + +#if defined( HAVE_GETTIMEOFDAY ) + // Return a very accurate estimate of the stream time by + // adding in the elapsed time since the last tick. + struct timeval then; + struct timeval now; + + if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 ) + return stream_.streamTime; + + gettimeofday( &now, NULL ); + then = stream_.lastTickTimestamp; + return stream_.streamTime + + ((now.tv_sec + 0.000001 * now.tv_usec) - + (then.tv_sec + 0.000001 * then.tv_usec)); +#else + return stream_.streamTime; +#endif +} + +void RtApi :: setStreamTime( double time ) +{ + verifyStream(); + + if ( time >= 0.0 ) + stream_.streamTime = time; +#if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); +#endif +} + +unsigned int RtApi :: getStreamSampleRate( void ) +{ + verifyStream(); + + return stream_.sampleRate; +} + + +// *************************************************** // +// +// OS/API-specific methods. +// +// *************************************************** // + +#if defined(__MACOSX_CORE__) + +#include + +// The OS X CoreAudio API is designed to use a separate callback +// procedure for each of its audio devices. A single RtAudio duplex +// stream using two different devices is supported here, though it +// cannot be guaranteed to always behave correctly because we cannot +// synchronize these two callbacks. +// +// A property listener is installed for over/underrun information. +// However, no functionality is currently provided to allow property +// listeners to trigger user handlers because it is unclear what could +// be done if a critical stream parameter (buffer size, sample rate, +// device disconnect) notification arrived. The listeners entail +// quite a bit of extra code and most likely, a user program wouldn't +// be prepared for the result anyway. However, we do provide a flag +// to the client callback function to inform of an over/underrun. + +// A structure to hold various information related to the CoreAudio API +// implementation. +struct CoreHandle { + AudioDeviceID id[2]; // device ids +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceIOProcID procId[2]; +#endif + UInt32 iStream[2]; // device stream index (or first if using multiple) + UInt32 nStreams[2]; // number of streams to use + bool xrun[2]; + char *deviceBuffer; + pthread_cond_t condition; + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + + CoreHandle() + :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +RtApiCore:: RtApiCore() +{ +#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER ) + // This is a largely undocumented but absolutely necessary + // requirement starting with OS-X 10.6. If not called, queries and + // updates to various audio device properties are not handled + // correctly. + CFRunLoopRef theRunLoop = NULL; + AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop); + if ( result != noErr ) { + errorText_ = "RtApiCore::RtApiCore: error setting run loop property!"; + error( RtAudioError::WARNING ); + } +#endif +} + +RtApiCore :: ~RtApiCore() +{ + // The subclass destructor gets called before the base class + // destructor, so close an existing stream before deallocating + // apiDeviceId memory. + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiCore :: getDeviceCount( void ) +{ + // Find out how many audio devices there are, if any. + UInt32 dataSize; + AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!"; + error( RtAudioError::WARNING ); + return 0; + } + + return dataSize / sizeof( AudioDeviceID ); +} + +unsigned int RtApiCore :: getDefaultInputDevice( void ) +{ + unsigned int nDevices = getDeviceCount(); + if ( nDevices <= 1 ) return 0; + + AudioDeviceID id; + UInt32 dataSize = sizeof( AudioDeviceID ); + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device."; + error( RtAudioError::WARNING ); + return 0; + } + + dataSize *= nDevices; + AudioDeviceID deviceList[ nDevices ]; + property.mSelector = kAudioHardwarePropertyDevices; + result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs."; + error( RtAudioError::WARNING ); + return 0; + } + + for ( unsigned int i=0; i= nDevices ) { + errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, + 0, NULL, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs."; + error( RtAudioError::WARNING ); + return info; + } + + AudioDeviceID id = deviceList[ device ]; + + // Get the device name. + info.name.erase(); + CFStringRef cfname; + dataSize = sizeof( CFStringRef ); + property.mSelector = kAudioObjectPropertyManufacturer; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); + int length = CFStringGetLength(cfname); + char *mname = (char *)malloc(length * 3 + 1); +#if defined( UNICODE ) || defined( _UNICODE ) + CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8); +#else + CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding()); +#endif + info.name.append( (const char *)mname, strlen(mname) ); + info.name.append( ": " ); + CFRelease( cfname ); + free(mname); + + property.mSelector = kAudioObjectPropertyName; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); + length = CFStringGetLength(cfname); + char *name = (char *)malloc(length * 3 + 1); +#if defined( UNICODE ) || defined( _UNICODE ) + CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8); +#else + CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding()); +#endif + info.name.append( (const char *)name, strlen(name) ); + CFRelease( cfname ); + free(name); + + // Get the output stream "configuration". + AudioBufferList *bufferList = nil; + property.mSelector = kAudioDevicePropertyStreamConfiguration; + property.mScope = kAudioDevicePropertyScopeOutput; + // property.mElement = kAudioObjectPropertyElementWildcard; + dataSize = 0; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != noErr || dataSize == 0 ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList."; + error( RtAudioError::WARNING ); + return info; + } + + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); + if ( result != noErr || dataSize == 0 ) { + free( bufferList ); + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Get output channel information. + unsigned int i, nStreams = bufferList->mNumberBuffers; + for ( i=0; imBuffers[i].mNumberChannels; + free( bufferList ); + + // Get the input stream "configuration". + property.mScope = kAudioDevicePropertyScopeInput; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != noErr || dataSize == 0 ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList."; + error( RtAudioError::WARNING ); + return info; + } + + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); + if (result != noErr || dataSize == 0) { + free( bufferList ); + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Get input channel information. + nStreams = bufferList->mNumberBuffers; + for ( i=0; imBuffers[i].mNumberChannels; + free( bufferList ); + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Probe the device sample rates. + bool isInput = false; + if ( info.outputChannels == 0 ) isInput = true; + + // Determine the supported sample rates. + property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates; + if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != kAudioHardwareNoError || dataSize == 0 ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + UInt32 nRanges = dataSize / sizeof( AudioValueRange ); + AudioValueRange rangeList[ nRanges ]; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList ); + if ( result != kAudioHardwareNoError ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // The sample rate reporting mechanism is a bit of a mystery. It + // seems that it can either return individual rates or a range of + // rates. I assume that if the min / max range values are the same, + // then that represents a single supported rate and if the min / max + // range values are different, the device supports an arbitrary + // range of values (though there might be multiple ranges, so we'll + // use the most conservative range). + Float64 minimumRate = 1.0, maximumRate = 10000000000.0; + bool haveValueRange = false; + info.sampleRates.clear(); + for ( UInt32 i=0; i info.preferredSampleRate ) ) + info.preferredSampleRate = tmpSr; + + } else { + haveValueRange = true; + if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum; + if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum; + } + } + + if ( haveValueRange ) { + for ( unsigned int k=0; k= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) { + info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + } + } + } + + // Sort and remove any redundant values + std::sort( info.sampleRates.begin(), info.sampleRates.end() ); + info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() ); + + if ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // CoreAudio always uses 32-bit floating point data for PCM streams. + // Thus, any other "physical" formats supported by the device are of + // no interest to the client. + info.nativeFormats = RTAUDIO_FLOAT32; + + if ( info.outputChannels > 0 ) + if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; + if ( info.inputChannels > 0 ) + if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; + + info.probed = true; + return info; +} + +static OSStatus callbackHandler( AudioDeviceID inDevice, + const AudioTimeStamp* /*inNow*/, + const AudioBufferList* inInputData, + const AudioTimeStamp* /*inInputTime*/, + AudioBufferList* outOutputData, + const AudioTimeStamp* /*inOutputTime*/, + void* infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + + RtApiCore *object = (RtApiCore *) info->object; + if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false ) + return kAudioHardwareUnspecifiedError; + else + return kAudioHardwareNoError; +} + +static OSStatus xrunListener( AudioObjectID /*inDevice*/, + UInt32 nAddresses, + const AudioObjectPropertyAddress properties[], + void* handlePointer ) +{ + CoreHandle *handle = (CoreHandle *) handlePointer; + for ( UInt32 i=0; ixrun[1] = true; + else + handle->xrun[0] = true; + } + } + + return kAudioHardwareNoError; +} + +static OSStatus rateListener( AudioObjectID inDevice, + UInt32 /*nAddresses*/, + const AudioObjectPropertyAddress /*properties*/[], + void* ratePointer ) +{ + Float64 *rate = (Float64 *) ratePointer; + UInt32 dataSize = sizeof( Float64 ); + AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate ); + return kAudioHardwareNoError; +} + +bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiCore::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, + 0, NULL, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs."; + return FAILURE; + } + + AudioDeviceID id = deviceList[ device ]; + + // Setup for stream mode. + bool isInput = false; + if ( mode == INPUT ) { + isInput = true; + property.mScope = kAudioDevicePropertyScopeInput; + } + else + property.mScope = kAudioDevicePropertyScopeOutput; + + // Get the stream "configuration". + AudioBufferList *bufferList = nil; + dataSize = 0; + property.mSelector = kAudioDevicePropertyStreamConfiguration; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != noErr || dataSize == 0 ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList."; + return FAILURE; + } + + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); + if (result != noErr || dataSize == 0) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Search for one or more streams that contain the desired number of + // channels. CoreAudio devices can have an arbitrary number of + // streams and each stream can have an arbitrary number of channels. + // For each stream, a single buffer of interleaved samples is + // provided. RtAudio prefers the use of one stream of interleaved + // data or multiple consecutive single-channel streams. However, we + // now support multiple consecutive multi-channel streams of + // interleaved data as well. + UInt32 iStream, offsetCounter = firstChannel; + UInt32 nStreams = bufferList->mNumberBuffers; + bool monoMode = false; + bool foundStream = false; + + // First check that the device supports the requested number of + // channels. + UInt32 deviceChannels = 0; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + + if ( deviceChannels < ( channels + firstChannel ) ) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Look for a single stream meeting our needs. + UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + if ( streamChannels >= channels + offsetCounter ) { + firstStream = iStream; + channelOffset = offsetCounter; + foundStream = true; + break; + } + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + } + + // If we didn't find a single stream above, then we should be able + // to meet the channel specification with multiple streams. + if ( foundStream == false ) { + monoMode = true; + offsetCounter = firstChannel; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + } + + firstStream = iStream; + channelOffset = offsetCounter; + Int32 channelCounter = channels + offsetCounter - streamChannels; + + if ( streamChannels > 1 ) monoMode = false; + while ( channelCounter > 0 ) { + streamChannels = bufferList->mBuffers[++iStream].mNumberChannels; + if ( streamChannels > 1 ) monoMode = false; + channelCounter -= streamChannels; + streamCount++; + } + } + + free( bufferList ); + + // Determine the buffer size. + AudioValueRange bufferRange; + dataSize = sizeof( AudioValueRange ); + property.mSelector = kAudioDevicePropertyBufferFrameSizeRange; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange ); + + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum; + else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum; + + // Set the buffer size. For multiple streams, I'm assuming we only + // need to make this setting for the master channel. + UInt32 theSize = (UInt32) *bufferSize; + dataSize = sizeof( UInt32 ); + property.mSelector = kAudioDevicePropertyBufferFrameSize; + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize ); + + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + *bufferSize = theSize; + if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 1; + + // Try to set "hog" mode ... it's not clear to me this is working. + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) { + pid_t hog_pid; + dataSize = sizeof( hog_pid ); + property.mSelector = kAudioDevicePropertyHogMode; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( hog_pid != getpid() ) { + hog_pid = getpid(); + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + } + + // Check and if necessary, change the sample rate for the device. + Float64 nominalRate; + dataSize = sizeof( Float64 ); + property.mSelector = kAudioDevicePropertyNominalSampleRate; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Only change the sample rate if off by more than 1 Hz. + if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) { + + // Set a property listener for the sample rate change + Float64 reportedRate = 0.0; + AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + nominalRate = (Float64) sampleRate; + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate ); + if ( result != noErr ) { + AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Now wait until the reported nominal rate is what we just set. + UInt32 microCounter = 0; + while ( reportedRate != nominalRate ) { + microCounter += 5000; + if ( microCounter > 5000000 ) break; + usleep( 5000 ); + } + + // Remove the property listener. + AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + + if ( microCounter > 5000000 ) { + errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Now set the stream format for all streams. Also, check the + // physical format of the device and change that if necessary. + AudioStreamBasicDescription description; + dataSize = sizeof( AudioStreamBasicDescription ); + property.mSelector = kAudioStreamPropertyVirtualFormat; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the sample rate and data format id. However, only make the + // change if the sample rate is not within 1.0 of the desired + // rate and the format is not linear pcm. + bool updateFormat = false; + if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) { + description.mSampleRate = (Float64) sampleRate; + updateFormat = true; + } + + if ( description.mFormatID != kAudioFormatLinearPCM ) { + description.mFormatID = kAudioFormatLinearPCM; + updateFormat = true; + } + + if ( updateFormat ) { + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Now check the physical format. + property.mSelector = kAudioStreamPropertyPhysicalFormat; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + //std::cout << "Current physical stream format:" << std::endl; + //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl; + //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl; + //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl; + //std::cout << " sample rate = " << description.mSampleRate << std::endl; + + if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) { + description.mFormatID = kAudioFormatLinearPCM; + //description.mSampleRate = (Float64) sampleRate; + AudioStreamBasicDescription testDescription = description; + UInt32 formatFlags; + + // We'll try higher bit rates first and then work our way down. + std::vector< std::pair > physicalFormats; + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger; + physicalFormats.push_back( std::pair( 32, formatFlags ) ); + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; + physicalFormats.push_back( std::pair( 32, formatFlags ) ); + physicalFormats.push_back( std::pair( 24, formatFlags ) ); // 24-bit packed + formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh ); + physicalFormats.push_back( std::pair( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low + formatFlags |= kAudioFormatFlagIsAlignedHigh; + physicalFormats.push_back( std::pair( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; + physicalFormats.push_back( std::pair( 16, formatFlags ) ); + physicalFormats.push_back( std::pair( 8, formatFlags ) ); + + bool setPhysicalFormat = false; + for( unsigned int i=0; iflags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( monoMode == true ) stream_.deviceInterleaved[mode] = false; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( streamCount == 1 ) { + if ( stream_.nUserChannels[mode] > 1 && + stream_.userInterleaved != stream_.deviceInterleaved[mode] ) + stream_.doConvertBuffer[mode] = true; + } + else if ( monoMode && stream_.userInterleaved ) + stream_.doConvertBuffer[mode] = true; + + // Allocate our CoreHandle structure for the stream. + CoreHandle *handle = 0; + if ( stream_.apiHandle == 0 ) { + try { + handle = new CoreHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory."; + goto error; + } + + if ( pthread_cond_init( &handle->condition, NULL ) ) { + errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; + } + else + handle = (CoreHandle *) stream_.apiHandle; + handle->iStream[mode] = firstStream; + handle->nStreams[mode] = streamCount; + handle->id[mode] = id; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) ); + memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + // If possible, we will make use of the CoreAudio stream buffers as + // "device buffers". However, we can't do this if using multiple + // streams. + if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) { + if ( streamCount > 1 ) setConvertInfo( mode, 0 ); + else setConvertInfo( mode, channelOffset ); + } + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device ) + // Only one callback procedure per device. + stream_.mode = DUPLEX; + else { +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] ); +#else + // deprecated in favor of AudioDeviceCreateIOProcID() + result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo ); +#endif + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ")."; + errorText_ = errorStream_.str(); + goto error; + } + if ( stream_.mode == OUTPUT && mode == INPUT ) + stream_.mode = DUPLEX; + else + stream_.mode = mode; + } + + // Setup the device property listener for over/underload. + property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; + result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle ); + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiCore :: closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if (handle) { + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + + property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; + if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) { + errorText_ = "RtApiCore::closeStream(): error removing property listener!"; + error( RtAudioError::WARNING ); + } + +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + if ( stream_.state == STREAM_RUNNING ) + AudioDeviceStop( handle->id[0], handle->procId[0] ); + AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] ); +#else // deprecated behaviour + if ( stream_.state == STREAM_RUNNING ) + AudioDeviceStop( handle->id[0], callbackHandler ); + AudioDeviceRemoveIOProc( handle->id[0], callbackHandler ); +#endif + } + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + if (handle) { + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + + property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; + if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) { + errorText_ = "RtApiCore::closeStream(): error removing property listener!"; + error( RtAudioError::WARNING ); + } + +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + if ( stream_.state == STREAM_RUNNING ) + AudioDeviceStop( handle->id[1], handle->procId[1] ); + AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] ); +#else // deprecated behaviour + if ( stream_.state == STREAM_RUNNING ) + AudioDeviceStop( handle->id[1], callbackHandler ); + AudioDeviceRemoveIOProc( handle->id[1], callbackHandler ); +#endif + } + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + // Destroy pthread condition variable. + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiCore :: startStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiCore::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + +#if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); +#endif + + OSStatus result = noErr; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + result = AudioDeviceStart( handle->id[0], handle->procId[0] ); +#else // deprecated behaviour + result = AudioDeviceStart( handle->id[0], callbackHandler ); +#endif + if ( result != noErr ) { + errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( stream_.mode == INPUT || + ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + result = AudioDeviceStart( handle->id[1], handle->procId[1] ); +#else // deprecated behaviour + result = AudioDeviceStart( handle->id[1], callbackHandler ); +#endif + if ( result != noErr ) { + errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + + unlock: + if ( result == noErr ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiCore :: stopStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiCore::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + OSStatus result = noErr; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled + } + +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + result = AudioDeviceStop( handle->id[0], handle->procId[0] ); +#else // deprecated behaviour + result = AudioDeviceStop( handle->id[0], callbackHandler ); +#endif + if ( result != noErr ) { + errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + result = AudioDeviceStop( handle->id[1], handle->procId[1] ); +#else // deprecated behaviour + result = AudioDeviceStop( handle->id[1], callbackHandler ); +#endif + if ( result != noErr ) { + errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + stream_.state = STREAM_STOPPED; + + unlock: + if ( result == noErr ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiCore :: abortStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiCore::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + stopStream(); +} + +// This function will be called by a spawned thread when the user +// callback function signals that the stream should be stopped or +// aborted. It is better to handle it this way because the +// callbackEvent() function probably should return before the AudioDeviceStop() +// function is called. +static void *coreStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiCore *object = (RtApiCore *) info->object; + + object->stopStream(); + pthread_exit( NULL ); +} + +bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, + const AudioBufferList *inBufferList, + const AudioBufferList *outBufferList ) +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + ThreadHandle threadId; + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == true ) + pthread_create( &threadId, NULL, coreStopStream, info ); + else // external call to stopStream() + pthread_cond_signal( &handle->condition ); + return SUCCESS; + } + + AudioDeviceID outputDevice = handle->id[0]; + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream or duplex mode AND the input/output devices are + // different AND this function is called for the input device. + if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + abortStream(); + return SUCCESS; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) { + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + if ( handle->nStreams[0] == 1 ) { + memset( outBufferList->mBuffers[handle->iStream[0]].mData, + 0, + outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); + } + else { // fill multiple streams with zeros + for ( unsigned int i=0; inStreams[0]; i++ ) { + memset( outBufferList->mBuffers[handle->iStream[0]+i].mData, + 0, + outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize ); + } + } + } + else if ( handle->nStreams[0] == 1 ) { + if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer + convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData, + stream_.userBuffer[0], stream_.convertInfo[0] ); + } + else { // copy from user buffer + memcpy( outBufferList->mBuffers[handle->iStream[0]].mData, + stream_.userBuffer[0], + outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); + } + } + else { // fill multiple streams + Float32 *inBuffer = (Float32 *) stream_.userBuffer[0]; + if ( stream_.doConvertBuffer[0] ) { + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + inBuffer = (Float32 *) stream_.deviceBuffer; + } + + if ( stream_.deviceInterleaved[0] == false ) { // mono mode + UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; + for ( unsigned int i=0; imBuffers[handle->iStream[0]+i].mData, + (void *)&inBuffer[i*stream_.bufferSize], bufferBytes ); + } + } + else { // fill multiple multi-channel streams with interleaved data + UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset; + Float32 *out, *in; + + bool inInterleaved = ( stream_.userInterleaved ) ? true : false; + UInt32 inChannels = stream_.nUserChannels[0]; + if ( stream_.doConvertBuffer[0] ) { + inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode + inChannels = stream_.nDeviceChannels[0]; + } + + if ( inInterleaved ) inOffset = 1; + else inOffset = stream_.bufferSize; + + channelsLeft = inChannels; + for ( unsigned int i=0; inStreams[0]; i++ ) { + in = inBuffer; + out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData; + streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels; + + outJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[0] > 0 ) { + streamChannels -= stream_.channelOffset[0]; + outJump = stream_.channelOffset[0]; + out += outJump; + } + + // Account for possible unfilled channels at end of the last stream + if ( streamChannels > channelsLeft ) { + outJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine input buffer offsets and skips + if ( inInterleaved ) { + inJump = inChannels; + in += inChannels - channelsLeft; + } + else { + inJump = 1; + in += (inChannels - channelsLeft) * inOffset; + } + + for ( unsigned int i=0; idrainCounter ) { + handle->drainCounter++; + goto unlock; + } + + AudioDeviceID inputDevice; + inputDevice = handle->id[1]; + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) { + + if ( handle->nStreams[1] == 1 ) { + if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer + convertBuffer( stream_.userBuffer[1], + (char *) inBufferList->mBuffers[handle->iStream[1]].mData, + stream_.convertInfo[1] ); + } + else { // copy to user buffer + memcpy( stream_.userBuffer[1], + inBufferList->mBuffers[handle->iStream[1]].mData, + inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); + } + } + else { // read from multiple streams + Float32 *outBuffer = (Float32 *) stream_.userBuffer[1]; + if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer; + + if ( stream_.deviceInterleaved[1] == false ) { // mono mode + UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize; + for ( unsigned int i=0; imBuffers[handle->iStream[1]+i].mData, bufferBytes ); + } + } + else { // read from multiple multi-channel streams + UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset; + Float32 *out, *in; + + bool outInterleaved = ( stream_.userInterleaved ) ? true : false; + UInt32 outChannels = stream_.nUserChannels[1]; + if ( stream_.doConvertBuffer[1] ) { + outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode + outChannels = stream_.nDeviceChannels[1]; + } + + if ( outInterleaved ) outOffset = 1; + else outOffset = stream_.bufferSize; + + channelsLeft = outChannels; + for ( unsigned int i=0; inStreams[1]; i++ ) { + out = outBuffer; + in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData; + streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels; + + inJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[1] > 0 ) { + streamChannels -= stream_.channelOffset[1]; + inJump = stream_.channelOffset[1]; + in += inJump; + } + + // Account for possible unread channels at end of the last stream + if ( streamChannels > channelsLeft ) { + inJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine output buffer offsets and skips + if ( outInterleaved ) { + outJump = outChannels; + out += outChannels - channelsLeft; + } + else { + outJump = 1; + out += (outChannels - channelsLeft) * outOffset; + } + + for ( unsigned int i=0; iid[0] != handle->id[1] && deviceId == handle->id[0] ) ) + RtApi::tickStreamTime(); + + return SUCCESS; +} + +const char* RtApiCore :: getErrorCode( OSStatus code ) +{ + switch( code ) { + + case kAudioHardwareNotRunningError: + return "kAudioHardwareNotRunningError"; + + case kAudioHardwareUnspecifiedError: + return "kAudioHardwareUnspecifiedError"; + + case kAudioHardwareUnknownPropertyError: + return "kAudioHardwareUnknownPropertyError"; + + case kAudioHardwareBadPropertySizeError: + return "kAudioHardwareBadPropertySizeError"; + + case kAudioHardwareIllegalOperationError: + return "kAudioHardwareIllegalOperationError"; + + case kAudioHardwareBadObjectError: + return "kAudioHardwareBadObjectError"; + + case kAudioHardwareBadDeviceError: + return "kAudioHardwareBadDeviceError"; + + case kAudioHardwareBadStreamError: + return "kAudioHardwareBadStreamError"; + + case kAudioHardwareUnsupportedOperationError: + return "kAudioHardwareUnsupportedOperationError"; + + case kAudioDeviceUnsupportedFormatError: + return "kAudioDeviceUnsupportedFormatError"; + + case kAudioDevicePermissionsError: + return "kAudioDevicePermissionsError"; + + default: + return "CoreAudio unknown error"; + } +} + + //******************** End of __MACOSX_CORE__ *********************// +#endif + +#if defined(__UNIX_JACK__) + +// JACK is a low-latency audio server, originally written for the +// GNU/Linux operating system and now also ported to OS-X. It can +// connect a number of different applications to an audio device, as +// well as allowing them to share audio between themselves. +// +// When using JACK with RtAudio, "devices" refer to JACK clients that +// have ports connected to the server. The JACK server is typically +// started in a terminal as follows: +// +// .jackd -d alsa -d hw:0 +// +// or through an interface program such as qjackctl. Many of the +// parameters normally set for a stream are fixed by the JACK server +// and can be specified when the JACK server is started. In +// particular, +// +// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4 +// +// specifies a sample rate of 44100 Hz, a buffer size of 512 sample +// frames, and number of buffers = 4. Once the server is running, it +// is not possible to override these values. If the values are not +// specified in the command-line, the JACK server uses default values. +// +// The JACK server does not have to be running when an instance of +// RtApiJack is created, though the function getDeviceCount() will +// report 0 devices found until JACK has been started. When no +// devices are available (i.e., the JACK server is not running), a +// stream cannot be opened. + +#include +#include +#include + +// A structure to hold various information related to the Jack API +// implementation. +struct JackHandle { + jack_client_t *client; + jack_port_t **ports[2]; + std::string deviceName[2]; + bool xrun[2]; + pthread_cond_t condition; + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + + JackHandle() + :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +#if !defined(__RTAUDIO_DEBUG__) +static void jackSilentError( const char * ) {}; +#endif + +RtApiJack :: RtApiJack() + :shouldAutoconnect_(true) { + // Nothing to do here. +#if !defined(__RTAUDIO_DEBUG__) + // Turn off Jack's internal error reporting. + jack_set_error_function( &jackSilentError ); +#endif +} + +RtApiJack :: ~RtApiJack() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiJack :: getDeviceCount( void ) +{ + // See if we can become a jack client. + jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption; + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); + if ( client == 0 ) return 0; + + const char **ports; + std::string port, previousPort; + unsigned int nChannels = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nChannels ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon + 1 ); + if ( port != previousPort ) { + nDevices++; + previousPort = port; + } + } + } while ( ports[++nChannels] ); + free( ports ); + } + + jack_client_close( client ); + return nDevices; +} + +RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); + if ( client == 0 ) { + errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; + error( RtAudioError::WARNING ); + return info; + } + + const char **ports; + std::string port, previousPort; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) info.name = port; + nDevices++; + previousPort = port; + } + } + } while ( ports[++nPorts] ); + free( ports ); + } + + if ( device >= nDevices ) { + jack_client_close( client ); + errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + // Get the current jack server sample rate. + info.sampleRates.clear(); + + info.preferredSampleRate = jack_get_sample_rate( client ); + info.sampleRates.push_back( info.preferredSampleRate ); + + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.outputChannels = nChannels; + } + + // Jack "output ports" equal RtAudio input channels. + nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.inputChannels = nChannels; + } + + if ( info.outputChannels == 0 && info.inputChannels == 0 ) { + jack_client_close(client); + errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; + error( RtAudioError::WARNING ); + return info; + } + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Jack always uses 32-bit floats. + info.nativeFormats = RTAUDIO_FLOAT32; + + // Jack doesn't provide default devices so we'll use the first available one. + if ( device == 0 && info.outputChannels > 0 ) + info.isDefaultOutput = true; + if ( device == 0 && info.inputChannels > 0 ) + info.isDefaultInput = true; + + jack_client_close(client); + info.probed = true; + return info; +} + +static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + + RtApiJack *object = (RtApiJack *) info->object; + if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; + + return 0; +} + +// This function will be called by a spawned thread when the Jack +// server signals that it is shutting down. It is necessary to handle +// it this way because the jackShutdown() function must return before +// the jack_deactivate() function (in closeStream()) will return. +static void *jackCloseStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiJack *object = (RtApiJack *) info->object; + + object->closeStream(); + + pthread_exit( NULL ); +} +static void jackShutdown( void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + RtApiJack *object = (RtApiJack *) info->object; + + // Check current stream state. If stopped, then we'll assume this + // was called as a result of a call to RtApiJack::stopStream (the + // deactivation of a client handle causes this function to be called). + // If not, we'll assume the Jack server is shutting down or some + // other problem occurred and we should close the stream. + if ( object->isStreamRunning() == false ) return; + + ThreadHandle threadId; + pthread_create( &threadId, NULL, jackCloseStream, info ); + std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; +} + +static int jackXrun( void *infoPointer ) +{ + JackHandle *handle = *((JackHandle **) infoPointer); + + if ( handle->ports[0] ) handle->xrun[0] = true; + if ( handle->ports[1] ) handle->xrun[1] = true; + + return 0; +} + +bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + JackHandle *handle = (JackHandle *) stream_.apiHandle; + + // Look for jack server and try to become a client (only do once per stream). + jack_client_t *client = 0; + if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { + jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption; + jack_status_t *status = NULL; + if ( options && !options->streamName.empty() ) + client = jack_client_open( options->streamName.c_str(), jackoptions, status ); + else + client = jack_client_open( "RtApiJack", jackoptions, status ); + if ( client == 0 ) { + errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + } + else { + // The handle must have been created on an earlier pass. + client = handle->client; + } + + const char **ports; + std::string port, previousPort, deviceName; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) deviceName = port; + nDevices++; + previousPort = port; + } + } + } while ( ports[++nPorts] ); + free( ports ); + } + + if ( device >= nDevices ) { + errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + unsigned long flag = JackPortIsInput; + if ( mode == INPUT ) flag = JackPortIsOutput; + + if ( ! (options && (options->flags & RTAUDIO_JACK_DONT_CONNECT)) ) { + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + } + // Compare the jack ports for specified client to the requested number of channels. + if ( nChannels < (channels + firstChannel) ) { + errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Check the jack server sample rate. + unsigned int jackRate = jack_get_sample_rate( client ); + if ( sampleRate != jackRate ) { + jack_client_close( client ); + errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = jackRate; + + // Get the latency of the JACK port. + ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag ); + if ( ports[ firstChannel ] ) { + // Added by Ge Wang + jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency); + // the range (usually the min and max are equal) + jack_latency_range_t latrange; latrange.min = latrange.max = 0; + // get the latency range + jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange ); + // be optimistic, use the min! + stream_.latency[mode] = latrange.min; + //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); + } + free( ports ); + + // The jack server always uses 32-bit floating-point data. + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + stream_.userFormat = format; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // Jack always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; + + // Jack always provides host byte-ordered data. + stream_.doByteSwap[mode] = false; + + // Get the buffer size. The buffer size and number of buffers + // (periods) is set when the jack server is started. + stream_.bufferSize = (int) jack_get_buffer_size( client ); + *bufferSize = stream_.bufferSize; + + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate our JackHandle structure for the stream. + if ( handle == 0 ) { + try { + handle = new JackHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory."; + goto error; + } + + if ( pthread_cond_init(&handle->condition, NULL) ) { + errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; + handle->client = client; + } + handle->deviceName[mode] = deviceName; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + if ( mode == OUTPUT ) + bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + else { // mode == INPUT + bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] ); + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]); + if ( bufferBytes < bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + // Allocate memory for the Jack ports (channels) identifiers. + handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels ); + if ( handle->ports[mode] == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory."; + goto error; + } + + stream_.device[mode] = device; + stream_.channelOffset[mode] = firstChannel; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; + + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up the stream for output. + stream_.mode = DUPLEX; + else { + stream_.mode = mode; + jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); + jack_set_xrun_callback( handle->client, jackXrun, (void *) &stream_.apiHandle ); + jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); + } + + // Register our ports. + char label[64]; + if ( mode == OUTPUT ) { + for ( unsigned int i=0; iports[0][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 ); + } + } + else { + for ( unsigned int i=0; iports[1][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 ); + } + } + + // Setup the buffer conversion information structure. We don't use + // buffers to do channel offsets, so we override that parameter + // here. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); + + if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false; + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &handle->condition ); + jack_client_close( handle->client ); + + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); + + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + return FAILURE; +} + +void RtApiJack :: closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiJack::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( handle ) { + + if ( stream_.state == STREAM_RUNNING ) + jack_deactivate( handle->client ); + + jack_client_close( handle->client ); + } + + if ( handle ) { + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiJack :: startStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiJack::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + int result = jack_activate( handle->client ); + if ( result ) { + errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; + goto unlock; + } + + const char **ports; + + // Get the list of available ports. + if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) { + result = 1; + ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; + goto unlock; + } + + // Now make the port connections. Since RtAudio wasn't designed to + // allow the user to select particular channels of a device, we'll + // just open the first "nChannels" ports with offset. + for ( unsigned int i=0; iclient, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting output ports!"; + goto unlock; + } + } + free(ports); + } + + if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) { + result = 1; + ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput ); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; + goto unlock; + } + + // Now make the port connections. See note above. + for ( unsigned int i=0; iclient, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting input ports!"; + goto unlock; + } + } + free(ports); + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + + unlock: + if ( result == 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiJack :: stopStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled + } + } + + jack_deactivate( handle->client ); + stream_.state = STREAM_STOPPED; +} + +void RtApiJack :: abortStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + stopStream(); +} + +// This function will be called by a spawned thread when the user +// callback function signals that the stream should be stopped or +// aborted. It is necessary to handle it this way because the +// callbackEvent() function must return before the jack_deactivate() +// function will return. +static void *jackStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiJack *object = (RtApiJack *) info->object; + + object->stopStream(); + pthread_exit( NULL ); +} + +bool RtApiJack :: callbackEvent( unsigned long nframes ) +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + if ( stream_.bufferSize != nframes ) { + errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + JackHandle *handle = (JackHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + ThreadHandle threadId; + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == true ) + pthread_create( &threadId, NULL, jackStopStream, info ); + else + pthread_cond_signal( &handle->condition ); + return SUCCESS; + } + + // Invoke user callback first, to get fresh output data. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + ThreadHandle id; + pthread_create( &id, NULL, jackStopStream, info ); + return SUCCESS; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + jack_default_audio_sample_t *jackbuffer; + unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memset( jackbuffer, 0, bufferBytes ); + } + + } + else if ( stream_.doConvertBuffer[0] ) { + + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); + } + } + else { // no buffer conversion + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); + } + } + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + if ( stream_.doConvertBuffer[1] ) { + for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes ); + } + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } + else { // no buffer conversion + for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes ); + } + } + } + + unlock: + RtApi::tickStreamTime(); + return SUCCESS; +} + //******************** End of __UNIX_JACK__ *********************// +#endif + +#if defined(__WINDOWS_ASIO__) // ASIO API on Windows + +// The ASIO API is designed around a callback scheme, so this +// implementation is similar to that used for OS-X CoreAudio and Linux +// Jack. The primary constraint with ASIO is that it only allows +// access to a single driver at a time. Thus, it is not possible to +// have more than one simultaneous RtAudio stream. +// +// This implementation also requires a number of external ASIO files +// and a few global variables. The ASIO callback scheme does not +// allow for the passing of user data, so we must create a global +// pointer to our callbackInfo structure. +// +// On unix systems, we make use of a pthread condition variable. +// Since there is no equivalent in Windows, I hacked something based +// on information found in +// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html. + +#include "asiosys.h" +#include "asio.h" +#include "iasiothiscallresolver.h" +#include "asiodrivers.h" +#include + +static AsioDrivers drivers; +static ASIOCallbacks asioCallbacks; +static ASIODriverInfo driverInfo; +static CallbackInfo *asioCallbackInfo; +static bool asioXRun; + +struct AsioHandle { + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + ASIOBufferInfo *bufferInfos; + HANDLE condition; + + AsioHandle() + :drainCounter(0), internalDrain(false), bufferInfos(0) {} +}; + +// Function declarations (definitions at end of section) +static const char* getAsioErrorString( ASIOError result ); +static void sampleRateChanged( ASIOSampleRate sRate ); +static long asioMessages( long selector, long value, void* message, double* opt ); + +RtApiAsio :: RtApiAsio() +{ + // ASIO cannot run on a multi-threaded appartment. You can call + // CoInitialize beforehand, but it must be for appartment threading + // (in which case, CoInitilialize will return S_FALSE here). + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( FAILED(hr) ) { + errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; + error( RtAudioError::WARNING ); + } + coInitialized_ = true; + + drivers.removeCurrentDriver(); + driverInfo.asioVersion = 2; + + // See note in DirectSound implementation about GetDesktopWindow(). + driverInfo.sysRef = GetForegroundWindow(); +} + +RtApiAsio :: ~RtApiAsio() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); + if ( coInitialized_ ) CoUninitialize(); +} + +unsigned int RtApiAsio :: getDeviceCount( void ) +{ + return (unsigned int) drivers.asioGetNumDev(); +} + +RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + // If a stream is already open, we cannot probe other devices. Thus, use the saved results. + if ( stream_.state != STREAM_CLOSED ) { + if ( device >= devices_.size() ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened."; + error( RtAudioError::WARNING ); + return info; + } + return devices_[ device ]; + } + + char driverName[32]; + ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + info.name = driverName; + + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Determine the device channel information. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + info.outputChannels = outputChannels; + info.inputChannels = inputChannels; + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Determine the supported sample rates. + info.sampleRates.clear(); + for ( unsigned int i=0; i info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[i]; + } + } + + // Determine supported data types ... just check first channel and assume rest are the same. + ASIOChannelInfo channelInfo; + channelInfo.channel = 0; + channelInfo.isInput = true; + if ( info.inputChannels <= 0 ) channelInfo.isInput = false; + result = ASIOGetChannelInfo( &channelInfo ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + info.nativeFormats = 0; + if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) + info.nativeFormats |= RTAUDIO_SINT16; + else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) + info.nativeFormats |= RTAUDIO_SINT32; + else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) + info.nativeFormats |= RTAUDIO_FLOAT32; + else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) + info.nativeFormats |= RTAUDIO_FLOAT64; + else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) + info.nativeFormats |= RTAUDIO_SINT24; + + if ( info.outputChannels > 0 ) + if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; + if ( info.inputChannels > 0 ) + if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; + + info.probed = true; + drivers.removeCurrentDriver(); + return info; +} + +static void bufferSwitch( long index, ASIOBool /*processNow*/ ) +{ + RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object; + object->callbackEvent( index ); +} + +void RtApiAsio :: saveDeviceInfo( void ) +{ + devices_.clear(); + + unsigned int nDevices = getDeviceCount(); + devices_.resize( nDevices ); + for ( unsigned int i=0; isaveDeviceInfo(); + + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // keep them before any "goto error", they are used for error cleanup + goto device boundary checks + bool buffersAllocated = false; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + unsigned int nChannels; + + + // Check the device channel count. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; + errorText_ = errorStream_.str(); + goto error; + } + + if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || + ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; + errorText_ = errorStream_.str(); + goto error; + } + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = firstChannel; + + // Verify the sample rate is supported. + result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + goto error; + } + + // Get the current sample rate + ASIOSampleRate currentRate; + result = ASIOGetSampleRate( ¤tRate ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; + errorText_ = errorStream_.str(); + goto error; + } + + // Set the sample rate only if necessary + if ( currentRate != sampleRate ) { + result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + goto error; + } + } + + // Determine the driver data type. + ASIOChannelInfo channelInfo; + channelInfo.channel = 0; + if ( mode == OUTPUT ) channelInfo.isInput = false; + else channelInfo.isInput = true; + result = ASIOGetChannelInfo( &channelInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; + errorText_ = errorStream_.str(); + goto error; + } + + // Assuming WINDOWS host is always little-endian. + stream_.doByteSwap[mode] = false; + stream_.userFormat = format; + stream_.deviceFormat[mode] = 0; + if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; + if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true; + } + + if ( stream_.deviceFormat[mode] == 0 ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + goto error; + } + + // Set the buffer size. For a duplex stream, this will end up + // setting the buffer size based on the input constraints, which + // should be ok. + long minSize, maxSize, preferSize, granularity; + result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; + errorText_ = errorStream_.str(); + goto error; + } + + if ( isDuplexInput ) { + // When this is the duplex input (output was opened before), then we have to use the same + // buffersize as the output, because it might use the preferred buffer size, which most + // likely wasn't passed as input to this. The buffer sizes have to be identically anyway, + // So instead of throwing an error, make them equal. The caller uses the reference + // to the "bufferSize" param as usual to set up processing buffers. + + *bufferSize = stream_.bufferSize; + + } else { + if ( *bufferSize == 0 ) *bufferSize = preferSize; + else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + else if ( granularity == -1 ) { + // Make sure bufferSize is a power of two. + int log2_of_min_size = 0; + int log2_of_max_size = 0; + + for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { + if ( minSize & ((long)1 << i) ) log2_of_min_size = i; + if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; + } + + long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); + int min_delta_num = log2_of_min_size; + + for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { + long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); + if (current_delta < min_delta) { + min_delta = current_delta; + min_delta_num = i; + } + } + + *bufferSize = ( (unsigned int)1 << min_delta_num ); + if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + } + else if ( granularity != 0 ) { + // Set to an even multiple of granularity, rounding up. + *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; + } + } + + /* + // we don't use it anymore, see above! + // Just left it here for the case... + if ( isDuplexInput && stream_.bufferSize != *bufferSize ) { + errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; + goto error; + } + */ + + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 2; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // ASIO always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; + + // Allocate, if necessary, our AsioHandle structure for the stream. + if ( handle == 0 ) { + try { + handle = new AsioHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; + goto error; + } + handle->bufferInfos = 0; + + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; + } + + // Create the ASIO internal buffers. Since RtAudio sets up input + // and output separately, we'll have to dispose of previously + // created output buffers for a duplex stream. + if ( mode == INPUT && stream_.mode == OUTPUT ) { + ASIODisposeBuffers(); + if ( handle->bufferInfos ) free( handle->bufferInfos ); + } + + // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. + unsigned int i; + nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); + if ( handle->bufferInfos == NULL ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + goto error; + } + + ASIOBufferInfo *infos; + infos = handle->bufferInfos; + for ( i=0; iisInput = ASIOFalse; + infos->channelNum = i + stream_.channelOffset[0]; + infos->buffers[0] = infos->buffers[1] = 0; + } + for ( i=0; iisInput = ASIOTrue; + infos->channelNum = i + stream_.channelOffset[1]; + infos->buffers[0] = infos->buffers[1] = 0; + } + + // prepare for callbacks + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.mode = isDuplexInput ? DUPLEX : mode; + + // store this class instance before registering callbacks, that are going to use it + asioCallbackInfo = &stream_.callbackInfo; + stream_.callbackInfo.object = (void *) this; + + // Set up the ASIO callback structure and create the ASIO data buffers. + asioCallbacks.bufferSwitch = &bufferSwitch; + asioCallbacks.sampleRateDidChange = &sampleRateChanged; + asioCallbacks.asioMessage = &asioMessages; + asioCallbacks.bufferSwitchTimeInfo = NULL; + result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); + if ( result != ASE_OK ) { + // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges + // but only accept the preferred buffer size as parameter for ASIOCreateBuffers (e.g. Creative's ASIO driver). + // In that case, let's be naïve and try that instead. + *bufferSize = preferSize; + stream_.bufferSize = *bufferSize; + result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); + } + + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; + errorText_ = errorStream_.str(); + goto error; + } + buffersAllocated = true; + stream_.state = STREAM_STOPPED; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( isDuplexInput && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + // Determine device latencies + long inputLatency, outputLatency; + result = ASIOGetLatencies( &inputLatency, &outputLatency ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING); // warn but don't fail + } + else { + stream_.latency[0] = outputLatency; + stream_.latency[1] = inputLatency; + } + + // Setup the buffer conversion information structure. We don't use + // buffers to do channel offsets, so we override that parameter + // here. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); + + return SUCCESS; + + error: + if ( !isDuplexInput ) { + // the cleanup for error in the duplex input, is done by RtApi::openStream + // So we clean up for single channel only + + if ( buffersAllocated ) + ASIODisposeBuffers(); + + drivers.removeCurrentDriver(); + + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + + delete handle; + stream_.apiHandle = 0; + } + + + if ( stream_.userBuffer[mode] ) { + free( stream_.userBuffer[mode] ); + stream_.userBuffer[mode] = 0; + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + } + + return FAILURE; +}//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +void RtApiAsio :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + ASIOStop(); + } + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +bool stopThreadCalled = false; + +void RtApiAsio :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAsio::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + ASIOError result = ASIOStart(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device."; + errorText_ = errorStream_.str(); + goto unlock; + } + + handle->drainCounter = 0; + handle->internalDrain = false; + ResetEvent( handle->condition ); + stream_.state = STREAM_RUNNING; + asioXRun = false; + + unlock: + stopThreadCalled = false; + + if ( result == ASE_OK ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAsio :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + WaitForSingleObject( handle->condition, INFINITE ); // block until signaled + } + } + + stream_.state = STREAM_STOPPED; + + ASIOError result = ASIOStop(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; + errorText_ = errorStream_.str(); + } + + if ( result == ASE_OK ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAsio :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + // The following lines were commented-out because some behavior was + // noted where the device buffers need to be zeroed to avoid + // continuing sound, even when the device buffers are completely + // disposed. So now, calling abort is the same as calling stop. + // AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + // handle->drainCounter = 2; + stopStream(); +} + +// This function will be called by a spawned thread when the user +// callback function signals that the stream should be stopped or +// aborted. It is necessary to handle it this way because the +// callbackEvent() function must return before the ASIOStop() +// function will return. +static unsigned __stdcall asioStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAsio *object = (RtApiAsio *) info->object; + + object->stopStream(); + _endthreadex( 0 ); + return 0; +} + +bool RtApiAsio :: callbackEvent( long bufferIndex ) +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal if finished. + if ( handle->drainCounter > 3 ) { + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else { // spawn a thread to stop the stream + unsigned threadId; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, + &stream_.callbackInfo, 0, &threadId ); + } + return SUCCESS; + } + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && asioXRun == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + asioXRun = false; + } + if ( stream_.mode != OUTPUT && asioXRun == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + asioXRun = false; + } + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + unsigned threadId; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, + &stream_.callbackInfo, 0, &threadId ); + return SUCCESS; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + unsigned int nChannels, bufferBytes, i, j; + nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); + } + + } + else if ( stream_.doConvertBuffer[0] ) { + + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[0], + stream_.deviceFormat[0] ); + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); + } + + } + else { + + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.userBuffer[0], + stream_.bufferSize * stream_.nUserChannels[0], + stream_.userFormat ); + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); + } + + } + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); + + if (stream_.doConvertBuffer[1]) { + + // Always interleave ASIO input data. + for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) + memcpy( &stream_.deviceBuffer[j++*bufferBytes], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } + + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[1], + stream_.deviceFormat[1] ); + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + + } + else { + for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) { + memcpy( &stream_.userBuffer[1][bufferBytes*j++], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } + } + + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.userBuffer[1], + stream_.bufferSize * stream_.nUserChannels[1], + stream_.userFormat ); + } + } + + unlock: + // The following call was suggested by Malte Clasen. While the API + // documentation indicates it should not be required, some device + // drivers apparently do not function correctly without it. + ASIOOutputReady(); + + RtApi::tickStreamTime(); + return SUCCESS; +} + +static void sampleRateChanged( ASIOSampleRate sRate ) +{ + // The ASIO documentation says that this usually only happens during + // external sync. Audio processing is not stopped by the driver, + // actual sample rate might not have even changed, maybe only the + // sample rate status of an AES/EBU or S/PDIF digital input at the + // audio device. + + RtApi *object = (RtApi *) asioCallbackInfo->object; + try { + object->stopStream(); + } + catch ( RtAudioError &exception ) { + std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; + return; + } + + std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; +} + +static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ ) +{ + long ret = 0; + + switch( selector ) { + case kAsioSelectorSupported: + if ( value == kAsioResetRequest + || value == kAsioEngineVersion + || value == kAsioResyncRequest + || value == kAsioLatenciesChanged + // The following three were added for ASIO 2.0, you don't + // necessarily have to support them. + || value == kAsioSupportsTimeInfo + || value == kAsioSupportsTimeCode + || value == kAsioSupportsInputMonitor) + ret = 1L; + break; + case kAsioResetRequest: + // Defer the task and perform the reset of the driver during the + // next "safe" situation. You cannot reset the driver right now, + // as this code is called from the driver. Reset the driver is + // done by completely destruct is. I.e. ASIOStop(), + // ASIODisposeBuffers(), Destruction Afterwards you initialize the + // driver again. + std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; + ret = 1L; + break; + case kAsioResyncRequest: + // This informs the application that the driver encountered some + // non-fatal data loss. It is used for synchronization purposes + // of different media. Added mainly to work around the Win16Mutex + // problems in Windows 95/98 with the Windows Multimedia system, + // which could lose data because the Mutex was held too long by + // another thread. However a driver can issue it in other + // situations, too. + // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; + asioXRun = true; + ret = 1L; + break; + case kAsioLatenciesChanged: + // This will inform the host application that the drivers were + // latencies changed. Beware, it this does not mean that the + // buffer sizes have changed! You might need to update internal + // delay data. + std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; + ret = 1L; + break; + case kAsioEngineVersion: + // Return the supported ASIO version of the host application. If + // a host application does not implement this selector, ASIO 1.0 + // is assumed by the driver. + ret = 2L; + break; + case kAsioSupportsTimeInfo: + // Informs the driver whether the + // asioCallbacks.bufferSwitchTimeInfo() callback is supported. + // For compatibility with ASIO 1.0 drivers the host application + // should always support the "old" bufferSwitch method, too. + ret = 0; + break; + case kAsioSupportsTimeCode: + // Informs the driver whether application is interested in time + // code info. If an application does not need to know about time + // code, the driver has less work to do. + ret = 0; + break; + } + return ret; +} + +static const char* getAsioErrorString( ASIOError result ) +{ + struct Messages + { + ASIOError value; + const char*message; + }; + + static const Messages m[] = + { + { ASE_NotPresent, "Hardware input or output is not present or available." }, + { ASE_HWMalfunction, "Hardware is malfunctioning." }, + { ASE_InvalidParameter, "Invalid input parameter." }, + { ASE_InvalidMode, "Invalid mode." }, + { ASE_SPNotAdvancing, "Sample position not advancing." }, + { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, + { ASE_NoMemory, "Not enough memory to complete the request." } + }; + + for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) + if ( m[i].value == result ) return m[i].message; + + return "Unknown error."; +} + +//******************** End of __WINDOWS_ASIO__ *********************// +#endif + + +#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API + +// Authored by Marcus Tomlinson , April 2014 +// - Introduces support for the Windows WASAPI API +// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required +// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface +// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user + +#ifndef INITGUID + #define INITGUID +#endif + +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#ifndef MF_E_TRANSFORM_NEED_MORE_INPUT + #define MF_E_TRANSFORM_NEED_MORE_INPUT _HRESULT_TYPEDEF_(0xc00d6d72) +#endif + +#ifndef MFSTARTUP_NOSOCKET + #define MFSTARTUP_NOSOCKET 0x1 +#endif + +#ifdef _MSC_VER + #pragma comment( lib, "ksuser" ) + #pragma comment( lib, "mfplat.lib" ) + #pragma comment( lib, "mfuuid.lib" ) + #pragma comment( lib, "wmcodecdspuuid" ) +#endif + +//============================================================================= + +#define SAFE_RELEASE( objectPtr )\ +if ( objectPtr )\ +{\ + objectPtr->Release();\ + objectPtr = NULL;\ +} + +typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex ); + +//----------------------------------------------------------------------------- + +// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size. +// Therefore we must perform all necessary conversions to user buffers in order to satisfy these +// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to +// provide intermediate storage for read / write synchronization. +class WasapiBuffer +{ +public: + WasapiBuffer() + : buffer_( NULL ), + bufferSize_( 0 ), + inIndex_( 0 ), + outIndex_( 0 ) {} + + ~WasapiBuffer() { + free( buffer_ ); + } + + // sets the length of the internal ring buffer + void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) { + free( buffer_ ); + + buffer_ = ( char* ) calloc( bufferSize, formatBytes ); + + bufferSize_ = bufferSize; + inIndex_ = 0; + outIndex_ = 0; + } + + // attempt to push a buffer into the ring buffer at the current "in" index + bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) + { + if ( !buffer || // incoming buffer is NULL + bufferSize == 0 || // incoming buffer has no data + bufferSize > bufferSize_ ) // incoming buffer too large + { + return false; + } + + unsigned int relOutIndex = outIndex_; + unsigned int inIndexEnd = inIndex_ + bufferSize; + if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) { + relOutIndex += bufferSize_; + } + + // the "IN" index CAN BEGIN at the "OUT" index + // the "IN" index CANNOT END at the "OUT" index + if ( inIndex_ < relOutIndex && inIndexEnd >= relOutIndex ) { + return false; // not enough space between "in" index and "out" index + } + + // copy buffer from external to internal + int fromZeroSize = inIndex_ + bufferSize - bufferSize_; + fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; + int fromInSize = bufferSize - fromZeroSize; + + switch( format ) + { + case RTAUDIO_SINT8: + memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) ); + memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) ); + break; + case RTAUDIO_SINT16: + memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) ); + memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) ); + break; + case RTAUDIO_SINT24: + memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) ); + memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) ); + break; + case RTAUDIO_SINT32: + memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) ); + memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) ); + break; + case RTAUDIO_FLOAT32: + memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) ); + memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) ); + break; + case RTAUDIO_FLOAT64: + memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) ); + memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) ); + break; + } + + // update "in" index + inIndex_ += bufferSize; + inIndex_ %= bufferSize_; + + return true; + } + + // attempt to pull a buffer from the ring buffer from the current "out" index + bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) + { + if ( !buffer || // incoming buffer is NULL + bufferSize == 0 || // incoming buffer has no data + bufferSize > bufferSize_ ) // incoming buffer too large + { + return false; + } + + unsigned int relInIndex = inIndex_; + unsigned int outIndexEnd = outIndex_ + bufferSize; + if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) { + relInIndex += bufferSize_; + } + + // the "OUT" index CANNOT BEGIN at the "IN" index + // the "OUT" index CAN END at the "IN" index + if ( outIndex_ <= relInIndex && outIndexEnd > relInIndex ) { + return false; // not enough space between "out" index and "in" index + } + + // copy buffer from internal to external + int fromZeroSize = outIndex_ + bufferSize - bufferSize_; + fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; + int fromOutSize = bufferSize - fromZeroSize; + + switch( format ) + { + case RTAUDIO_SINT8: + memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) ); + memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) ); + break; + case RTAUDIO_SINT16: + memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) ); + memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) ); + break; + case RTAUDIO_SINT24: + memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) ); + memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) ); + break; + case RTAUDIO_SINT32: + memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) ); + memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) ); + break; + case RTAUDIO_FLOAT32: + memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) ); + memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) ); + break; + case RTAUDIO_FLOAT64: + memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) ); + memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) ); + break; + } + + // update "out" index + outIndex_ += bufferSize; + outIndex_ %= bufferSize_; + + return true; + } + +private: + char* buffer_; + unsigned int bufferSize_; + unsigned int inIndex_; + unsigned int outIndex_; +}; + +//----------------------------------------------------------------------------- + +// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate +// between HW and the user. The WasapiResampler class is used to perform this conversion between +// HwIn->UserIn and UserOut->HwOut during the stream callback loop. +class WasapiResampler +{ +public: + WasapiResampler( bool isFloat, unsigned int bitsPerSample, unsigned int channelCount, + unsigned int inSampleRate, unsigned int outSampleRate ) + : _bytesPerSample( bitsPerSample / 8 ) + , _channelCount( channelCount ) + , _sampleRatio( ( float ) outSampleRate / inSampleRate ) + , _transformUnk( NULL ) + , _transform( NULL ) + , _mediaType( NULL ) + , _inputMediaType( NULL ) + , _outputMediaType( NULL ) + + #ifdef __IWMResamplerProps_FWD_DEFINED__ + , _resamplerProps( NULL ) + #endif + { + // 1. Initialization + + MFStartup( MF_VERSION, MFSTARTUP_NOSOCKET ); + + // 2. Create Resampler Transform Object + + CoCreateInstance( CLSID_CResamplerMediaObject, NULL, CLSCTX_INPROC_SERVER, + IID_IUnknown, ( void** ) &_transformUnk ); + + _transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) ); + + #ifdef __IWMResamplerProps_FWD_DEFINED__ + _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) ); + _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality + #endif + + // 3. Specify input / output format + + MFCreateMediaType( &_mediaType ); + _mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio ); + _mediaType->SetGUID( MF_MT_SUBTYPE, isFloat ? MFAudioFormat_Float : MFAudioFormat_PCM ); + _mediaType->SetUINT32( MF_MT_AUDIO_NUM_CHANNELS, channelCount ); + _mediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, inSampleRate ); + _mediaType->SetUINT32( MF_MT_AUDIO_BLOCK_ALIGNMENT, _bytesPerSample * channelCount ); + _mediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * inSampleRate ); + _mediaType->SetUINT32( MF_MT_AUDIO_BITS_PER_SAMPLE, bitsPerSample ); + _mediaType->SetUINT32( MF_MT_ALL_SAMPLES_INDEPENDENT, TRUE ); + + MFCreateMediaType( &_inputMediaType ); + _mediaType->CopyAllItems( _inputMediaType ); + + _transform->SetInputType( 0, _inputMediaType, 0 ); + + MFCreateMediaType( &_outputMediaType ); + _mediaType->CopyAllItems( _outputMediaType ); + + _outputMediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, outSampleRate ); + _outputMediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * outSampleRate ); + + _transform->SetOutputType( 0, _outputMediaType, 0 ); + + // 4. Send stream start messages to Resampler + + _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, 0 ); + _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, 0 ); + _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, 0 ); + } + + ~WasapiResampler() + { + // 8. Send stream stop messages to Resampler + + _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, 0 ); + _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, 0 ); + + // 9. Cleanup + + MFShutdown(); + + SAFE_RELEASE( _transformUnk ); + SAFE_RELEASE( _transform ); + SAFE_RELEASE( _mediaType ); + SAFE_RELEASE( _inputMediaType ); + SAFE_RELEASE( _outputMediaType ); + + #ifdef __IWMResamplerProps_FWD_DEFINED__ + SAFE_RELEASE( _resamplerProps ); + #endif + } + + void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount, int maxOutSampleCount = -1 ) + { + unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount; + if ( _sampleRatio == 1 ) + { + // no sample rate conversion required + memcpy( outBuffer, inBuffer, inputBufferSize ); + outSampleCount = inSampleCount; + return; + } + + unsigned int outputBufferSize = 0; + if ( maxOutSampleCount != -1 ) + { + outputBufferSize = _bytesPerSample * _channelCount * maxOutSampleCount; + } + else + { + outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount ); + } + + IMFMediaBuffer* rInBuffer; + IMFSample* rInSample; + BYTE* rInByteBuffer = NULL; + + // 5. Create Sample object from input data + + MFCreateMemoryBuffer( inputBufferSize, &rInBuffer ); + + rInBuffer->Lock( &rInByteBuffer, NULL, NULL ); + memcpy( rInByteBuffer, inBuffer, inputBufferSize ); + rInBuffer->Unlock(); + rInByteBuffer = NULL; + + rInBuffer->SetCurrentLength( inputBufferSize ); + + MFCreateSample( &rInSample ); + rInSample->AddBuffer( rInBuffer ); + + // 6. Pass input data to Resampler + + _transform->ProcessInput( 0, rInSample, 0 ); + + SAFE_RELEASE( rInBuffer ); + SAFE_RELEASE( rInSample ); + + // 7. Perform sample rate conversion + + IMFMediaBuffer* rOutBuffer = NULL; + BYTE* rOutByteBuffer = NULL; + + MFT_OUTPUT_DATA_BUFFER rOutDataBuffer; + DWORD rStatus; + DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput + + // 7.1 Create Sample object for output data + + memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer ); + MFCreateSample( &( rOutDataBuffer.pSample ) ); + MFCreateMemoryBuffer( rBytes, &rOutBuffer ); + rOutDataBuffer.pSample->AddBuffer( rOutBuffer ); + rOutDataBuffer.dwStreamID = 0; + rOutDataBuffer.dwStatus = 0; + rOutDataBuffer.pEvents = NULL; + + // 7.2 Get output data from Resampler + + if ( _transform->ProcessOutput( 0, 1, &rOutDataBuffer, &rStatus ) == MF_E_TRANSFORM_NEED_MORE_INPUT ) + { + outSampleCount = 0; + SAFE_RELEASE( rOutBuffer ); + SAFE_RELEASE( rOutDataBuffer.pSample ); + return; + } + + // 7.3 Write output data to outBuffer + + SAFE_RELEASE( rOutBuffer ); + rOutDataBuffer.pSample->ConvertToContiguousBuffer( &rOutBuffer ); + rOutBuffer->GetCurrentLength( &rBytes ); + + rOutBuffer->Lock( &rOutByteBuffer, NULL, NULL ); + memcpy( outBuffer, rOutByteBuffer, rBytes ); + rOutBuffer->Unlock(); + rOutByteBuffer = NULL; + + outSampleCount = rBytes / _bytesPerSample / _channelCount; + SAFE_RELEASE( rOutBuffer ); + SAFE_RELEASE( rOutDataBuffer.pSample ); + } + +private: + unsigned int _bytesPerSample; + unsigned int _channelCount; + float _sampleRatio; + + IUnknown* _transformUnk; + IMFTransform* _transform; + IMFMediaType* _mediaType; + IMFMediaType* _inputMediaType; + IMFMediaType* _outputMediaType; + + #ifdef __IWMResamplerProps_FWD_DEFINED__ + IWMResamplerProps* _resamplerProps; + #endif +}; + +//----------------------------------------------------------------------------- + +// A structure to hold various information related to the WASAPI implementation. +struct WasapiHandle +{ + IAudioClient* captureAudioClient; + IAudioClient* renderAudioClient; + IAudioCaptureClient* captureClient; + IAudioRenderClient* renderClient; + HANDLE captureEvent; + HANDLE renderEvent; + + WasapiHandle() + : captureAudioClient( NULL ), + renderAudioClient( NULL ), + captureClient( NULL ), + renderClient( NULL ), + captureEvent( NULL ), + renderEvent( NULL ) {} +}; + +//============================================================================= + +RtApiWasapi::RtApiWasapi() + : coInitialized_( false ), deviceEnumerator_( NULL ) +{ + // WASAPI can run either apartment or multi-threaded + HRESULT hr = CoInitialize( NULL ); + if ( !FAILED( hr ) ) + coInitialized_ = true; + + // Instantiate device enumerator + hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL, + CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ), + ( void** ) &deviceEnumerator_ ); + + // If this runs on an old Windows, it will fail. Ignore and proceed. + if ( FAILED( hr ) ) + deviceEnumerator_ = NULL; +} + +//----------------------------------------------------------------------------- + +RtApiWasapi::~RtApiWasapi() +{ + if ( stream_.state != STREAM_CLOSED ) + closeStream(); + + SAFE_RELEASE( deviceEnumerator_ ); + + // If this object previously called CoInitialize() + if ( coInitialized_ ) + CoUninitialize(); +} + +//============================================================================= + +unsigned int RtApiWasapi::getDeviceCount( void ) +{ + unsigned int captureDeviceCount = 0; + unsigned int renderDeviceCount = 0; + + IMMDeviceCollection* captureDevices = NULL; + IMMDeviceCollection* renderDevices = NULL; + + if ( !deviceEnumerator_ ) + return 0; + + // Count capture devices + errorText_.clear(); + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection."; + goto Exit; + } + + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count."; + goto Exit; + } + + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count."; + goto Exit; + } + +Exit: + // release all references + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); + + if ( errorText_.empty() ) + return captureDeviceCount + renderDeviceCount; + + error( RtAudioError::DRIVER_ERROR ); + return 0; +} + +//----------------------------------------------------------------------------- + +RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + unsigned int captureDeviceCount = 0; + unsigned int renderDeviceCount = 0; + std::string defaultDeviceName; + bool isCaptureDevice = false; + + PROPVARIANT deviceNameProp; + PROPVARIANT defaultDeviceNameProp; + + IMMDeviceCollection* captureDevices = NULL; + IMMDeviceCollection* renderDevices = NULL; + IMMDevice* devicePtr = NULL; + IMMDevice* defaultDevicePtr = NULL; + IAudioClient* audioClient = NULL; + IPropertyStore* devicePropStore = NULL; + IPropertyStore* defaultDevicePropStore = NULL; + + WAVEFORMATEX* deviceFormat = NULL; + WAVEFORMATEX* closestMatchFormat = NULL; + + // probed + info.probed = false; + + // Count capture devices + errorText_.clear(); + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection."; + goto Exit; + } + + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count."; + goto Exit; + } + + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count."; + goto Exit; + } + + // validate device index + if ( device >= captureDeviceCount + renderDeviceCount ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index."; + errorType = RtAudioError::INVALID_USE; + goto Exit; + } + + // determine whether index falls within capture or render devices + if ( device >= renderDeviceCount ) { + hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle."; + goto Exit; + } + isCaptureDevice = true; + } + else { + hr = renderDevices->Item( device, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle."; + goto Exit; + } + isCaptureDevice = false; + } + + // get default device name + if ( isCaptureDevice ) { + hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle."; + goto Exit; + } + } + else { + hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle."; + goto Exit; + } + } + + hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store."; + goto Exit; + } + PropVariantInit( &defaultDeviceNameProp ); + + hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName."; + goto Exit; + } + + defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal); + + // name + hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store."; + goto Exit; + } + + PropVariantInit( &deviceNameProp ); + + hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName."; + goto Exit; + } + + info.name =convertCharPointerToStdString(deviceNameProp.pwszVal); + + // is default + if ( isCaptureDevice ) { + info.isDefaultInput = info.name == defaultDeviceName; + info.isDefaultOutput = false; + } + else { + info.isDefaultInput = false; + info.isDefaultOutput = info.name == defaultDeviceName; + } + + // channel count + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client."; + goto Exit; + } + + hr = audioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format."; + goto Exit; + } + + if ( isCaptureDevice ) { + info.inputChannels = deviceFormat->nChannels; + info.outputChannels = 0; + info.duplexChannels = 0; + } + else { + info.inputChannels = 0; + info.outputChannels = deviceFormat->nChannels; + info.duplexChannels = 0; + } + + // sample rates + info.sampleRates.clear(); + + // allow support for all sample rates as we have a built-in sample rate converter + for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) { + info.sampleRates.push_back( SAMPLE_RATES[i] ); + } + info.preferredSampleRate = deviceFormat->nSamplesPerSec; + + // native format + info.nativeFormats = 0; + + if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT || + ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && + ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) ) + { + if ( deviceFormat->wBitsPerSample == 32 ) { + info.nativeFormats |= RTAUDIO_FLOAT32; + } + else if ( deviceFormat->wBitsPerSample == 64 ) { + info.nativeFormats |= RTAUDIO_FLOAT64; + } + } + else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM || + ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && + ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) ) + { + if ( deviceFormat->wBitsPerSample == 8 ) { + info.nativeFormats |= RTAUDIO_SINT8; + } + else if ( deviceFormat->wBitsPerSample == 16 ) { + info.nativeFormats |= RTAUDIO_SINT16; + } + else if ( deviceFormat->wBitsPerSample == 24 ) { + info.nativeFormats |= RTAUDIO_SINT24; + } + else if ( deviceFormat->wBitsPerSample == 32 ) { + info.nativeFormats |= RTAUDIO_SINT32; + } + } + + // probed + info.probed = true; + +Exit: + // release all references + PropVariantClear( &deviceNameProp ); + PropVariantClear( &defaultDeviceNameProp ); + + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); + SAFE_RELEASE( devicePtr ); + SAFE_RELEASE( defaultDevicePtr ); + SAFE_RELEASE( audioClient ); + SAFE_RELEASE( devicePropStore ); + SAFE_RELEASE( defaultDevicePropStore ); + + CoTaskMemFree( deviceFormat ); + CoTaskMemFree( closestMatchFormat ); + + if ( !errorText_.empty() ) + error( errorType ); + return info; +} + +//----------------------------------------------------------------------------- + +unsigned int RtApiWasapi::getDefaultOutputDevice( void ) +{ + for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { + if ( getDeviceInfo( i ).isDefaultOutput ) { + return i; + } + } + + return 0; +} + +//----------------------------------------------------------------------------- + +unsigned int RtApiWasapi::getDefaultInputDevice( void ) +{ + for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { + if ( getDeviceInfo( i ).isDefaultInput ) { + return i; + } + } + + return 0; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiWasapi::closeStream: No open stream to close."; + error( RtAudioError::WARNING ); + return; + } + + if ( stream_.state != STREAM_STOPPED ) + stopStream(); + + // clean up stream memory + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) + + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient ) + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient ) + + if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ) + CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ); + + if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ) + CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ); + + delete ( WasapiHandle* ) stream_.apiHandle; + stream_.apiHandle = NULL; + + for ( int i = 0; i < 2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + // update stream state + stream_.state = STREAM_CLOSED; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::startStream( void ) +{ + verifyStream(); + + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiWasapi::startStream: The stream is already running."; + error( RtAudioError::WARNING ); + return; + } + + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + + // update stream state + stream_.state = STREAM_RUNNING; + + // create WASAPI stream thread + stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL ); + + if ( !stream_.callbackInfo.thread ) { + errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread."; + error( RtAudioError::THREAD_ERROR ); + } + else { + SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority ); + ResumeThread( ( void* ) stream_.callbackInfo.thread ); + } +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::stopStream( void ) +{ + verifyStream(); + + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiWasapi::stopStream: The stream is already stopped."; + error( RtAudioError::WARNING ); + return; + } + + // inform stream thread by setting stream state to STREAM_STOPPING + stream_.state = STREAM_STOPPING; + + // wait until stream thread is stopped + while( stream_.state != STREAM_STOPPED ) { + Sleep( 1 ); + } + + // Wait for the last buffer to play before stopping. + Sleep( 1000 * stream_.bufferSize / stream_.sampleRate ); + + // close thread handle + if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { + errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread."; + error( RtAudioError::THREAD_ERROR ); + return; + } + + stream_.callbackInfo.thread = (ThreadHandle) NULL; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::abortStream( void ) +{ + verifyStream(); + + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiWasapi::abortStream: The stream is already stopped."; + error( RtAudioError::WARNING ); + return; + } + + // inform stream thread by setting stream state to STREAM_STOPPING + stream_.state = STREAM_STOPPING; + + // wait until stream thread is stopped + while ( stream_.state != STREAM_STOPPED ) { + Sleep( 1 ); + } + + // close thread handle + if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { + errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread."; + error( RtAudioError::THREAD_ERROR ); + return; + } + + stream_.callbackInfo.thread = (ThreadHandle) NULL; +} + +//----------------------------------------------------------------------------- + +bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int* bufferSize, + RtAudio::StreamOptions* options ) +{ + bool methodResult = FAILURE; + unsigned int captureDeviceCount = 0; + unsigned int renderDeviceCount = 0; + + IMMDeviceCollection* captureDevices = NULL; + IMMDeviceCollection* renderDevices = NULL; + IMMDevice* devicePtr = NULL; + WAVEFORMATEX* deviceFormat = NULL; + unsigned int bufferBytes; + stream_.state = STREAM_STOPPED; + + // create API Handle if not already created + if ( !stream_.apiHandle ) + stream_.apiHandle = ( void* ) new WasapiHandle(); + + // Count capture devices + errorText_.clear(); + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection."; + goto Exit; + } + + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count."; + goto Exit; + } + + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count."; + goto Exit; + } + + // validate device index + if ( device >= captureDeviceCount + renderDeviceCount ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index."; + goto Exit; + } + + // if device index falls within capture devices + if ( device >= renderDeviceCount ) { + if ( mode != INPUT ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device."; + goto Exit; + } + + // retrieve captureAudioClient from devicePtr + IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; + + hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle."; + goto Exit; + } + + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &captureAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device audio client."; + goto Exit; + } + + hr = captureAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device mix format."; + goto Exit; + } + + stream_.nDeviceChannels[mode] = deviceFormat->nChannels; + captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); + } + + // if device index falls within render devices and is configured for loopback + if ( device < renderDeviceCount && mode == INPUT ) + { + // if renderAudioClient is not initialised, initialise it now + IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; + if ( !renderAudioClient ) + { + probeDeviceOpen( device, OUTPUT, channels, firstChannel, sampleRate, format, bufferSize, options ); + } + + // retrieve captureAudioClient from devicePtr + IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; + + hr = renderDevices->Item( device, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle."; + goto Exit; + } + + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &captureAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client."; + goto Exit; + } + + hr = captureAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format."; + goto Exit; + } + + stream_.nDeviceChannels[mode] = deviceFormat->nChannels; + captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); + } + + // if device index falls within render devices and is configured for output + if ( device < renderDeviceCount && mode == OUTPUT ) + { + // if renderAudioClient is already initialised, don't initialise it again + IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; + if ( renderAudioClient ) + { + methodResult = SUCCESS; + goto Exit; + } + + hr = renderDevices->Item( device, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle."; + goto Exit; + } + + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &renderAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client."; + goto Exit; + } + + hr = renderAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format."; + goto Exit; + } + + stream_.nDeviceChannels[mode] = deviceFormat->nChannels; + renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); + } + + // fill stream data + if ( ( stream_.mode == OUTPUT && mode == INPUT ) || + ( stream_.mode == INPUT && mode == OUTPUT ) ) { + stream_.mode = DUPLEX; + } + else { + stream_.mode = mode; + } + + stream_.device[mode] = device; + stream_.doByteSwap[mode] = false; + stream_.sampleRate = sampleRate; + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 1; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = firstChannel; + stream_.userFormat = format; + stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) + stream_.userInterleaved = false; + else + stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] || + stream_.nUserChannels[0] != stream_.nDeviceChannels[0] || + stream_.nUserChannels[1] != stream_.nDeviceChannels[1] ) + stream_.doConvertBuffer[mode] = true; + else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + if ( stream_.doConvertBuffer[mode] ) + setConvertInfo( mode, firstChannel ); + + // Allocate necessary internal buffers + bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat ); + + stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 ); + if ( !stream_.userBuffer[mode] ) { + errorType = RtAudioError::MEMORY_ERROR; + errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory."; + goto Exit; + } + + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) + stream_.callbackInfo.priority = 15; + else + stream_.callbackInfo.priority = 0; + + ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback + ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode + + methodResult = SUCCESS; + +Exit: + //clean up + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); + SAFE_RELEASE( devicePtr ); + CoTaskMemFree( deviceFormat ); + + // if method failed, close the stream + if ( methodResult == FAILURE ) + closeStream(); + + if ( !errorText_.empty() ) + error( errorType ); + return methodResult; +} + +//============================================================================= + +DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread(); + + return 0; +} + +DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->stopStream(); + + return 0; +} + +DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->abortStream(); + + return 0; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::wasapiThread() +{ + // as this is a new thread, we must CoInitialize it + CoInitialize( NULL ); + + HRESULT hr; + + IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; + IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; + IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient; + IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient; + HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent; + HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent; + + WAVEFORMATEX* captureFormat = NULL; + WAVEFORMATEX* renderFormat = NULL; + float captureSrRatio = 0.0f; + float renderSrRatio = 0.0f; + WasapiBuffer captureBuffer; + WasapiBuffer renderBuffer; + WasapiResampler* captureResampler = NULL; + WasapiResampler* renderResampler = NULL; + + // declare local stream variables + RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback; + BYTE* streamBuffer = NULL; + DWORD captureFlags = 0; + unsigned int bufferFrameCount = 0; + unsigned int numFramesPadding = 0; + unsigned int convBufferSize = 0; + bool loopbackEnabled = stream_.device[INPUT] == stream_.device[OUTPUT]; + bool callbackPushed = true; + bool callbackPulled = false; + bool callbackStopped = false; + int callbackResult = 0; + + // convBuffer is used to store converted buffers between WASAPI and the user + char* convBuffer = NULL; + unsigned int convBuffSize = 0; + unsigned int deviceBuffSize = 0; + + std::string errorText; + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + + // Attempt to assign "Pro Audio" characteristic to thread + HMODULE AvrtDll = LoadLibraryW( L"AVRT.dll" ); + if ( AvrtDll ) { + DWORD taskIndex = 0; + TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = + ( TAvSetMmThreadCharacteristicsPtr ) (void(*)()) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" ); + AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex ); + FreeLibrary( AvrtDll ); + } + + // start capture stream if applicable + if ( captureAudioClient ) { + hr = captureAudioClient->GetMixFormat( &captureFormat ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; + goto Exit; + } + + // init captureResampler + captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64, + formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT], + captureFormat->nSamplesPerSec, stream_.sampleRate ); + + captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate ); + + if ( !captureClient ) { + hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, + loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK, + 0, + 0, + captureFormat, + NULL ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client."; + goto Exit; + } + + hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ), + ( void** ) &captureClient ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle."; + goto Exit; + } + + // don't configure captureEvent if in loopback mode + if ( !loopbackEnabled ) + { + // configure captureEvent to trigger on every available capture buffer + captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); + if ( !captureEvent ) { + errorType = RtAudioError::SYSTEM_ERROR; + errorText = "RtApiWasapi::wasapiThread: Unable to create capture event."; + goto Exit; + } + + hr = captureAudioClient->SetEventHandle( captureEvent ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to set capture event handle."; + goto Exit; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient; + + // reset the capture stream + hr = captureAudioClient->Reset(); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to reset capture stream."; + goto Exit; + } + + // start the capture stream + hr = captureAudioClient->Start(); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to start capture stream."; + goto Exit; + } + } + + unsigned int inBufferSize = 0; + hr = captureAudioClient->GetBufferSize( &inBufferSize ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to get capture buffer size."; + goto Exit; + } + + // scale outBufferSize according to stream->user sample rate ratio + unsigned int outBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT]; + inBufferSize *= stream_.nDeviceChannels[INPUT]; + + // set captureBuffer size + captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) ); + } + + // start render stream if applicable + if ( renderAudioClient ) { + hr = renderAudioClient->GetMixFormat( &renderFormat ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; + goto Exit; + } + + // init renderResampler + renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64, + formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT], + stream_.sampleRate, renderFormat->nSamplesPerSec ); + + renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate ); + + if ( !renderClient ) { + hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, + AUDCLNT_STREAMFLAGS_EVENTCALLBACK, + 0, + 0, + renderFormat, + NULL ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to initialize render audio client."; + goto Exit; + } + + hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ), + ( void** ) &renderClient ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle."; + goto Exit; + } + + // configure renderEvent to trigger on every available render buffer + renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); + if ( !renderEvent ) { + errorType = RtAudioError::SYSTEM_ERROR; + errorText = "RtApiWasapi::wasapiThread: Unable to create render event."; + goto Exit; + } + + hr = renderAudioClient->SetEventHandle( renderEvent ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to set render event handle."; + goto Exit; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient; + ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent; + + // reset the render stream + hr = renderAudioClient->Reset(); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to reset render stream."; + goto Exit; + } + + // start the render stream + hr = renderAudioClient->Start(); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to start render stream."; + goto Exit; + } + } + + unsigned int outBufferSize = 0; + hr = renderAudioClient->GetBufferSize( &outBufferSize ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to get render buffer size."; + goto Exit; + } + + // scale inBufferSize according to user->stream sample rate ratio + unsigned int inBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT]; + outBufferSize *= stream_.nDeviceChannels[OUTPUT]; + + // set renderBuffer size + renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) ); + } + + // malloc buffer memory + if ( stream_.mode == INPUT ) + { + using namespace std; // for ceilf + convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); + deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); + } + else if ( stream_.mode == OUTPUT ) + { + convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); + deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); + } + else if ( stream_.mode == DUPLEX ) + { + convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), + ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); + deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), + stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); + } + + convBuffSize *= 2; // allow overflow for *SrRatio remainders + convBuffer = ( char* ) calloc( convBuffSize, 1 ); + stream_.deviceBuffer = ( char* ) calloc( deviceBuffSize, 1 ); + if ( !convBuffer || !stream_.deviceBuffer ) { + errorType = RtAudioError::MEMORY_ERROR; + errorText = "RtApiWasapi::wasapiThread: Error allocating device buffer memory."; + goto Exit; + } + + // stream process loop + while ( stream_.state != STREAM_STOPPING ) { + if ( !callbackPulled ) { + // Callback Input + // ============== + // 1. Pull callback buffer from inputBuffer + // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count + // Convert callback buffer to user format + + if ( captureAudioClient ) + { + int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio ); + + convBufferSize = 0; + while ( convBufferSize < stream_.bufferSize ) + { + // Pull callback buffer from inputBuffer + callbackPulled = captureBuffer.pullBuffer( convBuffer, + samplesToPull * stream_.nDeviceChannels[INPUT], + stream_.deviceFormat[INPUT] ); + + if ( !callbackPulled ) + { + break; + } + + // Convert callback buffer to user sample rate + unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); + unsigned int convSamples = 0; + + captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset, + convBuffer, + samplesToPull, + convSamples, + convBufferSize == 0 ? -1 : stream_.bufferSize - convBufferSize ); + + convBufferSize += convSamples; + samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples + } + + if ( callbackPulled ) + { + if ( stream_.doConvertBuffer[INPUT] ) { + // Convert callback buffer to user format + convertBuffer( stream_.userBuffer[INPUT], + stream_.deviceBuffer, + stream_.convertInfo[INPUT] ); + } + else { + // no further conversion, simple copy deviceBuffer to userBuffer + memcpy( stream_.userBuffer[INPUT], + stream_.deviceBuffer, + stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) ); + } + } + } + else { + // if there is no capture stream, set callbackPulled flag + callbackPulled = true; + } + + // Execute Callback + // ================ + // 1. Execute user callback method + // 2. Handle return value from callback + + // if callback has not requested the stream to stop + if ( callbackPulled && !callbackStopped ) { + // Execute user callback method + callbackResult = callback( stream_.userBuffer[OUTPUT], + stream_.userBuffer[INPUT], + stream_.bufferSize, + getStreamTime(), + captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0, + stream_.callbackInfo.userData ); + + // tick stream time + RtApi::tickStreamTime(); + + // Handle return value from callback + if ( callbackResult == 1 ) { + // instantiate a thread to stop this thread + HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL ); + if ( !threadHandle ) { + errorType = RtAudioError::THREAD_ERROR; + errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread."; + goto Exit; + } + else if ( !CloseHandle( threadHandle ) ) { + errorType = RtAudioError::THREAD_ERROR; + errorText = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle."; + goto Exit; + } + + callbackStopped = true; + } + else if ( callbackResult == 2 ) { + // instantiate a thread to stop this thread + HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL ); + if ( !threadHandle ) { + errorType = RtAudioError::THREAD_ERROR; + errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread."; + goto Exit; + } + else if ( !CloseHandle( threadHandle ) ) { + errorType = RtAudioError::THREAD_ERROR; + errorText = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle."; + goto Exit; + } + + callbackStopped = true; + } + } + } + + // Callback Output + // =============== + // 1. Convert callback buffer to stream format + // 2. Convert callback buffer to stream sample rate and channel count + // 3. Push callback buffer into outputBuffer + + if ( renderAudioClient && callbackPulled ) + { + // if the last call to renderBuffer.PushBuffer() was successful + if ( callbackPushed || convBufferSize == 0 ) + { + if ( stream_.doConvertBuffer[OUTPUT] ) + { + // Convert callback buffer to stream format + convertBuffer( stream_.deviceBuffer, + stream_.userBuffer[OUTPUT], + stream_.convertInfo[OUTPUT] ); + + } + else { + // no further conversion, simple copy userBuffer to deviceBuffer + memcpy( stream_.deviceBuffer, + stream_.userBuffer[OUTPUT], + stream_.bufferSize * stream_.nUserChannels[OUTPUT] * formatBytes( stream_.userFormat ) ); + } + + // Convert callback buffer to stream sample rate + renderResampler->Convert( convBuffer, + stream_.deviceBuffer, + stream_.bufferSize, + convBufferSize ); + } + + // Push callback buffer into outputBuffer + callbackPushed = renderBuffer.pushBuffer( convBuffer, + convBufferSize * stream_.nDeviceChannels[OUTPUT], + stream_.deviceFormat[OUTPUT] ); + } + else { + // if there is no render stream, set callbackPushed flag + callbackPushed = true; + } + + // Stream Capture + // ============== + // 1. Get capture buffer from stream + // 2. Push capture buffer into inputBuffer + // 3. If 2. was successful: Release capture buffer + + if ( captureAudioClient ) { + // if the callback input buffer was not pulled from captureBuffer, wait for next capture event + if ( !callbackPulled ) { + WaitForSingleObject( loopbackEnabled ? renderEvent : captureEvent, INFINITE ); + } + + // Get capture buffer from stream + hr = captureClient->GetBuffer( &streamBuffer, + &bufferFrameCount, + &captureFlags, NULL, NULL ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer."; + goto Exit; + } + + if ( bufferFrameCount != 0 ) { + // Push capture buffer into inputBuffer + if ( captureBuffer.pushBuffer( ( char* ) streamBuffer, + bufferFrameCount * stream_.nDeviceChannels[INPUT], + stream_.deviceFormat[INPUT] ) ) + { + // Release capture buffer + hr = captureClient->ReleaseBuffer( bufferFrameCount ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + goto Exit; + } + } + else + { + // Inform WASAPI that capture was unsuccessful + hr = captureClient->ReleaseBuffer( 0 ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + goto Exit; + } + } + } + else + { + // Inform WASAPI that capture was unsuccessful + hr = captureClient->ReleaseBuffer( 0 ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + goto Exit; + } + } + } + + // Stream Render + // ============= + // 1. Get render buffer from stream + // 2. Pull next buffer from outputBuffer + // 3. If 2. was successful: Fill render buffer with next buffer + // Release render buffer + + if ( renderAudioClient ) { + // if the callback output buffer was not pushed to renderBuffer, wait for next render event + if ( callbackPulled && !callbackPushed ) { + WaitForSingleObject( renderEvent, INFINITE ); + } + + // Get render buffer from stream + hr = renderAudioClient->GetBufferSize( &bufferFrameCount ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size."; + goto Exit; + } + + hr = renderAudioClient->GetCurrentPadding( &numFramesPadding ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding."; + goto Exit; + } + + bufferFrameCount -= numFramesPadding; + + if ( bufferFrameCount != 0 ) { + hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer."; + goto Exit; + } + + // Pull next buffer from outputBuffer + // Fill render buffer with next buffer + if ( renderBuffer.pullBuffer( ( char* ) streamBuffer, + bufferFrameCount * stream_.nDeviceChannels[OUTPUT], + stream_.deviceFormat[OUTPUT] ) ) + { + // Release render buffer + hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + goto Exit; + } + } + else + { + // Inform WASAPI that render was unsuccessful + hr = renderClient->ReleaseBuffer( 0, 0 ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + goto Exit; + } + } + } + else + { + // Inform WASAPI that render was unsuccessful + hr = renderClient->ReleaseBuffer( 0, 0 ); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + goto Exit; + } + } + } + + // if the callback buffer was pushed renderBuffer reset callbackPulled flag + if ( callbackPushed ) { + // unsetting the callbackPulled flag lets the stream know that + // the audio device is ready for another callback output buffer. + callbackPulled = false; + } + + } + +Exit: + // clean up + CoTaskMemFree( captureFormat ); + CoTaskMemFree( renderFormat ); + + free ( convBuffer ); + delete renderResampler; + delete captureResampler; + + CoUninitialize(); + + // update stream state + stream_.state = STREAM_STOPPED; + + if ( !errorText.empty() ) + { + errorText_ = errorText; + error( errorType ); + } +} + +//******************** End of __WINDOWS_WASAPI__ *********************// +#endif + + +#if defined(__WINDOWS_DS__) // Windows DirectSound API + +// Modified by Robin Davies, October 2005 +// - Improvements to DirectX pointer chasing. +// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30. +// - Auto-call CoInitialize for DSOUND and ASIO platforms. +// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007 +// Changed device query structure for RtAudio 4.0.7, January 2010 + +#include +#include +#include +#include +#include +#include +#include + +#if defined(__MINGW32__) + // missing from latest mingw winapi +#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ +#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ +#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ +#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */ +#endif + +#define MINIMUM_DEVICE_BUFFER_SIZE 32768 + +#ifdef _MSC_VER // if Microsoft Visual C++ +#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. +#endif + +static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) +{ + if ( pointer > bufferSize ) pointer -= bufferSize; + if ( laterPointer < earlierPointer ) laterPointer += bufferSize; + if ( pointer < earlierPointer ) pointer += bufferSize; + return pointer >= earlierPointer && pointer < laterPointer; +} + +// A structure to hold various information related to the DirectSound +// API implementation. +struct DsHandle { + unsigned int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + void *id[2]; + void *buffer[2]; + bool xrun[2]; + UINT bufferPointer[2]; + DWORD dsBufferSize[2]; + DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. + HANDLE condition; + + DsHandle() + :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } +}; + +// Declarations for utility functions, callbacks, and structures +// specific to the DirectSound implementation. +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR module, + LPVOID lpContext ); + +static const char* getErrorString( int code ); + +static unsigned __stdcall callbackHandler( void *ptr ); + +struct DsDevice { + LPGUID id[2]; + bool validId[2]; + bool found; + std::string name; + + DsDevice() + : found(false) { validId[0] = false; validId[1] = false; } +}; + +struct DsProbeData { + bool isInput; + std::vector* dsDevices; +}; + +RtApiDs :: RtApiDs() +{ + // Dsound will run both-threaded. If CoInitialize fails, then just + // accept whatever the mainline chose for a threading model. + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( !FAILED( hr ) ) coInitialized_ = true; +} + +RtApiDs :: ~RtApiDs() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); + if ( coInitialized_ ) CoUninitialize(); // balanced call. +} + +// The DirectSound default output is always the first device. +unsigned int RtApiDs :: getDefaultOutputDevice( void ) +{ + return 0; +} + +// The DirectSound default input is always the first input device, +// which is the first capture device enumerated. +unsigned int RtApiDs :: getDefaultInputDevice( void ) +{ + return 0; +} + +unsigned int RtApiDs :: getDeviceCount( void ) +{ + // Set query flag for previously found devices to false, so that we + // can check for any devices that have disappeared. + for ( unsigned int i=0; i(dsDevices.size()); +} + +RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + if ( dsDevices.size() == 0 ) { + // Force a query of all devices + getDeviceCount(); + if ( dsDevices.size() == 0 ) { + errorText_ = "RtApiDs::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + } + + if ( device >= dsDevices.size() ) { + errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + HRESULT result; + if ( dsDevices[ device ].validId[0] == false ) goto probeInput; + + LPDIRECTSOUND output; + DSCAPS outCaps; + result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto probeInput; + } + + outCaps.dwSize = sizeof( outCaps ); + result = output->GetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto probeInput; + } + + // Get output channel information. + info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; + + // Get sample rate information. + info.sampleRates.clear(); + for ( unsigned int k=0; k= (unsigned int) outCaps.dwMinSecondarySampleRate && + SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) { + info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + } + } + + // Get format information. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; + if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; + + output->Release(); + + if ( getDefaultOutputDevice() == device ) + info.isDefaultOutput = true; + + if ( dsDevices[ device ].validId[1] == false ) { + info.name = dsDevices[ device ].name; + info.probed = true; + return info; + } + + probeInput: + + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Get input channel information. + info.inputChannels = inCaps.dwChannels; + + // Get sample rate and format information. + std::vector rates; + if ( inCaps.dwChannels >= 2 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; + + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 ); + } + } + else if ( inCaps.dwChannels == 1 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; + + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 ); + } + } + else info.inputChannels = 0; // technically, this would be an error + + input->Release(); + + if ( info.inputChannels == 0 ) return info; + + // Copy the supported rates to the info structure but avoid duplication. + bool found; + for ( unsigned int i=0; i 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + if ( device == 0 ) info.isDefaultInput = true; + + // Copy name and return. + info.name = dsDevices[ device ].name; + info.probed = true; + return info; +} + +bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + if ( channels + firstChannel > 2 ) { + errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; + return FAILURE; + } + + size_t nDevices = dsDevices.size(); + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiDs::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + if ( mode == OUTPUT ) { + if ( dsDevices[ device ].validId[0] == false ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + else { // mode == INPUT + if ( dsDevices[ device ].validId[1] == false ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // According to a note in PortAudio, using GetDesktopWindow() + // instead of GetForegroundWindow() is supposed to avoid problems + // that occur when the application's window is not the foreground + // window. Also, if the application window closes before the + // DirectSound buffer, DirectSound can crash. In the past, I had + // problems when using GetDesktopWindow() but it seems fine now + // (January 2010). I'll leave it commented here. + // HWND hWnd = GetForegroundWindow(); + HWND hWnd = GetDesktopWindow(); + + // Check the numberOfBuffers parameter and limit the lowest value to + // two. This is a judgement call and a value of two is probably too + // low for capture, but it should work for playback. + int nBuffers = 0; + if ( options ) nBuffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; + if ( nBuffers < 2 ) nBuffers = 3; + + // Check the lower range of the user-specified buffer size and set + // (arbitrarily) to a lower bound of 32. + if ( *bufferSize < 32 ) *bufferSize = 32; + + // Create the wave format structure. The data format setting will + // be determined later. + WAVEFORMATEX waveFormat; + ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); + waveFormat.wFormatTag = WAVE_FORMAT_PCM; + waveFormat.nChannels = channels + firstChannel; + waveFormat.nSamplesPerSec = (unsigned long) sampleRate; + + // Determine the device buffer size. By default, we'll use the value + // defined above (32K), but we will grow it to make allowances for + // very large software buffer sizes. + DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE; + DWORD dsPointerLeadTime = 0; + + void *ohandle = 0, *bhandle = 0; + HRESULT result; + if ( mode == OUTPUT ) { + + LPDIRECTSOUND output; + result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + DSCAPS outCaps; + outCaps.dwSize = sizeof( outCaps ); + result = output->GetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check channel information. + if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check format information. Use 16-bit format unless not + // supported or user requests 8-bit. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && + !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + stream_.userFormat = format; + + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; + + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > dsBufferSize ) + dsBufferSize *= 2; + + // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. + // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); + // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. + result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Even though we will write to the secondary buffer, we need to + // access the primary buffer to set the correct output format + // (since the default is 8-bit, 22 kHz!). Setup the DS primary + // buffer description. + DSBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; + + // Obtain the primary buffer + LPDIRECTSOUNDBUFFER buffer; + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the primary DS buffer sound format. + result = buffer->SetFormat( &waveFormat ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Setup the secondary DS buffer description. + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCHARDWARE ); // Force hardware mixing + bufferDescription.dwBufferBytes = dsBufferSize; + bufferDescription.lpwfxFormat = &waveFormat; + + // Try to create the secondary DS buffer. If that doesn't work, + // try to use software mixing. Otherwise, there's a problem. + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCSOFTWARE ); // Force software mixing + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Get the buffer size ... might be different from what we specified. + DSBCAPS dsbcaps; + dsbcaps.dwSize = sizeof( DSBCAPS ); + result = buffer->GetCaps( &dsbcaps ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + dsBufferSize = dsbcaps.dwBufferBytes; + + // Lock the DS buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + ohandle = (void *) output; + bhandle = (void *) buffer; + } + + if ( mode == INPUT ) { + + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check channel information. + if ( inCaps.dwChannels < channels + firstChannel ) { + errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; + return FAILURE; + } + + // Check format information. Use 16-bit format unless user + // requests 8-bit. + DWORD deviceFormats; + if ( channels + firstChannel == 2 ) { + deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + } + else { // channel == 1 + deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + } + stream_.userFormat = format; + + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; + + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > dsBufferSize ) + dsBufferSize *= 2; + + // Setup the secondary DS buffer description. + DSCBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); + bufferDescription.dwFlags = 0; + bufferDescription.dwReserved = 0; + bufferDescription.dwBufferBytes = dsBufferSize; + bufferDescription.lpwfxFormat = &waveFormat; + + // Create the capture buffer. + LPDIRECTSOUNDCAPTUREBUFFER buffer; + result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Get the buffer size ... might be different from what we specified. + DSCBCAPS dscbcaps; + dscbcaps.dwSize = sizeof( DSCBCAPS ); + result = buffer->GetCaps( &dscbcaps ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + dsBufferSize = dscbcaps.dwBufferBytes; + + // NOTE: We could have a problem here if this is a duplex stream + // and the play and capture hardware buffer sizes are different + // (I'm actually not sure if that is a problem or not). + // Currently, we are not verifying that. + + // Lock the capture buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Zero the buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + ohandle = (void *) input; + bhandle = (void *) buffer; + } + + // Set various stream parameters + DsHandle *handle = 0; + stream_.nDeviceChannels[mode] = channels + firstChannel; + stream_.nUserChannels[mode] = channels; + stream_.bufferSize = *bufferSize; + stream_.channelOffset[mode] = firstChannel; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // Set flag for buffer conversion + stream_.doConvertBuffer[mode] = false; + if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) + stream_.doConvertBuffer[mode] = true; + if (stream_.userFormat != stream_.deviceFormat[mode]) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + // Allocate our DsHandle structures for the stream. + if ( stream_.apiHandle == 0 ) { + try { + handle = new DsHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; + goto error; + } + + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; + } + else + handle = (DsHandle *) stream_.apiHandle; + handle->id[mode] = ohandle; + handle->buffer[mode] = bhandle; + handle->dsBufferSize[mode] = dsBufferSize; + handle->dsPointerLeadTime[mode] = dsPointerLeadTime; + + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream_.mode = DUPLEX; + else + stream_.mode = mode; + stream_.nBuffers = nBuffers; + stream_.sampleRate = sampleRate; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + // Setup the callback thread. + if ( stream_.callbackInfo.isRunning == false ) { + unsigned threadId; + stream_.callbackInfo.isRunning = true; + stream_.callbackInfo.object = (void *) this; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, + &stream_.callbackInfo, 0, &threadId ); + if ( stream_.callbackInfo.thread == 0 ) { + errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; + goto error; + } + + // Boost DS thread priority + SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); + } + return SUCCESS; + + error: + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) buffer->Release(); + object->Release(); + } + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) buffer->Release(); + object->Release(); + } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiDs :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + // Stop the callback thread. + stream_.callbackInfo.isRunning = false; + WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); + CloseHandle( (HANDLE) stream_.callbackInfo.thread ); + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiDs :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiDs::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + + // Increase scheduler frequency on lesser windows (a side-effect of + // increasing timer accuracy). On greater windows (Win2K or later), + // this is already in effect. + timeBeginPeriod( 1 ); + + buffersRolling = false; + duplexPrerollBytes = 0; + + if ( stream_.mode == DUPLEX ) { + // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize. + duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] ); + } + + HRESULT result = 0; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + result = buffer->Start( DSCBSTART_LOOPING ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + handle->drainCounter = 0; + handle->internalDrain = false; + ResetEvent( handle->condition ); + stream_.state = STREAM_RUNNING; + + unlock: + if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiDs :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + HRESULT result = 0; + LPVOID audioPtr; + DWORD dataLen; + DsHandle *handle = (DsHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + WaitForSingleObject( handle->condition, INFINITE ); // block until signaled + } + + stream_.state = STREAM_STOPPED; + + MUTEX_LOCK( &stream_.mutex ); + + // Stop the buffer and clear memory + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // If we start playing again, we must begin at beginning of buffer. + handle->bufferPointer[0] = 0; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + audioPtr = NULL; + dataLen = 0; + + stream_.state = STREAM_STOPPED; + + if ( stream_.mode != DUPLEX ) + MUTEX_LOCK( &stream_.mutex ); + + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // If we start recording again, we must begin at beginning of buffer. + handle->bufferPointer[1] = 0; + } + + unlock: + timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. + MUTEX_UNLOCK( &stream_.mutex ); + + if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiDs :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + stopStream(); +} + +void RtApiDs :: callbackEvent() +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) { + Sleep( 50 ); // sleep 50 milliseconds + return; + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + DsHandle *handle = (DsHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > stream_.nBuffers + 2 ) { + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else + stopStream(); + return; + } + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + abortStream(); + return; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + HRESULT result; + DWORD currentWritePointer, safeWritePointer; + DWORD currentReadPointer, safeReadPointer; + UINT nextWritePointer; + + LPVOID buffer1 = NULL; + LPVOID buffer2 = NULL; + DWORD bufferSize1 = 0; + DWORD bufferSize2 = 0; + + char *buffer; + long bufferBytes; + + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + if ( buffersRolling == false ) { + if ( stream_.mode == DUPLEX ) { + //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + + // It takes a while for the devices to get rolling. As a result, + // there's no guarantee that the capture and write device pointers + // will move in lockstep. Wait here for both devices to start + // rolling, and then set our buffer pointers accordingly. + // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 + // bytes later than the write buffer. + + // Stub: a serious risk of having a pre-emptive scheduling round + // take place between the two GetCurrentPosition calls... but I'm + // really not sure how to solve the problem. Temporarily boost to + // Realtime priority, maybe; but I'm not sure what priority the + // DirectSound service threads run at. We *should* be roughly + // within a ms or so of correct. + + LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + + DWORD startSafeWritePointer, startSafeReadPointer; + + result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + while ( true ) { + result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break; + Sleep( 1 ); + } + + //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + + handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; + if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; + handle->bufferPointer[1] = safeReadPointer; + } + else if ( stream_.mode == OUTPUT ) { + + // Set the proper nextWritePosition after initial startup. + LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; + if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; + } + + buffersRolling = true; + } + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + memset( stream_.userBuffer[0], 0, bufferBytes ); + } + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; + bufferBytes *= formatBytes( stream_.deviceFormat[0] ); + } + else { + buffer = stream_.userBuffer[0]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + } + + // No byte swapping necessary in DirectSound implementation. + + // Ahhh ... windoze. 16-bit data is signed but 8-bit data is + // unsigned. So, we need to convert our signed 8-bit data here to + // unsigned. + if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) + for ( int i=0; idsBufferSize[0]; + nextWritePointer = handle->bufferPointer[0]; + + DWORD endWrite, leadPointer; + while ( true ) { + // Find out where the read and "safe write" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + // We will copy our output buffer into the region between + // safeWritePointer and leadPointer. If leadPointer is not + // beyond the next endWrite position, wait until it is. + leadPointer = safeWritePointer + handle->dsPointerLeadTime[0]; + //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl; + if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize; + if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset + endWrite = nextWritePointer + bufferBytes; + + // Check whether the entire write region is behind the play pointer. + if ( leadPointer >= endWrite ) break; + + // If we are here, then we must wait until the leadPointer advances + // beyond the end of our next write region. We use the + // Sleep() function to suspend operation until that happens. + double millis = ( endWrite - leadPointer ) * 1000.0; + millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + } + + if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize ) + || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { + // We've strayed into the forbidden zone ... resync the read pointer. + handle->xrun[0] = true; + nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes; + if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize; + handle->bufferPointer[0] = nextWritePointer; + endWrite = nextWritePointer + bufferBytes; + } + + // Lock free space in the buffer + result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + // Copy our buffer into the DS buffer + CopyMemory( buffer1, buffer, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); + + // Update our buffer offset and unlock sound buffer + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize; + handle->bufferPointer[0] = nextWritePointer; + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; + bufferBytes *= formatBytes( stream_.deviceFormat[1] ); + } + else { + buffer = stream_.userBuffer[1]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; + bufferBytes *= formatBytes( stream_.userFormat ); + } + + LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + long nextReadPointer = handle->bufferPointer[1]; + DWORD dsBufferSize = handle->dsBufferSize[1]; + + // Find out where the write and "safe read" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset + DWORD endRead = nextReadPointer + bufferBytes; + + // Handling depends on whether we are INPUT or DUPLEX. + // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, + // then a wait here will drag the write pointers into the forbidden zone. + // + // In DUPLEX mode, rather than wait, we will back off the read pointer until + // it's in a safe position. This causes dropouts, but it seems to be the only + // practical way to sync up the read and write pointers reliably, given the + // the very complex relationship between phase and increment of the read and write + // pointers. + // + // In order to minimize audible dropouts in DUPLEX mode, we will + // provide a pre-roll period of 0.5 seconds in which we return + // zeros from the read buffer while the pointers sync up. + + if ( stream_.mode == DUPLEX ) { + if ( safeReadPointer < endRead ) { + if ( duplexPrerollBytes <= 0 ) { + // Pre-roll time over. Be more agressive. + int adjustment = endRead-safeReadPointer; + + handle->xrun[1] = true; + // Two cases: + // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, + // and perform fine adjustments later. + // - small adjustments: back off by twice as much. + if ( adjustment >= 2*bufferBytes ) + nextReadPointer = safeReadPointer-2*bufferBytes; + else + nextReadPointer = safeReadPointer-bufferBytes-adjustment; + + if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; + + } + else { + // In pre=roll time. Just do it. + nextReadPointer = safeReadPointer - bufferBytes; + while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; + } + endRead = nextReadPointer + bufferBytes; + } + } + else { // mode == INPUT + while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) { + // See comments for playback. + double millis = (endRead - safeReadPointer) * 1000.0; + millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + + // Wake up and find out where we are now. + result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset + } + } + + // Lock free space in the buffer + result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + if ( duplexPrerollBytes <= 0 ) { + // Copy our buffer into the DS buffer + CopyMemory( buffer, buffer1, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); + } + else { + memset( buffer, 0, bufferSize1 ); + if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); + duplexPrerollBytes -= bufferSize1 + bufferSize2; + } + + // Update our buffer offset and unlock sound buffer + nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize; + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + handle->bufferPointer[1] = nextReadPointer; + + // No byte swapping necessary in DirectSound implementation. + + // If necessary, convert 8-bit data from unsigned to signed. + if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 ) + for ( int j=0; jobject; + bool* isRunning = &info->isRunning; + + while ( *isRunning == true ) { + object->callbackEvent(); + } + + _endthreadex( 0 ); + return 0; +} + +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR /*module*/, + LPVOID lpContext ) +{ + struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext; + std::vector& dsDevices = *probeInfo.dsDevices; + + HRESULT hr; + bool validDevice = false; + if ( probeInfo.isInput == true ) { + DSCCAPS caps; + LPDIRECTSOUNDCAPTURE object; + + hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; + + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) + validDevice = true; + } + object->Release(); + } + else { + DSCAPS caps; + LPDIRECTSOUND object; + hr = DirectSoundCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; + + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) + validDevice = true; + } + object->Release(); + } + + // If good device, then save its name and guid. + std::string name = convertCharPointerToStdString( description ); + //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" ) + if ( lpguid == NULL ) + name = "Default Device"; + if ( validDevice ) { + for ( unsigned int i=0; i +#include + + // A structure to hold various information related to the ALSA API + // implementation. +struct AlsaHandle { + snd_pcm_t *handles[2]; + bool synchronized; + bool xrun[2]; + pthread_cond_t runnable_cv; + bool runnable; + + AlsaHandle() +#if _cplusplus >= 201103L + :handles{nullptr, nullptr}, synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; } +#else + : synchronized(false), runnable(false) { handles[0] = NULL; handles[1] = NULL; xrun[0] = false; xrun[1] = false; } +#endif +}; + +static void *alsaCallbackHandler( void * ptr ); + +RtApiAlsa :: RtApiAlsa() +{ + // Nothing to do here. +} + +RtApiAlsa :: ~RtApiAlsa() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiAlsa :: getDeviceCount( void ) +{ + unsigned nDevices = 0; + int result, subdevice, card; + char name[64]; + snd_ctl_t *handle = 0; + + strcpy(name, "default"); + result = snd_ctl_open( &handle, "default", 0 ); + if (result == 0) { + nDevices++; + snd_ctl_close( handle ); + } + + // Count cards and devices + card = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &handle, name, 0 ); + if ( result < 0 ) { + handle = 0; + errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto nextcard; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( handle, &subdevice ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + break; + } + if ( subdevice < 0 ) + break; + nDevices++; + } + nextcard: + if ( handle ) + snd_ctl_close( handle ); + snd_card_next( &card ); + } + + return nDevices; +} + +RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + unsigned nDevices = 0; + int result=-1, subdevice=-1, card=-1; + char name[64]; + snd_ctl_t *chandle = 0; + + result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); + if ( result == 0 ) { + if ( nDevices++ == device ) { + strcpy( name, "default" ); + goto foundDevice; + } + } + if ( chandle ) + snd_ctl_close( chandle ); + + // Count cards and devices + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + if ( result < 0 ) { + chandle = 0; + errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto nextcard; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( chandle, &subdevice ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + break; + } + if ( subdevice < 0 ) break; + if ( nDevices == device ) { + sprintf( name, "hw:%d,%d", card, subdevice ); + goto foundDevice; + } + nDevices++; + } + nextcard: + if ( chandle ) + snd_ctl_close( chandle ); + snd_card_next( &card ); + } + + if ( nDevices == 0 ) { + errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + foundDevice: + + // If a stream is already open, we cannot probe the stream devices. + // Thus, use the saved results. + if ( stream_.state != STREAM_CLOSED && + ( stream_.device[0] == device || stream_.device[1] == device ) ) { + snd_ctl_close( chandle ); + if ( device >= devices_.size() ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; + error( RtAudioError::WARNING ); + return info; + } + return devices_[ device ]; + } + + int openMode = SND_PCM_ASYNC; + snd_pcm_stream_t stream; + snd_pcm_info_t *pcminfo; + snd_pcm_info_alloca( &pcminfo ); + snd_pcm_t *phandle; + snd_pcm_hw_params_t *params; + snd_pcm_hw_params_alloca( ¶ms ); + + // First try for playback unless default device (which has subdev -1) + stream = SND_PCM_STREAM_PLAYBACK; + snd_pcm_info_set_stream( pcminfo, stream ); + if ( subdevice != -1 ) { + snd_pcm_info_set_device( pcminfo, subdevice ); + snd_pcm_info_set_subdevice( pcminfo, 0 ); + + result = snd_ctl_pcm_info( chandle, pcminfo ); + if ( result < 0 ) { + // Device probably doesn't support playback. + goto captureProbe; + } + } + + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto captureProbe; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto captureProbe; + } + + // Get output channel information. + unsigned int value; + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto captureProbe; + } + info.outputChannels = value; + snd_pcm_close( phandle ); + + captureProbe: + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); + + // Now try for capture unless default device (with subdev = -1) + if ( subdevice != -1 ) { + result = snd_ctl_pcm_info( chandle, pcminfo ); + snd_ctl_close( chandle ); + if ( result < 0 ) { + // Device probably doesn't support capture. + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + } + else + snd_ctl_close( chandle ); + + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + info.inputChannels = value; + snd_pcm_close( phandle ); + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // ALSA doesn't provide default devices so we'll use the first available one. + if ( device == 0 && info.outputChannels > 0 ) + info.isDefaultOutput = true; + if ( device == 0 && info.inputChannels > 0 ) + info.isDefaultInput = true; + + probeParameters: + // At this point, we just need to figure out the supported data + // formats and sample rates. We'll proceed by opening the device in + // the direction with the maximum number of channels, or playback if + // they are equal. This might limit our sample rate options, but so + // be it. + + if ( info.outputChannels >= info.inputChannels ) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); + + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Test our discrete set of sample rate values. + info.sampleRates.clear(); + for ( unsigned int i=0; i info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[i]; + } + } + if ( info.sampleRates.size() == 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Probe the supported data formats ... we don't care about endian-ness just yet + snd_pcm_format_t format; + info.nativeFormats = 0; + format = SND_PCM_FORMAT_S8; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT8; + format = SND_PCM_FORMAT_S16; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT16; + format = SND_PCM_FORMAT_S24; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT24; + format = SND_PCM_FORMAT_S32; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT32; + format = SND_PCM_FORMAT_FLOAT; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_FLOAT32; + format = SND_PCM_FORMAT_FLOAT64; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_FLOAT64; + + // Check that we have at least one supported format + if ( info.nativeFormats == 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Get the device name + if (strncmp(name, "default", 7)!=0) { + char *cardname; + result = snd_card_get_name( card, &cardname ); + if ( result >= 0 ) { + sprintf( name, "hw:%s,%d", cardname, subdevice ); + free( cardname ); + } + } + info.name = name; + + // That's all ... close the device and return + snd_pcm_close( phandle ); + info.probed = true; + return info; +} + +void RtApiAlsa :: saveDeviceInfo( void ) +{ + devices_.clear(); + + unsigned int nDevices = getDeviceCount(); + devices_.resize( nDevices ); + for ( unsigned int i=0; iflags & RTAUDIO_ALSA_USE_DEFAULT) ) + { + strcpy(name, "default"); + result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); + if ( result == 0 ) { + if ( nDevices == device ) { + strcpy( name, "default" ); + snd_ctl_close( chandle ); + goto foundDevice; + } + nDevices++; + } + } + + else { + nDevices++; + // Count cards and devices + card = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( chandle, &subdevice ); + if ( result < 0 ) break; + if ( subdevice < 0 ) break; + if ( nDevices == device ) { + sprintf( name, "hw:%d,%d", card, subdevice ); + snd_ctl_close( chandle ); + goto foundDevice; + } + nDevices++; + } + snd_ctl_close( chandle ); + snd_card_next( &card ); + } + + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + } + + foundDevice: + + // The getDeviceInfo() function will not work for a device that is + // already open. Thus, we'll probe the system before opening a + // stream and save the results for use by getDeviceInfo(). + if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once + this->saveDeviceInfo(); + + snd_pcm_stream_t stream; + if ( mode == OUTPUT ) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; + + snd_pcm_t *phandle; + int openMode = SND_PCM_ASYNC; + result = snd_pcm_open( &phandle, name, stream, openMode ); + if ( result < 0 ) { + if ( mode == OUTPUT ) + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output."; + else + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Fill the parameter structure. + snd_pcm_hw_params_t *hw_params; + snd_pcm_hw_params_alloca( &hw_params ); + result = snd_pcm_hw_params_any( phandle, hw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" ); + snd_pcm_hw_params_dump( hw_params, out ); +#endif + + // Set access ... check user preference. + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) { + stream_.userInterleaved = false; + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); + if ( result < 0 ) { + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); + stream_.deviceInterleaved[mode] = true; + } + else + stream_.deviceInterleaved[mode] = false; + } + else { + stream_.userInterleaved = true; + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); + if ( result < 0 ) { + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); + stream_.deviceInterleaved[mode] = false; + } + else + stream_.deviceInterleaved[mode] = true; + } + + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine how to set the device format. + stream_.userFormat = format; + snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN; + + if ( format == RTAUDIO_SINT8 ) + deviceFormat = SND_PCM_FORMAT_S8; + else if ( format == RTAUDIO_SINT16 ) + deviceFormat = SND_PCM_FORMAT_S16; + else if ( format == RTAUDIO_SINT24 ) + deviceFormat = SND_PCM_FORMAT_S24; + else if ( format == RTAUDIO_SINT32 ) + deviceFormat = SND_PCM_FORMAT_S32; + else if ( format == RTAUDIO_FLOAT32 ) + deviceFormat = SND_PCM_FORMAT_FLOAT; + else if ( format == RTAUDIO_FLOAT64 ) + deviceFormat = SND_PCM_FORMAT_FLOAT64; + + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) { + stream_.deviceFormat[mode] = format; + goto setFormat; + } + + // The user requested format is not natively supported by the device. + deviceFormat = SND_PCM_FORMAT_FLOAT64; + if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_FLOAT; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S32; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S24; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S16; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S8; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + goto setFormat; + } + + // If we get here, no supported format was found. + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; + + setFormat: + result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine whether byte-swaping is necessary. + stream_.doByteSwap[mode] = false; + if ( deviceFormat != SND_PCM_FORMAT_S8 ) { + result = snd_pcm_format_cpu_endian( deviceFormat ); + if ( result == 0 ) + stream_.doByteSwap[mode] = true; + else if (result < 0) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Set the sample rate. + result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine the number of channels for this device. We support a possible + // minimum device channel number > than the value requested by the user. + stream_.nUserChannels[mode] = channels; + unsigned int value; + result = snd_pcm_hw_params_get_channels_max( hw_params, &value ); + unsigned int deviceChannels = value; + if ( result < 0 || deviceChannels < channels + firstChannel ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + result = snd_pcm_hw_params_get_channels_min( hw_params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + deviceChannels = value; + if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel; + stream_.nDeviceChannels[mode] = deviceChannels; + + // Set the device channels. + result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the buffer (or period) size. + int dir = 0; + snd_pcm_uframes_t periodSize = *bufferSize; + result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + *bufferSize = periodSize; + + // Set the buffer number, which in ALSA is referred to as the "period". + unsigned int periods = 0; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; + if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers; + if ( periods < 2 ) periods = 4; // a fairly safe default value + result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + stream_.bufferSize = *bufferSize; + + // Install the hardware configuration + result = snd_pcm_hw_params( phandle, hw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n"); + snd_pcm_hw_params_dump( hw_params, out ); +#endif + + // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns. + snd_pcm_sw_params_t *sw_params = NULL; + snd_pcm_sw_params_alloca( &sw_params ); + snd_pcm_sw_params_current( phandle, sw_params ); + snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); + snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); + snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); + + // The following two settings were suggested by Theo Veenker + //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); + //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); + + // here are two options for a fix + //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); + snd_pcm_uframes_t val; + snd_pcm_sw_params_get_boundary( sw_params, &val ); + snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); + + result = snd_pcm_sw_params( phandle, sw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n"); + snd_pcm_sw_params_dump( sw_params, out ); +#endif + + // Set flags for buffer conversion + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate the ApiHandle if necessary and then save. + AlsaHandle *apiInfo = 0; + if ( stream_.apiHandle == 0 ) { + try { + apiInfo = (AlsaHandle *) new AlsaHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; + goto error; + } + + if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + + stream_.apiHandle = (void *) apiInfo; + apiInfo->handles[0] = 0; + apiInfo->handles[1] = 0; + } + else { + apiInfo = (AlsaHandle *) stream_.apiHandle; + } + apiInfo->handles[mode] = phandle; + phandle = 0; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.sampleRate = sampleRate; + stream_.nBuffers = periods; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + // Setup thread if necessary. + if ( stream_.mode == OUTPUT && mode == INPUT ) { + // We had already set up an output stream. + stream_.mode = DUPLEX; + // Link the streams if possible. + apiInfo->synchronized = false; + if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 ) + apiInfo->synchronized = true; + else { + errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; + error( RtAudioError::WARNING ); + } + } + else { + stream_.mode = mode; + + // Setup callback thread. + stream_.callbackInfo.object = (void *) this; + + // Set the thread attributes for joinable and realtime scheduling + // priority (optional). The higher priority will only take affect + // if the program is run as root or suid. Note, under Linux + // processes with CAP_SYS_NICE privilege, a user can change + // scheduling policy and priority (thus need not be root). See + // POSIX "capabilities". + pthread_attr_t attr; + pthread_attr_init( &attr ); + pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); +#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + stream_.callbackInfo.doRealtime = true; + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + + // Set the policy BEFORE the priority. Otherwise it fails. + pthread_attr_setschedpolicy(&attr, SCHED_RR); + pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM); + // This is definitely required. Otherwise it fails. + pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED); + pthread_attr_setschedparam(&attr, ¶m); + } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#endif + + stream_.callbackInfo.isRunning = true; + result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); + pthread_attr_destroy( &attr ); + if ( result ) { + // Failed. Try instead with default attributes. + result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiAlsa::error creating callback thread!"; + goto error; + } + } + } + + return SUCCESS; + + error: + if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable_cv ); + if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); + if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); + delete apiInfo; + stream_.apiHandle = 0; + } + + if ( phandle) snd_pcm_close( phandle ); + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiAlsa :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + apiInfo->runnable = true; + pthread_cond_signal( &apiInfo->runnable_cv ); + } + MUTEX_UNLOCK( &stream_.mutex ); + pthread_join( stream_.callbackInfo.thread, NULL ); + + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + snd_pcm_drop( apiInfo->handles[0] ); + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) + snd_pcm_drop( apiInfo->handles[1] ); + } + + if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable_cv ); + if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); + if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); + delete apiInfo; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiAlsa :: startStream() +{ + // This method calls snd_pcm_prepare if the device isn't already in that state. + + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + + int result = 0; + snd_pcm_state_t state; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + state = snd_pcm_state( handle[0] ); + if ( state != SND_PCM_STATE_PREPARED ) { + result = snd_pcm_prepare( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + } + + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open + state = snd_pcm_state( handle[1] ); + if ( state != SND_PCM_STATE_PREPARED ) { + result = snd_pcm_prepare( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + } + + stream_.state = STREAM_RUNNING; + + unlock: + apiInfo->runnable = true; + pthread_cond_signal( &apiInfo->runnable_cv ); + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( apiInfo->synchronized ) + result = snd_pcm_drop( handle[0] ); + else + result = snd_pcm_drain( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + apiInfo->runnable = false; // fixes high CPU usage when stopped + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = snd_pcm_drop( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + apiInfo->runnable = false; // fixes high CPU usage when stopped + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: callbackEvent() +{ + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + while ( !apiInfo->runnable ) + pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex ); + + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; + } + + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + apiInfo->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + apiInfo->xrun[1] = false; + } + doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + + if ( doStopStream == 2 ) { + abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; + + int result; + char *buffer; + int channels; + snd_pcm_t **handle; + snd_pcm_sframes_t frames; + RtAudioFormat format; + handle = (snd_pcm_t **) apiInfo->handles; + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + channels = stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; + } + else { + buffer = stream_.userBuffer[1]; + channels = stream_.nUserChannels[1]; + format = stream_.userFormat; + } + + // Read samples from device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[1] ) + result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; ixrun[1] = true; + result = snd_pcm_prepare( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtAudioError::WARNING ); + goto tryOutput; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); + + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + + // Check stream latency + result = snd_pcm_delay( handle[1], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; + } + + tryOutput: + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + channels = stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + channels = stream_.nUserChannels[0]; + format = stream_.userFormat; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer(buffer, stream_.bufferSize * channels, format); + + // Write samples to device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[0] ) + result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; ixrun[0] = true; + result = snd_pcm_prepare( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + else + errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun."; + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtAudioError::WARNING ); + goto unlock; + } + + // Check stream latency + result = snd_pcm_delay( handle[0], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} + +static void *alsaCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAlsa *object = (RtApiAlsa *) info->object; + bool *isRunning = &info->isRunning; + +#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) + if ( info->doRealtime ) { + std::cerr << "RtAudio alsa: " << + (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << + "running realtime scheduling" << std::endl; + } +#endif + + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); + } + + pthread_exit( NULL ); +} + +//******************** End of __LINUX_ALSA__ *********************// +#endif + +#if defined(__LINUX_PULSE__) + +// Code written by Peter Meerwald, pmeerw@pmeerw.net +// and Tristan Matthews. + +#include +#include +#include +#include + +static pa_mainloop_api *rt_pa_mainloop_api = NULL; +struct PaDeviceInfo { + PaDeviceInfo() : sink_index(-1), source_index(-1) {} + int sink_index; + int source_index; + std::string sink_name; + std::string source_name; + RtAudio::DeviceInfo info; +}; +static struct { + std::vector dev; + std::string default_sink_name; + std::string default_source_name; + int default_rate; +} rt_pa_info; + +static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000, + 44100, 48000, 96000, 0}; + +struct rtaudio_pa_format_mapping_t { + RtAudioFormat rtaudio_format; + pa_sample_format_t pa_format; +}; + +static const rtaudio_pa_format_mapping_t supported_sampleformats[] = { + {RTAUDIO_SINT16, PA_SAMPLE_S16LE}, + {RTAUDIO_SINT24, PA_SAMPLE_S24LE}, + {RTAUDIO_SINT32, PA_SAMPLE_S32LE}, + {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE}, + {0, PA_SAMPLE_INVALID}}; + +struct PulseAudioHandle { + pa_simple *s_play; + pa_simple *s_rec; + pthread_t thread; + pthread_cond_t runnable_cv; + bool runnable; + PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { } +}; + +static void rt_pa_mainloop_api_quit(int ret) { + rt_pa_mainloop_api->quit(rt_pa_mainloop_api, ret); +} + +static void rt_pa_server_callback(pa_context *context, const pa_server_info *info, void *data){ + (void)context; + (void)data; + pa_sample_spec ss; + + if (!info) + rt_pa_mainloop_api_quit(1); + + ss = info->sample_spec; + + rt_pa_info.default_rate = ss.rate; + rt_pa_info.default_sink_name = info->default_sink_name; + rt_pa_info.default_source_name = info->default_source_name; + rt_pa_mainloop_api_quit(0); +} + +static void rt_pa_sink_info_cb(pa_context * /*c*/, const pa_sink_info *i, + int eol, void * /*userdata*/) +{ + if (eol) return; + PaDeviceInfo inf; + inf.info.name = pa_proplist_gets(i->proplist, "device.description"); + inf.info.probed = true; + inf.info.outputChannels = i->sample_spec.channels; + inf.info.preferredSampleRate = i->sample_spec.rate; + inf.info.isDefaultOutput = (rt_pa_info.default_sink_name == i->name); + inf.sink_index = i->index; + inf.sink_name = i->name; + for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) + inf.info.sampleRates.push_back( *sr ); + for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats; + fm->rtaudio_format; ++fm ) + inf.info.nativeFormats |= fm->rtaudio_format; + for (size_t i=0; i < rt_pa_info.dev.size(); i++) + { + /* Attempt to match up sink and source records by device description. */ + if (rt_pa_info.dev[i].info.name == inf.info.name) { + rt_pa_info.dev[i].sink_index = inf.sink_index; + rt_pa_info.dev[i].sink_name = inf.sink_name; + rt_pa_info.dev[i].info.outputChannels = inf.info.outputChannels; + rt_pa_info.dev[i].info.isDefaultOutput = inf.info.isDefaultOutput; + /* Assume duplex channels are minimum of input and output channels. */ + /* Uncomment if we add support for DUPLEX + if (rt_pa_info.dev[i].source_index > -1) + (inf.info.outputChannels < rt_pa_info.dev[i].info.inputChannels) + ? inf.info.outputChannels : rt_pa_info.dev[i].info.inputChannels; + */ + return; + } + } + /* try to ensure device #0 is the default */ + if (inf.info.isDefaultOutput) + rt_pa_info.dev.insert(rt_pa_info.dev.begin(), inf); + else + rt_pa_info.dev.push_back(inf); +} + +static void rt_pa_source_info_cb(pa_context * /*c*/, const pa_source_info *i, + int eol, void * /*userdata*/) +{ + if (eol) return; + PaDeviceInfo inf; + inf.info.name = pa_proplist_gets(i->proplist, "device.description"); + inf.info.probed = true; + inf.info.inputChannels = i->sample_spec.channels; + inf.info.preferredSampleRate = i->sample_spec.rate; + inf.info.isDefaultInput = (rt_pa_info.default_source_name == i->name); + inf.source_index = i->index; + inf.source_name = i->name; + for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) + inf.info.sampleRates.push_back( *sr ); + for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats; + fm->rtaudio_format; ++fm ) + inf.info.nativeFormats |= fm->rtaudio_format; + + for (size_t i=0; i < rt_pa_info.dev.size(); i++) + { + /* Attempt to match up sink and source records by device description. */ + if (rt_pa_info.dev[i].info.name == inf.info.name) { + rt_pa_info.dev[i].source_index = inf.source_index; + rt_pa_info.dev[i].source_name = inf.source_name; + rt_pa_info.dev[i].info.inputChannels = inf.info.inputChannels; + rt_pa_info.dev[i].info.isDefaultInput = inf.info.isDefaultInput; + /* Assume duplex channels are minimum of input and output channels. */ + /* Uncomment if we add support for DUPLEX + if (rt_pa_info.dev[i].sink_index > -1) { + rt_pa_info.dev[i].info.duplexChannels = + (inf.info.inputChannels < rt_pa_info.dev[i].info.outputChannels) + ? inf.info.inputChannels : rt_pa_info.dev[i].info.outputChannels; + } + */ + return; + } + } + /* try to ensure device #0 is the default */ + if (inf.info.isDefaultInput) + rt_pa_info.dev.insert(rt_pa_info.dev.begin(), inf); + else + rt_pa_info.dev.push_back(inf); +} + +static void rt_pa_context_state_callback(pa_context *context, void *userdata) { + (void)userdata; + + switch (pa_context_get_state(context)) { + case PA_CONTEXT_CONNECTING: + case PA_CONTEXT_AUTHORIZING: + case PA_CONTEXT_SETTING_NAME: + break; + + case PA_CONTEXT_READY: + rt_pa_info.dev.clear(); + pa_context_get_server_info(context, rt_pa_server_callback, NULL); + pa_context_get_sink_info_list(context, rt_pa_sink_info_cb, NULL); + pa_context_get_source_info_list(context, rt_pa_source_info_cb, NULL); + break; + + case PA_CONTEXT_TERMINATED: + rt_pa_mainloop_api_quit(0); + break; + + case PA_CONTEXT_FAILED: + default: + rt_pa_mainloop_api_quit(1); + } +} + +RtApiPulse::~RtApiPulse() +{ + if ( stream_.state != STREAM_CLOSED ) + closeStream(); +} + +void RtApiPulse::collectDeviceInfo( void ) +{ + pa_context *context = NULL; + pa_mainloop *m = NULL; + char *server = NULL; + int ret = 1; + + if (!(m = pa_mainloop_new())) { + errorStream_ << "RtApiPulse::DeviceInfo pa_mainloop_new() failed."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto quit; + } + + rt_pa_mainloop_api = pa_mainloop_get_api(m); + + if (!(context = pa_context_new_with_proplist(rt_pa_mainloop_api, NULL, NULL))) { + errorStream_ << "pa_context_new() failed."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto quit; + } + + pa_context_set_state_callback(context, rt_pa_context_state_callback, NULL); + + if (pa_context_connect(context, server, PA_CONTEXT_NOFLAGS, NULL) < 0) { + errorStream_ << "RtApiPulse::DeviceInfo pa_context_connect() failed: " + << pa_strerror(pa_context_errno(context)); + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto quit; + } + + if (pa_mainloop_run(m, &ret) < 0) { + errorStream_ << "pa_mainloop_run() failed."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto quit; + } + +quit: + if (context) + pa_context_unref(context); + + if (m) { + pa_mainloop_free(m); + } + + pa_xfree(server); +} + +unsigned int RtApiPulse::getDeviceCount( void ) +{ + collectDeviceInfo(); + return rt_pa_info.dev.size(); +} + +RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int device ) +{ + if (rt_pa_info.dev.size()==0) + collectDeviceInfo(); + if (device < rt_pa_info.dev.size()) + return rt_pa_info.dev[device].info; + return RtAudio::DeviceInfo(); +} + +static void *pulseaudio_callback( void * user ) +{ + CallbackInfo *cbi = static_cast( user ); + RtApiPulse *context = static_cast( cbi->object ); + volatile bool *isRunning = &cbi->isRunning; + +#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) + if (cbi->doRealtime) { + std::cerr << "RtAudio pulse: " << + (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << + "running realtime scheduling" << std::endl; + } +#endif + + while ( *isRunning ) { + pthread_testcancel(); + context->callbackEvent(); + } + + pthread_exit( NULL ); +} + +void RtApiPulse::closeStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + stream_.callbackInfo.isRunning = false; + if ( pah ) { + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + pah->runnable = true; + pthread_cond_signal( &pah->runnable_cv ); + } + MUTEX_UNLOCK( &stream_.mutex ); + + pthread_join( pah->thread, 0 ); + if ( pah->s_play ) { + pa_simple_flush( pah->s_play, NULL ); + pa_simple_free( pah->s_play ); + } + if ( pah->s_rec ) + pa_simple_free( pah->s_rec ); + + pthread_cond_destroy( &pah->runnable_cv ); + delete pah; + stream_.apiHandle = 0; + } + + if ( stream_.userBuffer[0] ) { + free( stream_.userBuffer[0] ); + stream_.userBuffer[0] = 0; + } + if ( stream_.userBuffer[1] ) { + free( stream_.userBuffer[1] ); + stream_.userBuffer[1] = 0; + } + + stream_.state = STREAM_CLOSED; + stream_.mode = UNINITIALIZED; +} + +void RtApiPulse::callbackEvent( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + while ( !pah->runnable ) + pthread_cond_wait( &pah->runnable_cv, &stream_.mutex ); + + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... " + "this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; + } + + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT], + stream_.bufferSize, streamTime, status, + stream_.callbackInfo.userData ); + + if ( doStopStream == 2 ) { + abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT]; + void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT]; + + if ( stream_.state != STREAM_RUNNING ) + goto unlock; + + int pa_error; + size_t bytes; + if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( stream_.doConvertBuffer[OUTPUT] ) { + convertBuffer( stream_.deviceBuffer, + stream_.userBuffer[OUTPUT], + stream_.convertInfo[OUTPUT] ); + bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize * + formatBytes( stream_.deviceFormat[OUTPUT] ); + } else + bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize * + formatBytes( stream_.userFormat ); + + if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::callbackEvent: audio write error, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX) { + if ( stream_.doConvertBuffer[INPUT] ) + bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize * + formatBytes( stream_.deviceFormat[INPUT] ); + else + bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize * + formatBytes( stream_.userFormat ); + + if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::callbackEvent: audio read error, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + if ( stream_.doConvertBuffer[INPUT] ) { + convertBuffer( stream_.userBuffer[INPUT], + stream_.deviceBuffer, + stream_.convertInfo[INPUT] ); + } + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + RtApi::tickStreamTime(); + + if ( doStopStream == 1 ) + stopStream(); +} + +void RtApiPulse::startStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::startStream(): the stream is not open!"; + error( RtAudioError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiPulse::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + + stream_.state = STREAM_RUNNING; + + pah->runnable = true; + pthread_cond_signal( &pah->runnable_cv ); + MUTEX_UNLOCK( &stream_.mutex ); +} + +void RtApiPulse::stopStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::stopStream(): the stream is not open!"; + error( RtAudioError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + if ( pah ) { + pah->runnable = false; + if ( pah->s_play ) { + int pa_error; + if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::stopStream: error draining output device, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + } + + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); +} + +void RtApiPulse::abortStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::abortStream(): the stream is not open!"; + error( RtAudioError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + if ( pah ) { + pah->runnable = false; + if ( pah->s_play ) { + int pa_error; + if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::abortStream: error flushing output device, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + } + + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); +} + +bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, + unsigned int channels, unsigned int firstChannel, + unsigned int sampleRate, RtAudioFormat format, + unsigned int *bufferSize, RtAudio::StreamOptions *options ) +{ + PulseAudioHandle *pah = 0; + unsigned long bufferBytes = 0; + pa_sample_spec ss; + + if ( device >= rt_pa_info.dev.size() ) return false; + if ( firstChannel != 0 ) { + errorText_ = "PulseAudio does not support channel offset mapping."; + return false; + } + + /* these may be NULL for default, but we've already got the names */ + const char *dev_input = NULL; + const char *dev_output = NULL; + if (!rt_pa_info.dev[device].source_name.empty()) + dev_input = rt_pa_info.dev[device].source_name.c_str(); + if (!rt_pa_info.dev[device].sink_name.empty()) + dev_output = rt_pa_info.dev[device].sink_name.c_str(); + + if (mode==INPUT && rt_pa_info.dev[device].info.inputChannels == 0) { + errorText_ = "PulseAudio device does not support input."; + return false; + } + if (mode==OUTPUT && rt_pa_info.dev[device].info.outputChannels == 0) { + errorText_ = "PulseAudio device does not support output."; + return false; + } + if (mode==DUPLEX && rt_pa_info.dev[device].info.duplexChannels == 0) { + /* Note: will always error, DUPLEX not yet supported */ + errorText_ = "PulseAudio device does not support duplex."; + return false; + } + + if (mode==INPUT && rt_pa_info.dev[device].info.inputChannels < channels) { + errorText_ = "PulseAudio: unsupported number of input channels."; + return false; + } + + if (mode==OUTPUT && rt_pa_info.dev[device].info.outputChannels < channels) { + errorText_ = "PulseAudio: unsupported number of output channels."; + return false; + } + + if (mode==DUPLEX && rt_pa_info.dev[device].info.duplexChannels < channels) { + /* Note: will always error, DUPLEX not yet supported */ + errorText_ = "PulseAudio: unsupported number of duplex channels."; + return false; + } + + ss.channels = channels; + + bool sr_found = false; + for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) { + if ( sampleRate == *sr ) { + sr_found = true; + stream_.sampleRate = sampleRate; + ss.rate = sampleRate; + break; + } + } + if ( !sr_found ) { + errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate."; + return false; + } + + bool sf_found = 0; + for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats; + sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) { + if ( format == sf->rtaudio_format ) { + sf_found = true; + stream_.userFormat = sf->rtaudio_format; + stream_.deviceFormat[mode] = stream_.userFormat; + ss.format = sf->pa_format; + break; + } + } + if ( !sf_found ) { // Use internal data format conversion. + stream_.userFormat = format; + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + ss.format = PA_SAMPLE_FLOAT32LE; + } + + // Set other stream parameters. + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + stream_.nBuffers = options ? options->numberOfBuffers : 1; + stream_.doByteSwap[mode] = false; + stream_.nUserChannels[mode] = channels; + stream_.nDeviceChannels[mode] = channels + firstChannel; + stream_.channelOffset[mode] = 0; + std::string streamName = "RtAudio"; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] ) + stream_.doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers. + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + stream_.bufferSize = *bufferSize; + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.device[mode] = device; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + if ( !stream_.apiHandle ) { + PulseAudioHandle *pah = new PulseAudioHandle; + if ( !pah ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle."; + goto error; + } + + stream_.apiHandle = pah; + if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable."; + goto error; + } + } + pah = static_cast( stream_.apiHandle ); + + int error; + if ( options && !options->streamName.empty() ) streamName = options->streamName; + switch ( mode ) { + pa_buffer_attr buffer_attr; + case INPUT: + buffer_attr.fragsize = bufferBytes; + buffer_attr.maxlength = -1; + + pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, + dev_input, "Record", &ss, NULL, &buffer_attr, &error ); + if ( !pah->s_rec ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server."; + goto error; + } + break; + case OUTPUT: { + pa_buffer_attr * attr_ptr; + + if ( options && options->numberOfBuffers > 0 ) { + // pa_buffer_attr::fragsize is recording-only. + // Hopefully PortAudio won't access uninitialized fields. + buffer_attr.maxlength = bufferBytes * options->numberOfBuffers; + buffer_attr.minreq = -1; + buffer_attr.prebuf = -1; + buffer_attr.tlength = -1; + attr_ptr = &buffer_attr; + } else { + attr_ptr = nullptr; + } + + pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, + dev_output, "Playback", &ss, NULL, attr_ptr, &error ); + if ( !pah->s_play ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server."; + goto error; + } + break; + } + case DUPLEX: + /* Note: We could add DUPLEX by synchronizing multiple streams, + but it would mean moving from Simple API to Asynchronous API: + https://freedesktop.org/software/pulseaudio/doxygen/streams.html#sync_streams */ + errorText_ = "RtApiPulse::probeDeviceOpen: duplex not supported for PulseAudio."; + goto error; + default: + goto error; + } + + if ( stream_.mode == UNINITIALIZED ) + stream_.mode = mode; + else if ( stream_.mode == mode ) + goto error; + else + stream_.mode = DUPLEX; + + if ( !stream_.callbackInfo.isRunning ) { + stream_.callbackInfo.object = this; + + stream_.state = STREAM_STOPPED; + // Set the thread attributes for joinable and realtime scheduling + // priority (optional). The higher priority will only take affect + // if the program is run as root or suid. Note, under Linux + // processes with CAP_SYS_NICE privilege, a user can change + // scheduling policy and priority (thus need not be root). See + // POSIX "capabilities". + pthread_attr_t attr; + pthread_attr_init( &attr ); + pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); +#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + stream_.callbackInfo.doRealtime = true; + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + + // Set the policy BEFORE the priority. Otherwise it fails. + pthread_attr_setschedpolicy(&attr, SCHED_RR); + pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM); + // This is definitely required. Otherwise it fails. + pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED); + pthread_attr_setschedparam(&attr, ¶m); + } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#endif + + stream_.callbackInfo.isRunning = true; + int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo); + pthread_attr_destroy(&attr); + if(result != 0) { + // Failed. Try instead with default attributes. + result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo); + if(result != 0) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread."; + goto error; + } + } + } + + return SUCCESS; + + error: + if ( pah && stream_.callbackInfo.isRunning ) { + pthread_cond_destroy( &pah->runnable_cv ); + delete pah; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +//******************** End of __LINUX_PULSE__ *********************// +#endif + +#if defined(__LINUX_OSS__) + +#include +#include +#include +#include +#include +#include +#include + +static void *ossCallbackHandler(void * ptr); + +// A structure to hold various information related to the OSS API +// implementation. +struct OssHandle { + int id[2]; // device ids + bool xrun[2]; + bool triggered; + pthread_cond_t runnable; + + OssHandle() + :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +RtApiOss :: RtApiOss() +{ + // Nothing to do here. +} + +RtApiOss :: ~RtApiOss() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiOss :: getDeviceCount( void ) +{ + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; + error( RtAudioError::WARNING ); + return 0; + } + + oss_sysinfo sysinfo; + if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtAudioError::WARNING ); + return 0; + } + + close( mixerfd ); + return sysinfo.numaudios; +} + +RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; + error( RtAudioError::WARNING ); + return info; + } + + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtAudioError::WARNING ); + return info; + } + + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + oss_audioinfo ainfo; + ainfo.dev = device; + result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); + close( mixerfd ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Probe channels + if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels; + if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels; + if ( ainfo.caps & PCM_CAP_DUPLEX ) { + if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + } + + // Probe data formats ... do for input + unsigned long mask = ainfo.iformats; + if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE ) + info.nativeFormats |= RTAUDIO_SINT16; + if ( mask & AFMT_S8 ) + info.nativeFormats |= RTAUDIO_SINT8; + if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE ) + info.nativeFormats |= RTAUDIO_SINT32; +#ifdef AFMT_FLOAT + if ( mask & AFMT_FLOAT ) + info.nativeFormats |= RTAUDIO_FLOAT32; +#endif + if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE ) + info.nativeFormats |= RTAUDIO_SINT24; + + // Check that we have at least one supported format + if ( info.nativeFormats == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Probe the supported sample rates. + info.sampleRates.clear(); + if ( ainfo.nrates ) { + for ( unsigned int i=0; i info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + + break; + } + } + } + } + else { + // Check min and max rate values; + for ( unsigned int k=0; k= (int) SAMPLE_RATES[k] ) { + info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + } + } + } + + if ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + else { + info.probed = true; + info.name = ainfo.name; + } + + return info; +} + + +bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; + return FAILURE; + } + + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; + return FAILURE; + } + + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + oss_audioinfo ainfo; + ainfo.dev = device; + result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); + close( mixerfd ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check if device supports input or output + if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || + ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { + if ( mode == OUTPUT ) + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + int flags = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( mode == OUTPUT ) + flags |= O_WRONLY; + else { // mode == INPUT + if (stream_.mode == OUTPUT && stream_.device[0] == device) { + // We just set the same device for playback ... close and reopen for duplex (OSS only). + close( handle->id[0] ); + handle->id[0] = 0; + if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; + errorText_ = errorStream_.str(); + return FAILURE; + } + // Check that the number previously set channels is the same. + if ( stream_.nUserChannels[0] != channels ) { + errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + flags |= O_RDWR; + } + else + flags |= O_RDONLY; + } + + // Set exclusive access if specified. + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; + + // Try to open the device. + int fd; + fd = open( ainfo.devnode, flags, 0 ); + if ( fd == -1 ) { + if ( errno == EBUSY ) + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // For duplex operation, specifically set this mode (this doesn't seem to work). + /* + if ( flags | O_RDWR ) { + result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); + if ( result == -1) { + errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + */ + + // Check the device channel support. + stream_.nUserChannels[mode] = channels; + if ( ainfo.max_channels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the number of channels. + int deviceChannels = channels + firstChannel; + result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); + if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nDeviceChannels[mode] = deviceChannels; + + // Get the data format mask + int mask; + result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine how to set the device format. + stream_.userFormat = format; + int deviceFormat = -1; + stream_.doByteSwap[mode] = false; + if ( format == RTAUDIO_SINT8 ) { + if ( mask & AFMT_S8 ) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + } + else if ( format == RTAUDIO_SINT16 ) { + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; + } + } + else if ( format == RTAUDIO_SINT24 ) { + if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; + } + } + else if ( format == RTAUDIO_SINT32 ) { + if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + } + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; + } + } + + if ( deviceFormat == -1 ) { + // The user requested format is not natively supported by the device. + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + } + else if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S8) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + } + + if ( stream_.deviceFormat[mode] == 0 ) { + // This really shouldn't happen ... + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the data format. + int temp = deviceFormat; + result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); + if ( result == -1 || deviceFormat != temp ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Attempt to set the buffer size. According to OSS, the minimum + // number of buffers is two. The supposed minimum buffer size is 16 + // bytes, so that will be our lower bound. The argument to this + // call is in the form 0xMMMMSSSS (hex), where the buffer size (in + // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. + // We'll check the actual value used near the end of the setup + // procedure. + int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; + if ( ossBufferBytes < 16 ) ossBufferBytes = 16; + int buffers = 0; + if ( options ) buffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; + if ( buffers < 2 ) buffers = 3; + temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); + result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nBuffers = buffers; + + // Save buffer size (in sample frames). + *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); + stream_.bufferSize = *bufferSize; + + // Set the sample rate. + int srate = sampleRate; + result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Verify the sample rate setup worked. + if ( abs( srate - (int)sampleRate ) > 100 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = sampleRate; + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { + // We're doing duplex setup here. + stream_.deviceFormat[0] = stream_.deviceFormat[1]; + stream_.nDeviceChannels[0] = deviceChannels; + } + + // Set interleaving parameters. + stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) + stream_.userInterleaved = false; + + // Set flags for buffer conversion + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate the stream handles if necessary and then save. + if ( stream_.apiHandle == 0 ) { + try { + handle = new OssHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; + goto error; + } + + if ( pthread_cond_init( &handle->runnable, NULL ) ) { + errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + + stream_.apiHandle = (void *) handle; + } + else { + handle = (OssHandle *) stream_.apiHandle; + } + handle->id[mode] = fd; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + // Setup thread if necessary. + if ( stream_.mode == OUTPUT && mode == INPUT ) { + // We had already set up an output stream. + stream_.mode = DUPLEX; + if ( stream_.device[0] == device ) handle->id[0] = fd; + } + else { + stream_.mode = mode; + + // Setup callback thread. + stream_.callbackInfo.object = (void *) this; + + // Set the thread attributes for joinable and realtime scheduling + // priority. The higher priority will only take affect if the + // program is run as root or suid. + pthread_attr_t attr; + pthread_attr_init( &attr ); + pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); +#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + stream_.callbackInfo.doRealtime = true; + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + + // Set the policy BEFORE the priority. Otherwise it fails. + pthread_attr_setschedpolicy(&attr, SCHED_RR); + pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM); + // This is definitely required. Otherwise it fails. + pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED); + pthread_attr_setschedparam(&attr, ¶m); + } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#endif + + stream_.callbackInfo.isRunning = true; + result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo ); + pthread_attr_destroy( &attr ); + if ( result ) { + // Failed. Try instead with default attributes. + result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiOss::error creating callback thread!"; + goto error; + } + } + } + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &handle->runnable ); + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiOss :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + OssHandle *handle = (OssHandle *) stream_.apiHandle; + stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) + pthread_cond_signal( &handle->runnable ); + MUTEX_UNLOCK( &stream_.mutex ); + pthread_join( stream_.callbackInfo.thread, NULL ); + + if ( stream_.state == STREAM_RUNNING ) { + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + else + ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + stream_.state = STREAM_STOPPED; + } + + if ( handle ) { + pthread_cond_destroy( &handle->runnable ); + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiOss :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiOss::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + + stream_.state = STREAM_RUNNING; + + // No need to do anything else here ... OSS automatically starts + // when fed samples. + + MUTEX_UNLOCK( &stream_.mutex ); + + OssHandle *handle = (OssHandle *) stream_.apiHandle; + pthread_cond_signal( &handle->runnable ); +} + +void RtApiOss :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // Flush the output with zeros a few times. + char *buffer; + int samples; + RtAudioFormat format; + + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + samples = stream_.bufferSize * stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + samples = stream_.bufferSize * stream_.nUserChannels[0]; + format = stream_.userFormat; + } + + memset( buffer, 0, samples * formatBytes(format) ); + for ( unsigned int i=0; iid[0], buffer, samples * formatBytes(format) ); + if ( result == -1 ) { + errorText_ = "RtApiOss::stopStream: audio write error."; + error( RtAudioError::WARNING ); + } + } + + result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + handle->triggered = false; + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { + result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result != -1 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiOss :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + handle->triggered = false; + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { + result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result != -1 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiOss :: callbackEvent() +{ + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + pthread_cond_wait( &handle->runnable, &stream_.mutex ); + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; + } + + // Invoke user callback to get fresh output data. + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + if ( doStopStream == 2 ) { + this->abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; + + int result; + char *buffer; + int samples; + RtAudioFormat format; + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + samples = stream_.bufferSize * stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + samples = stream_.bufferSize * stream_.nUserChannels[0]; + format = stream_.userFormat; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( buffer, samples, format ); + + if ( stream_.mode == DUPLEX && handle->triggered == false ) { + int trig = 0; + ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); + result = write( handle->id[0], buffer, samples * formatBytes(format) ); + trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT; + ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); + handle->triggered = true; + } + else + // Write samples to device. + result = write( handle->id[0], buffer, samples * formatBytes(format) ); + + if ( result == -1 ) { + // We'll assume this is an underrun, though there isn't a + // specific means for determining that. + handle->xrun[0] = true; + errorText_ = "RtApiOss::callbackEvent: audio write error."; + error( RtAudioError::WARNING ); + // Continue on to input section. + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + samples = stream_.bufferSize * stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; + } + else { + buffer = stream_.userBuffer[1]; + samples = stream_.bufferSize * stream_.nUserChannels[1]; + format = stream_.userFormat; + } + + // Read samples from device. + result = read( handle->id[1], buffer, samples * formatBytes(format) ); + + if ( result == -1 ) { + // We'll assume this is an overrun, though there isn't a + // specific means for determining that. + handle->xrun[1] = true; + errorText_ = "RtApiOss::callbackEvent: audio read error."; + error( RtAudioError::WARNING ); + goto unlock; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, samples, format ); + + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} + +static void *ossCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiOss *object = (RtApiOss *) info->object; + bool *isRunning = &info->isRunning; + +#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) + if (info->doRealtime) { + std::cerr << "RtAudio oss: " << + (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << + "running realtime scheduling" << std::endl; + } +#endif + + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); + } + + pthread_exit( NULL ); +} + +//******************** End of __LINUX_OSS__ *********************// +#endif + + +// *************************************************** // +// +// Protected common (OS-independent) RtAudio methods. +// +// *************************************************** // + +// This method can be modified to control the behavior of error +// message printing. +void RtApi :: error( RtAudioError::Type type ) +{ + errorStream_.str(""); // clear the ostringstream + + RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback; + if ( errorCallback ) { + // abortStream() can generate new error messages. Ignore them. Just keep original one. + + if ( firstErrorOccurred_ ) + return; + + firstErrorOccurred_ = true; + const std::string errorMessage = errorText_; + + if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) { + stream_.callbackInfo.isRunning = false; // exit from the thread + abortStream(); + } + + errorCallback( type, errorMessage ); + firstErrorOccurred_ = false; + return; + } + + if ( type == RtAudioError::WARNING && showWarnings_ == true ) + std::cerr << '\n' << errorText_ << "\n\n"; + else if ( type != RtAudioError::WARNING ) + throw( RtAudioError( errorText_, type ) ); +} + +void RtApi :: verifyStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApi:: a stream is not open!"; + error( RtAudioError::INVALID_USE ); + } +} + +void RtApi :: clearStreamInfo() +{ + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; + stream_.sampleRate = 0; + stream_.bufferSize = 0; + stream_.nBuffers = 0; + stream_.userFormat = 0; + stream_.userInterleaved = true; + stream_.streamTime = 0.0; + stream_.apiHandle = 0; + stream_.deviceBuffer = 0; + stream_.callbackInfo.callback = 0; + stream_.callbackInfo.userData = 0; + stream_.callbackInfo.isRunning = false; + stream_.callbackInfo.errorCallback = 0; + for ( int i=0; i<2; i++ ) { + stream_.device[i] = 11111; + stream_.doConvertBuffer[i] = false; + stream_.deviceInterleaved[i] = true; + stream_.doByteSwap[i] = false; + stream_.nUserChannels[i] = 0; + stream_.nDeviceChannels[i] = 0; + stream_.channelOffset[i] = 0; + stream_.deviceFormat[i] = 0; + stream_.latency[i] = 0; + stream_.userBuffer[i] = 0; + stream_.convertInfo[i].channels = 0; + stream_.convertInfo[i].inJump = 0; + stream_.convertInfo[i].outJump = 0; + stream_.convertInfo[i].inFormat = 0; + stream_.convertInfo[i].outFormat = 0; + stream_.convertInfo[i].inOffset.clear(); + stream_.convertInfo[i].outOffset.clear(); + } +} + +unsigned int RtApi :: formatBytes( RtAudioFormat format ) +{ + if ( format == RTAUDIO_SINT16 ) + return 2; + else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 ) + return 4; + else if ( format == RTAUDIO_FLOAT64 ) + return 8; + else if ( format == RTAUDIO_SINT24 ) + return 3; + else if ( format == RTAUDIO_SINT8 ) + return 1; + + errorText_ = "RtApi::formatBytes: undefined format."; + error( RtAudioError::WARNING ); + + return 0; +} + +void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel ) +{ + if ( mode == INPUT ) { // convert device to user buffer + stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1]; + stream_.convertInfo[mode].outJump = stream_.nUserChannels[1]; + stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1]; + stream_.convertInfo[mode].outFormat = stream_.userFormat; + } + else { // convert user to device buffer + stream_.convertInfo[mode].inJump = stream_.nUserChannels[0]; + stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0]; + stream_.convertInfo[mode].inFormat = stream_.userFormat; + stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0]; + } + + if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump ) + stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump; + else + stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump; + + // Set up the interleave/deinterleave offsets. + if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) { + if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) || + ( mode == INPUT && stream_.userInterleaved ) ) { + for ( int k=0; k 0 ) { + if ( stream_.deviceInterleaved[mode] ) { + if ( mode == OUTPUT ) { + for ( int k=0; k info.inJump ) + memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) ); + + int j; + if (info.outFormat == RTAUDIO_FLOAT64) { + Float64 *out = (Float64 *)outBuffer; + + if (info.inFormat == RTAUDIO_SINT8) { + signed char *in = (signed char *)inBuffer; + for (unsigned int i=0; i> 8); + //out[info.outOffset[j]] >>= 8; + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_FLOAT32) { + Float32 *in = (Float32 *)inBuffer; + for (unsigned int i=0; i> 8); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 16) & 0x0000ffff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_FLOAT32) { + Float32 *in = (Float32 *)inBuffer; + for (unsigned int i=0; i> 8) & 0x00ff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_SINT24) { + Int24 *in = (Int24 *)inBuffer; + for (unsigned int i=0; i> 16); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 24) & 0x000000ff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_FLOAT32) { + Float32 *in = (Float32 *)inBuffer; + for (unsigned int i=0; i>8) | (x<<8); } +//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } +//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } + +void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) +{ + char val; + char *ptr; + + ptr = buffer; + if ( format == RTAUDIO_SINT16 ) { + for ( unsigned int i=0; i= 4 + #define RTAUDIO_DLL_PUBLIC __attribute__( (visibility( "default" )) ) + #else + #define RTAUDIO_DLL_PUBLIC + #endif +#endif + +#include +#include +#include +#include + +/*! \typedef typedef unsigned long RtAudioFormat; + \brief RtAudio data format type. + + Support for signed integers and floats. Audio data fed to/from an + RtAudio stream is assumed to ALWAYS be in host byte order. The + internal routines will automatically take care of any necessary + byte-swapping between the host format and the soundcard. Thus, + endian-ness is not a concern in the following format definitions. + + - \e RTAUDIO_SINT8: 8-bit signed integer. + - \e RTAUDIO_SINT16: 16-bit signed integer. + - \e RTAUDIO_SINT24: 24-bit signed integer. + - \e RTAUDIO_SINT32: 32-bit signed integer. + - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0. + - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0. +*/ +typedef unsigned long RtAudioFormat; +static const RtAudioFormat RTAUDIO_SINT8 = 0x1; // 8-bit signed integer. +static const RtAudioFormat RTAUDIO_SINT16 = 0x2; // 16-bit signed integer. +static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // 24-bit signed integer. +static const RtAudioFormat RTAUDIO_SINT32 = 0x8; // 32-bit signed integer. +static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0. +static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0. + +/*! \typedef typedef unsigned long RtAudioStreamFlags; + \brief RtAudio stream option flags. + + The following flags can be OR'ed together to allow a client to + make changes to the default stream behavior: + + - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). + - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. + - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. + - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only). + - \e RTAUDIO_JACK_DONT_CONNECT: Do not automatically connect ports (JACK only). + + By default, RtAudio streams pass and receive audio data from the + client in an interleaved format. By passing the + RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio + data will instead be presented in non-interleaved buffers. In + this case, each buffer argument in the RtAudioCallback function + will point to a single array of data, with \c nFrames samples for + each channel concatenated back-to-back. For example, the first + sample of data for the second channel would be located at index \c + nFrames (assuming the \c buffer pointer was recast to the correct + data type for the stream). + + Certain audio APIs offer a number of parameters that influence the + I/O latency of a stream. By default, RtAudio will attempt to set + these parameters internally for robust (glitch-free) performance + (though some APIs, like Windows DirectSound, make this difficult). + By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() + function, internal stream settings will be influenced in an attempt + to minimize stream latency, though possibly at the expense of stream + performance. + + If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to + open the input and/or output stream device(s) for exclusive use. + Note that this is not possible with all supported audio APIs. + + If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt + to select realtime scheduling (round-robin) for the callback thread. + + If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to + open the "default" PCM device when using the ALSA API. Note that this + will override any specified input or output device id. + + If the RTAUDIO_JACK_DONT_CONNECT flag is set, RtAudio will not attempt + to automatically connect the ports of the client to the audio device. +*/ +typedef unsigned int RtAudioStreamFlags; +static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved). +static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency. +static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others. +static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread. +static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only). +static const RtAudioStreamFlags RTAUDIO_JACK_DONT_CONNECT = 0x20; // Do not automatically connect ports (JACK only). + +/*! \typedef typedef unsigned long RtAudioStreamStatus; + \brief RtAudio stream status (over- or underflow) flags. + + Notification of a stream over- or underflow is indicated by a + non-zero stream \c status argument in the RtAudioCallback function. + The stream status can be one of the following two options, + depending on whether the stream is open for output and/or input: + + - \e RTAUDIO_INPUT_OVERFLOW: Input data was discarded because of an overflow condition at the driver. + - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound. +*/ +typedef unsigned int RtAudioStreamStatus; +static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1; // Input data was discarded because of an overflow condition at the driver. +static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2; // The output buffer ran low, likely causing a gap in the output sound. + +//! RtAudio callback function prototype. +/*! + All RtAudio clients must create a function of type RtAudioCallback + to read and/or write data from/to the audio stream. When the + underlying audio system is ready for new input or output data, this + function will be invoked. + + \param outputBuffer For output (or duplex) streams, the client + should write \c nFrames of audio sample frames into this + buffer. This argument should be recast to the datatype + specified when the stream was opened. For input-only + streams, this argument will be NULL. + + \param inputBuffer For input (or duplex) streams, this buffer will + hold \c nFrames of input audio sample frames. This + argument should be recast to the datatype specified when the + stream was opened. For output-only streams, this argument + will be NULL. + + \param nFrames The number of sample frames of input or output + data in the buffers. The actual buffer size in bytes is + dependent on the data type and number of channels in use. + + \param streamTime The number of seconds that have elapsed since the + stream was started. + + \param status If non-zero, this argument indicates a data overflow + or underflow condition for the stream. The particular + condition can be determined by comparison with the + RtAudioStreamStatus flags. + + \param userData A pointer to optional data provided by the client + when opening the stream (default = NULL). + + \return + To continue normal stream operation, the RtAudioCallback function + should return a value of zero. To stop the stream and drain the + output buffer, the function should return a value of one. To abort + the stream immediately, the client should return a value of two. + */ +typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer, + unsigned int nFrames, + double streamTime, + RtAudioStreamStatus status, + void *userData ); + +/************************************************************************/ +/*! \class RtAudioError + \brief Exception handling class for RtAudio. + + The RtAudioError class is quite simple but it does allow errors to be + "caught" by RtAudioError::Type. See the RtAudio documentation to know + which methods can throw an RtAudioError. +*/ +/************************************************************************/ + +class RTAUDIO_DLL_PUBLIC RtAudioError : public std::runtime_error +{ + public: + //! Defined RtAudioError types. + enum Type { + WARNING, /*!< A non-critical error. */ + DEBUG_WARNING, /*!< A non-critical error which might be useful for debugging. */ + UNSPECIFIED, /*!< The default, unspecified error type. */ + NO_DEVICES_FOUND, /*!< No devices found on system. */ + INVALID_DEVICE, /*!< An invalid device ID was specified. */ + MEMORY_ERROR, /*!< An error occured during memory allocation. */ + INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */ + INVALID_USE, /*!< The function was called incorrectly. */ + DRIVER_ERROR, /*!< A system driver error occured. */ + SYSTEM_ERROR, /*!< A system error occured. */ + THREAD_ERROR /*!< A thread error occured. */ + }; + + //! The constructor. + RtAudioError( const std::string& message, + Type type = RtAudioError::UNSPECIFIED ) + : std::runtime_error(message), type_(type) {} + + //! Prints thrown error message to stderr. + virtual void printMessage( void ) const + { std::cerr << '\n' << what() << "\n\n"; } + + //! Returns the thrown error message type. + virtual const Type& getType(void) const { return type_; } + + //! Returns the thrown error message string. + virtual const std::string getMessage(void) const + { return std::string(what()); } + + protected: + Type type_; +}; + +//! RtAudio error callback function prototype. +/*! + \param type Type of error. + \param errorText Error description. + */ +typedef void (*RtAudioErrorCallback)( RtAudioError::Type type, const std::string &errorText ); + +// **************************************************************** // +// +// RtAudio class declaration. +// +// RtAudio is a "controller" used to select an available audio i/o +// interface. It presents a common API for the user to call but all +// functionality is implemented by the class RtApi and its +// subclasses. RtAudio creates an instance of an RtApi subclass +// based on the user's API choice. If no choice is made, RtAudio +// attempts to make a "logical" API selection. +// +// **************************************************************** // + +class RtApi; + +class RTAUDIO_DLL_PUBLIC RtAudio +{ + public: + + //! Audio API specifier arguments. + enum Api { + UNSPECIFIED, /*!< Search for a working compiled API. */ + LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */ + LINUX_PULSE, /*!< The Linux PulseAudio API. */ + LINUX_OSS, /*!< The Linux Open Sound System API. */ + UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */ + MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */ + WINDOWS_WASAPI, /*!< The Microsoft WASAPI API. */ + WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */ + WINDOWS_DS, /*!< The Microsoft DirectSound API. */ + RTAUDIO_DUMMY, /*!< A compilable but non-functional API. */ + NUM_APIS /*!< Number of values in this enum. */ + }; + + //! The public device information structure for returning queried values. + struct DeviceInfo { + bool probed; /*!< true if the device capabilities were successfully probed. */ + std::string name; /*!< Character string device identifier. */ + unsigned int outputChannels{}; /*!< Maximum output channels supported by device. */ + unsigned int inputChannels{}; /*!< Maximum input channels supported by device. */ + unsigned int duplexChannels{}; /*!< Maximum simultaneous input/output channels supported by device. */ + bool isDefaultOutput{false}; /*!< true if this is the default output device. */ + bool isDefaultInput{false}; /*!< true if this is the default input device. */ + std::vector sampleRates; /*!< Supported sample rates (queried from list of standard rates). */ + unsigned int preferredSampleRate{}; /*!< Preferred sample rate, e.g. for WASAPI the system sample rate. */ + RtAudioFormat nativeFormats{}; /*!< Bit mask of supported data formats. */ + }; + + //! The structure for specifying input or ouput stream parameters. + struct StreamParameters { + unsigned int deviceId{}; /*!< Device index (0 to getDeviceCount() - 1). */ + unsigned int nChannels{}; /*!< Number of channels. */ + unsigned int firstChannel{}; /*!< First channel index on device (default = 0). */ + }; + + //! The structure for specifying stream options. + /*! + The following flags can be OR'ed together to allow a client to + make changes to the default stream behavior: + + - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). + - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. + - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. + - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread. + - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only). + + By default, RtAudio streams pass and receive audio data from the + client in an interleaved format. By passing the + RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio + data will instead be presented in non-interleaved buffers. In + this case, each buffer argument in the RtAudioCallback function + will point to a single array of data, with \c nFrames samples for + each channel concatenated back-to-back. For example, the first + sample of data for the second channel would be located at index \c + nFrames (assuming the \c buffer pointer was recast to the correct + data type for the stream). + + Certain audio APIs offer a number of parameters that influence the + I/O latency of a stream. By default, RtAudio will attempt to set + these parameters internally for robust (glitch-free) performance + (though some APIs, like Windows DirectSound, make this difficult). + By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() + function, internal stream settings will be influenced in an attempt + to minimize stream latency, though possibly at the expense of stream + performance. + + If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to + open the input and/or output stream device(s) for exclusive use. + Note that this is not possible with all supported audio APIs. + + If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt + to select realtime scheduling (round-robin) for the callback thread. + The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME + flag is set. It defines the thread's realtime priority. + + If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to + open the "default" PCM device when using the ALSA API. Note that this + will override any specified input or output device id. + + The \c numberOfBuffers parameter can be used to control stream + latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs + only. A value of two is usually the smallest allowed. Larger + numbers can potentially result in more robust stream performance, + though likely at the cost of stream latency. The value set by the + user is replaced during execution of the RtAudio::openStream() + function by the value actually used by the system. + + The \c streamName parameter can be used to set the client name + when using the Jack API. By default, the client name is set to + RtApiJack. However, if you wish to create multiple instances of + RtAudio with Jack, each instance must have a unique client name. + */ + struct StreamOptions { + RtAudioStreamFlags flags{}; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */ + unsigned int numberOfBuffers{}; /*!< Number of stream buffers. */ + std::string streamName; /*!< A stream name (currently used only in Jack). */ + int priority{}; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */ + }; + + //! A static function to determine the current RtAudio version. + static std::string getVersion( void ); + + //! A static function to determine the available compiled audio APIs. + /*! + The values returned in the std::vector can be compared against + the enumerated list values. Note that there can be more than one + API compiled for certain operating systems. + */ + static void getCompiledApi( std::vector &apis ); + + //! Return the name of a specified compiled audio API. + /*! + This obtains a short lower-case name used for identification purposes. + This value is guaranteed to remain identical across library versions. + If the API is unknown, this function will return the empty string. + */ + static std::string getApiName( RtAudio::Api api ); + + //! Return the display name of a specified compiled audio API. + /*! + This obtains a long name used for display purposes. + If the API is unknown, this function will return the empty string. + */ + static std::string getApiDisplayName( RtAudio::Api api ); + + //! Return the compiled audio API having the given name. + /*! + A case insensitive comparison will check the specified name + against the list of compiled APIs, and return the one which + matches. On failure, the function returns UNSPECIFIED. + */ + static RtAudio::Api getCompiledApiByName( const std::string &name ); + + //! The class constructor. + /*! + The constructor performs minor initialization tasks. An exception + can be thrown if no API support is compiled. + + If no API argument is specified and multiple API support has been + compiled, the default order of use is JACK, ALSA, OSS (Linux + systems) and ASIO, DS (Windows systems). + */ + RtAudio( RtAudio::Api api=UNSPECIFIED ); + + //! The destructor. + /*! + If a stream is running or open, it will be stopped and closed + automatically. + */ + ~RtAudio(); + + //! Returns the audio API specifier for the current instance of RtAudio. + RtAudio::Api getCurrentApi( void ); + + //! A public function that queries for the number of audio devices available. + /*! + This function performs a system query of available devices each time it + is called, thus supporting devices connected \e after instantiation. If + a system error occurs during processing, a warning will be issued. + */ + unsigned int getDeviceCount( void ); + + //! Return an RtAudio::DeviceInfo structure for a specified device number. + /*! + + Any device integer between 0 and getDeviceCount() - 1 is valid. + If an invalid argument is provided, an RtAudioError (type = INVALID_USE) + will be thrown. If a device is busy or otherwise unavailable, the + structure member "probed" will have a value of "false" and all + other members are undefined. If the specified device is the + current default input or output device, the corresponding + "isDefault" member will have a value of "true". + */ + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + + //! A function that returns the index of the default output device. + /*! + If the underlying audio API does not provide a "default + device", or if no devices are available, the return value will be + 0. Note that this is a valid device identifier and it is the + client's responsibility to verify that a device is available + before attempting to open a stream. + */ + unsigned int getDefaultOutputDevice( void ); + + //! A function that returns the index of the default input device. + /*! + If the underlying audio API does not provide a "default + device", or if no devices are available, the return value will be + 0. Note that this is a valid device identifier and it is the + client's responsibility to verify that a device is available + before attempting to open a stream. + */ + unsigned int getDefaultInputDevice( void ); + + //! A public function for opening a stream with the specified parameters. + /*! + An RtAudioError (type = SYSTEM_ERROR) is thrown if a stream cannot be + opened with the specified parameters or an error occurs during + processing. An RtAudioError (type = INVALID_USE) is thrown if any + invalid device ID or channel number parameters are specified. + + \param outputParameters Specifies output stream parameters to use + when opening a stream, including a device ID, number of channels, + and starting channel number. For input-only streams, this + argument should be NULL. The device ID is an index value between + 0 and getDeviceCount() - 1. + \param inputParameters Specifies input stream parameters to use + when opening a stream, including a device ID, number of channels, + and starting channel number. For output-only streams, this + argument should be NULL. The device ID is an index value between + 0 and getDeviceCount() - 1. + \param format An RtAudioFormat specifying the desired sample data format. + \param sampleRate The desired sample rate (sample frames per second). + \param *bufferFrames A pointer to a value indicating the desired + internal buffer size in sample frames. The actual value + used by the device is returned via the same pointer. A + value of zero can be specified, in which case the lowest + allowable value is determined. + \param callback A client-defined function that will be invoked + when input data is available and/or output data is needed. + \param userData An optional pointer to data that can be accessed + from within the callback function. + \param options An optional pointer to a structure containing various + global stream options, including a list of OR'ed RtAudioStreamFlags + and a suggested number of stream buffers that can be used to + control stream latency. More buffers typically result in more + robust performance, though at a cost of greater latency. If a + value of zero is specified, a system-specific median value is + chosen. If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the + lowest allowable value is used. The actual value used is + returned via the structure argument. The parameter is API dependent. + \param errorCallback A client-defined function that will be invoked + when an error has occured. + */ + void openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, RtAudioCallback callback, + void *userData = NULL, RtAudio::StreamOptions *options = NULL, RtAudioErrorCallback errorCallback = NULL ); + + //! A function that closes a stream and frees any associated stream memory. + /*! + If a stream is not open, this function issues a warning and + returns (no exception is thrown). + */ + void closeStream( void ); + + //! A function that starts a stream. + /*! + An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtAudioError (type = INVALID_USE) is thrown if a + stream is not open. A warning is issued if the stream is already + running. + */ + void startStream( void ); + + //! Stop a stream, allowing any samples remaining in the output queue to be played. + /*! + An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtAudioError (type = INVALID_USE) is thrown if a + stream is not open. A warning is issued if the stream is already + stopped. + */ + void stopStream( void ); + + //! Stop a stream, discarding any samples remaining in the input/output queue. + /*! + An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtAudioError (type = INVALID_USE) is thrown if a + stream is not open. A warning is issued if the stream is already + stopped. + */ + void abortStream( void ); + + //! Returns true if a stream is open and false if not. + bool isStreamOpen( void ) const; + + //! Returns true if the stream is running and false if it is stopped or not open. + bool isStreamRunning( void ) const; + + //! Returns the number of elapsed seconds since the stream was started. + /*! + If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. + */ + double getStreamTime( void ); + + //! Set the stream time to a time in seconds greater than or equal to 0.0. + /*! + If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. + */ + void setStreamTime( double time ); + + //! Returns the internal stream latency in sample frames. + /*! + The stream latency refers to delay in audio input and/or output + caused by internal buffering by the audio system and/or hardware. + For duplex streams, the returned value will represent the sum of + the input and output latencies. If a stream is not open, an + RtAudioError (type = INVALID_USE) will be thrown. If the API does not + report latency, the return value will be zero. + */ + long getStreamLatency( void ); + + //! Returns actual sample rate in use by the stream. + /*! + On some systems, the sample rate used may be slightly different + than that specified in the stream parameters. If a stream is not + open, an RtAudioError (type = INVALID_USE) will be thrown. + */ + unsigned int getStreamSampleRate( void ); + + //! Specify whether warning messages should be printed to stderr. + void showWarnings( bool value = true ); + + protected: + + void openRtApi( RtAudio::Api api ); + RtApi *rtapi_; +}; + +// Operating system dependent thread functionality. +#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) + + #ifndef NOMINMAX + #define NOMINMAX + #endif + #include + #include + #include + + typedef uintptr_t ThreadHandle; + typedef CRITICAL_SECTION StreamMutex; + +#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) + // Using pthread library for various flavors of unix. + #include + + typedef pthread_t ThreadHandle; + typedef pthread_mutex_t StreamMutex; + +#else // Setup for "dummy" behavior + + #define __RTAUDIO_DUMMY__ + typedef int ThreadHandle; + typedef int StreamMutex; + +#endif + +// This global structure type is used to pass callback information +// between the private RtAudio stream structure and global callback +// handling functions. +struct CallbackInfo { + void *object{}; // Used as a "this" pointer. + ThreadHandle thread{}; + void *callback{}; + void *userData{}; + void *errorCallback{}; + void *apiInfo{}; // void pointer for API specific callback information + bool isRunning{false}; + bool doRealtime{false}; + int priority{}; +}; + +// **************************************************************** // +// +// RtApi class declaration. +// +// Subclasses of RtApi contain all API- and OS-specific code necessary +// to fully implement the RtAudio API. +// +// Note that RtApi is an abstract base class and cannot be +// explicitly instantiated. The class RtAudio will create an +// instance of an RtApi subclass (RtApiOss, RtApiAlsa, +// RtApiJack, RtApiCore, RtApiDs, or RtApiAsio). +// +// **************************************************************** // + +#pragma pack(push, 1) +class S24 { + + protected: + unsigned char c3[3]; + + public: + S24() {} + + S24& operator = ( const int& i ) { + c3[0] = (i & 0x000000ff); + c3[1] = (i & 0x0000ff00) >> 8; + c3[2] = (i & 0x00ff0000) >> 16; + return *this; + } + + S24( const double& d ) { *this = (int) d; } + S24( const float& f ) { *this = (int) f; } + S24( const signed short& s ) { *this = (int) s; } + S24( const char& c ) { *this = (int) c; } + + int asInt() { + int i = c3[0] | (c3[1] << 8) | (c3[2] << 16); + if (i & 0x800000) i |= ~0xffffff; + return i; + } +}; +#pragma pack(pop) + +#if defined( HAVE_GETTIMEOFDAY ) + #include +#endif + +#include + +class RTAUDIO_DLL_PUBLIC RtApi +{ +public: + + RtApi(); + virtual ~RtApi(); + virtual RtAudio::Api getCurrentApi( void ) = 0; + virtual unsigned int getDeviceCount( void ) = 0; + virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0; + virtual unsigned int getDefaultInputDevice( void ); + virtual unsigned int getDefaultOutputDevice( void ); + void openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, RtAudioCallback callback, + void *userData, RtAudio::StreamOptions *options, + RtAudioErrorCallback errorCallback ); + virtual void closeStream( void ); + virtual void startStream( void ) = 0; + virtual void stopStream( void ) = 0; + virtual void abortStream( void ) = 0; + long getStreamLatency( void ); + unsigned int getStreamSampleRate( void ); + virtual double getStreamTime( void ); + virtual void setStreamTime( double time ); + bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; } + bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; } + void showWarnings( bool value ) { showWarnings_ = value; } + + +protected: + + static const unsigned int MAX_SAMPLE_RATES; + static const unsigned int SAMPLE_RATES[]; + + enum { FAILURE, SUCCESS }; + + enum StreamState { + STREAM_STOPPED, + STREAM_STOPPING, + STREAM_RUNNING, + STREAM_CLOSED = -50 + }; + + enum StreamMode { + OUTPUT, + INPUT, + DUPLEX, + UNINITIALIZED = -75 + }; + + // A protected structure used for buffer conversion. + struct ConvertInfo { + int channels; + int inJump, outJump; + RtAudioFormat inFormat, outFormat; + std::vector inOffset; + std::vector outOffset; + }; + + // A protected structure for audio streams. + struct RtApiStream { + unsigned int device[2]; // Playback and record, respectively. + void *apiHandle; // void pointer for API specific stream handle information + StreamMode mode; // OUTPUT, INPUT, or DUPLEX. + StreamState state; // STOPPED, RUNNING, or CLOSED + char *userBuffer[2]; // Playback and record, respectively. + char *deviceBuffer; + bool doConvertBuffer[2]; // Playback and record, respectively. + bool userInterleaved; + bool deviceInterleaved[2]; // Playback and record, respectively. + bool doByteSwap[2]; // Playback and record, respectively. + unsigned int sampleRate; + unsigned int bufferSize; + unsigned int nBuffers; + unsigned int nUserChannels[2]; // Playback and record, respectively. + unsigned int nDeviceChannels[2]; // Playback and record channels, respectively. + unsigned int channelOffset[2]; // Playback and record, respectively. + unsigned long latency[2]; // Playback and record, respectively. + RtAudioFormat userFormat; + RtAudioFormat deviceFormat[2]; // Playback and record, respectively. + StreamMutex mutex; + CallbackInfo callbackInfo; + ConvertInfo convertInfo[2]; + double streamTime; // Number of elapsed seconds since the stream started. + +#if defined(HAVE_GETTIMEOFDAY) + struct timeval lastTickTimestamp; +#endif + + RtApiStream() + :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; } + }; + + typedef S24 Int24; + typedef signed short Int16; + typedef signed int Int32; + typedef float Float32; + typedef double Float64; + + std::ostringstream errorStream_; + std::string errorText_; + bool showWarnings_; + RtApiStream stream_; + bool firstErrorOccurred_; + + /*! + Protected, api-specific method that attempts to open a device + with the given parameters. This function MUST be implemented by + all subclasses. If an error is encountered during the probe, a + "warning" message is reported and FAILURE is returned. A + successful probe is indicated by a return value of SUCCESS. + */ + virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); + + //! A protected function used to increment the stream time. + void tickStreamTime( void ); + + //! Protected common method to clear an RtApiStream structure. + void clearStreamInfo(); + + /*! + Protected common method that throws an RtAudioError (type = + INVALID_USE) if a stream is not open. + */ + void verifyStream( void ); + + //! Protected common error method to allow global control over error handling. + void error( RtAudioError::Type type ); + + /*! + Protected method used to perform format, channel number, and/or interleaving + conversions between the user and device buffers. + */ + void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info ); + + //! Protected common method used to perform byte-swapping on buffers. + void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ); + + //! Protected common method that returns the number of bytes for a given format. + unsigned int formatBytes( RtAudioFormat format ); + + //! Protected common method that sets up the parameters for buffer conversion. + void setConvertInfo( StreamMode mode, unsigned int firstChannel ); +}; + +// **************************************************************** // +// +// Inline RtAudio definitions. +// +// **************************************************************** // + +inline RtAudio::Api RtAudio :: getCurrentApi( void ) { return rtapi_->getCurrentApi(); } +inline unsigned int RtAudio :: getDeviceCount( void ) { return rtapi_->getDeviceCount(); } +inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); } +inline unsigned int RtAudio :: getDefaultInputDevice( void ) { return rtapi_->getDefaultInputDevice(); } +inline unsigned int RtAudio :: getDefaultOutputDevice( void ) { return rtapi_->getDefaultOutputDevice(); } +inline void RtAudio :: closeStream( void ) { return rtapi_->closeStream(); } +inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); } +inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); } +inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); } +inline bool RtAudio :: isStreamOpen( void ) const { return rtapi_->isStreamOpen(); } +inline bool RtAudio :: isStreamRunning( void ) const { return rtapi_->isStreamRunning(); } +inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); } +inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); } +inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); } +inline void RtAudio :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); } +inline void RtAudio :: showWarnings( bool value ) { rtapi_->showWarnings( value ); } + +// RtApi Subclass prototypes. + +#if defined(__MACOSX_CORE__) + +#include + +class RtApiCore: public RtApi +{ +public: + + RtApiCore(); + ~RtApiCore(); + RtAudio::Api getCurrentApi( void ) override { return RtAudio::MACOSX_CORE; } + unsigned int getDeviceCount( void ) override; + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; + unsigned int getDefaultOutputDevice( void ) override; + unsigned int getDefaultInputDevice( void ) override; + void closeStream( void ) override; + void startStream( void ) override; + void stopStream( void ) override; + void abortStream( void ) override; + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + bool callbackEvent( AudioDeviceID deviceId, + const AudioBufferList *inBufferList, + const AudioBufferList *outBufferList ); + + private: + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) override; + static const char* getErrorCode( OSStatus code ); +}; + +#endif + +#if defined(__UNIX_JACK__) + +class RtApiJack: public RtApi +{ +public: + + RtApiJack(); + ~RtApiJack(); + RtAudio::Api getCurrentApi( void ) override { return RtAudio::UNIX_JACK; } + unsigned int getDeviceCount( void ) override; + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; + void closeStream( void ) override; + void startStream( void ) override; + void stopStream( void ) override; + void abortStream( void ) override; + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + bool callbackEvent( unsigned long nframes ); + + private: + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) override; + + bool shouldAutoconnect_; +}; + +#endif + +#if defined(__WINDOWS_ASIO__) + +class RtApiAsio: public RtApi +{ +public: + + RtApiAsio(); + ~RtApiAsio(); + RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_ASIO; } + unsigned int getDeviceCount( void ) override; + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; + void closeStream( void ) override; + void startStream( void ) override; + void stopStream( void ) override; + void abortStream( void ) override; + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + bool callbackEvent( long bufferIndex ); + + private: + + std::vector devices_; + void saveDeviceInfo( void ); + bool coInitialized_; + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) override; +}; + +#endif + +#if defined(__WINDOWS_DS__) + +class RtApiDs: public RtApi +{ +public: + + RtApiDs(); + ~RtApiDs(); + RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_DS; } + unsigned int getDeviceCount( void ) override; + unsigned int getDefaultOutputDevice( void ) override; + unsigned int getDefaultInputDevice( void ) override; + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; + void closeStream( void ) override; + void startStream( void ) override; + void stopStream( void ) override; + void abortStream( void ) override; + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + void callbackEvent( void ); + + private: + + bool coInitialized_; + bool buffersRolling; + long duplexPrerollBytes; + std::vector dsDevices; + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) override; +}; + +#endif + +#if defined(__WINDOWS_WASAPI__) + +struct IMMDeviceEnumerator; + +class RtApiWasapi : public RtApi +{ +public: + RtApiWasapi(); + virtual ~RtApiWasapi(); + + RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_WASAPI; } + unsigned int getDeviceCount( void ) override; + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; + unsigned int getDefaultOutputDevice( void ) override; + unsigned int getDefaultInputDevice( void ) override; + void closeStream( void ) override; + void startStream( void ) override; + void stopStream( void ) override; + void abortStream( void ) override; + +private: + bool coInitialized_; + IMMDeviceEnumerator* deviceEnumerator_; + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int* bufferSize, + RtAudio::StreamOptions* options ) override; + + static DWORD WINAPI runWasapiThread( void* wasapiPtr ); + static DWORD WINAPI stopWasapiThread( void* wasapiPtr ); + static DWORD WINAPI abortWasapiThread( void* wasapiPtr ); + void wasapiThread(); +}; + +#endif + +#if defined(__LINUX_ALSA__) + +class RtApiAlsa: public RtApi +{ +public: + + RtApiAlsa(); + ~RtApiAlsa(); + RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_ALSA; } + unsigned int getDeviceCount( void ) override; + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; + void closeStream( void ) override; + void startStream( void ) override; + void stopStream( void ) override; + void abortStream( void ) override; + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + void callbackEvent( void ); + + private: + + std::vector devices_; + void saveDeviceInfo( void ); + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) override; +}; + +#endif + +#if defined(__LINUX_PULSE__) + +class RtApiPulse: public RtApi +{ +public: + ~RtApiPulse(); + RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_PULSE; } + unsigned int getDeviceCount( void ) override; + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; + void closeStream( void ) override; + void startStream( void ) override; + void stopStream( void ) override; + void abortStream( void ) override; + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + void callbackEvent( void ); + + private: + + void collectDeviceInfo( void ); + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) override; +}; + +#endif + +#if defined(__LINUX_OSS__) + +class RtApiOss: public RtApi +{ +public: + + RtApiOss(); + ~RtApiOss(); + RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_OSS; } + unsigned int getDeviceCount( void ) override; + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; + void closeStream( void ) override; + void startStream( void ) override; + void stopStream( void ) override; + void abortStream( void ) override; + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + void callbackEvent( void ); + + private: + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) override; +}; + +#endif + +#if defined(__RTAUDIO_DUMMY__) + +class RtApiDummy: public RtApi +{ +public: + + RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtAudioError::WARNING ); } + RtAudio::Api getCurrentApi( void ) override { return RtAudio::RTAUDIO_DUMMY; } + unsigned int getDeviceCount( void ) override { return 0; } + RtAudio::DeviceInfo getDeviceInfo( unsigned int /*device*/ ) override { RtAudio::DeviceInfo info; return info; } + void closeStream( void ) override {} + void startStream( void ) override {} + void stopStream( void ) override {} + void abortStream( void ) override {} + + private: + + bool probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/, + unsigned int /*firstChannel*/, unsigned int /*sampleRate*/, + RtAudioFormat /*format*/, unsigned int * /*bufferSize*/, + RtAudio::StreamOptions * /*options*/ ) override { return false; } +}; + +#endif + +#endif + +// Indentation settings for Vim and Emacs +// +// Local Variables: +// c-basic-offset: 2 +// indent-tabs-mode: nil +// End: +// +// vim: et sts=2 sw=2 diff --git a/wfmain.cpp b/wfmain.cpp index 8893c7fb..e2e5d53f 100644 --- a/wfmain.cpp +++ b/wfmain.cpp @@ -158,6 +158,49 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, QWidget *parent connect(keyM, SIGNAL(activated()), this, SLOT(shortcutM())); // Enumerate audio devices, need to do before settings are loaded. +#ifdef USE_RTAUDIO + std::map apiMap; + apiMap[RtAudio::MACOSX_CORE] = "OS-X Core Audio"; + apiMap[RtAudio::WINDOWS_ASIO] = "Windows ASIO"; + apiMap[RtAudio::WINDOWS_DS] = "Windows DirectSound"; + apiMap[RtAudio::WINDOWS_WASAPI] = "Windows WASAPI"; + apiMap[RtAudio::UNIX_JACK] = "Jack Client"; + apiMap[RtAudio::LINUX_ALSA] = "Linux ALSA"; + apiMap[RtAudio::LINUX_PULSE] = "Linux PulseAudio"; + apiMap[RtAudio::LINUX_OSS] = "Linux OSS"; + apiMap[RtAudio::RTAUDIO_DUMMY] = "RtAudio Dummy"; + + std::vector< RtAudio::Api > apis; + RtAudio::getCompiledApi(apis); + + qInfo(logAudio()) << "RtAudio Version " << QString::fromStdString(RtAudio::getVersion()); + + qInfo(logAudio()) << "Compiled APIs:"; + for (unsigned int i = 0; i < apis.size(); i++) { + qInfo(logAudio()) << " " << QString::fromStdString(apiMap[apis[i]]); + } + + RtAudio::DeviceInfo info; + qInfo(logAudio()) << "Current API: " << QString::fromStdString(apiMap[audio.getCurrentApi()]); + + unsigned int devices = audio.getDeviceCount(); + qInfo(logAudio()) << "Found " << devices << " audio device(s) ..."; + + for (unsigned int i = 0; i < devices; i++) { + info = audio.getDeviceInfo(i); + if (info.outputChannels > 0) { + qInfo(logAudio()) << (info.isDefaultOutput ? "*" : " ") << "Output Device : " << QString::fromStdString(info.name); + ui->audioOutputCombo->addItem(QString::fromStdString(info.name),i); + } + } + for (unsigned int i = 0; i < devices; i++) { + info = audio.getDeviceInfo(i); + if (info.inputChannels > 0) { + qInfo(logAudio()) << (info.isDefaultInput ? "*" : " ") << "Input Device : " << QString::fromStdString(info.name); + ui->audioInputCombo->addItem(QString::fromStdString(info.name), i); + } + } +#else if defined(USE_QTAUDIO) const auto audioOutputs = QAudioDeviceInfo::availableDevices(QAudio::AudioOutput); for (const QAudioDeviceInfo& deviceInfo : audioOutputs) { ui->audioOutputCombo->addItem(deviceInfo.deviceName()); @@ -166,6 +209,7 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, QWidget *parent for (const QAudioDeviceInfo& deviceInfo : audioInputs) { ui->audioInputCombo->addItem(deviceInfo.deviceName()); } +#endif ui->serialDeviceListCombo->blockSignals(true); ui->serialDeviceListCombo->addItem("Auto", 0); @@ -860,8 +904,13 @@ void wfmain::setDefPrefs() udpDefPrefs.audioLANPort = 50003; udpDefPrefs.username = QString(""); udpDefPrefs.password = QString(""); +#if defined(USE_RTAUDIO) + udpDefPrefs.audioOutput = QString::fromStdString(audio.getDeviceInfo(audio.getDefaultOutputDevice()).name); + udpDefPrefs.audioInput = QString::fromStdString(audio.getDeviceInfo(audio.getDefaultInputDevice()).name); +#elif defined(USE_QTAUDIO) udpDefPrefs.audioOutput = QAudioDeviceInfo::defaultOutputDevice().deviceName(); udpDefPrefs.audioInput = QAudioDeviceInfo::defaultInputDevice().deviceName(); +#endif udpDefPrefs.audioRXLatency = 150; udpDefPrefs.audioTXLatency = 150; udpDefPrefs.audioRXSampleRate = 48000; diff --git a/wfmain.h b/wfmain.h index d9f34496..796c605a 100644 --- a/wfmain.h +++ b/wfmain.h @@ -700,6 +700,8 @@ private: SERVERCONFIG serverConfig; + RtAudio audio; + }; Q_DECLARE_METATYPE(struct rigCapabilities) diff --git a/wfview.pro b/wfview.pro index f1838088..12b82230 100644 --- a/wfview.pro +++ b/wfview.pro @@ -33,6 +33,23 @@ DEFINES += QCUSTOMPLOT_COMPILE_LIBRARY DEFINES += OUTSIDE_SPEEX DEFINES += RANDOM_PREFIX=wf +# RTAudio defines +win32:DEFINES += __WINDOWS_WASAPI__ +#win32:DEFINES += __WINDOWS_DS__ # Requires DirectSound libraries +linux:DEFINES += __LINUX_ALSA__ +#linux:DEFINES += __LINUX_OSS__ +#linux:DEFINES += __LINUX_PULSE__ +macx:DEFINES += __MACOSX_CORE__ + +option(RTAUDIO_API_DS "Build DirectSound API" OFF) +option(RTAUDIO_API_ASIO "Build ASIO API" OFF) +option(RTAUDIO_API_WASAPI "Build WASAPI API" ${WIN32}) +option(RTAUDIO_API_OSS "Build OSS4 API" ${xBSD}) +option(RTAUDIO_API_ALSA "Build ALSA API" ${LINUX}) +option(RTAUDIO_API_PULSE "Build PulseAudio API" ${pulse_FOUND}) +option(RTAUDIO_API_JACK "Build JACK audio server API" ${HAVE_JACK}) +option(RTAUDIO_API_CORE "Build CoreAudio API" ${APPLE}) + macx:INCLUDEPATH += /usr/local/include /opt/local/include macx:LIBS += -L/usr/local/lib -L/opt/local/lib @@ -99,7 +116,8 @@ SOURCES += main.cpp\ pttyhandler.cpp \ resampler/resample.c \ repeatersetup.cpp \ - rigctld.cpp + rigctld.cpp \ + rtaudio/RTAudio.cpp HEADERS += wfmain.h \ commhandler.h \ @@ -122,7 +140,8 @@ HEADERS += wfmain.h \ resampler/resample_sse.h \ repeatersetup.h \ repeaterattributes.h \ - rigctld.h + rigctld.h \ + rtaudio/RTAudio.h FORMS += wfmain.ui \ diff --git a/wfview.vcxproj b/wfview.vcxproj index 3c50c5bc..6a56163e 100644 --- a/wfview.vcxproj +++ b/wfview.vcxproj @@ -325,7 +325,7 @@ Sync debug\ Disabled - _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;OUTSIDE_SPEEX;RANDOM_PREFIX=wf;%(PreprocessorDefinitions) + _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;OUTSIDE_SPEEX;RANDOM_PREFIX=wf;__WINDOWS_WASAPI__;%(PreprocessorDefinitions) false MultiThreadedDebugDLL true @@ -389,6 +389,7 @@ + @@ -396,6 +397,7 @@ + diff --git a/wfview.vcxproj.filters b/wfview.vcxproj.filters index efa5219e..afcab062 100644 --- a/wfview.vcxproj.filters +++ b/wfview.vcxproj.filters @@ -117,6 +117,9 @@ Source Files + + Source Files + @@ -350,6 +353,8 @@ + + @@ -369,5 +374,8 @@ Header Files + + Header Files + \ No newline at end of file From b4c10ce2a0d6babef8e1099c38e45efd2c54a63c Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Sun, 16 May 2021 21:16:59 +0100 Subject: [PATCH 02/47] More rtaudio stuff --- audiohandler.cpp | 749 +--------------------------------------- audiohandler.h | 755 ++++++++++++++++++++++++++++++++++++++++- audiohandler_rt.cpp | 421 +++++++++++++++++++++++ udphandler.cpp | 14 +- udphandler.h | 10 +- udpserver.cpp | 8 +- udpserver.h | 4 +- udpserversetup.h | 2 + wfmain.cpp | 35 +- wfmain.h | 8 +- wfview.vcxproj | 1 + wfview.vcxproj.filters | 3 + 12 files changed, 1236 insertions(+), 774 deletions(-) create mode 100644 audiohandler_rt.cpp diff --git a/audiohandler.cpp b/audiohandler.cpp index 8bbdff8b..daae0ee0 100644 --- a/audiohandler.cpp +++ b/audiohandler.cpp @@ -3,739 +3,20 @@ but as the setup/handling if output (RX) and input (TX) devices is so similar I have combined them. */ #include "audiohandler.h" + +#ifndef USE_RTAUDIO #include "logcategories.h" #define MULAW_BIAS 33 #define MULAW_MAX 0x1fff -static unsigned char ulaw_encode[8193] = - 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0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80 -}; - -static const qint16 ulaw_decode[256] = { - -32124, -31100, -30076, -29052, -28028, -27004, -25980, -24956, - -23932, -22908, -21884, -20860, -19836, -18812, -17788, -16764, - -15996, -15484, -14972, -14460, -13948, -13436, -12924, -12412, - -11900, -11388, -10876, -10364, -9852, -9340, -8828, -8316, - -7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140, - -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092, - -3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004, - -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980, - -1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436, - -1372, -1308, -1244, -1180, -1116, -1052, -988, -924, - -876, -844, -812, -780, -748, -716, -684, -652, - -620, -588, -556, -524, -492, -460, -428, -396, - -372, -356, -340, -324, -308, -292, -276, -260, - -244, -228, -212, -196, -180, -164, -148, -132, - -120, -112, -104, -96, -88, -80, -72, -64, - -56, -48, -40, -32, -24, -16, -8, 0, - 32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956, - 23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764, - 15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412, - 11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316, - 7932, 7676, 7420, 7164, 6908, 6652, 6396, 6140, - 5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092, - 3900, 3772, 3644, 3516, 3388, 3260, 3132, 3004, - 2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980, - 1884, 1820, 1756, 1692, 1628, 1564, 1500, 1436, - 1372, 1308, 1244, 1180, 1116, 1052, 988, 924, - 876, 844, 812, 780, 748, 716, 684, 652, - 620, 588, 556, 524, 492, 460, 428, 396, - 372, 356, 340, 324, 308, 292, 276, 260, - 244, 228, 212, 196, 180, 164, 148, 132, - 120, 112, 104, 96, 88, 80, 72, 64, - 56, 48, 40, 32, 24, 16, 8, 0 -}; - audioHandler::audioHandler(QObject* parent) : QIODevice(parent), isInitialized(false), audioOutput(Q_NULLPTR), audioInput(Q_NULLPTR), isUlaw(false), - latency(0), + audioLatency(0), isInput(0), chunkAvailable(false) { @@ -758,12 +39,13 @@ audioHandler::~audioHandler() } } -bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, QString port, quint8 resampleQuality) +bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, QString port, int device, quint8 resampleQuality) { if (isInitialized) { return false; } - /* Always use 16 bit 48K samples internally*/ + + /* Always use 16 bit 48K samples internally*/ format.setSampleSize(16); format.setChannelCount(channels); format.setSampleRate(INTERNAL_SAMPLE_RATE); @@ -771,7 +53,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 format.setByteOrder(QAudioFormat::LittleEndian); format.setSampleType(QAudioFormat::SignedInt); - this->latency = latency; + this->audioLatency = latency; this->isUlaw = ulaw; this->isInput = isinput; this->radioSampleBits = bits; @@ -787,7 +69,6 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 if (isInput) { resampler = wf_resampler_init(radioChannels, INTERNAL_SAMPLE_RATE, samplerate, resampleQuality, &resample_error); - const auto deviceInfos = QAudioDeviceInfo::availableDevices(QAudio::AudioInput); for (const QAudioDeviceInfo& deviceInfo : deviceInfos) { if (deviceInfo.deviceName() == port) { @@ -819,6 +100,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 } } + wf_resampler_get_ratio(resampler, &ratioNum, &ratioDen); qInfo(logAudio()) << (isInput ? "Input" : "Output") << "wf_resampler_init() returned: " << resample_error << " ratioNum" << ratioNum << " ratioDen" << ratioDen; @@ -829,14 +111,17 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 void audioHandler::setVolume(unsigned char volume) { //qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << (qreal)(volume/255.0) << ")"; + if (audioOutput != Q_NULLPTR) { audioOutput->setVolume((qreal)(volume / 255.0)); } else if (audioInput != Q_NULLPTR) { audioInput->setVolume((qreal)(volume / 255.0)); } + } + bool audioHandler::setDevice(QAudioDeviceInfo deviceInfo) { bool ret = true; @@ -988,11 +273,11 @@ qint64 audioHandler::readData(char* data, qint64 maxlen) { int timediff = packet->time.msecsTo(QTime::currentTime()); - if (timediff > (int)latency * 2) { + if (timediff > (int)audioLatency * 2) { qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet->seq << " arrived too late (increase output latency!) " << dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; - while (packet !=audioBuffer.end() && timediff > (int)latency) { + while (packet !=audioBuffer.end() && timediff > (int)audioLatency) { timediff = packet->time.msecsTo(QTime::currentTime()); lastSeq=packet->seq; packet = audioBuffer.erase(packet); // returns next packet @@ -1212,13 +497,13 @@ void audioHandler::incomingAudio(audioPacket data) void audioHandler::changeLatency(const quint16 newSize) { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Changing latency to: " << newSize << " from " << latency; - latency = newSize; + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Changing latency to: " << newSize << " from " << audioLatency; + audioLatency = newSize; } void audioHandler::getLatency() { - emit sendLatency(latency); + emit sendLatency(audioLatency); } bool audioHandler::isChunkAvailable() @@ -1312,5 +597,5 @@ void audioHandler::getNextAudioChunk(QByteArray& ret) return; } - +#endif diff --git a/audiohandler.h b/audiohandler.h index e7145c72..d65978b8 100644 --- a/audiohandler.h +++ b/audiohandler.h @@ -2,7 +2,7 @@ #define AUDIOHANDLER_H #define USE_RTAUDIO -#define USE_QTAUDIO +#undef USE_QTAUDIO #include @@ -21,6 +21,11 @@ #ifdef USE_RTAUDIO #include "rtaudio/RtAudio.h" + +typedef signed short MY_TYPE; +#define FORMAT RTAUDIO_SINT16 +#define SCALE 32767.0 + #endif #include @@ -35,6 +40,731 @@ #define INTERNAL_SAMPLE_RATE 48000 +#define MULAW_BIAS 33 +#define MULAW_MAX 0x1fff + +static unsigned char ulaw_encode[8193] = +{ 0xff, 0xfe, 0xfe, 0xfd, 0xfd, 0xfc, 0xfc, 0xfb, 0xfb, 0xfa, 0xfa, 0xf9, + 0xf9, 0xf8, 0xf8, 0xf7, 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0x82, 0x82, + 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, 0x82, + 0x82, 0x82, 0x82, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80 +}; + +static const qint16 ulaw_decode[256] = { + -32124, -31100, -30076, -29052, -28028, -27004, -25980, -24956, + -23932, -22908, -21884, -20860, -19836, -18812, -17788, -16764, + -15996, -15484, -14972, -14460, -13948, -13436, -12924, -12412, + -11900, -11388, -10876, -10364, -9852, -9340, -8828, -8316, + -7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140, + -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092, + -3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004, + -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980, + -1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436, + -1372, -1308, -1244, -1180, -1116, -1052, -988, -924, + -876, -844, -812, -780, -748, -716, -684, -652, + -620, -588, -556, -524, -492, -460, -428, -396, + -372, -356, -340, -324, -308, -292, -276, -260, + -244, -228, -212, -196, -180, -164, -148, -132, + -120, -112, -104, -96, -88, -80, -72, -64, + -56, -48, -40, -32, -24, -16, -8, 0, + 32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956, + 23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764, + 15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412, + 11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316, + 7932, 7676, 7420, 7164, 6908, 6652, 6396, 6140, + 5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092, + 3900, 3772, 3644, 3516, 3388, 3260, 3132, 3004, + 2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980, + 1884, 1820, 1756, 1692, 1628, 1564, 1500, 1436, + 1372, 1308, 1244, 1180, 1116, 1052, 988, 924, + 876, 844, 812, 780, 748, 716, 684, 652, + 620, 588, 556, 524, 492, 460, 428, 396, + 372, 356, 340, 324, 308, 292, 276, 260, + 244, 228, 212, 196, 180, 164, 148, 132, + 120, 112, 104, 96, 88, 80, 72, 64, + 56, 48, 40, 32, 24, 16, 8, 0 +}; + + struct audioPacket { quint32 seq; QTime time; @@ -68,7 +798,7 @@ public: void getNextAudioChunk(QByteArray &data); bool isChunkAvailable(); public slots: - bool init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isulaw, const bool isinput, QString port, quint8 resampleQuality); + bool init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isulaw, const bool isinput, QString port, int device, quint8 resampleQuality); void incomingAudio(const audioPacket data); void changeLatency(const quint16 newSize); @@ -89,24 +819,33 @@ private: QMutex mutex; bool isInitialized; - QAudioOutput* audioOutput; - QAudioInput* audioInput; +#ifdef USE_RTAUDIO + RtAudio audio; + int audioDevice = 0; + RtAudio::StreamParameters aParams; + RtAudio::DeviceInfo info; + +#elif defined(USE_QTAUDIO) + QAudioOutput* audioOutput; + QAudioInput* audioInput; + QAudioFormat format; + QAudioDeviceInfo deviceInfo; +#endif + SpeexResamplerState* resampler; + bool isUlaw; - quint16 latency; + quint16 audioLatency; bool isInput; // Used to determine whether input or output audio int chunkSize; bool chunkAvailable; quint32 lastSeq; - QAudioFormat format; - QAudioDeviceInfo deviceInfo; quint16 radioSampleRate; quint8 radioSampleBits; quint8 radioChannels; QVector audioBuffer; - SpeexResamplerState* resampler; unsigned int ratioNum; unsigned int ratioDen; }; diff --git a/audiohandler_rt.cpp b/audiohandler_rt.cpp new file mode 100644 index 00000000..e4a44d19 --- /dev/null +++ b/audiohandler_rt.cpp @@ -0,0 +1,421 @@ +/* + This class handles both RX and TX audio, each is created as a seperate instance of the class + but as the setup/handling if output (RX) and input (TX) devices is so similar I have combined them. +*/ +#include "audiohandler.h" + +#include "logcategories.h" + + +audioHandler::audioHandler(QObject* parent) : + QIODevice(parent), + isInitialized(false), + isUlaw(false), + audioLatency(0), + isInput(0), + chunkAvailable(false) +{ +} + +audioHandler::~audioHandler() +{ + //stop(); + + if (resampler) { + speex_resampler_destroy(resampler); + } +} + +bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, QString port, int device, quint8 resampleQuality) +{ + if (isInitialized) { + return false; + } + + this->audioDevice = device; + this->audioLatency = latency; + this->isUlaw = ulaw; + this->isInput = isinput; + this->radioSampleBits = bits; + this->radioSampleRate = samplerate; + this->radioChannels = channels; + + // chunk size is always relative to Internal Sample Rate. + this->chunkSize = (INTERNAL_SAMPLE_RATE / 25) * radioChannels; + + + if (device != 0) { + aParams.deviceId = device; + } + else if (isInput) { + aParams.deviceId = audio.getDefaultInputDevice(); + } + else { + aParams.deviceId = audio.getDefaultOutputDevice(); + } + + info = audio.getDeviceInfo(aParams.deviceId); + + if (info.probed) + { + qInfo(logAudio()) << (isInput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") successfully probed"; + if (info.nativeFormats == 0) + qInfo(logAudio()) << " No natively supported data formats!"; + else { + qDebug(logAudio()) << " Supported formats:" << + (info.nativeFormats & RTAUDIO_SINT8 ? "8-bit int," : "") << + (info.nativeFormats & RTAUDIO_SINT16 ? "16-bit int," : "") << + (info.nativeFormats & RTAUDIO_SINT24 ? "24-bit int," : "") << + (info.nativeFormats & RTAUDIO_SINT32 ? "32-bit int," : "") << + (info.nativeFormats & RTAUDIO_FLOAT32 ? "32-bit float," : "") << + (info.nativeFormats & RTAUDIO_FLOAT64 ? "64-bit float," : ""); + qInfo(logAudio()) << " Preferred sample rate:" << info.preferredSampleRate; + } + + qInfo(logAudio()) << " chunkSize: " << chunkSize; + } + else + { + qCritical(logAudio()) << (isInput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") could not be probed, check audio configuration!"; + } + + + int resample_error = 0; + + if (isInput) { + resampler = wf_resampler_init(radioChannels, INTERNAL_SAMPLE_RATE, samplerate, resampleQuality, &resample_error); + //audio.openStream(&aParams, NULL, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &this->chunkSize, &output, (void*)&data); + } + else + { + resampler = wf_resampler_init(radioChannels, samplerate, INTERNAL_SAMPLE_RATE, resampleQuality, &resample_error); + //audio.openStream(&aParams, NULL, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &this->chunkSize, &output, (void*)&data); + } + + wf_resampler_get_ratio(resampler, &ratioNum, &ratioDen); + qInfo(logAudio()) << " wf_resampler_init() returned: " << resample_error << " ratioNum" << ratioNum << " ratioDen" << ratioDen; + + return isInitialized; +} + +void audioHandler::setVolume(unsigned char volume) +{ + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << (qreal)(volume/255.0) << ")"; +} + + + +/// +/// This function processes the incoming audio FROM the radio and pushes it into the playback buffer *data +/// +/// +/// +/// +qint64 audioHandler::readData(char* data, qint64 maxlen) +{ + // Calculate output length, always full samples + int sentlen = 0; + + //qInfo(logAudio()) << "Looking for: " << maxlen << " bytes"; + + // We must lock the mutex for the entire time that the buffer may be modified. + // Get next packet from buffer. + if (!audioBuffer.isEmpty()) + { + + // Output buffer is ALWAYS 16 bit. + QMutexLocker locker(&mutex); + auto packet = audioBuffer.begin(); + + while (packet != audioBuffer.end() && sentlen < maxlen) + { + int timediff = packet->time.msecsTo(QTime::currentTime()); + + if (timediff > (int)audioLatency * 2) { + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet->seq << + " arrived too late (increase output latency!) " << + dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; + while (packet != audioBuffer.end() && timediff > (int)audioLatency) { + timediff = packet->time.msecsTo(QTime::currentTime()); + lastSeq = packet->seq; + packet = audioBuffer.erase(packet); // returns next packet + } + if (packet == audioBuffer.end()) { + break; + } + } + + // If we got here then packet time must be within latency threshold + + if (packet->seq == lastSeq + 1 || packet->seq <= lastSeq) + { + int send = qMin((int)maxlen - sentlen, packet->dataout.length() - packet->sent); + lastSeq = packet->seq; + //qInfo(logAudio()) << "Packet " << hex << packet->seq << " arrived on time " << Qt::dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; + + memcpy(data + sentlen, packet->dataout.constData() + packet->sent, send); + + sentlen = sentlen + send; + + if (send == packet->dataout.length() - packet->sent) + { + //qInfo(logAudio()) << "Get next packet"; + packet = audioBuffer.erase(packet); // returns next packet + } + else + { + // Store sent amount (could be zero if audioOutput buffer full) then break. + packet->sent = send; + break; + } + } + else { + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet->seq - 1; + lastSeq = packet->seq; + } + } + } + else { + // Fool audio system into thinking it has valid data, this seems to be required + // for MacOS Built in audio but shouldn't cause any issues with other platforms. + memset(data, 0x0, maxlen); + return maxlen; + } + + return sentlen; +} + +qint64 audioHandler::writeData(const char* data, qint64 len) +{ + qint64 sentlen = 0; + QMutexLocker locker(&mutex); + audioPacket* current; + + while (sentlen < len) { + if (!audioBuffer.isEmpty()) + { + if (audioBuffer.last().sent == chunkSize) + { + audioBuffer.append(audioPacket()); + audioBuffer.last().sent = 0; + } + } + else + { + audioBuffer.append(audioPacket()); + audioBuffer.last().sent = 0; + } + current = &audioBuffer.last(); + + int send = qMin((int)(len - sentlen), (int)chunkSize - current->sent); + + current->datain.append(QByteArray::fromRawData(data + sentlen, send)); + + sentlen = sentlen + send; + + current->seq = 0; // Not used in TX + current->time = QTime::currentTime(); + current->sent = current->datain.length(); + + if (current->sent == chunkSize) + { + chunkAvailable = true; + } + else if (audioBuffer.length() <= 1 && current->sent != chunkSize) { + chunkAvailable = false; + } + + } + + return (sentlen); // Always return the same number as we received +} + +qint64 audioHandler::bytesAvailable() const +{ + return 0; +} + +bool audioHandler::isSequential() const +{ + return true; +} + +void audioHandler::notified() +{ +} + + +void audioHandler::stateChanged(QAudio::State state) +{ + +} + + + +void audioHandler::incomingAudio(audioPacket data) +{ + QMutexLocker locker(&mutex); + + // Incoming data is 8bits? + if (radioSampleBits == 8) + { + QByteArray inPacket((int)data.datain.length() * 2, (char)0xff); + qint16* in = (qint16*)inPacket.data(); + for (int f = 0; f < data.datain.length(); f++) + { + if (isUlaw) + { + in[f] = ulaw_decode[(quint8)data.datain[f]]; + } + else + { + // Convert 8-bit sample to 16-bit + in[f] = (qint16)(((quint8)data.datain[f] << 8) - 32640); + } + } + data.datain = inPacket; // Replace incoming data with converted. + } + + //qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.datain.length(); + + /* We now have an array of 16bit samples in the NATIVE samplerate of the radio + If the radio sample rate is below 48000, we need to resample. + */ + + if (ratioDen != 1) { + + // We need to resample + quint32 outFrames = ((data.datain.length() / 2) * ratioDen) / radioChannels; + quint32 inFrames = (data.datain.length() / 2) / radioChannels; + data.dataout.resize(outFrames * 2 * radioChannels); // Preset the output buffer size. + + int err = 0; + if (this->radioChannels == 1) { + err = wf_resampler_process_int(resampler, 0, (const qint16*)data.datain.constData(), &inFrames, (qint16*)data.dataout.data(), &outFrames); + } + else { + err = wf_resampler_process_interleaved_int(resampler, (const qint16*)data.datain.constData(), &inFrames, (qint16*)data.dataout.data(), &outFrames); + } + if (err) { + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; + } + } + else { + data.dataout = data.datain; + } + + audioBuffer.push_back(data); + + // Sort the buffer by seq number. This is important and audio packets may have arrived out-of-order + std::sort(audioBuffer.begin(), audioBuffer.end(), + [](const audioPacket& a, const audioPacket& b) -> bool + { + return a.seq < b.seq; + }); + +} + +void audioHandler::changeLatency(const quint16 newSize) +{ + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Changing latency to: " << newSize << " from " << audioLatency; + audioLatency = newSize; +} + +void audioHandler::getLatency() +{ + emit sendLatency(audioLatency); +} + +bool audioHandler::isChunkAvailable() +{ + return (chunkAvailable); +} + +void audioHandler::getNextAudioChunk(QByteArray& ret) +{ + if (!audioBuffer.isEmpty() && chunkAvailable) + { + + QMutexLocker locker(&mutex); + + // Skip through audio buffer deleting any old entry. + auto packet = audioBuffer.begin(); + while (packet != audioBuffer.end()) + { + if (packet->time.msecsTo(QTime::currentTime()) > 100) { + //qInfo(logAudio()) << "TX Packet too old " << dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; + packet = audioBuffer.erase(packet); // returns next packet + } + else { + if (packet->datain.length() == chunkSize && ret.length() == 0) + { + /* We now have an array of samples in the computer native format (48000) + If the radio sample rate is below 48000, we need to resample. + */ + + if (ratioNum != 1) + { + // We need to resample (we are STILL 16 bit!) + quint32 outFrames = ((packet->datain.length() / 2) / ratioNum) / radioChannels; + quint32 inFrames = (packet->datain.length() / 2) / radioChannels; + packet->dataout.resize(outFrames * 2 * radioChannels); // Preset the output buffer size. + + const qint16* in = (qint16*)packet->datain.constData(); + qint16* out = (qint16*)packet->dataout.data(); + int err = 0; + if (this->radioChannels == 1) { + err = wf_resampler_process_int(resampler, 0, in, &inFrames, out, &outFrames); + } + else { + err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames); + } + if (err) { + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; + } + //qInfo(logAudio()) << "Resampler run " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; + //qInfo(logAudio()) << "Resampler run inLen:" << packet->datain.length() << " outLen:" << packet->dataout.length(); + if (radioSampleBits == 8) + { + packet->datain = packet->dataout; // Copy output packet back to input buffer. + packet->dataout.clear(); // Buffer MUST be cleared ready to be re-filled by the upsampling below. + } + } + else if (radioSampleBits == 16) { + // Only copy buffer if radioSampleBits is 16, as it will be handled below otherwise. + packet->dataout = packet->datain; + } + + // Do we need to convert 16-bit to 8-bit? + if (radioSampleBits == 8) { + packet->dataout.resize(packet->datain.length() / 2); + qint16* in = (qint16*)packet->datain.data(); + for (int f = 0; f < packet->dataout.length(); f++) + { + quint8 outdata = 0; + if (isUlaw) { + qint16 enc = qFromLittleEndian(in + f); + if (enc >= 0) + outdata = ulaw_encode[enc]; + else + outdata = 0x7f & ulaw_encode[-enc]; + } + else { + outdata = (quint8)(((qFromLittleEndian(in + f) >> 8) ^ 0x80) & 0xff); + } + packet->dataout[f] = (char)outdata; + } + } + ret = packet->dataout; + packet = audioBuffer.erase(packet); // returns next packet + } + else { + packet++; + } + } + } + } + return; +} + + + diff --git a/udphandler.cpp b/udphandler.cpp index 0db339a2..514c34a1 100644 --- a/udphandler.cpp +++ b/udphandler.cpp @@ -16,6 +16,8 @@ udpHandler::udpHandler(udpPreferences prefs) : txCodec(prefs.audioTXCodec), audioInputPort(prefs.audioInput), audioOutputPort(prefs.audioOutput), + audioInputDevice(prefs.audioInputDevice), + audioOutputDevice(prefs.audioOutputDevice), resampleQuality(prefs.resampleQuality) { @@ -313,7 +315,7 @@ void udpHandler::dataReceived() } else { civ = new udpCivData(localIP, radioIP, civPort); - audio = new udpAudio(localIP, radioIP, audioPort, rxLatency, txLatency, rxSampleRate, rxCodec, txSampleRate, txCodec, audioOutputPort, audioInputPort,resampleQuality); + audio = new udpAudio(localIP, radioIP, audioPort, rxLatency, txLatency, rxSampleRate, rxCodec, txSampleRate, txCodec, audioOutputPort, audioOutputDevice, audioInputPort,audioInputDevice, resampleQuality); QObject::connect(civ, SIGNAL(receive(QByteArray)), this, SLOT(receiveFromCivStream(QByteArray))); QObject::connect(audio, SIGNAL(haveAudioData(audioPacket)), this, SLOT(receiveAudioData(audioPacket))); @@ -669,7 +671,7 @@ void udpCivData::dataReceived() // Audio stream -udpAudio::udpAudio(QHostAddress local, QHostAddress ip, quint16 audioPort, quint16 rxlatency, quint16 txlatency, quint16 rxsample, quint8 rxcodec, quint16 txsample, quint8 txcodec, QString outputPort, QString inputPort,quint8 resampleQuality) +udpAudio::udpAudio(QHostAddress local, QHostAddress ip, quint16 audioPort, quint16 rxlatency, quint16 txlatency, quint16 rxsample, quint8 rxcodec, quint16 txsample, quint8 txcodec, QString outputPort, int outputDevice, QString inputPort, int inputDevice, quint8 resampleQuality) { qInfo(logUdp()) << "Starting udpAudio"; this->localIP = local; @@ -714,7 +716,7 @@ udpAudio::udpAudio(QHostAddress local, QHostAddress ip, quint16 audioPort, quint rxAudioThread->start(); - connect(this, SIGNAL(setupRxAudio(quint8, quint8, quint16, quint16, bool, bool, QString, quint8)), rxaudio, SLOT(init(quint8, quint8, quint16, quint16, bool, bool,QString, quint8))); + connect(this, SIGNAL(setupRxAudio(quint8, quint8, quint16, quint16, bool, bool, QString,int, quint8)), rxaudio, SLOT(init(quint8, quint8, quint16, quint16, bool, bool,QString, int, quint8))); qRegisterMetaType(); connect(this, SIGNAL(haveAudioData(audioPacket)), rxaudio, SLOT(incomingAudio(audioPacket))); @@ -736,7 +738,7 @@ udpAudio::udpAudio(QHostAddress local, QHostAddress ip, quint16 audioPort, quint txAudioThread->start(); - connect(this, SIGNAL(setupTxAudio(quint8, quint8, quint16, quint16, bool, bool,QString,quint8)), txaudio, SLOT(init(quint8, quint8, quint16, quint16, bool, bool,QString,quint8))); + connect(this, SIGNAL(setupTxAudio(quint8, quint8, quint16, quint16, bool, bool,QString, int,quint8)), txaudio, SLOT(init(quint8, quint8, quint16, quint16, bool, bool,QString,int,quint8))); connect(txAudioThread, SIGNAL(finished()), txaudio, SLOT(deleteLater())); sendControl(false, 0x03, 0x00); // First connect packet @@ -745,8 +747,8 @@ udpAudio::udpAudio(QHostAddress local, QHostAddress ip, quint16 audioPort, quint connect(pingTimer, &QTimer::timeout, this, &udpBase::sendPing); pingTimer->start(PING_PERIOD); // send ping packets every 100ms - emit setupTxAudio(txNumSamples, txChannelCount, txSampleRate, txLatency, txIsUlawCodec, true, inputPort,resampleQuality); - emit setupRxAudio(rxNumSamples, rxChannelCount, rxSampleRate, txLatency, rxIsUlawCodec, false, outputPort,resampleQuality); + emit setupTxAudio(txNumSamples, txChannelCount, txSampleRate, txLatency, txIsUlawCodec, true, inputPort,inputDevice, resampleQuality); + emit setupRxAudio(rxNumSamples, rxChannelCount, rxSampleRate, txLatency, rxIsUlawCodec, false, outputPort,outputDevice, resampleQuality); watchdogTimer = new QTimer(); connect(watchdogTimer, &QTimer::timeout, this, &udpAudio::watchdog); diff --git a/udphandler.h b/udphandler.h index dc4bc488..77eed8f4 100644 --- a/udphandler.h +++ b/udphandler.h @@ -42,6 +42,8 @@ struct udpPreferences { QString password; QString audioOutput; QString audioInput; + int audioInputDevice; + int audioOutputDevice; quint16 audioRXLatency; quint16 audioTXLatency; quint16 audioRXSampleRate; @@ -170,14 +172,14 @@ class udpAudio : public udpBase Q_OBJECT public: - udpAudio(QHostAddress local, QHostAddress ip, quint16 aport, quint16 rxlatency, quint16 txlatency, quint16 rxsample, quint8 rxcodec, quint16 txsample, quint8 txcodec, QString outputPort, QString inputPort,quint8 resampleQuality); + udpAudio(QHostAddress local, QHostAddress ip, quint16 aport, quint16 rxlatency, quint16 txlatency, quint16 rxsample, quint8 rxcodec, quint16 txsample, quint8 txcodec, QString outputPort, int outputDevice, QString inputPort, int inputDevice, quint8 resampleQuality); ~udpAudio(); signals: void haveAudioData(audioPacket data); - void setupTxAudio(const quint8 samples, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isUlaw, const bool isInput, QString port, quint8 resampleQuality); - void setupRxAudio(const quint8 samples, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isUlaw, const bool isInput, QString port, quint8 resampleQuality); + void setupTxAudio(const quint8 samples, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isUlaw, const bool isInput, QString port, int device, quint8 resampleQuality); + void setupRxAudio(const quint8 samples, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isUlaw, const bool isInput, QString port, int device, quint8 resampleQuality); void haveChangeLatency(quint16 value); void haveSetVolume(unsigned char value); @@ -284,6 +286,8 @@ private: QString audioInputPort; QString audioOutputPort; + int audioInputDevice; + int audioOutputDevice; quint8 resampleQuality; diff --git a/udpserver.cpp b/udpserver.cpp index 160a6ba0..9c73cffc 100644 --- a/udpserver.cpp +++ b/udpserver.cpp @@ -392,10 +392,10 @@ void udpServer::controlReceived() txAudioThread->start(); - connect(this, SIGNAL(setupTxAudio(quint8, quint8, quint16, quint16, bool, bool, QString, quint8)), txaudio, SLOT(init(quint8, quint8, quint16, quint16, bool, bool, QString, quint8))); + connect(this, SIGNAL(setupTxAudio(quint8, quint8, quint16, quint16, bool, bool, QString, int, quint8)), txaudio, SLOT(init(quint8, quint8, quint16, quint16, bool, bool, QString, int, quint8))); connect(txAudioThread, SIGNAL(finished()), txaudio, SLOT(deleteLater())); - emit setupTxAudio(samples, channels, current->txSampleRate, in->txbuffer, uLaw, false, config.audioOutput, config.resampleQuality); + emit setupTxAudio(samples, channels, current->txSampleRate, in->txbuffer, uLaw, false, config.audioOutput,config.audioOutputDevice, config.resampleQuality); hasTxAudio=datagram.senderAddress(); connect(this, SIGNAL(haveAudioData(audioPacket)), txaudio, SLOT(incomingAudio(audioPacket))); @@ -425,10 +425,10 @@ void udpServer::controlReceived() rxaudio->moveToThread(rxAudioThread); rxAudioThread->start(); - connect(this, SIGNAL(setupRxAudio(quint8, quint8, quint16, quint16, bool, bool, QString, quint8)), rxaudio, SLOT(init(quint8, quint8, quint16, quint16, bool, bool, QString, quint8))); + connect(this, SIGNAL(setupRxAudio(quint8, quint8, quint16, quint16, bool, bool, QString, int, quint8)), rxaudio, SLOT(init(quint8, quint8, quint16, quint16, bool, bool, QString, int, quint8))); connect(rxAudioThread, SIGNAL(finished()), txaudio, SLOT(deleteLater())); - emit setupRxAudio(samples, channels, current->rxSampleRate, 150, uLaw, true, config.audioInput, config.resampleQuality); + emit setupRxAudio(samples, channels, current->rxSampleRate, 150, uLaw, true, config.audioInput, config.audioInputDevice, config.resampleQuality); rxAudioTimer = new QTimer(); connect(rxAudioTimer, &QTimer::timeout, this, std::bind(&udpServer::sendRxAudio, this)); diff --git a/udpserver.h b/udpserver.h index abefe8e6..3b295d25 100644 --- a/udpserver.h +++ b/udpserver.h @@ -50,8 +50,8 @@ public slots: signals: void haveDataFromServer(QByteArray); void haveAudioData(audioPacket data); - void setupTxAudio(const quint8 samples, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isUlaw, const bool isInput, QString port, quint8 resampleQuality); - void setupRxAudio(const quint8 samples, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isUlaw, const bool isInput, QString port, quint8 resampleQuality); + void setupTxAudio(const quint8 samples, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isUlaw, const bool isInput, QString port, int device, quint8 resampleQuality); + void setupRxAudio(const quint8 samples, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isUlaw, const bool isInput, QString port, int device, quint8 resampleQuality); private: diff --git a/udpserversetup.h b/udpserversetup.h index b7074013..4f3f83f2 100644 --- a/udpserversetup.h +++ b/udpserversetup.h @@ -22,6 +22,8 @@ struct SERVERCONFIG { quint16 audioPort; QString audioOutput; QString audioInput; + int audioInputDevice; + int audioOutputDevice; quint8 resampleQuality; QList users; diff --git a/wfmain.cpp b/wfmain.cpp index e2e5d53f..98e5865c 100644 --- a/wfmain.cpp +++ b/wfmain.cpp @@ -189,25 +189,24 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, QWidget *parent for (unsigned int i = 0; i < devices; i++) { info = audio.getDeviceInfo(i); if (info.outputChannels > 0) { - qInfo(logAudio()) << (info.isDefaultOutput ? "*" : " ") << "Output Device : " << QString::fromStdString(info.name); - ui->audioOutputCombo->addItem(QString::fromStdString(info.name),i); + qInfo(logAudio()) << (info.isDefaultOutput ? "*" : " ") << "(" << i << ") Output Device : " << QString::fromStdString(info.name); + ui->audioOutputCombo->addItem(QString::fromStdString(info.name), i); } - } - for (unsigned int i = 0; i < devices; i++) { - info = audio.getDeviceInfo(i); - if (info.inputChannels > 0) { - qInfo(logAudio()) << (info.isDefaultInput ? "*" : " ") << "Input Device : " << QString::fromStdString(info.name); + else if (info.inputChannels > 0) { + qInfo(logAudio()) << (info.isDefaultInput ? "*" : " ") << "(" << i << ") Input Device : " << QString::fromStdString(info.name); ui->audioInputCombo->addItem(QString::fromStdString(info.name), i); } } #else if defined(USE_QTAUDIO) const auto audioOutputs = QAudioDeviceInfo::availableDevices(QAudio::AudioOutput); - for (const QAudioDeviceInfo& deviceInfo : audioOutputs) { - ui->audioOutputCombo->addItem(deviceInfo.deviceName()); + for (const QAudioDeviceInfo deviceInfo : audioOutputs) { + qInfo(logAudio()) << "Output Device : " << deviceInfo.deviceName(); + ui->audioOutputCombo->addItem(deviceInfo.deviceName(), 0); } const auto audioInputs = QAudioDeviceInfo::availableDevices(QAudio::AudioInput); - for (const QAudioDeviceInfo& deviceInfo : audioInputs) { - ui->audioInputCombo->addItem(deviceInfo.deviceName()); + for (const QAudioDeviceInfo deviceInfo : audioInputs) { + qInfo(logAudio()) << "Input Device : " << deviceInfo.deviceName(); + ui->audioInputCombo->addItem(deviceInfo.deviceName(), 0); } #endif @@ -911,6 +910,8 @@ void wfmain::setDefPrefs() udpDefPrefs.audioOutput = QAudioDeviceInfo::defaultOutputDevice().deviceName(); udpDefPrefs.audioInput = QAudioDeviceInfo::defaultInputDevice().deviceName(); #endif + udpDefPrefs.audioInputDevice = 0; + udpDefPrefs.audioOutputDevice = 0; udpDefPrefs.audioRXLatency = 150; udpDefPrefs.audioTXLatency = 150; udpDefPrefs.audioRXSampleRate = 48000; @@ -1079,6 +1080,7 @@ void wfmain::loadSettings() int audioOutputIndex = ui->audioOutputCombo->findText(udpPrefs.audioOutput); if (audioOutputIndex != -1) { ui->audioOutputCombo->setCurrentIndex(audioOutputIndex); + udpPrefs.audioOutputDevice = ui->audioOutputCombo->itemData(audioOutputIndex).toInt(); } udpPrefs.audioInput = settings.value("AudioInput", udpDefPrefs.audioInput).toString(); @@ -1087,6 +1089,7 @@ void wfmain::loadSettings() int audioInputIndex = ui->audioInputCombo->findText(udpPrefs.audioInput); if (audioInputIndex != -1) { ui->audioInputCombo->setCurrentIndex(audioInputIndex); + udpPrefs.audioInputDevice = ui->audioInputCombo->itemData(audioInputIndex).toInt(); } udpPrefs.resampleQuality = settings.value("ResampleQuality", udpDefPrefs.resampleQuality).toInt(); @@ -3465,14 +3468,16 @@ void wfmain::on_passwordTxt_textChanged(QString text) udpPrefs.password = text; } -void wfmain::on_audioOutputCombo_currentIndexChanged(QString text) +void wfmain::on_audioOutputCombo_currentIndexChanged(int value) { - udpPrefs.audioOutput = text; + udpPrefs.audioOutput = ui->audioOutputCombo->currentText(); + udpPrefs.audioOutputDevice = ui->audioOutputCombo->itemData(value).toInt(); } -void wfmain::on_audioInputCombo_currentIndexChanged(QString text) +void wfmain::on_audioInputCombo_currentIndexChanged(int value) { - udpPrefs.audioInput = text; + udpPrefs.audioInput = ui->audioInputCombo->currentText(); + udpPrefs.audioInputDevice = ui->audioOutputCombo->itemData(value).toInt(); } void wfmain::on_audioSampleRateCombo_currentIndexChanged(QString text) diff --git a/wfmain.h b/wfmain.h index 796c605a..500c0b2d 100644 --- a/wfmain.h +++ b/wfmain.h @@ -344,9 +344,9 @@ private slots: void on_passwordTxt_textChanged(QString text); - void on_audioOutputCombo_currentIndexChanged(QString text); + void on_audioOutputCombo_currentIndexChanged(int value); - void on_audioInputCombo_currentIndexChanged(QString text); + void on_audioInputCombo_currentIndexChanged(int value); void on_toFixedBtn_clicked(); @@ -699,9 +699,9 @@ private: SERVERCONFIG serverConfig; - +#ifdef USE_RTAUDIO RtAudio audio; - +#endif }; Q_DECLARE_METATYPE(struct rigCapabilities) diff --git a/wfview.vcxproj b/wfview.vcxproj index 6a56163e..9255089f 100644 --- a/wfview.vcxproj +++ b/wfview.vcxproj @@ -374,6 +374,7 @@ + diff --git a/wfview.vcxproj.filters b/wfview.vcxproj.filters index afcab062..90b9e3c5 100644 --- a/wfview.vcxproj.filters +++ b/wfview.vcxproj.filters @@ -120,6 +120,9 @@ Source Files + + Source Files + From dee0dbd0a5a7c076761b4a62f318f8c36727e41a Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Mon, 17 May 2021 07:46:10 +0100 Subject: [PATCH 03/47] Quick fix for VS project --- wfview.vcxproj | 2 +- wfview.vcxproj.filters | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/wfview.vcxproj b/wfview.vcxproj index 9255089f..586dbc17 100644 --- a/wfview.vcxproj +++ b/wfview.vcxproj @@ -203,7 +203,7 @@ Sync release\ MaxSpeed - _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;NDEBUG;QT_NO_DEBUG;OUTSIDE_SPEEX;RANDOM_PREFIX=wf;%(PreprocessorDefinitions) + _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;NDEBUG;QT_NO_DEBUG;OUTSIDE_SPEEX;RANDOM_PREFIX=wf;__WINDOWS_WASAPI__;%(PreprocessorDefinitions) false diff --git a/wfview.vcxproj.filters b/wfview.vcxproj.filters index 90b9e3c5..021b0ad1 100644 --- a/wfview.vcxproj.filters +++ b/wfview.vcxproj.filters @@ -358,6 +358,7 @@ + From a2219585afb8519ec438ad6df6de579e8f81f079 Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Sun, 23 May 2021 16:09:41 +0100 Subject: [PATCH 04/47] Non working test --- audiohandler.cpp | 463 ++++++++++++----------------------------- audiohandler.h | 57 ++--- audiohandler_rt.cpp | 421 ------------------------------------- udphandler.h | 2 + udpserver.cpp | 4 +- wfmain.cpp | 50 ++--- wfmain.h | 4 +- wfview.vcxproj | 8 +- wfview.vcxproj.filters | 18 +- 9 files changed, 187 insertions(+), 840 deletions(-) delete mode 100644 audiohandler_rt.cpp diff --git a/audiohandler.cpp b/audiohandler.cpp index 4cb85d0e..415efa51 100644 --- a/audiohandler.cpp +++ b/audiohandler.cpp @@ -1,247 +1,130 @@ /* - This class handles both RX and TX audio, each is created as a seperate instance of the class - but as the setup/handling if output (RX) and input (TX) devices is so similar I have combined them. + This class handles both RX and TX audio, each is created as a seperate instance of the class + but as the setup/handling if output (RX) and input (TX) devices is so similar I have combined them. */ #include "audiohandler.h" -#ifndef USE_RTAUDIO #include "logcategories.h" -#define MULAW_BIAS 33 -#define MULAW_MAX 0x1fff audioHandler::audioHandler(QObject* parent) : - QIODevice(parent), - isInitialized(false), - audioOutput(Q_NULLPTR), - audioInput(Q_NULLPTR), - isUlaw(false), - audioLatency(0), - isInput(0), + isInitialized(false), + isUlaw(false), + audioLatency(0), + isInput(0), chunkAvailable(false) { } audioHandler::~audioHandler() { - stop(); - if (audioOutput != Q_NULLPTR) { - audioOutput->stop(); - delete audioOutput; - } - if (audioInput != Q_NULLPTR) { - audioInput->stop(); - delete audioInput; - } + //stop(); - if (resampler != NULL) { + if (resampler != Q_NULLPTR) { speex_resampler_destroy(resampler); } + if (audio.isStreamRunning()) + { + audio.stopStream(); + } + delete buf; } -bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 audioLatency, const bool ulaw, const bool isinput, int device, quint8 resampleQuality) +bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, int port, quint8 resampleQuality) { - if (isInitialized) { - return false; - } - - /* Always use 16 bit 48K samples internally*/ - format.setSampleSize(16); - format.setChannelCount(channels); - format.setSampleRate(INTERNAL_SAMPLE_RATE); - format.setCodec("audio/pcm"); - format.setByteOrder(QAudioFormat::LittleEndian); - format.setSampleType(QAudioFormat::SignedInt); + if (isInitialized) { + return false; + } this->audioLatency = latency; this->isUlaw = ulaw; - this->isInput = isinput; - this->radioSampleBits = bits; - this->radioSampleRate = samplerate; + this->isInput = isinput; + this->radioSampleBits = bits; + this->radioSampleRate = samplerate; this->radioChannels = channels; // chunk size is always relative to Internal Sample Rate. this->chunkSize = (INTERNAL_SAMPLE_RATE / 25) * radioChannels; - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "chunkSize: " << this->chunkSize; - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "bufferLength (latency): " << this->latency; + if (port > 0) { + aParams.deviceId = port; + } + else if (isInput) { + aParams.deviceId = audio.getDefaultInputDevice(); + } + else { + aParams.deviceId = audio.getDefaultOutputDevice(); + } + aParams.nChannels = channels; + aParams.firstChannel = 0; - int resample_error=0; + info = audio.getDeviceInfo(aParams.deviceId); + + buf = new quint16[960]; + + if (info.probed) + { + qInfo(logAudio()) << (isInput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") successfully probed"; + if (info.nativeFormats == 0) + qInfo(logAudio()) << " No natively supported data formats!"; + else { + qDebug(logAudio()) << " Supported formats:" << + (info.nativeFormats & RTAUDIO_SINT8 ? "8-bit int," : "") << + (info.nativeFormats & RTAUDIO_SINT16 ? "16-bit int," : "") << + (info.nativeFormats & RTAUDIO_SINT24 ? "24-bit int," : "") << + (info.nativeFormats & RTAUDIO_SINT32 ? "32-bit int," : "") << + (info.nativeFormats & RTAUDIO_FLOAT32 ? "32-bit float," : "") << + (info.nativeFormats & RTAUDIO_FLOAT64 ? "64-bit float," : ""); + qInfo(logAudio()) << " Preferred sample rate:" << info.preferredSampleRate; + } + + qInfo(logAudio()) << " chunkSize: " << chunkSize; + } + else + { + qCritical(logAudio()) << (isInput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") could not be probed, check audio configuration!"; + } + + int resample_error = 0; if (isInput) { resampler = wf_resampler_init(radioChannels, INTERNAL_SAMPLE_RATE, samplerate, resampleQuality, &resample_error); - - isInitialized = setDevice(port); - - if (!isInitialized) { - qInfo(logAudio()) << "Input device " << port.deviceName() << " not found, using default"; - isInitialized = setDevice(QAudioDeviceInfo::defaultInputDevice()); + try { + audio.openStream(NULL, &aParams, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &this->chunkSize, &staticWrite); + audio.startStream(); + } + catch (RtAudioError& e) { + qInfo(logAudio()) << "Error opening:" << QString::fromStdString(e.getMessage()); + return false; } } else { resampler = wf_resampler_init(radioChannels, samplerate, INTERNAL_SAMPLE_RATE, resampleQuality, &resample_error); - - isInitialized = setDevice(port); - - if (!isInitialized) { - qInfo(logAudio()) << "Output device " << deviceInfo.deviceName() << " not found, using default"; - isInitialized = setDevice(QAudioDeviceInfo::defaultOutputDevice()); + try { + unsigned int length = chunkSize / 2; + audio.openStream(&aParams, NULL, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &length, &staticRead); + audio.startStream(); + } + catch (RtAudioError& e) { + qInfo(logAudio()) << "Error opening:" << QString::fromStdString(e.getMessage()); + return false; } } - + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "device successfully opened"; wf_resampler_get_ratio(resampler, &ratioNum, &ratioDen); - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "wf_resampler_init() returned: " << resample_error << " ratioNum" << ratioNum << " ratioDen" << ratioDen; + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "wf_resampler_init() returned: " << resample_error << " ratioNum" << ratioNum << " ratioDen" << ratioDen; - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "audio port name: " << deviceInfo.deviceName(); - return isInitialized; + return isInitialized; } void audioHandler::setVolume(unsigned char volume) { - //qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << (qreal)(volume/255.0) << ")"; - - if (audioOutput != Q_NULLPTR) { - audioOutput->setVolume((qreal)(volume / 255.0)); - } - else if (audioInput != Q_NULLPTR) { - audioInput->setVolume((qreal)(volume / 255.0)); - } - + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << (qreal)(volume/255.0) << ")"; } -bool audioHandler::setDevice(QAudioDeviceInfo deviceInfo) -{ - bool ret = true; - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setDevice() running :" << deviceInfo.deviceName(); - if (!deviceInfo.isFormatSupported(format)) { - if (deviceInfo.isNull()) - { - qInfo(logAudio()) << "No audio device was found. You probably need to install libqt5multimedia-plugins."; - ret = false; - } - else { - /* - qInfo(logAudio()) << "Audio Devices found: "; - const auto deviceInfos = QAudioDeviceInfo::availableDevices(QAudio::AudioOutput); - for (const QAudioDeviceInfo& deviceInfo : deviceInfos) - { - qInfo(logAudio()) << "Device name: " << deviceInfo.deviceName(); - qInfo(logAudio()) << "is null (probably not good):" << deviceInfo.isNull(); - qInfo(logAudio()) << "channel count:" << deviceInfo.supportedChannelCounts(); - qInfo(logAudio()) << "byte order:" << deviceInfo.supportedByteOrders(); - qInfo(logAudio()) << "supported codecs:" << deviceInfo.supportedCodecs(); - qInfo(logAudio()) << "sample rates:" << deviceInfo.supportedSampleRates(); - qInfo(logAudio()) << "sample sizes:" << deviceInfo.supportedSampleSizes(); - qInfo(logAudio()) << "sample types:" << deviceInfo.supportedSampleTypes(); - } - qInfo(logAudio()) << "----- done with audio info -----"; - */ - ret=false; - } - - qInfo(logAudio()) << "Format not supported, choosing nearest supported format - which may not work!"; - deviceInfo.nearestFormat(format); - } - this->deviceInfo = deviceInfo; - this->reinit(); - return ret; -} - -void audioHandler::reinit() -{ - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "reinit() running"; - if (audioOutput != Q_NULLPTR && audioOutput->state() != QAudio::StoppedState) { - this->stop(); - } - - if (this->isInput) - { - // (Re)initialize audio input - if (audioInput != Q_NULLPTR) - delete audioInput; - audioInput = new QAudioInput(deviceInfo, format, this); - - connect(audioInput, SIGNAL(notify()), SLOT(notified())); - connect(audioInput, SIGNAL(stateChanged(QAudio::State)), SLOT(stateChanged(QAudio::State))); - - } - else { - // (Re)initialize audio output - if (audioOutput != Q_NULLPTR) - delete audioOutput; - audioOutput = Q_NULLPTR; - audioOutput = new QAudioOutput(deviceInfo, format, this); - - // This seems to only be needed on Linux but is critical in aligning buffer sizes. -//#ifdef Q_OS_MAC - audioOutput->setBufferSize(chunkSize*8); -//#else -// audioOutput->setBufferSize(chunkSize*4); -//#endif - - connect(audioOutput, SIGNAL(notify()), SLOT(notified())); - connect(audioOutput, SIGNAL(stateChanged(QAudio::State)), SLOT(stateChanged(QAudio::State))); - } - - this->start(); - this->flush(); -} - -void audioHandler::start() -{ - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "start() running"; - - if ((audioOutput == Q_NULLPTR || audioOutput->state() != QAudio::StoppedState) && - (audioInput == Q_NULLPTR || audioInput->state() != QAudio::StoppedState) ) { - return; - } - - if (isInput) { - //this->open(QIODevice::WriteOnly | QIODevice::Unbuffered); - this->open(QIODevice::WriteOnly); - audioInput->start(this); - } - else { - //this->open(QIODevice::ReadOnly | QIODevice::Unbuffered); - this->open(QIODevice::ReadOnly); - audioOutput->start(this); - } -} - - -void audioHandler::flush() -{ - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "flush() running"; - this->stop(); - if (isInput) { - audioInput->reset(); - } - else { - audioOutput->reset(); - } - - QMutexLocker locker(&mutex); - audioBuffer.clear(); - this->start(); -} - -void audioHandler::stop() -{ - if (audioOutput != Q_NULLPTR && audioOutput->state() != QAudio::StoppedState) { - // Stop audio output - audioOutput->stop(); - this->close(); - } - - if (audioInput != Q_NULLPTR && audioInput->state() != QAudio::StoppedState) { - // Stop audio output - audioInput->stop(); - this->close(); - } -} /// /// This function processes the incoming audio FROM the radio and pushes it into the playback buffer *data @@ -249,85 +132,34 @@ void audioHandler::stop() /// /// /// -qint64 audioHandler::readData(char* data, qint64 maxlen) +int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status) { - // Calculate output length, always full samples + // Calculate output length, always full samples int sentlen = 0; + qint16* buffer = (qint16*)outputBuffer; + qDebug(logAudio()) << "looking for: " << nFrames; - //qInfo(logAudio()) << "Looking for: " << maxlen << " bytes"; - - // We must lock the mutex for the entire time that the buffer may be modified. - // Get next packet from buffer. - if (!inputBuffer.isEmpty()) + for (int f = 0; f < nFrames; f++) { - - // Output buffer is ALWAYS 16 bit. - QMutexLocker locker(&mutex); - auto packet = inputBuffer.begin(); - - while (packet != inputBuffer.end() && sentlen < maxlen) - { - int timediff = packet->time.msecsTo(QTime::currentTime()); - - if (timediff > (int)audioLatency * 2) { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet->seq << - " arrived too late (increase output latency!) " << - dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; - - while (packet !=audioBuffer.end() && timediff > (int)audioLatency) { - timediff = packet->time.msecsTo(QTime::currentTime()); - lastSeq=packet->seq; - packet = inputBuffer.erase(packet); // returns next packet - } - if (packet == inputBuffer.end()) { - break; - } - } - - // If we got here then packet time must be within latency threshold - - if (packet->seq == lastSeq+1 || packet->seq <= lastSeq) - { - int send = qMin((int)maxlen-sentlen, packet->dataout.length() - packet->sent); - lastSeq = packet->seq; - //qInfo(logAudio()) << "Packet " << hex << packet->seq << " arrived on time " << Qt::dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; - - memcpy(data + sentlen, packet->dataout.constData() + packet->sent, send); - - sentlen = sentlen + send; - - if (send == packet->dataout.length() - packet->sent) - { - //qInfo(logAudio()) << "Get next packet"; - packet = inputBuffer.erase(packet); // returns next packet - } - else - { - // Store sent amount (could be zero if audioOutput buffer full) then break. - packet->sent = send; - break; - } - } else { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet->seq - 1; - lastSeq = packet->seq; - } - } - } - else { - // Fool audio system into thinking it has valid data, this seems to be required - // for MacOS Built in audio but shouldn't cause any issues with other platforms. - memset(data, 0x0, maxlen); - return maxlen; + qDebug(logAudio()) << "*"; + + *buffer++ = getBuffer(f); + //*buffer++ = f; } - return sentlen; + return 0; } -qint64 audioHandler::writeData(const char* data, qint64 len) +quint16 audioHandler::getBuffer(int i) { + return buf[i]; +} + +int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status) { - qint64 sentlen = 0; + int sentlen = 0; + /* QMutexLocker locker(&mutex); - audioPacket *current; + audioPacket* current; while (sentlen < len) { if (!audioBuffer.isEmpty()) @@ -345,37 +177,37 @@ qint64 audioHandler::writeData(const char* data, qint64 len) } current = &audioBuffer.last(); - int send = qMin((int)(len - sentlen), (int)chunkSize-current->sent); + int send = qMin((int)(len - sentlen), (int)chunkSize - current->sent); - current->datain.append(QByteArray::fromRawData(data + sentlen, send )); + current->datain.append(QByteArray::fromRawData(data + sentlen, send)); sentlen = sentlen + send; current->seq = 0; // Not used in TX current->time = QTime::currentTime(); current->sent = current->datain.length(); - + if (current->sent == chunkSize) { chunkAvailable = true; } - else if (audioBuffer.length()<=1 && current->sent != chunkSize) { + else if (audioBuffer.length() <= 1 && current->sent != chunkSize) { chunkAvailable = false; } - - } - return (sentlen); // Always return the same number as we received + } + */ + return (sentlen); // Always return the same number as we received } qint64 audioHandler::bytesAvailable() const { - return 0; + return 0; } bool audioHandler::isSequential() const { - return true; + return true; } void audioHandler::notified() @@ -385,46 +217,13 @@ void audioHandler::notified() void audioHandler::stateChanged(QAudio::State state) { - // Process the state - switch (state) - { - case QAudio::IdleState: - { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Audio now in idle state: " << audioBuffer.length() << " packets in buffer"; - if (audioOutput != Q_NULLPTR && audioOutput->error() == QAudio::UnderrunError) - { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "buffer underrun"; - audioOutput->suspend(); - } - break; - } - case QAudio::ActiveState: - { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Audio now in active state: " << audioBuffer.length() << " packets in buffer"; - break; - } - case QAudio::SuspendedState: - { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Audio now in suspended state: " << audioBuffer.length() << " packets in buffer"; - break; - } - case QAudio::StoppedState: - { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Audio now in stopped state: " << audioBuffer.length() << " packets in buffer"; - break; - } - default: { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Unhandled audio state: " << audioBuffer.length() << " packets in buffer"; - } - } + } void audioHandler::incomingAudio(audioPacket data) { - if (audioOutput != Q_NULLPTR && audioOutput->state() != QAudio::StoppedState) { - QMutexLocker locker(&mutex); // Incoming data is 8bits? if (radioSampleBits == 8) @@ -446,7 +245,7 @@ void audioHandler::incomingAudio(audioPacket data) data.datain = inPacket; // Replace incoming data with converted. } - //qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.datain.length(); + qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.datain.length(); /* We now have an array of 16bit samples in the NATIVE samplerate of the radio If the radio sample rate is below 48000, we need to resample. @@ -467,42 +266,37 @@ void audioHandler::incomingAudio(audioPacket data) err = wf_resampler_process_interleaved_int(resampler, (const qint16*)data.datain.constData(), &inFrames, (qint16*)data.dataout.data(), &outFrames); } if (err) { - qInfo(logAudio()) <<(isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; } } else { - data.dataout = data.datain; + data.dataout = data.datain; } - inputBuffer.insert(data.seq, data); - - // Restart playback - if (audioOutput->state() == QAudio::SuspendedState) - { - qInfo(logAudio()) << "Output Audio Suspended, Resuming..."; - audioOutput->resume(); - } - } + memcpy(buf, data.dataout.constData(), data.dataout.length()); + qDebug(logAudio()) << "Got data: " << data.dataout.length(); + //audioBuffer.insert({ data.seq, data }); } void audioHandler::changeLatency(const quint16 newSize) { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Changing latency to: " << newSize << " from " << audioLatency; - audioLatency = newSize; + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Changing latency to: " << newSize << " from " << audioLatency; + audioLatency = newSize; } void audioHandler::getLatency() { - emit sendLatency(audioLatency); + emit sendLatency(audioLatency); } bool audioHandler::isChunkAvailable() { - return (chunkAvailable); + return (chunkAvailable); } void audioHandler::getNextAudioChunk(QByteArray& ret) { + /* if (!audioBuffer.isEmpty() && chunkAvailable) { @@ -512,18 +306,18 @@ void audioHandler::getNextAudioChunk(QByteArray& ret) auto packet = audioBuffer.begin(); while (packet != audioBuffer.end()) { - if (packet->time.msecsTo(QTime::currentTime()) > latency) { - //qInfo(logAudio()) << "TX Packet too old " << dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; + if (packet->time.msecsTo(QTime::currentTime()) > 100) { + //qInfo(logAudio()) << "TX Packet too old " << dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; packet = audioBuffer.erase(packet); // returns next packet } else { if (packet->datain.length() == chunkSize && ret.length() == 0) { - /* We now have an array of samples in the computer native format (48000) - If the radio sample rate is below 48000, we need to resample. - */ + // We now have an array of samples in the computer native format (48000) + // If the radio sample rate is below 48000, we need to resample. + - if (ratioNum != 1) + if (ratioNum != 1) { // We need to resample (we are STILL 16 bit!) quint32 outFrames = ((packet->datain.length() / 2) / ratioNum) / radioChannels; @@ -546,11 +340,11 @@ void audioHandler::getNextAudioChunk(QByteArray& ret) //qInfo(logAudio()) << "Resampler run inLen:" << packet->datain.length() << " outLen:" << packet->dataout.length(); if (radioSampleBits == 8) { - packet->datain=packet->dataout; // Copy output packet back to input buffer. + packet->datain = packet->dataout; // Copy output packet back to input buffer. packet->dataout.clear(); // Buffer MUST be cleared ready to be re-filled by the upsampling below. } } - else if (radioSampleBits == 16 ){ + else if (radioSampleBits == 16) { // Only copy buffer if radioSampleBits is 16, as it will be handled below otherwise. packet->dataout = packet->datain; } @@ -575,7 +369,7 @@ void audioHandler::getNextAudioChunk(QByteArray& ret) packet->dataout[f] = (char)outdata; } } - ret=packet->dataout; + ret = packet->dataout; packet = audioBuffer.erase(packet); // returns next packet } else { @@ -584,8 +378,9 @@ void audioHandler::getNextAudioChunk(QByteArray& ret) } } } - return; + */ + return; } -#endif + diff --git a/audiohandler.h b/audiohandler.h index ef0ee9e9..714b390a 100644 --- a/audiohandler.h +++ b/audiohandler.h @@ -1,32 +1,17 @@ #ifndef AUDIOHANDLER_H #define AUDIOHANDLER_H -#define USE_RTAUDIO -#undef USE_QTAUDIO - #include #include -#include #include -#include -#ifdef USE_QTAUDIO -#include -#include -#include -#include -#include -#endif - -#ifdef USE_RTAUDIO #include "rtaudio/RtAudio.h" typedef signed short MY_TYPE; #define FORMAT RTAUDIO_SINT16 #define SCALE 32767.0 -#endif #include #include @@ -774,12 +759,10 @@ struct audioPacket { QByteArray dataout; }; - -class audioHandler : public QIODevice +class audioHandler : public QObject { Q_OBJECT - public: audioHandler(QObject* parent = 0); ~audioHandler(); @@ -792,17 +775,27 @@ public: void flush(); void stop(); - qint64 readData(char* data, qint64 maxlen); - qint64 writeData(const char* data, qint64 len); + int readData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status); + + static int staticRead(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void* userData) { + return static_cast(userData)->readData(outputBuffer, inputBuffer, nFrames, streamTime, status); + } + + int writeData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status); + + static int staticWrite(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void* userData) { + return static_cast(userData)->writeData(outputBuffer, inputBuffer, nFrames, streamTime, status); + } + qint64 bytesAvailable() const; bool isSequential() const; void getNextAudioChunk(QByteArray &data); bool isChunkAvailable(); + +private slots: bool init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isulaw, const bool isinput, int port, quint8 resampleQuality); void incomingAudio(const audioPacket data); void changeLatency(const quint16 newSize); - -private slots: void notified(); void stateChanged(QAudio::State state); void setVolume(unsigned char volume); @@ -815,28 +808,22 @@ signals: private: void reinit(); + quint16 getBuffer(int); - QMutex mutex; + std::mutex mutex; bool isInitialized; -#ifdef USE_RTAUDIO RtAudio audio; int audioDevice = 0; RtAudio::StreamParameters aParams; RtAudio::DeviceInfo info; -#elif defined(USE_QTAUDIO) - QAudioOutput* audioOutput; - QAudioInput* audioInput; - QAudioFormat format; - QAudioDeviceInfo deviceInfo; -#endif - SpeexResamplerState* resampler; + SpeexResamplerState* resampler = Q_NULLPTR; bool isUlaw; quint16 audioLatency; bool isInput; // Used to determine whether input or output audio - int chunkSize; + unsigned int chunkSize; bool chunkAvailable; quint32 lastSeq; @@ -844,10 +831,10 @@ private: quint16 radioSampleRate; quint8 radioSampleBits; quint8 radioChannels; - QVector audioBuffer; - QMapinputBuffer; - SpeexResamplerState* resampler=NULL; + quint16 *buf; + + std::mapaudioBuffer; unsigned int ratioNum; unsigned int ratioDen; diff --git a/audiohandler_rt.cpp b/audiohandler_rt.cpp deleted file mode 100644 index e4a44d19..00000000 --- a/audiohandler_rt.cpp +++ /dev/null @@ -1,421 +0,0 @@ -/* - This class handles both RX and TX audio, each is created as a seperate instance of the class - but as the setup/handling if output (RX) and input (TX) devices is so similar I have combined them. -*/ -#include "audiohandler.h" - -#include "logcategories.h" - - -audioHandler::audioHandler(QObject* parent) : - QIODevice(parent), - isInitialized(false), - isUlaw(false), - audioLatency(0), - isInput(0), - chunkAvailable(false) -{ -} - -audioHandler::~audioHandler() -{ - //stop(); - - if (resampler) { - speex_resampler_destroy(resampler); - } -} - -bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, QString port, int device, quint8 resampleQuality) -{ - if (isInitialized) { - return false; - } - - this->audioDevice = device; - this->audioLatency = latency; - this->isUlaw = ulaw; - this->isInput = isinput; - this->radioSampleBits = bits; - this->radioSampleRate = samplerate; - this->radioChannels = channels; - - // chunk size is always relative to Internal Sample Rate. - this->chunkSize = (INTERNAL_SAMPLE_RATE / 25) * radioChannels; - - - if (device != 0) { - aParams.deviceId = device; - } - else if (isInput) { - aParams.deviceId = audio.getDefaultInputDevice(); - } - else { - aParams.deviceId = audio.getDefaultOutputDevice(); - } - - info = audio.getDeviceInfo(aParams.deviceId); - - if (info.probed) - { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") successfully probed"; - if (info.nativeFormats == 0) - qInfo(logAudio()) << " No natively supported data formats!"; - else { - qDebug(logAudio()) << " Supported formats:" << - (info.nativeFormats & RTAUDIO_SINT8 ? "8-bit int," : "") << - (info.nativeFormats & RTAUDIO_SINT16 ? "16-bit int," : "") << - (info.nativeFormats & RTAUDIO_SINT24 ? "24-bit int," : "") << - (info.nativeFormats & RTAUDIO_SINT32 ? "32-bit int," : "") << - (info.nativeFormats & RTAUDIO_FLOAT32 ? "32-bit float," : "") << - (info.nativeFormats & RTAUDIO_FLOAT64 ? "64-bit float," : ""); - qInfo(logAudio()) << " Preferred sample rate:" << info.preferredSampleRate; - } - - qInfo(logAudio()) << " chunkSize: " << chunkSize; - } - else - { - qCritical(logAudio()) << (isInput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") could not be probed, check audio configuration!"; - } - - - int resample_error = 0; - - if (isInput) { - resampler = wf_resampler_init(radioChannels, INTERNAL_SAMPLE_RATE, samplerate, resampleQuality, &resample_error); - //audio.openStream(&aParams, NULL, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &this->chunkSize, &output, (void*)&data); - } - else - { - resampler = wf_resampler_init(radioChannels, samplerate, INTERNAL_SAMPLE_RATE, resampleQuality, &resample_error); - //audio.openStream(&aParams, NULL, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &this->chunkSize, &output, (void*)&data); - } - - wf_resampler_get_ratio(resampler, &ratioNum, &ratioDen); - qInfo(logAudio()) << " wf_resampler_init() returned: " << resample_error << " ratioNum" << ratioNum << " ratioDen" << ratioDen; - - return isInitialized; -} - -void audioHandler::setVolume(unsigned char volume) -{ - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << (qreal)(volume/255.0) << ")"; -} - - - -/// -/// This function processes the incoming audio FROM the radio and pushes it into the playback buffer *data -/// -/// -/// -/// -qint64 audioHandler::readData(char* data, qint64 maxlen) -{ - // Calculate output length, always full samples - int sentlen = 0; - - //qInfo(logAudio()) << "Looking for: " << maxlen << " bytes"; - - // We must lock the mutex for the entire time that the buffer may be modified. - // Get next packet from buffer. - if (!audioBuffer.isEmpty()) - { - - // Output buffer is ALWAYS 16 bit. - QMutexLocker locker(&mutex); - auto packet = audioBuffer.begin(); - - while (packet != audioBuffer.end() && sentlen < maxlen) - { - int timediff = packet->time.msecsTo(QTime::currentTime()); - - if (timediff > (int)audioLatency * 2) { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet->seq << - " arrived too late (increase output latency!) " << - dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; - while (packet != audioBuffer.end() && timediff > (int)audioLatency) { - timediff = packet->time.msecsTo(QTime::currentTime()); - lastSeq = packet->seq; - packet = audioBuffer.erase(packet); // returns next packet - } - if (packet == audioBuffer.end()) { - break; - } - } - - // If we got here then packet time must be within latency threshold - - if (packet->seq == lastSeq + 1 || packet->seq <= lastSeq) - { - int send = qMin((int)maxlen - sentlen, packet->dataout.length() - packet->sent); - lastSeq = packet->seq; - //qInfo(logAudio()) << "Packet " << hex << packet->seq << " arrived on time " << Qt::dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; - - memcpy(data + sentlen, packet->dataout.constData() + packet->sent, send); - - sentlen = sentlen + send; - - if (send == packet->dataout.length() - packet->sent) - { - //qInfo(logAudio()) << "Get next packet"; - packet = audioBuffer.erase(packet); // returns next packet - } - else - { - // Store sent amount (could be zero if audioOutput buffer full) then break. - packet->sent = send; - break; - } - } - else { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet->seq - 1; - lastSeq = packet->seq; - } - } - } - else { - // Fool audio system into thinking it has valid data, this seems to be required - // for MacOS Built in audio but shouldn't cause any issues with other platforms. - memset(data, 0x0, maxlen); - return maxlen; - } - - return sentlen; -} - -qint64 audioHandler::writeData(const char* data, qint64 len) -{ - qint64 sentlen = 0; - QMutexLocker locker(&mutex); - audioPacket* current; - - while (sentlen < len) { - if (!audioBuffer.isEmpty()) - { - if (audioBuffer.last().sent == chunkSize) - { - audioBuffer.append(audioPacket()); - audioBuffer.last().sent = 0; - } - } - else - { - audioBuffer.append(audioPacket()); - audioBuffer.last().sent = 0; - } - current = &audioBuffer.last(); - - int send = qMin((int)(len - sentlen), (int)chunkSize - current->sent); - - current->datain.append(QByteArray::fromRawData(data + sentlen, send)); - - sentlen = sentlen + send; - - current->seq = 0; // Not used in TX - current->time = QTime::currentTime(); - current->sent = current->datain.length(); - - if (current->sent == chunkSize) - { - chunkAvailable = true; - } - else if (audioBuffer.length() <= 1 && current->sent != chunkSize) { - chunkAvailable = false; - } - - } - - return (sentlen); // Always return the same number as we received -} - -qint64 audioHandler::bytesAvailable() const -{ - return 0; -} - -bool audioHandler::isSequential() const -{ - return true; -} - -void audioHandler::notified() -{ -} - - -void audioHandler::stateChanged(QAudio::State state) -{ - -} - - - -void audioHandler::incomingAudio(audioPacket data) -{ - QMutexLocker locker(&mutex); - - // Incoming data is 8bits? - if (radioSampleBits == 8) - { - QByteArray inPacket((int)data.datain.length() * 2, (char)0xff); - qint16* in = (qint16*)inPacket.data(); - for (int f = 0; f < data.datain.length(); f++) - { - if (isUlaw) - { - in[f] = ulaw_decode[(quint8)data.datain[f]]; - } - else - { - // Convert 8-bit sample to 16-bit - in[f] = (qint16)(((quint8)data.datain[f] << 8) - 32640); - } - } - data.datain = inPacket; // Replace incoming data with converted. - } - - //qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.datain.length(); - - /* We now have an array of 16bit samples in the NATIVE samplerate of the radio - If the radio sample rate is below 48000, we need to resample. - */ - - if (ratioDen != 1) { - - // We need to resample - quint32 outFrames = ((data.datain.length() / 2) * ratioDen) / radioChannels; - quint32 inFrames = (data.datain.length() / 2) / radioChannels; - data.dataout.resize(outFrames * 2 * radioChannels); // Preset the output buffer size. - - int err = 0; - if (this->radioChannels == 1) { - err = wf_resampler_process_int(resampler, 0, (const qint16*)data.datain.constData(), &inFrames, (qint16*)data.dataout.data(), &outFrames); - } - else { - err = wf_resampler_process_interleaved_int(resampler, (const qint16*)data.datain.constData(), &inFrames, (qint16*)data.dataout.data(), &outFrames); - } - if (err) { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; - } - } - else { - data.dataout = data.datain; - } - - audioBuffer.push_back(data); - - // Sort the buffer by seq number. This is important and audio packets may have arrived out-of-order - std::sort(audioBuffer.begin(), audioBuffer.end(), - [](const audioPacket& a, const audioPacket& b) -> bool - { - return a.seq < b.seq; - }); - -} - -void audioHandler::changeLatency(const quint16 newSize) -{ - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Changing latency to: " << newSize << " from " << audioLatency; - audioLatency = newSize; -} - -void audioHandler::getLatency() -{ - emit sendLatency(audioLatency); -} - -bool audioHandler::isChunkAvailable() -{ - return (chunkAvailable); -} - -void audioHandler::getNextAudioChunk(QByteArray& ret) -{ - if (!audioBuffer.isEmpty() && chunkAvailable) - { - - QMutexLocker locker(&mutex); - - // Skip through audio buffer deleting any old entry. - auto packet = audioBuffer.begin(); - while (packet != audioBuffer.end()) - { - if (packet->time.msecsTo(QTime::currentTime()) > 100) { - //qInfo(logAudio()) << "TX Packet too old " << dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; - packet = audioBuffer.erase(packet); // returns next packet - } - else { - if (packet->datain.length() == chunkSize && ret.length() == 0) - { - /* We now have an array of samples in the computer native format (48000) - If the radio sample rate is below 48000, we need to resample. - */ - - if (ratioNum != 1) - { - // We need to resample (we are STILL 16 bit!) - quint32 outFrames = ((packet->datain.length() / 2) / ratioNum) / radioChannels; - quint32 inFrames = (packet->datain.length() / 2) / radioChannels; - packet->dataout.resize(outFrames * 2 * radioChannels); // Preset the output buffer size. - - const qint16* in = (qint16*)packet->datain.constData(); - qint16* out = (qint16*)packet->dataout.data(); - int err = 0; - if (this->radioChannels == 1) { - err = wf_resampler_process_int(resampler, 0, in, &inFrames, out, &outFrames); - } - else { - err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames); - } - if (err) { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; - } - //qInfo(logAudio()) << "Resampler run " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; - //qInfo(logAudio()) << "Resampler run inLen:" << packet->datain.length() << " outLen:" << packet->dataout.length(); - if (radioSampleBits == 8) - { - packet->datain = packet->dataout; // Copy output packet back to input buffer. - packet->dataout.clear(); // Buffer MUST be cleared ready to be re-filled by the upsampling below. - } - } - else if (radioSampleBits == 16) { - // Only copy buffer if radioSampleBits is 16, as it will be handled below otherwise. - packet->dataout = packet->datain; - } - - // Do we need to convert 16-bit to 8-bit? - if (radioSampleBits == 8) { - packet->dataout.resize(packet->datain.length() / 2); - qint16* in = (qint16*)packet->datain.data(); - for (int f = 0; f < packet->dataout.length(); f++) - { - quint8 outdata = 0; - if (isUlaw) { - qint16 enc = qFromLittleEndian(in + f); - if (enc >= 0) - outdata = ulaw_encode[enc]; - else - outdata = 0x7f & ulaw_encode[-enc]; - } - else { - outdata = (quint8)(((qFromLittleEndian(in + f) >> 8) ^ 0x80) & 0xff); - } - packet->dataout[f] = (char)outdata; - } - } - ret = packet->dataout; - packet = audioBuffer.erase(packet); // returns next packet - } - else { - packet++; - } - } - } - } - return; -} - - - diff --git a/udphandler.h b/udphandler.h index 98a79340..59e7328a 100644 --- a/udphandler.h +++ b/udphandler.h @@ -43,6 +43,8 @@ struct udpPreferences { QString password; int audioOutput; int audioInput; + QString audioOutputName; + QString audioInputName; quint16 audioRXLatency; quint16 audioTXLatency; diff --git a/udpserver.cpp b/udpserver.cpp index 4fbed160..d392c7e3 100644 --- a/udpserver.cpp +++ b/udpserver.cpp @@ -395,7 +395,7 @@ void udpServer::controlReceived() connect(this, SIGNAL(setupTxAudio(quint8,quint8,quint16,quint16,bool,bool,int,quint8)), txaudio, SLOT(init(quint8,quint8,quint16,quint16,bool,bool,int,quint8))); connect(txAudioThread, SIGNAL(finished()), txaudio, SLOT(deleteLater())); - emit setupTxAudio(samples, channels, current->txSampleRate, in->txbuffer, uLaw, false, config.audioOutput,config.audioOutputDevice, config.resampleQuality); + emit setupTxAudio(samples, channels, current->txSampleRate, in->txbuffer, uLaw, false, config.audioOutput, config.resampleQuality); hasTxAudio=datagram.senderAddress(); connect(this, SIGNAL(haveAudioData(audioPacket)), txaudio, SLOT(incomingAudio(audioPacket))); @@ -428,7 +428,7 @@ void udpServer::controlReceived() connect(this, SIGNAL(setupRxAudio(quint8,quint8,quint16,quint16,bool,bool,QString,int,quint8)), rxaudio, SLOT(init(quint8,quint8,quint16,quint16,bool,bool,int,quint8))); connect(rxAudioThread, SIGNAL(finished()), txaudio, SLOT(deleteLater())); - emit setupRxAudio(samples, channels, current->rxSampleRate, 150, uLaw, true, config.audioInput, config.audioInputDevice, config.resampleQuality); + emit setupRxAudio(samples, channels, current->rxSampleRate, 150, uLaw, true, config.audioInput, config.resampleQuality); rxAudioTimer = new QTimer(); rxAudioTimer->setTimerType(Qt::PreciseTimer); diff --git a/wfmain.cpp b/wfmain.cpp index 6ef432df..89e98be4 100644 --- a/wfmain.cpp +++ b/wfmain.cpp @@ -169,7 +169,6 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s // Enumerate audio devices, need to do before settings are loaded. -#ifdef USE_RTAUDIO std::map apiMap; apiMap[RtAudio::MACOSX_CORE] = "OS-X Core Audio"; apiMap[RtAudio::WINDOWS_ASIO] = "Windows ASIO"; @@ -197,7 +196,7 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s unsigned int devices = audio.getDeviceCount(); qInfo(logAudio()) << "Found " << devices << " audio device(s) ..."; - for (unsigned int i = 0; i < devices; i++) { + for (unsigned int i = 1; i < devices; i++) { info = audio.getDeviceInfo(i); if (info.outputChannels > 0) { qInfo(logAudio()) << (info.isDefaultOutput ? "*" : " ") << "(" << i << ") Output Device : " << QString::fromStdString(info.name); @@ -297,8 +296,8 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s serverConfig.audioInput = udpPrefs.audioInput; serverConfig.audioOutput = udpPrefs.audioOutput; serverConfig.baudRate = prefs.serialPortBaud; - serverConfig.inputDevice = udpPrefs.inputDevice; - serverConfig.outputDevice = udpPrefs.outputDevice; + serverConfig.audioInput = udpPrefs.audioInput; + serverConfig.audioOutput = udpPrefs.audioOutput; } udp = new udpServer(serverConfig); @@ -952,15 +951,8 @@ void wfmain::setDefPrefs() udpDefPrefs.audioLANPort = 50003; udpDefPrefs.username = QString(""); udpDefPrefs.password = QString(""); -#if defined(USE_RTAUDIO) - udpDefPrefs.audioOutput = QString::fromStdString(audio.getDeviceInfo(audio.getDefaultOutputDevice()).name); - udpDefPrefs.audioInput = QString::fromStdString(audio.getDeviceInfo(audio.getDefaultInputDevice()).name); -#elif defined(USE_QTAUDIO) - udpDefPrefs.audioOutput = QAudioDeviceInfo::defaultOutputDevice().deviceName(); - udpDefPrefs.audioInput = QAudioDeviceInfo::defaultInputDevice().deviceName(); -#endif - udpDefPrefs.audioInputDevice = 0; - udpDefPrefs.audioOutputDevice = 0; + udpDefPrefs.audioOutput = 0; + udpDefPrefs.audioInput = 0; udpDefPrefs.audioRXLatency = 150; udpDefPrefs.audioTXLatency = 150; udpDefPrefs.audioRXSampleRate = 48000; @@ -1123,7 +1115,7 @@ void wfmain::loadSettings() ui->audioSampleRateCombo->setEnabled(ui->lanEnableBtn->isChecked()); int audioSampleRateIndex = ui->audioSampleRateCombo->findText(QString::number(udpDefPrefs.audioRXSampleRate)); if (audioSampleRateIndex != -1) { - ui->audioOutputCombo->setCurrentIndex(audioSampleRateIndex); + ui->audioSampleRateCombo->setCurrentIndex(audioSampleRateIndex); } // Add codec combobox items here so that we can add userdata! @@ -1150,22 +1142,20 @@ void wfmain::loadSettings() if (ui->audioTXCodecCombo->itemData(f).toInt() == udpPrefs.audioTXCodec) ui->audioTXCodecCombo->setCurrentIndex(f); - udpPrefs.audioOutput = settings->value("AudioOutput", udpDefPrefs.audioOutput).toString(); - qInfo(logGui()) << "Got Audio Output: " << udpPrefs.audioOutput; - //ui->audioOutputCombo->setEnabled(ui->lanEnableBtn->isChecked()); - int audioOutputIndex = ui->audioOutputCombo->findText(udpPrefs.audioOutput); + udpPrefs.audioOutputName = settings->value("AudioOutput", udpDefPrefs.audioOutputName).toString(); + qInfo(logGui()) << "Got Audio Output: " << udpPrefs.audioOutputName; + int audioOutputIndex = ui->audioOutputCombo->findText(udpPrefs.audioOutputName); if (audioOutputIndex != -1) { ui->audioOutputCombo->setCurrentIndex(audioOutputIndex); - udpPrefs.audioOutputDevice = ui->audioOutputCombo->itemData(audioOutputIndex).toInt(); + udpPrefs.audioOutput = ui->audioOutputCombo->itemData(audioOutputIndex).toInt(); } - udpPrefs.audioInput = settings->value("AudioInput", udpDefPrefs.audioInput).toString(); - qInfo(logGui()) << "Got Audio Input: " << udpPrefs.audioInput; - //ui->audioInputCombo->setEnabled(ui->lanEnableBtn->isChecked()); - int audioInputIndex = ui->audioInputCombo->findText(udpPrefs.audioInput); + udpPrefs.audioInputName = settings->value("AudioInput", udpDefPrefs.audioInputName).toString(); + qInfo(logGui()) << "Got Audio Input: " << udpPrefs.audioInputName; + int audioInputIndex = ui->audioInputCombo->findText(udpPrefs.audioInputName); if (audioInputIndex != -1) { ui->audioInputCombo->setCurrentIndex(audioInputIndex); - udpPrefs.audioInputDevice = ui->audioInputCombo->itemData(audioInputIndex).toInt(); + udpPrefs.audioInput = ui->audioInputCombo->itemData(audioInputIndex).toInt(); } udpPrefs.resampleQuality = settings->value("ResampleQuality", udpDefPrefs.resampleQuality).toInt(); @@ -1279,8 +1269,8 @@ void wfmain::saveSettings() settings->setValue("AudioRXCodec", udpPrefs.audioRXCodec); settings->setValue("AudioTXSampleRate", udpPrefs.audioRXSampleRate); settings->setValue("AudioTXCodec", udpPrefs.audioTXCodec); - settings->setValue("AudioOutput", udpPrefs.audioOutput); - settings->setValue("AudioInput", udpPrefs.audioInput); + settings->setValue("AudioOutput", udpPrefs.audioOutputName); + settings->setValue("AudioInput", udpPrefs.audioInputName); settings->setValue("ResampleQuality", udpPrefs.resampleQuality); settings->setValue("ClientName", udpPrefs.clientName); settings->endGroup(); @@ -3577,14 +3567,14 @@ void wfmain::on_passwordTxt_textChanged(QString text) void wfmain::on_audioOutputCombo_currentIndexChanged(int value) { - udpPrefs.audioOutput = ui->audioOutputCombo->currentText(); - udpPrefs.audioOutputDevice = ui->audioOutputCombo->itemData(value).toInt(); + udpPrefs.audioOutput = ui->audioOutputCombo->itemData(value).toInt(); + udpPrefs.audioOutputName = ui->audioOutputCombo->itemText(value); } void wfmain::on_audioInputCombo_currentIndexChanged(int value) { - udpPrefs.audioInput = ui->audioInputCombo->currentText(); - udpPrefs.audioInputDevice = ui->audioOutputCombo->itemData(value).toInt(); + udpPrefs.audioInput = ui->audioInputCombo->itemData(value).toInt(); + udpPrefs.audioInputName = ui->audioInputCombo->itemText(value); } void wfmain::on_audioSampleRateCombo_currentIndexChanged(QString text) diff --git a/wfmain.h b/wfmain.h index 23e74d31..de091e45 100644 --- a/wfmain.h +++ b/wfmain.h @@ -26,7 +26,7 @@ #include "qledlabel.h" #include "rigctld.h" -#include +#include <../qcustomplot/qcustomplot.h> #include namespace Ui { @@ -705,9 +705,7 @@ private: SERVERCONFIG serverConfig; -#ifdef USE_RTAUDIO RtAudio audio; -#endif }; Q_DECLARE_METATYPE(struct rigCapabilities) diff --git a/wfview.vcxproj b/wfview.vcxproj index 586dbc17..36bfb877 100644 --- a/wfview.vcxproj +++ b/wfview.vcxproj @@ -374,7 +374,8 @@ - + + @@ -382,14 +383,12 @@ - - @@ -398,6 +397,7 @@ + @@ -409,8 +409,6 @@ - - diff --git a/wfview.vcxproj.filters b/wfview.vcxproj.filters index 021b0ad1..f16a5ae5 100644 --- a/wfview.vcxproj.filters +++ b/wfview.vcxproj.filters @@ -69,9 +69,6 @@ Source Files - - Source Files - Source Files @@ -84,9 +81,6 @@ Source Files - - Source Files - Source Files @@ -120,7 +114,10 @@ Source Files - + + Source Files + + Source Files @@ -134,9 +131,6 @@ Header Files - - Header Files - Header Files @@ -179,6 +173,9 @@ Header Files + + Header Files + @@ -359,6 +356,7 @@ + From 45f5fdf3445a5e88d1c5ca07aad8b16326508183 Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Sun, 23 May 2021 22:45:10 +0100 Subject: [PATCH 05/47] First working rtaudio (output only) --- audiohandler.cpp | 84 ++++++++++++++++++++++++++++++++++++++++-------- audiohandler.h | 2 +- 2 files changed, 72 insertions(+), 14 deletions(-) diff --git a/audiohandler.cpp b/audiohandler.cpp index 415efa51..031d70aa 100644 --- a/audiohandler.cpp +++ b/audiohandler.cpp @@ -90,7 +90,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 if (isInput) { resampler = wf_resampler_init(radioChannels, INTERNAL_SAMPLE_RATE, samplerate, resampleQuality, &resample_error); try { - audio.openStream(NULL, &aParams, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &this->chunkSize, &staticWrite); + audio.openStream(NULL, &aParams, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &this->chunkSize, &staticWrite, this); audio.startStream(); } catch (RtAudioError& e) { @@ -103,7 +103,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 resampler = wf_resampler_init(radioChannels, samplerate, INTERNAL_SAMPLE_RATE, resampleQuality, &resample_error); try { unsigned int length = chunkSize / 2; - audio.openStream(&aParams, NULL, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &length, &staticRead); + audio.openStream(&aParams, NULL, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &length, &staticRead, this); audio.startStream(); } catch (RtAudioError& e) { @@ -137,15 +137,69 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n // Calculate output length, always full samples int sentlen = 0; qint16* buffer = (qint16*)outputBuffer; - qDebug(logAudio()) << "looking for: " << nFrames; + //qDebug(logAudio()) << "looking for: " << nFrames << this->audioBuffer.size(); - for (int f = 0; f < nFrames; f++) + + if (!audioBuffer.isEmpty()) { - qDebug(logAudio()) << "*"; - - *buffer++ = getBuffer(f); - //*buffer++ = f; + + // Output buffer is ALWAYS 16 bit. + auto packet = audioBuffer.begin(); + + while (packet != audioBuffer.end() && sentlen < nFrames/2) + { + int timediff = packet->time.msecsTo(QTime::currentTime()); + + if (timediff > (int)audioLatency * 2) { + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet->seq << + " arrived too late (increase output latency!) " << + dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; + while (packet != audioBuffer.end() && timediff > (int)audioLatency) { + timediff = packet->time.msecsTo(QTime::currentTime()); + lastSeq = packet->seq; + packet = audioBuffer.erase(packet); // returns next packet + } + if (packet == audioBuffer.end()) { + break; + } + } + + // If we got here then packet time must be within latency threshold + + if (packet->seq == lastSeq + 1 || packet->seq <= lastSeq) + { + int send = qMin((int)nFrames*2 - sentlen, packet->dataout.length() - packet->sent); + lastSeq = packet->seq; + //qInfo(logAudio()) << "Packet " << hex << packet->seq << " arrived on time " << Qt::dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; + + memcpy(buffer + sentlen, packet->dataout.constData() + packet->sent, send); + + sentlen = sentlen + send; + + if (send == packet->dataout.length() - packet->sent) + { + //qInfo(logAudio()) << "Get next packet"; + packet = audioBuffer.erase(packet); // returns next packet + } + else + { + // Store sent amount (could be zero if audioOutput buffer full) then break. + packet->sent = send; + break; + } + } + else { + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet->seq - 1; + lastSeq = packet->seq; + } + } } + else { + // Fool audio system into thinking it has valid data, this seems to be required + // for MacOS Built in audio but shouldn't cause any issues with other platforms. + return 0; + } + return 0; } @@ -224,7 +278,11 @@ void audioHandler::stateChanged(QAudio::State state) void audioHandler::incomingAudio(audioPacket data) { - + // No point buffering audio until stream is actually running. + if (!audio.isStreamRunning()) + { + return; + } // Incoming data is 8bits? if (radioSampleBits == 8) { @@ -245,7 +303,7 @@ void audioHandler::incomingAudio(audioPacket data) data.datain = inPacket; // Replace incoming data with converted. } - qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.datain.length(); + //qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.datain.length(); /* We now have an array of 16bit samples in the NATIVE samplerate of the radio If the radio sample rate is below 48000, we need to resample. @@ -273,9 +331,9 @@ void audioHandler::incomingAudio(audioPacket data) data.dataout = data.datain; } - memcpy(buf, data.dataout.constData(), data.dataout.length()); - qDebug(logAudio()) << "Got data: " << data.dataout.length(); - //audioBuffer.insert({ data.seq, data }); + //memcpy(buf, data.dataout.constData(), data.dataout.length()); + //qDebug(logAudio()) << "Got data: " << data.dataout.length(); + audioBuffer.insert( data.seq, data ); } void audioHandler::changeLatency(const quint16 newSize) diff --git a/audiohandler.h b/audiohandler.h index 714b390a..6f3ffb94 100644 --- a/audiohandler.h +++ b/audiohandler.h @@ -834,7 +834,7 @@ private: quint16 *buf; - std::mapaudioBuffer; + QMapaudioBuffer; unsigned int ratioNum; unsigned int ratioDen; From fd23f36977d46721db5a65816d0bf105780a4ec5 Mon Sep 17 00:00:00 2001 From: M0VSE Date: Sun, 23 May 2021 23:04:35 +0100 Subject: [PATCH 06/47] Fixes for linux build --- audiohandler.h | 2 -- wfmain.h | 2 +- wfview.pro | 20 ++++++++++---------- 3 files changed, 11 insertions(+), 13 deletions(-) diff --git a/audiohandler.h b/audiohandler.h index 6f3ffb94..1b3d3d9c 100644 --- a/audiohandler.h +++ b/audiohandler.h @@ -810,8 +810,6 @@ private: void reinit(); quint16 getBuffer(int); - std::mutex mutex; - bool isInitialized; RtAudio audio; int audioDevice = 0; diff --git a/wfmain.h b/wfmain.h index de091e45..44e996a5 100644 --- a/wfmain.h +++ b/wfmain.h @@ -26,7 +26,7 @@ #include "qledlabel.h" #include "rigctld.h" -#include <../qcustomplot/qcustomplot.h> +#include #include namespace Ui { diff --git a/wfview.pro b/wfview.pro index a01190f3..fe9be6b4 100644 --- a/wfview.pro +++ b/wfview.pro @@ -41,14 +41,14 @@ linux:DEFINES += __LINUX_ALSA__ #linux:DEFINES += __LINUX_PULSE__ macx:DEFINES += __MACOSX_CORE__ -option(RTAUDIO_API_DS "Build DirectSound API" OFF) -option(RTAUDIO_API_ASIO "Build ASIO API" OFF) -option(RTAUDIO_API_WASAPI "Build WASAPI API" ${WIN32}) -option(RTAUDIO_API_OSS "Build OSS4 API" ${xBSD}) -option(RTAUDIO_API_ALSA "Build ALSA API" ${LINUX}) -option(RTAUDIO_API_PULSE "Build PulseAudio API" ${pulse_FOUND}) -option(RTAUDIO_API_JACK "Build JACK audio server API" ${HAVE_JACK}) -option(RTAUDIO_API_CORE "Build CoreAudio API" ${APPLE}) +#option(RTAUDIO_API_DS "Build DirectSound API" OFF) +#option(RTAUDIO_API_ASIO "Build ASIO API" OFF) +#option(RTAUDIO_API_WASAPI "Build WASAPI API" ${WIN32}) +#option(RTAUDIO_API_OSS "Build OSS4 API" ${xBSD}) +#option(RTAUDIO_API_ALSA "Build ALSA API" ${LINUX}) +#option(RTAUDIO_API_PULSE "Build PulseAudio API" ${pulse_FOUND}) +#option(RTAUDIO_API_JACK "Build JACK audio server API" ${HAVE_JACK}) +#option(RTAUDIO_API_CORE "Build CoreAudio API" ${APPLE}) macx:INCLUDEPATH += /usr/local/include /opt/local/include macx:LIBS += -L/usr/local/lib -L/opt/local/lib @@ -121,7 +121,7 @@ SOURCES += main.cpp\ resampler/resample.c \ repeatersetup.cpp \ rigctld.cpp \ - rtaudio/RTAudio.cpp + rtaudio/RtAudio.cpp HEADERS += wfmain.h \ commhandler.h \ @@ -145,7 +145,7 @@ HEADERS += wfmain.h \ repeatersetup.h \ repeaterattributes.h \ rigctld.h \ - rtaudio/RTAudio.h + rtaudio/RtAudio.h FORMS += wfmain.ui \ From bf838076f5132efb0581a129003e6653b15fd346 Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Mon, 24 May 2021 09:27:18 +0100 Subject: [PATCH 07/47] Add mutex for audio buffer --- audiohandler.cpp | 15 ++++++--------- audiohandler.h | 4 +++- wfview.vcxproj | 2 +- wfview.vcxproj.filters | 1 + 4 files changed, 11 insertions(+), 11 deletions(-) diff --git a/audiohandler.cpp b/audiohandler.cpp index 031d70aa..3997fdb3 100644 --- a/audiohandler.cpp +++ b/audiohandler.cpp @@ -138,11 +138,13 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n int sentlen = 0; qint16* buffer = (qint16*)outputBuffer; //qDebug(logAudio()) << "looking for: " << nFrames << this->audioBuffer.size(); + if (status == RTAUDIO_OUTPUT_UNDERFLOW) + qDebug(logAudio()) << "Underflow detected"; if (!audioBuffer.isEmpty()) { - + mutex.lock(); // Output buffer is ALWAYS 16 bit. auto packet = audioBuffer.begin(); @@ -193,13 +195,8 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n lastSeq = packet->seq; } } + mutex.unlock(); } - else { - // Fool audio system into thinking it has valid data, this seems to be required - // for MacOS Built in audio but shouldn't cause any issues with other platforms. - return 0; - } - return 0; } @@ -331,9 +328,9 @@ void audioHandler::incomingAudio(audioPacket data) data.dataout = data.datain; } - //memcpy(buf, data.dataout.constData(), data.dataout.length()); - //qDebug(logAudio()) << "Got data: " << data.dataout.length(); + mutex.lock(); audioBuffer.insert( data.seq, data ); + mutex.unlock(); } void audioHandler::changeLatency(const quint16 newSize) diff --git a/audiohandler.h b/audiohandler.h index 1b3d3d9c..54bb0442 100644 --- a/audiohandler.h +++ b/audiohandler.h @@ -5,7 +5,7 @@ #include #include - +#include #include "rtaudio/RtAudio.h" typedef signed short MY_TYPE; @@ -836,6 +836,8 @@ private: unsigned int ratioNum; unsigned int ratioDen; + QMutex mutex; + volatile bool lock=false; }; #endif // AUDIOHANDLER_H diff --git a/wfview.vcxproj b/wfview.vcxproj index 36bfb877..816a3a0b 100644 --- a/wfview.vcxproj +++ b/wfview.vcxproj @@ -95,7 +95,7 @@ wfview true - $(IncludePath) + C:\Users\Phil\source\repos\qcustomplot;$(IncludePath) $(LibraryPath) diff --git a/wfview.vcxproj.filters b/wfview.vcxproj.filters index f16a5ae5..94c1d1e9 100644 --- a/wfview.vcxproj.filters +++ b/wfview.vcxproj.filters @@ -357,6 +357,7 @@ + From 940f68cfe74188a0cc9edf994df928fd11a6403e Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Mon, 24 May 2021 16:28:01 +0100 Subject: [PATCH 08/47] add opus-tools as submodule --- .gitmodules | 3 + opus-tools | 1 + rtaudio/ChangeLog | 10 - rtaudio/LICENSE | 27 - rtaudio/README.md | 38 - rtaudio/RtAudio.cpp | 10911 ------------------------------------------ rtaudio/RtAudio.h | 1185 ----- 7 files changed, 4 insertions(+), 12171 deletions(-) create mode 100644 .gitmodules create mode 160000 opus-tools delete mode 100644 rtaudio/ChangeLog delete mode 100644 rtaudio/LICENSE delete mode 100644 rtaudio/README.md delete mode 100644 rtaudio/RtAudio.cpp delete mode 100644 rtaudio/RtAudio.h diff --git a/.gitmodules b/.gitmodules new file mode 100644 index 00000000..d9385a3f --- /dev/null +++ b/.gitmodules @@ -0,0 +1,3 @@ +[submodule "opus-tools"] + path = opus-tools + url = https://github.com/xiph/opus-tools.git diff --git a/opus-tools b/opus-tools new file mode 160000 index 00000000..ae5d6d59 --- /dev/null +++ b/opus-tools @@ -0,0 +1 @@ +Subproject commit ae5d6d59e82ef40300a4dece7897499685f87184 diff --git a/rtaudio/ChangeLog b/rtaudio/ChangeLog deleted file mode 100644 index 6282a761..00000000 --- a/rtaudio/ChangeLog +++ /dev/null @@ -1,10 +0,0 @@ -30.04.2018 - * Additions and fixes for realtime operation: (Tim E. Real of MusE) - - Added realtime operation to Pulse driver. - - Fixed ALSA realtime support. Attributes are once again all set - in probeDeviceOpen(). - - Fixed OSS realtime support. Same mods as done to ALSA driver. - OSS untested, but should work, it's the same code. - - A diagnostic message (streamed to cerr) in each of the callback - handlers informs the user if realtime is really running. - diff --git a/rtaudio/LICENSE b/rtaudio/LICENSE deleted file mode 100644 index c55dbfa2..00000000 --- a/rtaudio/LICENSE +++ /dev/null @@ -1,27 +0,0 @@ - -RtAudio: a set of realtime audio i/o C++ classes -Copyright (c) 2001-2019 Gary P. Scavone - -Permission is hereby granted, free of charge, to any person -obtaining a copy of this software and associated documentation files -(the "Software"), to deal in the Software without restriction, -including without limitation the rights to use, copy, modify, merge, -publish, distribute, sublicense, and/or sell copies of the Software, -and to permit persons to whom the Software is furnished to do so, -subject to the following conditions: - -The above copyright notice and this permission notice shall be -included in all copies or substantial portions of the Software. - -Any person wishing to distribute modifications to the Software is -asked to send the modifications to the original developer so that -they can be incorporated into the canonical version. This is, -however, not a binding provision of this license. - -THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, -EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF -MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. -IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR -ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF -CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION -WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. diff --git a/rtaudio/README.md b/rtaudio/README.md deleted file mode 100644 index d58883b9..00000000 --- a/rtaudio/README.md +++ /dev/null @@ -1,38 +0,0 @@ -# RtAudio - -[![Build Status](https://travis-ci.org/thestk/rtaudio.svg?branch=master)](https://travis-ci.org/thestk/rtaudio) - -A set of C++ classes that provide a common API for realtime audio input/output across Linux (native ALSA, JACK, PulseAudio and OSS), Macintosh OS X (CoreAudio and JACK), and Windows (DirectSound, ASIO and WASAPI) operating systems. - -By Gary P. Scavone, 2001-2019 (and many other developers!) - -This distribution of RtAudio contains the following: - -- doc: RtAudio documentation (see doc/html/index.html) -- tests: example RtAudio programs -- include: header and source files necessary for ASIO, DS & OSS compilation -- tests/Windows: Visual C++ .net test program workspace and projects - -## Overview - -RtAudio is a set of C++ classes that provides a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, JACK, PulseAudio and OSS), Macintosh OS X and Windows (DirectSound, ASIO and WASAPI) operating systems. RtAudio significantly simplifies the process of interacting with computer audio hardware. It was designed with the following objectives: - - - object-oriented C++ design - - simple, common API across all supported platforms - - only one source and one header file for easy inclusion in programming projects - - allow simultaneous multi-api support - - support dynamic connection of devices - - provide extensive audio device parameter control - - allow audio device capability probing - - automatic internal conversion for data format, channel number compensation, (de)interleaving, and byte-swapping - -RtAudio incorporates the concept of audio streams, which represent audio output (playback) and/or input (recording). Available audio devices and their capabilities can be enumerated and then specified when opening a stream. Where applicable, multiple API support can be compiled and a particular API specified when creating an RtAudio instance. See the \ref apinotes section for information specific to each of the supported audio APIs. - -## Further Reading - -For complete documentation on RtAudio, see the doc directory of the distribution or surf to http://www.music.mcgill.ca/~gary/rtaudio/. - - -## Legal and ethical: - -The RtAudio license is similar to the MIT License. Please see [LICENSE](LICENSE). diff --git a/rtaudio/RtAudio.cpp b/rtaudio/RtAudio.cpp deleted file mode 100644 index 3e64b85f..00000000 --- a/rtaudio/RtAudio.cpp +++ /dev/null @@ -1,10911 +0,0 @@ -/************************************************************************/ -/*! \class RtAudio - \brief Realtime audio i/o C++ classes. - - RtAudio provides a common API (Application Programming Interface) - for realtime audio input/output across Linux (native ALSA, Jack, - and OSS), Macintosh OS X (CoreAudio and Jack), and Windows - (DirectSound, ASIO and WASAPI) operating systems. - - RtAudio GitHub site: https://github.com/thestk/rtaudio - RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ - - RtAudio: realtime audio i/o C++ classes - Copyright (c) 2001-2019 Gary P. Scavone - - Permission is hereby granted, free of charge, to any person - obtaining a copy of this software and associated documentation files - (the "Software"), to deal in the Software without restriction, - including without limitation the rights to use, copy, modify, merge, - publish, distribute, sublicense, and/or sell copies of the Software, - and to permit persons to whom the Software is furnished to do so, - subject to the following conditions: - - The above copyright notice and this permission notice shall be - included in all copies or substantial portions of the Software. - - Any person wishing to distribute modifications to the Software is - asked to send the modifications to the original developer so that - they can be incorporated into the canonical version. This is, - however, not a binding provision of this license. - - THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. -*/ -/************************************************************************/ - -// RtAudio: Version 5.1.0 - -#include "RtAudio.h" -#include -#include -#include -#include -#include -#include - -// Static variable definitions. -const unsigned int RtApi::MAX_SAMPLE_RATES = 14; -const unsigned int RtApi::SAMPLE_RATES[] = { - 4000, 5512, 8000, 9600, 11025, 16000, 22050, - 32000, 44100, 48000, 88200, 96000, 176400, 192000 -}; - -#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) - #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) - #define MUTEX_DESTROY(A) DeleteCriticalSection(A) - #define MUTEX_LOCK(A) EnterCriticalSection(A) - #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) - - #include "tchar.h" - - template inline - std::string convertCharPointerToStdString(const T *text); - - template<> inline - std::string convertCharPointerToStdString(const char *text) - { - return std::string(text); - } - - template<> inline - std::string convertCharPointerToStdString(const wchar_t *text) - { - int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL); - std::string s( length-1, '\0' ); - WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL); - return s; - } - -#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) - // pthread API - #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) - #define MUTEX_DESTROY(A) pthread_mutex_destroy(A) - #define MUTEX_LOCK(A) pthread_mutex_lock(A) - #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A) -#else - #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions - #define MUTEX_DESTROY(A) abs(*A) // dummy definitions -#endif - -// *************************************************** // -// -// RtAudio definitions. -// -// *************************************************** // - -std::string RtAudio :: getVersion( void ) -{ - return RTAUDIO_VERSION; -} - -// Define API names and display names. -// Must be in same order as API enum. -extern "C" { -const char* rtaudio_api_names[][2] = { - { "unspecified" , "Unknown" }, - { "alsa" , "ALSA" }, - { "pulse" , "Pulse" }, - { "oss" , "OpenSoundSystem" }, - { "jack" , "Jack" }, - { "core" , "CoreAudio" }, - { "wasapi" , "WASAPI" }, - { "asio" , "ASIO" }, - { "ds" , "DirectSound" }, - { "dummy" , "Dummy" }, -}; -const unsigned int rtaudio_num_api_names = - sizeof(rtaudio_api_names)/sizeof(rtaudio_api_names[0]); - -// The order here will control the order of RtAudio's API search in -// the constructor. -extern "C" const RtAudio::Api rtaudio_compiled_apis[] = { -#if defined(__UNIX_JACK__) - RtAudio::UNIX_JACK, -#endif -#if defined(__LINUX_PULSE__) - RtAudio::LINUX_PULSE, -#endif -#if defined(__LINUX_ALSA__) - RtAudio::LINUX_ALSA, -#endif -#if defined(__LINUX_OSS__) - RtAudio::LINUX_OSS, -#endif -#if defined(__WINDOWS_ASIO__) - RtAudio::WINDOWS_ASIO, -#endif -#if defined(__WINDOWS_WASAPI__) - RtAudio::WINDOWS_WASAPI, -#endif -#if defined(__WINDOWS_DS__) - RtAudio::WINDOWS_DS, -#endif -#if defined(__MACOSX_CORE__) - RtAudio::MACOSX_CORE, -#endif -#if defined(__RTAUDIO_DUMMY__) - RtAudio::RTAUDIO_DUMMY, -#endif - RtAudio::UNSPECIFIED, -}; -extern "C" const unsigned int rtaudio_num_compiled_apis = - sizeof(rtaudio_compiled_apis)/sizeof(rtaudio_compiled_apis[0])-1; -} - -// This is a compile-time check that rtaudio_num_api_names == RtAudio::NUM_APIS. -// If the build breaks here, check that they match. -template class StaticAssert { private: StaticAssert() {} }; -template<> class StaticAssert{ public: StaticAssert() {} }; -class StaticAssertions { StaticAssertions() { - StaticAssert(); -}}; - -void RtAudio :: getCompiledApi( std::vector &apis ) -{ - apis = std::vector(rtaudio_compiled_apis, - rtaudio_compiled_apis + rtaudio_num_compiled_apis); -} - -std::string RtAudio :: getApiName( RtAudio::Api api ) -{ - if (api < 0 || api >= RtAudio::NUM_APIS) - return ""; - return rtaudio_api_names[api][0]; -} - -std::string RtAudio :: getApiDisplayName( RtAudio::Api api ) -{ - if (api < 0 || api >= RtAudio::NUM_APIS) - return "Unknown"; - return rtaudio_api_names[api][1]; -} - -RtAudio::Api RtAudio :: getCompiledApiByName( const std::string &name ) -{ - unsigned int i=0; - for (i = 0; i < rtaudio_num_compiled_apis; ++i) - if (name == rtaudio_api_names[rtaudio_compiled_apis[i]][0]) - return rtaudio_compiled_apis[i]; - return RtAudio::UNSPECIFIED; -} - -void RtAudio :: openRtApi( RtAudio::Api api ) -{ - if ( rtapi_ ) - delete rtapi_; - rtapi_ = 0; - -#if defined(__UNIX_JACK__) - if ( api == UNIX_JACK ) - rtapi_ = new RtApiJack(); -#endif -#if defined(__LINUX_ALSA__) - if ( api == LINUX_ALSA ) - rtapi_ = new RtApiAlsa(); -#endif -#if defined(__LINUX_PULSE__) - if ( api == LINUX_PULSE ) - rtapi_ = new RtApiPulse(); -#endif -#if defined(__LINUX_OSS__) - if ( api == LINUX_OSS ) - rtapi_ = new RtApiOss(); -#endif -#if defined(__WINDOWS_ASIO__) - if ( api == WINDOWS_ASIO ) - rtapi_ = new RtApiAsio(); -#endif -#if defined(__WINDOWS_WASAPI__) - if ( api == WINDOWS_WASAPI ) - rtapi_ = new RtApiWasapi(); -#endif -#if defined(__WINDOWS_DS__) - if ( api == WINDOWS_DS ) - rtapi_ = new RtApiDs(); -#endif -#if defined(__MACOSX_CORE__) - if ( api == MACOSX_CORE ) - rtapi_ = new RtApiCore(); -#endif -#if defined(__RTAUDIO_DUMMY__) - if ( api == RTAUDIO_DUMMY ) - rtapi_ = new RtApiDummy(); -#endif -} - -RtAudio :: RtAudio( RtAudio::Api api ) -{ - rtapi_ = 0; - - if ( api != UNSPECIFIED ) { - // Attempt to open the specified API. - openRtApi( api ); - if ( rtapi_ ) return; - - // No compiled support for specified API value. Issue a debug - // warning and continue as if no API was specified. - std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl; - } - - // Iterate through the compiled APIs and return as soon as we find - // one with at least one device or we reach the end of the list. - std::vector< RtAudio::Api > apis; - getCompiledApi( apis ); - for ( unsigned int i=0; igetDeviceCount() ) break; - } - - if ( rtapi_ ) return; - - // It should not be possible to get here because the preprocessor - // definition __RTAUDIO_DUMMY__ is automatically defined if no - // API-specific definitions are passed to the compiler. But just in - // case something weird happens, we'll thow an error. - std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n"; - throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) ); -} - -RtAudio :: ~RtAudio() -{ - if ( rtapi_ ) - delete rtapi_; -} - -void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, - RtAudio::StreamParameters *inputParameters, - RtAudioFormat format, unsigned int sampleRate, - unsigned int *bufferFrames, - RtAudioCallback callback, void *userData, - RtAudio::StreamOptions *options, - RtAudioErrorCallback errorCallback ) -{ - return rtapi_->openStream( outputParameters, inputParameters, format, - sampleRate, bufferFrames, callback, - userData, options, errorCallback ); -} - -// *************************************************** // -// -// Public RtApi definitions (see end of file for -// private or protected utility functions). -// -// *************************************************** // - -RtApi :: RtApi() -{ - stream_.state = STREAM_CLOSED; - stream_.mode = UNINITIALIZED; - stream_.apiHandle = 0; - stream_.userBuffer[0] = 0; - stream_.userBuffer[1] = 0; - MUTEX_INITIALIZE( &stream_.mutex ); - showWarnings_ = true; - firstErrorOccurred_ = false; -} - -RtApi :: ~RtApi() -{ - MUTEX_DESTROY( &stream_.mutex ); -} - -void RtApi :: openStream( RtAudio::StreamParameters *oParams, - RtAudio::StreamParameters *iParams, - RtAudioFormat format, unsigned int sampleRate, - unsigned int *bufferFrames, - RtAudioCallback callback, void *userData, - RtAudio::StreamOptions *options, - RtAudioErrorCallback errorCallback ) -{ - if ( stream_.state != STREAM_CLOSED ) { - errorText_ = "RtApi::openStream: a stream is already open!"; - error( RtAudioError::INVALID_USE ); - return; - } - - // Clear stream information potentially left from a previously open stream. - clearStreamInfo(); - - if ( oParams && oParams->nChannels < 1 ) { - errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one."; - error( RtAudioError::INVALID_USE ); - return; - } - - if ( iParams && iParams->nChannels < 1 ) { - errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one."; - error( RtAudioError::INVALID_USE ); - return; - } - - if ( oParams == NULL && iParams == NULL ) { - errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!"; - error( RtAudioError::INVALID_USE ); - return; - } - - if ( formatBytes(format) == 0 ) { - errorText_ = "RtApi::openStream: 'format' parameter value is undefined."; - error( RtAudioError::INVALID_USE ); - return; - } - - unsigned int nDevices = getDeviceCount(); - unsigned int oChannels = 0; - if ( oParams ) { - oChannels = oParams->nChannels; - if ( oParams->deviceId >= nDevices ) { - errorText_ = "RtApi::openStream: output device parameter value is invalid."; - error( RtAudioError::INVALID_USE ); - return; - } - } - - unsigned int iChannels = 0; - if ( iParams ) { - iChannels = iParams->nChannels; - if ( iParams->deviceId >= nDevices ) { - errorText_ = "RtApi::openStream: input device parameter value is invalid."; - error( RtAudioError::INVALID_USE ); - return; - } - } - - bool result; - - if ( oChannels > 0 ) { - - result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel, - sampleRate, format, bufferFrames, options ); - if ( result == false ) { - error( RtAudioError::SYSTEM_ERROR ); - return; - } - } - - if ( iChannels > 0 ) { - - result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel, - sampleRate, format, bufferFrames, options ); - if ( result == false ) { - if ( oChannels > 0 ) closeStream(); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - } - - stream_.callbackInfo.callback = (void *) callback; - stream_.callbackInfo.userData = userData; - stream_.callbackInfo.errorCallback = (void *) errorCallback; - - if ( options ) options->numberOfBuffers = stream_.nBuffers; - stream_.state = STREAM_STOPPED; -} - -unsigned int RtApi :: getDefaultInputDevice( void ) -{ - // Should be implemented in subclasses if possible. - return 0; -} - -unsigned int RtApi :: getDefaultOutputDevice( void ) -{ - // Should be implemented in subclasses if possible. - return 0; -} - -void RtApi :: closeStream( void ) -{ - // MUST be implemented in subclasses! - return; -} - -bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/, - unsigned int /*firstChannel*/, unsigned int /*sampleRate*/, - RtAudioFormat /*format*/, unsigned int * /*bufferSize*/, - RtAudio::StreamOptions * /*options*/ ) -{ - // MUST be implemented in subclasses! - return FAILURE; -} - -void RtApi :: tickStreamTime( void ) -{ - // Subclasses that do not provide their own implementation of - // getStreamTime should call this function once per buffer I/O to - // provide basic stream time support. - - stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate ); - -#if defined( HAVE_GETTIMEOFDAY ) - gettimeofday( &stream_.lastTickTimestamp, NULL ); -#endif -} - -long RtApi :: getStreamLatency( void ) -{ - verifyStream(); - - long totalLatency = 0; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - totalLatency = stream_.latency[0]; - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) - totalLatency += stream_.latency[1]; - - return totalLatency; -} - -double RtApi :: getStreamTime( void ) -{ - verifyStream(); - -#if defined( HAVE_GETTIMEOFDAY ) - // Return a very accurate estimate of the stream time by - // adding in the elapsed time since the last tick. - struct timeval then; - struct timeval now; - - if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 ) - return stream_.streamTime; - - gettimeofday( &now, NULL ); - then = stream_.lastTickTimestamp; - return stream_.streamTime + - ((now.tv_sec + 0.000001 * now.tv_usec) - - (then.tv_sec + 0.000001 * then.tv_usec)); -#else - return stream_.streamTime; -#endif -} - -void RtApi :: setStreamTime( double time ) -{ - verifyStream(); - - if ( time >= 0.0 ) - stream_.streamTime = time; -#if defined( HAVE_GETTIMEOFDAY ) - gettimeofday( &stream_.lastTickTimestamp, NULL ); -#endif -} - -unsigned int RtApi :: getStreamSampleRate( void ) -{ - verifyStream(); - - return stream_.sampleRate; -} - - -// *************************************************** // -// -// OS/API-specific methods. -// -// *************************************************** // - -#if defined(__MACOSX_CORE__) - -#include - -// The OS X CoreAudio API is designed to use a separate callback -// procedure for each of its audio devices. A single RtAudio duplex -// stream using two different devices is supported here, though it -// cannot be guaranteed to always behave correctly because we cannot -// synchronize these two callbacks. -// -// A property listener is installed for over/underrun information. -// However, no functionality is currently provided to allow property -// listeners to trigger user handlers because it is unclear what could -// be done if a critical stream parameter (buffer size, sample rate, -// device disconnect) notification arrived. The listeners entail -// quite a bit of extra code and most likely, a user program wouldn't -// be prepared for the result anyway. However, we do provide a flag -// to the client callback function to inform of an over/underrun. - -// A structure to hold various information related to the CoreAudio API -// implementation. -struct CoreHandle { - AudioDeviceID id[2]; // device ids -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - AudioDeviceIOProcID procId[2]; -#endif - UInt32 iStream[2]; // device stream index (or first if using multiple) - UInt32 nStreams[2]; // number of streams to use - bool xrun[2]; - char *deviceBuffer; - pthread_cond_t condition; - int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - - CoreHandle() - :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } -}; - -RtApiCore:: RtApiCore() -{ -#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER ) - // This is a largely undocumented but absolutely necessary - // requirement starting with OS-X 10.6. If not called, queries and - // updates to various audio device properties are not handled - // correctly. - CFRunLoopRef theRunLoop = NULL; - AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop); - if ( result != noErr ) { - errorText_ = "RtApiCore::RtApiCore: error setting run loop property!"; - error( RtAudioError::WARNING ); - } -#endif -} - -RtApiCore :: ~RtApiCore() -{ - // The subclass destructor gets called before the base class - // destructor, so close an existing stream before deallocating - // apiDeviceId memory. - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiCore :: getDeviceCount( void ) -{ - // Find out how many audio devices there are, if any. - UInt32 dataSize; - AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; - OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize ); - if ( result != noErr ) { - errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!"; - error( RtAudioError::WARNING ); - return 0; - } - - return dataSize / sizeof( AudioDeviceID ); -} - -unsigned int RtApiCore :: getDefaultInputDevice( void ) -{ - unsigned int nDevices = getDeviceCount(); - if ( nDevices <= 1 ) return 0; - - AudioDeviceID id; - UInt32 dataSize = sizeof( AudioDeviceID ); - AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; - OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id ); - if ( result != noErr ) { - errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device."; - error( RtAudioError::WARNING ); - return 0; - } - - dataSize *= nDevices; - AudioDeviceID deviceList[ nDevices ]; - property.mSelector = kAudioHardwarePropertyDevices; - result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList ); - if ( result != noErr ) { - errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs."; - error( RtAudioError::WARNING ); - return 0; - } - - for ( unsigned int i=0; i= nDevices ) { - errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - AudioDeviceID deviceList[ nDevices ]; - UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; - AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, - 0, NULL, &dataSize, (void *) &deviceList ); - if ( result != noErr ) { - errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs."; - error( RtAudioError::WARNING ); - return info; - } - - AudioDeviceID id = deviceList[ device ]; - - // Get the device name. - info.name.erase(); - CFStringRef cfname; - dataSize = sizeof( CFStringRef ); - property.mSelector = kAudioObjectPropertyManufacturer; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); - int length = CFStringGetLength(cfname); - char *mname = (char *)malloc(length * 3 + 1); -#if defined( UNICODE ) || defined( _UNICODE ) - CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8); -#else - CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding()); -#endif - info.name.append( (const char *)mname, strlen(mname) ); - info.name.append( ": " ); - CFRelease( cfname ); - free(mname); - - property.mSelector = kAudioObjectPropertyName; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); - length = CFStringGetLength(cfname); - char *name = (char *)malloc(length * 3 + 1); -#if defined( UNICODE ) || defined( _UNICODE ) - CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8); -#else - CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding()); -#endif - info.name.append( (const char *)name, strlen(name) ); - CFRelease( cfname ); - free(name); - - // Get the output stream "configuration". - AudioBufferList *bufferList = nil; - property.mSelector = kAudioDevicePropertyStreamConfiguration; - property.mScope = kAudioDevicePropertyScopeOutput; - // property.mElement = kAudioObjectPropertyElementWildcard; - dataSize = 0; - result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); - if ( result != noErr || dataSize == 0 ) { - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Allocate the AudioBufferList. - bufferList = (AudioBufferList *) malloc( dataSize ); - if ( bufferList == NULL ) { - errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList."; - error( RtAudioError::WARNING ); - return info; - } - - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); - if ( result != noErr || dataSize == 0 ) { - free( bufferList ); - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Get output channel information. - unsigned int i, nStreams = bufferList->mNumberBuffers; - for ( i=0; imBuffers[i].mNumberChannels; - free( bufferList ); - - // Get the input stream "configuration". - property.mScope = kAudioDevicePropertyScopeInput; - result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); - if ( result != noErr || dataSize == 0 ) { - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Allocate the AudioBufferList. - bufferList = (AudioBufferList *) malloc( dataSize ); - if ( bufferList == NULL ) { - errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList."; - error( RtAudioError::WARNING ); - return info; - } - - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); - if (result != noErr || dataSize == 0) { - free( bufferList ); - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Get input channel information. - nStreams = bufferList->mNumberBuffers; - for ( i=0; imBuffers[i].mNumberChannels; - free( bufferList ); - - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // Probe the device sample rates. - bool isInput = false; - if ( info.outputChannels == 0 ) isInput = true; - - // Determine the supported sample rates. - property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates; - if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput; - result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); - if ( result != kAudioHardwareNoError || dataSize == 0 ) { - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - UInt32 nRanges = dataSize / sizeof( AudioValueRange ); - AudioValueRange rangeList[ nRanges ]; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList ); - if ( result != kAudioHardwareNoError ) { - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // The sample rate reporting mechanism is a bit of a mystery. It - // seems that it can either return individual rates or a range of - // rates. I assume that if the min / max range values are the same, - // then that represents a single supported rate and if the min / max - // range values are different, the device supports an arbitrary - // range of values (though there might be multiple ranges, so we'll - // use the most conservative range). - Float64 minimumRate = 1.0, maximumRate = 10000000000.0; - bool haveValueRange = false; - info.sampleRates.clear(); - for ( UInt32 i=0; i info.preferredSampleRate ) ) - info.preferredSampleRate = tmpSr; - - } else { - haveValueRange = true; - if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum; - if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum; - } - } - - if ( haveValueRange ) { - for ( unsigned int k=0; k= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) { - info.sampleRates.push_back( SAMPLE_RATES[k] ); - - if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) - info.preferredSampleRate = SAMPLE_RATES[k]; - } - } - } - - // Sort and remove any redundant values - std::sort( info.sampleRates.begin(), info.sampleRates.end() ); - info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() ); - - if ( info.sampleRates.size() == 0 ) { - errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // CoreAudio always uses 32-bit floating point data for PCM streams. - // Thus, any other "physical" formats supported by the device are of - // no interest to the client. - info.nativeFormats = RTAUDIO_FLOAT32; - - if ( info.outputChannels > 0 ) - if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; - if ( info.inputChannels > 0 ) - if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; - - info.probed = true; - return info; -} - -static OSStatus callbackHandler( AudioDeviceID inDevice, - const AudioTimeStamp* /*inNow*/, - const AudioBufferList* inInputData, - const AudioTimeStamp* /*inInputTime*/, - AudioBufferList* outOutputData, - const AudioTimeStamp* /*inOutputTime*/, - void* infoPointer ) -{ - CallbackInfo *info = (CallbackInfo *) infoPointer; - - RtApiCore *object = (RtApiCore *) info->object; - if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false ) - return kAudioHardwareUnspecifiedError; - else - return kAudioHardwareNoError; -} - -static OSStatus xrunListener( AudioObjectID /*inDevice*/, - UInt32 nAddresses, - const AudioObjectPropertyAddress properties[], - void* handlePointer ) -{ - CoreHandle *handle = (CoreHandle *) handlePointer; - for ( UInt32 i=0; ixrun[1] = true; - else - handle->xrun[0] = true; - } - } - - return kAudioHardwareNoError; -} - -static OSStatus rateListener( AudioObjectID inDevice, - UInt32 /*nAddresses*/, - const AudioObjectPropertyAddress /*properties*/[], - void* ratePointer ) -{ - Float64 *rate = (Float64 *) ratePointer; - UInt32 dataSize = sizeof( Float64 ); - AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate ); - return kAudioHardwareNoError; -} - -bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - // Get device ID - unsigned int nDevices = getDeviceCount(); - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiCore::probeDeviceOpen: no devices found!"; - return FAILURE; - } - - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - AudioDeviceID deviceList[ nDevices ]; - UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; - AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, - 0, NULL, &dataSize, (void *) &deviceList ); - if ( result != noErr ) { - errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs."; - return FAILURE; - } - - AudioDeviceID id = deviceList[ device ]; - - // Setup for stream mode. - bool isInput = false; - if ( mode == INPUT ) { - isInput = true; - property.mScope = kAudioDevicePropertyScopeInput; - } - else - property.mScope = kAudioDevicePropertyScopeOutput; - - // Get the stream "configuration". - AudioBufferList *bufferList = nil; - dataSize = 0; - property.mSelector = kAudioDevicePropertyStreamConfiguration; - result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); - if ( result != noErr || dataSize == 0 ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Allocate the AudioBufferList. - bufferList = (AudioBufferList *) malloc( dataSize ); - if ( bufferList == NULL ) { - errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList."; - return FAILURE; - } - - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); - if (result != noErr || dataSize == 0) { - free( bufferList ); - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Search for one or more streams that contain the desired number of - // channels. CoreAudio devices can have an arbitrary number of - // streams and each stream can have an arbitrary number of channels. - // For each stream, a single buffer of interleaved samples is - // provided. RtAudio prefers the use of one stream of interleaved - // data or multiple consecutive single-channel streams. However, we - // now support multiple consecutive multi-channel streams of - // interleaved data as well. - UInt32 iStream, offsetCounter = firstChannel; - UInt32 nStreams = bufferList->mNumberBuffers; - bool monoMode = false; - bool foundStream = false; - - // First check that the device supports the requested number of - // channels. - UInt32 deviceChannels = 0; - for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; - - if ( deviceChannels < ( channels + firstChannel ) ) { - free( bufferList ); - errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Look for a single stream meeting our needs. - UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0; - for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; - if ( streamChannels >= channels + offsetCounter ) { - firstStream = iStream; - channelOffset = offsetCounter; - foundStream = true; - break; - } - if ( streamChannels > offsetCounter ) break; - offsetCounter -= streamChannels; - } - - // If we didn't find a single stream above, then we should be able - // to meet the channel specification with multiple streams. - if ( foundStream == false ) { - monoMode = true; - offsetCounter = firstChannel; - for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; - if ( streamChannels > offsetCounter ) break; - offsetCounter -= streamChannels; - } - - firstStream = iStream; - channelOffset = offsetCounter; - Int32 channelCounter = channels + offsetCounter - streamChannels; - - if ( streamChannels > 1 ) monoMode = false; - while ( channelCounter > 0 ) { - streamChannels = bufferList->mBuffers[++iStream].mNumberChannels; - if ( streamChannels > 1 ) monoMode = false; - channelCounter -= streamChannels; - streamCount++; - } - } - - free( bufferList ); - - // Determine the buffer size. - AudioValueRange bufferRange; - dataSize = sizeof( AudioValueRange ); - property.mSelector = kAudioDevicePropertyBufferFrameSizeRange; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange ); - - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum; - else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum; - - // Set the buffer size. For multiple streams, I'm assuming we only - // need to make this setting for the master channel. - UInt32 theSize = (UInt32) *bufferSize; - dataSize = sizeof( UInt32 ); - property.mSelector = kAudioDevicePropertyBufferFrameSize; - result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize ); - - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // If attempting to setup a duplex stream, the bufferSize parameter - // MUST be the same in both directions! - *bufferSize = theSize; - if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - stream_.bufferSize = *bufferSize; - stream_.nBuffers = 1; - - // Try to set "hog" mode ... it's not clear to me this is working. - if ( options && options->flags & RTAUDIO_HOG_DEVICE ) { - pid_t hog_pid; - dataSize = sizeof( hog_pid ); - property.mSelector = kAudioDevicePropertyHogMode; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - if ( hog_pid != getpid() ) { - hog_pid = getpid(); - result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - } - - // Check and if necessary, change the sample rate for the device. - Float64 nominalRate; - dataSize = sizeof( Float64 ); - property.mSelector = kAudioDevicePropertyNominalSampleRate; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Only change the sample rate if off by more than 1 Hz. - if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) { - - // Set a property listener for the sample rate change - Float64 reportedRate = 0.0; - AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; - result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - nominalRate = (Float64) sampleRate; - result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate ); - if ( result != noErr ) { - AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Now wait until the reported nominal rate is what we just set. - UInt32 microCounter = 0; - while ( reportedRate != nominalRate ) { - microCounter += 5000; - if ( microCounter > 5000000 ) break; - usleep( 5000 ); - } - - // Remove the property listener. - AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); - - if ( microCounter > 5000000 ) { - errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Now set the stream format for all streams. Also, check the - // physical format of the device and change that if necessary. - AudioStreamBasicDescription description; - dataSize = sizeof( AudioStreamBasicDescription ); - property.mSelector = kAudioStreamPropertyVirtualFormat; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the sample rate and data format id. However, only make the - // change if the sample rate is not within 1.0 of the desired - // rate and the format is not linear pcm. - bool updateFormat = false; - if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) { - description.mSampleRate = (Float64) sampleRate; - updateFormat = true; - } - - if ( description.mFormatID != kAudioFormatLinearPCM ) { - description.mFormatID = kAudioFormatLinearPCM; - updateFormat = true; - } - - if ( updateFormat ) { - result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Now check the physical format. - property.mSelector = kAudioStreamPropertyPhysicalFormat; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - //std::cout << "Current physical stream format:" << std::endl; - //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl; - //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl; - //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl; - //std::cout << " sample rate = " << description.mSampleRate << std::endl; - - if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) { - description.mFormatID = kAudioFormatLinearPCM; - //description.mSampleRate = (Float64) sampleRate; - AudioStreamBasicDescription testDescription = description; - UInt32 formatFlags; - - // We'll try higher bit rates first and then work our way down. - std::vector< std::pair > physicalFormats; - formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger; - physicalFormats.push_back( std::pair( 32, formatFlags ) ); - formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; - physicalFormats.push_back( std::pair( 32, formatFlags ) ); - physicalFormats.push_back( std::pair( 24, formatFlags ) ); // 24-bit packed - formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh ); - physicalFormats.push_back( std::pair( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low - formatFlags |= kAudioFormatFlagIsAlignedHigh; - physicalFormats.push_back( std::pair( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high - formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; - physicalFormats.push_back( std::pair( 16, formatFlags ) ); - physicalFormats.push_back( std::pair( 8, formatFlags ) ); - - bool setPhysicalFormat = false; - for( unsigned int i=0; iflags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - stream_.deviceInterleaved[mode] = true; - if ( monoMode == true ) stream_.deviceInterleaved[mode] = false; - - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( streamCount == 1 ) { - if ( stream_.nUserChannels[mode] > 1 && - stream_.userInterleaved != stream_.deviceInterleaved[mode] ) - stream_.doConvertBuffer[mode] = true; - } - else if ( monoMode && stream_.userInterleaved ) - stream_.doConvertBuffer[mode] = true; - - // Allocate our CoreHandle structure for the stream. - CoreHandle *handle = 0; - if ( stream_.apiHandle == 0 ) { - try { - handle = new CoreHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory."; - goto error; - } - - if ( pthread_cond_init( &handle->condition, NULL ) ) { - errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - stream_.apiHandle = (void *) handle; - } - else - handle = (CoreHandle *) stream_.apiHandle; - handle->iStream[mode] = firstStream; - handle->nStreams[mode] = streamCount; - handle->id[mode] = id; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) ); - memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - // If possible, we will make use of the CoreAudio stream buffers as - // "device buffers". However, we can't do this if using multiple - // streams. - if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.sampleRate = sampleRate; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - stream_.callbackInfo.object = (void *) this; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) { - if ( streamCount > 1 ) setConvertInfo( mode, 0 ); - else setConvertInfo( mode, channelOffset ); - } - - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device ) - // Only one callback procedure per device. - stream_.mode = DUPLEX; - else { -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] ); -#else - // deprecated in favor of AudioDeviceCreateIOProcID() - result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo ); -#endif - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ")."; - errorText_ = errorStream_.str(); - goto error; - } - if ( stream_.mode == OUTPUT && mode == INPUT ) - stream_.mode = DUPLEX; - else - stream_.mode = mode; - } - - // Setup the device property listener for over/underload. - property.mSelector = kAudioDeviceProcessorOverload; - property.mScope = kAudioObjectPropertyScopeGlobal; - result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle ); - - return SUCCESS; - - error: - if ( handle ) { - pthread_cond_destroy( &handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.state = STREAM_CLOSED; - return FAILURE; -} - -void RtApiCore :: closeStream( void ) -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiCore::closeStream(): no open stream to close!"; - error( RtAudioError::WARNING ); - return; - } - - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if (handle) { - AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - - property.mSelector = kAudioDeviceProcessorOverload; - property.mScope = kAudioObjectPropertyScopeGlobal; - if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) { - errorText_ = "RtApiCore::closeStream(): error removing property listener!"; - error( RtAudioError::WARNING ); - } - -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - if ( stream_.state == STREAM_RUNNING ) - AudioDeviceStop( handle->id[0], handle->procId[0] ); - AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] ); -#else // deprecated behaviour - if ( stream_.state == STREAM_RUNNING ) - AudioDeviceStop( handle->id[0], callbackHandler ); - AudioDeviceRemoveIOProc( handle->id[0], callbackHandler ); -#endif - } - } - - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { - if (handle) { - AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - - property.mSelector = kAudioDeviceProcessorOverload; - property.mScope = kAudioObjectPropertyScopeGlobal; - if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) { - errorText_ = "RtApiCore::closeStream(): error removing property listener!"; - error( RtAudioError::WARNING ); - } - -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - if ( stream_.state == STREAM_RUNNING ) - AudioDeviceStop( handle->id[1], handle->procId[1] ); - AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] ); -#else // deprecated behaviour - if ( stream_.state == STREAM_RUNNING ) - AudioDeviceStop( handle->id[1], callbackHandler ); - AudioDeviceRemoveIOProc( handle->id[1], callbackHandler ); -#endif - } - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - // Destroy pthread condition variable. - pthread_cond_destroy( &handle->condition ); - delete handle; - stream_.apiHandle = 0; - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiCore :: startStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiCore::startStream(): the stream is already running!"; - error( RtAudioError::WARNING ); - return; - } - -#if defined( HAVE_GETTIMEOFDAY ) - gettimeofday( &stream_.lastTickTimestamp, NULL ); -#endif - - OSStatus result = noErr; - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - result = AudioDeviceStart( handle->id[0], handle->procId[0] ); -#else // deprecated behaviour - result = AudioDeviceStart( handle->id[0], callbackHandler ); -#endif - if ( result != noErr ) { - errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( stream_.mode == INPUT || - ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { - -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - result = AudioDeviceStart( handle->id[1], handle->procId[1] ); -#else // deprecated behaviour - result = AudioDeviceStart( handle->id[1], callbackHandler ); -#endif - if ( result != noErr ) { - errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; - - unlock: - if ( result == noErr ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiCore :: stopStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiCore::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - OSStatus result = noErr; - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 2; - pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled - } - -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - result = AudioDeviceStop( handle->id[0], handle->procId[0] ); -#else // deprecated behaviour - result = AudioDeviceStop( handle->id[0], callbackHandler ); -#endif - if ( result != noErr ) { - errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { - -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - result = AudioDeviceStop( handle->id[1], handle->procId[1] ); -#else // deprecated behaviour - result = AudioDeviceStop( handle->id[1], callbackHandler ); -#endif - if ( result != noErr ) { - errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - stream_.state = STREAM_STOPPED; - - unlock: - if ( result == noErr ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiCore :: abortStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiCore::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - handle->drainCounter = 2; - - stopStream(); -} - -// This function will be called by a spawned thread when the user -// callback function signals that the stream should be stopped or -// aborted. It is better to handle it this way because the -// callbackEvent() function probably should return before the AudioDeviceStop() -// function is called. -static void *coreStopStream( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiCore *object = (RtApiCore *) info->object; - - object->stopStream(); - pthread_exit( NULL ); -} - -bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, - const AudioBufferList *inBufferList, - const AudioBufferList *outBufferList ) -{ - if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtAudioError::WARNING ); - return FAILURE; - } - - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > 3 ) { - ThreadHandle threadId; - - stream_.state = STREAM_STOPPING; - if ( handle->internalDrain == true ) - pthread_create( &threadId, NULL, coreStopStream, info ); - else // external call to stopStream() - pthread_cond_signal( &handle->condition ); - return SUCCESS; - } - - AudioDeviceID outputDevice = handle->id[0]; - - // Invoke user callback to get fresh output data UNLESS we are - // draining stream or duplex mode AND the input/output devices are - // different AND this function is called for the input device. - if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - - int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( cbReturnValue == 2 ) { - stream_.state = STREAM_STOPPING; - handle->drainCounter = 2; - abortStream(); - return SUCCESS; - } - else if ( cbReturnValue == 1 ) { - handle->drainCounter = 1; - handle->internalDrain = true; - } - } - - if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) { - - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - - if ( handle->nStreams[0] == 1 ) { - memset( outBufferList->mBuffers[handle->iStream[0]].mData, - 0, - outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); - } - else { // fill multiple streams with zeros - for ( unsigned int i=0; inStreams[0]; i++ ) { - memset( outBufferList->mBuffers[handle->iStream[0]+i].mData, - 0, - outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize ); - } - } - } - else if ( handle->nStreams[0] == 1 ) { - if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer - convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData, - stream_.userBuffer[0], stream_.convertInfo[0] ); - } - else { // copy from user buffer - memcpy( outBufferList->mBuffers[handle->iStream[0]].mData, - stream_.userBuffer[0], - outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); - } - } - else { // fill multiple streams - Float32 *inBuffer = (Float32 *) stream_.userBuffer[0]; - if ( stream_.doConvertBuffer[0] ) { - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - inBuffer = (Float32 *) stream_.deviceBuffer; - } - - if ( stream_.deviceInterleaved[0] == false ) { // mono mode - UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; - for ( unsigned int i=0; imBuffers[handle->iStream[0]+i].mData, - (void *)&inBuffer[i*stream_.bufferSize], bufferBytes ); - } - } - else { // fill multiple multi-channel streams with interleaved data - UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset; - Float32 *out, *in; - - bool inInterleaved = ( stream_.userInterleaved ) ? true : false; - UInt32 inChannels = stream_.nUserChannels[0]; - if ( stream_.doConvertBuffer[0] ) { - inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode - inChannels = stream_.nDeviceChannels[0]; - } - - if ( inInterleaved ) inOffset = 1; - else inOffset = stream_.bufferSize; - - channelsLeft = inChannels; - for ( unsigned int i=0; inStreams[0]; i++ ) { - in = inBuffer; - out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData; - streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels; - - outJump = 0; - // Account for possible channel offset in first stream - if ( i == 0 && stream_.channelOffset[0] > 0 ) { - streamChannels -= stream_.channelOffset[0]; - outJump = stream_.channelOffset[0]; - out += outJump; - } - - // Account for possible unfilled channels at end of the last stream - if ( streamChannels > channelsLeft ) { - outJump = streamChannels - channelsLeft; - streamChannels = channelsLeft; - } - - // Determine input buffer offsets and skips - if ( inInterleaved ) { - inJump = inChannels; - in += inChannels - channelsLeft; - } - else { - inJump = 1; - in += (inChannels - channelsLeft) * inOffset; - } - - for ( unsigned int i=0; idrainCounter ) { - handle->drainCounter++; - goto unlock; - } - - AudioDeviceID inputDevice; - inputDevice = handle->id[1]; - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) { - - if ( handle->nStreams[1] == 1 ) { - if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer - convertBuffer( stream_.userBuffer[1], - (char *) inBufferList->mBuffers[handle->iStream[1]].mData, - stream_.convertInfo[1] ); - } - else { // copy to user buffer - memcpy( stream_.userBuffer[1], - inBufferList->mBuffers[handle->iStream[1]].mData, - inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); - } - } - else { // read from multiple streams - Float32 *outBuffer = (Float32 *) stream_.userBuffer[1]; - if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer; - - if ( stream_.deviceInterleaved[1] == false ) { // mono mode - UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize; - for ( unsigned int i=0; imBuffers[handle->iStream[1]+i].mData, bufferBytes ); - } - } - else { // read from multiple multi-channel streams - UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset; - Float32 *out, *in; - - bool outInterleaved = ( stream_.userInterleaved ) ? true : false; - UInt32 outChannels = stream_.nUserChannels[1]; - if ( stream_.doConvertBuffer[1] ) { - outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode - outChannels = stream_.nDeviceChannels[1]; - } - - if ( outInterleaved ) outOffset = 1; - else outOffset = stream_.bufferSize; - - channelsLeft = outChannels; - for ( unsigned int i=0; inStreams[1]; i++ ) { - out = outBuffer; - in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData; - streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels; - - inJump = 0; - // Account for possible channel offset in first stream - if ( i == 0 && stream_.channelOffset[1] > 0 ) { - streamChannels -= stream_.channelOffset[1]; - inJump = stream_.channelOffset[1]; - in += inJump; - } - - // Account for possible unread channels at end of the last stream - if ( streamChannels > channelsLeft ) { - inJump = streamChannels - channelsLeft; - streamChannels = channelsLeft; - } - - // Determine output buffer offsets and skips - if ( outInterleaved ) { - outJump = outChannels; - out += outChannels - channelsLeft; - } - else { - outJump = 1; - out += (outChannels - channelsLeft) * outOffset; - } - - for ( unsigned int i=0; iid[0] != handle->id[1] && deviceId == handle->id[0] ) ) - RtApi::tickStreamTime(); - - return SUCCESS; -} - -const char* RtApiCore :: getErrorCode( OSStatus code ) -{ - switch( code ) { - - case kAudioHardwareNotRunningError: - return "kAudioHardwareNotRunningError"; - - case kAudioHardwareUnspecifiedError: - return "kAudioHardwareUnspecifiedError"; - - case kAudioHardwareUnknownPropertyError: - return "kAudioHardwareUnknownPropertyError"; - - case kAudioHardwareBadPropertySizeError: - return "kAudioHardwareBadPropertySizeError"; - - case kAudioHardwareIllegalOperationError: - return "kAudioHardwareIllegalOperationError"; - - case kAudioHardwareBadObjectError: - return "kAudioHardwareBadObjectError"; - - case kAudioHardwareBadDeviceError: - return "kAudioHardwareBadDeviceError"; - - case kAudioHardwareBadStreamError: - return "kAudioHardwareBadStreamError"; - - case kAudioHardwareUnsupportedOperationError: - return "kAudioHardwareUnsupportedOperationError"; - - case kAudioDeviceUnsupportedFormatError: - return "kAudioDeviceUnsupportedFormatError"; - - case kAudioDevicePermissionsError: - return "kAudioDevicePermissionsError"; - - default: - return "CoreAudio unknown error"; - } -} - - //******************** End of __MACOSX_CORE__ *********************// -#endif - -#if defined(__UNIX_JACK__) - -// JACK is a low-latency audio server, originally written for the -// GNU/Linux operating system and now also ported to OS-X. It can -// connect a number of different applications to an audio device, as -// well as allowing them to share audio between themselves. -// -// When using JACK with RtAudio, "devices" refer to JACK clients that -// have ports connected to the server. The JACK server is typically -// started in a terminal as follows: -// -// .jackd -d alsa -d hw:0 -// -// or through an interface program such as qjackctl. Many of the -// parameters normally set for a stream are fixed by the JACK server -// and can be specified when the JACK server is started. In -// particular, -// -// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4 -// -// specifies a sample rate of 44100 Hz, a buffer size of 512 sample -// frames, and number of buffers = 4. Once the server is running, it -// is not possible to override these values. If the values are not -// specified in the command-line, the JACK server uses default values. -// -// The JACK server does not have to be running when an instance of -// RtApiJack is created, though the function getDeviceCount() will -// report 0 devices found until JACK has been started. When no -// devices are available (i.e., the JACK server is not running), a -// stream cannot be opened. - -#include -#include -#include - -// A structure to hold various information related to the Jack API -// implementation. -struct JackHandle { - jack_client_t *client; - jack_port_t **ports[2]; - std::string deviceName[2]; - bool xrun[2]; - pthread_cond_t condition; - int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - - JackHandle() - :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } -}; - -#if !defined(__RTAUDIO_DEBUG__) -static void jackSilentError( const char * ) {}; -#endif - -RtApiJack :: RtApiJack() - :shouldAutoconnect_(true) { - // Nothing to do here. -#if !defined(__RTAUDIO_DEBUG__) - // Turn off Jack's internal error reporting. - jack_set_error_function( &jackSilentError ); -#endif -} - -RtApiJack :: ~RtApiJack() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiJack :: getDeviceCount( void ) -{ - // See if we can become a jack client. - jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption; - jack_status_t *status = NULL; - jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); - if ( client == 0 ) return 0; - - const char **ports; - std::string port, previousPort; - unsigned int nChannels = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nChannels ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon + 1 ); - if ( port != previousPort ) { - nDevices++; - previousPort = port; - } - } - } while ( ports[++nChannels] ); - free( ports ); - } - - jack_client_close( client ); - return nDevices; -} - -RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption - jack_status_t *status = NULL; - jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); - if ( client == 0 ) { - errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; - error( RtAudioError::WARNING ); - return info; - } - - const char **ports; - std::string port, previousPort; - unsigned int nPorts = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nPorts ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon ); - if ( port != previousPort ) { - if ( nDevices == device ) info.name = port; - nDevices++; - previousPort = port; - } - } - } while ( ports[++nPorts] ); - free( ports ); - } - - if ( device >= nDevices ) { - jack_client_close( client ); - errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - // Get the current jack server sample rate. - info.sampleRates.clear(); - - info.preferredSampleRate = jack_get_sample_rate( client ); - info.sampleRates.push_back( info.preferredSampleRate ); - - // Count the available ports containing the client name as device - // channels. Jack "input ports" equal RtAudio output channels. - unsigned int nChannels = 0; - ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - info.outputChannels = nChannels; - } - - // Jack "output ports" equal RtAudio input channels. - nChannels = 0; - ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - info.inputChannels = nChannels; - } - - if ( info.outputChannels == 0 && info.inputChannels == 0 ) { - jack_client_close(client); - errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; - error( RtAudioError::WARNING ); - return info; - } - - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // Jack always uses 32-bit floats. - info.nativeFormats = RTAUDIO_FLOAT32; - - // Jack doesn't provide default devices so we'll use the first available one. - if ( device == 0 && info.outputChannels > 0 ) - info.isDefaultOutput = true; - if ( device == 0 && info.inputChannels > 0 ) - info.isDefaultInput = true; - - jack_client_close(client); - info.probed = true; - return info; -} - -static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) -{ - CallbackInfo *info = (CallbackInfo *) infoPointer; - - RtApiJack *object = (RtApiJack *) info->object; - if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; - - return 0; -} - -// This function will be called by a spawned thread when the Jack -// server signals that it is shutting down. It is necessary to handle -// it this way because the jackShutdown() function must return before -// the jack_deactivate() function (in closeStream()) will return. -static void *jackCloseStream( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiJack *object = (RtApiJack *) info->object; - - object->closeStream(); - - pthread_exit( NULL ); -} -static void jackShutdown( void *infoPointer ) -{ - CallbackInfo *info = (CallbackInfo *) infoPointer; - RtApiJack *object = (RtApiJack *) info->object; - - // Check current stream state. If stopped, then we'll assume this - // was called as a result of a call to RtApiJack::stopStream (the - // deactivation of a client handle causes this function to be called). - // If not, we'll assume the Jack server is shutting down or some - // other problem occurred and we should close the stream. - if ( object->isStreamRunning() == false ) return; - - ThreadHandle threadId; - pthread_create( &threadId, NULL, jackCloseStream, info ); - std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; -} - -static int jackXrun( void *infoPointer ) -{ - JackHandle *handle = *((JackHandle **) infoPointer); - - if ( handle->ports[0] ) handle->xrun[0] = true; - if ( handle->ports[1] ) handle->xrun[1] = true; - - return 0; -} - -bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - JackHandle *handle = (JackHandle *) stream_.apiHandle; - - // Look for jack server and try to become a client (only do once per stream). - jack_client_t *client = 0; - if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { - jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption; - jack_status_t *status = NULL; - if ( options && !options->streamName.empty() ) - client = jack_client_open( options->streamName.c_str(), jackoptions, status ); - else - client = jack_client_open( "RtApiJack", jackoptions, status ); - if ( client == 0 ) { - errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; - error( RtAudioError::WARNING ); - return FAILURE; - } - } - else { - // The handle must have been created on an earlier pass. - client = handle->client; - } - - const char **ports; - std::string port, previousPort, deviceName; - unsigned int nPorts = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nPorts ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon ); - if ( port != previousPort ) { - if ( nDevices == device ) deviceName = port; - nDevices++; - previousPort = port; - } - } - } while ( ports[++nPorts] ); - free( ports ); - } - - if ( device >= nDevices ) { - errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - unsigned long flag = JackPortIsInput; - if ( mode == INPUT ) flag = JackPortIsOutput; - - if ( ! (options && (options->flags & RTAUDIO_JACK_DONT_CONNECT)) ) { - // Count the available ports containing the client name as device - // channels. Jack "input ports" equal RtAudio output channels. - unsigned int nChannels = 0; - ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - } - // Compare the jack ports for specified client to the requested number of channels. - if ( nChannels < (channels + firstChannel) ) { - errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Check the jack server sample rate. - unsigned int jackRate = jack_get_sample_rate( client ); - if ( sampleRate != jackRate ) { - jack_client_close( client ); - errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.sampleRate = jackRate; - - // Get the latency of the JACK port. - ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag ); - if ( ports[ firstChannel ] ) { - // Added by Ge Wang - jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency); - // the range (usually the min and max are equal) - jack_latency_range_t latrange; latrange.min = latrange.max = 0; - // get the latency range - jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange ); - // be optimistic, use the min! - stream_.latency[mode] = latrange.min; - //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); - } - free( ports ); - - // The jack server always uses 32-bit floating-point data. - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - stream_.userFormat = format; - - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - - // Jack always uses non-interleaved buffers. - stream_.deviceInterleaved[mode] = false; - - // Jack always provides host byte-ordered data. - stream_.doByteSwap[mode] = false; - - // Get the buffer size. The buffer size and number of buffers - // (periods) is set when the jack server is started. - stream_.bufferSize = (int) jack_get_buffer_size( client ); - *bufferSize = stream_.bufferSize; - - stream_.nDeviceChannels[mode] = channels; - stream_.nUserChannels[mode] = channels; - - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate our JackHandle structure for the stream. - if ( handle == 0 ) { - try { - handle = new JackHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory."; - goto error; - } - - if ( pthread_cond_init(&handle->condition, NULL) ) { - errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - stream_.apiHandle = (void *) handle; - handle->client = client; - } - handle->deviceName[mode] = deviceName; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - if ( mode == OUTPUT ) - bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - else { // mode == INPUT - bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] ); - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]); - if ( bufferBytes < bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - // Allocate memory for the Jack ports (channels) identifiers. - handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels ); - if ( handle->ports[mode] == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory."; - goto error; - } - - stream_.device[mode] = device; - stream_.channelOffset[mode] = firstChannel; - stream_.state = STREAM_STOPPED; - stream_.callbackInfo.object = (void *) this; - - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up the stream for output. - stream_.mode = DUPLEX; - else { - stream_.mode = mode; - jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); - jack_set_xrun_callback( handle->client, jackXrun, (void *) &stream_.apiHandle ); - jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); - } - - // Register our ports. - char label[64]; - if ( mode == OUTPUT ) { - for ( unsigned int i=0; iports[0][i] = jack_port_register( handle->client, (const char *)label, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 ); - } - } - else { - for ( unsigned int i=0; iports[1][i] = jack_port_register( handle->client, (const char *)label, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 ); - } - } - - // Setup the buffer conversion information structure. We don't use - // buffers to do channel offsets, so we override that parameter - // here. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); - - if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false; - - return SUCCESS; - - error: - if ( handle ) { - pthread_cond_destroy( &handle->condition ); - jack_client_close( handle->client ); - - if ( handle->ports[0] ) free( handle->ports[0] ); - if ( handle->ports[1] ) free( handle->ports[1] ); - - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - return FAILURE; -} - -void RtApiJack :: closeStream( void ) -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiJack::closeStream(): no open stream to close!"; - error( RtAudioError::WARNING ); - return; - } - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - if ( handle ) { - - if ( stream_.state == STREAM_RUNNING ) - jack_deactivate( handle->client ); - - jack_client_close( handle->client ); - } - - if ( handle ) { - if ( handle->ports[0] ) free( handle->ports[0] ); - if ( handle->ports[1] ) free( handle->ports[1] ); - pthread_cond_destroy( &handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiJack :: startStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiJack::startStream(): the stream is already running!"; - error( RtAudioError::WARNING ); - return; - } - - #if defined( HAVE_GETTIMEOFDAY ) - gettimeofday( &stream_.lastTickTimestamp, NULL ); - #endif - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - int result = jack_activate( handle->client ); - if ( result ) { - errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; - goto unlock; - } - - const char **ports; - - // Get the list of available ports. - if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) { - result = 1; - ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput); - if ( ports == NULL) { - errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; - goto unlock; - } - - // Now make the port connections. Since RtAudio wasn't designed to - // allow the user to select particular channels of a device, we'll - // just open the first "nChannels" ports with offset. - for ( unsigned int i=0; iclient, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); - if ( result ) { - free( ports ); - errorText_ = "RtApiJack::startStream(): error connecting output ports!"; - goto unlock; - } - } - free(ports); - } - - if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) { - result = 1; - ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput ); - if ( ports == NULL) { - errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; - goto unlock; - } - - // Now make the port connections. See note above. - for ( unsigned int i=0; iclient, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) ); - if ( result ) { - free( ports ); - errorText_ = "RtApiJack::startStream(): error connecting input ports!"; - goto unlock; - } - } - free(ports); - } - - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; - - unlock: - if ( result == 0 ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiJack :: stopStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 2; - pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled - } - } - - jack_deactivate( handle->client ); - stream_.state = STREAM_STOPPED; -} - -void RtApiJack :: abortStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - handle->drainCounter = 2; - - stopStream(); -} - -// This function will be called by a spawned thread when the user -// callback function signals that the stream should be stopped or -// aborted. It is necessary to handle it this way because the -// callbackEvent() function must return before the jack_deactivate() -// function will return. -static void *jackStopStream( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiJack *object = (RtApiJack *) info->object; - - object->stopStream(); - pthread_exit( NULL ); -} - -bool RtApiJack :: callbackEvent( unsigned long nframes ) -{ - if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtAudioError::WARNING ); - return FAILURE; - } - if ( stream_.bufferSize != nframes ) { - errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; - error( RtAudioError::WARNING ); - return FAILURE; - } - - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - JackHandle *handle = (JackHandle *) stream_.apiHandle; - - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > 3 ) { - ThreadHandle threadId; - - stream_.state = STREAM_STOPPING; - if ( handle->internalDrain == true ) - pthread_create( &threadId, NULL, jackStopStream, info ); - else - pthread_cond_signal( &handle->condition ); - return SUCCESS; - } - - // Invoke user callback first, to get fresh output data. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( cbReturnValue == 2 ) { - stream_.state = STREAM_STOPPING; - handle->drainCounter = 2; - ThreadHandle id; - pthread_create( &id, NULL, jackStopStream, info ); - return SUCCESS; - } - else if ( cbReturnValue == 1 ) { - handle->drainCounter = 1; - handle->internalDrain = true; - } - } - - jack_default_audio_sample_t *jackbuffer; - unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - - for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); - memset( jackbuffer, 0, bufferBytes ); - } - - } - else if ( stream_.doConvertBuffer[0] ) { - - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - - for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); - memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); - } - } - else { // no buffer conversion - for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); - memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); - } - } - } - - // Don't bother draining input - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - if ( stream_.doConvertBuffer[1] ) { - for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); - memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes ); - } - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - } - else { // no buffer conversion - for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); - memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes ); - } - } - } - - unlock: - RtApi::tickStreamTime(); - return SUCCESS; -} - //******************** End of __UNIX_JACK__ *********************// -#endif - -#if defined(__WINDOWS_ASIO__) // ASIO API on Windows - -// The ASIO API is designed around a callback scheme, so this -// implementation is similar to that used for OS-X CoreAudio and Linux -// Jack. The primary constraint with ASIO is that it only allows -// access to a single driver at a time. Thus, it is not possible to -// have more than one simultaneous RtAudio stream. -// -// This implementation also requires a number of external ASIO files -// and a few global variables. The ASIO callback scheme does not -// allow for the passing of user data, so we must create a global -// pointer to our callbackInfo structure. -// -// On unix systems, we make use of a pthread condition variable. -// Since there is no equivalent in Windows, I hacked something based -// on information found in -// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html. - -#include "asiosys.h" -#include "asio.h" -#include "iasiothiscallresolver.h" -#include "asiodrivers.h" -#include - -static AsioDrivers drivers; -static ASIOCallbacks asioCallbacks; -static ASIODriverInfo driverInfo; -static CallbackInfo *asioCallbackInfo; -static bool asioXRun; - -struct AsioHandle { - int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - ASIOBufferInfo *bufferInfos; - HANDLE condition; - - AsioHandle() - :drainCounter(0), internalDrain(false), bufferInfos(0) {} -}; - -// Function declarations (definitions at end of section) -static const char* getAsioErrorString( ASIOError result ); -static void sampleRateChanged( ASIOSampleRate sRate ); -static long asioMessages( long selector, long value, void* message, double* opt ); - -RtApiAsio :: RtApiAsio() -{ - // ASIO cannot run on a multi-threaded appartment. You can call - // CoInitialize beforehand, but it must be for appartment threading - // (in which case, CoInitilialize will return S_FALSE here). - coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); - if ( FAILED(hr) ) { - errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; - error( RtAudioError::WARNING ); - } - coInitialized_ = true; - - drivers.removeCurrentDriver(); - driverInfo.asioVersion = 2; - - // See note in DirectSound implementation about GetDesktopWindow(). - driverInfo.sysRef = GetForegroundWindow(); -} - -RtApiAsio :: ~RtApiAsio() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); - if ( coInitialized_ ) CoUninitialize(); -} - -unsigned int RtApiAsio :: getDeviceCount( void ) -{ - return (unsigned int) drivers.asioGetNumDev(); -} - -RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - // Get device ID - unsigned int nDevices = getDeviceCount(); - if ( nDevices == 0 ) { - errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - if ( device >= nDevices ) { - errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - // If a stream is already open, we cannot probe other devices. Thus, use the saved results. - if ( stream_.state != STREAM_CLOSED ) { - if ( device >= devices_.size() ) { - errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened."; - error( RtAudioError::WARNING ); - return info; - } - return devices_[ device ]; - } - - char driverName[32]; - ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - info.name = driverName; - - if ( !drivers.loadDriver( driverName ) ) { - errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - result = ASIOInit( &driverInfo ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Determine the device channel information. - long inputChannels, outputChannels; - result = ASIOGetChannels( &inputChannels, &outputChannels ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - info.outputChannels = outputChannels; - info.inputChannels = inputChannels; - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // Determine the supported sample rates. - info.sampleRates.clear(); - for ( unsigned int i=0; i info.preferredSampleRate ) ) - info.preferredSampleRate = SAMPLE_RATES[i]; - } - } - - // Determine supported data types ... just check first channel and assume rest are the same. - ASIOChannelInfo channelInfo; - channelInfo.channel = 0; - channelInfo.isInput = true; - if ( info.inputChannels <= 0 ) channelInfo.isInput = false; - result = ASIOGetChannelInfo( &channelInfo ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - info.nativeFormats = 0; - if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) - info.nativeFormats |= RTAUDIO_SINT16; - else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) - info.nativeFormats |= RTAUDIO_SINT32; - else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) - info.nativeFormats |= RTAUDIO_FLOAT32; - else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) - info.nativeFormats |= RTAUDIO_FLOAT64; - else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) - info.nativeFormats |= RTAUDIO_SINT24; - - if ( info.outputChannels > 0 ) - if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; - if ( info.inputChannels > 0 ) - if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; - - info.probed = true; - drivers.removeCurrentDriver(); - return info; -} - -static void bufferSwitch( long index, ASIOBool /*processNow*/ ) -{ - RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object; - object->callbackEvent( index ); -} - -void RtApiAsio :: saveDeviceInfo( void ) -{ - devices_.clear(); - - unsigned int nDevices = getDeviceCount(); - devices_.resize( nDevices ); - for ( unsigned int i=0; isaveDeviceInfo(); - - if ( !drivers.loadDriver( driverName ) ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - result = ASIOInit( &driverInfo ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // keep them before any "goto error", they are used for error cleanup + goto device boundary checks - bool buffersAllocated = false; - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - unsigned int nChannels; - - - // Check the device channel count. - long inputChannels, outputChannels; - result = ASIOGetChannels( &inputChannels, &outputChannels ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; - errorText_ = errorStream_.str(); - goto error; - } - - if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || - ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; - errorText_ = errorStream_.str(); - goto error; - } - stream_.nDeviceChannels[mode] = channels; - stream_.nUserChannels[mode] = channels; - stream_.channelOffset[mode] = firstChannel; - - // Verify the sample rate is supported. - result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - goto error; - } - - // Get the current sample rate - ASIOSampleRate currentRate; - result = ASIOGetSampleRate( ¤tRate ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; - errorText_ = errorStream_.str(); - goto error; - } - - // Set the sample rate only if necessary - if ( currentRate != sampleRate ) { - result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - goto error; - } - } - - // Determine the driver data type. - ASIOChannelInfo channelInfo; - channelInfo.channel = 0; - if ( mode == OUTPUT ) channelInfo.isInput = false; - else channelInfo.isInput = true; - result = ASIOGetChannelInfo( &channelInfo ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; - errorText_ = errorStream_.str(); - goto error; - } - - // Assuming WINDOWS host is always little-endian. - stream_.doByteSwap[mode] = false; - stream_.userFormat = format; - stream_.deviceFormat[mode] = 0; - if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true; - } - else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true; - } - else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true; - } - else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; - if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true; - } - else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true; - } - - if ( stream_.deviceFormat[mode] == 0 ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - goto error; - } - - // Set the buffer size. For a duplex stream, this will end up - // setting the buffer size based on the input constraints, which - // should be ok. - long minSize, maxSize, preferSize, granularity; - result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; - errorText_ = errorStream_.str(); - goto error; - } - - if ( isDuplexInput ) { - // When this is the duplex input (output was opened before), then we have to use the same - // buffersize as the output, because it might use the preferred buffer size, which most - // likely wasn't passed as input to this. The buffer sizes have to be identically anyway, - // So instead of throwing an error, make them equal. The caller uses the reference - // to the "bufferSize" param as usual to set up processing buffers. - - *bufferSize = stream_.bufferSize; - - } else { - if ( *bufferSize == 0 ) *bufferSize = preferSize; - else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; - else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - else if ( granularity == -1 ) { - // Make sure bufferSize is a power of two. - int log2_of_min_size = 0; - int log2_of_max_size = 0; - - for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { - if ( minSize & ((long)1 << i) ) log2_of_min_size = i; - if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; - } - - long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); - int min_delta_num = log2_of_min_size; - - for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { - long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); - if (current_delta < min_delta) { - min_delta = current_delta; - min_delta_num = i; - } - } - - *bufferSize = ( (unsigned int)1 << min_delta_num ); - if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; - else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - } - else if ( granularity != 0 ) { - // Set to an even multiple of granularity, rounding up. - *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; - } - } - - /* - // we don't use it anymore, see above! - // Just left it here for the case... - if ( isDuplexInput && stream_.bufferSize != *bufferSize ) { - errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; - goto error; - } - */ - - stream_.bufferSize = *bufferSize; - stream_.nBuffers = 2; - - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - - // ASIO always uses non-interleaved buffers. - stream_.deviceInterleaved[mode] = false; - - // Allocate, if necessary, our AsioHandle structure for the stream. - if ( handle == 0 ) { - try { - handle = new AsioHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; - goto error; - } - handle->bufferInfos = 0; - - // Create a manual-reset event. - handle->condition = CreateEvent( NULL, // no security - TRUE, // manual-reset - FALSE, // non-signaled initially - NULL ); // unnamed - stream_.apiHandle = (void *) handle; - } - - // Create the ASIO internal buffers. Since RtAudio sets up input - // and output separately, we'll have to dispose of previously - // created output buffers for a duplex stream. - if ( mode == INPUT && stream_.mode == OUTPUT ) { - ASIODisposeBuffers(); - if ( handle->bufferInfos ) free( handle->bufferInfos ); - } - - // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. - unsigned int i; - nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; - handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); - if ( handle->bufferInfos == NULL ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - goto error; - } - - ASIOBufferInfo *infos; - infos = handle->bufferInfos; - for ( i=0; iisInput = ASIOFalse; - infos->channelNum = i + stream_.channelOffset[0]; - infos->buffers[0] = infos->buffers[1] = 0; - } - for ( i=0; iisInput = ASIOTrue; - infos->channelNum = i + stream_.channelOffset[1]; - infos->buffers[0] = infos->buffers[1] = 0; - } - - // prepare for callbacks - stream_.sampleRate = sampleRate; - stream_.device[mode] = device; - stream_.mode = isDuplexInput ? DUPLEX : mode; - - // store this class instance before registering callbacks, that are going to use it - asioCallbackInfo = &stream_.callbackInfo; - stream_.callbackInfo.object = (void *) this; - - // Set up the ASIO callback structure and create the ASIO data buffers. - asioCallbacks.bufferSwitch = &bufferSwitch; - asioCallbacks.sampleRateDidChange = &sampleRateChanged; - asioCallbacks.asioMessage = &asioMessages; - asioCallbacks.bufferSwitchTimeInfo = NULL; - result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); - if ( result != ASE_OK ) { - // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges - // but only accept the preferred buffer size as parameter for ASIOCreateBuffers (e.g. Creative's ASIO driver). - // In that case, let's be naïve and try that instead. - *bufferSize = preferSize; - stream_.bufferSize = *bufferSize; - result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); - } - - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; - errorText_ = errorStream_.str(); - goto error; - } - buffersAllocated = true; - stream_.state = STREAM_STOPPED; - - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( isDuplexInput && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - // Determine device latencies - long inputLatency, outputLatency; - result = ASIOGetLatencies( &inputLatency, &outputLatency ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING); // warn but don't fail - } - else { - stream_.latency[0] = outputLatency; - stream_.latency[1] = inputLatency; - } - - // Setup the buffer conversion information structure. We don't use - // buffers to do channel offsets, so we override that parameter - // here. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); - - return SUCCESS; - - error: - if ( !isDuplexInput ) { - // the cleanup for error in the duplex input, is done by RtApi::openStream - // So we clean up for single channel only - - if ( buffersAllocated ) - ASIODisposeBuffers(); - - drivers.removeCurrentDriver(); - - if ( handle ) { - CloseHandle( handle->condition ); - if ( handle->bufferInfos ) - free( handle->bufferInfos ); - - delete handle; - stream_.apiHandle = 0; - } - - - if ( stream_.userBuffer[mode] ) { - free( stream_.userBuffer[mode] ); - stream_.userBuffer[mode] = 0; - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - } - - return FAILURE; -}//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// - -void RtApiAsio :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; - error( RtAudioError::WARNING ); - return; - } - - if ( stream_.state == STREAM_RUNNING ) { - stream_.state = STREAM_STOPPED; - ASIOStop(); - } - ASIODisposeBuffers(); - drivers.removeCurrentDriver(); - - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( handle ) { - CloseHandle( handle->condition ); - if ( handle->bufferInfos ) - free( handle->bufferInfos ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -bool stopThreadCalled = false; - -void RtApiAsio :: startStream() -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiAsio::startStream(): the stream is already running!"; - error( RtAudioError::WARNING ); - return; - } - - #if defined( HAVE_GETTIMEOFDAY ) - gettimeofday( &stream_.lastTickTimestamp, NULL ); - #endif - - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - ASIOError result = ASIOStart(); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device."; - errorText_ = errorStream_.str(); - goto unlock; - } - - handle->drainCounter = 0; - handle->internalDrain = false; - ResetEvent( handle->condition ); - stream_.state = STREAM_RUNNING; - asioXRun = false; - - unlock: - stopThreadCalled = false; - - if ( result == ASE_OK ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiAsio :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 2; - WaitForSingleObject( handle->condition, INFINITE ); // block until signaled - } - } - - stream_.state = STREAM_STOPPED; - - ASIOError result = ASIOStop(); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; - errorText_ = errorStream_.str(); - } - - if ( result == ASE_OK ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiAsio :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - // The following lines were commented-out because some behavior was - // noted where the device buffers need to be zeroed to avoid - // continuing sound, even when the device buffers are completely - // disposed. So now, calling abort is the same as calling stop. - // AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - // handle->drainCounter = 2; - stopStream(); -} - -// This function will be called by a spawned thread when the user -// callback function signals that the stream should be stopped or -// aborted. It is necessary to handle it this way because the -// callbackEvent() function must return before the ASIOStop() -// function will return. -static unsigned __stdcall asioStopStream( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiAsio *object = (RtApiAsio *) info->object; - - object->stopStream(); - _endthreadex( 0 ); - return 0; -} - -bool RtApiAsio :: callbackEvent( long bufferIndex ) -{ - if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtAudioError::WARNING ); - return FAILURE; - } - - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - - // Check if we were draining the stream and signal if finished. - if ( handle->drainCounter > 3 ) { - - stream_.state = STREAM_STOPPING; - if ( handle->internalDrain == false ) - SetEvent( handle->condition ); - else { // spawn a thread to stop the stream - unsigned threadId; - stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, - &stream_.callbackInfo, 0, &threadId ); - } - return SUCCESS; - } - - // Invoke user callback to get fresh output data UNLESS we are - // draining stream. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && asioXRun == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - asioXRun = false; - } - if ( stream_.mode != OUTPUT && asioXRun == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - asioXRun = false; - } - int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( cbReturnValue == 2 ) { - stream_.state = STREAM_STOPPING; - handle->drainCounter = 2; - unsigned threadId; - stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, - &stream_.callbackInfo, 0, &threadId ); - return SUCCESS; - } - else if ( cbReturnValue == 1 ) { - handle->drainCounter = 1; - handle->internalDrain = true; - } - } - - unsigned int nChannels, bufferBytes, i, j; - nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); - - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - - for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) - memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); - } - - } - else if ( stream_.doConvertBuffer[0] ) { - - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( stream_.deviceBuffer, - stream_.bufferSize * stream_.nDeviceChannels[0], - stream_.deviceFormat[0] ); - - for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) - memcpy( handle->bufferInfos[i].buffers[bufferIndex], - &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); - } - - } - else { - - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( stream_.userBuffer[0], - stream_.bufferSize * stream_.nUserChannels[0], - stream_.userFormat ); - - for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) - memcpy( handle->bufferInfos[i].buffers[bufferIndex], - &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); - } - - } - } - - // Don't bother draining input - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); - - if (stream_.doConvertBuffer[1]) { - - // Always interleave ASIO input data. - for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) - memcpy( &stream_.deviceBuffer[j++*bufferBytes], - handle->bufferInfos[i].buffers[bufferIndex], - bufferBytes ); - } - - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( stream_.deviceBuffer, - stream_.bufferSize * stream_.nDeviceChannels[1], - stream_.deviceFormat[1] ); - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - - } - else { - for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) { - memcpy( &stream_.userBuffer[1][bufferBytes*j++], - handle->bufferInfos[i].buffers[bufferIndex], - bufferBytes ); - } - } - - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( stream_.userBuffer[1], - stream_.bufferSize * stream_.nUserChannels[1], - stream_.userFormat ); - } - } - - unlock: - // The following call was suggested by Malte Clasen. While the API - // documentation indicates it should not be required, some device - // drivers apparently do not function correctly without it. - ASIOOutputReady(); - - RtApi::tickStreamTime(); - return SUCCESS; -} - -static void sampleRateChanged( ASIOSampleRate sRate ) -{ - // The ASIO documentation says that this usually only happens during - // external sync. Audio processing is not stopped by the driver, - // actual sample rate might not have even changed, maybe only the - // sample rate status of an AES/EBU or S/PDIF digital input at the - // audio device. - - RtApi *object = (RtApi *) asioCallbackInfo->object; - try { - object->stopStream(); - } - catch ( RtAudioError &exception ) { - std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; - return; - } - - std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; -} - -static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ ) -{ - long ret = 0; - - switch( selector ) { - case kAsioSelectorSupported: - if ( value == kAsioResetRequest - || value == kAsioEngineVersion - || value == kAsioResyncRequest - || value == kAsioLatenciesChanged - // The following three were added for ASIO 2.0, you don't - // necessarily have to support them. - || value == kAsioSupportsTimeInfo - || value == kAsioSupportsTimeCode - || value == kAsioSupportsInputMonitor) - ret = 1L; - break; - case kAsioResetRequest: - // Defer the task and perform the reset of the driver during the - // next "safe" situation. You cannot reset the driver right now, - // as this code is called from the driver. Reset the driver is - // done by completely destruct is. I.e. ASIOStop(), - // ASIODisposeBuffers(), Destruction Afterwards you initialize the - // driver again. - std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; - ret = 1L; - break; - case kAsioResyncRequest: - // This informs the application that the driver encountered some - // non-fatal data loss. It is used for synchronization purposes - // of different media. Added mainly to work around the Win16Mutex - // problems in Windows 95/98 with the Windows Multimedia system, - // which could lose data because the Mutex was held too long by - // another thread. However a driver can issue it in other - // situations, too. - // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; - asioXRun = true; - ret = 1L; - break; - case kAsioLatenciesChanged: - // This will inform the host application that the drivers were - // latencies changed. Beware, it this does not mean that the - // buffer sizes have changed! You might need to update internal - // delay data. - std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; - ret = 1L; - break; - case kAsioEngineVersion: - // Return the supported ASIO version of the host application. If - // a host application does not implement this selector, ASIO 1.0 - // is assumed by the driver. - ret = 2L; - break; - case kAsioSupportsTimeInfo: - // Informs the driver whether the - // asioCallbacks.bufferSwitchTimeInfo() callback is supported. - // For compatibility with ASIO 1.0 drivers the host application - // should always support the "old" bufferSwitch method, too. - ret = 0; - break; - case kAsioSupportsTimeCode: - // Informs the driver whether application is interested in time - // code info. If an application does not need to know about time - // code, the driver has less work to do. - ret = 0; - break; - } - return ret; -} - -static const char* getAsioErrorString( ASIOError result ) -{ - struct Messages - { - ASIOError value; - const char*message; - }; - - static const Messages m[] = - { - { ASE_NotPresent, "Hardware input or output is not present or available." }, - { ASE_HWMalfunction, "Hardware is malfunctioning." }, - { ASE_InvalidParameter, "Invalid input parameter." }, - { ASE_InvalidMode, "Invalid mode." }, - { ASE_SPNotAdvancing, "Sample position not advancing." }, - { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, - { ASE_NoMemory, "Not enough memory to complete the request." } - }; - - for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) - if ( m[i].value == result ) return m[i].message; - - return "Unknown error."; -} - -//******************** End of __WINDOWS_ASIO__ *********************// -#endif - - -#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API - -// Authored by Marcus Tomlinson , April 2014 -// - Introduces support for the Windows WASAPI API -// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required -// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface -// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user - -#ifndef INITGUID - #define INITGUID -#endif - -#include -#include -#include -#include -#include - -#include -#include -#include -#include - -#ifndef MF_E_TRANSFORM_NEED_MORE_INPUT - #define MF_E_TRANSFORM_NEED_MORE_INPUT _HRESULT_TYPEDEF_(0xc00d6d72) -#endif - -#ifndef MFSTARTUP_NOSOCKET - #define MFSTARTUP_NOSOCKET 0x1 -#endif - -#ifdef _MSC_VER - #pragma comment( lib, "ksuser" ) - #pragma comment( lib, "mfplat.lib" ) - #pragma comment( lib, "mfuuid.lib" ) - #pragma comment( lib, "wmcodecdspuuid" ) -#endif - -//============================================================================= - -#define SAFE_RELEASE( objectPtr )\ -if ( objectPtr )\ -{\ - objectPtr->Release();\ - objectPtr = NULL;\ -} - -typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex ); - -//----------------------------------------------------------------------------- - -// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size. -// Therefore we must perform all necessary conversions to user buffers in order to satisfy these -// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to -// provide intermediate storage for read / write synchronization. -class WasapiBuffer -{ -public: - WasapiBuffer() - : buffer_( NULL ), - bufferSize_( 0 ), - inIndex_( 0 ), - outIndex_( 0 ) {} - - ~WasapiBuffer() { - free( buffer_ ); - } - - // sets the length of the internal ring buffer - void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) { - free( buffer_ ); - - buffer_ = ( char* ) calloc( bufferSize, formatBytes ); - - bufferSize_ = bufferSize; - inIndex_ = 0; - outIndex_ = 0; - } - - // attempt to push a buffer into the ring buffer at the current "in" index - bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) - { - if ( !buffer || // incoming buffer is NULL - bufferSize == 0 || // incoming buffer has no data - bufferSize > bufferSize_ ) // incoming buffer too large - { - return false; - } - - unsigned int relOutIndex = outIndex_; - unsigned int inIndexEnd = inIndex_ + bufferSize; - if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) { - relOutIndex += bufferSize_; - } - - // the "IN" index CAN BEGIN at the "OUT" index - // the "IN" index CANNOT END at the "OUT" index - if ( inIndex_ < relOutIndex && inIndexEnd >= relOutIndex ) { - return false; // not enough space between "in" index and "out" index - } - - // copy buffer from external to internal - int fromZeroSize = inIndex_ + bufferSize - bufferSize_; - fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; - int fromInSize = bufferSize - fromZeroSize; - - switch( format ) - { - case RTAUDIO_SINT8: - memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) ); - memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) ); - break; - case RTAUDIO_SINT16: - memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) ); - memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) ); - break; - case RTAUDIO_SINT24: - memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) ); - memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) ); - break; - case RTAUDIO_SINT32: - memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) ); - memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) ); - break; - case RTAUDIO_FLOAT32: - memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) ); - memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) ); - break; - case RTAUDIO_FLOAT64: - memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) ); - memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) ); - break; - } - - // update "in" index - inIndex_ += bufferSize; - inIndex_ %= bufferSize_; - - return true; - } - - // attempt to pull a buffer from the ring buffer from the current "out" index - bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) - { - if ( !buffer || // incoming buffer is NULL - bufferSize == 0 || // incoming buffer has no data - bufferSize > bufferSize_ ) // incoming buffer too large - { - return false; - } - - unsigned int relInIndex = inIndex_; - unsigned int outIndexEnd = outIndex_ + bufferSize; - if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) { - relInIndex += bufferSize_; - } - - // the "OUT" index CANNOT BEGIN at the "IN" index - // the "OUT" index CAN END at the "IN" index - if ( outIndex_ <= relInIndex && outIndexEnd > relInIndex ) { - return false; // not enough space between "out" index and "in" index - } - - // copy buffer from internal to external - int fromZeroSize = outIndex_ + bufferSize - bufferSize_; - fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; - int fromOutSize = bufferSize - fromZeroSize; - - switch( format ) - { - case RTAUDIO_SINT8: - memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) ); - memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) ); - break; - case RTAUDIO_SINT16: - memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) ); - memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) ); - break; - case RTAUDIO_SINT24: - memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) ); - memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) ); - break; - case RTAUDIO_SINT32: - memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) ); - memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) ); - break; - case RTAUDIO_FLOAT32: - memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) ); - memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) ); - break; - case RTAUDIO_FLOAT64: - memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) ); - memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) ); - break; - } - - // update "out" index - outIndex_ += bufferSize; - outIndex_ %= bufferSize_; - - return true; - } - -private: - char* buffer_; - unsigned int bufferSize_; - unsigned int inIndex_; - unsigned int outIndex_; -}; - -//----------------------------------------------------------------------------- - -// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate -// between HW and the user. The WasapiResampler class is used to perform this conversion between -// HwIn->UserIn and UserOut->HwOut during the stream callback loop. -class WasapiResampler -{ -public: - WasapiResampler( bool isFloat, unsigned int bitsPerSample, unsigned int channelCount, - unsigned int inSampleRate, unsigned int outSampleRate ) - : _bytesPerSample( bitsPerSample / 8 ) - , _channelCount( channelCount ) - , _sampleRatio( ( float ) outSampleRate / inSampleRate ) - , _transformUnk( NULL ) - , _transform( NULL ) - , _mediaType( NULL ) - , _inputMediaType( NULL ) - , _outputMediaType( NULL ) - - #ifdef __IWMResamplerProps_FWD_DEFINED__ - , _resamplerProps( NULL ) - #endif - { - // 1. Initialization - - MFStartup( MF_VERSION, MFSTARTUP_NOSOCKET ); - - // 2. Create Resampler Transform Object - - CoCreateInstance( CLSID_CResamplerMediaObject, NULL, CLSCTX_INPROC_SERVER, - IID_IUnknown, ( void** ) &_transformUnk ); - - _transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) ); - - #ifdef __IWMResamplerProps_FWD_DEFINED__ - _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) ); - _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality - #endif - - // 3. Specify input / output format - - MFCreateMediaType( &_mediaType ); - _mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio ); - _mediaType->SetGUID( MF_MT_SUBTYPE, isFloat ? MFAudioFormat_Float : MFAudioFormat_PCM ); - _mediaType->SetUINT32( MF_MT_AUDIO_NUM_CHANNELS, channelCount ); - _mediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, inSampleRate ); - _mediaType->SetUINT32( MF_MT_AUDIO_BLOCK_ALIGNMENT, _bytesPerSample * channelCount ); - _mediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * inSampleRate ); - _mediaType->SetUINT32( MF_MT_AUDIO_BITS_PER_SAMPLE, bitsPerSample ); - _mediaType->SetUINT32( MF_MT_ALL_SAMPLES_INDEPENDENT, TRUE ); - - MFCreateMediaType( &_inputMediaType ); - _mediaType->CopyAllItems( _inputMediaType ); - - _transform->SetInputType( 0, _inputMediaType, 0 ); - - MFCreateMediaType( &_outputMediaType ); - _mediaType->CopyAllItems( _outputMediaType ); - - _outputMediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, outSampleRate ); - _outputMediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * outSampleRate ); - - _transform->SetOutputType( 0, _outputMediaType, 0 ); - - // 4. Send stream start messages to Resampler - - _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, 0 ); - _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, 0 ); - _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, 0 ); - } - - ~WasapiResampler() - { - // 8. Send stream stop messages to Resampler - - _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, 0 ); - _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, 0 ); - - // 9. Cleanup - - MFShutdown(); - - SAFE_RELEASE( _transformUnk ); - SAFE_RELEASE( _transform ); - SAFE_RELEASE( _mediaType ); - SAFE_RELEASE( _inputMediaType ); - SAFE_RELEASE( _outputMediaType ); - - #ifdef __IWMResamplerProps_FWD_DEFINED__ - SAFE_RELEASE( _resamplerProps ); - #endif - } - - void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount, int maxOutSampleCount = -1 ) - { - unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount; - if ( _sampleRatio == 1 ) - { - // no sample rate conversion required - memcpy( outBuffer, inBuffer, inputBufferSize ); - outSampleCount = inSampleCount; - return; - } - - unsigned int outputBufferSize = 0; - if ( maxOutSampleCount != -1 ) - { - outputBufferSize = _bytesPerSample * _channelCount * maxOutSampleCount; - } - else - { - outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount ); - } - - IMFMediaBuffer* rInBuffer; - IMFSample* rInSample; - BYTE* rInByteBuffer = NULL; - - // 5. Create Sample object from input data - - MFCreateMemoryBuffer( inputBufferSize, &rInBuffer ); - - rInBuffer->Lock( &rInByteBuffer, NULL, NULL ); - memcpy( rInByteBuffer, inBuffer, inputBufferSize ); - rInBuffer->Unlock(); - rInByteBuffer = NULL; - - rInBuffer->SetCurrentLength( inputBufferSize ); - - MFCreateSample( &rInSample ); - rInSample->AddBuffer( rInBuffer ); - - // 6. Pass input data to Resampler - - _transform->ProcessInput( 0, rInSample, 0 ); - - SAFE_RELEASE( rInBuffer ); - SAFE_RELEASE( rInSample ); - - // 7. Perform sample rate conversion - - IMFMediaBuffer* rOutBuffer = NULL; - BYTE* rOutByteBuffer = NULL; - - MFT_OUTPUT_DATA_BUFFER rOutDataBuffer; - DWORD rStatus; - DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput - - // 7.1 Create Sample object for output data - - memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer ); - MFCreateSample( &( rOutDataBuffer.pSample ) ); - MFCreateMemoryBuffer( rBytes, &rOutBuffer ); - rOutDataBuffer.pSample->AddBuffer( rOutBuffer ); - rOutDataBuffer.dwStreamID = 0; - rOutDataBuffer.dwStatus = 0; - rOutDataBuffer.pEvents = NULL; - - // 7.2 Get output data from Resampler - - if ( _transform->ProcessOutput( 0, 1, &rOutDataBuffer, &rStatus ) == MF_E_TRANSFORM_NEED_MORE_INPUT ) - { - outSampleCount = 0; - SAFE_RELEASE( rOutBuffer ); - SAFE_RELEASE( rOutDataBuffer.pSample ); - return; - } - - // 7.3 Write output data to outBuffer - - SAFE_RELEASE( rOutBuffer ); - rOutDataBuffer.pSample->ConvertToContiguousBuffer( &rOutBuffer ); - rOutBuffer->GetCurrentLength( &rBytes ); - - rOutBuffer->Lock( &rOutByteBuffer, NULL, NULL ); - memcpy( outBuffer, rOutByteBuffer, rBytes ); - rOutBuffer->Unlock(); - rOutByteBuffer = NULL; - - outSampleCount = rBytes / _bytesPerSample / _channelCount; - SAFE_RELEASE( rOutBuffer ); - SAFE_RELEASE( rOutDataBuffer.pSample ); - } - -private: - unsigned int _bytesPerSample; - unsigned int _channelCount; - float _sampleRatio; - - IUnknown* _transformUnk; - IMFTransform* _transform; - IMFMediaType* _mediaType; - IMFMediaType* _inputMediaType; - IMFMediaType* _outputMediaType; - - #ifdef __IWMResamplerProps_FWD_DEFINED__ - IWMResamplerProps* _resamplerProps; - #endif -}; - -//----------------------------------------------------------------------------- - -// A structure to hold various information related to the WASAPI implementation. -struct WasapiHandle -{ - IAudioClient* captureAudioClient; - IAudioClient* renderAudioClient; - IAudioCaptureClient* captureClient; - IAudioRenderClient* renderClient; - HANDLE captureEvent; - HANDLE renderEvent; - - WasapiHandle() - : captureAudioClient( NULL ), - renderAudioClient( NULL ), - captureClient( NULL ), - renderClient( NULL ), - captureEvent( NULL ), - renderEvent( NULL ) {} -}; - -//============================================================================= - -RtApiWasapi::RtApiWasapi() - : coInitialized_( false ), deviceEnumerator_( NULL ) -{ - // WASAPI can run either apartment or multi-threaded - HRESULT hr = CoInitialize( NULL ); - if ( !FAILED( hr ) ) - coInitialized_ = true; - - // Instantiate device enumerator - hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL, - CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ), - ( void** ) &deviceEnumerator_ ); - - // If this runs on an old Windows, it will fail. Ignore and proceed. - if ( FAILED( hr ) ) - deviceEnumerator_ = NULL; -} - -//----------------------------------------------------------------------------- - -RtApiWasapi::~RtApiWasapi() -{ - if ( stream_.state != STREAM_CLOSED ) - closeStream(); - - SAFE_RELEASE( deviceEnumerator_ ); - - // If this object previously called CoInitialize() - if ( coInitialized_ ) - CoUninitialize(); -} - -//============================================================================= - -unsigned int RtApiWasapi::getDeviceCount( void ) -{ - unsigned int captureDeviceCount = 0; - unsigned int renderDeviceCount = 0; - - IMMDeviceCollection* captureDevices = NULL; - IMMDeviceCollection* renderDevices = NULL; - - if ( !deviceEnumerator_ ) - return 0; - - // Count capture devices - errorText_.clear(); - HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection."; - goto Exit; - } - - hr = captureDevices->GetCount( &captureDeviceCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count."; - goto Exit; - } - - // Count render devices - hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection."; - goto Exit; - } - - hr = renderDevices->GetCount( &renderDeviceCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count."; - goto Exit; - } - -Exit: - // release all references - SAFE_RELEASE( captureDevices ); - SAFE_RELEASE( renderDevices ); - - if ( errorText_.empty() ) - return captureDeviceCount + renderDeviceCount; - - error( RtAudioError::DRIVER_ERROR ); - return 0; -} - -//----------------------------------------------------------------------------- - -RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - unsigned int captureDeviceCount = 0; - unsigned int renderDeviceCount = 0; - std::string defaultDeviceName; - bool isCaptureDevice = false; - - PROPVARIANT deviceNameProp; - PROPVARIANT defaultDeviceNameProp; - - IMMDeviceCollection* captureDevices = NULL; - IMMDeviceCollection* renderDevices = NULL; - IMMDevice* devicePtr = NULL; - IMMDevice* defaultDevicePtr = NULL; - IAudioClient* audioClient = NULL; - IPropertyStore* devicePropStore = NULL; - IPropertyStore* defaultDevicePropStore = NULL; - - WAVEFORMATEX* deviceFormat = NULL; - WAVEFORMATEX* closestMatchFormat = NULL; - - // probed - info.probed = false; - - // Count capture devices - errorText_.clear(); - RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; - HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection."; - goto Exit; - } - - hr = captureDevices->GetCount( &captureDeviceCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count."; - goto Exit; - } - - // Count render devices - hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection."; - goto Exit; - } - - hr = renderDevices->GetCount( &renderDeviceCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count."; - goto Exit; - } - - // validate device index - if ( device >= captureDeviceCount + renderDeviceCount ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index."; - errorType = RtAudioError::INVALID_USE; - goto Exit; - } - - // determine whether index falls within capture or render devices - if ( device >= renderDeviceCount ) { - hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle."; - goto Exit; - } - isCaptureDevice = true; - } - else { - hr = renderDevices->Item( device, &devicePtr ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle."; - goto Exit; - } - isCaptureDevice = false; - } - - // get default device name - if ( isCaptureDevice ) { - hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle."; - goto Exit; - } - } - else { - hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle."; - goto Exit; - } - } - - hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store."; - goto Exit; - } - PropVariantInit( &defaultDeviceNameProp ); - - hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName."; - goto Exit; - } - - defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal); - - // name - hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store."; - goto Exit; - } - - PropVariantInit( &deviceNameProp ); - - hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName."; - goto Exit; - } - - info.name =convertCharPointerToStdString(deviceNameProp.pwszVal); - - // is default - if ( isCaptureDevice ) { - info.isDefaultInput = info.name == defaultDeviceName; - info.isDefaultOutput = false; - } - else { - info.isDefaultInput = false; - info.isDefaultOutput = info.name == defaultDeviceName; - } - - // channel count - hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client."; - goto Exit; - } - - hr = audioClient->GetMixFormat( &deviceFormat ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format."; - goto Exit; - } - - if ( isCaptureDevice ) { - info.inputChannels = deviceFormat->nChannels; - info.outputChannels = 0; - info.duplexChannels = 0; - } - else { - info.inputChannels = 0; - info.outputChannels = deviceFormat->nChannels; - info.duplexChannels = 0; - } - - // sample rates - info.sampleRates.clear(); - - // allow support for all sample rates as we have a built-in sample rate converter - for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) { - info.sampleRates.push_back( SAMPLE_RATES[i] ); - } - info.preferredSampleRate = deviceFormat->nSamplesPerSec; - - // native format - info.nativeFormats = 0; - - if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT || - ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && - ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) ) - { - if ( deviceFormat->wBitsPerSample == 32 ) { - info.nativeFormats |= RTAUDIO_FLOAT32; - } - else if ( deviceFormat->wBitsPerSample == 64 ) { - info.nativeFormats |= RTAUDIO_FLOAT64; - } - } - else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM || - ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && - ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) ) - { - if ( deviceFormat->wBitsPerSample == 8 ) { - info.nativeFormats |= RTAUDIO_SINT8; - } - else if ( deviceFormat->wBitsPerSample == 16 ) { - info.nativeFormats |= RTAUDIO_SINT16; - } - else if ( deviceFormat->wBitsPerSample == 24 ) { - info.nativeFormats |= RTAUDIO_SINT24; - } - else if ( deviceFormat->wBitsPerSample == 32 ) { - info.nativeFormats |= RTAUDIO_SINT32; - } - } - - // probed - info.probed = true; - -Exit: - // release all references - PropVariantClear( &deviceNameProp ); - PropVariantClear( &defaultDeviceNameProp ); - - SAFE_RELEASE( captureDevices ); - SAFE_RELEASE( renderDevices ); - SAFE_RELEASE( devicePtr ); - SAFE_RELEASE( defaultDevicePtr ); - SAFE_RELEASE( audioClient ); - SAFE_RELEASE( devicePropStore ); - SAFE_RELEASE( defaultDevicePropStore ); - - CoTaskMemFree( deviceFormat ); - CoTaskMemFree( closestMatchFormat ); - - if ( !errorText_.empty() ) - error( errorType ); - return info; -} - -//----------------------------------------------------------------------------- - -unsigned int RtApiWasapi::getDefaultOutputDevice( void ) -{ - for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { - if ( getDeviceInfo( i ).isDefaultOutput ) { - return i; - } - } - - return 0; -} - -//----------------------------------------------------------------------------- - -unsigned int RtApiWasapi::getDefaultInputDevice( void ) -{ - for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { - if ( getDeviceInfo( i ).isDefaultInput ) { - return i; - } - } - - return 0; -} - -//----------------------------------------------------------------------------- - -void RtApiWasapi::closeStream( void ) -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiWasapi::closeStream: No open stream to close."; - error( RtAudioError::WARNING ); - return; - } - - if ( stream_.state != STREAM_STOPPED ) - stopStream(); - - // clean up stream memory - SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) - SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) - - SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient ) - SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient ) - - if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ) - CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ); - - if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ) - CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ); - - delete ( WasapiHandle* ) stream_.apiHandle; - stream_.apiHandle = NULL; - - for ( int i = 0; i < 2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - // update stream state - stream_.state = STREAM_CLOSED; -} - -//----------------------------------------------------------------------------- - -void RtApiWasapi::startStream( void ) -{ - verifyStream(); - - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiWasapi::startStream: The stream is already running."; - error( RtAudioError::WARNING ); - return; - } - - #if defined( HAVE_GETTIMEOFDAY ) - gettimeofday( &stream_.lastTickTimestamp, NULL ); - #endif - - // update stream state - stream_.state = STREAM_RUNNING; - - // create WASAPI stream thread - stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL ); - - if ( !stream_.callbackInfo.thread ) { - errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread."; - error( RtAudioError::THREAD_ERROR ); - } - else { - SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority ); - ResumeThread( ( void* ) stream_.callbackInfo.thread ); - } -} - -//----------------------------------------------------------------------------- - -void RtApiWasapi::stopStream( void ) -{ - verifyStream(); - - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiWasapi::stopStream: The stream is already stopped."; - error( RtAudioError::WARNING ); - return; - } - - // inform stream thread by setting stream state to STREAM_STOPPING - stream_.state = STREAM_STOPPING; - - // wait until stream thread is stopped - while( stream_.state != STREAM_STOPPED ) { - Sleep( 1 ); - } - - // Wait for the last buffer to play before stopping. - Sleep( 1000 * stream_.bufferSize / stream_.sampleRate ); - - // close thread handle - if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { - errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread."; - error( RtAudioError::THREAD_ERROR ); - return; - } - - stream_.callbackInfo.thread = (ThreadHandle) NULL; -} - -//----------------------------------------------------------------------------- - -void RtApiWasapi::abortStream( void ) -{ - verifyStream(); - - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiWasapi::abortStream: The stream is already stopped."; - error( RtAudioError::WARNING ); - return; - } - - // inform stream thread by setting stream state to STREAM_STOPPING - stream_.state = STREAM_STOPPING; - - // wait until stream thread is stopped - while ( stream_.state != STREAM_STOPPED ) { - Sleep( 1 ); - } - - // close thread handle - if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { - errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread."; - error( RtAudioError::THREAD_ERROR ); - return; - } - - stream_.callbackInfo.thread = (ThreadHandle) NULL; -} - -//----------------------------------------------------------------------------- - -bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int* bufferSize, - RtAudio::StreamOptions* options ) -{ - bool methodResult = FAILURE; - unsigned int captureDeviceCount = 0; - unsigned int renderDeviceCount = 0; - - IMMDeviceCollection* captureDevices = NULL; - IMMDeviceCollection* renderDevices = NULL; - IMMDevice* devicePtr = NULL; - WAVEFORMATEX* deviceFormat = NULL; - unsigned int bufferBytes; - stream_.state = STREAM_STOPPED; - - // create API Handle if not already created - if ( !stream_.apiHandle ) - stream_.apiHandle = ( void* ) new WasapiHandle(); - - // Count capture devices - errorText_.clear(); - RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; - HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection."; - goto Exit; - } - - hr = captureDevices->GetCount( &captureDeviceCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count."; - goto Exit; - } - - // Count render devices - hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection."; - goto Exit; - } - - hr = renderDevices->GetCount( &renderDeviceCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count."; - goto Exit; - } - - // validate device index - if ( device >= captureDeviceCount + renderDeviceCount ) { - errorType = RtAudioError::INVALID_USE; - errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index."; - goto Exit; - } - - // if device index falls within capture devices - if ( device >= renderDeviceCount ) { - if ( mode != INPUT ) { - errorType = RtAudioError::INVALID_USE; - errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device."; - goto Exit; - } - - // retrieve captureAudioClient from devicePtr - IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; - - hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle."; - goto Exit; - } - - hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, - NULL, ( void** ) &captureAudioClient ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device audio client."; - goto Exit; - } - - hr = captureAudioClient->GetMixFormat( &deviceFormat ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device mix format."; - goto Exit; - } - - stream_.nDeviceChannels[mode] = deviceFormat->nChannels; - captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); - } - - // if device index falls within render devices and is configured for loopback - if ( device < renderDeviceCount && mode == INPUT ) - { - // if renderAudioClient is not initialised, initialise it now - IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; - if ( !renderAudioClient ) - { - probeDeviceOpen( device, OUTPUT, channels, firstChannel, sampleRate, format, bufferSize, options ); - } - - // retrieve captureAudioClient from devicePtr - IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; - - hr = renderDevices->Item( device, &devicePtr ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle."; - goto Exit; - } - - hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, - NULL, ( void** ) &captureAudioClient ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client."; - goto Exit; - } - - hr = captureAudioClient->GetMixFormat( &deviceFormat ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format."; - goto Exit; - } - - stream_.nDeviceChannels[mode] = deviceFormat->nChannels; - captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); - } - - // if device index falls within render devices and is configured for output - if ( device < renderDeviceCount && mode == OUTPUT ) - { - // if renderAudioClient is already initialised, don't initialise it again - IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; - if ( renderAudioClient ) - { - methodResult = SUCCESS; - goto Exit; - } - - hr = renderDevices->Item( device, &devicePtr ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle."; - goto Exit; - } - - hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, - NULL, ( void** ) &renderAudioClient ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client."; - goto Exit; - } - - hr = renderAudioClient->GetMixFormat( &deviceFormat ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format."; - goto Exit; - } - - stream_.nDeviceChannels[mode] = deviceFormat->nChannels; - renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); - } - - // fill stream data - if ( ( stream_.mode == OUTPUT && mode == INPUT ) || - ( stream_.mode == INPUT && mode == OUTPUT ) ) { - stream_.mode = DUPLEX; - } - else { - stream_.mode = mode; - } - - stream_.device[mode] = device; - stream_.doByteSwap[mode] = false; - stream_.sampleRate = sampleRate; - stream_.bufferSize = *bufferSize; - stream_.nBuffers = 1; - stream_.nUserChannels[mode] = channels; - stream_.channelOffset[mode] = firstChannel; - stream_.userFormat = format; - stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats; - - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) - stream_.userInterleaved = false; - else - stream_.userInterleaved = true; - stream_.deviceInterleaved[mode] = true; - - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] || - stream_.nUserChannels[0] != stream_.nDeviceChannels[0] || - stream_.nUserChannels[1] != stream_.nDeviceChannels[1] ) - stream_.doConvertBuffer[mode] = true; - else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - if ( stream_.doConvertBuffer[mode] ) - setConvertInfo( mode, firstChannel ); - - // Allocate necessary internal buffers - bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat ); - - stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 ); - if ( !stream_.userBuffer[mode] ) { - errorType = RtAudioError::MEMORY_ERROR; - errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory."; - goto Exit; - } - - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) - stream_.callbackInfo.priority = 15; - else - stream_.callbackInfo.priority = 0; - - ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback - ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode - - methodResult = SUCCESS; - -Exit: - //clean up - SAFE_RELEASE( captureDevices ); - SAFE_RELEASE( renderDevices ); - SAFE_RELEASE( devicePtr ); - CoTaskMemFree( deviceFormat ); - - // if method failed, close the stream - if ( methodResult == FAILURE ) - closeStream(); - - if ( !errorText_.empty() ) - error( errorType ); - return methodResult; -} - -//============================================================================= - -DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr ) -{ - if ( wasapiPtr ) - ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread(); - - return 0; -} - -DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr ) -{ - if ( wasapiPtr ) - ( ( RtApiWasapi* ) wasapiPtr )->stopStream(); - - return 0; -} - -DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr ) -{ - if ( wasapiPtr ) - ( ( RtApiWasapi* ) wasapiPtr )->abortStream(); - - return 0; -} - -//----------------------------------------------------------------------------- - -void RtApiWasapi::wasapiThread() -{ - // as this is a new thread, we must CoInitialize it - CoInitialize( NULL ); - - HRESULT hr; - - IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; - IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; - IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient; - IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient; - HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent; - HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent; - - WAVEFORMATEX* captureFormat = NULL; - WAVEFORMATEX* renderFormat = NULL; - float captureSrRatio = 0.0f; - float renderSrRatio = 0.0f; - WasapiBuffer captureBuffer; - WasapiBuffer renderBuffer; - WasapiResampler* captureResampler = NULL; - WasapiResampler* renderResampler = NULL; - - // declare local stream variables - RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback; - BYTE* streamBuffer = NULL; - DWORD captureFlags = 0; - unsigned int bufferFrameCount = 0; - unsigned int numFramesPadding = 0; - unsigned int convBufferSize = 0; - bool loopbackEnabled = stream_.device[INPUT] == stream_.device[OUTPUT]; - bool callbackPushed = true; - bool callbackPulled = false; - bool callbackStopped = false; - int callbackResult = 0; - - // convBuffer is used to store converted buffers between WASAPI and the user - char* convBuffer = NULL; - unsigned int convBuffSize = 0; - unsigned int deviceBuffSize = 0; - - std::string errorText; - RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; - - // Attempt to assign "Pro Audio" characteristic to thread - HMODULE AvrtDll = LoadLibraryW( L"AVRT.dll" ); - if ( AvrtDll ) { - DWORD taskIndex = 0; - TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = - ( TAvSetMmThreadCharacteristicsPtr ) (void(*)()) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" ); - AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex ); - FreeLibrary( AvrtDll ); - } - - // start capture stream if applicable - if ( captureAudioClient ) { - hr = captureAudioClient->GetMixFormat( &captureFormat ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; - goto Exit; - } - - // init captureResampler - captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64, - formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT], - captureFormat->nSamplesPerSec, stream_.sampleRate ); - - captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate ); - - if ( !captureClient ) { - hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, - loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK, - 0, - 0, - captureFormat, - NULL ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client."; - goto Exit; - } - - hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ), - ( void** ) &captureClient ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle."; - goto Exit; - } - - // don't configure captureEvent if in loopback mode - if ( !loopbackEnabled ) - { - // configure captureEvent to trigger on every available capture buffer - captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); - if ( !captureEvent ) { - errorType = RtAudioError::SYSTEM_ERROR; - errorText = "RtApiWasapi::wasapiThread: Unable to create capture event."; - goto Exit; - } - - hr = captureAudioClient->SetEventHandle( captureEvent ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to set capture event handle."; - goto Exit; - } - - ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent; - } - - ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient; - - // reset the capture stream - hr = captureAudioClient->Reset(); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to reset capture stream."; - goto Exit; - } - - // start the capture stream - hr = captureAudioClient->Start(); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to start capture stream."; - goto Exit; - } - } - - unsigned int inBufferSize = 0; - hr = captureAudioClient->GetBufferSize( &inBufferSize ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to get capture buffer size."; - goto Exit; - } - - // scale outBufferSize according to stream->user sample rate ratio - unsigned int outBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT]; - inBufferSize *= stream_.nDeviceChannels[INPUT]; - - // set captureBuffer size - captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) ); - } - - // start render stream if applicable - if ( renderAudioClient ) { - hr = renderAudioClient->GetMixFormat( &renderFormat ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; - goto Exit; - } - - // init renderResampler - renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64, - formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT], - stream_.sampleRate, renderFormat->nSamplesPerSec ); - - renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate ); - - if ( !renderClient ) { - hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, - AUDCLNT_STREAMFLAGS_EVENTCALLBACK, - 0, - 0, - renderFormat, - NULL ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to initialize render audio client."; - goto Exit; - } - - hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ), - ( void** ) &renderClient ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle."; - goto Exit; - } - - // configure renderEvent to trigger on every available render buffer - renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); - if ( !renderEvent ) { - errorType = RtAudioError::SYSTEM_ERROR; - errorText = "RtApiWasapi::wasapiThread: Unable to create render event."; - goto Exit; - } - - hr = renderAudioClient->SetEventHandle( renderEvent ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to set render event handle."; - goto Exit; - } - - ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient; - ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent; - - // reset the render stream - hr = renderAudioClient->Reset(); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to reset render stream."; - goto Exit; - } - - // start the render stream - hr = renderAudioClient->Start(); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to start render stream."; - goto Exit; - } - } - - unsigned int outBufferSize = 0; - hr = renderAudioClient->GetBufferSize( &outBufferSize ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to get render buffer size."; - goto Exit; - } - - // scale inBufferSize according to user->stream sample rate ratio - unsigned int inBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT]; - outBufferSize *= stream_.nDeviceChannels[OUTPUT]; - - // set renderBuffer size - renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) ); - } - - // malloc buffer memory - if ( stream_.mode == INPUT ) - { - using namespace std; // for ceilf - convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); - deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); - } - else if ( stream_.mode == OUTPUT ) - { - convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); - deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); - } - else if ( stream_.mode == DUPLEX ) - { - convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), - ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); - deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), - stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); - } - - convBuffSize *= 2; // allow overflow for *SrRatio remainders - convBuffer = ( char* ) calloc( convBuffSize, 1 ); - stream_.deviceBuffer = ( char* ) calloc( deviceBuffSize, 1 ); - if ( !convBuffer || !stream_.deviceBuffer ) { - errorType = RtAudioError::MEMORY_ERROR; - errorText = "RtApiWasapi::wasapiThread: Error allocating device buffer memory."; - goto Exit; - } - - // stream process loop - while ( stream_.state != STREAM_STOPPING ) { - if ( !callbackPulled ) { - // Callback Input - // ============== - // 1. Pull callback buffer from inputBuffer - // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count - // Convert callback buffer to user format - - if ( captureAudioClient ) - { - int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio ); - - convBufferSize = 0; - while ( convBufferSize < stream_.bufferSize ) - { - // Pull callback buffer from inputBuffer - callbackPulled = captureBuffer.pullBuffer( convBuffer, - samplesToPull * stream_.nDeviceChannels[INPUT], - stream_.deviceFormat[INPUT] ); - - if ( !callbackPulled ) - { - break; - } - - // Convert callback buffer to user sample rate - unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); - unsigned int convSamples = 0; - - captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset, - convBuffer, - samplesToPull, - convSamples, - convBufferSize == 0 ? -1 : stream_.bufferSize - convBufferSize ); - - convBufferSize += convSamples; - samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples - } - - if ( callbackPulled ) - { - if ( stream_.doConvertBuffer[INPUT] ) { - // Convert callback buffer to user format - convertBuffer( stream_.userBuffer[INPUT], - stream_.deviceBuffer, - stream_.convertInfo[INPUT] ); - } - else { - // no further conversion, simple copy deviceBuffer to userBuffer - memcpy( stream_.userBuffer[INPUT], - stream_.deviceBuffer, - stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) ); - } - } - } - else { - // if there is no capture stream, set callbackPulled flag - callbackPulled = true; - } - - // Execute Callback - // ================ - // 1. Execute user callback method - // 2. Handle return value from callback - - // if callback has not requested the stream to stop - if ( callbackPulled && !callbackStopped ) { - // Execute user callback method - callbackResult = callback( stream_.userBuffer[OUTPUT], - stream_.userBuffer[INPUT], - stream_.bufferSize, - getStreamTime(), - captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0, - stream_.callbackInfo.userData ); - - // tick stream time - RtApi::tickStreamTime(); - - // Handle return value from callback - if ( callbackResult == 1 ) { - // instantiate a thread to stop this thread - HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL ); - if ( !threadHandle ) { - errorType = RtAudioError::THREAD_ERROR; - errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread."; - goto Exit; - } - else if ( !CloseHandle( threadHandle ) ) { - errorType = RtAudioError::THREAD_ERROR; - errorText = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle."; - goto Exit; - } - - callbackStopped = true; - } - else if ( callbackResult == 2 ) { - // instantiate a thread to stop this thread - HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL ); - if ( !threadHandle ) { - errorType = RtAudioError::THREAD_ERROR; - errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread."; - goto Exit; - } - else if ( !CloseHandle( threadHandle ) ) { - errorType = RtAudioError::THREAD_ERROR; - errorText = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle."; - goto Exit; - } - - callbackStopped = true; - } - } - } - - // Callback Output - // =============== - // 1. Convert callback buffer to stream format - // 2. Convert callback buffer to stream sample rate and channel count - // 3. Push callback buffer into outputBuffer - - if ( renderAudioClient && callbackPulled ) - { - // if the last call to renderBuffer.PushBuffer() was successful - if ( callbackPushed || convBufferSize == 0 ) - { - if ( stream_.doConvertBuffer[OUTPUT] ) - { - // Convert callback buffer to stream format - convertBuffer( stream_.deviceBuffer, - stream_.userBuffer[OUTPUT], - stream_.convertInfo[OUTPUT] ); - - } - else { - // no further conversion, simple copy userBuffer to deviceBuffer - memcpy( stream_.deviceBuffer, - stream_.userBuffer[OUTPUT], - stream_.bufferSize * stream_.nUserChannels[OUTPUT] * formatBytes( stream_.userFormat ) ); - } - - // Convert callback buffer to stream sample rate - renderResampler->Convert( convBuffer, - stream_.deviceBuffer, - stream_.bufferSize, - convBufferSize ); - } - - // Push callback buffer into outputBuffer - callbackPushed = renderBuffer.pushBuffer( convBuffer, - convBufferSize * stream_.nDeviceChannels[OUTPUT], - stream_.deviceFormat[OUTPUT] ); - } - else { - // if there is no render stream, set callbackPushed flag - callbackPushed = true; - } - - // Stream Capture - // ============== - // 1. Get capture buffer from stream - // 2. Push capture buffer into inputBuffer - // 3. If 2. was successful: Release capture buffer - - if ( captureAudioClient ) { - // if the callback input buffer was not pulled from captureBuffer, wait for next capture event - if ( !callbackPulled ) { - WaitForSingleObject( loopbackEnabled ? renderEvent : captureEvent, INFINITE ); - } - - // Get capture buffer from stream - hr = captureClient->GetBuffer( &streamBuffer, - &bufferFrameCount, - &captureFlags, NULL, NULL ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer."; - goto Exit; - } - - if ( bufferFrameCount != 0 ) { - // Push capture buffer into inputBuffer - if ( captureBuffer.pushBuffer( ( char* ) streamBuffer, - bufferFrameCount * stream_.nDeviceChannels[INPUT], - stream_.deviceFormat[INPUT] ) ) - { - // Release capture buffer - hr = captureClient->ReleaseBuffer( bufferFrameCount ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; - goto Exit; - } - } - else - { - // Inform WASAPI that capture was unsuccessful - hr = captureClient->ReleaseBuffer( 0 ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; - goto Exit; - } - } - } - else - { - // Inform WASAPI that capture was unsuccessful - hr = captureClient->ReleaseBuffer( 0 ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; - goto Exit; - } - } - } - - // Stream Render - // ============= - // 1. Get render buffer from stream - // 2. Pull next buffer from outputBuffer - // 3. If 2. was successful: Fill render buffer with next buffer - // Release render buffer - - if ( renderAudioClient ) { - // if the callback output buffer was not pushed to renderBuffer, wait for next render event - if ( callbackPulled && !callbackPushed ) { - WaitForSingleObject( renderEvent, INFINITE ); - } - - // Get render buffer from stream - hr = renderAudioClient->GetBufferSize( &bufferFrameCount ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size."; - goto Exit; - } - - hr = renderAudioClient->GetCurrentPadding( &numFramesPadding ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding."; - goto Exit; - } - - bufferFrameCount -= numFramesPadding; - - if ( bufferFrameCount != 0 ) { - hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer."; - goto Exit; - } - - // Pull next buffer from outputBuffer - // Fill render buffer with next buffer - if ( renderBuffer.pullBuffer( ( char* ) streamBuffer, - bufferFrameCount * stream_.nDeviceChannels[OUTPUT], - stream_.deviceFormat[OUTPUT] ) ) - { - // Release render buffer - hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer."; - goto Exit; - } - } - else - { - // Inform WASAPI that render was unsuccessful - hr = renderClient->ReleaseBuffer( 0, 0 ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer."; - goto Exit; - } - } - } - else - { - // Inform WASAPI that render was unsuccessful - hr = renderClient->ReleaseBuffer( 0, 0 ); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer."; - goto Exit; - } - } - } - - // if the callback buffer was pushed renderBuffer reset callbackPulled flag - if ( callbackPushed ) { - // unsetting the callbackPulled flag lets the stream know that - // the audio device is ready for another callback output buffer. - callbackPulled = false; - } - - } - -Exit: - // clean up - CoTaskMemFree( captureFormat ); - CoTaskMemFree( renderFormat ); - - free ( convBuffer ); - delete renderResampler; - delete captureResampler; - - CoUninitialize(); - - // update stream state - stream_.state = STREAM_STOPPED; - - if ( !errorText.empty() ) - { - errorText_ = errorText; - error( errorType ); - } -} - -//******************** End of __WINDOWS_WASAPI__ *********************// -#endif - - -#if defined(__WINDOWS_DS__) // Windows DirectSound API - -// Modified by Robin Davies, October 2005 -// - Improvements to DirectX pointer chasing. -// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30. -// - Auto-call CoInitialize for DSOUND and ASIO platforms. -// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007 -// Changed device query structure for RtAudio 4.0.7, January 2010 - -#include -#include -#include -#include -#include -#include -#include - -#if defined(__MINGW32__) - // missing from latest mingw winapi -#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ -#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ -#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ -#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */ -#endif - -#define MINIMUM_DEVICE_BUFFER_SIZE 32768 - -#ifdef _MSC_VER // if Microsoft Visual C++ -#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. -#endif - -static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) -{ - if ( pointer > bufferSize ) pointer -= bufferSize; - if ( laterPointer < earlierPointer ) laterPointer += bufferSize; - if ( pointer < earlierPointer ) pointer += bufferSize; - return pointer >= earlierPointer && pointer < laterPointer; -} - -// A structure to hold various information related to the DirectSound -// API implementation. -struct DsHandle { - unsigned int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - void *id[2]; - void *buffer[2]; - bool xrun[2]; - UINT bufferPointer[2]; - DWORD dsBufferSize[2]; - DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. - HANDLE condition; - - DsHandle() - :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } -}; - -// Declarations for utility functions, callbacks, and structures -// specific to the DirectSound implementation. -static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, - LPCTSTR description, - LPCTSTR module, - LPVOID lpContext ); - -static const char* getErrorString( int code ); - -static unsigned __stdcall callbackHandler( void *ptr ); - -struct DsDevice { - LPGUID id[2]; - bool validId[2]; - bool found; - std::string name; - - DsDevice() - : found(false) { validId[0] = false; validId[1] = false; } -}; - -struct DsProbeData { - bool isInput; - std::vector* dsDevices; -}; - -RtApiDs :: RtApiDs() -{ - // Dsound will run both-threaded. If CoInitialize fails, then just - // accept whatever the mainline chose for a threading model. - coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); - if ( !FAILED( hr ) ) coInitialized_ = true; -} - -RtApiDs :: ~RtApiDs() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); - if ( coInitialized_ ) CoUninitialize(); // balanced call. -} - -// The DirectSound default output is always the first device. -unsigned int RtApiDs :: getDefaultOutputDevice( void ) -{ - return 0; -} - -// The DirectSound default input is always the first input device, -// which is the first capture device enumerated. -unsigned int RtApiDs :: getDefaultInputDevice( void ) -{ - return 0; -} - -unsigned int RtApiDs :: getDeviceCount( void ) -{ - // Set query flag for previously found devices to false, so that we - // can check for any devices that have disappeared. - for ( unsigned int i=0; i(dsDevices.size()); -} - -RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - if ( dsDevices.size() == 0 ) { - // Force a query of all devices - getDeviceCount(); - if ( dsDevices.size() == 0 ) { - errorText_ = "RtApiDs::getDeviceInfo: no devices found!"; - error( RtAudioError::INVALID_USE ); - return info; - } - } - - if ( device >= dsDevices.size() ) { - errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - HRESULT result; - if ( dsDevices[ device ].validId[0] == false ) goto probeInput; - - LPDIRECTSOUND output; - DSCAPS outCaps; - result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto probeInput; - } - - outCaps.dwSize = sizeof( outCaps ); - result = output->GetCaps( &outCaps ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto probeInput; - } - - // Get output channel information. - info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; - - // Get sample rate information. - info.sampleRates.clear(); - for ( unsigned int k=0; k= (unsigned int) outCaps.dwMinSecondarySampleRate && - SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) { - info.sampleRates.push_back( SAMPLE_RATES[k] ); - - if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) - info.preferredSampleRate = SAMPLE_RATES[k]; - } - } - - // Get format information. - if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; - if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; - - output->Release(); - - if ( getDefaultOutputDevice() == device ) - info.isDefaultOutput = true; - - if ( dsDevices[ device ].validId[1] == false ) { - info.name = dsDevices[ device ].name; - info.probed = true; - return info; - } - - probeInput: - - LPDIRECTSOUNDCAPTURE input; - result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - DSCCAPS inCaps; - inCaps.dwSize = sizeof( inCaps ); - result = input->GetCaps( &inCaps ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Get input channel information. - info.inputChannels = inCaps.dwChannels; - - // Get sample rate and format information. - std::vector rates; - if ( inCaps.dwChannels >= 2 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; - - if ( info.nativeFormats & RTAUDIO_SINT16 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 ); - } - else if ( info.nativeFormats & RTAUDIO_SINT8 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 ); - } - } - else if ( inCaps.dwChannels == 1 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; - - if ( info.nativeFormats & RTAUDIO_SINT16 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 ); - } - else if ( info.nativeFormats & RTAUDIO_SINT8 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 ); - } - } - else info.inputChannels = 0; // technically, this would be an error - - input->Release(); - - if ( info.inputChannels == 0 ) return info; - - // Copy the supported rates to the info structure but avoid duplication. - bool found; - for ( unsigned int i=0; i 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - if ( device == 0 ) info.isDefaultInput = true; - - // Copy name and return. - info.name = dsDevices[ device ].name; - info.probed = true; - return info; -} - -bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - if ( channels + firstChannel > 2 ) { - errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; - return FAILURE; - } - - size_t nDevices = dsDevices.size(); - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiDs::probeDeviceOpen: no devices found!"; - return FAILURE; - } - - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - if ( mode == OUTPUT ) { - if ( dsDevices[ device ].validId[0] == false ) { - errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - else { // mode == INPUT - if ( dsDevices[ device ].validId[1] == false ) { - errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // According to a note in PortAudio, using GetDesktopWindow() - // instead of GetForegroundWindow() is supposed to avoid problems - // that occur when the application's window is not the foreground - // window. Also, if the application window closes before the - // DirectSound buffer, DirectSound can crash. In the past, I had - // problems when using GetDesktopWindow() but it seems fine now - // (January 2010). I'll leave it commented here. - // HWND hWnd = GetForegroundWindow(); - HWND hWnd = GetDesktopWindow(); - - // Check the numberOfBuffers parameter and limit the lowest value to - // two. This is a judgement call and a value of two is probably too - // low for capture, but it should work for playback. - int nBuffers = 0; - if ( options ) nBuffers = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; - if ( nBuffers < 2 ) nBuffers = 3; - - // Check the lower range of the user-specified buffer size and set - // (arbitrarily) to a lower bound of 32. - if ( *bufferSize < 32 ) *bufferSize = 32; - - // Create the wave format structure. The data format setting will - // be determined later. - WAVEFORMATEX waveFormat; - ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); - waveFormat.wFormatTag = WAVE_FORMAT_PCM; - waveFormat.nChannels = channels + firstChannel; - waveFormat.nSamplesPerSec = (unsigned long) sampleRate; - - // Determine the device buffer size. By default, we'll use the value - // defined above (32K), but we will grow it to make allowances for - // very large software buffer sizes. - DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE; - DWORD dsPointerLeadTime = 0; - - void *ohandle = 0, *bhandle = 0; - HRESULT result; - if ( mode == OUTPUT ) { - - LPDIRECTSOUND output; - result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - DSCAPS outCaps; - outCaps.dwSize = sizeof( outCaps ); - result = output->GetCaps( &outCaps ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check channel information. - if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check format information. Use 16-bit format unless not - // supported or user requests 8-bit. - if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && - !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - stream_.userFormat = format; - - // Update wave format structure and buffer information. - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; - - // If the user wants an even bigger buffer, increase the device buffer size accordingly. - while ( dsPointerLeadTime * 2U > dsBufferSize ) - dsBufferSize *= 2; - - // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. - // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); - // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. - result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Even though we will write to the secondary buffer, we need to - // access the primary buffer to set the correct output format - // (since the default is 8-bit, 22 kHz!). Setup the DS primary - // buffer description. - DSBUFFERDESC bufferDescription; - ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSBUFFERDESC ); - bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; - - // Obtain the primary buffer - LPDIRECTSOUNDBUFFER buffer; - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the primary DS buffer sound format. - result = buffer->SetFormat( &waveFormat ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Setup the secondary DS buffer description. - ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSBUFFERDESC ); - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GLOBALFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCHARDWARE ); // Force hardware mixing - bufferDescription.dwBufferBytes = dsBufferSize; - bufferDescription.lpwfxFormat = &waveFormat; - - // Try to create the secondary DS buffer. If that doesn't work, - // try to use software mixing. Otherwise, there's a problem. - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GLOBALFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCSOFTWARE ); // Force software mixing - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Get the buffer size ... might be different from what we specified. - DSBCAPS dsbcaps; - dsbcaps.dwSize = sizeof( DSBCAPS ); - result = buffer->GetCaps( &dsbcaps ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - dsBufferSize = dsbcaps.dwBufferBytes; - - // Lock the DS buffer - LPVOID audioPtr; - DWORD dataLen; - result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); - - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - ohandle = (void *) output; - bhandle = (void *) buffer; - } - - if ( mode == INPUT ) { - - LPDIRECTSOUNDCAPTURE input; - result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - DSCCAPS inCaps; - inCaps.dwSize = sizeof( inCaps ); - result = input->GetCaps( &inCaps ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check channel information. - if ( inCaps.dwChannels < channels + firstChannel ) { - errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; - return FAILURE; - } - - // Check format information. Use 16-bit format unless user - // requests 8-bit. - DWORD deviceFormats; - if ( channels + firstChannel == 2 ) { - deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; - if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - else { // assume 16-bit is supported - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - } - else { // channel == 1 - deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; - if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - else { // assume 16-bit is supported - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - } - stream_.userFormat = format; - - // Update wave format structure and buffer information. - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; - - // If the user wants an even bigger buffer, increase the device buffer size accordingly. - while ( dsPointerLeadTime * 2U > dsBufferSize ) - dsBufferSize *= 2; - - // Setup the secondary DS buffer description. - DSCBUFFERDESC bufferDescription; - ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); - bufferDescription.dwFlags = 0; - bufferDescription.dwReserved = 0; - bufferDescription.dwBufferBytes = dsBufferSize; - bufferDescription.lpwfxFormat = &waveFormat; - - // Create the capture buffer. - LPDIRECTSOUNDCAPTUREBUFFER buffer; - result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Get the buffer size ... might be different from what we specified. - DSCBCAPS dscbcaps; - dscbcaps.dwSize = sizeof( DSCBCAPS ); - result = buffer->GetCaps( &dscbcaps ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - dsBufferSize = dscbcaps.dwBufferBytes; - - // NOTE: We could have a problem here if this is a duplex stream - // and the play and capture hardware buffer sizes are different - // (I'm actually not sure if that is a problem or not). - // Currently, we are not verifying that. - - // Lock the capture buffer - LPVOID audioPtr; - DWORD dataLen; - result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Zero the buffer - ZeroMemory( audioPtr, dataLen ); - - // Unlock the buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - ohandle = (void *) input; - bhandle = (void *) buffer; - } - - // Set various stream parameters - DsHandle *handle = 0; - stream_.nDeviceChannels[mode] = channels + firstChannel; - stream_.nUserChannels[mode] = channels; - stream_.bufferSize = *bufferSize; - stream_.channelOffset[mode] = firstChannel; - stream_.deviceInterleaved[mode] = true; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - - // Set flag for buffer conversion - stream_.doConvertBuffer[mode] = false; - if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) - stream_.doConvertBuffer[mode] = true; - if (stream_.userFormat != stream_.deviceFormat[mode]) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers - long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - // Allocate our DsHandle structures for the stream. - if ( stream_.apiHandle == 0 ) { - try { - handle = new DsHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; - goto error; - } - - // Create a manual-reset event. - handle->condition = CreateEvent( NULL, // no security - TRUE, // manual-reset - FALSE, // non-signaled initially - NULL ); // unnamed - stream_.apiHandle = (void *) handle; - } - else - handle = (DsHandle *) stream_.apiHandle; - handle->id[mode] = ohandle; - handle->buffer[mode] = bhandle; - handle->dsBufferSize[mode] = dsBufferSize; - handle->dsPointerLeadTime[mode] = dsPointerLeadTime; - - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up an output stream. - stream_.mode = DUPLEX; - else - stream_.mode = mode; - stream_.nBuffers = nBuffers; - stream_.sampleRate = sampleRate; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - - // Setup the callback thread. - if ( stream_.callbackInfo.isRunning == false ) { - unsigned threadId; - stream_.callbackInfo.isRunning = true; - stream_.callbackInfo.object = (void *) this; - stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, - &stream_.callbackInfo, 0, &threadId ); - if ( stream_.callbackInfo.thread == 0 ) { - errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; - goto error; - } - - // Boost DS thread priority - SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); - } - return SUCCESS; - - error: - if ( handle ) { - if ( handle->buffer[0] ) { // the object pointer can be NULL and valid - LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - if ( buffer ) buffer->Release(); - object->Release(); - } - if ( handle->buffer[1] ) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - if ( buffer ) buffer->Release(); - object->Release(); - } - CloseHandle( handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.state = STREAM_CLOSED; - return FAILURE; -} - -void RtApiDs :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiDs::closeStream(): no open stream to close!"; - error( RtAudioError::WARNING ); - return; - } - - // Stop the callback thread. - stream_.callbackInfo.isRunning = false; - WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); - CloseHandle( (HANDLE) stream_.callbackInfo.thread ); - - DsHandle *handle = (DsHandle *) stream_.apiHandle; - if ( handle ) { - if ( handle->buffer[0] ) { // the object pointer can be NULL and valid - LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - if ( buffer ) { - buffer->Stop(); - buffer->Release(); - } - object->Release(); - } - if ( handle->buffer[1] ) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - if ( buffer ) { - buffer->Stop(); - buffer->Release(); - } - object->Release(); - } - CloseHandle( handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiDs :: startStream() -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiDs::startStream(): the stream is already running!"; - error( RtAudioError::WARNING ); - return; - } - - #if defined( HAVE_GETTIMEOFDAY ) - gettimeofday( &stream_.lastTickTimestamp, NULL ); - #endif - - DsHandle *handle = (DsHandle *) stream_.apiHandle; - - // Increase scheduler frequency on lesser windows (a side-effect of - // increasing timer accuracy). On greater windows (Win2K or later), - // this is already in effect. - timeBeginPeriod( 1 ); - - buffersRolling = false; - duplexPrerollBytes = 0; - - if ( stream_.mode == DUPLEX ) { - // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize. - duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] ); - } - - HRESULT result = 0; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - result = buffer->Start( DSCBSTART_LOOPING ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - handle->drainCounter = 0; - handle->internalDrain = false; - ResetEvent( handle->condition ); - stream_.state = STREAM_RUNNING; - - unlock: - if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiDs :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - HRESULT result = 0; - LPVOID audioPtr; - DWORD dataLen; - DsHandle *handle = (DsHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 2; - WaitForSingleObject( handle->condition, INFINITE ); // block until signaled - } - - stream_.state = STREAM_STOPPED; - - MUTEX_LOCK( &stream_.mutex ); - - // Stop the buffer and clear memory - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = buffer->Stop(); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); - - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // If we start playing again, we must begin at beginning of buffer. - handle->bufferPointer[0] = 0; - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - audioPtr = NULL; - dataLen = 0; - - stream_.state = STREAM_STOPPED; - - if ( stream_.mode != DUPLEX ) - MUTEX_LOCK( &stream_.mutex ); - - result = buffer->Stop(); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); - - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // If we start recording again, we must begin at beginning of buffer. - handle->bufferPointer[1] = 0; - } - - unlock: - timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. - MUTEX_UNLOCK( &stream_.mutex ); - - if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiDs :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - DsHandle *handle = (DsHandle *) stream_.apiHandle; - handle->drainCounter = 2; - - stopStream(); -} - -void RtApiDs :: callbackEvent() -{ - if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) { - Sleep( 50 ); // sleep 50 milliseconds - return; - } - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtAudioError::WARNING ); - return; - } - - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - DsHandle *handle = (DsHandle *) stream_.apiHandle; - - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > stream_.nBuffers + 2 ) { - - stream_.state = STREAM_STOPPING; - if ( handle->internalDrain == false ) - SetEvent( handle->condition ); - else - stopStream(); - return; - } - - // Invoke user callback to get fresh output data UNLESS we are - // draining stream. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( cbReturnValue == 2 ) { - stream_.state = STREAM_STOPPING; - handle->drainCounter = 2; - abortStream(); - return; - } - else if ( cbReturnValue == 1 ) { - handle->drainCounter = 1; - handle->internalDrain = true; - } - } - - HRESULT result; - DWORD currentWritePointer, safeWritePointer; - DWORD currentReadPointer, safeReadPointer; - UINT nextWritePointer; - - LPVOID buffer1 = NULL; - LPVOID buffer2 = NULL; - DWORD bufferSize1 = 0; - DWORD bufferSize2 = 0; - - char *buffer; - long bufferBytes; - - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - if ( buffersRolling == false ) { - if ( stream_.mode == DUPLEX ) { - //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); - - // It takes a while for the devices to get rolling. As a result, - // there's no guarantee that the capture and write device pointers - // will move in lockstep. Wait here for both devices to start - // rolling, and then set our buffer pointers accordingly. - // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 - // bytes later than the write buffer. - - // Stub: a serious risk of having a pre-emptive scheduling round - // take place between the two GetCurrentPosition calls... but I'm - // really not sure how to solve the problem. Temporarily boost to - // Realtime priority, maybe; but I'm not sure what priority the - // DirectSound service threads run at. We *should* be roughly - // within a ms or so of correct. - - LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - - DWORD startSafeWritePointer, startSafeReadPointer; - - result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - while ( true ) { - result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break; - Sleep( 1 ); - } - - //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); - - handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; - if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; - handle->bufferPointer[1] = safeReadPointer; - } - else if ( stream_.mode == OUTPUT ) { - - // Set the proper nextWritePosition after initial startup. - LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; - if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; - } - - buffersRolling = true; - } - - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - memset( stream_.userBuffer[0], 0, bufferBytes ); - } - - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; - bufferBytes *= formatBytes( stream_.deviceFormat[0] ); - } - else { - buffer = stream_.userBuffer[0]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - } - - // No byte swapping necessary in DirectSound implementation. - - // Ahhh ... windoze. 16-bit data is signed but 8-bit data is - // unsigned. So, we need to convert our signed 8-bit data here to - // unsigned. - if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) - for ( int i=0; idsBufferSize[0]; - nextWritePointer = handle->bufferPointer[0]; - - DWORD endWrite, leadPointer; - while ( true ) { - // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - - // We will copy our output buffer into the region between - // safeWritePointer and leadPointer. If leadPointer is not - // beyond the next endWrite position, wait until it is. - leadPointer = safeWritePointer + handle->dsPointerLeadTime[0]; - //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl; - if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize; - if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset - endWrite = nextWritePointer + bufferBytes; - - // Check whether the entire write region is behind the play pointer. - if ( leadPointer >= endWrite ) break; - - // If we are here, then we must wait until the leadPointer advances - // beyond the end of our next write region. We use the - // Sleep() function to suspend operation until that happens. - double millis = ( endWrite - leadPointer ) * 1000.0; - millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - } - - if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize ) - || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { - // We've strayed into the forbidden zone ... resync the read pointer. - handle->xrun[0] = true; - nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes; - if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize; - handle->bufferPointer[0] = nextWritePointer; - endWrite = nextWritePointer + bufferBytes; - } - - // Lock free space in the buffer - result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - - // Copy our buffer into the DS buffer - CopyMemory( buffer1, buffer, bufferSize1 ); - if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); - - // Update our buffer offset and unlock sound buffer - dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize; - handle->bufferPointer[0] = nextWritePointer; - } - - // Don't bother draining input - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; - bufferBytes *= formatBytes( stream_.deviceFormat[1] ); - } - else { - buffer = stream_.userBuffer[1]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; - bufferBytes *= formatBytes( stream_.userFormat ); - } - - LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - long nextReadPointer = handle->bufferPointer[1]; - DWORD dsBufferSize = handle->dsBufferSize[1]; - - // Find out where the write and "safe read" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - - if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset - DWORD endRead = nextReadPointer + bufferBytes; - - // Handling depends on whether we are INPUT or DUPLEX. - // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, - // then a wait here will drag the write pointers into the forbidden zone. - // - // In DUPLEX mode, rather than wait, we will back off the read pointer until - // it's in a safe position. This causes dropouts, but it seems to be the only - // practical way to sync up the read and write pointers reliably, given the - // the very complex relationship between phase and increment of the read and write - // pointers. - // - // In order to minimize audible dropouts in DUPLEX mode, we will - // provide a pre-roll period of 0.5 seconds in which we return - // zeros from the read buffer while the pointers sync up. - - if ( stream_.mode == DUPLEX ) { - if ( safeReadPointer < endRead ) { - if ( duplexPrerollBytes <= 0 ) { - // Pre-roll time over. Be more agressive. - int adjustment = endRead-safeReadPointer; - - handle->xrun[1] = true; - // Two cases: - // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, - // and perform fine adjustments later. - // - small adjustments: back off by twice as much. - if ( adjustment >= 2*bufferBytes ) - nextReadPointer = safeReadPointer-2*bufferBytes; - else - nextReadPointer = safeReadPointer-bufferBytes-adjustment; - - if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; - - } - else { - // In pre=roll time. Just do it. - nextReadPointer = safeReadPointer - bufferBytes; - while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; - } - endRead = nextReadPointer + bufferBytes; - } - } - else { // mode == INPUT - while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) { - // See comments for playback. - double millis = (endRead - safeReadPointer) * 1000.0; - millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - - // Wake up and find out where we are now. - result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - - if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset - } - } - - // Lock free space in the buffer - result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - - if ( duplexPrerollBytes <= 0 ) { - // Copy our buffer into the DS buffer - CopyMemory( buffer, buffer1, bufferSize1 ); - if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); - } - else { - memset( buffer, 0, bufferSize1 ); - if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); - duplexPrerollBytes -= bufferSize1 + bufferSize2; - } - - // Update our buffer offset and unlock sound buffer - nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize; - dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - handle->bufferPointer[1] = nextReadPointer; - - // No byte swapping necessary in DirectSound implementation. - - // If necessary, convert 8-bit data from unsigned to signed. - if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 ) - for ( int j=0; jobject; - bool* isRunning = &info->isRunning; - - while ( *isRunning == true ) { - object->callbackEvent(); - } - - _endthreadex( 0 ); - return 0; -} - -static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, - LPCTSTR description, - LPCTSTR /*module*/, - LPVOID lpContext ) -{ - struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext; - std::vector& dsDevices = *probeInfo.dsDevices; - - HRESULT hr; - bool validDevice = false; - if ( probeInfo.isInput == true ) { - DSCCAPS caps; - LPDIRECTSOUNDCAPTURE object; - - hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); - if ( hr != DS_OK ) return TRUE; - - caps.dwSize = sizeof(caps); - hr = object->GetCaps( &caps ); - if ( hr == DS_OK ) { - if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) - validDevice = true; - } - object->Release(); - } - else { - DSCAPS caps; - LPDIRECTSOUND object; - hr = DirectSoundCreate( lpguid, &object, NULL ); - if ( hr != DS_OK ) return TRUE; - - caps.dwSize = sizeof(caps); - hr = object->GetCaps( &caps ); - if ( hr == DS_OK ) { - if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) - validDevice = true; - } - object->Release(); - } - - // If good device, then save its name and guid. - std::string name = convertCharPointerToStdString( description ); - //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" ) - if ( lpguid == NULL ) - name = "Default Device"; - if ( validDevice ) { - for ( unsigned int i=0; i -#include - - // A structure to hold various information related to the ALSA API - // implementation. -struct AlsaHandle { - snd_pcm_t *handles[2]; - bool synchronized; - bool xrun[2]; - pthread_cond_t runnable_cv; - bool runnable; - - AlsaHandle() -#if _cplusplus >= 201103L - :handles{nullptr, nullptr}, synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; } -#else - : synchronized(false), runnable(false) { handles[0] = NULL; handles[1] = NULL; xrun[0] = false; xrun[1] = false; } -#endif -}; - -static void *alsaCallbackHandler( void * ptr ); - -RtApiAlsa :: RtApiAlsa() -{ - // Nothing to do here. -} - -RtApiAlsa :: ~RtApiAlsa() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiAlsa :: getDeviceCount( void ) -{ - unsigned nDevices = 0; - int result, subdevice, card; - char name[64]; - snd_ctl_t *handle = 0; - - strcpy(name, "default"); - result = snd_ctl_open( &handle, "default", 0 ); - if (result == 0) { - nDevices++; - snd_ctl_close( handle ); - } - - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &handle, name, 0 ); - if ( result < 0 ) { - handle = 0; - errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto nextcard; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( handle, &subdevice ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - break; - } - if ( subdevice < 0 ) - break; - nDevices++; - } - nextcard: - if ( handle ) - snd_ctl_close( handle ); - snd_card_next( &card ); - } - - return nDevices; -} - -RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - unsigned nDevices = 0; - int result=-1, subdevice=-1, card=-1; - char name[64]; - snd_ctl_t *chandle = 0; - - result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); - if ( result == 0 ) { - if ( nDevices++ == device ) { - strcpy( name, "default" ); - goto foundDevice; - } - } - if ( chandle ) - snd_ctl_close( chandle ); - - // Count cards and devices - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); - if ( result < 0 ) { - chandle = 0; - errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto nextcard; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - break; - } - if ( subdevice < 0 ) break; - if ( nDevices == device ) { - sprintf( name, "hw:%d,%d", card, subdevice ); - goto foundDevice; - } - nDevices++; - } - nextcard: - if ( chandle ) - snd_ctl_close( chandle ); - snd_card_next( &card ); - } - - if ( nDevices == 0 ) { - errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - if ( device >= nDevices ) { - errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - foundDevice: - - // If a stream is already open, we cannot probe the stream devices. - // Thus, use the saved results. - if ( stream_.state != STREAM_CLOSED && - ( stream_.device[0] == device || stream_.device[1] == device ) ) { - snd_ctl_close( chandle ); - if ( device >= devices_.size() ) { - errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; - error( RtAudioError::WARNING ); - return info; - } - return devices_[ device ]; - } - - int openMode = SND_PCM_ASYNC; - snd_pcm_stream_t stream; - snd_pcm_info_t *pcminfo; - snd_pcm_info_alloca( &pcminfo ); - snd_pcm_t *phandle; - snd_pcm_hw_params_t *params; - snd_pcm_hw_params_alloca( ¶ms ); - - // First try for playback unless default device (which has subdev -1) - stream = SND_PCM_STREAM_PLAYBACK; - snd_pcm_info_set_stream( pcminfo, stream ); - if ( subdevice != -1 ) { - snd_pcm_info_set_device( pcminfo, subdevice ); - snd_pcm_info_set_subdevice( pcminfo, 0 ); - - result = snd_ctl_pcm_info( chandle, pcminfo ); - if ( result < 0 ) { - // Device probably doesn't support playback. - goto captureProbe; - } - } - - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto captureProbe; - } - - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto captureProbe; - } - - // Get output channel information. - unsigned int value; - result = snd_pcm_hw_params_get_channels_max( params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto captureProbe; - } - info.outputChannels = value; - snd_pcm_close( phandle ); - - captureProbe: - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream( pcminfo, stream ); - - // Now try for capture unless default device (with subdev = -1) - if ( subdevice != -1 ) { - result = snd_ctl_pcm_info( chandle, pcminfo ); - snd_ctl_close( chandle ); - if ( result < 0 ) { - // Device probably doesn't support capture. - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - } - else - snd_ctl_close( chandle ); - - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - - result = snd_pcm_hw_params_get_channels_max( params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - info.inputChannels = value; - snd_pcm_close( phandle ); - - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // ALSA doesn't provide default devices so we'll use the first available one. - if ( device == 0 && info.outputChannels > 0 ) - info.isDefaultOutput = true; - if ( device == 0 && info.inputChannels > 0 ) - info.isDefaultInput = true; - - probeParameters: - // At this point, we just need to figure out the supported data - // formats and sample rates. We'll proceed by opening the device in - // the direction with the maximum number of channels, or playback if - // they are equal. This might limit our sample rate options, but so - // be it. - - if ( info.outputChannels >= info.inputChannels ) - stream = SND_PCM_STREAM_PLAYBACK; - else - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream( pcminfo, stream ); - - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Test our discrete set of sample rate values. - info.sampleRates.clear(); - for ( unsigned int i=0; i info.preferredSampleRate ) ) - info.preferredSampleRate = SAMPLE_RATES[i]; - } - } - if ( info.sampleRates.size() == 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Probe the supported data formats ... we don't care about endian-ness just yet - snd_pcm_format_t format; - info.nativeFormats = 0; - format = SND_PCM_FORMAT_S8; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT8; - format = SND_PCM_FORMAT_S16; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT16; - format = SND_PCM_FORMAT_S24; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT24; - format = SND_PCM_FORMAT_S32; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT32; - format = SND_PCM_FORMAT_FLOAT; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_FLOAT32; - format = SND_PCM_FORMAT_FLOAT64; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_FLOAT64; - - // Check that we have at least one supported format - if ( info.nativeFormats == 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Get the device name - if (strncmp(name, "default", 7)!=0) { - char *cardname; - result = snd_card_get_name( card, &cardname ); - if ( result >= 0 ) { - sprintf( name, "hw:%s,%d", cardname, subdevice ); - free( cardname ); - } - } - info.name = name; - - // That's all ... close the device and return - snd_pcm_close( phandle ); - info.probed = true; - return info; -} - -void RtApiAlsa :: saveDeviceInfo( void ) -{ - devices_.clear(); - - unsigned int nDevices = getDeviceCount(); - devices_.resize( nDevices ); - for ( unsigned int i=0; iflags & RTAUDIO_ALSA_USE_DEFAULT) ) - { - strcpy(name, "default"); - result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); - if ( result == 0 ) { - if ( nDevices == device ) { - strcpy( name, "default" ); - snd_ctl_close( chandle ); - goto foundDevice; - } - nDevices++; - } - } - - else { - nDevices++; - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) break; - if ( subdevice < 0 ) break; - if ( nDevices == device ) { - sprintf( name, "hw:%d,%d", card, subdevice ); - snd_ctl_close( chandle ); - goto foundDevice; - } - nDevices++; - } - snd_ctl_close( chandle ); - snd_card_next( &card ); - } - - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!"; - return FAILURE; - } - - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - } - - foundDevice: - - // The getDeviceInfo() function will not work for a device that is - // already open. Thus, we'll probe the system before opening a - // stream and save the results for use by getDeviceInfo(). - if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once - this->saveDeviceInfo(); - - snd_pcm_stream_t stream; - if ( mode == OUTPUT ) - stream = SND_PCM_STREAM_PLAYBACK; - else - stream = SND_PCM_STREAM_CAPTURE; - - snd_pcm_t *phandle; - int openMode = SND_PCM_ASYNC; - result = snd_pcm_open( &phandle, name, stream, openMode ); - if ( result < 0 ) { - if ( mode == OUTPUT ) - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output."; - else - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Fill the parameter structure. - snd_pcm_hw_params_t *hw_params; - snd_pcm_hw_params_alloca( &hw_params ); - result = snd_pcm_hw_params_any( phandle, hw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - -#if defined(__RTAUDIO_DEBUG__) - fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" ); - snd_pcm_hw_params_dump( hw_params, out ); -#endif - - // Set access ... check user preference. - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) { - stream_.userInterleaved = false; - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); - if ( result < 0 ) { - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); - stream_.deviceInterleaved[mode] = true; - } - else - stream_.deviceInterleaved[mode] = false; - } - else { - stream_.userInterleaved = true; - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); - if ( result < 0 ) { - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); - stream_.deviceInterleaved[mode] = false; - } - else - stream_.deviceInterleaved[mode] = true; - } - - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Determine how to set the device format. - stream_.userFormat = format; - snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN; - - if ( format == RTAUDIO_SINT8 ) - deviceFormat = SND_PCM_FORMAT_S8; - else if ( format == RTAUDIO_SINT16 ) - deviceFormat = SND_PCM_FORMAT_S16; - else if ( format == RTAUDIO_SINT24 ) - deviceFormat = SND_PCM_FORMAT_S24; - else if ( format == RTAUDIO_SINT32 ) - deviceFormat = SND_PCM_FORMAT_S32; - else if ( format == RTAUDIO_FLOAT32 ) - deviceFormat = SND_PCM_FORMAT_FLOAT; - else if ( format == RTAUDIO_FLOAT64 ) - deviceFormat = SND_PCM_FORMAT_FLOAT64; - - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) { - stream_.deviceFormat[mode] = format; - goto setFormat; - } - - // The user requested format is not natively supported by the device. - deviceFormat = SND_PCM_FORMAT_FLOAT64; - if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_FLOAT; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_S32; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_S24; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_S16; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_S8; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - goto setFormat; - } - - // If we get here, no supported format was found. - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; - - setFormat: - result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Determine whether byte-swaping is necessary. - stream_.doByteSwap[mode] = false; - if ( deviceFormat != SND_PCM_FORMAT_S8 ) { - result = snd_pcm_format_cpu_endian( deviceFormat ); - if ( result == 0 ) - stream_.doByteSwap[mode] = true; - else if (result < 0) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Set the sample rate. - result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Determine the number of channels for this device. We support a possible - // minimum device channel number > than the value requested by the user. - stream_.nUserChannels[mode] = channels; - unsigned int value; - result = snd_pcm_hw_params_get_channels_max( hw_params, &value ); - unsigned int deviceChannels = value; - if ( result < 0 || deviceChannels < channels + firstChannel ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - result = snd_pcm_hw_params_get_channels_min( hw_params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - deviceChannels = value; - if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel; - stream_.nDeviceChannels[mode] = deviceChannels; - - // Set the device channels. - result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the buffer (or period) size. - int dir = 0; - snd_pcm_uframes_t periodSize = *bufferSize; - result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - *bufferSize = periodSize; - - // Set the buffer number, which in ALSA is referred to as the "period". - unsigned int periods = 0; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; - if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers; - if ( periods < 2 ) periods = 4; // a fairly safe default value - result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // If attempting to setup a duplex stream, the bufferSize parameter - // MUST be the same in both directions! - if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - stream_.bufferSize = *bufferSize; - - // Install the hardware configuration - result = snd_pcm_hw_params( phandle, hw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - -#if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n"); - snd_pcm_hw_params_dump( hw_params, out ); -#endif - - // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns. - snd_pcm_sw_params_t *sw_params = NULL; - snd_pcm_sw_params_alloca( &sw_params ); - snd_pcm_sw_params_current( phandle, sw_params ); - snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); - snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); - snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); - - // The following two settings were suggested by Theo Veenker - //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); - //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); - - // here are two options for a fix - //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); - snd_pcm_uframes_t val; - snd_pcm_sw_params_get_boundary( sw_params, &val ); - snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); - - result = snd_pcm_sw_params( phandle, sw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - -#if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n"); - snd_pcm_sw_params_dump( sw_params, out ); -#endif - - // Set flags for buffer conversion - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate the ApiHandle if necessary and then save. - AlsaHandle *apiInfo = 0; - if ( stream_.apiHandle == 0 ) { - try { - apiInfo = (AlsaHandle *) new AlsaHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; - goto error; - } - - if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - - stream_.apiHandle = (void *) apiInfo; - apiInfo->handles[0] = 0; - apiInfo->handles[1] = 0; - } - else { - apiInfo = (AlsaHandle *) stream_.apiHandle; - } - apiInfo->handles[mode] = phandle; - phandle = 0; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.sampleRate = sampleRate; - stream_.nBuffers = periods; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - - // Setup thread if necessary. - if ( stream_.mode == OUTPUT && mode == INPUT ) { - // We had already set up an output stream. - stream_.mode = DUPLEX; - // Link the streams if possible. - apiInfo->synchronized = false; - if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 ) - apiInfo->synchronized = true; - else { - errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; - error( RtAudioError::WARNING ); - } - } - else { - stream_.mode = mode; - - // Setup callback thread. - stream_.callbackInfo.object = (void *) this; - - // Set the thread attributes for joinable and realtime scheduling - // priority (optional). The higher priority will only take affect - // if the program is run as root or suid. Note, under Linux - // processes with CAP_SYS_NICE privilege, a user can change - // scheduling policy and priority (thus need not be root). See - // POSIX "capabilities". - pthread_attr_t attr; - pthread_attr_init( &attr ); - pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); -#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { - stream_.callbackInfo.doRealtime = true; - struct sched_param param; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - param.sched_priority = priority; - - // Set the policy BEFORE the priority. Otherwise it fails. - pthread_attr_setschedpolicy(&attr, SCHED_RR); - pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM); - // This is definitely required. Otherwise it fails. - pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED); - pthread_attr_setschedparam(&attr, ¶m); - } - else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#endif - - stream_.callbackInfo.isRunning = true; - result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); - pthread_attr_destroy( &attr ); - if ( result ) { - // Failed. Try instead with default attributes. - result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo ); - if ( result ) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiAlsa::error creating callback thread!"; - goto error; - } - } - } - - return SUCCESS; - - error: - if ( apiInfo ) { - pthread_cond_destroy( &apiInfo->runnable_cv ); - if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); - if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); - delete apiInfo; - stream_.apiHandle = 0; - } - - if ( phandle) snd_pcm_close( phandle ); - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.state = STREAM_CLOSED; - return FAILURE; -} - -void RtApiAlsa :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; - error( RtAudioError::WARNING ); - return; - } - - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - stream_.callbackInfo.isRunning = false; - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) { - apiInfo->runnable = true; - pthread_cond_signal( &apiInfo->runnable_cv ); - } - MUTEX_UNLOCK( &stream_.mutex ); - pthread_join( stream_.callbackInfo.thread, NULL ); - - if ( stream_.state == STREAM_RUNNING ) { - stream_.state = STREAM_STOPPED; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - snd_pcm_drop( apiInfo->handles[0] ); - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) - snd_pcm_drop( apiInfo->handles[1] ); - } - - if ( apiInfo ) { - pthread_cond_destroy( &apiInfo->runnable_cv ); - if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); - if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); - delete apiInfo; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiAlsa :: startStream() -{ - // This method calls snd_pcm_prepare if the device isn't already in that state. - - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; - error( RtAudioError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - #if defined( HAVE_GETTIMEOFDAY ) - gettimeofday( &stream_.lastTickTimestamp, NULL ); - #endif - - int result = 0; - snd_pcm_state_t state; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - state = snd_pcm_state( handle[0] ); - if ( state != SND_PCM_STATE_PREPARED ) { - result = snd_pcm_prepare( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - } - - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open - state = snd_pcm_state( handle[1] ); - if ( state != SND_PCM_STATE_PREPARED ) { - result = snd_pcm_prepare( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - } - - stream_.state = STREAM_RUNNING; - - unlock: - apiInfo->runnable = true; - pthread_cond_signal( &apiInfo->runnable_cv ); - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result >= 0 ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiAlsa :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); - - int result = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( apiInfo->synchronized ) - result = snd_pcm_drop( handle[0] ); - else - result = snd_pcm_drain( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - unlock: - apiInfo->runnable = false; // fixes high CPU usage when stopped - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result >= 0 ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiAlsa :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); - - int result = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - result = snd_pcm_drop( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - unlock: - apiInfo->runnable = false; // fixes high CPU usage when stopped - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result >= 0 ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiAlsa :: callbackEvent() -{ - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_LOCK( &stream_.mutex ); - while ( !apiInfo->runnable ) - pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex ); - - if ( stream_.state != STREAM_RUNNING ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - MUTEX_UNLOCK( &stream_.mutex ); - } - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtAudioError::WARNING ); - return; - } - - int doStopStream = 0; - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - apiInfo->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - apiInfo->xrun[1] = false; - } - doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); - - if ( doStopStream == 2 ) { - abortStream(); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; - - int result; - char *buffer; - int channels; - snd_pcm_t **handle; - snd_pcm_sframes_t frames; - RtAudioFormat format; - handle = (snd_pcm_t **) apiInfo->handles; - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - channels = stream_.nDeviceChannels[1]; - format = stream_.deviceFormat[1]; - } - else { - buffer = stream_.userBuffer[1]; - channels = stream_.nUserChannels[1]; - format = stream_.userFormat; - } - - // Read samples from device in interleaved/non-interleaved format. - if ( stream_.deviceInterleaved[1] ) - result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); - else { - void *bufs[channels]; - size_t offset = stream_.bufferSize * formatBytes( format ); - for ( int i=0; ixrun[1] = true; - result = snd_pcm_prepare( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - error( RtAudioError::WARNING ); - goto tryOutput; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); - - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - - // Check stream latency - result = snd_pcm_delay( handle[1], &frames ); - if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; - } - - tryOutput: - - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - channels = stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - channels = stream_.nUserChannels[0]; - format = stream_.userFormat; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[0] ) - byteSwapBuffer(buffer, stream_.bufferSize * channels, format); - - // Write samples to device in interleaved/non-interleaved format. - if ( stream_.deviceInterleaved[0] ) - result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); - else { - void *bufs[channels]; - size_t offset = stream_.bufferSize * formatBytes( format ); - for ( int i=0; ixrun[0] = true; - result = snd_pcm_prepare( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - else - errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun."; - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - error( RtAudioError::WARNING ); - goto unlock; - } - - // Check stream latency - result = snd_pcm_delay( handle[0], &frames ); - if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; - } - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - RtApi::tickStreamTime(); - if ( doStopStream == 1 ) this->stopStream(); -} - -static void *alsaCallbackHandler( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiAlsa *object = (RtApiAlsa *) info->object; - bool *isRunning = &info->isRunning; - -#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) - if ( info->doRealtime ) { - std::cerr << "RtAudio alsa: " << - (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << - "running realtime scheduling" << std::endl; - } -#endif - - while ( *isRunning == true ) { - pthread_testcancel(); - object->callbackEvent(); - } - - pthread_exit( NULL ); -} - -//******************** End of __LINUX_ALSA__ *********************// -#endif - -#if defined(__LINUX_PULSE__) - -// Code written by Peter Meerwald, pmeerw@pmeerw.net -// and Tristan Matthews. - -#include -#include -#include -#include - -static pa_mainloop_api *rt_pa_mainloop_api = NULL; -struct PaDeviceInfo { - PaDeviceInfo() : sink_index(-1), source_index(-1) {} - int sink_index; - int source_index; - std::string sink_name; - std::string source_name; - RtAudio::DeviceInfo info; -}; -static struct { - std::vector dev; - std::string default_sink_name; - std::string default_source_name; - int default_rate; -} rt_pa_info; - -static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000, - 44100, 48000, 96000, 0}; - -struct rtaudio_pa_format_mapping_t { - RtAudioFormat rtaudio_format; - pa_sample_format_t pa_format; -}; - -static const rtaudio_pa_format_mapping_t supported_sampleformats[] = { - {RTAUDIO_SINT16, PA_SAMPLE_S16LE}, - {RTAUDIO_SINT24, PA_SAMPLE_S24LE}, - {RTAUDIO_SINT32, PA_SAMPLE_S32LE}, - {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE}, - {0, PA_SAMPLE_INVALID}}; - -struct PulseAudioHandle { - pa_simple *s_play; - pa_simple *s_rec; - pthread_t thread; - pthread_cond_t runnable_cv; - bool runnable; - PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { } -}; - -static void rt_pa_mainloop_api_quit(int ret) { - rt_pa_mainloop_api->quit(rt_pa_mainloop_api, ret); -} - -static void rt_pa_server_callback(pa_context *context, const pa_server_info *info, void *data){ - (void)context; - (void)data; - pa_sample_spec ss; - - if (!info) - rt_pa_mainloop_api_quit(1); - - ss = info->sample_spec; - - rt_pa_info.default_rate = ss.rate; - rt_pa_info.default_sink_name = info->default_sink_name; - rt_pa_info.default_source_name = info->default_source_name; - rt_pa_mainloop_api_quit(0); -} - -static void rt_pa_sink_info_cb(pa_context * /*c*/, const pa_sink_info *i, - int eol, void * /*userdata*/) -{ - if (eol) return; - PaDeviceInfo inf; - inf.info.name = pa_proplist_gets(i->proplist, "device.description"); - inf.info.probed = true; - inf.info.outputChannels = i->sample_spec.channels; - inf.info.preferredSampleRate = i->sample_spec.rate; - inf.info.isDefaultOutput = (rt_pa_info.default_sink_name == i->name); - inf.sink_index = i->index; - inf.sink_name = i->name; - for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) - inf.info.sampleRates.push_back( *sr ); - for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats; - fm->rtaudio_format; ++fm ) - inf.info.nativeFormats |= fm->rtaudio_format; - for (size_t i=0; i < rt_pa_info.dev.size(); i++) - { - /* Attempt to match up sink and source records by device description. */ - if (rt_pa_info.dev[i].info.name == inf.info.name) { - rt_pa_info.dev[i].sink_index = inf.sink_index; - rt_pa_info.dev[i].sink_name = inf.sink_name; - rt_pa_info.dev[i].info.outputChannels = inf.info.outputChannels; - rt_pa_info.dev[i].info.isDefaultOutput = inf.info.isDefaultOutput; - /* Assume duplex channels are minimum of input and output channels. */ - /* Uncomment if we add support for DUPLEX - if (rt_pa_info.dev[i].source_index > -1) - (inf.info.outputChannels < rt_pa_info.dev[i].info.inputChannels) - ? inf.info.outputChannels : rt_pa_info.dev[i].info.inputChannels; - */ - return; - } - } - /* try to ensure device #0 is the default */ - if (inf.info.isDefaultOutput) - rt_pa_info.dev.insert(rt_pa_info.dev.begin(), inf); - else - rt_pa_info.dev.push_back(inf); -} - -static void rt_pa_source_info_cb(pa_context * /*c*/, const pa_source_info *i, - int eol, void * /*userdata*/) -{ - if (eol) return; - PaDeviceInfo inf; - inf.info.name = pa_proplist_gets(i->proplist, "device.description"); - inf.info.probed = true; - inf.info.inputChannels = i->sample_spec.channels; - inf.info.preferredSampleRate = i->sample_spec.rate; - inf.info.isDefaultInput = (rt_pa_info.default_source_name == i->name); - inf.source_index = i->index; - inf.source_name = i->name; - for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) - inf.info.sampleRates.push_back( *sr ); - for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats; - fm->rtaudio_format; ++fm ) - inf.info.nativeFormats |= fm->rtaudio_format; - - for (size_t i=0; i < rt_pa_info.dev.size(); i++) - { - /* Attempt to match up sink and source records by device description. */ - if (rt_pa_info.dev[i].info.name == inf.info.name) { - rt_pa_info.dev[i].source_index = inf.source_index; - rt_pa_info.dev[i].source_name = inf.source_name; - rt_pa_info.dev[i].info.inputChannels = inf.info.inputChannels; - rt_pa_info.dev[i].info.isDefaultInput = inf.info.isDefaultInput; - /* Assume duplex channels are minimum of input and output channels. */ - /* Uncomment if we add support for DUPLEX - if (rt_pa_info.dev[i].sink_index > -1) { - rt_pa_info.dev[i].info.duplexChannels = - (inf.info.inputChannels < rt_pa_info.dev[i].info.outputChannels) - ? inf.info.inputChannels : rt_pa_info.dev[i].info.outputChannels; - } - */ - return; - } - } - /* try to ensure device #0 is the default */ - if (inf.info.isDefaultInput) - rt_pa_info.dev.insert(rt_pa_info.dev.begin(), inf); - else - rt_pa_info.dev.push_back(inf); -} - -static void rt_pa_context_state_callback(pa_context *context, void *userdata) { - (void)userdata; - - switch (pa_context_get_state(context)) { - case PA_CONTEXT_CONNECTING: - case PA_CONTEXT_AUTHORIZING: - case PA_CONTEXT_SETTING_NAME: - break; - - case PA_CONTEXT_READY: - rt_pa_info.dev.clear(); - pa_context_get_server_info(context, rt_pa_server_callback, NULL); - pa_context_get_sink_info_list(context, rt_pa_sink_info_cb, NULL); - pa_context_get_source_info_list(context, rt_pa_source_info_cb, NULL); - break; - - case PA_CONTEXT_TERMINATED: - rt_pa_mainloop_api_quit(0); - break; - - case PA_CONTEXT_FAILED: - default: - rt_pa_mainloop_api_quit(1); - } -} - -RtApiPulse::~RtApiPulse() -{ - if ( stream_.state != STREAM_CLOSED ) - closeStream(); -} - -void RtApiPulse::collectDeviceInfo( void ) -{ - pa_context *context = NULL; - pa_mainloop *m = NULL; - char *server = NULL; - int ret = 1; - - if (!(m = pa_mainloop_new())) { - errorStream_ << "RtApiPulse::DeviceInfo pa_mainloop_new() failed."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto quit; - } - - rt_pa_mainloop_api = pa_mainloop_get_api(m); - - if (!(context = pa_context_new_with_proplist(rt_pa_mainloop_api, NULL, NULL))) { - errorStream_ << "pa_context_new() failed."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto quit; - } - - pa_context_set_state_callback(context, rt_pa_context_state_callback, NULL); - - if (pa_context_connect(context, server, PA_CONTEXT_NOFLAGS, NULL) < 0) { - errorStream_ << "RtApiPulse::DeviceInfo pa_context_connect() failed: " - << pa_strerror(pa_context_errno(context)); - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto quit; - } - - if (pa_mainloop_run(m, &ret) < 0) { - errorStream_ << "pa_mainloop_run() failed."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto quit; - } - -quit: - if (context) - pa_context_unref(context); - - if (m) { - pa_mainloop_free(m); - } - - pa_xfree(server); -} - -unsigned int RtApiPulse::getDeviceCount( void ) -{ - collectDeviceInfo(); - return rt_pa_info.dev.size(); -} - -RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int device ) -{ - if (rt_pa_info.dev.size()==0) - collectDeviceInfo(); - if (device < rt_pa_info.dev.size()) - return rt_pa_info.dev[device].info; - return RtAudio::DeviceInfo(); -} - -static void *pulseaudio_callback( void * user ) -{ - CallbackInfo *cbi = static_cast( user ); - RtApiPulse *context = static_cast( cbi->object ); - volatile bool *isRunning = &cbi->isRunning; - -#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) - if (cbi->doRealtime) { - std::cerr << "RtAudio pulse: " << - (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << - "running realtime scheduling" << std::endl; - } -#endif - - while ( *isRunning ) { - pthread_testcancel(); - context->callbackEvent(); - } - - pthread_exit( NULL ); -} - -void RtApiPulse::closeStream( void ) -{ - PulseAudioHandle *pah = static_cast( stream_.apiHandle ); - - stream_.callbackInfo.isRunning = false; - if ( pah ) { - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) { - pah->runnable = true; - pthread_cond_signal( &pah->runnable_cv ); - } - MUTEX_UNLOCK( &stream_.mutex ); - - pthread_join( pah->thread, 0 ); - if ( pah->s_play ) { - pa_simple_flush( pah->s_play, NULL ); - pa_simple_free( pah->s_play ); - } - if ( pah->s_rec ) - pa_simple_free( pah->s_rec ); - - pthread_cond_destroy( &pah->runnable_cv ); - delete pah; - stream_.apiHandle = 0; - } - - if ( stream_.userBuffer[0] ) { - free( stream_.userBuffer[0] ); - stream_.userBuffer[0] = 0; - } - if ( stream_.userBuffer[1] ) { - free( stream_.userBuffer[1] ); - stream_.userBuffer[1] = 0; - } - - stream_.state = STREAM_CLOSED; - stream_.mode = UNINITIALIZED; -} - -void RtApiPulse::callbackEvent( void ) -{ - PulseAudioHandle *pah = static_cast( stream_.apiHandle ); - - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_LOCK( &stream_.mutex ); - while ( !pah->runnable ) - pthread_cond_wait( &pah->runnable_cv, &stream_.mutex ); - - if ( stream_.state != STREAM_RUNNING ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - MUTEX_UNLOCK( &stream_.mutex ); - } - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... " - "this shouldn't happen!"; - error( RtAudioError::WARNING ); - return; - } - - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT], - stream_.bufferSize, streamTime, status, - stream_.callbackInfo.userData ); - - if ( doStopStream == 2 ) { - abortStream(); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT]; - void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT]; - - if ( stream_.state != STREAM_RUNNING ) - goto unlock; - - int pa_error; - size_t bytes; - if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( stream_.doConvertBuffer[OUTPUT] ) { - convertBuffer( stream_.deviceBuffer, - stream_.userBuffer[OUTPUT], - stream_.convertInfo[OUTPUT] ); - bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize * - formatBytes( stream_.deviceFormat[OUTPUT] ); - } else - bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize * - formatBytes( stream_.userFormat ); - - if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) { - errorStream_ << "RtApiPulse::callbackEvent: audio write error, " << - pa_strerror( pa_error ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - } - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX) { - if ( stream_.doConvertBuffer[INPUT] ) - bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize * - formatBytes( stream_.deviceFormat[INPUT] ); - else - bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize * - formatBytes( stream_.userFormat ); - - if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) { - errorStream_ << "RtApiPulse::callbackEvent: audio read error, " << - pa_strerror( pa_error ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - } - if ( stream_.doConvertBuffer[INPUT] ) { - convertBuffer( stream_.userBuffer[INPUT], - stream_.deviceBuffer, - stream_.convertInfo[INPUT] ); - } - } - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - RtApi::tickStreamTime(); - - if ( doStopStream == 1 ) - stopStream(); -} - -void RtApiPulse::startStream( void ) -{ - PulseAudioHandle *pah = static_cast( stream_.apiHandle ); - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiPulse::startStream(): the stream is not open!"; - error( RtAudioError::INVALID_USE ); - return; - } - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiPulse::startStream(): the stream is already running!"; - error( RtAudioError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - #if defined( HAVE_GETTIMEOFDAY ) - gettimeofday( &stream_.lastTickTimestamp, NULL ); - #endif - - stream_.state = STREAM_RUNNING; - - pah->runnable = true; - pthread_cond_signal( &pah->runnable_cv ); - MUTEX_UNLOCK( &stream_.mutex ); -} - -void RtApiPulse::stopStream( void ) -{ - PulseAudioHandle *pah = static_cast( stream_.apiHandle ); - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiPulse::stopStream(): the stream is not open!"; - error( RtAudioError::INVALID_USE ); - return; - } - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); - - if ( pah ) { - pah->runnable = false; - if ( pah->s_play ) { - int pa_error; - if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) { - errorStream_ << "RtApiPulse::stopStream: error draining output device, " << - pa_strerror( pa_error ) << "."; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - } - } - - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); -} - -void RtApiPulse::abortStream( void ) -{ - PulseAudioHandle *pah = static_cast( stream_.apiHandle ); - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiPulse::abortStream(): the stream is not open!"; - error( RtAudioError::INVALID_USE ); - return; - } - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); - - if ( pah ) { - pah->runnable = false; - if ( pah->s_play ) { - int pa_error; - if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) { - errorStream_ << "RtApiPulse::abortStream: error flushing output device, " << - pa_strerror( pa_error ) << "."; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - } - } - - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); -} - -bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, - unsigned int channels, unsigned int firstChannel, - unsigned int sampleRate, RtAudioFormat format, - unsigned int *bufferSize, RtAudio::StreamOptions *options ) -{ - PulseAudioHandle *pah = 0; - unsigned long bufferBytes = 0; - pa_sample_spec ss; - - if ( device >= rt_pa_info.dev.size() ) return false; - if ( firstChannel != 0 ) { - errorText_ = "PulseAudio does not support channel offset mapping."; - return false; - } - - /* these may be NULL for default, but we've already got the names */ - const char *dev_input = NULL; - const char *dev_output = NULL; - if (!rt_pa_info.dev[device].source_name.empty()) - dev_input = rt_pa_info.dev[device].source_name.c_str(); - if (!rt_pa_info.dev[device].sink_name.empty()) - dev_output = rt_pa_info.dev[device].sink_name.c_str(); - - if (mode==INPUT && rt_pa_info.dev[device].info.inputChannels == 0) { - errorText_ = "PulseAudio device does not support input."; - return false; - } - if (mode==OUTPUT && rt_pa_info.dev[device].info.outputChannels == 0) { - errorText_ = "PulseAudio device does not support output."; - return false; - } - if (mode==DUPLEX && rt_pa_info.dev[device].info.duplexChannels == 0) { - /* Note: will always error, DUPLEX not yet supported */ - errorText_ = "PulseAudio device does not support duplex."; - return false; - } - - if (mode==INPUT && rt_pa_info.dev[device].info.inputChannels < channels) { - errorText_ = "PulseAudio: unsupported number of input channels."; - return false; - } - - if (mode==OUTPUT && rt_pa_info.dev[device].info.outputChannels < channels) { - errorText_ = "PulseAudio: unsupported number of output channels."; - return false; - } - - if (mode==DUPLEX && rt_pa_info.dev[device].info.duplexChannels < channels) { - /* Note: will always error, DUPLEX not yet supported */ - errorText_ = "PulseAudio: unsupported number of duplex channels."; - return false; - } - - ss.channels = channels; - - bool sr_found = false; - for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) { - if ( sampleRate == *sr ) { - sr_found = true; - stream_.sampleRate = sampleRate; - ss.rate = sampleRate; - break; - } - } - if ( !sr_found ) { - errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate."; - return false; - } - - bool sf_found = 0; - for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats; - sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) { - if ( format == sf->rtaudio_format ) { - sf_found = true; - stream_.userFormat = sf->rtaudio_format; - stream_.deviceFormat[mode] = stream_.userFormat; - ss.format = sf->pa_format; - break; - } - } - if ( !sf_found ) { // Use internal data format conversion. - stream_.userFormat = format; - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - ss.format = PA_SAMPLE_FLOAT32LE; - } - - // Set other stream parameters. - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - stream_.deviceInterleaved[mode] = true; - stream_.nBuffers = options ? options->numberOfBuffers : 1; - stream_.doByteSwap[mode] = false; - stream_.nUserChannels[mode] = channels; - stream_.nDeviceChannels[mode] = channels + firstChannel; - stream_.channelOffset[mode] = 0; - std::string streamName = "RtAudio"; - - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] ) - stream_.doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers. - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - stream_.bufferSize = *bufferSize; - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.device[mode] = device; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - - if ( !stream_.apiHandle ) { - PulseAudioHandle *pah = new PulseAudioHandle; - if ( !pah ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle."; - goto error; - } - - stream_.apiHandle = pah; - if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable."; - goto error; - } - } - pah = static_cast( stream_.apiHandle ); - - int error; - if ( options && !options->streamName.empty() ) streamName = options->streamName; - switch ( mode ) { - pa_buffer_attr buffer_attr; - case INPUT: - buffer_attr.fragsize = bufferBytes; - buffer_attr.maxlength = -1; - - pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, - dev_input, "Record", &ss, NULL, &buffer_attr, &error ); - if ( !pah->s_rec ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server."; - goto error; - } - break; - case OUTPUT: { - pa_buffer_attr * attr_ptr; - - if ( options && options->numberOfBuffers > 0 ) { - // pa_buffer_attr::fragsize is recording-only. - // Hopefully PortAudio won't access uninitialized fields. - buffer_attr.maxlength = bufferBytes * options->numberOfBuffers; - buffer_attr.minreq = -1; - buffer_attr.prebuf = -1; - buffer_attr.tlength = -1; - attr_ptr = &buffer_attr; - } else { - attr_ptr = nullptr; - } - - pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, - dev_output, "Playback", &ss, NULL, attr_ptr, &error ); - if ( !pah->s_play ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server."; - goto error; - } - break; - } - case DUPLEX: - /* Note: We could add DUPLEX by synchronizing multiple streams, - but it would mean moving from Simple API to Asynchronous API: - https://freedesktop.org/software/pulseaudio/doxygen/streams.html#sync_streams */ - errorText_ = "RtApiPulse::probeDeviceOpen: duplex not supported for PulseAudio."; - goto error; - default: - goto error; - } - - if ( stream_.mode == UNINITIALIZED ) - stream_.mode = mode; - else if ( stream_.mode == mode ) - goto error; - else - stream_.mode = DUPLEX; - - if ( !stream_.callbackInfo.isRunning ) { - stream_.callbackInfo.object = this; - - stream_.state = STREAM_STOPPED; - // Set the thread attributes for joinable and realtime scheduling - // priority (optional). The higher priority will only take affect - // if the program is run as root or suid. Note, under Linux - // processes with CAP_SYS_NICE privilege, a user can change - // scheduling policy and priority (thus need not be root). See - // POSIX "capabilities". - pthread_attr_t attr; - pthread_attr_init( &attr ); - pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); -#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { - stream_.callbackInfo.doRealtime = true; - struct sched_param param; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - param.sched_priority = priority; - - // Set the policy BEFORE the priority. Otherwise it fails. - pthread_attr_setschedpolicy(&attr, SCHED_RR); - pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM); - // This is definitely required. Otherwise it fails. - pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED); - pthread_attr_setschedparam(&attr, ¶m); - } - else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#endif - - stream_.callbackInfo.isRunning = true; - int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo); - pthread_attr_destroy(&attr); - if(result != 0) { - // Failed. Try instead with default attributes. - result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo); - if(result != 0) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread."; - goto error; - } - } - } - - return SUCCESS; - - error: - if ( pah && stream_.callbackInfo.isRunning ) { - pthread_cond_destroy( &pah->runnable_cv ); - delete pah; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.state = STREAM_CLOSED; - return FAILURE; -} - -//******************** End of __LINUX_PULSE__ *********************// -#endif - -#if defined(__LINUX_OSS__) - -#include -#include -#include -#include -#include -#include -#include - -static void *ossCallbackHandler(void * ptr); - -// A structure to hold various information related to the OSS API -// implementation. -struct OssHandle { - int id[2]; // device ids - bool xrun[2]; - bool triggered; - pthread_cond_t runnable; - - OssHandle() - :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } -}; - -RtApiOss :: RtApiOss() -{ - // Nothing to do here. -} - -RtApiOss :: ~RtApiOss() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiOss :: getDeviceCount( void ) -{ - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; - error( RtAudioError::WARNING ); - return 0; - } - - oss_sysinfo sysinfo; - if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; - error( RtAudioError::WARNING ); - return 0; - } - - close( mixerfd ); - return sysinfo.numaudios; -} - -RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; - error( RtAudioError::WARNING ); - return info; - } - - oss_sysinfo sysinfo; - int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); - if ( result == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; - error( RtAudioError::WARNING ); - return info; - } - - unsigned nDevices = sysinfo.numaudios; - if ( nDevices == 0 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - if ( device >= nDevices ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - oss_audioinfo ainfo; - ainfo.dev = device; - result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); - close( mixerfd ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Probe channels - if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels; - if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels; - if ( ainfo.caps & PCM_CAP_DUPLEX ) { - if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - } - - // Probe data formats ... do for input - unsigned long mask = ainfo.iformats; - if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE ) - info.nativeFormats |= RTAUDIO_SINT16; - if ( mask & AFMT_S8 ) - info.nativeFormats |= RTAUDIO_SINT8; - if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE ) - info.nativeFormats |= RTAUDIO_SINT32; -#ifdef AFMT_FLOAT - if ( mask & AFMT_FLOAT ) - info.nativeFormats |= RTAUDIO_FLOAT32; -#endif - if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE ) - info.nativeFormats |= RTAUDIO_SINT24; - - // Check that we have at least one supported format - if ( info.nativeFormats == 0 ) { - errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Probe the supported sample rates. - info.sampleRates.clear(); - if ( ainfo.nrates ) { - for ( unsigned int i=0; i info.preferredSampleRate ) ) - info.preferredSampleRate = SAMPLE_RATES[k]; - - break; - } - } - } - } - else { - // Check min and max rate values; - for ( unsigned int k=0; k= (int) SAMPLE_RATES[k] ) { - info.sampleRates.push_back( SAMPLE_RATES[k] ); - - if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) - info.preferredSampleRate = SAMPLE_RATES[k]; - } - } - } - - if ( info.sampleRates.size() == 0 ) { - errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - } - else { - info.probed = true; - info.name = ainfo.name; - } - - return info; -} - - -bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; - return FAILURE; - } - - oss_sysinfo sysinfo; - int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); - if ( result == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; - return FAILURE; - } - - unsigned nDevices = sysinfo.numaudios; - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; - return FAILURE; - } - - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - oss_audioinfo ainfo; - ainfo.dev = device; - result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); - close( mixerfd ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check if device supports input or output - if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || - ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { - if ( mode == OUTPUT ) - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; - else - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - int flags = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( mode == OUTPUT ) - flags |= O_WRONLY; - else { // mode == INPUT - if (stream_.mode == OUTPUT && stream_.device[0] == device) { - // We just set the same device for playback ... close and reopen for duplex (OSS only). - close( handle->id[0] ); - handle->id[0] = 0; - if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; - errorText_ = errorStream_.str(); - return FAILURE; - } - // Check that the number previously set channels is the same. - if ( stream_.nUserChannels[0] != channels ) { - errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - flags |= O_RDWR; - } - else - flags |= O_RDONLY; - } - - // Set exclusive access if specified. - if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; - - // Try to open the device. - int fd; - fd = open( ainfo.devnode, flags, 0 ); - if ( fd == -1 ) { - if ( errno == EBUSY ) - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; - else - errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // For duplex operation, specifically set this mode (this doesn't seem to work). - /* - if ( flags | O_RDWR ) { - result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); - if ( result == -1) { - errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - */ - - // Check the device channel support. - stream_.nUserChannels[mode] = channels; - if ( ainfo.max_channels < (int)(channels + firstChannel) ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the number of channels. - int deviceChannels = channels + firstChannel; - result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); - if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.nDeviceChannels[mode] = deviceChannels; - - // Get the data format mask - int mask; - result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Determine how to set the device format. - stream_.userFormat = format; - int deviceFormat = -1; - stream_.doByteSwap[mode] = false; - if ( format == RTAUDIO_SINT8 ) { - if ( mask & AFMT_S8 ) { - deviceFormat = AFMT_S8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - } - else if ( format == RTAUDIO_SINT16 ) { - if ( mask & AFMT_S16_NE ) { - deviceFormat = AFMT_S16_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else if ( mask & AFMT_S16_OE ) { - deviceFormat = AFMT_S16_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - stream_.doByteSwap[mode] = true; - } - } - else if ( format == RTAUDIO_SINT24 ) { - if ( mask & AFMT_S24_NE ) { - deviceFormat = AFMT_S24_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - } - else if ( mask & AFMT_S24_OE ) { - deviceFormat = AFMT_S24_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - stream_.doByteSwap[mode] = true; - } - } - else if ( format == RTAUDIO_SINT32 ) { - if ( mask & AFMT_S32_NE ) { - deviceFormat = AFMT_S32_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - } - else if ( mask & AFMT_S32_OE ) { - deviceFormat = AFMT_S32_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - stream_.doByteSwap[mode] = true; - } - } - - if ( deviceFormat == -1 ) { - // The user requested format is not natively supported by the device. - if ( mask & AFMT_S16_NE ) { - deviceFormat = AFMT_S16_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else if ( mask & AFMT_S32_NE ) { - deviceFormat = AFMT_S32_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - } - else if ( mask & AFMT_S24_NE ) { - deviceFormat = AFMT_S24_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - } - else if ( mask & AFMT_S16_OE ) { - deviceFormat = AFMT_S16_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S32_OE ) { - deviceFormat = AFMT_S32_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S24_OE ) { - deviceFormat = AFMT_S24_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S8) { - deviceFormat = AFMT_S8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - } - - if ( stream_.deviceFormat[mode] == 0 ) { - // This really shouldn't happen ... - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the data format. - int temp = deviceFormat; - result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); - if ( result == -1 || deviceFormat != temp ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Attempt to set the buffer size. According to OSS, the minimum - // number of buffers is two. The supposed minimum buffer size is 16 - // bytes, so that will be our lower bound. The argument to this - // call is in the form 0xMMMMSSSS (hex), where the buffer size (in - // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. - // We'll check the actual value used near the end of the setup - // procedure. - int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; - if ( ossBufferBytes < 16 ) ossBufferBytes = 16; - int buffers = 0; - if ( options ) buffers = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; - if ( buffers < 2 ) buffers = 3; - temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); - result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.nBuffers = buffers; - - // Save buffer size (in sample frames). - *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); - stream_.bufferSize = *bufferSize; - - // Set the sample rate. - int srate = sampleRate; - result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Verify the sample rate setup worked. - if ( abs( srate - (int)sampleRate ) > 100 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.sampleRate = sampleRate; - - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { - // We're doing duplex setup here. - stream_.deviceFormat[0] = stream_.deviceFormat[1]; - stream_.nDeviceChannels[0] = deviceChannels; - } - - // Set interleaving parameters. - stream_.userInterleaved = true; - stream_.deviceInterleaved[mode] = true; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) - stream_.userInterleaved = false; - - // Set flags for buffer conversion - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate the stream handles if necessary and then save. - if ( stream_.apiHandle == 0 ) { - try { - handle = new OssHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; - goto error; - } - - if ( pthread_cond_init( &handle->runnable, NULL ) ) { - errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - - stream_.apiHandle = (void *) handle; - } - else { - handle = (OssHandle *) stream_.apiHandle; - } - handle->id[mode] = fd; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - - // Setup thread if necessary. - if ( stream_.mode == OUTPUT && mode == INPUT ) { - // We had already set up an output stream. - stream_.mode = DUPLEX; - if ( stream_.device[0] == device ) handle->id[0] = fd; - } - else { - stream_.mode = mode; - - // Setup callback thread. - stream_.callbackInfo.object = (void *) this; - - // Set the thread attributes for joinable and realtime scheduling - // priority. The higher priority will only take affect if the - // program is run as root or suid. - pthread_attr_t attr; - pthread_attr_init( &attr ); - pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); -#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { - stream_.callbackInfo.doRealtime = true; - struct sched_param param; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - param.sched_priority = priority; - - // Set the policy BEFORE the priority. Otherwise it fails. - pthread_attr_setschedpolicy(&attr, SCHED_RR); - pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM); - // This is definitely required. Otherwise it fails. - pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED); - pthread_attr_setschedparam(&attr, ¶m); - } - else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#endif - - stream_.callbackInfo.isRunning = true; - result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo ); - pthread_attr_destroy( &attr ); - if ( result ) { - // Failed. Try instead with default attributes. - result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo ); - if ( result ) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiOss::error creating callback thread!"; - goto error; - } - } - } - - return SUCCESS; - - error: - if ( handle ) { - pthread_cond_destroy( &handle->runnable ); - if ( handle->id[0] ) close( handle->id[0] ); - if ( handle->id[1] ) close( handle->id[1] ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.state = STREAM_CLOSED; - return FAILURE; -} - -void RtApiOss :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiOss::closeStream(): no open stream to close!"; - error( RtAudioError::WARNING ); - return; - } - - OssHandle *handle = (OssHandle *) stream_.apiHandle; - stream_.callbackInfo.isRunning = false; - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) - pthread_cond_signal( &handle->runnable ); - MUTEX_UNLOCK( &stream_.mutex ); - pthread_join( stream_.callbackInfo.thread, NULL ); - - if ( stream_.state == STREAM_RUNNING ) { - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - else - ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - stream_.state = STREAM_STOPPED; - } - - if ( handle ) { - pthread_cond_destroy( &handle->runnable ); - if ( handle->id[0] ) close( handle->id[0] ); - if ( handle->id[1] ) close( handle->id[1] ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiOss :: startStream() -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiOss::startStream(): the stream is already running!"; - error( RtAudioError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - #if defined( HAVE_GETTIMEOFDAY ) - gettimeofday( &stream_.lastTickTimestamp, NULL ); - #endif - - stream_.state = STREAM_RUNNING; - - // No need to do anything else here ... OSS automatically starts - // when fed samples. - - MUTEX_UNLOCK( &stream_.mutex ); - - OssHandle *handle = (OssHandle *) stream_.apiHandle; - pthread_cond_signal( &handle->runnable ); -} - -void RtApiOss :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - int result = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - // Flush the output with zeros a few times. - char *buffer; - int samples; - RtAudioFormat format; - - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - samples = stream_.bufferSize * stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - samples = stream_.bufferSize * stream_.nUserChannels[0]; - format = stream_.userFormat; - } - - memset( buffer, 0, samples * formatBytes(format) ); - for ( unsigned int i=0; iid[0], buffer, samples * formatBytes(format) ); - if ( result == -1 ) { - errorText_ = "RtApiOss::stopStream: audio write error."; - error( RtAudioError::WARNING ); - } - } - - result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - handle->triggered = false; - } - - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { - result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - unlock: - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result != -1 ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiOss :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - int result = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - handle->triggered = false; - } - - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { - result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - unlock: - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result != -1 ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiOss :: callbackEvent() -{ - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_LOCK( &stream_.mutex ); - pthread_cond_wait( &handle->runnable, &stream_.mutex ); - if ( stream_.state != STREAM_RUNNING ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - MUTEX_UNLOCK( &stream_.mutex ); - } - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtAudioError::WARNING ); - return; - } - - // Invoke user callback to get fresh output data. - int doStopStream = 0; - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); - if ( doStopStream == 2 ) { - this->abortStream(); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; - - int result; - char *buffer; - int samples; - RtAudioFormat format; - - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - samples = stream_.bufferSize * stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - samples = stream_.bufferSize * stream_.nUserChannels[0]; - format = stream_.userFormat; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( buffer, samples, format ); - - if ( stream_.mode == DUPLEX && handle->triggered == false ) { - int trig = 0; - ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); - result = write( handle->id[0], buffer, samples * formatBytes(format) ); - trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT; - ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); - handle->triggered = true; - } - else - // Write samples to device. - result = write( handle->id[0], buffer, samples * formatBytes(format) ); - - if ( result == -1 ) { - // We'll assume this is an underrun, though there isn't a - // specific means for determining that. - handle->xrun[0] = true; - errorText_ = "RtApiOss::callbackEvent: audio write error."; - error( RtAudioError::WARNING ); - // Continue on to input section. - } - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - samples = stream_.bufferSize * stream_.nDeviceChannels[1]; - format = stream_.deviceFormat[1]; - } - else { - buffer = stream_.userBuffer[1]; - samples = stream_.bufferSize * stream_.nUserChannels[1]; - format = stream_.userFormat; - } - - // Read samples from device. - result = read( handle->id[1], buffer, samples * formatBytes(format) ); - - if ( result == -1 ) { - // We'll assume this is an overrun, though there isn't a - // specific means for determining that. - handle->xrun[1] = true; - errorText_ = "RtApiOss::callbackEvent: audio read error."; - error( RtAudioError::WARNING ); - goto unlock; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( buffer, samples, format ); - - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - } - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - RtApi::tickStreamTime(); - if ( doStopStream == 1 ) this->stopStream(); -} - -static void *ossCallbackHandler( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiOss *object = (RtApiOss *) info->object; - bool *isRunning = &info->isRunning; - -#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) - if (info->doRealtime) { - std::cerr << "RtAudio oss: " << - (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << - "running realtime scheduling" << std::endl; - } -#endif - - while ( *isRunning == true ) { - pthread_testcancel(); - object->callbackEvent(); - } - - pthread_exit( NULL ); -} - -//******************** End of __LINUX_OSS__ *********************// -#endif - - -// *************************************************** // -// -// Protected common (OS-independent) RtAudio methods. -// -// *************************************************** // - -// This method can be modified to control the behavior of error -// message printing. -void RtApi :: error( RtAudioError::Type type ) -{ - errorStream_.str(""); // clear the ostringstream - - RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback; - if ( errorCallback ) { - // abortStream() can generate new error messages. Ignore them. Just keep original one. - - if ( firstErrorOccurred_ ) - return; - - firstErrorOccurred_ = true; - const std::string errorMessage = errorText_; - - if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) { - stream_.callbackInfo.isRunning = false; // exit from the thread - abortStream(); - } - - errorCallback( type, errorMessage ); - firstErrorOccurred_ = false; - return; - } - - if ( type == RtAudioError::WARNING && showWarnings_ == true ) - std::cerr << '\n' << errorText_ << "\n\n"; - else if ( type != RtAudioError::WARNING ) - throw( RtAudioError( errorText_, type ) ); -} - -void RtApi :: verifyStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApi:: a stream is not open!"; - error( RtAudioError::INVALID_USE ); - } -} - -void RtApi :: clearStreamInfo() -{ - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; - stream_.sampleRate = 0; - stream_.bufferSize = 0; - stream_.nBuffers = 0; - stream_.userFormat = 0; - stream_.userInterleaved = true; - stream_.streamTime = 0.0; - stream_.apiHandle = 0; - stream_.deviceBuffer = 0; - stream_.callbackInfo.callback = 0; - stream_.callbackInfo.userData = 0; - stream_.callbackInfo.isRunning = false; - stream_.callbackInfo.errorCallback = 0; - for ( int i=0; i<2; i++ ) { - stream_.device[i] = 11111; - stream_.doConvertBuffer[i] = false; - stream_.deviceInterleaved[i] = true; - stream_.doByteSwap[i] = false; - stream_.nUserChannels[i] = 0; - stream_.nDeviceChannels[i] = 0; - stream_.channelOffset[i] = 0; - stream_.deviceFormat[i] = 0; - stream_.latency[i] = 0; - stream_.userBuffer[i] = 0; - stream_.convertInfo[i].channels = 0; - stream_.convertInfo[i].inJump = 0; - stream_.convertInfo[i].outJump = 0; - stream_.convertInfo[i].inFormat = 0; - stream_.convertInfo[i].outFormat = 0; - stream_.convertInfo[i].inOffset.clear(); - stream_.convertInfo[i].outOffset.clear(); - } -} - -unsigned int RtApi :: formatBytes( RtAudioFormat format ) -{ - if ( format == RTAUDIO_SINT16 ) - return 2; - else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 ) - return 4; - else if ( format == RTAUDIO_FLOAT64 ) - return 8; - else if ( format == RTAUDIO_SINT24 ) - return 3; - else if ( format == RTAUDIO_SINT8 ) - return 1; - - errorText_ = "RtApi::formatBytes: undefined format."; - error( RtAudioError::WARNING ); - - return 0; -} - -void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel ) -{ - if ( mode == INPUT ) { // convert device to user buffer - stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1]; - stream_.convertInfo[mode].outJump = stream_.nUserChannels[1]; - stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1]; - stream_.convertInfo[mode].outFormat = stream_.userFormat; - } - else { // convert user to device buffer - stream_.convertInfo[mode].inJump = stream_.nUserChannels[0]; - stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0]; - stream_.convertInfo[mode].inFormat = stream_.userFormat; - stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0]; - } - - if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump ) - stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump; - else - stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump; - - // Set up the interleave/deinterleave offsets. - if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) { - if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) || - ( mode == INPUT && stream_.userInterleaved ) ) { - for ( int k=0; k 0 ) { - if ( stream_.deviceInterleaved[mode] ) { - if ( mode == OUTPUT ) { - for ( int k=0; k info.inJump ) - memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) ); - - int j; - if (info.outFormat == RTAUDIO_FLOAT64) { - Float64 *out = (Float64 *)outBuffer; - - if (info.inFormat == RTAUDIO_SINT8) { - signed char *in = (signed char *)inBuffer; - for (unsigned int i=0; i> 8); - //out[info.outOffset[j]] >>= 8; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i> 8); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i> 16) & 0x0000ffff); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i> 8) & 0x00ff); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT24) { - Int24 *in = (Int24 *)inBuffer; - for (unsigned int i=0; i> 16); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i> 24) & 0x000000ff); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i>8) | (x<<8); } -//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } -//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } - -void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) -{ - char val; - char *ptr; - - ptr = buffer; - if ( format == RTAUDIO_SINT16 ) { - for ( unsigned int i=0; i= 4 - #define RTAUDIO_DLL_PUBLIC __attribute__( (visibility( "default" )) ) - #else - #define RTAUDIO_DLL_PUBLIC - #endif -#endif - -#include -#include -#include -#include - -/*! \typedef typedef unsigned long RtAudioFormat; - \brief RtAudio data format type. - - Support for signed integers and floats. Audio data fed to/from an - RtAudio stream is assumed to ALWAYS be in host byte order. The - internal routines will automatically take care of any necessary - byte-swapping between the host format and the soundcard. Thus, - endian-ness is not a concern in the following format definitions. - - - \e RTAUDIO_SINT8: 8-bit signed integer. - - \e RTAUDIO_SINT16: 16-bit signed integer. - - \e RTAUDIO_SINT24: 24-bit signed integer. - - \e RTAUDIO_SINT32: 32-bit signed integer. - - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0. - - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0. -*/ -typedef unsigned long RtAudioFormat; -static const RtAudioFormat RTAUDIO_SINT8 = 0x1; // 8-bit signed integer. -static const RtAudioFormat RTAUDIO_SINT16 = 0x2; // 16-bit signed integer. -static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // 24-bit signed integer. -static const RtAudioFormat RTAUDIO_SINT32 = 0x8; // 32-bit signed integer. -static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0. -static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0. - -/*! \typedef typedef unsigned long RtAudioStreamFlags; - \brief RtAudio stream option flags. - - The following flags can be OR'ed together to allow a client to - make changes to the default stream behavior: - - - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). - - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. - - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. - - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only). - - \e RTAUDIO_JACK_DONT_CONNECT: Do not automatically connect ports (JACK only). - - By default, RtAudio streams pass and receive audio data from the - client in an interleaved format. By passing the - RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio - data will instead be presented in non-interleaved buffers. In - this case, each buffer argument in the RtAudioCallback function - will point to a single array of data, with \c nFrames samples for - each channel concatenated back-to-back. For example, the first - sample of data for the second channel would be located at index \c - nFrames (assuming the \c buffer pointer was recast to the correct - data type for the stream). - - Certain audio APIs offer a number of parameters that influence the - I/O latency of a stream. By default, RtAudio will attempt to set - these parameters internally for robust (glitch-free) performance - (though some APIs, like Windows DirectSound, make this difficult). - By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() - function, internal stream settings will be influenced in an attempt - to minimize stream latency, though possibly at the expense of stream - performance. - - If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to - open the input and/or output stream device(s) for exclusive use. - Note that this is not possible with all supported audio APIs. - - If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt - to select realtime scheduling (round-robin) for the callback thread. - - If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to - open the "default" PCM device when using the ALSA API. Note that this - will override any specified input or output device id. - - If the RTAUDIO_JACK_DONT_CONNECT flag is set, RtAudio will not attempt - to automatically connect the ports of the client to the audio device. -*/ -typedef unsigned int RtAudioStreamFlags; -static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved). -static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency. -static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others. -static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread. -static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only). -static const RtAudioStreamFlags RTAUDIO_JACK_DONT_CONNECT = 0x20; // Do not automatically connect ports (JACK only). - -/*! \typedef typedef unsigned long RtAudioStreamStatus; - \brief RtAudio stream status (over- or underflow) flags. - - Notification of a stream over- or underflow is indicated by a - non-zero stream \c status argument in the RtAudioCallback function. - The stream status can be one of the following two options, - depending on whether the stream is open for output and/or input: - - - \e RTAUDIO_INPUT_OVERFLOW: Input data was discarded because of an overflow condition at the driver. - - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound. -*/ -typedef unsigned int RtAudioStreamStatus; -static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1; // Input data was discarded because of an overflow condition at the driver. -static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2; // The output buffer ran low, likely causing a gap in the output sound. - -//! RtAudio callback function prototype. -/*! - All RtAudio clients must create a function of type RtAudioCallback - to read and/or write data from/to the audio stream. When the - underlying audio system is ready for new input or output data, this - function will be invoked. - - \param outputBuffer For output (or duplex) streams, the client - should write \c nFrames of audio sample frames into this - buffer. This argument should be recast to the datatype - specified when the stream was opened. For input-only - streams, this argument will be NULL. - - \param inputBuffer For input (or duplex) streams, this buffer will - hold \c nFrames of input audio sample frames. This - argument should be recast to the datatype specified when the - stream was opened. For output-only streams, this argument - will be NULL. - - \param nFrames The number of sample frames of input or output - data in the buffers. The actual buffer size in bytes is - dependent on the data type and number of channels in use. - - \param streamTime The number of seconds that have elapsed since the - stream was started. - - \param status If non-zero, this argument indicates a data overflow - or underflow condition for the stream. The particular - condition can be determined by comparison with the - RtAudioStreamStatus flags. - - \param userData A pointer to optional data provided by the client - when opening the stream (default = NULL). - - \return - To continue normal stream operation, the RtAudioCallback function - should return a value of zero. To stop the stream and drain the - output buffer, the function should return a value of one. To abort - the stream immediately, the client should return a value of two. - */ -typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer, - unsigned int nFrames, - double streamTime, - RtAudioStreamStatus status, - void *userData ); - -/************************************************************************/ -/*! \class RtAudioError - \brief Exception handling class for RtAudio. - - The RtAudioError class is quite simple but it does allow errors to be - "caught" by RtAudioError::Type. See the RtAudio documentation to know - which methods can throw an RtAudioError. -*/ -/************************************************************************/ - -class RTAUDIO_DLL_PUBLIC RtAudioError : public std::runtime_error -{ - public: - //! Defined RtAudioError types. - enum Type { - WARNING, /*!< A non-critical error. */ - DEBUG_WARNING, /*!< A non-critical error which might be useful for debugging. */ - UNSPECIFIED, /*!< The default, unspecified error type. */ - NO_DEVICES_FOUND, /*!< No devices found on system. */ - INVALID_DEVICE, /*!< An invalid device ID was specified. */ - MEMORY_ERROR, /*!< An error occured during memory allocation. */ - INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */ - INVALID_USE, /*!< The function was called incorrectly. */ - DRIVER_ERROR, /*!< A system driver error occured. */ - SYSTEM_ERROR, /*!< A system error occured. */ - THREAD_ERROR /*!< A thread error occured. */ - }; - - //! The constructor. - RtAudioError( const std::string& message, - Type type = RtAudioError::UNSPECIFIED ) - : std::runtime_error(message), type_(type) {} - - //! Prints thrown error message to stderr. - virtual void printMessage( void ) const - { std::cerr << '\n' << what() << "\n\n"; } - - //! Returns the thrown error message type. - virtual const Type& getType(void) const { return type_; } - - //! Returns the thrown error message string. - virtual const std::string getMessage(void) const - { return std::string(what()); } - - protected: - Type type_; -}; - -//! RtAudio error callback function prototype. -/*! - \param type Type of error. - \param errorText Error description. - */ -typedef void (*RtAudioErrorCallback)( RtAudioError::Type type, const std::string &errorText ); - -// **************************************************************** // -// -// RtAudio class declaration. -// -// RtAudio is a "controller" used to select an available audio i/o -// interface. It presents a common API for the user to call but all -// functionality is implemented by the class RtApi and its -// subclasses. RtAudio creates an instance of an RtApi subclass -// based on the user's API choice. If no choice is made, RtAudio -// attempts to make a "logical" API selection. -// -// **************************************************************** // - -class RtApi; - -class RTAUDIO_DLL_PUBLIC RtAudio -{ - public: - - //! Audio API specifier arguments. - enum Api { - UNSPECIFIED, /*!< Search for a working compiled API. */ - LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */ - LINUX_PULSE, /*!< The Linux PulseAudio API. */ - LINUX_OSS, /*!< The Linux Open Sound System API. */ - UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */ - MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */ - WINDOWS_WASAPI, /*!< The Microsoft WASAPI API. */ - WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */ - WINDOWS_DS, /*!< The Microsoft DirectSound API. */ - RTAUDIO_DUMMY, /*!< A compilable but non-functional API. */ - NUM_APIS /*!< Number of values in this enum. */ - }; - - //! The public device information structure for returning queried values. - struct DeviceInfo { - bool probed; /*!< true if the device capabilities were successfully probed. */ - std::string name; /*!< Character string device identifier. */ - unsigned int outputChannels{}; /*!< Maximum output channels supported by device. */ - unsigned int inputChannels{}; /*!< Maximum input channels supported by device. */ - unsigned int duplexChannels{}; /*!< Maximum simultaneous input/output channels supported by device. */ - bool isDefaultOutput{false}; /*!< true if this is the default output device. */ - bool isDefaultInput{false}; /*!< true if this is the default input device. */ - std::vector sampleRates; /*!< Supported sample rates (queried from list of standard rates). */ - unsigned int preferredSampleRate{}; /*!< Preferred sample rate, e.g. for WASAPI the system sample rate. */ - RtAudioFormat nativeFormats{}; /*!< Bit mask of supported data formats. */ - }; - - //! The structure for specifying input or ouput stream parameters. - struct StreamParameters { - unsigned int deviceId{}; /*!< Device index (0 to getDeviceCount() - 1). */ - unsigned int nChannels{}; /*!< Number of channels. */ - unsigned int firstChannel{}; /*!< First channel index on device (default = 0). */ - }; - - //! The structure for specifying stream options. - /*! - The following flags can be OR'ed together to allow a client to - make changes to the default stream behavior: - - - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). - - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. - - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. - - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread. - - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only). - - By default, RtAudio streams pass and receive audio data from the - client in an interleaved format. By passing the - RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio - data will instead be presented in non-interleaved buffers. In - this case, each buffer argument in the RtAudioCallback function - will point to a single array of data, with \c nFrames samples for - each channel concatenated back-to-back. For example, the first - sample of data for the second channel would be located at index \c - nFrames (assuming the \c buffer pointer was recast to the correct - data type for the stream). - - Certain audio APIs offer a number of parameters that influence the - I/O latency of a stream. By default, RtAudio will attempt to set - these parameters internally for robust (glitch-free) performance - (though some APIs, like Windows DirectSound, make this difficult). - By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() - function, internal stream settings will be influenced in an attempt - to minimize stream latency, though possibly at the expense of stream - performance. - - If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to - open the input and/or output stream device(s) for exclusive use. - Note that this is not possible with all supported audio APIs. - - If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt - to select realtime scheduling (round-robin) for the callback thread. - The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME - flag is set. It defines the thread's realtime priority. - - If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to - open the "default" PCM device when using the ALSA API. Note that this - will override any specified input or output device id. - - The \c numberOfBuffers parameter can be used to control stream - latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs - only. A value of two is usually the smallest allowed. Larger - numbers can potentially result in more robust stream performance, - though likely at the cost of stream latency. The value set by the - user is replaced during execution of the RtAudio::openStream() - function by the value actually used by the system. - - The \c streamName parameter can be used to set the client name - when using the Jack API. By default, the client name is set to - RtApiJack. However, if you wish to create multiple instances of - RtAudio with Jack, each instance must have a unique client name. - */ - struct StreamOptions { - RtAudioStreamFlags flags{}; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */ - unsigned int numberOfBuffers{}; /*!< Number of stream buffers. */ - std::string streamName; /*!< A stream name (currently used only in Jack). */ - int priority{}; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */ - }; - - //! A static function to determine the current RtAudio version. - static std::string getVersion( void ); - - //! A static function to determine the available compiled audio APIs. - /*! - The values returned in the std::vector can be compared against - the enumerated list values. Note that there can be more than one - API compiled for certain operating systems. - */ - static void getCompiledApi( std::vector &apis ); - - //! Return the name of a specified compiled audio API. - /*! - This obtains a short lower-case name used for identification purposes. - This value is guaranteed to remain identical across library versions. - If the API is unknown, this function will return the empty string. - */ - static std::string getApiName( RtAudio::Api api ); - - //! Return the display name of a specified compiled audio API. - /*! - This obtains a long name used for display purposes. - If the API is unknown, this function will return the empty string. - */ - static std::string getApiDisplayName( RtAudio::Api api ); - - //! Return the compiled audio API having the given name. - /*! - A case insensitive comparison will check the specified name - against the list of compiled APIs, and return the one which - matches. On failure, the function returns UNSPECIFIED. - */ - static RtAudio::Api getCompiledApiByName( const std::string &name ); - - //! The class constructor. - /*! - The constructor performs minor initialization tasks. An exception - can be thrown if no API support is compiled. - - If no API argument is specified and multiple API support has been - compiled, the default order of use is JACK, ALSA, OSS (Linux - systems) and ASIO, DS (Windows systems). - */ - RtAudio( RtAudio::Api api=UNSPECIFIED ); - - //! The destructor. - /*! - If a stream is running or open, it will be stopped and closed - automatically. - */ - ~RtAudio(); - - //! Returns the audio API specifier for the current instance of RtAudio. - RtAudio::Api getCurrentApi( void ); - - //! A public function that queries for the number of audio devices available. - /*! - This function performs a system query of available devices each time it - is called, thus supporting devices connected \e after instantiation. If - a system error occurs during processing, a warning will be issued. - */ - unsigned int getDeviceCount( void ); - - //! Return an RtAudio::DeviceInfo structure for a specified device number. - /*! - - Any device integer between 0 and getDeviceCount() - 1 is valid. - If an invalid argument is provided, an RtAudioError (type = INVALID_USE) - will be thrown. If a device is busy or otherwise unavailable, the - structure member "probed" will have a value of "false" and all - other members are undefined. If the specified device is the - current default input or output device, the corresponding - "isDefault" member will have a value of "true". - */ - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); - - //! A function that returns the index of the default output device. - /*! - If the underlying audio API does not provide a "default - device", or if no devices are available, the return value will be - 0. Note that this is a valid device identifier and it is the - client's responsibility to verify that a device is available - before attempting to open a stream. - */ - unsigned int getDefaultOutputDevice( void ); - - //! A function that returns the index of the default input device. - /*! - If the underlying audio API does not provide a "default - device", or if no devices are available, the return value will be - 0. Note that this is a valid device identifier and it is the - client's responsibility to verify that a device is available - before attempting to open a stream. - */ - unsigned int getDefaultInputDevice( void ); - - //! A public function for opening a stream with the specified parameters. - /*! - An RtAudioError (type = SYSTEM_ERROR) is thrown if a stream cannot be - opened with the specified parameters or an error occurs during - processing. An RtAudioError (type = INVALID_USE) is thrown if any - invalid device ID or channel number parameters are specified. - - \param outputParameters Specifies output stream parameters to use - when opening a stream, including a device ID, number of channels, - and starting channel number. For input-only streams, this - argument should be NULL. The device ID is an index value between - 0 and getDeviceCount() - 1. - \param inputParameters Specifies input stream parameters to use - when opening a stream, including a device ID, number of channels, - and starting channel number. For output-only streams, this - argument should be NULL. The device ID is an index value between - 0 and getDeviceCount() - 1. - \param format An RtAudioFormat specifying the desired sample data format. - \param sampleRate The desired sample rate (sample frames per second). - \param *bufferFrames A pointer to a value indicating the desired - internal buffer size in sample frames. The actual value - used by the device is returned via the same pointer. A - value of zero can be specified, in which case the lowest - allowable value is determined. - \param callback A client-defined function that will be invoked - when input data is available and/or output data is needed. - \param userData An optional pointer to data that can be accessed - from within the callback function. - \param options An optional pointer to a structure containing various - global stream options, including a list of OR'ed RtAudioStreamFlags - and a suggested number of stream buffers that can be used to - control stream latency. More buffers typically result in more - robust performance, though at a cost of greater latency. If a - value of zero is specified, a system-specific median value is - chosen. If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the - lowest allowable value is used. The actual value used is - returned via the structure argument. The parameter is API dependent. - \param errorCallback A client-defined function that will be invoked - when an error has occured. - */ - void openStream( RtAudio::StreamParameters *outputParameters, - RtAudio::StreamParameters *inputParameters, - RtAudioFormat format, unsigned int sampleRate, - unsigned int *bufferFrames, RtAudioCallback callback, - void *userData = NULL, RtAudio::StreamOptions *options = NULL, RtAudioErrorCallback errorCallback = NULL ); - - //! A function that closes a stream and frees any associated stream memory. - /*! - If a stream is not open, this function issues a warning and - returns (no exception is thrown). - */ - void closeStream( void ); - - //! A function that starts a stream. - /*! - An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs - during processing. An RtAudioError (type = INVALID_USE) is thrown if a - stream is not open. A warning is issued if the stream is already - running. - */ - void startStream( void ); - - //! Stop a stream, allowing any samples remaining in the output queue to be played. - /*! - An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs - during processing. An RtAudioError (type = INVALID_USE) is thrown if a - stream is not open. A warning is issued if the stream is already - stopped. - */ - void stopStream( void ); - - //! Stop a stream, discarding any samples remaining in the input/output queue. - /*! - An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs - during processing. An RtAudioError (type = INVALID_USE) is thrown if a - stream is not open. A warning is issued if the stream is already - stopped. - */ - void abortStream( void ); - - //! Returns true if a stream is open and false if not. - bool isStreamOpen( void ) const; - - //! Returns true if the stream is running and false if it is stopped or not open. - bool isStreamRunning( void ) const; - - //! Returns the number of elapsed seconds since the stream was started. - /*! - If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. - */ - double getStreamTime( void ); - - //! Set the stream time to a time in seconds greater than or equal to 0.0. - /*! - If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. - */ - void setStreamTime( double time ); - - //! Returns the internal stream latency in sample frames. - /*! - The stream latency refers to delay in audio input and/or output - caused by internal buffering by the audio system and/or hardware. - For duplex streams, the returned value will represent the sum of - the input and output latencies. If a stream is not open, an - RtAudioError (type = INVALID_USE) will be thrown. If the API does not - report latency, the return value will be zero. - */ - long getStreamLatency( void ); - - //! Returns actual sample rate in use by the stream. - /*! - On some systems, the sample rate used may be slightly different - than that specified in the stream parameters. If a stream is not - open, an RtAudioError (type = INVALID_USE) will be thrown. - */ - unsigned int getStreamSampleRate( void ); - - //! Specify whether warning messages should be printed to stderr. - void showWarnings( bool value = true ); - - protected: - - void openRtApi( RtAudio::Api api ); - RtApi *rtapi_; -}; - -// Operating system dependent thread functionality. -#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) - - #ifndef NOMINMAX - #define NOMINMAX - #endif - #include - #include - #include - - typedef uintptr_t ThreadHandle; - typedef CRITICAL_SECTION StreamMutex; - -#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) - // Using pthread library for various flavors of unix. - #include - - typedef pthread_t ThreadHandle; - typedef pthread_mutex_t StreamMutex; - -#else // Setup for "dummy" behavior - - #define __RTAUDIO_DUMMY__ - typedef int ThreadHandle; - typedef int StreamMutex; - -#endif - -// This global structure type is used to pass callback information -// between the private RtAudio stream structure and global callback -// handling functions. -struct CallbackInfo { - void *object{}; // Used as a "this" pointer. - ThreadHandle thread{}; - void *callback{}; - void *userData{}; - void *errorCallback{}; - void *apiInfo{}; // void pointer for API specific callback information - bool isRunning{false}; - bool doRealtime{false}; - int priority{}; -}; - -// **************************************************************** // -// -// RtApi class declaration. -// -// Subclasses of RtApi contain all API- and OS-specific code necessary -// to fully implement the RtAudio API. -// -// Note that RtApi is an abstract base class and cannot be -// explicitly instantiated. The class RtAudio will create an -// instance of an RtApi subclass (RtApiOss, RtApiAlsa, -// RtApiJack, RtApiCore, RtApiDs, or RtApiAsio). -// -// **************************************************************** // - -#pragma pack(push, 1) -class S24 { - - protected: - unsigned char c3[3]; - - public: - S24() {} - - S24& operator = ( const int& i ) { - c3[0] = (i & 0x000000ff); - c3[1] = (i & 0x0000ff00) >> 8; - c3[2] = (i & 0x00ff0000) >> 16; - return *this; - } - - S24( const double& d ) { *this = (int) d; } - S24( const float& f ) { *this = (int) f; } - S24( const signed short& s ) { *this = (int) s; } - S24( const char& c ) { *this = (int) c; } - - int asInt() { - int i = c3[0] | (c3[1] << 8) | (c3[2] << 16); - if (i & 0x800000) i |= ~0xffffff; - return i; - } -}; -#pragma pack(pop) - -#if defined( HAVE_GETTIMEOFDAY ) - #include -#endif - -#include - -class RTAUDIO_DLL_PUBLIC RtApi -{ -public: - - RtApi(); - virtual ~RtApi(); - virtual RtAudio::Api getCurrentApi( void ) = 0; - virtual unsigned int getDeviceCount( void ) = 0; - virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0; - virtual unsigned int getDefaultInputDevice( void ); - virtual unsigned int getDefaultOutputDevice( void ); - void openStream( RtAudio::StreamParameters *outputParameters, - RtAudio::StreamParameters *inputParameters, - RtAudioFormat format, unsigned int sampleRate, - unsigned int *bufferFrames, RtAudioCallback callback, - void *userData, RtAudio::StreamOptions *options, - RtAudioErrorCallback errorCallback ); - virtual void closeStream( void ); - virtual void startStream( void ) = 0; - virtual void stopStream( void ) = 0; - virtual void abortStream( void ) = 0; - long getStreamLatency( void ); - unsigned int getStreamSampleRate( void ); - virtual double getStreamTime( void ); - virtual void setStreamTime( double time ); - bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; } - bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; } - void showWarnings( bool value ) { showWarnings_ = value; } - - -protected: - - static const unsigned int MAX_SAMPLE_RATES; - static const unsigned int SAMPLE_RATES[]; - - enum { FAILURE, SUCCESS }; - - enum StreamState { - STREAM_STOPPED, - STREAM_STOPPING, - STREAM_RUNNING, - STREAM_CLOSED = -50 - }; - - enum StreamMode { - OUTPUT, - INPUT, - DUPLEX, - UNINITIALIZED = -75 - }; - - // A protected structure used for buffer conversion. - struct ConvertInfo { - int channels; - int inJump, outJump; - RtAudioFormat inFormat, outFormat; - std::vector inOffset; - std::vector outOffset; - }; - - // A protected structure for audio streams. - struct RtApiStream { - unsigned int device[2]; // Playback and record, respectively. - void *apiHandle; // void pointer for API specific stream handle information - StreamMode mode; // OUTPUT, INPUT, or DUPLEX. - StreamState state; // STOPPED, RUNNING, or CLOSED - char *userBuffer[2]; // Playback and record, respectively. - char *deviceBuffer; - bool doConvertBuffer[2]; // Playback and record, respectively. - bool userInterleaved; - bool deviceInterleaved[2]; // Playback and record, respectively. - bool doByteSwap[2]; // Playback and record, respectively. - unsigned int sampleRate; - unsigned int bufferSize; - unsigned int nBuffers; - unsigned int nUserChannels[2]; // Playback and record, respectively. - unsigned int nDeviceChannels[2]; // Playback and record channels, respectively. - unsigned int channelOffset[2]; // Playback and record, respectively. - unsigned long latency[2]; // Playback and record, respectively. - RtAudioFormat userFormat; - RtAudioFormat deviceFormat[2]; // Playback and record, respectively. - StreamMutex mutex; - CallbackInfo callbackInfo; - ConvertInfo convertInfo[2]; - double streamTime; // Number of elapsed seconds since the stream started. - -#if defined(HAVE_GETTIMEOFDAY) - struct timeval lastTickTimestamp; -#endif - - RtApiStream() - :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; } - }; - - typedef S24 Int24; - typedef signed short Int16; - typedef signed int Int32; - typedef float Float32; - typedef double Float64; - - std::ostringstream errorStream_; - std::string errorText_; - bool showWarnings_; - RtApiStream stream_; - bool firstErrorOccurred_; - - /*! - Protected, api-specific method that attempts to open a device - with the given parameters. This function MUST be implemented by - all subclasses. If an error is encountered during the probe, a - "warning" message is reported and FAILURE is returned. A - successful probe is indicated by a return value of SUCCESS. - */ - virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ); - - //! A protected function used to increment the stream time. - void tickStreamTime( void ); - - //! Protected common method to clear an RtApiStream structure. - void clearStreamInfo(); - - /*! - Protected common method that throws an RtAudioError (type = - INVALID_USE) if a stream is not open. - */ - void verifyStream( void ); - - //! Protected common error method to allow global control over error handling. - void error( RtAudioError::Type type ); - - /*! - Protected method used to perform format, channel number, and/or interleaving - conversions between the user and device buffers. - */ - void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info ); - - //! Protected common method used to perform byte-swapping on buffers. - void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ); - - //! Protected common method that returns the number of bytes for a given format. - unsigned int formatBytes( RtAudioFormat format ); - - //! Protected common method that sets up the parameters for buffer conversion. - void setConvertInfo( StreamMode mode, unsigned int firstChannel ); -}; - -// **************************************************************** // -// -// Inline RtAudio definitions. -// -// **************************************************************** // - -inline RtAudio::Api RtAudio :: getCurrentApi( void ) { return rtapi_->getCurrentApi(); } -inline unsigned int RtAudio :: getDeviceCount( void ) { return rtapi_->getDeviceCount(); } -inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); } -inline unsigned int RtAudio :: getDefaultInputDevice( void ) { return rtapi_->getDefaultInputDevice(); } -inline unsigned int RtAudio :: getDefaultOutputDevice( void ) { return rtapi_->getDefaultOutputDevice(); } -inline void RtAudio :: closeStream( void ) { return rtapi_->closeStream(); } -inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); } -inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); } -inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); } -inline bool RtAudio :: isStreamOpen( void ) const { return rtapi_->isStreamOpen(); } -inline bool RtAudio :: isStreamRunning( void ) const { return rtapi_->isStreamRunning(); } -inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); } -inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); } -inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); } -inline void RtAudio :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); } -inline void RtAudio :: showWarnings( bool value ) { rtapi_->showWarnings( value ); } - -// RtApi Subclass prototypes. - -#if defined(__MACOSX_CORE__) - -#include - -class RtApiCore: public RtApi -{ -public: - - RtApiCore(); - ~RtApiCore(); - RtAudio::Api getCurrentApi( void ) override { return RtAudio::MACOSX_CORE; } - unsigned int getDeviceCount( void ) override; - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; - unsigned int getDefaultOutputDevice( void ) override; - unsigned int getDefaultInputDevice( void ) override; - void closeStream( void ) override; - void startStream( void ) override; - void stopStream( void ) override; - void abortStream( void ) override; - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - bool callbackEvent( AudioDeviceID deviceId, - const AudioBufferList *inBufferList, - const AudioBufferList *outBufferList ); - - private: - - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) override; - static const char* getErrorCode( OSStatus code ); -}; - -#endif - -#if defined(__UNIX_JACK__) - -class RtApiJack: public RtApi -{ -public: - - RtApiJack(); - ~RtApiJack(); - RtAudio::Api getCurrentApi( void ) override { return RtAudio::UNIX_JACK; } - unsigned int getDeviceCount( void ) override; - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; - void closeStream( void ) override; - void startStream( void ) override; - void stopStream( void ) override; - void abortStream( void ) override; - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - bool callbackEvent( unsigned long nframes ); - - private: - - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) override; - - bool shouldAutoconnect_; -}; - -#endif - -#if defined(__WINDOWS_ASIO__) - -class RtApiAsio: public RtApi -{ -public: - - RtApiAsio(); - ~RtApiAsio(); - RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_ASIO; } - unsigned int getDeviceCount( void ) override; - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; - void closeStream( void ) override; - void startStream( void ) override; - void stopStream( void ) override; - void abortStream( void ) override; - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - bool callbackEvent( long bufferIndex ); - - private: - - std::vector devices_; - void saveDeviceInfo( void ); - bool coInitialized_; - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) override; -}; - -#endif - -#if defined(__WINDOWS_DS__) - -class RtApiDs: public RtApi -{ -public: - - RtApiDs(); - ~RtApiDs(); - RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_DS; } - unsigned int getDeviceCount( void ) override; - unsigned int getDefaultOutputDevice( void ) override; - unsigned int getDefaultInputDevice( void ) override; - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; - void closeStream( void ) override; - void startStream( void ) override; - void stopStream( void ) override; - void abortStream( void ) override; - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - void callbackEvent( void ); - - private: - - bool coInitialized_; - bool buffersRolling; - long duplexPrerollBytes; - std::vector dsDevices; - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) override; -}; - -#endif - -#if defined(__WINDOWS_WASAPI__) - -struct IMMDeviceEnumerator; - -class RtApiWasapi : public RtApi -{ -public: - RtApiWasapi(); - virtual ~RtApiWasapi(); - - RtAudio::Api getCurrentApi( void ) override { return RtAudio::WINDOWS_WASAPI; } - unsigned int getDeviceCount( void ) override; - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; - unsigned int getDefaultOutputDevice( void ) override; - unsigned int getDefaultInputDevice( void ) override; - void closeStream( void ) override; - void startStream( void ) override; - void stopStream( void ) override; - void abortStream( void ) override; - -private: - bool coInitialized_; - IMMDeviceEnumerator* deviceEnumerator_; - - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int* bufferSize, - RtAudio::StreamOptions* options ) override; - - static DWORD WINAPI runWasapiThread( void* wasapiPtr ); - static DWORD WINAPI stopWasapiThread( void* wasapiPtr ); - static DWORD WINAPI abortWasapiThread( void* wasapiPtr ); - void wasapiThread(); -}; - -#endif - -#if defined(__LINUX_ALSA__) - -class RtApiAlsa: public RtApi -{ -public: - - RtApiAlsa(); - ~RtApiAlsa(); - RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_ALSA; } - unsigned int getDeviceCount( void ) override; - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; - void closeStream( void ) override; - void startStream( void ) override; - void stopStream( void ) override; - void abortStream( void ) override; - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - void callbackEvent( void ); - - private: - - std::vector devices_; - void saveDeviceInfo( void ); - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) override; -}; - -#endif - -#if defined(__LINUX_PULSE__) - -class RtApiPulse: public RtApi -{ -public: - ~RtApiPulse(); - RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_PULSE; } - unsigned int getDeviceCount( void ) override; - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; - void closeStream( void ) override; - void startStream( void ) override; - void stopStream( void ) override; - void abortStream( void ) override; - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - void callbackEvent( void ); - - private: - - void collectDeviceInfo( void ); - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) override; -}; - -#endif - -#if defined(__LINUX_OSS__) - -class RtApiOss: public RtApi -{ -public: - - RtApiOss(); - ~RtApiOss(); - RtAudio::Api getCurrentApi() override { return RtAudio::LINUX_OSS; } - unsigned int getDeviceCount( void ) override; - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) override; - void closeStream( void ) override; - void startStream( void ) override; - void stopStream( void ) override; - void abortStream( void ) override; - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - void callbackEvent( void ); - - private: - - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) override; -}; - -#endif - -#if defined(__RTAUDIO_DUMMY__) - -class RtApiDummy: public RtApi -{ -public: - - RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtAudioError::WARNING ); } - RtAudio::Api getCurrentApi( void ) override { return RtAudio::RTAUDIO_DUMMY; } - unsigned int getDeviceCount( void ) override { return 0; } - RtAudio::DeviceInfo getDeviceInfo( unsigned int /*device*/ ) override { RtAudio::DeviceInfo info; return info; } - void closeStream( void ) override {} - void startStream( void ) override {} - void stopStream( void ) override {} - void abortStream( void ) override {} - - private: - - bool probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/, - unsigned int /*firstChannel*/, unsigned int /*sampleRate*/, - RtAudioFormat /*format*/, unsigned int * /*bufferSize*/, - RtAudio::StreamOptions * /*options*/ ) override { return false; } -}; - -#endif - -#endif - -// Indentation settings for Vim and Emacs -// -// Local Variables: -// c-basic-offset: 2 -// indent-tabs-mode: nil -// End: -// -// vim: et sts=2 sw=2 From 0795a3e2abe16f62bcb4fe317e0261744effe185 Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Mon, 24 May 2021 16:28:35 +0100 Subject: [PATCH 09/47] add rtaudio as submodule --- .gitmodules | 3 +++ rtaudio | 1 + 2 files changed, 4 insertions(+) create mode 160000 rtaudio diff --git a/.gitmodules b/.gitmodules index d9385a3f..0e24a440 100644 --- a/.gitmodules +++ b/.gitmodules @@ -1,3 +1,6 @@ [submodule "opus-tools"] path = opus-tools url = https://github.com/xiph/opus-tools.git +[submodule "rtaudio"] + path = rtaudio + url = https://github.com/thestk/rtaudio.git diff --git a/rtaudio b/rtaudio new file mode 160000 index 00000000..c9bf99d4 --- /dev/null +++ b/rtaudio @@ -0,0 +1 @@ +Subproject commit c9bf99d414cf81d19ef0ddd00212a4a58ccd99c6 From 65a3141b3cb4fe5d341bfc1a1d50818b07c65811 Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Mon, 24 May 2021 18:00:38 +0100 Subject: [PATCH 10/47] move ulaw to dedicated header --- audiohandler.cpp | 17 +- audiohandler.h | 749 +---------------------------------------- ulaw.h | 728 +++++++++++++++++++++++++++++++++++++++ wfview.pro | 18 +- wfview.vcxproj | 305 ++++------------- wfview.vcxproj.filters | 194 +++++------ 6 files changed, 920 insertions(+), 1091 deletions(-) create mode 100644 ulaw.h diff --git a/audiohandler.cpp b/audiohandler.cpp index 3997fdb3..fff49c04 100644 --- a/audiohandler.cpp +++ b/audiohandler.cpp @@ -5,7 +5,7 @@ #include "audiohandler.h" #include "logcategories.h" - +#include "ulaw.h" audioHandler::audioHandler(QObject* parent) : isInitialized(false), @@ -26,8 +26,8 @@ audioHandler::~audioHandler() if (audio.isStreamRunning()) { audio.stopStream(); + audio.closeStream(); } - delete buf; } bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, int port, quint8 resampleQuality) @@ -58,9 +58,13 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 aParams.nChannels = channels; aParams.firstChannel = 0; - info = audio.getDeviceInfo(aParams.deviceId); - - buf = new quint16[960]; + try { + info = audio.getDeviceInfo(aParams.deviceId); + } + catch (RtAudioError& e) { + qInfo(logAudio()) << "Device error:" << aParams.deviceId << ":" << QString::fromStdString(e.getMessage()); + return false; + } if (info.probed) { @@ -201,9 +205,6 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n return 0; } -quint16 audioHandler::getBuffer(int i) { - return buf[i]; -} int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status) { diff --git a/audiohandler.h b/audiohandler.h index 54bb0442..539c991b 100644 --- a/audiohandler.h +++ b/audiohandler.h @@ -29,728 +29,6 @@ typedef signed short MY_TYPE; #define MULAW_BIAS 33 #define MULAW_MAX 0x1fff -static unsigned char ulaw_encode[8193] = -{ 0xff, 0xfe, 0xfe, 0xfd, 0xfd, 0xfc, 0xfc, 0xfb, 0xfb, 0xfa, 0xfa, 0xf9, - 0xf9, 0xf8, 0xf8, 0xf7, 0xf7, 0xf6, 0xf6, 0xf5, 0xf5, 0xf4, 0xf4, 0xf3, - 0xf3, 0xf2, 0xf2, 0xf1, 0xf1, 0xf0, 0xf0, 0xef, 0xef, 0xef, 0xef, 0xee, - 0xee, 0xee, 0xee, 0xed, 0xed, 0xed, 0xed, 0xec, 0xec, 0xec, 0xec, 0xeb, - 0xeb, 0xeb, 0xeb, 0xea, 0xea, 0xea, 0xea, 0xe9, 0xe9, 0xe9, 0xe9, 0xe8, - 0xe8, 0xe8, 0xe8, 0xe7, 0xe7, 0xe7, 0xe7, 0xe6, 0xe6, 0xe6, 0xe6, 0xe5, - 0xe5, 0xe5, 0xe5, 0xe4, 0xe4, 0xe4, 0xe4, 0xe3, 0xe3, 0xe3, 0xe3, 0xe2, - 0xe2, 0xe2, 0xe2, 0xe1, 0xe1, 0xe1, 0xe1, 0xe0, 0xe0, 0xe0, 0xe0, 0xdf, - 0xdf, 0xdf, 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0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, - 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80 -}; - -static const qint16 ulaw_decode[256] = { - -32124, -31100, -30076, -29052, -28028, -27004, -25980, -24956, - -23932, -22908, -21884, -20860, -19836, -18812, -17788, -16764, - -15996, -15484, -14972, -14460, -13948, -13436, -12924, -12412, - -11900, -11388, -10876, -10364, -9852, -9340, -8828, -8316, - -7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140, - -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092, - -3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004, - -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980, - -1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436, - -1372, -1308, -1244, -1180, -1116, -1052, -988, -924, - -876, -844, -812, -780, -748, -716, -684, -652, - -620, -588, -556, -524, -492, -460, -428, -396, - -372, -356, -340, -324, -308, -292, -276, -260, - -244, -228, -212, -196, -180, -164, -148, -132, - -120, -112, -104, -96, -88, -80, -72, -64, - -56, -48, -40, -32, -24, -16, -8, 0, - 32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956, - 23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764, - 15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412, - 11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316, - 7932, 7676, 7420, 7164, 6908, 6652, 6396, 6140, - 5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092, - 3900, 3772, 3644, 3516, 3388, 3260, 3132, 3004, - 2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980, - 1884, 1820, 1756, 1692, 1628, 1564, 1500, 1436, - 1372, 1308, 1244, 1180, 1116, 1052, 988, 924, - 876, 844, 812, 780, 748, 716, 684, 652, - 620, 588, 556, 524, 492, 460, 428, 396, - 372, 356, 340, 324, 308, 292, 276, 260, - 244, 228, 212, 196, 180, 164, 148, 132, - 120, 112, 104, 96, 88, 80, 72, 64, - 56, 48, 40, 32, 24, 16, 8, 0 -}; - - struct audioPacket { quint32 seq; QTime time; @@ -775,18 +53,6 @@ public: void flush(); void stop(); - int readData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status); - - static int staticRead(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void* userData) { - return static_cast(userData)->readData(outputBuffer, inputBuffer, nFrames, streamTime, status); - } - - int writeData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status); - - static int staticWrite(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void* userData) { - return static_cast(userData)->writeData(outputBuffer, inputBuffer, nFrames, streamTime, status); - } - qint64 bytesAvailable() const; bool isSequential() const; void getNextAudioChunk(QByteArray &data); @@ -807,8 +73,19 @@ signals: private: + int readData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status); + + static int staticRead(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void* userData) { + return static_cast(userData)->readData(outputBuffer, inputBuffer, nFrames, streamTime, status); + } + + int writeData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status); + + static int staticWrite(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void* userData) { + return static_cast(userData)->writeData(outputBuffer, inputBuffer, nFrames, streamTime, status); + } + void reinit(); - quint16 getBuffer(int); bool isInitialized; RtAudio audio; @@ -830,8 +107,6 @@ private: quint8 radioSampleBits; quint8 radioChannels; - quint16 *buf; - QMapaudioBuffer; unsigned int ratioNum; diff --git a/ulaw.h b/ulaw.h new file mode 100644 index 00000000..9d673a51 --- /dev/null +++ b/ulaw.h @@ -0,0 +1,728 @@ +#ifndef ULAW_H +#define ULAW_H + + +static unsigned char ulaw_encode[8193] = +{ 0xff, 0xfe, 0xfe, 0xfd, 0xfd, 0xfc, 0xfc, 0xfb, 0xfb, 0xfa, 0xfa, 0xf9, + 0xf9, 0xf8, 0xf8, 0xf7, 0xf7, 0xf6, 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0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, + 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x81, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, + 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80 +}; + +static const qint16 ulaw_decode[256] = { + -32124, -31100, -30076, -29052, -28028, -27004, -25980, -24956, + -23932, -22908, -21884, -20860, -19836, -18812, -17788, -16764, + -15996, -15484, -14972, -14460, -13948, -13436, -12924, -12412, + -11900, -11388, -10876, -10364, -9852, -9340, -8828, -8316, + -7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140, + -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092, + -3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004, + -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980, + -1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436, + -1372, -1308, -1244, -1180, -1116, -1052, -988, -924, + -876, -844, -812, -780, -748, -716, -684, -652, + -620, -588, -556, -524, -492, -460, -428, -396, + -372, -356, -340, -324, -308, -292, -276, -260, + -244, -228, -212, -196, -180, -164, -148, -132, + -120, -112, -104, -96, -88, -80, -72, -64, + -56, -48, -40, -32, -24, -16, -8, 0, + 32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956, + 23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764, + 15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412, + 11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316, + 7932, 7676, 7420, 7164, 6908, 6652, 6396, 6140, + 5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092, + 3900, 3772, 3644, 3516, 3388, 3260, 3132, 3004, + 2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980, + 1884, 1820, 1756, 1692, 1628, 1564, 1500, 1436, + 1372, 1308, 1244, 1180, 1116, 1052, 988, 924, + 876, 844, 812, 780, 748, 716, 684, 652, + 620, 588, 556, 524, 492, 460, 428, 396, + 372, 356, 340, 324, 308, 292, 276, 260, + 244, 228, 212, 196, 180, 164, 148, 132, + 120, 112, 104, 96, 88, 80, 72, 64, + 56, 48, 40, 32, 24, 16, 8, 0 +}; + + + +#endif diff --git a/wfview.pro b/wfview.pro index fe9be6b4..644a8d64 100644 --- a/wfview.pro +++ b/wfview.pro @@ -16,10 +16,10 @@ CONFIG(debug, release|debug) { } else { # For Release builds only: -QMAKE_CXXFLAGS += -s +linux:QMAKE_CXXFLAGS += -s QMAKE_CXXFLAGS += -fvisibility=hidden QMAKE_CXXFLAGS += -fvisibility-inlines-hidden -QMAKE_LFLAGS += -O2 -s +linux:QMAKE_LFLAGS += -O2 -s } # The following define makes your compiler emit warnings if you use @@ -102,6 +102,9 @@ linux:LIBS += -L./ -l$$QCPLIB !linux:HEADERS += ../qcustomplot/qcustomplot.h !linux:INCLUDEPATH += ../qcustomplot +INCLUDEPATH += opus-tools/src +INCLUDEPATH += rtaudio + SOURCES += main.cpp\ wfmain.cpp \ commhandler.cpp \ @@ -118,7 +121,7 @@ SOURCES += main.cpp\ meter.cpp \ qledlabel.cpp \ pttyhandler.cpp \ - resampler/resample.c \ + opus-tools/src/resample.c \ repeatersetup.cpp \ rigctld.cpp \ rtaudio/RtAudio.cpp @@ -139,13 +142,14 @@ HEADERS += wfmain.h \ meter.h \ qledlabel.h \ pttyhandler.h \ - resampler/speex_resampler.h \ - resampler/arch.h \ - resampler/resample_sse.h \ + opus-tools/src/speex_resampler.h \ + opus-tools/src/arch.h \ + opus-tools/src/resample_sse.h \ repeatersetup.h \ repeaterattributes.h \ rigctld.h \ - rtaudio/RtAudio.h + rtaudio/RtAudio.h \ + ulaw.h FORMS += wfmain.ui \ diff --git a/wfview.vcxproj b/wfview.vcxproj index 816a3a0b..1dc80f4d 100644 --- a/wfview.vcxproj +++ b/wfview.vcxproj @@ -1,41 +1,24 @@  - - Debug - Win32 - Release Win32 - - Release - x64 - - + Debug - x64 + Win32 {326108AD-FA9D-3AAF-8D3E-062C4DDC34E2} wfview - QtVS_v303 - 10.0.18362.0 - 10.0.18362.0 + QtVS_v304 + 10.0.19041.0 + 10.0.19041.0 $(MSBuildProjectDirectory)\QtMsBuild - - v142 - release\ - false - NotSet - Application - release\ - wfview - v142 release\ @@ -45,15 +28,6 @@ release\ wfview - - v142 - debug\ - false - NotSet - Application - debug\ - wfview - v142 debug\ @@ -68,15 +42,9 @@ - - - - - - @@ -84,118 +52,34 @@ - + debug\ debug\ wfview true - $(IncludePath) - $(LibraryPath) - - wfview - true - C:\Users\Phil\source\repos\qcustomplot;$(IncludePath) - $(LibraryPath) - - + release\ release\ wfview true false - - wfview - true - false - - - msvc2019_64_5 - core;network;gui;multimedia;widgets;serialport;printsupport - - + msvc2019 core;network;gui;multimedia;widgets;serialport;printsupport - - msvc2019_64_5 - core;opengl;network;gui;multimedia;widgets;serialport;printsupport - - + msvc2019 - core;opengl;network;gui;multimedia;widgets;serialport;printsupport + core;network;gui;multimedia;widgets;serialport;printsupport - - - GeneratedFiles\$(ConfigurationName);GeneratedFiles;.;c:\qcustomplot;release;/include;%(AdditionalIncludeDirectories) - -Zc:rvalueCast -Zc:inline -Zc:strictStrings -Zc:throwingNew -Zc:referenceBinding -Zc:__cplusplus -s -w34100 -w34189 -w44996 -w44456 -w44457 -w44458 %(AdditionalOptions) - release\ - false - None - 4577;4467;%(DisableSpecificWarnings) - Sync - release\ - MaxSpeed - _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;NDEBUG;QT_NO_DEBUG;OUTSIDE_SPEEX;RANDOM_PREFIX=wf;%(PreprocessorDefinitions) - false - - - MultiThreadedDLL - true - true - Level3 - true - - - OpenGL32.lib;shell32.lib;%(AdditionalDependencies) - C:\openssl\lib;C:\Utils\my_sql\mysql-5.7.25-winx64\lib;C:\Utils\postgresql\pgsql\lib;%(AdditionalLibraryDirectories) - -O2 -s "/MANIFESTDEPENDENCY:type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' publicKeyToken='6595b64144ccf1df' language='*' processorArchitecture='*'" %(AdditionalOptions) - true - false - true - false - true - $(OutDir)\wfview.exe - true - Windows - true - - - Unsigned - None - 0 - - - _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;NDEBUG;QT_NO_DEBUG;QT_MULTIMEDIA_LIB;QT_OPENGL_LIB;QT_PRINTSUPPORT_LIB;QT_WIDGETS_LIB;QT_GUI_LIB;QT_SERIALPORT_LIB;QT_NETWORK_LIB;QT_CORE_LIB;%(PreprocessorDefinitions) - - - msvc - ./$(Configuration)/moc_predefs.h - Moc'ing %(Identity)... - output - $(Configuration) - moc_%(Filename).cpp - - - default - Rcc'ing %(Identity)... - $(Configuration) - qrc_%(Filename).cpp - - - Uic'ing %(Identity)... - $(ProjectDir) - ui_%(Filename).h - - - GeneratedFiles\$(ConfigurationName);GeneratedFiles;.;c:\qcustomplot;release;/include;%(AdditionalIncludeDirectories) - -Zc:rvalueCast -Zc:inline -Zc:strictStrings -Zc:throwingNew -Zc:referenceBinding -Zc:__cplusplus -s -w34100 -w34189 -w44996 -w44456 -w44457 -w44458 %(AdditionalOptions) + .;..\qcustomplot;opus-tools\src;rtaudio;release;/include;%(AdditionalIncludeDirectories) + -Zc:rvalueCast -Zc:inline -Zc:strictStrings -Zc:throwingNew -Zc:referenceBinding -Zc:__cplusplus -w34100 -w34189 -w44996 -w44456 -w44457 -w44458 %(AdditionalOptions) release\ false None @@ -203,7 +87,7 @@ Sync release\ MaxSpeed - _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;NDEBUG;QT_NO_DEBUG;OUTSIDE_SPEEX;RANDOM_PREFIX=wf;__WINDOWS_WASAPI__;%(PreprocessorDefinitions) + _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_COMPILE_LIBRARY;OUTSIDE_SPEEX;RANDOM_PREFIX=wf;__WINDOWS_WASAPI__;GITSHORT="2db02fb";HOST="wfview.org";UNAME="build";NDEBUG;QT_NO_DEBUG;%(PreprocessorDefinitions) false @@ -214,9 +98,9 @@ true - OpenGL32.lib;shell32.lib;%(AdditionalDependencies) - C:\openssl\lib;C:\Utils\my_sql\mysql-5.7.25-winx64\lib;C:\Utils\postgresql\pgsql\lib;%(AdditionalLibraryDirectories) - -O2 -s "/MANIFESTDEPENDENCY:type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' publicKeyToken='6595b64144ccf1df' language='*' processorArchitecture='*'" %(AdditionalOptions) + shell32.lib;%(AdditionalDependencies) + C:\opensslx86\lib;C:\Utils\my_sql\mysql-5.7.25-win32\lib;C:\Utils\postgresqlx86\pgsql\lib;%(AdditionalLibraryDirectories) + "/MANIFESTDEPENDENCY:type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' publicKeyToken='6595b64144ccf1df' language='*' processorArchitecture='*'" %(AdditionalOptions) true false true @@ -233,66 +117,7 @@ 0 - _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;NDEBUG;QT_NO_DEBUG;QT_MULTIMEDIA_LIB;QT_OPENGL_LIB;QT_PRINTSUPPORT_LIB;QT_WIDGETS_LIB;QT_GUI_LIB;QT_SERIALPORT_LIB;QT_NETWORK_LIB;QT_CORE_LIB;%(PreprocessorDefinitions) - - - msvc - ./$(Configuration)/moc_predefs.h - Moc'ing %(Identity)... - output - $(Configuration) - moc_%(Filename).cpp - - - default - Rcc'ing %(Identity)... - $(Configuration) - qrc_%(Filename).cpp - - - Uic'ing %(Identity)... - $(ProjectDir) - ui_%(Filename).h - - - - - GeneratedFiles\$(ConfigurationName);GeneratedFiles;.;c:\qcustomplot;debug;/include;%(AdditionalIncludeDirectories) - -Zc:rvalueCast -Zc:inline -Zc:strictStrings -Zc:throwingNew -Zc:referenceBinding -Zc:__cplusplus -w34100 -w34189 -w44996 -w44456 -w44457 -w44458 %(AdditionalOptions) - debug\ - false - ProgramDatabase - 4577;4467;%(DisableSpecificWarnings) - Sync - debug\ - Disabled - _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;OUTSIDE_SPEEX;RANDOM_PREFIX=wf;%(PreprocessorDefinitions) - false - MultiThreadedDebugDLL - true - true - Level3 - true - - - OpenGL32.lib;shell32.lib;%(AdditionalDependencies) - C:\openssl\lib;C:\Utils\my_sql\mysql-5.7.25-winx64\lib;C:\Utils\postgresql\pgsql\lib;%(AdditionalLibraryDirectories) - "/MANIFESTDEPENDENCY:type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' publicKeyToken='6595b64144ccf1df' language='*' processorArchitecture='*'" %(AdditionalOptions) - true - true - true - $(OutDir)\wfview.exe - true - Windows - true - - - Unsigned - None - 0 - - - _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;QT_MULTIMEDIA_LIB;QT_OPENGL_LIB;QT_PRINTSUPPORT_LIB;QT_WIDGETS_LIB;QT_GUI_LIB;QT_SERIALPORT_LIB;QT_NETWORK_LIB;QT_CORE_LIB;_DEBUG;%(PreprocessorDefinitions) + _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_COMPILE_LIBRARY;OUTSIDE_SPEEX;RANDOM_PREFIX=wf;__WINDOWS_WASAPI__;GITSHORT=\"2db02fb\";HOST=\"wfview.org\";UNAME=\"build\";NDEBUG;QT_NO_DEBUG;QT_MULTIMEDIA_LIB;QT_PRINTSUPPORT_LIB;QT_WIDGETS_LIB;QT_GUI_LIB;QT_SERIALPORT_LIB;QT_NETWORK_LIB;QT_CORE_LIB;%(PreprocessorDefinitions) msvc @@ -316,8 +141,8 @@ - GeneratedFiles\$(ConfigurationName);GeneratedFiles;.;c:\qcustomplot;debug;/include;%(AdditionalIncludeDirectories) - -Zc:rvalueCast -Zc:inline -Zc:strictStrings -Zc:throwingNew -Zc:referenceBinding -Zc:__cplusplus -w34100 -w34189 -w44996 -w44456 -w44457 -w44458 %(AdditionalOptions) + .;..\qcustomplot;opus-tools\src;rtaudio;debug;/include;%(AdditionalIncludeDirectories) + -Zc:rvalueCast -Zc:inline -Zc:strictStrings -Zc:throwingNew -Zc:referenceBinding -Zc:__cplusplus -w34100 -w34189 -w44996 -w44456 -w44457 -w44458 %(AdditionalOptions) debug\ false ProgramDatabase @@ -325,7 +150,7 @@ Sync debug\ Disabled - _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;OUTSIDE_SPEEX;RANDOM_PREFIX=wf;__WINDOWS_WASAPI__;%(PreprocessorDefinitions) + _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_COMPILE_LIBRARY;OUTSIDE_SPEEX;RANDOM_PREFIX=wf;__WINDOWS_WASAPI__;GITSHORT="2db02fb";HOST="wfview.org";UNAME="build";%(PreprocessorDefinitions) false MultiThreadedDebugDLL true @@ -334,8 +159,8 @@ true - OpenGL32.lib;shell32.lib;%(AdditionalDependencies) - C:\openssl\lib;C:\Utils\my_sql\mysql-5.7.25-winx64\lib;C:\Utils\postgresql\pgsql\lib;%(AdditionalLibraryDirectories) + shell32.lib;%(AdditionalDependencies) + C:\opensslx86\lib;C:\Utils\my_sql\mysql-5.7.25-win32\lib;C:\Utils\postgresqlx86\pgsql\lib;%(AdditionalLibraryDirectories) "/MANIFESTDEPENDENCY:type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' publicKeyToken='6595b64144ccf1df' language='*' processorArchitecture='*'" %(AdditionalOptions) true true @@ -351,7 +176,7 @@ 0 - _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;WIN64;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_USE_OPENGL;HOST=1;UNAME=1;GITSHORT=1;QT_MULTIMEDIA_LIB;QT_OPENGL_LIB;QT_PRINTSUPPORT_LIB;QT_WIDGETS_LIB;QT_GUI_LIB;QT_SERIALPORT_LIB;QT_NETWORK_LIB;QT_CORE_LIB;_DEBUG;%(PreprocessorDefinitions) + _WINDOWS;UNICODE;_UNICODE;WIN32;_ENABLE_EXTENDED_ALIGNED_STORAGE;QT_DEPRECATED_WARNINGS;QCUSTOMPLOT_COMPILE_LIBRARY;OUTSIDE_SPEEX;RANDOM_PREFIX=wf;__WINDOWS_WASAPI__;GITSHORT=\"2db02fb\";HOST=\"wfview.org\";UNAME=\"build\";QT_MULTIMEDIA_LIB;QT_PRINTSUPPORT_LIB;QT_WIDGETS_LIB;QT_GUI_LIB;QT_SERIALPORT_LIB;QT_NETWORK_LIB;QT_CORE_LIB;_DEBUG;%(PreprocessorDefinitions) msvc @@ -374,22 +199,22 @@ - + - - + + + - + - @@ -397,65 +222,74 @@ - - - - - + + + + + + - - + + + + + + + + + + + + + - - - - - - + + + + + + + + + Document - true true - $(QTDIR)\mkspecs\features\data\dummy.cpp;%(AdditionalInputs) $(QTDIR)\mkspecs\features\data\dummy.cpp;%(AdditionalInputs) - cl -Bx"$(QTDIR)\bin\qmake.exe" -nologo -Zc:wchar_t -FS -Zc:rvalueCast -Zc:inline -Zc:strictStrings -Zc:throwingNew -Zc:referenceBinding -Zc:__cplusplus -Zi -MDd -W3 -w34100 -w34189 -w44996 -w44456 -w44457 -w44458 -wd4577 -wd4467 -E $(QTDIR)\mkspecs\features\data\dummy.cpp 2>NUL >debug\moc_predefs.h - cl -Bx"$(QTDIR)\bin\qmake.exe" -nologo -Zc:wchar_t -FS -Zc:rvalueCast -Zc:inline -Zc:strictStrings -Zc:throwingNew -Zc:referenceBinding -Zc:__cplusplus -Zi -MDd -W3 -w34100 -w34189 -w44996 -w44456 -w44457 -w44458 -wd4577 -wd4467 -E $(QTDIR)\mkspecs\features\data\dummy.cpp 2>NUL >debug\moc_predefs.h - Generate moc_predefs.h + cl -Bx"$(QTDIR)\bin\qmake.exe" -nologo -Zc:wchar_t -FS -Zc:rvalueCast -Zc:inline -Zc:strictStrings -Zc:throwingNew -Zc:referenceBinding -Zc:__cplusplus -Zi -MDd -W3 -w34100 -w34189 -w44996 -w44456 -w44457 -w44458 -wd4577 -wd4467 -E $(QTDIR)\mkspecs\features\data\dummy.cpp 2>NUL >debug\moc_predefs.h Generate moc_predefs.h - debug\moc_predefs.h;%(Outputs) debug\moc_predefs.h;%(Outputs) Document - $(QTDIR)\mkspecs\features\data\dummy.cpp;%(AdditionalInputs) $(QTDIR)\mkspecs\features\data\dummy.cpp;%(AdditionalInputs) - cl -Bx"$(QTDIR)\bin\qmake.exe" -nologo -Zc:wchar_t -FS -Zc:rvalueCast -Zc:inline -Zc:strictStrings -Zc:throwingNew -Zc:referenceBinding -Zc:__cplusplus -s -fvisibility=hidden -fvisibility-inlines-hidden -O2 -MD -W3 -w34100 -w34189 -w44996 -w44456 -w44457 -w44458 -wd4577 -wd4467 -E $(QTDIR)\mkspecs\features\data\dummy.cpp 2>NUL >release\moc_predefs.h - cl -Bx"$(QTDIR)\bin\qmake.exe" -nologo -Zc:wchar_t -FS -Zc:rvalueCast -Zc:inline -Zc:strictStrings -Zc:throwingNew -Zc:referenceBinding -Zc:__cplusplus -s -fvisibility=hidden -fvisibility-inlines-hidden -O2 -MD -W3 -w34100 -w34189 -w44996 -w44456 -w44457 -w44458 -wd4577 -wd4467 -E $(QTDIR)\mkspecs\features\data\dummy.cpp 2>NUL >release\moc_predefs.h - Generate moc_predefs.h + cl -Bx"$(QTDIR)\bin\qmake.exe" -nologo -Zc:wchar_t -FS -Zc:rvalueCast -Zc:inline -Zc:strictStrings -Zc:throwingNew -Zc:referenceBinding -Zc:__cplusplus -fvisibility=hidden -fvisibility-inlines-hidden -O2 -MD -W3 -w34100 -w34189 -w44996 -w44456 -w44457 -w44458 -wd4577 -wd4467 -E $(QTDIR)\mkspecs\features\data\dummy.cpp 2>NUL >release\moc_predefs.h Generate moc_predefs.h - release\moc_predefs.h;%(Outputs) release\moc_predefs.h;%(Outputs) - true true - - - - + + + + + + + + @@ -490,18 +324,14 @@ - resources resources - resources resources - style style - style style @@ -519,6 +349,9 @@ + + + diff --git a/wfview.vcxproj.filters b/wfview.vcxproj.filters index 94c1d1e9..e09d8dbb 100644 --- a/wfview.vcxproj.filters +++ b/wfview.vcxproj.filters @@ -57,6 +57,15 @@ + + Source Files + + + Source Files + + + Source Files + Source Files @@ -69,59 +78,62 @@ Source Files - - Source Files - - - Source Files - - - Source Files - - - Source Files - - - Source Files - - - Source Files - - - Source Files - - - Source Files - Source Files - - Source Files - Source Files - + + Source Files + + Source Files Source Files + + Source Files + + + Source Files + Source Files - + Source Files - + Source Files - + + Source Files + + + Source Files + + + Source Files + + Source Files + + Header Files + + + Header Files + + + Header Files + + + Header Files + Header Files @@ -131,51 +143,60 @@ Header Files - - Header Files - - - Header Files - - - Header Files - - - Header Files - - - Header Files - - - Header Files - - - Header Files - - - Header Files - - - Header Files - Header Files - + Header Files - + Header Files - - Header Files - - - Header Files - Header Files + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + + + Header Files + @@ -186,10 +207,10 @@ - + Form Files - + Form Files @@ -198,7 +219,7 @@ Form Files - + Form Files @@ -333,31 +354,6 @@ Resource Files - - - - - - - - - - - - - - - - - - - - - - - - - @@ -371,14 +367,6 @@ - - Header Files - - - Header Files - - - Header Files - + \ No newline at end of file From f567d5ab04f83de7b6a76f9d852465ffbdf88e6e Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Thu, 27 May 2021 11:41:08 +0100 Subject: [PATCH 11/47] Use ring buffer with rtaudio to eliminate mutexes --- audiohandler.cpp | 220 ++++++++++++--------- audiohandler.h | 14 +- ring/LICENSE | 21 ++ ring/README.md | 10 + ring/ring.cpp | 290 +++++++++++++++++++++++++++ ring/ring.h | 440 +++++++++++++++++++++++++++++++++++++++++ udphandler.cpp | 17 +- udphandler.h | 2 + udpserver.cpp | 12 +- wfview.vcxproj | 2 + wfview.vcxproj.filters | 7 + 11 files changed, 923 insertions(+), 112 deletions(-) create mode 100644 ring/LICENSE create mode 100644 ring/README.md create mode 100644 ring/ring.cpp create mode 100644 ring/ring.h diff --git a/audiohandler.cpp b/audiohandler.cpp index fff49c04..39265f12 100644 --- a/audiohandler.cpp +++ b/audiohandler.cpp @@ -28,6 +28,8 @@ audioHandler::~audioHandler() audio.stopStream(); audio.closeStream(); } + if (ringBuf != Q_NULLPTR) + delete ringBuf; } bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, int port, quint8 resampleQuality) @@ -46,6 +48,9 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 // chunk size is always relative to Internal Sample Rate. this->chunkSize = (INTERNAL_SAMPLE_RATE / 25) * radioChannels; + ringBuf = new wilt::Ring(100); // Should be customizable. + tempBuf.sent = 0; + if (port > 0) { aParams.deviceId = port; } @@ -55,7 +60,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 else { aParams.deviceId = audio.getDefaultOutputDevice(); } - aParams.nChannels = channels; + aParams.nChannels = 2; // Internally this is always 2 channels aParams.firstChannel = 0; try { @@ -140,67 +145,70 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n { // Calculate output length, always full samples int sentlen = 0; - qint16* buffer = (qint16*)outputBuffer; - //qDebug(logAudio()) << "looking for: " << nFrames << this->audioBuffer.size(); + + quint8* buffer = (quint8*)outputBuffer; if (status == RTAUDIO_OUTPUT_UNDERFLOW) qDebug(logAudio()) << "Underflow detected"; + unsigned int nBytes = nFrames * 2 * 2; // This is ALWAYS 2 bytes per sample and 2 channels - if (!audioBuffer.isEmpty()) + if (ringBuf->size()>0) { - mutex.lock(); // Output buffer is ALWAYS 16 bit. - auto packet = audioBuffer.begin(); - while (packet != audioBuffer.end() && sentlen < nFrames/2) + while (sentlen < nBytes) { - int timediff = packet->time.msecsTo(QTime::currentTime()); - - if (timediff > (int)audioLatency * 2) { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet->seq << - " arrived too late (increase output latency!) " << - dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; - while (packet != audioBuffer.end() && timediff > (int)audioLatency) { - timediff = packet->time.msecsTo(QTime::currentTime()); - lastSeq = packet->seq; - packet = audioBuffer.erase(packet); // returns next packet - } - if (packet == audioBuffer.end()) { - break; - } - } - - // If we got here then packet time must be within latency threshold - - if (packet->seq == lastSeq + 1 || packet->seq <= lastSeq) + audioPacket packet; + if (!ringBuf->try_read(packet)) { - int send = qMin((int)nFrames*2 - sentlen, packet->dataout.length() - packet->sent); - lastSeq = packet->seq; - //qInfo(logAudio()) << "Packet " << hex << packet->seq << " arrived on time " << Qt::dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; - - memcpy(buffer + sentlen, packet->dataout.constData() + packet->sent, send); + qDebug() << "No more data available but buffer is not full! sentlen:" << sentlen << " nBytes:" << nBytes ; + return 0; + } + currentLatency = packet.time.msecsTo(QTime::currentTime()); + // This shouldn't be required but if we did output a partial packet + // This will add the remaining packet data to the output buffer. + if (tempBuf.sent != tempBuf.data.length()) + { + int send = qMin((int)nBytes - sentlen, tempBuf.data.length() - tempBuf.sent); + memcpy(buffer + sentlen, tempBuf.data.constData() + tempBuf.sent, send); + tempBuf.sent = tempBuf.sent + send; sentlen = sentlen + send; + qDebug(logAudio()) << "Adding partial:" << send; + } - if (send == packet->dataout.length() - packet->sent) - { - //qInfo(logAudio()) << "Get next packet"; - packet = audioBuffer.erase(packet); // returns next packet - } - else - { - // Store sent amount (could be zero if audioOutput buffer full) then break. - packet->sent = send; - break; - } + while (currentLatency > (int)audioLatency) { + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet.seq << + " arrived too late (increase output latency!) " << + dec << packet.time.msecsTo(QTime::currentTime()) << "ms"; + lastSeq = packet.seq; + if (!ringBuf->try_read(packet)) + return sentlen; + currentLatency = packet.time.msecsTo(QTime::currentTime()); } - else { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet->seq - 1; - lastSeq = packet->seq; + + int send = qMin((int)nBytes - sentlen, packet.data.length()); + memcpy(buffer + sentlen, packet.data.constData(), send); + sentlen = sentlen + send; + if (send < packet.data.length()) + { + qDebug(logAudio()) << "Asking for partial, sent:" << send << "packet length" << packet.data.length(); + tempBuf = packet; + tempBuf.sent = tempBuf.sent + send; + //lastSeq = packet.seq; + //break; } + + if (packet.seq <= lastSeq) { + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Duplicate/early audio packet: " << hex << lastSeq << " got " << hex << packet.seq; + } + else if (packet.seq != lastSeq + 1) { + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet.seq - 1; + } + lastSeq = packet.seq; } - mutex.unlock(); } + //qDebug(logAudio()) << "looking for: " << nBytes << " got: " << sentlen; return 0; } @@ -231,13 +239,13 @@ int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int int send = qMin((int)(len - sentlen), (int)chunkSize - current->sent); - current->datain.append(QByteArray::fromRawData(data + sentlen, send)); + current->data.append(QByteArray::fromRawData(data + sentlen, send)); sentlen = sentlen + send; current->seq = 0; // Not used in TX current->time = QTime::currentTime(); - current->sent = current->datain.length(); + current->sent = current->data.length(); if (current->sent == chunkSize) { @@ -274,64 +282,82 @@ void audioHandler::stateChanged(QAudio::State state) -void audioHandler::incomingAudio(audioPacket data) +int audioHandler::incomingAudio(audioPacket data) { // No point buffering audio until stream is actually running. if (!audio.isStreamRunning()) { - return; + qDebug(logAudio()) << "Packet received before stream was started"; + return currentLatency; } - // Incoming data is 8bits? - if (radioSampleBits == 8) + // Incoming data is 8bits? + if (radioSampleBits == 8) + { + QByteArray outPacket((int)data.data.length() * 2, (char)0xff); + qint16* out = (qint16*)outPacket.data(); + for (int f = 0; f < data.data.length(); f++) { - QByteArray inPacket((int)data.datain.length() * 2, (char)0xff); - qint16* in = (qint16*)inPacket.data(); - for (int f = 0; f < data.datain.length(); f++) + if (isUlaw) { - if (isUlaw) - { - in[f] = ulaw_decode[(quint8)data.datain[f]]; - } - else - { - // Convert 8-bit sample to 16-bit - in[f] = (qint16)(((quint8)data.datain[f] << 8) - 32640); - } + out[f] = ulaw_decode[(quint8)data.data[f]]; + } + else + { + // Convert 8-bit sample to 16-bit + out[f] = (qint16)(((quint8)data.data[f] << 8) - 32640); } - data.datain = inPacket; // Replace incoming data with converted. } + data.data.clear(); + data.data = outPacket; // Replace incoming data with converted. + } - //qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.datain.length(); + //qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.data.length(); - /* We now have an array of 16bit samples in the NATIVE samplerate of the radio - If the radio sample rate is below 48000, we need to resample. - */ + /* We now have an array of 16bit samples in the NATIVE samplerate of the radio + If the radio sample rate is below 48000, we need to resample. + */ - if (ratioDen != 1) { + if (ratioDen != 1) { - // We need to resample - quint32 outFrames = ((data.datain.length() / 2) * ratioDen) / radioChannels; - quint32 inFrames = (data.datain.length() / 2) / radioChannels; - data.dataout.resize(outFrames * 2 * radioChannels); // Preset the output buffer size. + // We need to resample + quint32 outFrames = ((data.data.length() / 2) * ratioDen) / radioChannels; + quint32 inFrames = (data.data.length() / 2) / radioChannels; + QByteArray outPacket(outFrames * 2 * radioChannels, 0xff); // Preset the output buffer size. - int err = 0; - if (this->radioChannels == 1) { - err = wf_resampler_process_int(resampler, 0, (const qint16*)data.datain.constData(), &inFrames, (qint16*)data.dataout.data(), &outFrames); - } - else { - err = wf_resampler_process_interleaved_int(resampler, (const qint16*)data.datain.constData(), &inFrames, (qint16*)data.dataout.data(), &outFrames); - } - if (err) { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; - } + int err = 0; + if (this->radioChannels == 1) { + err = wf_resampler_process_int(resampler, 0, (const qint16*)data.data.constData(), &inFrames, (qint16*)outPacket.data(), &outFrames); } else { - data.dataout = data.datain; + err = wf_resampler_process_interleaved_int(resampler, (const qint16*)data.data.constData(), &inFrames, (qint16*)outPacket.data(), &outFrames); } + if (err) { + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; + } + data.data.clear(); + data.data = outPacket; // Replace incoming data with converted. + } - mutex.lock(); - audioBuffer.insert( data.seq, data ); - mutex.unlock(); + if (radioChannels == 1) + { + // Convert to stereo + QByteArray outPacket(data.data.length()*2, 0xff); // Preset the output buffer size. + qint16* in = (qint16*)data.data.data(); + qint16* out = (qint16*)outPacket.data(); + for (int f = 0; f < data.data.length()/2; f++) + { + *out++ = *in; + *out++ = *in++; + } + data.data.clear(); + data.data = outPacket; // Replace incoming data with converted. + } + + if (!ringBuf->try_write(data)) + { + qDebug(logAudio()) << "Buffer full! capacity:" << ringBuf->capacity() << "length" << ringBuf->size(); + } + return currentLatency; } void audioHandler::changeLatency(const quint16 newSize) @@ -367,7 +393,7 @@ void audioHandler::getNextAudioChunk(QByteArray& ret) packet = audioBuffer.erase(packet); // returns next packet } else { - if (packet->datain.length() == chunkSize && ret.length() == 0) + if (packet->data.length() == chunkSize && ret.length() == 0) { // We now have an array of samples in the computer native format (48000) // If the radio sample rate is below 48000, we need to resample. @@ -376,11 +402,11 @@ void audioHandler::getNextAudioChunk(QByteArray& ret) if (ratioNum != 1) { // We need to resample (we are STILL 16 bit!) - quint32 outFrames = ((packet->datain.length() / 2) / ratioNum) / radioChannels; - quint32 inFrames = (packet->datain.length() / 2) / radioChannels; + quint32 outFrames = ((packet->data.length() / 2) / ratioNum) / radioChannels; + quint32 inFrames = (packet->data.length() / 2) / radioChannels; packet->dataout.resize(outFrames * 2 * radioChannels); // Preset the output buffer size. - const qint16* in = (qint16*)packet->datain.constData(); + const qint16* in = (qint16*)packet->data.constData(); qint16* out = (qint16*)packet->dataout.data(); int err = 0; if (this->radioChannels == 1) { @@ -393,22 +419,22 @@ void audioHandler::getNextAudioChunk(QByteArray& ret) qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; } //qInfo(logAudio()) << "Resampler run " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; - //qInfo(logAudio()) << "Resampler run inLen:" << packet->datain.length() << " outLen:" << packet->dataout.length(); + //qInfo(logAudio()) << "Resampler run inLen:" << packet->data.length() << " outLen:" << packet->dataout.length(); if (radioSampleBits == 8) { - packet->datain = packet->dataout; // Copy output packet back to input buffer. + packet->data = packet->dataout; // Copy output packet back to input buffer. packet->dataout.clear(); // Buffer MUST be cleared ready to be re-filled by the upsampling below. } } else if (radioSampleBits == 16) { // Only copy buffer if radioSampleBits is 16, as it will be handled below otherwise. - packet->dataout = packet->datain; + packet->dataout = packet->data; } // Do we need to convert 16-bit to 8-bit? if (radioSampleBits == 8) { - packet->dataout.resize(packet->datain.length() / 2); - qint16* in = (qint16*)packet->datain.data(); + packet->dataout.resize(packet->data.length() / 2); + qint16* in = (qint16*)packet->data.data(); for (int f = 0; f < packet->dataout.length(); f++) { quint8 outdata = 0; diff --git a/audiohandler.h b/audiohandler.h index 539c991b..dda7c7c3 100644 --- a/audiohandler.h +++ b/audiohandler.h @@ -18,6 +18,7 @@ typedef signed short MY_TYPE; #include #include #include "resampler/speex_resampler.h" +#include "ring/ring.h" #include @@ -33,8 +34,7 @@ struct audioPacket { quint32 seq; QTime time; quint16 sent; - QByteArray datain; - QByteArray dataout; + QByteArray data; }; class audioHandler : public QObject @@ -57,10 +57,10 @@ public: bool isSequential() const; void getNextAudioChunk(QByteArray &data); bool isChunkAvailable(); + int incomingAudio(const audioPacket data); private slots: bool init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isulaw, const bool isinput, int port, quint8 resampleQuality); - void incomingAudio(const audioPacket data); void changeLatency(const quint16 newSize); void notified(); void stateChanged(QAudio::State state); @@ -111,8 +111,12 @@ private: unsigned int ratioNum; unsigned int ratioDen; - QMutex mutex; - volatile bool lock=false; + + wilt::Ring *ringBuf=Q_NULLPTR; + volatile bool ready = false; + audioPacket tempBuf; + quint16 currentLatency; + }; #endif // AUDIOHANDLER_H diff --git a/ring/LICENSE b/ring/LICENSE new file mode 100644 index 00000000..b9a4d39d --- /dev/null +++ b/ring/LICENSE @@ -0,0 +1,21 @@ +MIT License + +Copyright (c) 2018 Trevor Wilson + +Permission is hereby granted, free of charge, to any person obtaining a copy +of this software and associated documentation files (the "Software"), to deal +in the Software without restriction, including without limitation the rights +to use, copy, modify, merge, publish, distribute, sublicense, and/or sell +copies of the Software, and to permit persons to whom the Software is +furnished to do so, subject to the following conditions: + +The above copyright notice and this permission notice shall be included in all +copies or substantial portions of the Software. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. diff --git a/ring/README.md b/ring/README.md new file mode 100644 index 00000000..2cdbbe9a --- /dev/null +++ b/ring/README.md @@ -0,0 +1,10 @@ +# Ring Library + +## Overview + +This library provides source for a multi-producer multi-consumer lock-free ring buffer. It provides a very simple interface for writing and reading from the buffer. The source includes a `Ring_` class, that provides the raw implementation and C-like facilities, as well as a templated `Ring` class for typed reads and writes. + + +## Contact + +If you have any questions, concerns, or recommendations please feel free to e-mail me at kmdreko@gmail.com. If you notice a bug or defect, create an issue to report it. diff --git a/ring/ring.cpp b/ring/ring.cpp new file mode 100644 index 00000000..d50555df --- /dev/null +++ b/ring/ring.cpp @@ -0,0 +1,290 @@ +//////////////////////////////////////////////////////////////////////////////// +// FILE: ring.cpp +// DATE: 2016-02-25 +// AUTH: Trevor Wilson +// DESC: Implements a lock-free, multi-consumer, multi-producer ring buffer +// class + +//////////////////////////////////////////////////////////////////////////////// +// Copyright (c) 2016 Trevor Wilson +// +// Permission is hereby granted, free of charge, to any person obtaining a copy +// of this software and associated documentation files(the "Software"), to deal +// in the Software without restriction, including without limitation the rights +// to use, copy, modify, merge, publish, distribute, sublicense, and / or sell +// copies of the Software, and to permit persons to whom the Software is +// furnished to do so, subject to the following conditions : +// +// The above copyright notice and this permission notice shall be included in +// all copies or substantial portions of the Software. +// +// THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +// IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +// FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.IN NO EVENT SHALL THE +// AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +// LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +// OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +// SOFTWARE. + +#include "ring.h" +using namespace wilt; + +#include +// - std::memcpy + +Ring_::Ring_() + : beg_(nullptr) + , end_(nullptr) +{ + std::atomic_init(&used_, static_cast(0)); + std::atomic_init(&free_, static_cast(0)); + std::atomic_init(&rbuf_, static_cast(0)); + std::atomic_init(&rptr_, static_cast(0)); + std::atomic_init(&wptr_, static_cast(0)); + std::atomic_init(&wbuf_, static_cast(0)); +} + +Ring_::Ring_(std::size_t size) + : beg_(new char[size]) + , end_(beg_ + size) +{ + std::atomic_init(&used_, static_cast(0)); + std::atomic_init(&free_, static_cast(size)); + std::atomic_init(&rbuf_, beg_); + std::atomic_init(&rptr_, beg_); + std::atomic_init(&wptr_, beg_); + std::atomic_init(&wbuf_, beg_); +} + +Ring_::Ring_(Ring_&& ring) + : beg_(ring.beg_) + , end_(ring.end_) +{ + std::atomic_init(&used_, ring.used_.load()); + std::atomic_init(&free_, ring.free_.load()); + std::atomic_init(&rbuf_, ring.rbuf_.load()); + std::atomic_init(&rptr_, ring.rptr_.load()); + std::atomic_init(&wptr_, ring.wptr_.load()); + std::atomic_init(&wbuf_, ring.wbuf_.load()); + + ring.beg_ = nullptr; + ring.end_ = nullptr; + + ring.used_.store(0); + ring.free_.store(0); + ring.rbuf_.store(nullptr); + ring.rptr_.store(nullptr); + ring.wptr_.store(nullptr); + ring.wbuf_.store(nullptr); +} + +Ring_& Ring_::operator= (Ring_&& ring) +{ + delete[] beg_; + + beg_ = ring.beg_; + end_ = ring.end_; + + used_.store(ring.used_.load()); + free_.store(ring.free_.load()); + rbuf_.store(ring.rbuf_.load()); + rptr_.store(ring.rptr_.load()); + wptr_.store(ring.wptr_.load()); + wbuf_.store(ring.wbuf_.load()); + + ring.beg_ = nullptr; + ring.end_ = nullptr; + + ring.used_.store(0); + ring.free_.store(0); + ring.rbuf_.store(nullptr); + ring.rptr_.store(nullptr); + ring.wptr_.store(nullptr); + ring.wbuf_.store(nullptr); + + return *this; +} + +Ring_::~Ring_() +{ + delete[] beg_; +} + +std::size_t Ring_::size() const +{ + // The 'used' space can be negative in an over-reserved case, but it can be + // clamped to 0 for simplicity. + + auto s = used_.load(); + return s < 0 ? 0 : static_cast(s); +} + +std::size_t Ring_::capacity() const +{ + return static_cast(end_ - beg_); +} + +void Ring_::read(void* data, std::size_t length) noexcept +{ + auto block = acquire_read_block_(length); + + copy_read_block_(block, (char*)data, length); + release_read_block_(block, length); +} + +void Ring_::write(const void* data, std::size_t length) noexcept +{ + auto block = acquire_write_block_(length); + + copy_write_block_(block, (const char*)data, length); + release_write_block_(block, length); +} + +bool Ring_::try_read(void* data, std::size_t length) noexcept +{ + auto block = try_acquire_read_block_(length); + if (block == nullptr) + return false; + + copy_read_block_(block, (char*)data, length); + release_read_block_(block, length); + + return true; +} + +bool Ring_::try_write(const void* data, std::size_t length) noexcept +{ + auto block = try_acquire_write_block_(length); + if (block == nullptr) + return false; + + copy_write_block_(block, (const char*)data, length); + release_write_block_(block, length); + + return true; +} + +char* Ring_::normalize_(char* ptr) +{ + return ptr < end_ ? ptr : ptr - capacity(); +} + +char* Ring_::acquire_read_block_(std::size_t length) +{ + auto size = static_cast(length); + while (true) // loop while conflict + { + auto old_rptr = rptr_.load(std::memory_order_consume); // read rptr + while (used_.load(std::memory_order_consume) < size) // check for data + ; // spin until success + + auto new_rptr = normalize_(old_rptr + size); // get block end + used_.fetch_sub(size); // reserve + if (rptr_.compare_exchange_strong(old_rptr, new_rptr)) // try commit + return old_rptr; // committed + + used_.fetch_add(size, std::memory_order_relaxed); // un-reserve + } +} + +char* Ring_::try_acquire_read_block_(std::size_t length) +{ + auto size = static_cast(length); + while (true) // loop while conflict + { + auto old_rptr = rptr_.load(std::memory_order_consume); // read rptr + if (used_.load(std::memory_order_consume) < size) // check for data + return nullptr; // return failure + + auto new_rptr = normalize_(old_rptr + size); // get block end + used_.fetch_sub(size); // reserve + if (rptr_.compare_exchange_strong(old_rptr, new_rptr)) // try commit + return old_rptr; // committed + + used_.fetch_add(size, std::memory_order_relaxed); // un-reserve + } +} + +void Ring_::copy_read_block_(const char* block, char* data, std::size_t length) +{ + if (block + length < end_) + { + std::memcpy(data, block, length); + } + else + { + auto first = end_ - block; + std::memcpy(data, block, first); + std::memcpy(data + first, beg_, length - first); + } +} + +void Ring_::release_read_block_(char* old_rptr, std::size_t length) +{ + auto new_rptr = normalize_(old_rptr + length); // get block end + while (rbuf_.load() != old_rptr) // check for earlier reads + ; // spin until reads complete + + rbuf_.store(new_rptr); // finish commit + free_.fetch_add(length, std::memory_order_relaxed); // add to free space +} + +char* Ring_::acquire_write_block_(std::size_t length) +{ + auto size = static_cast(length); + while (true) // loop while conflict + { + auto old_wbuf = wbuf_.load(std::memory_order_consume); // read wbuf + while (free_.load(std::memory_order_consume) < size) // check for space + ; // spin until success + + auto new_wbuf = normalize_(old_wbuf + size); // get block end + free_.fetch_sub(size); // reserve + if (wbuf_.compare_exchange_strong(old_wbuf, new_wbuf)) // try commit + return old_wbuf; // committed + + free_.fetch_add(size, std::memory_order_relaxed); // un-reserve + } +} + +char* Ring_::try_acquire_write_block_(std::size_t length) +{ + auto size = static_cast(length); + while (true) // loop while conflict + { + auto old_wbuf = wbuf_.load(std::memory_order_consume); // read wbuf + if (free_.load(std::memory_order_consume) < size) // check for space + return nullptr; // return failure + + auto new_wbuf = normalize_(old_wbuf + size); // get block end + free_.fetch_sub(size); // reserve + if (wbuf_.compare_exchange_strong(old_wbuf, new_wbuf)) // try commit + return old_wbuf; // committed + + free_.fetch_add(size, std::memory_order_relaxed); // un-reserve + } +} + +void Ring_::copy_write_block_(char* block, const char* data, std::size_t length) +{ + if (block + length < end_) + { + std::memcpy(block, data, length); + } + else + { + auto first = end_ - block; + std::memcpy(block, data, first); + std::memcpy(beg_, data + first, length - first); + } +} + +void Ring_::release_write_block_(char* old_wbuf, std::size_t length) +{ + auto new_wbuf = normalize_(old_wbuf + length); // get block end + while (wptr_.load() != old_wbuf) // wait for earlier writes + ; // spin until writes complete + + wptr_.store(new_wbuf); // finish commit + used_.fetch_add(length, std::memory_order_relaxed); // add to used space +} diff --git a/ring/ring.h b/ring/ring.h new file mode 100644 index 00000000..cec58526 --- /dev/null +++ b/ring/ring.h @@ -0,0 +1,440 @@ +//////////////////////////////////////////////////////////////////////////////// +// FILE: ring.h +// DATE: 2016-02-25 +// AUTH: Trevor Wilson +// DESC: Defines a lock-free, multi-consumer, multi-producer ring buffer class + +//////////////////////////////////////////////////////////////////////////////// +// Copyright (c) 2016 Trevor Wilson +// +// Permission is hereby granted, free of charge, to any person obtaining a copy +// of this software and associated documentation files(the "Software"), to deal +// in the Software without restriction, including without limitation the rights +// to use, copy, modify, merge, publish, distribute, sublicense, and / or sell +// copies of the Software, and to permit persons to whom the Software is +// furnished to do so, subject to the following conditions : +// +// The above copyright notice and this permission notice shall be included in +// all copies or substantial portions of the Software. +// +// THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +// IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +// FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.IN NO EVENT SHALL THE +// AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +// LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +// OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +// SOFTWARE. + +#ifndef WILT_RING_H +#define WILT_RING_H + +#include +// - std::atomic +#include +// - std::size_t +// - std::ptrdiff_t +#include +// - ::new(ptr) +#include +// - std::is_nothrow_copy_constructible +// - std::is_nothrow_move_constructible +// - std::is_nothrow_move_assignable +// - std::is_nothrow_destructible +#include +// - std::move + +namespace wilt +{ + ////////////////////////////////////////////////////////////////////////////// + // This structure aims to access elements in a ring buffer from multiple + // concurrent readers and writers in a lock-free manner. + // + // The class works by allocating the array and storing two pointers (for the + // beginning and end of the allocated space). Two atomic pointers are used to + // track the beginning and end of the currently used storage space. To + // facilitate concurrent reads and writes, theres a read buffer pointer before + // the read pointer for data currently being read, and a corresponding write + // buffer pointer beyond the write pointer for data currently being written. + // These buffer pointers cannot overlap. Just using these pointers suffer from + // some minute inefficiencies and a few ABA problems. Therfore, atomic + // integers are used to store the currently used and currently free sizes. + // + // It allows multiple readers and multiple writers by implementing a reserve- + // commit system. A thread wanting to read will check the used size to see if + // there's enough data. If there is, it subtracts from the used size to + // 'reserve' the read. It then does a compare-exchange to 'commit' by + // increasing the read pointer. If that fails, then it backs out ('un- + // reserves') by adding back to the used size and tries again. If it + // succeeds, then it proceeds to read the data. In order to complete, the + // reader must update the read buffer pointer to where it just finished + // reading from. However, because other readers that started before may not be + // done yet, the reader must wait until the read buffer pointer points to + // where the read started. Only, then is the read buffer pointer updated, and + // the free size increased. So while this implementation is lock-free, it is + // not wait-free. This same principle works the same when writing (ammended + // for the appropriate pointers). + // + // If two readers try to read at the same time and there is only enough data + // for one of them. The used size MAY be negative because they both 'reserve' + // the data. This is an over-reserved state. But the compare-exchange will + // only allow one reader to 'commit' to the read and the other will 'un- + // reserve' the read. + // + // |beg |rptr used=5 |wbuf - unused + // |----|----|++++|====|====|====|====|====|++++|----| + modifying + // free=3 |rbuf |wptr |end = used + // + // The diagram above shows a buffer of size 10 storing 5 bytes with a reader + // reading one byte and one writer reading one byte. + // + // Out of the box, the class works by reading and writing raw bytes from POD + // data types and arrays. A wrapper could allow for a nicer interface for + // pushing and popping elements. As it stands, this structure cannot be easily + // modified to store types of variable size. + + class Ring_ + { + private: + //////////////////////////////////////////////////////////////////////////// + // TYPE DEFINITIONS + //////////////////////////////////////////////////////////////////////////// + + typedef char* data_ptr; + typedef std::atomic size_type; + typedef std::atomic atom_ptr; + + private: + //////////////////////////////////////////////////////////////////////////// + // PRIVATE MEMBERS + //////////////////////////////////////////////////////////////////////////// + // Beginning and end pointers don't need to be atomic because they don't + // change. used_ and free_ can be negative in certain cases (and that's ok). + + data_ptr beg_; // pointer to beginning of data block + data_ptr end_; // pointer to end of data block + + alignas(64) + size_type used_; // size of unreserved used space + + alignas(64) + size_type free_; // size of unreserved free space + + alignas(64) + atom_ptr rbuf_; // pointer to beginning of data being read + atom_ptr rptr_; // pointer to beginning of data + + alignas(64) + atom_ptr wptr_; // pointer to end of data + atom_ptr wbuf_; // pointer to end of data being written + + public: + //////////////////////////////////////////////////////////////////////////// + // CONSTRUCTORS AND DESTRUCTORS + //////////////////////////////////////////////////////////////////////////// + + // Constructs a ring without a buffer (capacity() == 0) + Ring_(); + + // Constructs a ring with a buffer with a size + Ring_(std::size_t size); + + // Moves the buffer between rings, assumes no concurrent operations + Ring_(Ring_&& ring); + + // Moves the buffer between rings, assumes no concurrent operations on + // either ring. Deallocates the buffer + Ring_& operator= (Ring_&& ring); + + // No copying + Ring_(const Ring_&) = delete; + Ring_& operator= (const Ring_&) = delete; + + // Deallocates the buffer + ~Ring_(); + + public: + //////////////////////////////////////////////////////////////////////////// + // QUERY FUNCTIONS + //////////////////////////////////////////////////////////////////////////// + // Functions only report on the state of the ring + + // Returns the current amount of non-reserved used space (amount of written + // data that a read hasn't yet reserved). Over-reserved scenarios mean this + // number is not the ultimate source of truth with concurrent operations, + // but its the closest safe approximation. This, of course, doesn't report + // writes that have not completed. + std::size_t size() const; + + // Maximum amount of data that can be held + std::size_t capacity() const; + + public: + //////////////////////////////////////////////////////////////////////////// + // ACCESSORS AND MODIFIERS + //////////////////////////////////////////////////////////////////////////// + // All operations assume object has not been moved. Blocking operations run + // until operation is completed. Non-blocking operations fail if there is + // not enough space + + void read(void* data, std::size_t length) noexcept; + void write(const void* data, std::size_t length) noexcept; + bool try_read(void* data, std::size_t length) noexcept; + bool try_write(const void* data, std::size_t length) noexcept; + + protected: + //////////////////////////////////////////////////////////////////////////// + // PROTECTED FUNCTIONS + //////////////////////////////////////////////////////////////////////////// + // Helper functions + + // Wraps a pointer within the array. Assumes 'beg_ <= ptr < end_+capacity()' + char* normalize_(char*); + + char* acquire_read_block_(std::size_t length); + char* try_acquire_read_block_(std::size_t length); + void copy_read_block_(const char* block, char* data, std::size_t length); + void release_read_block_(char* block, std::size_t length); + + char* acquire_write_block_(std::size_t length); + char* try_acquire_write_block_(std::size_t length); + void copy_write_block_(char* block, const char* data, std::size_t length); + void release_write_block_(char* block, std::size_t length); + + char* begin_alloc_() { return beg_; } + const char* begin_alloc_() const { return beg_; } + char* end_alloc_() { return end_; } + const char* end_alloc_() const { return end_; } + char* begin_data_() { return rptr_; } + const char* begin_data_() const { return rptr_; } + char* end_data_() { return wptr_; } + const char* end_data_() const { return wptr_; } + + }; // class Ring_ + + template + class Ring : protected Ring_ + { + public: + //////////////////////////////////////////////////////////////////////////// + // CONSTRUCTORS AND DESTRUCTORS + //////////////////////////////////////////////////////////////////////////// + + // Constructs a ring without a buffer (capacity() == 0) + Ring(); + + // Constructs a ring with a buffer with a size + Ring(std::size_t size); + + // Moves the buffer between rings, assumes no concurrent operations + Ring(Ring&& ring); + + // Moves the buffer between rings, assumes no concurrent operations on + // either ring. Deallocates the buffer + Ring& operator= (Ring&& ring); + + // No copying + Ring(const Ring_&) = delete; + Ring& operator= (const Ring_&) = delete; + + // Deallocates the buffer, destructs stored data. + ~Ring(); + + public: + //////////////////////////////////////////////////////////////////////////// + // QUERY FUNCTIONS + //////////////////////////////////////////////////////////////////////////// + // Functions only report on the state of the ring + + // Returns the current amount of non-reserved used space (amount of written + // data that a read hasn't yet reserved). Over-reserved scenarios mean this + // number is not the ultimate source of truth with concurrent operations, + // but its the closest safe approximation. This, of course, doesn't report + // writes that have not completed. + std::size_t size() const; + + // Maximum amount of data that can be held + std::size_t capacity() const; + + public: + //////////////////////////////////////////////////////////////////////////// + // ACCESSORS AND MODIFIERS + //////////////////////////////////////////////////////////////////////////// + // All operations assume object has not been moved. Blocking operations run + // until operation is completed. Non-blocking operations fail if there is + // not enough space + + void read(T& data) noexcept; // blocking read + void write(const T& data) noexcept; // blocking write + void write(T&& data) noexcept; // blocking write + bool try_read(T& data) noexcept; // non-blocking read + bool try_write(const T& data) noexcept; // non-blocking write + bool try_write(T&& data) noexcept; // non-blocking write + + private: + //////////////////////////////////////////////////////////////////////////// + // PRIVATE HELPER FUNCTIONS + //////////////////////////////////////////////////////////////////////////// + + void destruct_(); + + }; // class Ring + + template + Ring::Ring() + : Ring_() + { } + + template + Ring::Ring(std::size_t size) + : Ring_(size * sizeof(T)) + { } + + template + Ring::Ring(Ring&& ring) + : Ring_(std::move(ring)) + { } + + template + Ring& Ring::operator= (Ring&& ring) + { + destruct_(); + + Ring_::operator= (ring); + + return *this; + } + + template + Ring::~Ring() + { + destruct_(); + } + + template + void Ring::destruct_() + { + if (size() == 0) + return; + + auto itr = begin_data_(); + auto end = end_data_(); + do + { + auto t = reinterpret_cast(itr); + t->~T(); + + itr = normalize_(itr + sizeof(T)); + } while (itr != end); + } + + template + std::size_t Ring::size() const + { + return Ring_::size() / sizeof(T); + } + + template + std::size_t Ring::capacity() const + { + return Ring_::capacity() / sizeof(T); + } + + template + void Ring::read(T& data) noexcept + { + static_assert(std::is_nothrow_move_assignable::value, "T move assignment must not throw"); + static_assert(std::is_nothrow_destructible::value, "T destructor must not throw"); + + auto block = acquire_read_block_(sizeof(T)); + + // critical section + auto t = reinterpret_cast(block); + data = std::move(*t); + t->~T(); + + release_read_block_(block, sizeof(T)); + } + + template + void Ring::write(const T& data) noexcept + { + static_assert(std::is_nothrow_copy_constructible::value, "T copy constructor must not throw"); + + auto block = acquire_write_block_(sizeof(T)); + + // critical section + new(block) T(data); + + release_write_block_(block, sizeof(T)); + } + + template + void Ring::write(T&& data) noexcept + { + static_assert(std::is_nothrow_move_constructible::value, "T move constructor must not throw"); + + auto block = acquire_write_block_(sizeof(T)); + + // critical section + new(block) T(std::move(data)); + + release_write_block_(block, sizeof(T)); + } + + template + bool Ring::try_read(T& data) noexcept + { + static_assert(std::is_nothrow_move_assignable::value, "T move assignment must not throw"); + static_assert(std::is_nothrow_destructible::value, "T destructor must not throw"); + + auto block = try_acquire_read_block_(sizeof(T)); + if (block == nullptr) + return false; + + // critical section + auto t = reinterpret_cast(block); + data = std::move(*t); + t->~T(); + + release_read_block_(block, sizeof(T)); + + return true; + } + + template + bool Ring::try_write(const T& data) noexcept + { + static_assert(std::is_nothrow_copy_constructible::value, "T copy constructor must not throw"); + + auto block = try_acquire_write_block_(sizeof(T)); + if (block == nullptr) + return false; + + // critical section + new(block) T(data); + + release_write_block_(block, sizeof(T)); + + return true; + } + + template + bool Ring::try_write(T&& data) noexcept + { + static_assert(std::is_nothrow_move_constructible::value, "T move constructor must not throw"); + + auto block = try_acquire_write_block_(sizeof(T)); + if (block == nullptr) + return false; + + // critical section + new(block) T(std::move(data)); + + release_write_block_(block, sizeof(T)); + + return true; + } + +} // namespace wilt + +#endif // !WILT_RING_H \ No newline at end of file diff --git a/udphandler.cpp b/udphandler.cpp index 62202253..22b248b5 100644 --- a/udphandler.cpp +++ b/udphandler.cpp @@ -189,7 +189,16 @@ void udpHandler::dataReceived() totallost = totallost + civ->packetsLost; } - emit haveNetworkStatus(" rtt: " + QString::number(latency) + " ms, loss: (" + QString::number(totallost) + "/" + QString::number(totalsent) + ")"); + QString tempLatency; + if (rxLatency > audio->audioLatency) + { + tempLatency = QString::number(audio->audioLatency) + "ms"; + } + else { + tempLatency = "" + QString::number(audio->audioLatency) + "ms"; + } + + emit haveNetworkStatus("rx latency: " + tempLatency + " ms / rtt: " + QString::number(latency) + " ms, loss: (" + QString::number(totallost) + "/" + QString::number(totalsent) + ")"); } break; } @@ -936,11 +945,11 @@ void udpAudio::dataReceived() tempAudio.seq = (quint32)seqPrefix << 16 | in->seq; tempAudio.time = lastReceived; tempAudio.sent = 0; - tempAudio.datain = r.mid(0x18); + tempAudio.data = r.mid(0x18); // Prefer signal/slot to forward audio as it is thread/safe // Need to do more testing but latency appears fine. - emit haveAudioData(tempAudio); - //rxaudio->incomingAudio(tempAudio); + //emit haveAudioData(tempAudio); + audioLatency=rxaudio->incomingAudio(tempAudio); } break; } diff --git a/udphandler.h b/udphandler.h index 59e7328a..a835d957 100644 --- a/udphandler.h +++ b/udphandler.h @@ -178,6 +178,8 @@ public: udpAudio(QHostAddress local, QHostAddress ip, quint16 aport, quint16 rxlatency, quint16 txlatency, quint16 rxsample, quint8 rxcodec, quint16 txsample, quint8 txcodec, int outputPort, int inputPort, quint8 resampleQuality); ~udpAudio(); + int audioLatency = 0; + signals: void haveAudioData(audioPacket data); diff --git a/udpserver.cpp b/udpserver.cpp index d392c7e3..23a1dff0 100644 --- a/udpserver.cpp +++ b/udpserver.cpp @@ -695,7 +695,7 @@ void udpServer::audioReceived() tempAudio.seq = (quint32)current->seqPrefix << 16 | in->seq; tempAudio.time = QTime::currentTime();; tempAudio.sent = 0; - tempAudio.datain = r.mid(0x18); + tempAudio.data = r.mid(0x18); //qInfo(logUdpServer()) << "sending tx audio " << in->seq; emit haveAudioData(tempAudio); } @@ -1305,9 +1305,9 @@ void udpServer::sendRxAudio() while (len < audio.length()) { audioPacket partial; - partial.datain = audio.mid(len, 1364); + partial.data = audio.mid(len, 1364); receiveAudioData(partial); - len = len + partial.datain.length(); + len = len + partial.data.length(); } } } @@ -1322,15 +1322,15 @@ void udpServer::receiveAudioData(const audioPacket &d) if (client != Q_NULLPTR && client->connected) { audio_packet p; memset(p.packet, 0x0, sizeof(p)); // We can't be sure it is initialized with 0x00! - p.len = sizeof(p) + d.datain.length(); + p.len = sizeof(p) + d.data.length(); p.sentid = client->myId; p.rcvdid = client->remoteId; p.ident = 0x0080; // audio is always this? - p.datalen = (quint16)qToBigEndian((quint16)d.datain.length()); + p.datalen = (quint16)qToBigEndian((quint16)d.data.length()); p.sendseq = (quint16)qToBigEndian((quint16)client->sendAudioSeq); // THIS IS BIG ENDIAN! p.seq = client->txSeq; QByteArray t = QByteArray::fromRawData((const char*)p.packet, sizeof(p)); - t.append(d.datain); + t.append(d.data); client->txMutex.lock(); client->txSeqBuf.append(SEQBUFENTRY()); client->txSeqBuf.last().seqNum = p.seq; diff --git a/wfview.vcxproj b/wfview.vcxproj index 1dc80f4d..d977fb5d 100644 --- a/wfview.vcxproj +++ b/wfview.vcxproj @@ -199,6 +199,7 @@ + @@ -222,6 +223,7 @@ + diff --git a/wfview.vcxproj.filters b/wfview.vcxproj.filters index e09d8dbb..06a9b64d 100644 --- a/wfview.vcxproj.filters +++ b/wfview.vcxproj.filters @@ -120,6 +120,9 @@ Source Files + + Source Files + @@ -197,6 +200,9 @@ Header Files + + Header Files + @@ -354,6 +360,7 @@ Resource Files + From 812e86d8ddd2ae79bb4365c237651180e6cc978a Mon Sep 17 00:00:00 2001 From: M0VSE Date: Thu, 27 May 2021 11:45:58 +0100 Subject: [PATCH 12/47] fix qmake file --- wfview.pro | 40 +++++++++++++++++++++------------------- 1 file changed, 21 insertions(+), 19 deletions(-) diff --git a/wfview.pro b/wfview.pro index 644a8d64..30c8503d 100644 --- a/wfview.pro +++ b/wfview.pro @@ -106,13 +106,13 @@ INCLUDEPATH += opus-tools/src INCLUDEPATH += rtaudio SOURCES += main.cpp\ - wfmain.cpp \ + wfmain.cpp \ commhandler.cpp \ rigcommander.cpp \ freqmemory.cpp \ rigidentities.cpp \ - udphandler.cpp \ - logcategories.cpp \ + udphandler.cpp \ + logcategories.cpp \ audiohandler.cpp \ calibrationwindow.cpp \ satellitesetup.cpp \ @@ -120,36 +120,38 @@ SOURCES += main.cpp\ udpserver.cpp \ meter.cpp \ qledlabel.cpp \ - pttyhandler.cpp \ - opus-tools/src/resample.c \ + pttyhandler.cpp \ + opus-tools/src/resample.c \ repeatersetup.cpp \ - rigctld.cpp \ - rtaudio/RtAudio.cpp + rigctld.cpp \ + rtaudio/RtAudio.cpp + ring/ring.cpp HEADERS += wfmain.h \ commhandler.h \ rigcommander.h \ freqmemory.h \ rigidentities.h \ - udphandler.h \ - logcategories.h \ + udphandler.h \ + logcategories.h \ audiohandler.h \ calibrationwindow.h \ satellitesetup.h \ udpserversetup.h \ - udpserver.h \ - packettypes.h \ + udpserver.h \ + packettypes.h \ meter.h \ - qledlabel.h \ - pttyhandler.h \ - opus-tools/src/speex_resampler.h \ - opus-tools/src/arch.h \ - opus-tools/src/resample_sse.h \ + qledlabel.h \ + pttyhandler.h \ + opus-tools/src/speex_resampler.h \ + opus-tools/src/arch.h \ + opus-tools/src/resample_sse.h \ repeatersetup.h \ repeaterattributes.h \ - rigctld.h \ - rtaudio/RtAudio.h \ - ulaw.h + rigctld.h \ + rtaudio/RtAudio.h \ + ulaw.h \ + ring/ring.h FORMS += wfmain.ui \ From abb207ffc405eb7521fe397790de295d8dbada44 Mon Sep 17 00:00:00 2001 From: M0VSE Date: Thu, 27 May 2021 11:48:51 +0100 Subject: [PATCH 13/47] Fix qmake (again) --- wfview.pro | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/wfview.pro b/wfview.pro index 30c8503d..2f2d03b3 100644 --- a/wfview.pro +++ b/wfview.pro @@ -124,7 +124,7 @@ SOURCES += main.cpp\ opus-tools/src/resample.c \ repeatersetup.cpp \ rigctld.cpp \ - rtaudio/RtAudio.cpp + rtaudio/RtAudio.cpp \ ring/ring.cpp HEADERS += wfmain.h \ From 1b5815a1c2a4b3ba9856c5f2881cba2f0d5b9186 Mon Sep 17 00:00:00 2001 From: M0VSE Date: Thu, 27 May 2021 11:51:57 +0100 Subject: [PATCH 14/47] add asound lib to linux build --- wfview.pro | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/wfview.pro b/wfview.pro index 2f2d03b3..fa03a50a 100644 --- a/wfview.pro +++ b/wfview.pro @@ -96,7 +96,7 @@ CONFIG(debug, release|debug) { linux: QCPLIB = qcustomplot } -linux:LIBS += -L./ -l$$QCPLIB +linux:LIBS += -L./ -l$$QCPLIB -lasound !linux:SOURCES += ../qcustomplot/qcustomplot.cpp !linux:HEADERS += ../qcustomplot/qcustomplot.h From 841f8aff9eac7944abb3fd7e011eef5cd97811a7 Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Thu, 27 May 2021 12:27:17 +0100 Subject: [PATCH 15/47] Fix for compiling on MacOS --- wfmain.cpp | 2 +- wfview.pro | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/wfmain.cpp b/wfmain.cpp index 89e98be4..b65fd6f4 100644 --- a/wfmain.cpp +++ b/wfmain.cpp @@ -202,7 +202,7 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s qInfo(logAudio()) << (info.isDefaultOutput ? "*" : " ") << "(" << i << ") Output Device : " << QString::fromStdString(info.name); ui->audioOutputCombo->addItem(QString::fromStdString(info.name), i); } - else if (info.inputChannels > 0) { + if (info.inputChannels > 0) { qInfo(logAudio()) << (info.isDefaultInput ? "*" : " ") << "(" << i << ") Input Device : " << QString::fromStdString(info.name); ui->audioInputCombo->addItem(QString::fromStdString(info.name), i); } diff --git a/wfview.pro b/wfview.pro index fa03a50a..124a6add 100644 --- a/wfview.pro +++ b/wfview.pro @@ -97,6 +97,7 @@ CONFIG(debug, release|debug) { } linux:LIBS += -L./ -l$$QCPLIB -lasound +macx:LIBS += -framework CoreAudio -framework CoreFoundation -lpthread !linux:SOURCES += ../qcustomplot/qcustomplot.cpp !linux:HEADERS += ../qcustomplot/qcustomplot.h From d7afcf03097a360a3c18add434e6b747969c0729 Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Thu, 27 May 2021 13:54:52 +0100 Subject: [PATCH 16/47] Add tx audio --- audiohandler.cpp | 191 +++++++++++++++++++---------------------------- audiohandler.h | 2 + udphandler.cpp | 10 ++- 3 files changed, 85 insertions(+), 118 deletions(-) diff --git a/audiohandler.cpp b/audiohandler.cpp index 39265f12..940d080e 100644 --- a/audiohandler.cpp +++ b/audiohandler.cpp @@ -60,7 +60,6 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 else { aParams.deviceId = audio.getDefaultOutputDevice(); } - aParams.nChannels = 2; // Internally this is always 2 channels aParams.firstChannel = 0; try { @@ -99,6 +98,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 if (isInput) { resampler = wf_resampler_init(radioChannels, INTERNAL_SAMPLE_RATE, samplerate, resampleQuality, &resample_error); try { + aParams.nChannels = 1; // Internally this is always 2 channels audio.openStream(NULL, &aParams, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &this->chunkSize, &staticWrite, this); audio.startStream(); } @@ -112,6 +112,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 resampler = wf_resampler_init(radioChannels, samplerate, INTERNAL_SAMPLE_RATE, resampleQuality, &resample_error); try { unsigned int length = chunkSize / 2; + aParams.nChannels = 2; // Internally this is always 2 channels audio.openStream(&aParams, NULL, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &length, &staticRead, this); audio.startStream(); } @@ -131,6 +132,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 void audioHandler::setVolume(unsigned char volume) { qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << (qreal)(volume/255.0) << ")"; + this->volume = (qreal)(volume / 255.0); } @@ -217,47 +219,29 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status) { int sentlen = 0; - /* - QMutexLocker locker(&mutex); - audioPacket* current; - - while (sentlen < len) { - if (!audioBuffer.isEmpty()) + unsigned int nBytes = nFrames * 2; // This is ALWAYS 2 bytes per sample and 1 channels + const char* data = (const char*)inputBuffer; + while (sentlen < nBytes) { + if (tempBuf.sent != chunkSize) { - if (audioBuffer.last().sent == chunkSize) + int send = qMin((int)(nBytes - sentlen), (int)chunkSize - tempBuf.sent); + tempBuf.data.append(QByteArray::fromRawData(data + sentlen, send)); + sentlen = sentlen + send; + tempBuf.seq = 0; // Not used in TX + tempBuf.time = QTime::currentTime(); + tempBuf.sent = tempBuf.data.length(); + } + else { + if (!ringBuf->try_write(tempBuf)) { - audioBuffer.append(audioPacket()); - audioBuffer.last().sent = 0; + qDebug(logAudio()) << "outgoing audio buffer full!"; } + tempBuf.data.clear(); + tempBuf.sent = 0; } - else - { - audioBuffer.append(audioPacket()); - audioBuffer.last().sent = 0; - } - current = &audioBuffer.last(); - - int send = qMin((int)(len - sentlen), (int)chunkSize - current->sent); - - current->data.append(QByteArray::fromRawData(data + sentlen, send)); - - sentlen = sentlen + send; - - current->seq = 0; // Not used in TX - current->time = QTime::currentTime(); - current->sent = current->data.length(); - - if (current->sent == chunkSize) - { - chunkAvailable = true; - } - else if (audioBuffer.length() <= 1 && current->sent != chunkSize) { - chunkAvailable = false; - } - } - */ - return (sentlen); // Always return the same number as we received + + return 0; } qint64 audioHandler::bytesAvailable() const @@ -338,21 +322,28 @@ int audioHandler::incomingAudio(audioPacket data) data.data = outPacket; // Replace incoming data with converted. } - if (radioChannels == 1) - { - // Convert to stereo + if (radioChannels == 1) { + // Convert to stereo and set volume. QByteArray outPacket(data.data.length()*2, 0xff); // Preset the output buffer size. qint16* in = (qint16*)data.data.data(); qint16* out = (qint16*)outPacket.data(); for (int f = 0; f < data.data.length()/2; f++) { - *out++ = *in; - *out++ = *in++; + *out++ = *in * volume; + *out++ = *in++ * volume; } data.data.clear(); data.data = outPacket; // Replace incoming data with converted. + } else { + // We already have two channels so just update volume. + qint16* in = (qint16*)data.data.data(); + for (int f = 0; f < data.data.length() / 2; f++) + { + in[f] = in[f] * volume; + } } + if (!ringBuf->try_write(data)) { qDebug(logAudio()) << "Buffer full! capacity:" << ringBuf->capacity() << "length" << ringBuf->size(); @@ -378,90 +369,62 @@ bool audioHandler::isChunkAvailable() void audioHandler::getNextAudioChunk(QByteArray& ret) { - /* - if (!audioBuffer.isEmpty() && chunkAvailable) + audioPacket packet; + packet.sent = 0; + if (ringBuf != Q_NULLPTR && ringBuf->try_read(packet)) { - - QMutexLocker locker(&mutex); - - // Skip through audio buffer deleting any old entry. - auto packet = audioBuffer.begin(); - while (packet != audioBuffer.end()) + if (ratioNum != 1) { - if (packet->time.msecsTo(QTime::currentTime()) > 100) { - //qInfo(logAudio()) << "TX Packet too old " << dec << packet->time.msecsTo(QTime::currentTime()) << "ms"; - packet = audioBuffer.erase(packet); // returns next packet + // We need to resample (we are STILL 16 bit!) + quint32 outFrames = ((packet.data.length() / 2) / ratioNum) / radioChannels; + quint32 inFrames = (packet.data.length() / 2) / radioChannels; + QByteArray inPacket((int)outFrames * 2 * radioChannels, (char)0xff); + + const qint16* in = (qint16*)packet.data.constData(); + qint16* out = (qint16*)inPacket.data(); + int err = 0; + if (this->radioChannels == 1) { + err = wf_resampler_process_int(resampler, 0, in, &inFrames, out, &outFrames); } else { - if (packet->data.length() == chunkSize && ret.length() == 0) - { - // We now have an array of samples in the computer native format (48000) - // If the radio sample rate is below 48000, we need to resample. - + err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames); + } + if (err) { + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; + } + //qInfo(logAudio()) << "Resampler run " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; + //qInfo(logAudio()) << "Resampler run inLen:" << packet->datain.length() << " outLen:" << packet->dataout.length(); + packet.data.clear(); + packet.data = inPacket; // Copy output packet back to input buffer. + } - if (ratioNum != 1) - { - // We need to resample (we are STILL 16 bit!) - quint32 outFrames = ((packet->data.length() / 2) / ratioNum) / radioChannels; - quint32 inFrames = (packet->data.length() / 2) / radioChannels; - packet->dataout.resize(outFrames * 2 * radioChannels); // Preset the output buffer size. - - const qint16* in = (qint16*)packet->data.constData(); - qint16* out = (qint16*)packet->dataout.data(); - int err = 0; - if (this->radioChannels == 1) { - err = wf_resampler_process_int(resampler, 0, in, &inFrames, out, &outFrames); - } - else { - err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames); - } - if (err) { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; - } - //qInfo(logAudio()) << "Resampler run " << err << " inFrames:" << inFrames << " outFrames:" << outFrames; - //qInfo(logAudio()) << "Resampler run inLen:" << packet->data.length() << " outLen:" << packet->dataout.length(); - if (radioSampleBits == 8) - { - packet->data = packet->dataout; // Copy output packet back to input buffer. - packet->dataout.clear(); // Buffer MUST be cleared ready to be re-filled by the upsampling below. - } - } - else if (radioSampleBits == 16) { - // Only copy buffer if radioSampleBits is 16, as it will be handled below otherwise. - packet->dataout = packet->data; - } - - // Do we need to convert 16-bit to 8-bit? - if (radioSampleBits == 8) { - packet->dataout.resize(packet->data.length() / 2); - qint16* in = (qint16*)packet->data.data(); - for (int f = 0; f < packet->dataout.length(); f++) - { - quint8 outdata = 0; - if (isUlaw) { - qint16 enc = qFromLittleEndian(in + f); - if (enc >= 0) - outdata = ulaw_encode[enc]; - else - outdata = 0x7f & ulaw_encode[-enc]; - } - else { - outdata = (quint8)(((qFromLittleEndian(in + f) >> 8) ^ 0x80) & 0xff); - } - packet->dataout[f] = (char)outdata; - } - } - ret = packet->dataout; - packet = audioBuffer.erase(packet); // returns next packet + // Do we need to convert 16-bit to 8-bit? + if (radioSampleBits == 8) { + QByteArray inPacket((int)packet.data.length() / 2, (char)0xff); + qint16* in = (qint16*)packet.data.data(); + for (int f = 0; f < inPacket.length(); f++) + { + quint8 outdata = 0; + if (isUlaw) { + qint16 enc = qFromLittleEndian(in + f); + if (enc >= 0) + outdata = ulaw_encode[enc]; + else + outdata = 0x7f & ulaw_encode[-enc]; } else { - packet++; + outdata = (quint8)(((qFromLittleEndian(in + f) >> 8) ^ 0x80) & 0xff); } + inPacket[f] = (char)outdata; } + packet.data.clear(); + packet.data = inPacket; // Copy output packet back to input buffer. } + ret = packet.data; } - */ + return; + } diff --git a/audiohandler.h b/audiohandler.h index dda7c7c3..26bd199b 100644 --- a/audiohandler.h +++ b/audiohandler.h @@ -6,6 +6,7 @@ #include #include #include +#include #include "rtaudio/RtAudio.h" typedef signed short MY_TYPE; @@ -116,6 +117,7 @@ private: volatile bool ready = false; audioPacket tempBuf; quint16 currentLatency; + qreal volume=1.0; }; diff --git a/udphandler.cpp b/udphandler.cpp index 22b248b5..11745245 100644 --- a/udphandler.cpp +++ b/udphandler.cpp @@ -858,10 +858,12 @@ void udpAudio::watchdog() void udpAudio::sendTxAudio() { - - if (txaudio && txaudio->isChunkAvailable()) { - QByteArray audio; - txaudio->getNextAudioChunk(audio); + if (txaudio == Q_NULLPTR) { + return; + } + QByteArray audio; + txaudio->getNextAudioChunk(audio); + if (audio.length() > 0) { int counter = 1; int len = 0; From 5ffb27ae78ca17cc6c74ea821f0c259f0b5dbf34 Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Thu, 27 May 2021 14:09:12 +0100 Subject: [PATCH 17/47] Fix for tx audio channels --- audiohandler.cpp | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/audiohandler.cpp b/audiohandler.cpp index 940d080e..e14282c0 100644 --- a/audiohandler.cpp +++ b/audiohandler.cpp @@ -219,7 +219,7 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status) { int sentlen = 0; - unsigned int nBytes = nFrames * 2; // This is ALWAYS 2 bytes per sample and 1 channels + unsigned int nBytes = nFrames * 2; // This is ALWAYS 2 bytes per sample and 1 channel const char* data = (const char*)inputBuffer; while (sentlen < nBytes) { if (tempBuf.sent != chunkSize) @@ -235,6 +235,7 @@ int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int if (!ringBuf->try_write(tempBuf)) { qDebug(logAudio()) << "outgoing audio buffer full!"; + break; } tempBuf.data.clear(); tempBuf.sent = 0; From 9d9298ff739d625d70d3137c6062ce48a0cdfc84 Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Thu, 27 May 2021 14:20:53 +0100 Subject: [PATCH 18/47] Merge from master --- CHANGELOG | 16 +++++++++ INSTALL_PREBUILT_BINARY.md | 14 ++++---- freqmemory.h | 19 +++++++++- rigcommander.cpp | 72 ++++++++++++++++++++++++++++++++++++-- rigcommander.h | 2 ++ rigidentities.cpp | 3 ++ rigidentities.h | 5 +++ udpserversetup.ui | 7 ++++ wfmain.cpp | 54 ++++++++++------------------ wfmain.ui | 36 ++++++++++++++++++- 10 files changed, 181 insertions(+), 47 deletions(-) diff --git a/CHANGELOG b/CHANGELOG index 50b48d6e..3b8d085c 100644 --- a/CHANGELOG +++ b/CHANGELOG @@ -1,7 +1,23 @@ # CHANGELOG +- 20210521 + + Add baud rate detection for remote rigs + + Correct propCIVAddr to work if less than 0xe0 + + Change audiohandler to use latency for tx audio + + - 20210520 + Added IC-756 Pro. Tested UI and back-end response with 7300 and fake + RigID reply. + + Added additional support for the IC-756 Pro III. Essentially untested. + + Cleaned up warning and UI help text. + Add more features to rigstate - 20210519 diff --git a/INSTALL_PREBUILT_BINARY.md b/INSTALL_PREBUILT_BINARY.md index 70948b5f..c8ecdff5 100644 --- a/INSTALL_PREBUILT_BINARY.md +++ b/INSTALL_PREBUILT_BINARY.md @@ -29,13 +29,13 @@ Ubuntu 20.04.2 ~~~ -### for all, the following is appicable: +### for all, the following is applicable: ~~~ download the tar.gz file here: https://wfview.org/download/test-linux-build/ the file below will unpack in ./dist -tar zxvf wfview-linux.tar.gz +tar zxvf wfview-linux.tar.gz (change the filename accordingly) cd dist sudo ./install.sh ~~~ @@ -48,34 +48,34 @@ Now for the system specifics; pick your version: ~~~ sudo apt install libqcustomplot2.0 libqt5multimedia5 libqt5serialport5 sudo ln -s /usr/lib/x86_64-linux-gnu/libqcustomplot.so.2.0.1 /usr/lib/x86_64-linux-gnu/libqcustomplot.so.2 -start wfview +wfview ~~~ ### Fedora 33/34: ~~~ sudo dnf install qcustomplot-qt5 qt5-qtmultimedia qt5-qtserialport sudo ln -s /usr/lib64/libqcustomplot-qt5.so.2 /usr/lib64/libqcustomplot.so.2 -start wfview +wfview ~~~ ### Mint 20.1 ~~~ sudo apt install libqcustomplot2.0 libqt5multimedia5 libqt5serialport5 sudo ln -s /usr/lib64/libqcustomplot-qt5.so.2 /usr/lib64/libqcustomplot.so.2 -start wfview +wfview ~~~ ### openSUSE/Tumbleweed/SLES: ~~~ sudo zypper in libqcustomplot2 libQt5SerialPort5 -start wfview +wfview ~~~ ### UBUNTU: ~~~ sudo apt install libqcustomplot2.0 libqt5multimedia5 libqt5serialport5 sudo ln -s /usr/lib/x86_64-linux-gnu/libqcustomplot.so.2.0.1 /usr/lib/x86_64-linux-gnu/libqcustomplot.so.2 -start wfview +wfview ~~~ diff --git a/freqmemory.h b/freqmemory.h index a06e5501..b803f4a8 100644 --- a/freqmemory.h +++ b/freqmemory.h @@ -20,7 +20,24 @@ enum mode_kind { modeLSB_D=0x80, modeUSB_D=0x81, modeDV=0x17, - modeDD=0x27 + modeDD=0x27, + modeWFM, + modeS_AMD, + modeS_AML, + modeS_AMU, + modeP25, + modedPMR, + modeNXDN_VN, + modeNXDN_N, + modeDCR, + modePSK, + modePSK_R +}; + +struct mode_info { + mode_kind mk; + unsigned char reg; + QString name; }; struct preset_kind { diff --git a/rigcommander.cpp b/rigcommander.cpp index c06a1205..f8b78a18 100644 --- a/rigcommander.cpp +++ b/rigcommander.cpp @@ -147,6 +147,7 @@ void rigCommander::commSetup(unsigned char rigCivAddr, udpPreferences prefs, QSt void rigCommander::closeComm() { + qDebug(logRig()) << "Closing rig comms"; if (comm != Q_NULLPTR) { delete comm; } @@ -2265,7 +2266,7 @@ void rigCommander::parseDetailedRegisters1A05() default: break; } - return; + break; case model7850: switch(subcmd) { @@ -2332,6 +2333,7 @@ void rigCommander::parseDetailedRegisters1A05() default: break; } + break; case model7610: switch(subcmd) { @@ -2407,6 +2409,7 @@ void rigCommander::parseDetailedRegisters1A05() default: break; } + break; case model705: switch(subcmd) { @@ -2525,6 +2528,16 @@ void rigCommander::parseWFData() } } + +mode_info rigCommander::createMode(mode_kind m, unsigned char reg, QString name) +{ + mode_info mode; + mode.mk = m; + mode.reg = reg; + mode.name = name; + return mode; +} + void rigCommander::determineRigCaps() { //TODO: Determine available bands (low priority, rig will reject out of band requests anyway) @@ -2539,6 +2552,14 @@ void rigCommander::determineRigCaps() standardVU = {band70cm, band2m}; + std::vector commonModes; + commonModes = { createMode(modeLSB, 0x00, "LSB"), createMode(modeUSB, 0x01, "USB"), + createMode(modeFM, 0x05, "FM"), createMode(modeAM, 0x02, "AM"), + createMode(modeCW, 0x03, "CW"), createMode(modeCW_R, 0x07, "CW-R"), + createMode(modeRTTY, 0x04, "RTTY"), createMode(modeRTTY_R, 0x08, "RTTY-R") + }; + + rigCaps.model = model; rigCaps.civ = incomingCIVAddr; @@ -2618,6 +2639,7 @@ void rigCommander::determineRigCaps() rigCaps.bands.push_back(bandGen); rigCaps.bands.push_back(band630m); rigCaps.bands.push_back(band2200m); + rigCaps.modes = commonModes; break; case modelR8600: rigCaps.modelName = QString("IC-R8600"); @@ -2643,6 +2665,13 @@ void rigCommander::determineRigCaps() rigCaps.bands = standardHF; rigCaps.bands.insert(rigCaps.bands.end(), standardVU.begin(), standardVU.end()); rigCaps.bands.insert(rigCaps.bands.end(), {band23cm, band4m, band630m, band2200m, bandGen}); + rigCaps.modes = commonModes; + rigCaps.modes.insert(rigCaps.modes.end(), { + createMode(modeWFM, 0x06, "WFM"), createMode(modeS_AMD, 0x11, "S-AM (D)"), + createMode(modeS_AML, 0x14, "S-AM(L)"), createMode(modeS_AMU, 0x15, "S-AM(U)"), + createMode(modeP25, 0x16, "P25"), createMode(modedPMR, 0x18, "dPMR"), + createMode(modeNXDN_VN, 0x19, "NXDN-VN"), createMode(modeNXDN_N, 0x20, "NXDN-N"), + createMode(modeDCR, 0x21, "DCR")}); break; case model9700: rigCaps.modelName = QString("IC-9700"); @@ -2667,6 +2696,9 @@ void rigCommander::determineRigCaps() rigCaps.bsr[band23cm] = 0x03; rigCaps.bsr[band70cm] = 0x02; rigCaps.bsr[band2m] = 0x01; + rigCaps.modes = commonModes; + rigCaps.modes.insert(rigCaps.modes.end(), {createMode(modeDV, 0x17, "DV"), + createMode(modeDD, 0x22, "DD")}); break; case model910h: rigCaps.modelName = QString("IC-910H"); @@ -2686,6 +2718,7 @@ void rigCommander::determineRigCaps() rigCaps.bsr[band23cm] = 0x03; rigCaps.bsr[band70cm] = 0x02; rigCaps.bsr[band2m] = 0x01; + rigCaps.modes = commonModes; break; case model7610: rigCaps.modelName = QString("IC-7610"); @@ -2714,6 +2747,7 @@ void rigCommander::determineRigCaps() rigCaps.bands.push_back(bandGen); rigCaps.bands.push_back(band630m); rigCaps.bands.push_back(band2200m); + rigCaps.modes = commonModes; break; case model7850: rigCaps.modelName = QString("IC-785x"); @@ -2741,6 +2775,9 @@ void rigCommander::determineRigCaps() rigCaps.bands.push_back(bandGen); rigCaps.bands.push_back(band630m); rigCaps.bands.push_back(band2200m); + rigCaps.modes = commonModes; + rigCaps.modes.insert(rigCaps.modes.end(), {createMode(modePSK, 0x12, "PSK"), + createMode(modePSK_R, 0x13, "PSK-R")}); break; case model705: rigCaps.modelName = QString("IC-705"); @@ -2773,6 +2810,9 @@ void rigCommander::determineRigCaps() rigCaps.bsr[bandGen] = 0x15; rigCaps.bands.push_back(band630m); rigCaps.bands.push_back(band2200m); + rigCaps.modes = commonModes; + rigCaps.modes.insert(rigCaps.modes.end(), {createMode(modeWFM, 0x06, "WFM"), + createMode(modeDV, 0x17, "DV")}); break; case model7000: rigCaps.modelName = QString("IC-7000"); @@ -2792,6 +2832,7 @@ void rigCommander::determineRigCaps() rigCaps.bsr[band2m] = 0x11; rigCaps.bsr[band70cm] = 0x12; rigCaps.bsr[bandGen] = 0x13; + rigCaps.modes = commonModes; break; case model7410: rigCaps.modelName = QString("IC-7410"); @@ -2810,6 +2851,7 @@ void rigCommander::determineRigCaps() rigCaps.bands = standardHF; rigCaps.bands.push_back(bandGen); rigCaps.bsr[bandGen] = 0x11; + rigCaps.modes = commonModes; break; case model7100: rigCaps.modelName = QString("IC-7100"); @@ -2832,6 +2874,9 @@ void rigCommander::determineRigCaps() rigCaps.bsr[band2m] = 0x11; rigCaps.bsr[band70cm] = 0x12; rigCaps.bsr[bandGen] = 0x13; + rigCaps.modes = commonModes; + rigCaps.modes.insert(rigCaps.modes.end(), {createMode(modeWFM, 0x06, "WFM"), + createMode(modeDV, 0x17, "DV")}); break; case model7200: rigCaps.modelName = QString("IC-7200"); @@ -2849,6 +2894,7 @@ void rigCommander::determineRigCaps() rigCaps.bands = standardHF; rigCaps.bands.push_back(bandGen); rigCaps.bsr[bandGen] = 0x11; + rigCaps.modes = commonModes; break; case model706: rigCaps.modelName = QString("IC-706"); @@ -2862,6 +2908,8 @@ void rigCommander::determineRigCaps() rigCaps.bands = standardHF; rigCaps.bands.insert(rigCaps.bands.end(), standardVU.begin(), standardVU.end()); rigCaps.bands.push_back(bandGen); + rigCaps.modes = commonModes; + rigCaps.modes.insert(rigCaps.modes.end(), createMode(modeWFM, 0x06, "WFM")); break; case model756pro: rigCaps.modelName = QString("IC-756 Pro"); @@ -2878,6 +2926,24 @@ void rigCommander::determineRigCaps() rigCaps.bands = standardHF; rigCaps.bands.push_back(bandGen); rigCaps.bsr[bandGen] = 0x11; + rigCaps.modes = commonModes; + break; + case model756proii: + rigCaps.modelName = QString("IC-756 Pro II"); + rigCaps.hasSpectrum = false; + rigCaps.inputs.clear(); + rigCaps.hasLan = false; + rigCaps.hasEthernet = false; + rigCaps.hasWiFi = false; + rigCaps.hasATU = true; + rigCaps.preamps.push_back('\x01'); + rigCaps.preamps.push_back('\x02'); + rigCaps.attenuators.insert(rigCaps.attenuators.end(),{ '\x06' , '\x12', '\x18'}); + rigCaps.antennas = {0x00, 0x01}; + rigCaps.bands = standardHF; + rigCaps.bands.push_back(bandGen); + rigCaps.bsr[bandGen] = 0x11; + rigCaps.modes = commonModes; break; case model756proiii: rigCaps.modelName = QString("IC-756 Pro III"); @@ -2894,6 +2960,7 @@ void rigCommander::determineRigCaps() rigCaps.bands = standardHF; rigCaps.bands.push_back(bandGen); rigCaps.bsr[bandGen] = 0x11; + rigCaps.modes = commonModes; break; default: rigCaps.modelName = QString("IC-0x%1").arg(rigCaps.modelID, 2, 16); @@ -2913,7 +2980,8 @@ void rigCommander::determineRigCaps() rigCaps.bands = standardHF; rigCaps.bands.insert(rigCaps.bands.end(), standardVU.begin(), standardVU.end()); rigCaps.bands.insert(rigCaps.bands.end(), {band23cm, band4m, band630m, band2200m, bandGen}); - qInfo(logRig()) << "Found unknown rig: " << rigCaps.modelID; + rigCaps.modes = commonModes; + qInfo(logRig()) << "Found unknown rig: 0x" << QString("%1").arg(rigCaps.modelID, 2, 16); break; } haveRigCaps = true; diff --git a/rigcommander.h b/rigcommander.h index bdb111c0..23f714ee 100644 --- a/rigcommander.h +++ b/rigcommander.h @@ -9,6 +9,7 @@ #include "udphandler.h" #include "rigidentities.h" #include "repeaterattributes.h" +#include "freqmemory.h" // This file figures out what to send to the comm and also // parses returns into useful things. @@ -375,6 +376,7 @@ private: void debugMe(); void printHex(const QByteArray &pdata); void printHex(const QByteArray &pdata, bool printVert, bool printHoriz); + mode_info createMode(mode_kind m, unsigned char reg, QString name); commHandler* comm = Q_NULLPTR; pttyHandler* ptty = Q_NULLPTR; diff --git a/rigidentities.cpp b/rigidentities.cpp index f335659d..15f62acc 100644 --- a/rigidentities.cpp +++ b/rigidentities.cpp @@ -58,6 +58,9 @@ model_kind determineRadioModel(unsigned char rigID) case model756pro: rig = model756pro; break; + case model756proii: + rig = model756proii; + break; default: rig = modelUnknown; break; diff --git a/rigidentities.h b/rigidentities.h index 21a8bf43..9ad26300 100644 --- a/rigidentities.h +++ b/rigidentities.h @@ -6,6 +6,8 @@ #include #include +#include "freqmemory.h" + // Credit for parts of CIV list: // http://www.docksideradio.com/Icom%20Radio%20Hex%20Addresses.htm @@ -27,6 +29,7 @@ enum model_kind { model705 = 0xA4, model706 = 0x58, model756pro = 0x5C, + model756proii = 0x64, model756proiii = 0x6E, model910h = 0x60, modelUnknown = 0xFF @@ -102,6 +105,8 @@ struct rigCapabilities { std::vector antennas; std::vector bands; unsigned char bsr[20] = {0}; + + std::vector modes; }; diff --git a/udpserversetup.ui b/udpserversetup.ui index edab45bd..28c0e8f3 100644 --- a/udpserversetup.ui +++ b/udpserversetup.ui @@ -341,6 +341,13 @@ + + + + Note: This allows other computers to connect to this computer's radio + + + diff --git a/wfmain.cpp b/wfmain.cpp index b65fd6f4..dcff89df 100644 --- a/wfmain.cpp +++ b/wfmain.cpp @@ -895,6 +895,9 @@ void wfmain::receiveFoundRigID(rigCapabilities rigCaps) delayedCommand->setInterval( msMinTiming * 2); // 20 byte message periodicPollingTimer->setInterval( msMinTiming ); // slower for s-meter poll + qInfo(logSystem()) << "Delay command interval timing: " << msMinTiming * 2 << "ms"; + qInfo(logSystem()) << "Periodic polling timer: " << msMinTiming << "ms"; + // Normal: delayedCmdIntervalLAN_ms = msMinTiming * 2; delayedCmdIntervalSerial_ms = msMinTiming * 2; @@ -1070,11 +1073,11 @@ void wfmain::loadSettings() { ui->baudRateCombo->setEnabled(false); ui->serialDeviceListCombo->setEnabled(false); - ui->udpServerSetupBtn->setEnabled(false); + //ui->udpServerSetupBtn->setEnabled(false); } else { ui->baudRateCombo->setEnabled(true); ui->serialDeviceListCombo->setEnabled(true); - ui->udpServerSetupBtn->setEnabled(true); + //ui->udpServerSetupBtn->setEnabled(true); } ui->lanEnableBtn->setChecked(prefs.enableLAN); @@ -2276,40 +2279,19 @@ void wfmain::receiveRigID(rigCapabilities rigCaps) // Added so that server receives rig capabilities. emit sendRigCaps(rigCaps); rpt->setRig(rigCaps); - if(rigCaps.model==model7850) + + // Set the mode combo box up: + + ui->modeSelectCombo->blockSignals(true); + ui->modeSelectCombo->clear(); + + for(unsigned int i=0; i < rigCaps.modes.size(); i++) { - ui->modeSelectCombo->addItem("PSK", 0x12); - ui->modeSelectCombo->addItem("PSK-R", 0x13); + ui->modeSelectCombo->addItem(rigCaps.modes.at(i).name, + rigCaps.modes.at(i).reg); } - if(rigCaps.hasDV) - { - ui->modeSelectCombo->addItem("DV", 0x17); - } - - if(rigCaps.hasDD) - { - ui->modeSelectCombo->addItem("DD", 0x22); - } - - if(rigCaps.model==modelR8600) - { - ui->modeSelectCombo->addItem("WFM", 0x06); - ui->modeSelectCombo->addItem("FSK-R", 0x08); - ui->modeSelectCombo->addItem("S-AM (D)", 0x11); - ui->modeSelectCombo->addItem("S-AM (L)", 0x14); - ui->modeSelectCombo->addItem("S-AM (U)", 0x15); - ui->modeSelectCombo->addItem("P25", 0x16); - ui->modeSelectCombo->addItem("dPMR", 0x18); - ui->modeSelectCombo->addItem("NXDN-VN", 0x19); - ui->modeSelectCombo->addItem("NXDN-N", 0x20); - ui->modeSelectCombo->addItem("DCR", 0x21); - } - - if (rigCaps.model == model705) - { - ui->modeSelectCombo->addItem("WFM", 0x06); - } + ui->modeSelectCombo->blockSignals(false); if(rigCaps.model == model9700) { @@ -3525,7 +3507,7 @@ void wfmain::on_serialEnableBtn_clicked(bool checked) ui->txLatencyValue->setEnabled(!checked); ui->baudRateCombo->setEnabled(checked); ui->serialDeviceListCombo->setEnabled(checked); - ui->udpServerSetupBtn->setEnabled(true); + //ui->udpServerSetupBtn->setEnabled(true); } void wfmain::on_lanEnableBtn_clicked(bool checked) @@ -3538,7 +3520,7 @@ void wfmain::on_lanEnableBtn_clicked(bool checked) ui->passwordTxt->setEnabled(checked); ui->baudRateCombo->setEnabled(!checked); ui->serialDeviceListCombo->setEnabled(!checked); - ui->udpServerSetupBtn->setEnabled(false); + //ui->udpServerSetupBtn->setEnabled(false); if(checked) { showStatusBarText("After filling in values, press Save Settings and re-start wfview."); @@ -4387,7 +4369,7 @@ void wfmain::on_rigCIVaddrHexLine_editingFinished() prefs.radioCIVAddr = propCIVAddr; showStatusBarText(QString("Setting radio CI-V address to: 0x%1. Press Save Settings to retain.").arg(propCIVAddr, 2, 16)); } else { - showStatusBarText(QString("Could not use provided CI-V address. Address must be < 0x7E")); + showStatusBarText(QString("Could not use provided CI-V address. Address must be < 0xE0")); } } diff --git a/wfmain.ui b/wfmain.ui index 330242df..33cf6f9a 100644 --- a/wfmain.ui +++ b/wfmain.ui @@ -18,7 +18,7 @@ - 0 + 3 @@ -2600,6 +2600,40 @@ + + + + 0 + + + + + Please note: Changing CI-V Address, Serial Device, and Baud rate requires pressing "Save Settings", closing wfview, and re-opening. + + + + + + + + + 0 + + + + + <html><head></head><body><p>Please see the <a href="https://wfview.org/wfview-user-manual/settings-tab/">User Manual</a> for more information. </p></body></html> + + + Qt::RichText + + + true + + + + + From 4ca16d7376b540e5fb5e0aa5c51cf588ef2bbace Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Thu, 27 May 2021 14:42:28 +0100 Subject: [PATCH 19/47] Fix udpserver --- udpserver.cpp | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/udpserver.cpp b/udpserver.cpp index 23a1dff0..548f4867 100644 --- a/udpserver.cpp +++ b/udpserver.cpp @@ -697,7 +697,8 @@ void udpServer::audioReceived() tempAudio.sent = 0; tempAudio.data = r.mid(0x18); //qInfo(logUdpServer()) << "sending tx audio " << in->seq; - emit haveAudioData(tempAudio); + //emit haveAudioData(tempAudio); + txaudio->incomingAudio(tempAudio); } } break; @@ -1298,11 +1299,10 @@ void udpServer::dataForServer(QByteArray d) void udpServer::sendRxAudio() { - if (rxaudio && rxaudio->isChunkAvailable()) { + if (rxaudio) { QByteArray audio; rxaudio->getNextAudioChunk(audio); int len = 0; - while (len < audio.length()) { audioPacket partial; partial.data = audio.mid(len, 1364); From 2ac8d3332c12c860d8f481ad3deede5594fffcb9 Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Thu, 27 May 2021 14:57:06 +0100 Subject: [PATCH 20/47] Update udpserver.cpp --- udpserver.cpp | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/udpserver.cpp b/udpserver.cpp index 548f4867..1103324e 100644 --- a/udpserver.cpp +++ b/udpserver.cpp @@ -395,7 +395,7 @@ void udpServer::controlReceived() connect(this, SIGNAL(setupTxAudio(quint8,quint8,quint16,quint16,bool,bool,int,quint8)), txaudio, SLOT(init(quint8,quint8,quint16,quint16,bool,bool,int,quint8))); connect(txAudioThread, SIGNAL(finished()), txaudio, SLOT(deleteLater())); - emit setupTxAudio(samples, channels, current->txSampleRate, in->txbuffer, uLaw, false, config.audioOutput, config.resampleQuality); + emit setupTxAudio(samples, channels, current->txSampleRate, current->txBufferLen, uLaw, false, config.audioOutput, config.resampleQuality); hasTxAudio=datagram.senderAddress(); connect(this, SIGNAL(haveAudioData(audioPacket)), txaudio, SLOT(incomingAudio(audioPacket))); From be81d8d767b1df5ec329c29bfc8aa2291a9c15d8 Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Thu, 27 May 2021 15:25:18 +0100 Subject: [PATCH 21/47] Update udpserver.cpp --- udpserver.cpp | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/udpserver.cpp b/udpserver.cpp index 1103324e..40b60527 100644 --- a/udpserver.cpp +++ b/udpserver.cpp @@ -398,7 +398,7 @@ void udpServer::controlReceived() emit setupTxAudio(samples, channels, current->txSampleRate, current->txBufferLen, uLaw, false, config.audioOutput, config.resampleQuality); hasTxAudio=datagram.senderAddress(); - connect(this, SIGNAL(haveAudioData(audioPacket)), txaudio, SLOT(incomingAudio(audioPacket))); + //connect(this, SIGNAL(haveAudioData(audioPacket)), txaudio, SLOT(incomingAudio(audioPacket))); } if (rxaudio == Q_NULLPTR) @@ -425,7 +425,7 @@ void udpServer::controlReceived() rxaudio->moveToThread(rxAudioThread); rxAudioThread->start(); - connect(this, SIGNAL(setupRxAudio(quint8,quint8,quint16,quint16,bool,bool,QString,int,quint8)), rxaudio, SLOT(init(quint8,quint8,quint16,quint16,bool,bool,int,quint8))); + connect(this, SIGNAL(setupRxAudio(quint8,quint8,quint16,quint16,bool,bool,int,quint8)), rxaudio, SLOT(init(quint8,quint8,quint16,quint16,bool,bool,int,quint8))); connect(rxAudioThread, SIGNAL(finished()), txaudio, SLOT(deleteLater())); emit setupRxAudio(samples, channels, current->rxSampleRate, 150, uLaw, true, config.audioInput, config.resampleQuality); From 49bcc40f2e9eec66aa0502a9aad3769cfdce823a Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Thu, 27 May 2021 17:59:01 +0100 Subject: [PATCH 22/47] udpServer fixes --- udphandler.cpp | 7 ++++--- udpserver.cpp | 1 + 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/udphandler.cpp b/udphandler.cpp index 11745245..52013559 100644 --- a/udphandler.cpp +++ b/udphandler.cpp @@ -747,7 +747,8 @@ udpAudio::udpAudio(QHostAddress local, QHostAddress ip, quint16 audioPort, quint connect(this, SIGNAL(setupRxAudio(quint8,quint8,quint16,quint16,bool,bool,int,quint8)), rxaudio, SLOT(init(quint8,quint8,quint16,quint16,bool,bool,int,quint8))); - connect(this, SIGNAL(haveAudioData(audioPacket)), rxaudio, SLOT(incomingAudio(audioPacket))); + // signal/slot not currently used. + //connect(this, SIGNAL(haveAudioData(audioPacket)), rxaudio, SLOT(incomingAudio(audioPacket))); connect(this, SIGNAL(haveChangeLatency(quint16)), rxaudio, SLOT(changeLatency(quint16))); connect(this, SIGNAL(haveSetVolume(unsigned char)), rxaudio, SLOT(setVolume(unsigned char))); connect(rxAudioThread, SIGNAL(finished()), rxaudio, SLOT(deleteLater())); @@ -950,8 +951,8 @@ void udpAudio::dataReceived() tempAudio.data = r.mid(0x18); // Prefer signal/slot to forward audio as it is thread/safe // Need to do more testing but latency appears fine. - //emit haveAudioData(tempAudio); - audioLatency=rxaudio->incomingAudio(tempAudio); + audioLatency = rxaudio->incomingAudio(tempAudio); + emit haveAudioData(tempAudio); } break; } diff --git a/udpserver.cpp b/udpserver.cpp index 40b60527..275b58f7 100644 --- a/udpserver.cpp +++ b/udpserver.cpp @@ -398,6 +398,7 @@ void udpServer::controlReceived() emit setupTxAudio(samples, channels, current->txSampleRate, current->txBufferLen, uLaw, false, config.audioOutput, config.resampleQuality); hasTxAudio=datagram.senderAddress(); + // signal/slot not currently used. //connect(this, SIGNAL(haveAudioData(audioPacket)), txaudio, SLOT(incomingAudio(audioPacket))); } From 1e917b3c3be9f7352299b4c54d08b161fe53976f Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Thu, 27 May 2021 18:34:44 +0100 Subject: [PATCH 23/47] udpserver fixes --- audiohandler.cpp | 14 +++++++------- audiohandler.h | 5 +++-- udphandler.cpp | 5 +++-- udpserver.cpp | 6 ++---- 4 files changed, 15 insertions(+), 15 deletions(-) diff --git a/audiohandler.cpp b/audiohandler.cpp index e14282c0..44da328d 100644 --- a/audiohandler.cpp +++ b/audiohandler.cpp @@ -131,8 +131,8 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 void audioHandler::setVolume(unsigned char volume) { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << (qreal)(volume/255.0) << ")"; - this->volume = (qreal)(volume / 255.0); + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << (qreal)(volume / 255.0) << ")"; + this->volume = (qreal)(volume/255.0); } @@ -267,13 +267,13 @@ void audioHandler::stateChanged(QAudio::State state) -int audioHandler::incomingAudio(audioPacket data) +void audioHandler::incomingAudio(audioPacket data) { // No point buffering audio until stream is actually running. if (!audio.isStreamRunning()) { qDebug(logAudio()) << "Packet received before stream was started"; - return currentLatency; + return; } // Incoming data is 8bits? if (radioSampleBits == 8) @@ -349,7 +349,7 @@ int audioHandler::incomingAudio(audioPacket data) { qDebug(logAudio()) << "Buffer full! capacity:" << ringBuf->capacity() << "length" << ringBuf->size(); } - return currentLatency; + return; } void audioHandler::changeLatency(const quint16 newSize) @@ -358,9 +358,9 @@ void audioHandler::changeLatency(const quint16 newSize) audioLatency = newSize; } -void audioHandler::getLatency() +int audioHandler::getLatency() { - emit sendLatency(audioLatency); + return currentLatency; } bool audioHandler::isChunkAvailable() diff --git a/audiohandler.h b/audiohandler.h index 26bd199b..1948c9c5 100644 --- a/audiohandler.h +++ b/audiohandler.h @@ -13,6 +13,7 @@ typedef signed short MY_TYPE; #define FORMAT RTAUDIO_SINT16 #define SCALE 32767.0 +#define LOG100 4.60517018599 #include #include @@ -46,7 +47,7 @@ public: audioHandler(QObject* parent = 0); ~audioHandler(); - void getLatency(); + int getLatency(); bool setDevice(QAudioDeviceInfo deviceInfo); @@ -58,7 +59,6 @@ public: bool isSequential() const; void getNextAudioChunk(QByteArray &data); bool isChunkAvailable(); - int incomingAudio(const audioPacket data); private slots: bool init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isulaw, const bool isinput, int port, quint8 resampleQuality); @@ -66,6 +66,7 @@ private slots: void notified(); void stateChanged(QAudio::State state); void setVolume(unsigned char volume); + void incomingAudio(const audioPacket data); signals: void audioMessage(QString message); diff --git a/udphandler.cpp b/udphandler.cpp index 52013559..c8278f6a 100644 --- a/udphandler.cpp +++ b/udphandler.cpp @@ -748,7 +748,7 @@ udpAudio::udpAudio(QHostAddress local, QHostAddress ip, quint16 audioPort, quint connect(this, SIGNAL(setupRxAudio(quint8,quint8,quint16,quint16,bool,bool,int,quint8)), rxaudio, SLOT(init(quint8,quint8,quint16,quint16,bool,bool,int,quint8))); // signal/slot not currently used. - //connect(this, SIGNAL(haveAudioData(audioPacket)), rxaudio, SLOT(incomingAudio(audioPacket))); + connect(this, SIGNAL(haveAudioData(audioPacket)), rxaudio, SLOT(incomingAudio(audioPacket))); connect(this, SIGNAL(haveChangeLatency(quint16)), rxaudio, SLOT(changeLatency(quint16))); connect(this, SIGNAL(haveSetVolume(unsigned char)), rxaudio, SLOT(setVolume(unsigned char))); connect(rxAudioThread, SIGNAL(finished()), rxaudio, SLOT(deleteLater())); @@ -951,8 +951,9 @@ void udpAudio::dataReceived() tempAudio.data = r.mid(0x18); // Prefer signal/slot to forward audio as it is thread/safe // Need to do more testing but latency appears fine. - audioLatency = rxaudio->incomingAudio(tempAudio); + //audioLatency = rxaudio->incomingAudio(tempAudio); emit haveAudioData(tempAudio); + audioLatency = rxaudio->getLatency(); } break; } diff --git a/udpserver.cpp b/udpserver.cpp index 275b58f7..f6c22a44 100644 --- a/udpserver.cpp +++ b/udpserver.cpp @@ -398,8 +398,7 @@ void udpServer::controlReceived() emit setupTxAudio(samples, channels, current->txSampleRate, current->txBufferLen, uLaw, false, config.audioOutput, config.resampleQuality); hasTxAudio=datagram.senderAddress(); - // signal/slot not currently used. - //connect(this, SIGNAL(haveAudioData(audioPacket)), txaudio, SLOT(incomingAudio(audioPacket))); + connect(this, SIGNAL(haveAudioData(audioPacket)), txaudio, SLOT(incomingAudio(audioPacket))); } if (rxaudio == Q_NULLPTR) @@ -698,8 +697,7 @@ void udpServer::audioReceived() tempAudio.sent = 0; tempAudio.data = r.mid(0x18); //qInfo(logUdpServer()) << "sending tx audio " << in->seq; - //emit haveAudioData(tempAudio); - txaudio->incomingAudio(tempAudio); + emit haveAudioData(tempAudio); } } break; From 8859f2d6ea57921652cb617f018e041b13cafba3 Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Thu, 27 May 2021 18:46:01 +0100 Subject: [PATCH 24/47] Allow higher latency --- audiohandler.cpp | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/audiohandler.cpp b/audiohandler.cpp index 44da328d..abbc3ffe 100644 --- a/audiohandler.cpp +++ b/audiohandler.cpp @@ -179,13 +179,13 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n qDebug(logAudio()) << "Adding partial:" << send; } - while (currentLatency > (int)audioLatency) { + if (currentLatency > (int)audioLatency) { qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet.seq << " arrived too late (increase output latency!) " << dec << packet.time.msecsTo(QTime::currentTime()) << "ms"; - lastSeq = packet.seq; - if (!ringBuf->try_read(packet)) - return sentlen; + //lastSeq = packet.seq; + //if (!ringBuf->try_read(packet)) + // return sentlen; currentLatency = packet.time.msecsTo(QTime::currentTime()); } From 2a61e7f849a39f4c847fd311a6a91fc1a0bee48e Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Thu, 27 May 2021 18:48:19 +0100 Subject: [PATCH 25/47] Latency reporting causing issue? --- audiohandler.cpp | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/audiohandler.cpp b/audiohandler.cpp index abbc3ffe..5860402e 100644 --- a/audiohandler.cpp +++ b/audiohandler.cpp @@ -180,9 +180,9 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n } if (currentLatency > (int)audioLatency) { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet.seq << - " arrived too late (increase output latency!) " << - dec << packet.time.msecsTo(QTime::currentTime()) << "ms"; + //qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet.seq << + // " arrived too late (increase output latency!) " << + // dec << packet.time.msecsTo(QTime::currentTime()) << "ms"; //lastSeq = packet.seq; //if (!ringBuf->try_read(packet)) // return sentlen; From 941b5159d50ba1460e3cd303b1f2d4718e1d407d Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Thu, 27 May 2021 23:23:23 +0100 Subject: [PATCH 26/47] Use preferred sample rate rather than force 48000 --- audiohandler.cpp | 42 +++++++++++++++++++++++------------------- 1 file changed, 23 insertions(+), 19 deletions(-) diff --git a/audiohandler.cpp b/audiohandler.cpp index 5860402e..d167d0ed 100644 --- a/audiohandler.cpp +++ b/audiohandler.cpp @@ -84,6 +84,10 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 (info.nativeFormats & RTAUDIO_FLOAT32 ? "32-bit float," : "") << (info.nativeFormats & RTAUDIO_FLOAT64 ? "64-bit float," : ""); qInfo(logAudio()) << " Preferred sample rate:" << info.preferredSampleRate; + if (isInput) + qInfo(logAudio()) << " Input Channels:" << info.inputChannels; + else + qInfo(logAudio()) << " Output Channels:" << info.outputChannels; } qInfo(logAudio()) << " chunkSize: " << chunkSize; @@ -96,10 +100,10 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 int resample_error = 0; if (isInput) { - resampler = wf_resampler_init(radioChannels, INTERNAL_SAMPLE_RATE, samplerate, resampleQuality, &resample_error); + resampler = wf_resampler_init(radioChannels, info.preferredSampleRate, samplerate, resampleQuality, &resample_error); try { - aParams.nChannels = 1; // Internally this is always 2 channels - audio.openStream(NULL, &aParams, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &this->chunkSize, &staticWrite, this); + aParams.nChannels = radioChannels; + audio.openStream(NULL, &aParams, RTAUDIO_SINT16, info.preferredSampleRate, &this->chunkSize, &staticWrite, this); audio.startStream(); } catch (RtAudioError& e) { @@ -109,11 +113,11 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 } else { - resampler = wf_resampler_init(radioChannels, samplerate, INTERNAL_SAMPLE_RATE, resampleQuality, &resample_error); + resampler = wf_resampler_init(radioChannels, samplerate, info.preferredSampleRate, resampleQuality, &resample_error); try { unsigned int length = chunkSize / 2; aParams.nChannels = 2; // Internally this is always 2 channels - audio.openStream(&aParams, NULL, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &length, &staticRead, this); + audio.openStream(&aParams, NULL, RTAUDIO_SINT16, info.preferredSampleRate, &length, &staticRead, this); audio.startStream(); } catch (RtAudioError& e) { @@ -176,16 +180,16 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n memcpy(buffer + sentlen, tempBuf.data.constData() + tempBuf.sent, send); tempBuf.sent = tempBuf.sent + send; sentlen = sentlen + send; - qDebug(logAudio()) << "Adding partial:" << send; + //qDebug(logAudio()) << "Adding partial:" << send; } if (currentLatency > (int)audioLatency) { - //qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet.seq << - // " arrived too late (increase output latency!) " << - // dec << packet.time.msecsTo(QTime::currentTime()) << "ms"; - //lastSeq = packet.seq; - //if (!ringBuf->try_read(packet)) - // return sentlen; + qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet.seq << + " arrived too late (increase output latency!) " << + dec << packet.time.msecsTo(QTime::currentTime()) << "ms"; + lastSeq = packet.seq; + if (!ringBuf->try_read(packet)) + return sentlen; currentLatency = packet.time.msecsTo(QTime::currentTime()); } @@ -194,18 +198,18 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n sentlen = sentlen + send; if (send < packet.data.length()) { - qDebug(logAudio()) << "Asking for partial, sent:" << send << "packet length" << packet.data.length(); + //qDebug(logAudio()) << "Asking for partial, sent:" << send << "packet length" << packet.data.length(); tempBuf = packet; tempBuf.sent = tempBuf.sent + send; - //lastSeq = packet.seq; - //break; + lastSeq = packet.seq; + break; } if (packet.seq <= lastSeq) { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Duplicate/early audio packet: " << hex << lastSeq << " got " << hex << packet.seq; + qDebug(logAudio()) << (isInput ? "Input" : "Output") << "Duplicate/early audio packet: " << hex << lastSeq << " got " << hex << packet.seq; } else if (packet.seq != lastSeq + 1) { - qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet.seq - 1; + qDebug(logAudio()) << (isInput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet.seq - 1; } lastSeq = packet.seq; } @@ -219,7 +223,7 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status) { int sentlen = 0; - unsigned int nBytes = nFrames * 2; // This is ALWAYS 2 bytes per sample and 1 channel + unsigned int nBytes = nFrames * 2 * radioChannels; // This is ALWAYS 2 bytes per sample const char* data = (const char*)inputBuffer; while (sentlen < nBytes) { if (tempBuf.sent != chunkSize) @@ -229,7 +233,7 @@ int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int sentlen = sentlen + send; tempBuf.seq = 0; // Not used in TX tempBuf.time = QTime::currentTime(); - tempBuf.sent = tempBuf.data.length(); + tempBuf.sent = tempBuf.sent + send; } else { if (!ringBuf->try_write(tempBuf)) From c7af6a3a3850953526c776db68bafffd584bdc25 Mon Sep 17 00:00:00 2001 From: Phil Taylor Date: Fri, 28 May 2021 10:45:09 +0100 Subject: [PATCH 27/47] Change toolbar display formatting --- udphandler.cpp | 7 +++---- udpserver.cpp | 5 +++-- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/udphandler.cpp b/udphandler.cpp index c8278f6a..29c9f5a7 100644 --- a/udphandler.cpp +++ b/udphandler.cpp @@ -192,13 +192,12 @@ void udpHandler::dataReceived() QString tempLatency; if (rxLatency > audio->audioLatency) { - tempLatency = QString::number(audio->audioLatency) + "ms"; + tempLatency = QString("%1 ms").arg(audio->audioLatency,3); } else { - tempLatency = "" + QString::number(audio->audioLatency) + "ms"; + tempLatency = QString("%1 ms").arg(audio->audioLatency,3); } - - emit haveNetworkStatus("rx latency: " + tempLatency + " ms / rtt: " + QString::number(latency) + " ms, loss: (" + QString::number(totallost) + "/" + QString::number(totalsent) + ")"); + emit haveNetworkStatus(QString("
rx latency: %1 ms / rtt: %2 ms / loss: %3/%4").arg(tempLatency).arg(latency, 3).arg(totallost,3).arg(totalsent,3));
                 }
                 break;
             }
diff --git a/udpserver.cpp b/udpserver.cpp
index f6c22a44..14ae346d 100644
--- a/udpserver.cpp
+++ b/udpserver.cpp
@@ -428,7 +428,7 @@ void udpServer::controlReceived()
                         connect(this, SIGNAL(setupRxAudio(quint8,quint8,quint16,quint16,bool,bool,int,quint8)), rxaudio, SLOT(init(quint8,quint8,quint16,quint16,bool,bool,int,quint8)));
                         connect(rxAudioThread, SIGNAL(finished()), txaudio, SLOT(deleteLater()));
 
-                        emit setupRxAudio(samples, channels, current->rxSampleRate, 150, uLaw, true, config.audioInput, config.resampleQuality);
+                        emit setupRxAudio(samples, channels, current->rxSampleRate,150, uLaw, true, config.audioInput, config.resampleQuality);
 
                         rxAudioTimer = new QTimer();
                         rxAudioTimer->setTimerType(Qt::PreciseTimer);
@@ -1298,8 +1298,9 @@ void udpServer::dataForServer(QByteArray d)
 
 void udpServer::sendRxAudio()
 {
+    QByteArray audio;
     if (rxaudio) {
-        QByteArray audio;
+        audio.clear();
         rxaudio->getNextAudioChunk(audio);
         int len = 0;
         while (len < audio.length()) {

From 8578ca63768f2ec8a82125ca71fc5908f90459c7 Mon Sep 17 00:00:00 2001
From: Phil Taylor 
Date: Fri, 28 May 2021 10:48:04 +0100
Subject: [PATCH 28/47] Update udphandler.cpp

---
 udphandler.cpp | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/udphandler.cpp b/udphandler.cpp
index 29c9f5a7..24e05d1e 100644
--- a/udphandler.cpp
+++ b/udphandler.cpp
@@ -197,7 +197,7 @@ void udpHandler::dataReceived()
                     else {
                         tempLatency = QString("%1 ms").arg(audio->audioLatency,3);
                     }
-                    emit haveNetworkStatus(QString("
rx latency: %1 ms / rtt: %2 ms / loss: %3/%4").arg(tempLatency).arg(latency, 3).arg(totallost,3).arg(totalsent,3));
+                    emit haveNetworkStatus(QString("
rx latency: %1 ms / rtt: %2 ms / loss: %3/%4
").arg(tempLatency).arg(latency, 3).arg(totallost,3).arg(totalsent,3));
                 }
                 break;
             }

From 344ab41930ca0fe928c6ab73f00c87c71b3123c8 Mon Sep 17 00:00:00 2001
From: Phil Taylor 
Date: Fri, 28 May 2021 18:13:08 +0100
Subject: [PATCH 29/47] Update audiohandler.cpp

---
 audiohandler.cpp | 5 +++++
 1 file changed, 5 insertions(+)

diff --git a/audiohandler.cpp b/audiohandler.cpp
index d167d0ed..a37085e6 100644
--- a/audiohandler.cpp
+++ b/audiohandler.cpp
@@ -180,6 +180,11 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
 				memcpy(buffer + sentlen, tempBuf.data.constData() + tempBuf.sent, send);
 				tempBuf.sent = tempBuf.sent + send;
 				sentlen = sentlen + send;
+				if (tempBuf.sent != tempBuf.data.length())
+				{
+					// We still don't have enough buffer space for this?
+					return 0;
+				}
 				//qDebug(logAudio()) << "Adding partial:" << send;
 			}
 

From abf47038559f1a983081c502e75f94e3ef85b6f1 Mon Sep 17 00:00:00 2001
From: Phil Taylor 
Date: Fri, 28 May 2021 18:33:40 +0100
Subject: [PATCH 30/47] Add some startup logging

---
 main.cpp | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

diff --git a/main.cpp b/main.cpp
index 24f50ea1..9a7143e7 100644
--- a/main.cpp
+++ b/main.cpp
@@ -117,12 +117,12 @@ int main(int argc, char *argv[])
     // Set handler
     qInstallMessageHandler(messageHandler);
 
-    qInfo(logSystem()) << "Starting wfview";
-
-
-    qDebug(logSystem()) << "SerialPortCL as set by parser: " << serialPortCL;
-    qDebug(logSystem()) << "remote host as set by parser: " << hostCL;
-    qDebug(logSystem()) << "CIV as set by parser: " << civCL;
+    qInfo(logSystem()) << QString("Starting wfview: build %1 on %2 at %3 by %5@%6").arg(GITSHORT).arg(__DATE__).arg(__TIME__).arg(UNAME).arg(HOST);
+    qInfo(logSystem()) << QString("Operating System: %1 (%2)").arg(QSysInfo::prettyProductName()).arg(QSysInfo::buildCpuArchitecture());
+    qInfo(logSystem()) << QString("Build Qt Version %1. Current Qt Version: %2").arg(QT_VERSION_STR).arg(qVersion());
+    qDebug(logSystem()) << QString("SerialPortCL as set by parser: %1").arg(serialPortCL);
+    qDebug(logSystem()) << QString("remote host as set by parser: %1").arg(hostCL);
+    qDebug(logSystem()) << QString("CIV as set by parser: %1").arg(civCL);
     a.setWheelScrollLines(1); // one line per wheel click
     wfmain w( serialPortCL, hostCL, settingsFile);
 

From f3e483a80e9ac2529ba9d18d3ef973e0205b5c9b Mon Sep 17 00:00:00 2001
From: Elliott Liggett 
Date: Fri, 28 May 2021 20:51:19 -0700
Subject: [PATCH 31/47] Cleaning up the main constructor for wfmain.

---
 wfmain.cpp | 958 ++++++++++++++++++++++++++++-------------------------
 wfmain.h   |   9 +
 2 files changed, 520 insertions(+), 447 deletions(-)

diff --git a/wfmain.cpp b/wfmain.cpp
index 50b674cd..8061abbe 100644
--- a/wfmain.cpp
+++ b/wfmain.cpp
@@ -35,138 +35,9 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
 
     haveRigCaps = false;
 
-    ui->bandStkLastUsedBtn->setVisible(false);
-    ui->bandStkVoiceBtn->setVisible(false);
-    ui->bandStkDataBtn->setVisible(false);
-    ui->bandStkCWBtn->setVisible(false);
-
-    keyF1 = new QShortcut(this);
-    keyF1->setKey(Qt::Key_F1);
-    connect(keyF1, SIGNAL(activated()), this, SLOT(shortcutF1()));
-
-    keyF2 = new QShortcut(this);
-    keyF2->setKey(Qt::Key_F2);
-    connect(keyF2, SIGNAL(activated()), this, SLOT(shortcutF2()));
-
-    keyF3 = new QShortcut(this);
-    keyF3->setKey(Qt::Key_F3);
-    connect(keyF3, SIGNAL(activated()), this, SLOT(shortcutF3()));
-
-    keyF4 = new QShortcut(this);
-    keyF4->setKey(Qt::Key_F4);
-    connect(keyF4, SIGNAL(activated()), this, SLOT(shortcutF4()));
-
-    keyF5 = new QShortcut(this);
-    keyF5->setKey(Qt::Key_F5);
-    connect(keyF5, SIGNAL(activated()), this, SLOT(shortcutF5()));
-
-    keyF6 = new QShortcut(this);
-    keyF6->setKey(Qt::Key_F6);
-    connect(keyF6, SIGNAL(activated()), this, SLOT(shortcutF6()));
-
-    keyF7 = new QShortcut(this);
-    keyF7->setKey(Qt::Key_F7);
-    connect(keyF7, SIGNAL(activated()), this, SLOT(shortcutF7()));
-
-    keyF8 = new QShortcut(this);
-    keyF8->setKey(Qt::Key_F8);
-    connect(keyF8, SIGNAL(activated()), this, SLOT(shortcutF8()));
-
-    keyF9 = new QShortcut(this);
-    keyF9->setKey(Qt::Key_F9);
-    connect(keyF9, SIGNAL(activated()), this, SLOT(shortcutF9()));
-
-    keyF10 = new QShortcut(this);
-    keyF10->setKey(Qt::Key_F10);
-    connect(keyF10, SIGNAL(activated()), this, SLOT(shortcutF10()));
-
-    keyF11 = new QShortcut(this);
-    keyF11->setKey(Qt::Key_F11);
-    connect(keyF11, SIGNAL(activated()), this, SLOT(shortcutF11()));
-
-    keyF12 = new QShortcut(this);
-    keyF12->setKey(Qt::Key_F12);
-    connect(keyF12, SIGNAL(activated()), this, SLOT(shortcutF12()));
-
-    keyControlT = new QShortcut(this);
-    keyControlT->setKey(Qt::CTRL + Qt::Key_T);
-    connect(keyControlT, SIGNAL(activated()), this, SLOT(shortcutControlT()));
-
-    keyControlR = new QShortcut(this);
-    keyControlR->setKey(Qt::CTRL + Qt::Key_R);
-    connect(keyControlR, SIGNAL(activated()), this, SLOT(shortcutControlR()));
-
-    keyControlI = new QShortcut(this);
-    keyControlI->setKey(Qt::CTRL + Qt::Key_I);
-    connect(keyControlI, SIGNAL(activated()), this, SLOT(shortcutControlI()));
-
-    keyControlU = new QShortcut(this);
-    keyControlU->setKey(Qt::CTRL + Qt::Key_U);
-    connect(keyControlU, SIGNAL(activated()), this, SLOT(shortcutControlU()));
-
-    keyStar = new QShortcut(this);
-    keyStar->setKey(Qt::Key_Asterisk);
-    connect(keyStar, SIGNAL(activated()), this, SLOT(shortcutStar()));
-
-    keySlash = new QShortcut(this);
-    keySlash->setKey(Qt::Key_Slash);
-    connect(keySlash, SIGNAL(activated()), this, SLOT(shortcutSlash()));
-
-    keyMinus = new QShortcut(this);
-    keyMinus->setKey(Qt::Key_Minus);
-    connect(keyMinus, SIGNAL(activated()), this, SLOT(shortcutMinus()));
-
-    keyPlus = new QShortcut(this);
-    keyPlus->setKey(Qt::Key_Plus);
-    connect(keyPlus, SIGNAL(activated()), this, SLOT(shortcutPlus()));
-
-    keyShiftMinus = new QShortcut(this);
-    keyShiftMinus->setKey(Qt::SHIFT + Qt::Key_Minus);
-    connect(keyShiftMinus, SIGNAL(activated()), this, SLOT(shortcutShiftMinus()));
-
-    keyShiftPlus = new QShortcut(this);
-    keyShiftPlus->setKey(Qt::SHIFT + Qt::Key_Plus);
-    connect(keyShiftPlus, SIGNAL(activated()), this, SLOT(shortcutShiftPlus()));
-
-    keyControlMinus = new QShortcut(this);
-    keyControlMinus->setKey(Qt::CTRL + Qt::Key_Minus);
-    connect(keyControlMinus, SIGNAL(activated()), this, SLOT(shortcutControlMinus()));
-
-    keyControlPlus = new QShortcut(this);
-    keyControlPlus->setKey(Qt::CTRL + Qt::Key_Plus);
-    connect(keyControlPlus, SIGNAL(activated()), this, SLOT(shortcutControlPlus()));
-
-    keyQuit = new QShortcut(this);
-    keyQuit->setKey(Qt::CTRL + Qt::Key_Q);
-    connect(keyQuit, SIGNAL(activated()), this, SLOT(on_exitBtn_clicked()));
-
-    keyPageUp = new QShortcut(this);
-    keyPageUp->setKey(Qt::Key_PageUp);
-    connect(keyPageUp, SIGNAL(activated()), this, SLOT(shortcutPageUp()));
-
-    keyPageDown = new QShortcut(this);
-    keyPageDown->setKey(Qt::Key_PageDown);
-    connect(keyPageDown, SIGNAL(activated()), this, SLOT(shortcutPageDown()));
-
-    keyF = new QShortcut(this);
-    keyF->setKey(Qt::Key_F);
-    connect(keyF, SIGNAL(activated()), this, SLOT(shortcutF()));
-
-    keyM = new QShortcut(this);
-    keyM->setKey(Qt::Key_M);
-    connect(keyM, SIGNAL(activated()), this, SLOT(shortcutM()));
-
-    ui->baudRateCombo->insertItem(0, QString("115200"), 115200);
-    ui->baudRateCombo->insertItem(1, QString("57600"), 57600);
-    ui->baudRateCombo->insertItem(2, QString("38400"), 38400);
-    ui->baudRateCombo->insertItem(3, QString("28800"), 28800);
-    ui->baudRateCombo->insertItem(4, QString("19200"), 19200);
-    ui->baudRateCombo->insertItem(5, QString("9600"), 9600);
-    ui->baudRateCombo->insertItem(6, QString("4800"), 4800);
-    ui->baudRateCombo->insertItem(7, QString("2400"), 2400);
-    ui->baudRateCombo->insertItem(8, QString("1200"), 1200);
-    ui->baudRateCombo->insertItem(9, QString("300"), 300);
+    setupKeyShortcuts();
 
+    setupMainUI();
 
     // Enumerate audio devices, need to do before settings are loaded.
     const auto audioOutputs = QAudioDeviceInfo::availableDevices(QAudio::AudioOutput);
@@ -228,6 +99,8 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
 
     setDefaultColors(); // set of UI colors with defaults populated
     setDefPrefs(); // other default options
+
+
     if (settingsFile.isNull()) {
         settings = new QSettings();
     }
@@ -250,13 +123,12 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
         qDebug() << "Loading settings from:" << path + file;
         settings = new QSettings(path + file, QSettings::Format::IniFormat);
     }
+
+    setupPlots();
     loadSettings(); // Look for saved preferences
     setTuningSteps(); // TODO: Combine into preferences
 
-    ui->spectrumModeCombo->addItem("Center", (spectrumMode)spectModeCenter);
-    ui->spectrumModeCombo->addItem("Fixed", (spectrumMode)spectModeFixed);
-    ui->spectrumModeCombo->addItem("Scroll-C", (spectrumMode)spectModeScrollC);
-    ui->spectrumModeCombo->addItem("Scroll-F", (spectrumMode)spectModeScrollF);
+    setUIToPrefs();
 
     // Start server if enabled in config
     if (serverConfig.enabled) {
@@ -287,92 +159,6 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
 
     }
 
-    plot = ui->plot; // rename it waterfall.
-    wf = ui->waterfall;
-
-    freqIndicatorLine = new QCPItemLine(plot);
-    freqIndicatorLine->setAntialiased(true);
-    freqIndicatorLine->setPen(QPen(Qt::blue));
-
-
-    ui->modeSelectCombo->addItem("LSB",  0x00);
-    ui->modeSelectCombo->addItem("USB",  0x01);
-    ui->modeSelectCombo->addItem("FM",   0x05);
-    ui->modeSelectCombo->addItem("AM",   0x02);
-    ui->modeSelectCombo->addItem("CW",   0x03);
-    ui->modeSelectCombo->addItem("CW-R", 0x07);
-    ui->modeSelectCombo->addItem("RTTY", 0x04);
-    ui->modeSelectCombo->addItem("RTTY-R", 0x08);
-
-
-    ui->modeFilterCombo->addItem("1", 1);
-    ui->modeFilterCombo->addItem("2", 2);
-    ui->modeFilterCombo->addItem("3", 3);
-    ui->modeFilterCombo->addItem("Setup...", 99);
-
-    ui->tuningStepCombo->blockSignals(true);
-
-    ui->tuningStepCombo->addItem("1 Hz",      (unsigned int)       1);
-    ui->tuningStepCombo->addItem("10 Hz",     (unsigned int)      10);
-    ui->tuningStepCombo->addItem("100 Hz",    (unsigned int)     100);
-    ui->tuningStepCombo->addItem("1 kHz",     (unsigned int)    1000);
-    ui->tuningStepCombo->addItem("2.5 kHz",   (unsigned int)    2500);
-    ui->tuningStepCombo->addItem("5 kHz",     (unsigned int)    5000);
-    ui->tuningStepCombo->addItem("6.125 kHz", (unsigned int)    6125);	// PMR 
-    ui->tuningStepCombo->addItem("8.333 kHz", (unsigned int)    8333);	// airband stepsize
-    ui->tuningStepCombo->addItem("9 kHz",     (unsigned int)    9000);	// European medium wave stepsize
-    ui->tuningStepCombo->addItem("10 kHz",    (unsigned int)   10000);
-    ui->tuningStepCombo->addItem("12.5 kHz",  (unsigned int)   12500);
-    ui->tuningStepCombo->addItem("100 kHz",   (unsigned int)  100000);
-    ui->tuningStepCombo->addItem("250 kHz",   (unsigned int)  250000);
-    ui->tuningStepCombo->addItem("1 MHz",     (unsigned int) 1000000);  //for 23 cm and HF 
-
-
-    ui->tuningStepCombo->setCurrentIndex(2);
-    ui->tuningStepCombo->blockSignals(false);
-
-    ui->wfthemeCombo->addItem("Jet", QCPColorGradient::gpJet);
-    ui->wfthemeCombo->addItem("Cold", QCPColorGradient::gpCold);
-    ui->wfthemeCombo->addItem("Hot", QCPColorGradient::gpHot);
-    ui->wfthemeCombo->addItem("Thermal", QCPColorGradient::gpThermal);
-    ui->wfthemeCombo->addItem("Night", QCPColorGradient::gpNight);
-    ui->wfthemeCombo->addItem("Ion", QCPColorGradient::gpIon);
-    ui->wfthemeCombo->addItem("Gray", QCPColorGradient::gpGrayscale);
-    ui->wfthemeCombo->addItem("Geography", QCPColorGradient::gpGeography);
-    ui->wfthemeCombo->addItem("Hues", QCPColorGradient::gpHues);
-    ui->wfthemeCombo->addItem("Polar", QCPColorGradient::gpPolar);
-    ui->wfthemeCombo->addItem("Spectrum", QCPColorGradient::gpSpectrum);
-    ui->wfthemeCombo->addItem("Candy", QCPColorGradient::gpCandy);
-
-    spans << "2.5k" << "5.0k" << "10k" << "25k";
-    spans << "50k" << "100k" << "250k" << "500k";
-    ui->scopeBWCombo->insertItems(0, spans);
-
-    edges << "1" << "2" << "3" << "4";
-    ui->scopeEdgeCombo->insertItems(0, edges);
-
-    ui->splitter->setHandleWidth(5);
-
-    // Set scroll wheel response (tick interval)
-    // and set arrow key response (single step)
-    ui->rfGainSlider->setTickInterval(100);
-    ui->rfGainSlider->setSingleStep(10);
-
-    ui->afGainSlider->setTickInterval(100);
-    ui->afGainSlider->setSingleStep(10);
-
-    ui->sqlSlider->setTickInterval(100);
-    ui->sqlSlider->setSingleStep(10);
-
-    ui->txPowerSlider->setTickInterval(100);
-    ui->txPowerSlider->setSingleStep(10);
-
-    ui->micGainSlider->setTickInterval(100);
-    ui->micGainSlider->setSingleStep(10);
-
-    ui->scopeRefLevelSlider->setTickInterval(50);
-    ui->scopeRefLevelSlider->setSingleStep(20);
-
     rigStatus = new QLabel(this);
     ui->statusBar->addPermanentWidget(rigStatus);
     ui->statusBar->showMessage("Connecting to rig...", 1000);
@@ -416,6 +202,146 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
     qRegisterMetaType();
     qRegisterMetaType();
 
+    rigConnections();
+
+
+
+    // Plot user interaction
+    connect(plot, SIGNAL(mouseDoubleClick(QMouseEvent*)), this, SLOT(handlePlotDoubleClick(QMouseEvent*)));
+    connect(wf, SIGNAL(mouseDoubleClick(QMouseEvent*)), this, SLOT(handleWFDoubleClick(QMouseEvent*)));
+    connect(plot, SIGNAL(mousePress(QMouseEvent*)), this, SLOT(handlePlotClick(QMouseEvent*)));
+    connect(wf, SIGNAL(mousePress(QMouseEvent*)), this, SLOT(handleWFClick(QMouseEvent*)));
+    connect(wf, SIGNAL(mouseWheel(QWheelEvent*)), this, SLOT(handleWFScroll(QWheelEvent*)));
+    connect(plot, SIGNAL(mouseWheel(QWheelEvent*)), this, SLOT(handlePlotScroll(QWheelEvent*)));
+
+
+    if (serverConfig.enabled && udp != Q_NULLPTR) {
+        // Server
+        connect(rig, SIGNAL(haveAudioData(audioPacket)), udp, SLOT(receiveAudioData(audioPacket)));
+        connect(rig, SIGNAL(haveDataForServer(QByteArray)), udp, SLOT(dataForServer(QByteArray)));
+        connect(udp, SIGNAL(haveDataFromServer(QByteArray)), rig, SLOT(dataFromServer(QByteArray)));
+    }
+
+
+    SMeterReadings.fill(0,30);
+    powerMeterReadings.fill(0,30);
+
+    pttTimer = new QTimer(this);
+    pttTimer->setInterval(180*1000); // 3 minute max transmit time in ms
+    pttTimer->setSingleShot(true);
+    connect(pttTimer, SIGNAL(timeout()), this, SLOT(handlePttLimit()));
+    amTransmitting = false;
+    ui->tuneLockChk->setChecked(false);
+    freqLock = false;
+
+    //getInitialRigState();
+    oldFreqDialVal = ui->freqDial->value();
+}
+
+wfmain::~wfmain()
+{
+    rigThread->quit();
+    rigThread->wait();
+    if (serverThread != Q_NULLPTR) {
+        serverThread->quit();
+        serverThread->wait();
+    }
+    if (rigCtl != Q_NULLPTR) {
+        delete rigCtl;
+    }
+    delete rpt;
+    delete ui;
+    delete settings;
+}
+
+void wfmain::closeEvent(QCloseEvent *event)
+{
+    // Are you sure?
+    QMessageBox::StandardButton resBtn = QMessageBox::question( this, QString("Confirm close"),
+                                                                tr("Are you sure you wish to exit?\n"),
+                                                                QMessageBox::No | QMessageBox::Yes,
+                                                                QMessageBox::Yes);
+    if (resBtn == QMessageBox::Yes) {
+        QApplication::exit();
+    } else {
+        event->ignore();
+    }
+}
+
+void wfmain::openRig()
+{
+    // This function is intended to handle opening a connection to the rig.
+    // the connection can be either serial or network,
+    // and this function is also responsible for initiating the search for a rig model and capabilities.
+    // Any errors, such as unable to open connection or unable to open port, are to be reported to the user.
+
+    //TODO: if(hasRunPreviously)
+
+    //TODO: if(useNetwork){...
+
+    // } else {
+
+    // if (prefs.fileWasNotFound) {
+    //     showRigSettings(); // rig setting dialog box for network/serial, CIV, hostname, port, baud rate, serial device, etc
+    // TODO: How do we know if the setting was loaded?
+
+
+    // TODO: Use these if they are found
+    if(!serialPortCL.isEmpty())
+    {
+        qDebug(logSystem()) << "Serial port specified by user: " << serialPortCL;
+    } else {
+        qDebug(logSystem()) << "Serial port not specified. ";
+    }
+
+    if(!hostCL.isEmpty())
+    {
+        qDebug(logSystem()) << "Remote host name specified by user: " << hostCL;
+    }
+
+    // Start rigctld
+    if (prefs.enableRigCtlD) {
+        rigCtl = new rigCtlD(this);
+
+        rigCtl->startServer(prefs.rigCtlPort);
+        connect(this, SIGNAL(sendRigCaps(rigCapabilities)), rigCtl, SLOT(receiveRigCaps(rigCapabilities)));
+    }
+
+    makeRig();
+
+    if (prefs.enableLAN)
+    {
+        ui->lanEnableBtn->setChecked(true);
+        emit sendCommSetup(prefs.radioCIVAddr, udpPrefs,prefs.virtualSerialPort);
+    } else {
+        ui->serialEnableBtn->setChecked(true);
+        if( (prefs.serialPortRadio.toLower() == QString("auto")) && (serialPortCL.isEmpty()))
+        {
+            findSerialPort();
+
+        } else {
+            if(serialPortCL.isEmpty())
+            {
+                serialPortRig = prefs.serialPortRadio;
+            } else {
+                serialPortRig = serialPortCL;
+            }
+        }
+
+        // Here, the radioCIVAddr is being set from a default preference, which is for the 7300.
+        // However, we will not use it initially. OTOH, if it is set explicitedly to a value in the prefs,
+        // then we skip auto detection.
+        emit sendCommSetup(prefs.radioCIVAddr, serialPortRig, prefs.serialPortBaud,prefs.virtualSerialPort);
+    }
+
+    ui->statusBar->showMessage(QString("Connecting to rig using serial port ").append(serialPortRig), 1000);
+
+}
+
+void wfmain::rigConnections()
+{
+    // from line 413..
+
     connect(this, SIGNAL(sendPowerOn()), rig, SLOT(powerOn()));
     connect(this, SIGNAL(sendPowerOff()), rig, SLOT(powerOff()));
 
@@ -558,178 +484,35 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
     connect(cal, SIGNAL(setRefAdjustCourse(unsigned char)), rig, SLOT(setRefAdjustCourse(unsigned char)));
     connect(cal, SIGNAL(setRefAdjustFine(unsigned char)), rig, SLOT(setRefAdjustFine(unsigned char)));
 
-
-    // Plot user interaction
-    connect(plot, SIGNAL(mouseDoubleClick(QMouseEvent*)), this, SLOT(handlePlotDoubleClick(QMouseEvent*)));
-    connect(wf, SIGNAL(mouseDoubleClick(QMouseEvent*)), this, SLOT(handleWFDoubleClick(QMouseEvent*)));
-    connect(plot, SIGNAL(mousePress(QMouseEvent*)), this, SLOT(handlePlotClick(QMouseEvent*)));
-    connect(wf, SIGNAL(mousePress(QMouseEvent*)), this, SLOT(handleWFClick(QMouseEvent*)));
-    connect(wf, SIGNAL(mouseWheel(QWheelEvent*)), this, SLOT(handleWFScroll(QWheelEvent*)));
-    connect(plot, SIGNAL(mouseWheel(QWheelEvent*)), this, SLOT(handlePlotScroll(QWheelEvent*)));
-
-
-    if (serverConfig.enabled && udp != Q_NULLPTR) {
-        // Server
-        connect(rig, SIGNAL(haveAudioData(audioPacket)), udp, SLOT(receiveAudioData(audioPacket)));
-        connect(rig, SIGNAL(haveDataForServer(QByteArray)), udp, SLOT(dataForServer(QByteArray)));
-        connect(udp, SIGNAL(haveDataFromServer(QByteArray)), rig, SLOT(dataFromServer(QByteArray)));
-    }
-
-    ui->plot->addGraph(); // primary
-    ui->plot->addGraph(0, 0); // secondary, peaks, same axis as first?
-    ui->waterfall->addGraph();
-
-
-
-    colorMap = new QCPColorMap(wf->xAxis, wf->yAxis);
-    colorMapData = NULL;
-
-#if QCUSTOMPLOT_VERSION < 0x020001
-    wf->addPlottable(colorMap);
-#endif
-
-    colorScale = new QCPColorScale(wf);
-
-    ui->tabWidget->setCurrentIndex(0);
-
-    QColor color(20+200/4.0*1,70*(1.6-1/4.0), 150, 150);
-    plot->graph(1)->setLineStyle(QCPGraph::lsLine);
-    plot->graph(1)->setPen(QPen(color.lighter(200)));
-    plot->graph(1)->setBrush(QBrush(color));
-
-    drawPeaks = false;
-
-    SMeterReadings.fill(0,30);
-    powerMeterReadings.fill(0,30);
-
-
-    ui->freqMhzLineEdit->setValidator( new QDoubleValidator(0, 100, 6, this));
-    ui->controlPortTxt->setValidator(new QIntValidator(this));
-
-    pttTimer = new QTimer(this);
-    pttTimer->setInterval(180*1000); // 3 minute max transmit time in ms
-    pttTimer->setSingleShot(true);
-    connect(pttTimer, SIGNAL(timeout()), this, SLOT(handlePttLimit()));
-    amTransmitting = false;
-    ui->tuneLockChk->setChecked(false);
-    freqLock = false;
-
-
-    freqIndicatorLine->start->setCoords(0.5,0);
-    freqIndicatorLine->end->setCoords(0.5,160);
-
-    qDebug(logSystem()) << "Running with debugging options enabled.";
-#ifdef QT_DEBUG
-    ui->debugBtn->setVisible(true);
-    ui->satOpsBtn->setVisible(true);
-#else
-    ui->debugBtn->setVisible(false);
-    ui->satOpsBtn->setVisible(false);
-#endif
-
-    // Initial state of UI:
-    ui->fullScreenChk->setChecked(prefs.useFullScreen);
-    on_fullScreenChk_clicked(prefs.useFullScreen);
-
-    ui->useDarkThemeChk->setChecked(prefs.useDarkMode);
-    on_useDarkThemeChk_clicked(prefs.useDarkMode);
-
-    ui->useSystemThemeChk->setChecked(prefs.useSystemTheme);
-    on_useSystemThemeChk_clicked(prefs.useSystemTheme);
-
-    ui->drawPeakChk->setChecked(prefs.drawPeaks);
-    on_drawPeakChk_clicked(prefs.drawPeaks);
-    drawPeaks = prefs.drawPeaks;
-
-    //getInitialRigState();
-    oldFreqDialVal = ui->freqDial->value();
 }
 
-wfmain::~wfmain()
+//void wfmain::removeRigConnections()
+//{
+
+//}
+
+void wfmain::makeRig()
 {
-    rigThread->quit();
-    rigThread->wait();
-    if (serverThread != Q_NULLPTR) {
-        serverThread->quit();
-        serverThread->wait();
-    }
-    if (rigCtl != Q_NULLPTR) {
-        delete rigCtl;
-    }
-    delete rpt;
-    delete ui;
-    delete settings;
-}
-
-void wfmain::closeEvent(QCloseEvent *event)
-{
-    // Are you sure?
-    QMessageBox::StandardButton resBtn = QMessageBox::question( this, QString("Confirm close"),
-                                                                tr("Are you sure you wish to exit?\n"),
-                                                                QMessageBox::No | QMessageBox::Yes,
-                                                                QMessageBox::Yes);
-    if (resBtn == QMessageBox::Yes) {
-        QApplication::exit();
-    } else {
-        event->ignore();
-    }
-}
-
-void wfmain::openRig()
-{
-    // This function is intended to handle opening a connection to the rig.
-    // the connection can be either serial or network,
-    // and this function is also responsible for initiating the search for a rig model and capabilities.
-    // Any errors, such as unable to open connection or unable to open port, are to be reported to the user.
-
-    //TODO: if(hasRunPreviously)
-
-    //TODO: if(useNetwork){...
-
-    // } else {
-
-    // if (prefs.fileWasNotFound) {
-    //     showRigSettings(); // rig setting dialog box for network/serial, CIV, hostname, port, baud rate, serial device, etc
-    // TODO: How do we know if the setting was loaded?
-
-
-    // TODO: Use these if they are found
-    if(!serialPortCL.isEmpty())
-    {
-        qDebug(logSystem()) << "Serial port specified by user: " << serialPortCL;
-    } else {
-        qDebug(logSystem()) << "Serial port not specified. ";
-    }
-
-    if(!hostCL.isEmpty())
-    {
-        qDebug(logSystem()) << "Remote host name specified by user: " << hostCL;
-    }
-
-    // Start rigctld
-    if (prefs.enableRigCtlD) {
-        rigCtl = new rigCtlD(this);
-
-        rigCtl->startServer(prefs.rigCtlPort);
-        connect(this, SIGNAL(sendRigCaps(rigCapabilities)), rigCtl, SLOT(receiveRigCaps(rigCapabilities)));
-    }
-
     if (rigThread == Q_NULLPTR)
     {
         rig = new rigCommander();
         rigThread = new QThread(this);
 
+        // Thread:
         rig->moveToThread(rigThread);
-
         connect(rigThread, SIGNAL(started()), rig, SLOT(process()));
         connect(rigThread, SIGNAL(finished()), rig, SLOT(deleteLater()));
         rigThread->start();
+
+        // Rig status and Errors:
         connect(rig, SIGNAL(haveSerialPortError(QString, QString)), this, SLOT(receiveSerialPortError(QString, QString)));
         connect(rig, SIGNAL(haveStatusUpdate(QString)), this, SLOT(receiveStatusUpdate(QString)));
-        
+
+        // Rig comm setup:
         connect(this, SIGNAL(sendCommSetup(unsigned char, udpPreferences, QString)), rig, SLOT(commSetup(unsigned char, udpPreferences, QString)));
         connect(this, SIGNAL(sendCommSetup(unsigned char, QString, quint32,QString)), rig, SLOT(commSetup(unsigned char, QString, quint32,QString)));
 
+
         connect(rig, SIGNAL(haveBaudRate(quint32)), this, SLOT(receiveBaudRate(quint32)));
 
         connect(this, SIGNAL(sendCloseComm()), rig, SLOT(closeComm()));
@@ -745,84 +528,78 @@ void wfmain::openRig()
             connect(rigCtl, SIGNAL(setPTT(bool)), rig, SLOT(setPTT(bool)));
         }
     }
+}
 
-    if (prefs.enableLAN)
+void wfmain::removeRig()
+{
+    if (rigThread != Q_NULLPTR)
     {
-        ui->lanEnableBtn->setChecked(true);
-        emit sendCommSetup(prefs.radioCIVAddr, udpPrefs,prefs.virtualSerialPort);
-    } else {
-        ui->serialEnableBtn->setChecked(true);
-        if( (prefs.serialPortRadio.toLower() == QString("auto")) && (serialPortCL.isEmpty()))
-        {
-            // Find the ICOM
-            // qInfo(logSystem()) << "Searching for serial port...";
-            QDirIterator it73("/dev/serial/by-id", QStringList() << "*IC-7300*", QDir::Files, QDirIterator::Subdirectories);
-            QDirIterator it97("/dev/serial", QStringList() << "*IC-9700*A*", QDir::Files, QDirIterator::Subdirectories);
-            QDirIterator it785x("/dev/serial", QStringList() << "*IC-785*A*", QDir::Files, QDirIterator::Subdirectories);
-            QDirIterator it705("/dev/serial", QStringList() << "*IC-705*A", QDir::Files, QDirIterator::Subdirectories);
-            QDirIterator it7610("/dev/serial", QStringList() << "*IC-7610*A", QDir::Files, QDirIterator::Subdirectories);
-            QDirIterator itR8600("/dev/serial", QStringList() << "*IC-R8600*A", QDir::Files, QDirIterator::Subdirectories);
-            QDirIterator itTest("/tmp/test", QStringList() << "*radio*", QDir::NoFilter, QDirIterator::Subdirectories);
-
-            qDebug() << "test iterator isEmpty: " << itTest.filePath().isEmpty();
-
-            if(!it73.filePath().isEmpty())
-            {
-                // IC-7300
-                serialPortRig = it73.filePath(); // first
-            } else if(!it97.filePath().isEmpty())
-            {
-                // IC-9700
-                serialPortRig = it97.filePath();
-            } else if(!it785x.filePath().isEmpty())
-            {
-                // IC-785x
-                serialPortRig = it785x.filePath();
-            } else if(!it705.filePath().isEmpty())
-            {
-                // IC-705
-                serialPortRig = it705.filePath();
-            } else if(!it7610.filePath().isEmpty())
-            {
-                // IC-7610
-                serialPortRig = it7610.filePath();
-            } else if(!itR8600.filePath().isEmpty())
-            {
-                // IC-R8600
-                serialPortRig = itR8600.filePath();
-            } else {
-                //fall back:
-
-
-                qInfo(logSystem()) << "Could not find Icom serial port. Falling back to OS default. Use --port to specify, or modify preferences.";
-#ifdef Q_OS_MAC
-                serialPortRig = QString("/dev/tty.SLAB_USBtoUART");
-#endif
-#ifdef Q_OS_LINUX
-                serialPortRig = QString("/dev/ttyUSB0");
-#endif
-#ifdef Q_OS_WIN
-                serialPortRig = QString("COM1");
-#endif
-            }
-
-        } else {
-            if(serialPortCL.isEmpty())
-            {
-                serialPortRig = prefs.serialPortRadio;
-            } else {
-                serialPortRig = serialPortCL;
-            }
+        if (rigCtl != Q_NULLPTR) {
+            rigCtl->disconnect();
         }
+        rigThread->disconnect();
+
+        rig->disconnect();
+
+        delete rigThread;
+        delete rig;
 
-        // Here, the radioCIVAddr is being set from a default preference, which is for the 7300.
-        // However, we will not use it initially. OTOH, if it is set explicitedly to a value in the prefs,
-        // then we skip auto detection.
-        emit sendCommSetup(prefs.radioCIVAddr, serialPortRig, prefs.serialPortBaud,prefs.virtualSerialPort);
     }
 
-    ui->statusBar->showMessage(QString("Connecting to rig using serial port ").append(serialPortRig), 1000);
+}
 
+
+void wfmain::findSerialPort()
+{
+    // Find the ICOM radio connected, or, if none, fall back to OS default.
+    // qInfo(logSystem()) << "Searching for serial port...";
+    QDirIterator it73("/dev/serial/by-id", QStringList() << "*IC-7300*", QDir::Files, QDirIterator::Subdirectories);
+    QDirIterator it97("/dev/serial", QStringList() << "*IC-9700*A*", QDir::Files, QDirIterator::Subdirectories);
+    QDirIterator it785x("/dev/serial", QStringList() << "*IC-785*A*", QDir::Files, QDirIterator::Subdirectories);
+    QDirIterator it705("/dev/serial", QStringList() << "*IC-705*A", QDir::Files, QDirIterator::Subdirectories);
+    QDirIterator it7610("/dev/serial", QStringList() << "*IC-7610*A", QDir::Files, QDirIterator::Subdirectories);
+    QDirIterator itR8600("/dev/serial", QStringList() << "*IC-R8600*A", QDir::Files, QDirIterator::Subdirectories);
+    QDirIterator itTest("/tmp/test", QStringList() << "*radio*", QDir::NoFilter, QDirIterator::Subdirectories);
+
+    qDebug() << "test iterator isEmpty: " << itTest.filePath().isEmpty();
+
+    if(!it73.filePath().isEmpty())
+    {
+        // IC-7300
+        serialPortRig = it73.filePath(); // first
+    } else if(!it97.filePath().isEmpty())
+    {
+        // IC-9700
+        serialPortRig = it97.filePath();
+    } else if(!it785x.filePath().isEmpty())
+    {
+        // IC-785x
+        serialPortRig = it785x.filePath();
+    } else if(!it705.filePath().isEmpty())
+    {
+        // IC-705
+        serialPortRig = it705.filePath();
+    } else if(!it7610.filePath().isEmpty())
+    {
+        // IC-7610
+        serialPortRig = it7610.filePath();
+    } else if(!itR8600.filePath().isEmpty())
+    {
+        // IC-R8600
+        serialPortRig = itR8600.filePath();
+    } else {
+        //fall back:
+        qInfo(logSystem()) << "Could not find Icom serial port. Falling back to OS default. Use --port to specify, or modify preferences.";
+#ifdef Q_OS_MAC
+        serialPortRig = QString("/dev/tty.SLAB_USBtoUART");
+#endif
+#ifdef Q_OS_LINUX
+        serialPortRig = QString("/dev/ttyUSB0");
+#endif
+#ifdef Q_OS_WIN
+        serialPortRig = QString("COM1");
+#endif
+    }
 }
 
 void wfmain::receiveCommReady()
@@ -902,6 +679,293 @@ void wfmain::receiveStatusUpdate(QString text)
     this->rigStatus->setText(text);
 }
 
+void wfmain::setupPlots()
+{
+
+// Line 290--
+    plot = ui->plot; // rename it waterfall.
+    wf = ui->waterfall;
+
+    freqIndicatorLine = new QCPItemLine(plot);
+    freqIndicatorLine->setAntialiased(true);
+    freqIndicatorLine->setPen(QPen(Qt::blue));
+//
+
+    ui->plot->addGraph(); // primary
+    ui->plot->addGraph(0, 0); // secondary, peaks, same axis as first?
+    ui->waterfall->addGraph();
+
+    colorMap = new QCPColorMap(wf->xAxis, wf->yAxis);
+    colorMapData = NULL;
+
+#if QCUSTOMPLOT_VERSION < 0x020001
+    wf->addPlottable(colorMap);
+#endif
+
+    colorScale = new QCPColorScale(wf);
+
+    ui->tabWidget->setCurrentIndex(0);
+
+    QColor color(20+200/4.0*1,70*(1.6-1/4.0), 150, 150);
+    plot->graph(1)->setLineStyle(QCPGraph::lsLine);
+    plot->graph(1)->setPen(QPen(color.lighter(200)));
+    plot->graph(1)->setBrush(QBrush(color));
+
+    freqIndicatorLine->start->setCoords(0.5,0);
+    freqIndicatorLine->end->setCoords(0.5,160);
+}
+
+void wfmain::setupMainUI()
+{
+    ui->bandStkLastUsedBtn->setVisible(false);
+    ui->bandStkVoiceBtn->setVisible(false);
+    ui->bandStkDataBtn->setVisible(false);
+    ui->bandStkCWBtn->setVisible(false);
+
+    ui->baudRateCombo->insertItem(0, QString("115200"), 115200);
+    ui->baudRateCombo->insertItem(1, QString("57600"), 57600);
+    ui->baudRateCombo->insertItem(2, QString("38400"), 38400);
+    ui->baudRateCombo->insertItem(3, QString("28800"), 28800);
+    ui->baudRateCombo->insertItem(4, QString("19200"), 19200);
+    ui->baudRateCombo->insertItem(5, QString("9600"), 9600);
+    ui->baudRateCombo->insertItem(6, QString("4800"), 4800);
+    ui->baudRateCombo->insertItem(7, QString("2400"), 2400);
+    ui->baudRateCombo->insertItem(8, QString("1200"), 1200);
+    ui->baudRateCombo->insertItem(9, QString("300"), 300);
+
+    ui->spectrumModeCombo->addItem("Center", (spectrumMode)spectModeCenter);
+    ui->spectrumModeCombo->addItem("Fixed", (spectrumMode)spectModeFixed);
+    ui->spectrumModeCombo->addItem("Scroll-C", (spectrumMode)spectModeScrollC);
+    ui->spectrumModeCombo->addItem("Scroll-F", (spectrumMode)spectModeScrollF);
+
+    ui->modeSelectCombo->addItem("LSB",  0x00);
+    ui->modeSelectCombo->addItem("USB",  0x01);
+    ui->modeSelectCombo->addItem("FM",   0x05);
+    ui->modeSelectCombo->addItem("AM",   0x02);
+    ui->modeSelectCombo->addItem("CW",   0x03);
+    ui->modeSelectCombo->addItem("CW-R", 0x07);
+    ui->modeSelectCombo->addItem("RTTY", 0x04);
+    ui->modeSelectCombo->addItem("RTTY-R", 0x08);
+
+
+    ui->modeFilterCombo->addItem("1", 1);
+    ui->modeFilterCombo->addItem("2", 2);
+    ui->modeFilterCombo->addItem("3", 3);
+    ui->modeFilterCombo->addItem("Setup...", 99);
+
+    ui->tuningStepCombo->blockSignals(true);
+
+    ui->tuningStepCombo->addItem("1 Hz",      (unsigned int)       1);
+    ui->tuningStepCombo->addItem("10 Hz",     (unsigned int)      10);
+    ui->tuningStepCombo->addItem("100 Hz",    (unsigned int)     100);
+    ui->tuningStepCombo->addItem("1 kHz",     (unsigned int)    1000);
+    ui->tuningStepCombo->addItem("2.5 kHz",   (unsigned int)    2500);
+    ui->tuningStepCombo->addItem("5 kHz",     (unsigned int)    5000);
+    ui->tuningStepCombo->addItem("6.125 kHz", (unsigned int)    6125);	// PMR
+    ui->tuningStepCombo->addItem("8.333 kHz", (unsigned int)    8333);	// airband stepsize
+    ui->tuningStepCombo->addItem("9 kHz",     (unsigned int)    9000);	// European medium wave stepsize
+    ui->tuningStepCombo->addItem("10 kHz",    (unsigned int)   10000);
+    ui->tuningStepCombo->addItem("12.5 kHz",  (unsigned int)   12500);
+    ui->tuningStepCombo->addItem("100 kHz",   (unsigned int)  100000);
+    ui->tuningStepCombo->addItem("250 kHz",   (unsigned int)  250000);
+    ui->tuningStepCombo->addItem("1 MHz",     (unsigned int) 1000000);  //for 23 cm and HF
+
+
+    ui->tuningStepCombo->setCurrentIndex(2);
+    ui->tuningStepCombo->blockSignals(false);
+
+    ui->wfthemeCombo->addItem("Jet", QCPColorGradient::gpJet);
+    ui->wfthemeCombo->addItem("Cold", QCPColorGradient::gpCold);
+    ui->wfthemeCombo->addItem("Hot", QCPColorGradient::gpHot);
+    ui->wfthemeCombo->addItem("Thermal", QCPColorGradient::gpThermal);
+    ui->wfthemeCombo->addItem("Night", QCPColorGradient::gpNight);
+    ui->wfthemeCombo->addItem("Ion", QCPColorGradient::gpIon);
+    ui->wfthemeCombo->addItem("Gray", QCPColorGradient::gpGrayscale);
+    ui->wfthemeCombo->addItem("Geography", QCPColorGradient::gpGeography);
+    ui->wfthemeCombo->addItem("Hues", QCPColorGradient::gpHues);
+    ui->wfthemeCombo->addItem("Polar", QCPColorGradient::gpPolar);
+    ui->wfthemeCombo->addItem("Spectrum", QCPColorGradient::gpSpectrum);
+    ui->wfthemeCombo->addItem("Candy", QCPColorGradient::gpCandy);
+
+    spans << "2.5k" << "5.0k" << "10k" << "25k";
+    spans << "50k" << "100k" << "250k" << "500k";
+    ui->scopeBWCombo->insertItems(0, spans);
+
+    edges << "1" << "2" << "3" << "4";
+    ui->scopeEdgeCombo->insertItems(0, edges);
+
+    ui->splitter->setHandleWidth(5);
+
+    // Set scroll wheel response (tick interval)
+    // and set arrow key response (single step)
+    ui->rfGainSlider->setTickInterval(100);
+    ui->rfGainSlider->setSingleStep(10);
+
+    ui->afGainSlider->setTickInterval(100);
+    ui->afGainSlider->setSingleStep(10);
+
+    ui->sqlSlider->setTickInterval(100);
+    ui->sqlSlider->setSingleStep(10);
+
+    ui->txPowerSlider->setTickInterval(100);
+    ui->txPowerSlider->setSingleStep(10);
+
+    ui->micGainSlider->setTickInterval(100);
+    ui->micGainSlider->setSingleStep(10);
+
+    ui->scopeRefLevelSlider->setTickInterval(50);
+    ui->scopeRefLevelSlider->setSingleStep(20);
+
+    ui->freqMhzLineEdit->setValidator( new QDoubleValidator(0, 100, 6, this));
+    ui->controlPortTxt->setValidator(new QIntValidator(this));
+
+    qDebug(logSystem()) << "Running with debugging options enabled.";
+#ifdef QT_DEBUG
+    ui->debugBtn->setVisible(true);
+    ui->satOpsBtn->setVisible(true);
+#else
+    ui->debugBtn->setVisible(false);
+    ui->satOpsBtn->setVisible(false);
+#endif
+
+}
+
+void wfmain::setUIToPrefs()
+{
+    ui->fullScreenChk->setChecked(prefs.useFullScreen);
+    on_fullScreenChk_clicked(prefs.useFullScreen);
+
+    ui->useDarkThemeChk->setChecked(prefs.useDarkMode);
+    on_useDarkThemeChk_clicked(prefs.useDarkMode);
+
+    ui->useSystemThemeChk->setChecked(prefs.useSystemTheme);
+    on_useSystemThemeChk_clicked(prefs.useSystemTheme);
+
+    ui->drawPeakChk->setChecked(prefs.drawPeaks);
+    on_drawPeakChk_clicked(prefs.drawPeaks);
+    drawPeaks = prefs.drawPeaks;
+
+}
+
+void wfmain::setupKeyShortcuts()
+{
+    keyF1 = new QShortcut(this);
+    keyF1->setKey(Qt::Key_F1);
+    connect(keyF1, SIGNAL(activated()), this, SLOT(shortcutF1()));
+
+    keyF2 = new QShortcut(this);
+    keyF2->setKey(Qt::Key_F2);
+    connect(keyF2, SIGNAL(activated()), this, SLOT(shortcutF2()));
+
+    keyF3 = new QShortcut(this);
+    keyF3->setKey(Qt::Key_F3);
+    connect(keyF3, SIGNAL(activated()), this, SLOT(shortcutF3()));
+
+    keyF4 = new QShortcut(this);
+    keyF4->setKey(Qt::Key_F4);
+    connect(keyF4, SIGNAL(activated()), this, SLOT(shortcutF4()));
+
+    keyF5 = new QShortcut(this);
+    keyF5->setKey(Qt::Key_F5);
+    connect(keyF5, SIGNAL(activated()), this, SLOT(shortcutF5()));
+
+    keyF6 = new QShortcut(this);
+    keyF6->setKey(Qt::Key_F6);
+    connect(keyF6, SIGNAL(activated()), this, SLOT(shortcutF6()));
+
+    keyF7 = new QShortcut(this);
+    keyF7->setKey(Qt::Key_F7);
+    connect(keyF7, SIGNAL(activated()), this, SLOT(shortcutF7()));
+
+    keyF8 = new QShortcut(this);
+    keyF8->setKey(Qt::Key_F8);
+    connect(keyF8, SIGNAL(activated()), this, SLOT(shortcutF8()));
+
+    keyF9 = new QShortcut(this);
+    keyF9->setKey(Qt::Key_F9);
+    connect(keyF9, SIGNAL(activated()), this, SLOT(shortcutF9()));
+
+    keyF10 = new QShortcut(this);
+    keyF10->setKey(Qt::Key_F10);
+    connect(keyF10, SIGNAL(activated()), this, SLOT(shortcutF10()));
+
+    keyF11 = new QShortcut(this);
+    keyF11->setKey(Qt::Key_F11);
+    connect(keyF11, SIGNAL(activated()), this, SLOT(shortcutF11()));
+
+    keyF12 = new QShortcut(this);
+    keyF12->setKey(Qt::Key_F12);
+    connect(keyF12, SIGNAL(activated()), this, SLOT(shortcutF12()));
+
+    keyControlT = new QShortcut(this);
+    keyControlT->setKey(Qt::CTRL + Qt::Key_T);
+    connect(keyControlT, SIGNAL(activated()), this, SLOT(shortcutControlT()));
+
+    keyControlR = new QShortcut(this);
+    keyControlR->setKey(Qt::CTRL + Qt::Key_R);
+    connect(keyControlR, SIGNAL(activated()), this, SLOT(shortcutControlR()));
+
+    keyControlI = new QShortcut(this);
+    keyControlI->setKey(Qt::CTRL + Qt::Key_I);
+    connect(keyControlI, SIGNAL(activated()), this, SLOT(shortcutControlI()));
+
+    keyControlU = new QShortcut(this);
+    keyControlU->setKey(Qt::CTRL + Qt::Key_U);
+    connect(keyControlU, SIGNAL(activated()), this, SLOT(shortcutControlU()));
+
+    keyStar = new QShortcut(this);
+    keyStar->setKey(Qt::Key_Asterisk);
+    connect(keyStar, SIGNAL(activated()), this, SLOT(shortcutStar()));
+
+    keySlash = new QShortcut(this);
+    keySlash->setKey(Qt::Key_Slash);
+    connect(keySlash, SIGNAL(activated()), this, SLOT(shortcutSlash()));
+
+    keyMinus = new QShortcut(this);
+    keyMinus->setKey(Qt::Key_Minus);
+    connect(keyMinus, SIGNAL(activated()), this, SLOT(shortcutMinus()));
+
+    keyPlus = new QShortcut(this);
+    keyPlus->setKey(Qt::Key_Plus);
+    connect(keyPlus, SIGNAL(activated()), this, SLOT(shortcutPlus()));
+
+    keyShiftMinus = new QShortcut(this);
+    keyShiftMinus->setKey(Qt::SHIFT + Qt::Key_Minus);
+    connect(keyShiftMinus, SIGNAL(activated()), this, SLOT(shortcutShiftMinus()));
+
+    keyShiftPlus = new QShortcut(this);
+    keyShiftPlus->setKey(Qt::SHIFT + Qt::Key_Plus);
+    connect(keyShiftPlus, SIGNAL(activated()), this, SLOT(shortcutShiftPlus()));
+
+    keyControlMinus = new QShortcut(this);
+    keyControlMinus->setKey(Qt::CTRL + Qt::Key_Minus);
+    connect(keyControlMinus, SIGNAL(activated()), this, SLOT(shortcutControlMinus()));
+
+    keyControlPlus = new QShortcut(this);
+    keyControlPlus->setKey(Qt::CTRL + Qt::Key_Plus);
+    connect(keyControlPlus, SIGNAL(activated()), this, SLOT(shortcutControlPlus()));
+
+    keyQuit = new QShortcut(this);
+    keyQuit->setKey(Qt::CTRL + Qt::Key_Q);
+    connect(keyQuit, SIGNAL(activated()), this, SLOT(on_exitBtn_clicked()));
+
+    keyPageUp = new QShortcut(this);
+    keyPageUp->setKey(Qt::Key_PageUp);
+    connect(keyPageUp, SIGNAL(activated()), this, SLOT(shortcutPageUp()));
+
+    keyPageDown = new QShortcut(this);
+    keyPageDown->setKey(Qt::Key_PageDown);
+    connect(keyPageDown, SIGNAL(activated()), this, SLOT(shortcutPageDown()));
+
+    keyF = new QShortcut(this);
+    keyF->setKey(Qt::Key_F);
+    connect(keyF, SIGNAL(activated()), this, SLOT(shortcutF()));
+
+    keyM = new QShortcut(this);
+    keyM->setKey(Qt::Key_M);
+    connect(keyM, SIGNAL(activated()), this, SLOT(shortcutM()));
+}
+
 void wfmain::setDefPrefs()
 {
     defPrefs.useFullScreen = false;
diff --git a/wfmain.h b/wfmain.h
index 06c1f341..cc8de8ae 100644
--- a/wfmain.h
+++ b/wfmain.h
@@ -513,6 +513,15 @@ private:
     QTimer * periodicPollingTimer;
     QTimer * pttTimer;
 
+    void setupPlots();
+    void makeRig();
+    void rigConnections();
+    void removeRig();
+    void findSerialPort();
+
+    void setupKeyShortcuts();
+    void setupMainUI();
+    void setUIToPrefs();
 
     QStringList modes;
     int currentModeIndex;

From f600212124d48780e079658f4e79c2bb553f75f1 Mon Sep 17 00:00:00 2001
From: Elliott Liggett 
Date: Fri, 28 May 2021 21:01:29 -0700
Subject: [PATCH 32/47] More arranging of the chairs. Also fixed a minor bug
 that prevented the "Manual" serial device entry on my system.

---
 wfmain.cpp | 177 +++++++++++++++++++++++++++--------------------------
 wfmain.h   |   2 +
 2 files changed, 93 insertions(+), 86 deletions(-)

diff --git a/wfmain.cpp b/wfmain.cpp
index 8061abbe..8182a4c0 100644
--- a/wfmain.cpp
+++ b/wfmain.cpp
@@ -32,6 +32,14 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
     connect(srv, SIGNAL(serverConfig(SERVERCONFIG, bool)), this, SLOT(serverConfigRequested(SERVERCONFIG, bool)));
        
     qRegisterMetaType(); // Needs to be registered early.
+    qRegisterMetaType();
+    qRegisterMetaType();
+    qRegisterMetaType();
+    qRegisterMetaType();
+    qRegisterMetaType();
+    qRegisterMetaType();
+    qRegisterMetaType();
+    qRegisterMetaType();
 
     haveRigCaps = false;
 
@@ -39,63 +47,9 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
 
     setupMainUI();
 
-    // Enumerate audio devices, need to do before settings are loaded.
-    const auto audioOutputs = QAudioDeviceInfo::availableDevices(QAudio::AudioOutput);
-    for (const QAudioDeviceInfo& deviceInfo : audioOutputs) {
-        ui->audioOutputCombo->addItem(deviceInfo.deviceName(),QVariant::fromValue(deviceInfo));
-    }
-    const auto audioInputs = QAudioDeviceInfo::availableDevices(QAudio::AudioInput);
-    for (const QAudioDeviceInfo& deviceInfo : audioInputs) {
-        ui->audioInputCombo->addItem(deviceInfo.deviceName(),QVariant::fromValue(deviceInfo));
-    }
-
-    ui->serialDeviceListCombo->blockSignals(true);
-    ui->serialDeviceListCombo->addItem("Auto", 0);
-    int i = 0;
-    foreach(const QSerialPortInfo & serialPortInfo, QSerialPortInfo::availablePorts())
-    {
-        portList.append(serialPortInfo.portName());
-#if defined(Q_OS_LINUX) || defined(Q_OS_MAC)
-        ui->serialDeviceListCombo->addItem(QString("/dev/")+serialPortInfo.portName(), i++);
-        ui->serialDeviceListCombo->addItem("Manual...", 256);
-#else
-        ui->serialDeviceListCombo->addItem(serialPortInfo.portName(), i++);
-#endif
-    }
-    ui->serialDeviceListCombo->blockSignals(false);
-
-    ui->vspCombo->blockSignals(true);
-
-#ifdef Q_OS_WIN
-    ui->vspCombo->addItem(QString("None"), i++);
-
-    foreach(const QSerialPortInfo & serialPortInfo, QSerialPortInfo::availablePorts())
-    {
-        ui->vspCombo->addItem(serialPortInfo.portName());
-    }
-#else
-    // Provide reasonable names for the symbolic link to the pty device
-#ifdef Q_OS_MAC
-    QString vspName = QStandardPaths::standardLocations(QStandardPaths::DownloadLocation)[0] + "/rig-pty";
-#else
-    QString vspName=QDir::homePath()+"/rig-pty";
-#endif
-    for (i=1;i<8;i++) {
-        ui->vspCombo->addItem(vspName + QString::number(i));
-
-        if (QFile::exists(vspName+QString::number(i))) {
-            auto * model = qobject_cast(ui->vspCombo->model());
-            auto *item = model->item(ui->vspCombo->count()-1);
-            item->setEnabled(false);
-        }
-    }
-    ui->vspCombo->addItem(vspName+QString::number(i));
-    ui->vspCombo->addItem(QString("None"), i++);
-
-#endif
-    ui->vspCombo->setEditable(true);
-    ui->vspCombo->blockSignals(false);
+    setSerialDevicesUI();
 
+    setAudioDevicesUI();
 
     setDefaultColors(); // set of UI colors with defaults populated
     setDefPrefs(); // other default options
@@ -159,22 +113,6 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
 
     }
 
-    rigStatus = new QLabel(this);
-    ui->statusBar->addPermanentWidget(rigStatus);
-    ui->statusBar->showMessage("Connecting to rig...", 1000);
-
-    pttLed = new QLedLabel(this);
-    ui->statusBar->addPermanentWidget(pttLed);
-    pttLed->setState(QLedLabel::State::StateOk);
-
-    connectedLed = new QLedLabel(this);
-    ui->statusBar->addPermanentWidget(connectedLed);
-
-    rigName = new QLabel(this);
-    ui->statusBar->addPermanentWidget(rigName);
-    rigName->setText("NONE");
-    rigName->setFixedWidth(50);
-
     delayedCmdIntervalLAN_ms = 10; // interval for regular delayed commands, including initial rig/UI state queries
     delayedCmdIntervalSerial_ms = 100; // interval for regular delayed commands, including initial rig/UI state queries
     delayedCmdStartupInterval_ms = 250; // interval for rigID polling
@@ -193,19 +131,8 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
 
     openRig();
 
-    qRegisterMetaType();
-    qRegisterMetaType();
-    qRegisterMetaType();
-    qRegisterMetaType();
-    qRegisterMetaType();
-    qRegisterMetaType();
-    qRegisterMetaType();
-    qRegisterMetaType();
-
     rigConnections();
 
-
-
     // Plot user interaction
     connect(plot, SIGNAL(mouseDoubleClick(QMouseEvent*)), this, SLOT(handlePlotDoubleClick(QMouseEvent*)));
     connect(wf, SIGNAL(mouseDoubleClick(QMouseEvent*)), this, SLOT(handleWFDoubleClick(QMouseEvent*)));
@@ -340,8 +267,6 @@ void wfmain::openRig()
 
 void wfmain::rigConnections()
 {
-    // from line 413..
-
     connect(this, SIGNAL(sendPowerOn()), rig, SLOT(powerOn()));
     connect(this, SIGNAL(sendPowerOff()), rig, SLOT(powerOff()));
 
@@ -828,6 +753,21 @@ void wfmain::setupMainUI()
     ui->satOpsBtn->setVisible(false);
 #endif
 
+    rigStatus = new QLabel(this);
+    ui->statusBar->addPermanentWidget(rigStatus);
+    ui->statusBar->showMessage("Connecting to rig...", 1000);
+
+    pttLed = new QLedLabel(this);
+    ui->statusBar->addPermanentWidget(pttLed);
+    pttLed->setState(QLedLabel::State::StateOk);
+
+    connectedLed = new QLedLabel(this);
+    ui->statusBar->addPermanentWidget(connectedLed);
+
+    rigName = new QLabel(this);
+    ui->statusBar->addPermanentWidget(rigName);
+    rigName->setText("NONE");
+    rigName->setFixedWidth(50);
 }
 
 void wfmain::setUIToPrefs()
@@ -844,9 +784,74 @@ void wfmain::setUIToPrefs()
     ui->drawPeakChk->setChecked(prefs.drawPeaks);
     on_drawPeakChk_clicked(prefs.drawPeaks);
     drawPeaks = prefs.drawPeaks;
-
 }
 
+void wfmain::setAudioDevicesUI()
+{
+    // Enumerate audio devices, need to do before settings are loaded.
+    const auto audioOutputs = QAudioDeviceInfo::availableDevices(QAudio::AudioOutput);
+    for (const QAudioDeviceInfo& deviceInfo : audioOutputs) {
+        ui->audioOutputCombo->addItem(deviceInfo.deviceName(),QVariant::fromValue(deviceInfo));
+    }
+    const auto audioInputs = QAudioDeviceInfo::availableDevices(QAudio::AudioInput);
+    for (const QAudioDeviceInfo& deviceInfo : audioInputs) {
+        ui->audioInputCombo->addItem(deviceInfo.deviceName(),QVariant::fromValue(deviceInfo));
+    }
+}
+
+void wfmain::setSerialDevicesUI()
+{
+    ui->serialDeviceListCombo->blockSignals(true);
+    ui->serialDeviceListCombo->addItem("Auto", 0);
+    int i = 0;
+    foreach(const QSerialPortInfo & serialPortInfo, QSerialPortInfo::availablePorts())
+    {
+        portList.append(serialPortInfo.portName());
+#if defined(Q_OS_LINUX) || defined(Q_OS_MAC)
+        ui->serialDeviceListCombo->addItem(QString("/dev/")+serialPortInfo.portName(), i++);
+#else
+        ui->serialDeviceListCombo->addItem(serialPortInfo.portName(), i++);
+#endif
+    }
+#if defined(Q_OS_LINUX) || defined(Q_OS_MAC)
+    ui->serialDeviceListCombo->addItem("Manual...", 256);
+#endif
+    ui->serialDeviceListCombo->blockSignals(false);
+
+    ui->vspCombo->blockSignals(true);
+
+#ifdef Q_OS_WIN
+    ui->vspCombo->addItem(QString("None"), i++);
+
+    foreach(const QSerialPortInfo & serialPortInfo, QSerialPortInfo::availablePorts())
+    {
+        ui->vspCombo->addItem(serialPortInfo.portName());
+    }
+#else
+    // Provide reasonable names for the symbolic link to the pty device
+#ifdef Q_OS_MAC
+    QString vspName = QStandardPaths::standardLocations(QStandardPaths::DownloadLocation)[0] + "/rig-pty";
+#else
+    QString vspName=QDir::homePath()+"/rig-pty";
+#endif
+    for (i=1;i<8;i++) {
+        ui->vspCombo->addItem(vspName + QString::number(i));
+
+        if (QFile::exists(vspName+QString::number(i))) {
+            auto * model = qobject_cast(ui->vspCombo->model());
+            auto *item = model->item(ui->vspCombo->count()-1);
+            item->setEnabled(false);
+        }
+    }
+    ui->vspCombo->addItem(vspName+QString::number(i));
+    ui->vspCombo->addItem(QString("None"), i++);
+
+#endif
+    ui->vspCombo->setEditable(true);
+    ui->vspCombo->blockSignals(false);
+}
+
+
 void wfmain::setupKeyShortcuts()
 {
     keyF1 = new QShortcut(this);
diff --git a/wfmain.h b/wfmain.h
index cc8de8ae..292b4928 100644
--- a/wfmain.h
+++ b/wfmain.h
@@ -522,6 +522,8 @@ private:
     void setupKeyShortcuts();
     void setupMainUI();
     void setUIToPrefs();
+    void setSerialDevicesUI();
+    void setAudioDevicesUI();
 
     QStringList modes;
     int currentModeIndex;

From 7660adb3a937684cc43646135b5312dc0b461bf4 Mon Sep 17 00:00:00 2001
From: Elliott Liggett 
Date: Fri, 28 May 2021 21:27:06 -0700
Subject: [PATCH 33/47] More chair movements.

---
 wfmain.cpp | 194 ++++++++++++++++++++++++++++-------------------------
 wfmain.h   |   3 +
 2 files changed, 107 insertions(+), 90 deletions(-)

diff --git a/wfmain.cpp b/wfmain.cpp
index 8182a4c0..59f6ada5 100644
--- a/wfmain.cpp
+++ b/wfmain.cpp
@@ -51,32 +51,10 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
 
     setAudioDevicesUI();
 
-    setDefaultColors(); // set of UI colors with defaults populated
-    setDefPrefs(); // other default options
+    setDefaultColors();
+    setDefPrefs();
 
-
-    if (settingsFile.isNull()) {
-        settings = new QSettings();
-    }
-    else
-    {
-        QString file = settingsFile;
-        QFile info(settingsFile);
-        QString path="";
-        if (!QFileInfo(info).isAbsolute())
-        {
-            path = QStandardPaths::writableLocation(QStandardPaths::AppDataLocation);
-            if (path.isEmpty())
-            {
-                path = QDir::homePath();
-            }
-            path = path + "/";
-            file = info.fileName();
-        }
-
-        qDebug() << "Loading settings from:" << path + file;
-        settings = new QSettings(path + file, QSettings::Format::IniFormat);
-    }
+    getSettingsFilePath(settingsFile);
 
     setupPlots();
     loadSettings(); // Look for saved preferences
@@ -84,64 +62,14 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
 
     setUIToPrefs();
 
-    // Start server if enabled in config
-    if (serverConfig.enabled) {
-        serverConfig.lan = prefs.enableLAN;
-        if (!serverConfig.lan) {
-            serverConfig.resampleQuality = udpPrefs.resampleQuality;
-            serverConfig.audioInput = udpPrefs.audioInput;
-            serverConfig.audioOutput = udpPrefs.audioOutput;
-            serverConfig.baudRate = prefs.serialPortBaud;
-            serverConfig.inputDevice = udpPrefs.inputDevice;
-            serverConfig.outputDevice = udpPrefs.outputDevice;
-        }
-        udp = new udpServer(serverConfig);
+    setServerToPrefs();
 
-        serverThread = new QThread(this);
-
-        udp->moveToThread(serverThread);
-
-        connect(this, SIGNAL(initServer()), udp, SLOT(init()));
-        connect(serverThread, SIGNAL(finished()), udp, SLOT(deleteLater()));
-
-
-        serverThread->start();
-
-        emit initServer();
-
-        connect(this, SIGNAL(sendRigCaps(rigCapabilities)), udp, SLOT(receiveRigCaps(rigCapabilities)));
-
-    }
-
-    delayedCmdIntervalLAN_ms = 10; // interval for regular delayed commands, including initial rig/UI state queries
-    delayedCmdIntervalSerial_ms = 100; // interval for regular delayed commands, including initial rig/UI state queries
-    delayedCmdStartupInterval_ms = 250; // interval for rigID polling
-    delayedCommand = new QTimer(this);
-    delayedCommand->setInterval(delayedCmdStartupInterval_ms); // 250ms until we find rig civ and id, then 100ms.
-    delayedCommand->setSingleShot(true);
-    connect(delayedCommand, SIGNAL(timeout()), this, SLOT(runDelayedCommand()));
-
-    periodicPollingTimer = new QTimer(this);
-    periodicPollingTimer->setInterval(10);
-    periodicPollingTimer->setSingleShot(false);
-    connect(periodicPollingTimer, SIGNAL(timeout()), this, SLOT(runPeriodicCommands()));
-
-    freq.MHzDouble = 0.0;
-    freq.Hz = 0;
+    setInitialTiming();
 
     openRig();
 
     rigConnections();
 
-    // Plot user interaction
-    connect(plot, SIGNAL(mouseDoubleClick(QMouseEvent*)), this, SLOT(handlePlotDoubleClick(QMouseEvent*)));
-    connect(wf, SIGNAL(mouseDoubleClick(QMouseEvent*)), this, SLOT(handleWFDoubleClick(QMouseEvent*)));
-    connect(plot, SIGNAL(mousePress(QMouseEvent*)), this, SLOT(handlePlotClick(QMouseEvent*)));
-    connect(wf, SIGNAL(mousePress(QMouseEvent*)), this, SLOT(handleWFClick(QMouseEvent*)));
-    connect(wf, SIGNAL(mouseWheel(QWheelEvent*)), this, SLOT(handleWFScroll(QWheelEvent*)));
-    connect(plot, SIGNAL(mouseWheel(QWheelEvent*)), this, SLOT(handlePlotScroll(QWheelEvent*)));
-
-
     if (serverConfig.enabled && udp != Q_NULLPTR) {
         // Server
         connect(rig, SIGNAL(haveAudioData(audioPacket)), udp, SLOT(receiveAudioData(audioPacket)));
@@ -149,20 +77,7 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
         connect(udp, SIGNAL(haveDataFromServer(QByteArray)), rig, SLOT(dataFromServer(QByteArray)));
     }
 
-
-    SMeterReadings.fill(0,30);
-    powerMeterReadings.fill(0,30);
-
-    pttTimer = new QTimer(this);
-    pttTimer->setInterval(180*1000); // 3 minute max transmit time in ms
-    pttTimer->setSingleShot(true);
-    connect(pttTimer, SIGNAL(timeout()), this, SLOT(handlePttLimit()));
     amTransmitting = false;
-    ui->tuneLockChk->setChecked(false);
-    freqLock = false;
-
-    //getInitialRigState();
-    oldFreqDialVal = ui->freqDial->value();
 }
 
 wfmain::~wfmain()
@@ -638,6 +553,15 @@ void wfmain::setupPlots()
 
     freqIndicatorLine->start->setCoords(0.5,0);
     freqIndicatorLine->end->setCoords(0.5,160);
+
+    // Plot user interaction
+    connect(plot, SIGNAL(mouseDoubleClick(QMouseEvent*)), this, SLOT(handlePlotDoubleClick(QMouseEvent*)));
+    connect(wf, SIGNAL(mouseDoubleClick(QMouseEvent*)), this, SLOT(handleWFDoubleClick(QMouseEvent*)));
+    connect(plot, SIGNAL(mousePress(QMouseEvent*)), this, SLOT(handlePlotClick(QMouseEvent*)));
+    connect(wf, SIGNAL(mousePress(QMouseEvent*)), this, SLOT(handleWFClick(QMouseEvent*)));
+    connect(wf, SIGNAL(mouseWheel(QWheelEvent*)), this, SLOT(handleWFScroll(QWheelEvent*)));
+    connect(plot, SIGNAL(mouseWheel(QWheelEvent*)), this, SLOT(handlePlotScroll(QWheelEvent*)));
+
 }
 
 void wfmain::setupMainUI()
@@ -768,6 +692,96 @@ void wfmain::setupMainUI()
     ui->statusBar->addPermanentWidget(rigName);
     rigName->setText("NONE");
     rigName->setFixedWidth(50);
+
+    SMeterReadings.fill(0,30);
+    powerMeterReadings.fill(0,30);
+
+    freq.MHzDouble = 0.0;
+    freq.Hz = 0;
+    oldFreqDialVal = ui->freqDial->value();
+
+    ui->tuneLockChk->setChecked(false);
+    freqLock = false;
+}
+
+void wfmain::getSettingsFilePath(QString settingsFile)
+{
+    if (settingsFile.isNull()) {
+        settings = new QSettings();
+    }
+    else
+    {
+        QString file = settingsFile;
+        QFile info(settingsFile);
+        QString path="";
+        if (!QFileInfo(info).isAbsolute())
+        {
+            path = QStandardPaths::writableLocation(QStandardPaths::AppDataLocation);
+            if (path.isEmpty())
+            {
+                path = QDir::homePath();
+            }
+            path = path + "/";
+            file = info.fileName();
+        }
+
+        qInfo(logSystem()) << "Loading settings from:" << path + file;
+        settings = new QSettings(path + file, QSettings::Format::IniFormat);
+    }
+}
+
+
+void wfmain::setInitialTiming()
+{
+    delayedCmdIntervalLAN_ms = 10; // interval for regular delayed commands, including initial rig/UI state queries
+    delayedCmdIntervalSerial_ms = 100; // interval for regular delayed commands, including initial rig/UI state queries
+    delayedCmdStartupInterval_ms = 250; // interval for rigID polling
+    delayedCommand = new QTimer(this);
+    delayedCommand->setInterval(delayedCmdStartupInterval_ms); // 250ms until we find rig civ and id, then 100ms.
+    delayedCommand->setSingleShot(true);
+    connect(delayedCommand, SIGNAL(timeout()), this, SLOT(runDelayedCommand()));
+
+    periodicPollingTimer = new QTimer(this);
+    periodicPollingTimer->setInterval(10);
+    periodicPollingTimer->setSingleShot(false);
+    connect(periodicPollingTimer, SIGNAL(timeout()), this, SLOT(runPeriodicCommands()));
+
+    pttTimer = new QTimer(this);
+    pttTimer->setInterval(180*1000); // 3 minute max transmit time in ms
+    pttTimer->setSingleShot(true);
+    connect(pttTimer, SIGNAL(timeout()), this, SLOT(handlePttLimit()));
+}
+
+void wfmain::setServerToPrefs()
+{
+    // Start server if enabled in config
+    if (serverConfig.enabled) {
+        serverConfig.lan = prefs.enableLAN;
+        if (!serverConfig.lan) {
+            serverConfig.resampleQuality = udpPrefs.resampleQuality;
+            serverConfig.audioInput = udpPrefs.audioInput;
+            serverConfig.audioOutput = udpPrefs.audioOutput;
+            serverConfig.baudRate = prefs.serialPortBaud;
+            serverConfig.inputDevice = udpPrefs.inputDevice;
+            serverConfig.outputDevice = udpPrefs.outputDevice;
+        }
+        udp = new udpServer(serverConfig);
+
+        serverThread = new QThread(this);
+
+        udp->moveToThread(serverThread);
+
+        connect(this, SIGNAL(initServer()), udp, SLOT(init()));
+        connect(serverThread, SIGNAL(finished()), udp, SLOT(deleteLater()));
+
+
+        serverThread->start();
+
+        emit initServer();
+
+        connect(this, SIGNAL(sendRigCaps(rigCapabilities)), udp, SLOT(receiveRigCaps(rigCapabilities)));
+
+    }
 }
 
 void wfmain::setUIToPrefs()
diff --git a/wfmain.h b/wfmain.h
index 292b4928..c004f3b1 100644
--- a/wfmain.h
+++ b/wfmain.h
@@ -524,6 +524,9 @@ private:
     void setUIToPrefs();
     void setSerialDevicesUI();
     void setAudioDevicesUI();
+    void setServerToPrefs();
+    void setInitialTiming();
+    void getSettingsFilePath(QString settingsFile);
 
     QStringList modes;
     int currentModeIndex;

From c45113b05db3ecde9c3e4654d373c3a04cebde8a Mon Sep 17 00:00:00 2001
From: Phil Taylor 
Date: Sat, 29 May 2021 09:16:28 +0100
Subject: [PATCH 34/47] Small change to show default audio devices

---
 wfmain.cpp | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/wfmain.cpp b/wfmain.cpp
index dcff89df..06f89781 100644
--- a/wfmain.cpp
+++ b/wfmain.cpp
@@ -194,7 +194,7 @@ wfmain::wfmain(const QString serialPortCL, const QString hostCL, const QString s
     qInfo(logAudio()) << "Current API: " << QString::fromStdString(apiMap[audio.getCurrentApi()]);
 
     unsigned int devices = audio.getDeviceCount();
-    qInfo(logAudio()) << "Found " << devices << " audio device(s) ...";
+    qInfo(logAudio()) << "Found " << devices << " audio device(s) *=default";
 
     for (unsigned int i = 1; i < devices; i++) {
         info = audio.getDeviceInfo(i);

From a3cbeacf5ee75e0b8f8f6913dfeaf5d0e9f470b0 Mon Sep 17 00:00:00 2001
From: Phil Taylor 
Date: Sat, 29 May 2021 18:59:45 +0100
Subject: [PATCH 35/47] Lots more changes for rtaudio compatibility

---
 audiohandler.cpp | 211 ++++++++++++++++++++++++++++++-----------------
 audiohandler.h   |   2 +-
 2 files changed, 135 insertions(+), 78 deletions(-)

diff --git a/audiohandler.cpp b/audiohandler.cpp
index a37085e6..a2a4ebe7 100644
--- a/audiohandler.cpp
+++ b/audiohandler.cpp
@@ -32,7 +32,7 @@ audioHandler::~audioHandler()
 		delete ringBuf;
 }
 
-bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, int port, quint8 resampleQuality)
+bool audioHandler::init(const quint8 bits, const quint8 radioChan, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, int port, quint8 resampleQuality)
 {
 	if (isInitialized) {
 		return false;
@@ -43,10 +43,9 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
 	this->isInput = isinput;
 	this->radioSampleBits = bits;
 	this->radioSampleRate = samplerate;
-	this->radioChannels = channels;
+	this->radioChannels = radioChan;
 
 	// chunk size is always relative to Internal Sample Rate.
-	this->chunkSize = (INTERNAL_SAMPLE_RATE / 25) * radioChannels;
 
 	ringBuf = new wilt::Ring(100); // Should be customizable.
 	tempBuf.sent = 0;
@@ -61,6 +60,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
 		aParams.deviceId = audio.getDefaultOutputDevice();
 	}
 	aParams.firstChannel = 0;
+	aParams.nChannels = 2; // Internally this is always 2 channels for TX and RX.
 
 	try {
 		info = audio.getDeviceInfo(aParams.deviceId);
@@ -72,6 +72,9 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
 
 	if (info.probed)
 	{
+		// Per channel chunk size.
+		this->chunkSize = (info.preferredSampleRate / 50);
+
 		qInfo(logAudio()) << (isInput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") successfully probed";
 		if (info.nativeFormats == 0)
 			qInfo(logAudio()) << "		No natively supported data formats!";
@@ -100,9 +103,8 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
 	int resample_error = 0;
 
 	if (isInput) {
-		resampler = wf_resampler_init(radioChannels, info.preferredSampleRate, samplerate, resampleQuality, &resample_error);
+		resampler = wf_resampler_init(2, info.preferredSampleRate, samplerate, resampleQuality, &resample_error);
 		try {
-			aParams.nChannels = radioChannels; 
 			audio.openStream(NULL, &aParams, RTAUDIO_SINT16, info.preferredSampleRate, &this->chunkSize, &staticWrite, this);
 			audio.startStream();
 		}
@@ -113,11 +115,9 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
 	}
 	else
 	{
-		resampler = wf_resampler_init(radioChannels, samplerate, info.preferredSampleRate, resampleQuality, &resample_error);
+		resampler = wf_resampler_init(2, samplerate, info.preferredSampleRate, resampleQuality, &resample_error);
 		try {
-			unsigned int length = chunkSize / 2;
-			aParams.nChannels = 2; // Internally this is always 2 channels
-			audio.openStream(&aParams, NULL, RTAUDIO_SINT16, info.preferredSampleRate, &length, &staticRead, this);
+			audio.openStream(&aParams, NULL, RTAUDIO_SINT16, info.preferredSampleRate, &this->chunkSize, &staticRead, this);
 			audio.startStream();
 		}
 		catch (RtAudioError& e) {
@@ -127,6 +127,8 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
 	}
 	qInfo(logAudio()) << (isInput ? "Input" : "Output") << "device successfully opened";
 
+	qInfo(logAudio()) << (isInput ? "Input" : "Output") << "detected latency:" <volume = (qreal)(volume/255.0);
+	this->volume = (qreal)volume/255.0;
+	qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << this->volume << ")";
 }
 
 
@@ -149,26 +151,27 @@ void audioHandler::setVolume(unsigned char volume)
 /// 
 int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status)
 {
+	Q_UNUSED(inputBuffer);
+	Q_UNUSED(streamTime);
 	// Calculate output length, always full samples
 	int sentlen = 0;
-
 	quint8* buffer = (quint8*)outputBuffer; 
 	if (status == RTAUDIO_OUTPUT_UNDERFLOW)
 		qDebug(logAudio()) << "Underflow detected";
 
-	unsigned int nBytes = nFrames * 2 * 2; // This is ALWAYS 2 bytes per sample and 2 channels
+	int nBytes = nFrames * 2 * 2; // This is ALWAYS 2 bytes per sample and 2 channels
 
 	if (ringBuf->size()>0)
 	{
 		// Output buffer is ALWAYS 16 bit.
-
+		//qDebug(logAudio()) << "Read: nFrames" << nFrames << "nBytes" << nBytes;
 		while (sentlen < nBytes)
 		{
 			audioPacket packet;
 			if (!ringBuf->try_read(packet))
 			{
 				qDebug() << "No more data available but buffer is not full! sentlen:" << sentlen << " nBytes:" << nBytes ;
-				return 0;
+				break;
 			}
 			currentLatency = packet.time.msecsTo(QTime::currentTime());
 
@@ -183,7 +186,7 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
 				if (tempBuf.sent != tempBuf.data.length())
 				{
 					// We still don't have enough buffer space for this?
-					return 0;
+					break;
 				}
 				//qDebug(logAudio()) << "Adding partial:" << send;
 			}
@@ -194,7 +197,7 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
 					dec << packet.time.msecsTo(QTime::currentTime()) << "ms";
 				lastSeq = packet.seq;
 				if (!ringBuf->try_read(packet))
-					return sentlen;
+					break;
 				currentLatency = packet.time.msecsTo(QTime::currentTime());
 			}
 
@@ -227,13 +230,16 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
 
 int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status)
 {
+	Q_UNUSED(outputBuffer);
+	Q_UNUSED(streamTime);
 	int sentlen = 0;
-	unsigned int nBytes = nFrames * 2 * radioChannels; // This is ALWAYS 2 bytes per sample
+	int nBytes = nFrames * 2 * 2; // This is ALWAYS 2 bytes per sample
 	const char* data = (const char*)inputBuffer;
+	//qDebug(logAudio()) << "nFrames" << nFrames << "nBytes" << nBytes;
 	while (sentlen < nBytes) {
-		if (tempBuf.sent != chunkSize)
+		if (tempBuf.sent != nBytes)
 		{
-			int send = qMin((int)(nBytes - sentlen), (int)chunkSize - tempBuf.sent);
+			int send = qMin((int)(nBytes - sentlen), (int)nBytes - tempBuf.sent);
 			tempBuf.data.append(QByteArray::fromRawData(data + sentlen, send));
 			sentlen = sentlen + send;
 			tempBuf.seq = 0; // Not used in TX
@@ -251,6 +257,8 @@ int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int
 		}
 	}
 
+	//qDebug(logAudio()) << "sentlen" << sentlen;
+
 	return 0; 
 }
 
@@ -276,85 +284,113 @@ void audioHandler::stateChanged(QAudio::State state)
 
 
 
-void audioHandler::incomingAudio(audioPacket data)
+void audioHandler::incomingAudio(audioPacket inPacket)
 {
 	// No point buffering audio until stream is actually running.
+	// Regardless of the radio stream format, the buffered audio will ALWAYS be
+	// 16bit sample interleaved stereo 48K (or whatever the native sample rate is)
+
 	if (!audio.isStreamRunning())
 	{
 		qDebug(logAudio()) << "Packet received before stream was started";
 		return;
 	}
+	//qDebug(logAudio()) << "Got" << radioSampleBits << "bits, length" << inPacket.data.length();
 	// Incoming data is 8bits?
 	if (radioSampleBits == 8)
 	{
-		QByteArray outPacket((int)data.data.length() * 2, (char)0xff);
+		// Current packet is 8bit so need to create a new buffer that is 16bit 
+		QByteArray outPacket((int)inPacket.data.length() * 2 *(2/radioChannels), (char)0xff);
 		qint16* out = (qint16*)outPacket.data();
-		for (int f = 0; f < data.data.length(); f++)
+		for (int f = 0; f < inPacket.data.length(); f++)
 		{
 			if (isUlaw)
 			{
-				out[f] = ulaw_decode[(quint8)data.data[f]];
+				if (radioChannels == 1)
+				{
+					*out++ = ulaw_decode[(quint8)inPacket.data[f]] * this->volume;
+					*out++ = ulaw_decode[(quint8)inPacket.data[f]] * this->volume;
+				}
+				else
+				{
+					// This is already 2 channel.
+					*out++ = ulaw_decode[(quint8)inPacket.data[f]] * this->volume;
+				}
 			}
 			else
 			{
 				// Convert 8-bit sample to 16-bit
-				out[f] = (qint16)(((quint8)data.data[f] << 8) - 32640);
+				if (radioChannels == 1)
+				{
+					*out++ = (qint16)(((quint8)inPacket.data[f] << 8) - 32640) * this->volume;
+					*out++ = (qint16)(((quint8)inPacket.data[f] << 8) - 32640) * this->volume;
+				}
+				else
+				{
+					*out++ = (qint16)(((quint8)inPacket.data[f] << 8) - 32640 * this->volume);
+				}
 			}
 		}
-		data.data.clear();
-		data.data = outPacket; // Replace incoming data with converted.
+		inPacket.data.clear();
+		inPacket.data = outPacket; // Replace incoming data with converted.
 	}
+	else 
+	{
+		// This is already a 16bit stream, do we need to convert to stereo?
+		if (radioChannels == 1) {
+			// Yes
+			QByteArray outPacket(inPacket.data.length() * 2, (char)0xff); // Preset the output buffer size.
+			qint16* in = (qint16*)inPacket.data.data();
+			qint16* out = (qint16*)outPacket.data();
+			for (int f = 0; f < inPacket.data.length() / 2; f++)
+			{
+				*out++ = (qint16)*in * this->volume;
+				*out++ = (qint16)*in++ * this->volume;
+			}
+			inPacket.data.clear();
+			inPacket.data = outPacket; // Replace incoming data with converted.
+		}
+		else 
+		{
+			// We already have two channels so just update volume.
+			qint16* in = (qint16*)inPacket.data.data();
+			for (int f = 0; f < inPacket.data.length() / 2; f++)
+			{
+				*in = *in++ * this->volume;
+			}
+		}
 
-	//qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.data.length();
+	}
 
 	/*	We now have an array of 16bit samples in the NATIVE samplerate of the radio
 		If the radio sample rate is below 48000, we need to resample.
 		*/
+	//qDebug(logAudio()) << "Now 16 bit stereo, length" << inPacket.data.length();
 
 	if (ratioDen != 1) {
 
 		// We need to resample
-		quint32 outFrames = ((data.data.length() / 2) * ratioDen) / radioChannels;
-		quint32 inFrames = (data.data.length() / 2) / radioChannels;
-		QByteArray outPacket(outFrames * 2 * radioChannels, 0xff); // Preset the output buffer size.
+		// We have a stereo 16bit stream.
+		quint32 outFrames = ((inPacket.data.length() / 4) * ratioDen);
+		quint32 inFrames = (inPacket.data.length() / 4);
+		QByteArray outPacket(outFrames * 4, (char)0xff); // Preset the output buffer size.
+
+		const qint16* in = (qint16*)inPacket.data.constData();
+		qint16* out = (qint16*)outPacket.data();
 
 		int err = 0;
-		if (this->radioChannels == 1) {
-			err = wf_resampler_process_int(resampler, 0, (const qint16*)data.data.constData(), &inFrames, (qint16*)outPacket.data(), &outFrames);
-		}
-		else {
-			err = wf_resampler_process_interleaved_int(resampler, (const qint16*)data.data.constData(), &inFrames, (qint16*)outPacket.data(), &outFrames);
-		}
+		err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
 		if (err) {
 			qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
 		}
-		data.data.clear();
-		data.data = outPacket; // Replace incoming data with converted.
+		inPacket.data.clear();
+		inPacket.data = outPacket; // Replace incoming data with converted.
 	}
 
-	if (radioChannels == 1) {
-		// Convert to stereo and set volume.
-		QByteArray outPacket(data.data.length()*2, 0xff); // Preset the output buffer size.
-		qint16* in = (qint16*)data.data.data();
-		qint16* out = (qint16*)outPacket.data();
-		for (int f = 0; f < data.data.length()/2; f++)
-		{
-			*out++ = *in * volume;
-			*out++ = *in++ * volume;
-		}
-		data.data.clear();
-		data.data = outPacket; // Replace incoming data with converted.
-	} else {
-		// We already have two channels so just update volume.
-		qint16* in = (qint16*)data.data.data();
-		for (int f = 0; f < data.data.length() / 2; f++)
-		{
-			in[f] = in[f] * volume;
-		}
-	}
+	//qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.data.length();
+	//qDebug(logAudio()) << "Now 16 bit stereo, length" << inPacket.data.length();
 
-
-	if (!ringBuf->try_write(data))
+	if (!ringBuf->try_write(inPacket))
 	{
 		qDebug(logAudio()) << "Buffer full! capacity:" << ringBuf->capacity() << "length" << ringBuf->size();
 	}
@@ -381,38 +417,57 @@ void audioHandler::getNextAudioChunk(QByteArray& ret)
 {
 	audioPacket packet;
 	packet.sent = 0;
+
 	if (ringBuf != Q_NULLPTR && ringBuf->try_read(packet))
 	{
+
+		//qDebug(logAudio) << "Chunksize" << this->chunkSize << "Packet size" << packet.data.length();
+		// Packet will arrive as stereo interleaved 16bit 48K
 		if (ratioNum != 1)
 		{
-			// We need to resample (we are STILL 16 bit!)
-			quint32 outFrames = ((packet.data.length() / 2) / ratioNum) / radioChannels;
-			quint32 inFrames = (packet.data.length() / 2) / radioChannels;
-			QByteArray inPacket((int)outFrames * 2 * radioChannels, (char)0xff);
+			quint32 outFrames = ((packet.data.length() / 4) / ratioNum);
+			quint32 inFrames = (packet.data.length() / 4);
+			QByteArray outPacket((int)outFrames * 4, (char)0xff);
 
 			const qint16* in = (qint16*)packet.data.constData();
-			qint16* out = (qint16*)inPacket.data();
+			qint16* out = (qint16*)outPacket.data();
+
 			int err = 0;
-			if (this->radioChannels == 1) {
-				err = wf_resampler_process_int(resampler, 0, in, &inFrames, out, &outFrames);
-			}
-			else {
-				err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
-			}
+			err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
 			if (err) {
 				qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
 			}
 			//qInfo(logAudio()) << "Resampler run " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
 			//qInfo(logAudio()) << "Resampler run inLen:" << packet->datain.length() << " outLen:" << packet->dataout.length();
 			packet.data.clear();
-			packet.data = inPacket; // Copy output packet back to input buffer.
+			packet.data = outPacket; // Copy output packet back to input buffer.
 		}
 
+		//qDebug(logAudio()) << "Now resampled, length" << packet.data.length();
+
+		// Do we need to convert mono to stereo?
+		if (radioChannels == 1)
+		{
+			// Strip out right channel?
+			QByteArray outPacket(packet.data.length()/2, (char)0xff);
+			const qint16* in = (qint16*)packet.data.constData();
+			qint16* out = (qint16*)outPacket.data();
+			for (int f = 0; f < outPacket.length()/2; f++)
+			{
+				*out++ = *in++;
+				*++in; // Skip each even channel.
+			}
+			packet.data.clear();
+			packet.data = outPacket; // Copy output packet back to input buffer.
+		}
+		
+		//qDebug(logAudio()) << "Now mono, length" << packet.data.length();
+
 		// Do we need to convert 16-bit to 8-bit?
 		if (radioSampleBits == 8) {
-			QByteArray inPacket((int)packet.data.length() / 2, (char)0xff);
+			QByteArray outPacket((int)packet.data.length() / 2, (char)0xff);
 			qint16* in = (qint16*)packet.data.data();
-			for (int f = 0; f < inPacket.length(); f++)
+			for (int f = 0; f < outPacket.length(); f++)
 			{
 				quint8 outdata = 0;
 				if (isUlaw) {
@@ -425,14 +480,16 @@ void audioHandler::getNextAudioChunk(QByteArray& ret)
 				else {
 					outdata = (quint8)(((qFromLittleEndian(in + f) >> 8) ^ 0x80) & 0xff);
 				}
-				inPacket[f] = (char)outdata;
+				outPacket[f] = (char)outdata;
 			}
 			packet.data.clear();
-			packet.data = inPacket; // Copy output packet back to input buffer.
+			packet.data = outPacket; // Copy output packet back to input buffer.
 		}
 		ret = packet.data;
+		//qDebug(logAudio()) << "Now radio format, length" << packet.data.length();
 	}
 
+
 	return;
 
 }
diff --git a/audiohandler.h b/audiohandler.h
index 1948c9c5..a30eec8d 100644
--- a/audiohandler.h
+++ b/audiohandler.h
@@ -7,6 +7,7 @@
 #include 
 #include 
 #include 
+#include 
 #include "rtaudio/RtAudio.h"
 
 typedef signed short  MY_TYPE;
@@ -119,7 +120,6 @@ private:
     audioPacket tempBuf;
     quint16 currentLatency;
     qreal volume=1.0;
-
 };
 
 #endif // AUDIOHANDLER_H

From 5c0c22c6f45da4c57835704989895368af706a4f Mon Sep 17 00:00:00 2001
From: Phil Taylor 
Date: Sat, 29 May 2021 20:13:52 +0100
Subject: [PATCH 36/47] Detect number of device channels and convert
 accordingly

---
 audiohandler.cpp | 73 ++++++++++++++++++++++--------------------------
 audiohandler.h   |  1 +
 2 files changed, 34 insertions(+), 40 deletions(-)

diff --git a/audiohandler.cpp b/audiohandler.cpp
index a2a4ebe7..7511cd18 100644
--- a/audiohandler.cpp
+++ b/audiohandler.cpp
@@ -60,7 +60,6 @@ bool audioHandler::init(const quint8 bits, const quint8 radioChan, const quint16
 		aParams.deviceId = audio.getDefaultOutputDevice();
 	}
 	aParams.firstChannel = 0;
-	aParams.nChannels = 2; // Internally this is always 2 channels for TX and RX.
 
 	try {
 		info = audio.getDeviceInfo(aParams.deviceId);
@@ -73,11 +72,15 @@ bool audioHandler::init(const quint8 bits, const quint8 radioChan, const quint16
 	if (info.probed)
 	{
 		// Per channel chunk size.
+			aParams.nChannels = 2; // Internally this is always 2 channels for TX and RX.
 		this->chunkSize = (info.preferredSampleRate / 50);
 
 		qInfo(logAudio()) << (isInput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") successfully probed";
 		if (info.nativeFormats == 0)
+		{
 			qInfo(logAudio()) << "		No natively supported data formats!";
+			return false;
+		}
 		else {
 			qDebug(logAudio()) << "		Supported formats:" <<
 				(info.nativeFormats & RTAUDIO_SINT8 ? "8-bit int," : "") <<
@@ -86,11 +89,19 @@ bool audioHandler::init(const quint8 bits, const quint8 radioChan, const quint16
 				(info.nativeFormats & RTAUDIO_SINT32 ? "32-bit int," : "") <<
 				(info.nativeFormats & RTAUDIO_FLOAT32 ? "32-bit float," : "") <<
 				(info.nativeFormats & RTAUDIO_FLOAT64 ? "64-bit float," : "");
+
 			qInfo(logAudio()) << "		Preferred sample rate:" << info.preferredSampleRate;
-			if (isInput)
-				qInfo(logAudio()) << "		Input Channels:" << info.inputChannels;
-			else
-				qInfo(logAudio()) << "		Output Channels:" << info.outputChannels;
+			if (isInput) {
+				devChannels = info.inputChannels;
+			} 
+			else {
+				devChannels = info.outputChannels;
+			}
+			qInfo(logAudio()) << "		Channels:" << devChannels;
+			if (devChannels > 2) {
+				devChannels = 2;
+			}
+			aParams.nChannels = devChannels;
 		}
 
 		qInfo(logAudio())  << "		chunkSize: " << chunkSize;
@@ -98,12 +109,13 @@ bool audioHandler::init(const quint8 bits, const quint8 radioChan, const quint16
 	else
 	{
 		qCritical(logAudio()) << (isInput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") could not be probed, check audio configuration!";
+		return false;
 	}
 
 	int resample_error = 0;
 
 	if (isInput) {
-		resampler = wf_resampler_init(2, info.preferredSampleRate, samplerate, resampleQuality, &resample_error);
+		resampler = wf_resampler_init(devChannels, info.preferredSampleRate, samplerate, resampleQuality, &resample_error);
 		try {
 			audio.openStream(NULL, &aParams, RTAUDIO_SINT16, info.preferredSampleRate, &this->chunkSize, &staticWrite, this);
 			audio.startStream();
@@ -115,7 +127,7 @@ bool audioHandler::init(const quint8 bits, const quint8 radioChan, const quint16
 	}
 	else
 	{
-		resampler = wf_resampler_init(2, samplerate, info.preferredSampleRate, resampleQuality, &resample_error);
+		resampler = wf_resampler_init(devChannels, samplerate, info.preferredSampleRate, resampleQuality, &resample_error);
 		try {
 			audio.openStream(&aParams, NULL, RTAUDIO_SINT16, info.preferredSampleRate, &this->chunkSize, &staticRead, this);
 			audio.startStream();
@@ -159,7 +171,7 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
 	if (status == RTAUDIO_OUTPUT_UNDERFLOW)
 		qDebug(logAudio()) << "Underflow detected";
 
-	int nBytes = nFrames * 2 * 2; // This is ALWAYS 2 bytes per sample and 2 channels
+	int nBytes = nFrames * devChannels * 2; // This is ALWAYS 2 bytes per sample and 2 channels
 
 	if (ringBuf->size()>0)
 	{
@@ -233,7 +245,7 @@ int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int
 	Q_UNUSED(outputBuffer);
 	Q_UNUSED(streamTime);
 	int sentlen = 0;
-	int nBytes = nFrames * 2 * 2; // This is ALWAYS 2 bytes per sample
+	int nBytes = nFrames * devChannels * 2; // This is ALWAYS 2 bytes per sample
 	const char* data = (const char*)inputBuffer;
 	//qDebug(logAudio()) << "nFrames" << nFrames << "nBytes" << nBytes;
 	while (sentlen < nBytes) {
@@ -300,35 +312,16 @@ void audioHandler::incomingAudio(audioPacket inPacket)
 	if (radioSampleBits == 8)
 	{
 		// Current packet is 8bit so need to create a new buffer that is 16bit 
-		QByteArray outPacket((int)inPacket.data.length() * 2 *(2/radioChannels), (char)0xff);
+		QByteArray outPacket((int)inPacket.data.length() * 2 *(devChannels/radioChannels), (char)0xff);
 		qint16* out = (qint16*)outPacket.data();
 		for (int f = 0; f < inPacket.data.length(); f++)
 		{
-			if (isUlaw)
+			for (int g = radioChannels; g <= devChannels; g++)
 			{
-				if (radioChannels == 1)
-				{
+				if (isUlaw)
 					*out++ = ulaw_decode[(quint8)inPacket.data[f]] * this->volume;
-					*out++ = ulaw_decode[(quint8)inPacket.data[f]] * this->volume;
-				}
 				else
-				{
-					// This is already 2 channel.
-					*out++ = ulaw_decode[(quint8)inPacket.data[f]] * this->volume;
-				}
-			}
-			else
-			{
-				// Convert 8-bit sample to 16-bit
-				if (radioChannels == 1)
-				{
-					*out++ = (qint16)(((quint8)inPacket.data[f] << 8) - 32640) * this->volume;
-					*out++ = (qint16)(((quint8)inPacket.data[f] << 8) - 32640) * this->volume;
-				}
-				else
-				{
 					*out++ = (qint16)(((quint8)inPacket.data[f] << 8) - 32640 * this->volume);
-				}
 			}
 		}
 		inPacket.data.clear();
@@ -337,7 +330,7 @@ void audioHandler::incomingAudio(audioPacket inPacket)
 	else 
 	{
 		// This is already a 16bit stream, do we need to convert to stereo?
-		if (radioChannels == 1) {
+		if (radioChannels == 1 && devChannels > 1) {
 			// Yes
 			QByteArray outPacket(inPacket.data.length() * 2, (char)0xff); // Preset the output buffer size.
 			qint16* in = (qint16*)inPacket.data.data();
@@ -352,7 +345,7 @@ void audioHandler::incomingAudio(audioPacket inPacket)
 		}
 		else 
 		{
-			// We already have two channels so just update volume.
+			// We already have the same number of channels so just update volume.
 			qint16* in = (qint16*)inPacket.data.data();
 			for (int f = 0; f < inPacket.data.length() / 2; f++)
 			{
@@ -371,8 +364,8 @@ void audioHandler::incomingAudio(audioPacket inPacket)
 
 		// We need to resample
 		// We have a stereo 16bit stream.
-		quint32 outFrames = ((inPacket.data.length() / 4) * ratioDen);
-		quint32 inFrames = (inPacket.data.length() / 4);
+		quint32 outFrames = ((inPacket.data.length() / 2 / devChannels) * ratioDen);
+		quint32 inFrames = (inPacket.data.length() / 2 / devChannels);
 		QByteArray outPacket(outFrames * 4, (char)0xff); // Preset the output buffer size.
 
 		const qint16* in = (qint16*)inPacket.data.constData();
@@ -425,9 +418,9 @@ void audioHandler::getNextAudioChunk(QByteArray& ret)
 		// Packet will arrive as stereo interleaved 16bit 48K
 		if (ratioNum != 1)
 		{
-			quint32 outFrames = ((packet.data.length() / 4) / ratioNum);
-			quint32 inFrames = (packet.data.length() / 4);
-			QByteArray outPacket((int)outFrames * 4, (char)0xff);
+			quint32 outFrames = ((packet.data.length() / 2 / devChannels) / ratioNum);
+			quint32 inFrames = (packet.data.length() / 2 / devChannels);
+			QByteArray outPacket((int)outFrames * 2 * devChannels, (char)0xff);
 
 			const qint16* in = (qint16*)packet.data.constData();
 			qint16* out = (qint16*)outPacket.data();
@@ -446,7 +439,7 @@ void audioHandler::getNextAudioChunk(QByteArray& ret)
 		//qDebug(logAudio()) << "Now resampled, length" << packet.data.length();
 
 		// Do we need to convert mono to stereo?
-		if (radioChannels == 1)
+		if (radioChannels == 1 && devChannels > 1)
 		{
 			// Strip out right channel?
 			QByteArray outPacket(packet.data.length()/2, (char)0xff);
@@ -455,7 +448,7 @@ void audioHandler::getNextAudioChunk(QByteArray& ret)
 			for (int f = 0; f < outPacket.length()/2; f++)
 			{
 				*out++ = *in++;
-				*++in; // Skip each even channel.
+				*in++; // Skip each even channel.
 			}
 			packet.data.clear();
 			packet.data = outPacket; // Copy output packet back to input buffer.
diff --git a/audiohandler.h b/audiohandler.h
index a30eec8d..eddc6255 100644
--- a/audiohandler.h
+++ b/audiohandler.h
@@ -120,6 +120,7 @@ private:
     audioPacket tempBuf;
     quint16 currentLatency;
     qreal volume=1.0;
+    int devChannels;
 };
 
 #endif // AUDIOHANDLER_H

From 94ecd5bb56be3e81adc096c96c91090e09b6974b Mon Sep 17 00:00:00 2001
From: Phil Taylor 
Date: Sat, 29 May 2021 20:50:27 +0100
Subject: [PATCH 37/47] fix for mac/linux compiler

---
 audiohandler.cpp | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

diff --git a/audiohandler.cpp b/audiohandler.cpp
index 7511cd18..79843fd9 100644
--- a/audiohandler.cpp
+++ b/audiohandler.cpp
@@ -234,7 +234,7 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
 			lastSeq = packet.seq;
 		}
 	}
-	//qDebug(logAudio()) << "looking for: " << nBytes << " got: " << sentlen;
+    //qDebug(logAudio()) << "looking for: " << nBytes << " got: " << sentlen;
 
 	return 0;
 }
@@ -307,7 +307,7 @@ void audioHandler::incomingAudio(audioPacket inPacket)
 		qDebug(logAudio()) << "Packet received before stream was started";
 		return;
 	}
-	//qDebug(logAudio()) << "Got" << radioSampleBits << "bits, length" << inPacket.data.length();
+    //qDebug(logAudio()) << "Got" << radioSampleBits << "bits, length" << inPacket.data.length();
 	// Incoming data is 8bits?
 	if (radioSampleBits == 8)
 	{
@@ -349,7 +349,8 @@ void audioHandler::incomingAudio(audioPacket inPacket)
 			qint16* in = (qint16*)inPacket.data.data();
 			for (int f = 0; f < inPacket.data.length() / 2; f++)
 			{
-				*in = *in++ * this->volume;
+                *in = *in * this->volume;
+                *in++;
 			}
 		}
 
@@ -358,7 +359,7 @@ void audioHandler::incomingAudio(audioPacket inPacket)
 	/*	We now have an array of 16bit samples in the NATIVE samplerate of the radio
 		If the radio sample rate is below 48000, we need to resample.
 		*/
-	//qDebug(logAudio()) << "Now 16 bit stereo, length" << inPacket.data.length();
+    //qDebug(logAudio()) << "Now 16 bit stereo, length" << inPacket.data.length();
 
 	if (ratioDen != 1) {
 
@@ -380,8 +381,7 @@ void audioHandler::incomingAudio(audioPacket inPacket)
 		inPacket.data = outPacket; // Replace incoming data with converted.
 	}
 
-	//qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.data.length();
-	//qDebug(logAudio()) << "Now 16 bit stereo, length" << inPacket.data.length();
+    //qDebug(logAudio()) << "Adding packet to buffer:" << inPacket.seq << ": " << inPacket.data.length();
 
 	if (!ringBuf->try_write(inPacket))
 	{

From bbf1ecb6d8fe87bec2039c74824a84d8748bf8d4 Mon Sep 17 00:00:00 2001
From: Phil Taylor 
Date: Sun, 30 May 2021 11:36:13 +0100
Subject: [PATCH 38/47] add silence if audio has stopped

---
 audiohandler.cpp | 4 ++++
 1 file changed, 4 insertions(+)

diff --git a/audiohandler.cpp b/audiohandler.cpp
index 79843fd9..dcf463d6 100644
--- a/audiohandler.cpp
+++ b/audiohandler.cpp
@@ -236,6 +236,10 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
 	}
     //qDebug(logAudio()) << "looking for: " << nBytes << " got: " << sentlen;
 
+    // fill the rest of the buffer with silence
+    if (nBytes > sentlen) {
+        memset(buffer+sentlen,0,nBytes-sentlen);
+    }
 	return 0;
 }
 

From dda1f91afaedad4917aae325de3e4d5965d0fabf Mon Sep 17 00:00:00 2001
From: Phil Taylor 
Date: Sun, 30 May 2021 12:07:51 +0100
Subject: [PATCH 39/47] Remove various compiler warnings and tidy up.

---
 audiohandler.cpp | 32 ++++----------------------------
 audiohandler.h   | 11 -----------
 2 files changed, 4 insertions(+), 39 deletions(-)

diff --git a/audiohandler.cpp b/audiohandler.cpp
index dcf463d6..5441db32 100644
--- a/audiohandler.cpp
+++ b/audiohandler.cpp
@@ -14,6 +14,7 @@ audioHandler::audioHandler(QObject* parent) :
 	isInput(0),
 	chunkAvailable(false)
 {
+	Q_UNUSED(parent)
 }
 
 audioHandler::~audioHandler()
@@ -248,6 +249,7 @@ int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int
 {
 	Q_UNUSED(outputBuffer);
 	Q_UNUSED(streamTime);
+	Q_UNUSED(status);
 	int sentlen = 0;
 	int nBytes = nFrames * devChannels * 2; // This is ALWAYS 2 bytes per sample
 	const char* data = (const char*)inputBuffer;
@@ -278,27 +280,6 @@ int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int
 	return 0; 
 }
 
-qint64 audioHandler::bytesAvailable() const
-{
-	return 0;
-}
-
-bool audioHandler::isSequential() const
-{
-	return true;
-}
-
-void audioHandler::notified()
-{
-}
-
-
-void audioHandler::stateChanged(QAudio::State state)
-{
-	
-}
-
-
 
 void audioHandler::incomingAudio(audioPacket inPacket)
 {
@@ -354,7 +335,7 @@ void audioHandler::incomingAudio(audioPacket inPacket)
 			for (int f = 0; f < inPacket.data.length() / 2; f++)
 			{
                 *in = *in * this->volume;
-                *in++;
+                in++;
 			}
 		}
 
@@ -405,11 +386,6 @@ int audioHandler::getLatency()
 	return currentLatency;
 }
 
-bool audioHandler::isChunkAvailable()
-{
-	return (chunkAvailable);
-}
-
 void audioHandler::getNextAudioChunk(QByteArray& ret)
 {
 	audioPacket packet;
@@ -452,7 +428,7 @@ void audioHandler::getNextAudioChunk(QByteArray& ret)
 			for (int f = 0; f < outPacket.length()/2; f++)
 			{
 				*out++ = *in++;
-				*in++; // Skip each even channel.
+				in++; // Skip each even channel.
 			}
 			packet.data.clear();
 			packet.data = outPacket; // Copy output packet back to input buffer.
diff --git a/audiohandler.h b/audiohandler.h
index eddc6255..d9786442 100644
--- a/audiohandler.h
+++ b/audiohandler.h
@@ -50,22 +50,11 @@ public:
 
     int getLatency();
 
-    bool setDevice(QAudioDeviceInfo deviceInfo);
-
-    void start();
-    void flush();
-    void stop();
-
-    qint64 bytesAvailable() const;
-    bool isSequential() const;
     void getNextAudioChunk(QByteArray &data);
-    bool isChunkAvailable();
 
 private slots:
     bool init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool isulaw, const bool isinput, int port, quint8 resampleQuality);
     void changeLatency(const quint16 newSize);
-    void notified();
-    void stateChanged(QAudio::State state);
     void setVolume(unsigned char volume);
     void incomingAudio(const audioPacket data);
 

From 2ad5e2c54cee6ba058e8863356c103d658e44423 Mon Sep 17 00:00:00 2001
From: Elliott Liggett 
Date: Sun, 30 May 2021 15:08:22 -0700
Subject: [PATCH 40/47] CIV may now be changed as-needed while running.

---
 rigcommander.cpp | 13 ++++++++-----
 wfmain.cpp       |  3 +++
 wfmain.h         |  3 +++
 3 files changed, 14 insertions(+), 5 deletions(-)

diff --git a/rigcommander.cpp b/rigcommander.cpp
index f8b78a18..ab7a5501 100644
--- a/rigcommander.cpp
+++ b/rigcommander.cpp
@@ -170,9 +170,7 @@ void rigCommander::setup()
     // common elements between the two constructors go here:
     setCIVAddr(civAddr);
     spectSeqMax = 0; // this is now set after rig ID determined
-    payloadPrefix = QByteArray("\xFE\xFE");
-    payloadPrefix.append(civAddr);
-    payloadPrefix.append((char)compCivAddr);
+
 
     payloadSuffix = QByteArray("\xFD");
 
@@ -183,6 +181,8 @@ void rigCommander::setup()
     pttAllowed = true; // This is for developing, set to false for "safe" debugging. Set to true for deployment.
 }
 
+
+
 void rigCommander::process()
 {
     // new thread enters here. Do nothing but do check for errors.
@@ -963,10 +963,13 @@ void rigCommander::setPTT(bool pttOn)
 
 void rigCommander::setCIVAddr(unsigned char civAddr)
 {
-    // Note: This is the radio's CIV address
+    // Note: This sets the radio's CIV address
     // the computer's CIV address is defined in the header file.
-    // TODO: this function *could* be written to re-write the CIV preamble.
+
     this->civAddr = civAddr;
+    payloadPrefix = QByteArray("\xFE\xFE");
+    payloadPrefix.append(civAddr);
+    payloadPrefix.append((char)compCivAddr);
 }
 
 void rigCommander::handleNewData(const QByteArray& data)
diff --git a/wfmain.cpp b/wfmain.cpp
index 59f6ada5..bda9cb09 100644
--- a/wfmain.cpp
+++ b/wfmain.cpp
@@ -182,6 +182,8 @@ void wfmain::openRig()
 
 void wfmain::rigConnections()
 {
+    connect(this, SIGNAL(setCIVAddr(unsigned char)), rig, SLOT(setCIVAddr(unsigned char)));
+
     connect(this, SIGNAL(sendPowerOn()), rig, SLOT(powerOn()));
     connect(this, SIGNAL(sendPowerOff()), rig, SLOT(powerOff()));
 
@@ -4426,6 +4428,7 @@ void wfmain::on_rigCIVaddrHexLine_editingFinished()
     if(okconvert && (propCIVAddr < 0xe0) && (propCIVAddr != 0))
     {
         prefs.radioCIVAddr = propCIVAddr;
+        emit setCIVAddr(propCIVAddr);
         showStatusBarText(QString("Setting radio CI-V address to: 0x%1. Press Save Settings to retain.").arg(propCIVAddr, 2, 16));
     } else {
         showStatusBarText(QString("Could not use provided CI-V address. Address must be < 0xE0"));
diff --git a/wfmain.h b/wfmain.h
index c004f3b1..e0a55688 100644
--- a/wfmain.h
+++ b/wfmain.h
@@ -44,6 +44,9 @@ public:
     ~wfmain();
 
 signals:
+    // Basic to rig:
+    void setCIVAddr(unsigned char);
+
     // Power
     void sendPowerOn();
     void sendPowerOff();

From 5174c03d116594391dd80475823bed8757880064 Mon Sep 17 00:00:00 2001
From: Elliott Liggett 
Date: Sun, 30 May 2021 23:26:36 -0700
Subject: [PATCH 41/47] Waterfal length may now be adjusted. Let's see what
 range of length seems good and limit the control accordingly. Also there may
 be a memory leak in the prepareWf() function where the colormap is created
 when the image is resized.

---
 rigcommander.cpp |   1 -
 wfmain.cpp       | 125 +++++++++++++++++++++++++++++++----------------
 wfmain.h         |   8 +--
 wfmain.ui        |  56 ++++++++++++++++++++-
 4 files changed, 142 insertions(+), 48 deletions(-)

diff --git a/rigcommander.cpp b/rigcommander.cpp
index ab7a5501..a96378ad 100644
--- a/rigcommander.cpp
+++ b/rigcommander.cpp
@@ -171,7 +171,6 @@ void rigCommander::setup()
     setCIVAddr(civAddr);
     spectSeqMax = 0; // this is now set after rig ID determined
 
-
     payloadSuffix = QByteArray("\xFD");
 
     lookingForRig = false;
diff --git a/wfmain.cpp b/wfmain.cpp
index bda9cb09..2e35fdc7 100644
--- a/wfmain.cpp
+++ b/wfmain.cpp
@@ -142,7 +142,7 @@ void wfmain::openRig()
     }
 
     // Start rigctld
-    if (prefs.enableRigCtlD) {
+    if (prefs.enableRigCtlD && rigCtl == Q_NULLPTR) {
         rigCtl = new rigCtlD(this);
 
         rigCtl->startServer(prefs.rigCtlPort);
@@ -525,6 +525,7 @@ void wfmain::setupPlots()
 {
 
 // Line 290--
+    spectrumDrawLock = true;
     plot = ui->plot; // rename it waterfall.
     wf = ui->waterfall;
 
@@ -563,7 +564,7 @@ void wfmain::setupPlots()
     connect(wf, SIGNAL(mousePress(QMouseEvent*)), this, SLOT(handleWFClick(QMouseEvent*)));
     connect(wf, SIGNAL(mouseWheel(QWheelEvent*)), this, SLOT(handleWFScroll(QWheelEvent*)));
     connect(plot, SIGNAL(mouseWheel(QWheelEvent*)), this, SLOT(handlePlotScroll(QWheelEvent*)));
-
+    spectrumDrawLock = false;
 }
 
 void wfmain::setupMainUI()
@@ -1136,7 +1137,7 @@ void wfmain::loadSettings()
     }
 
     ui->lanEnableBtn->setChecked(prefs.enableLAN);
-    ui->connectBtn->setEnabled(prefs.enableLAN);
+    ui->connectBtn->setEnabled(true);
 
     prefs.enableRigCtlD = settings->value("EnableRigCtlD", defPrefs.enableRigCtlD).toBool();
     prefs.rigCtlPort = settings->value("RigCtlPort", defPrefs.rigCtlPort).toInt();
@@ -1422,35 +1423,66 @@ void wfmain::saveSettings()
     settings->sync(); // Automatic, not needed (supposedly)
 }
 
+
 void wfmain::prepareWf()
+{
+    prepareWf(160);
+}
+
+void wfmain::prepareWf(unsigned int wfLength)
 {
     // All this code gets moved in from the constructor of wfmain.
 
     if(haveRigCaps)
     {
-        // do things
-        spectWidth = rigCaps.spectLenMax; // was fixed at 475
-        wfLength = 160; // fixed for now, time-length of waterfall
+        // TODO: Lock the function that draws on the spectrum while we are updating.
+        spectrumDrawLock = true;
+
+        spectWidth = rigCaps.spectLenMax;
+        this->wfLength = wfLength; // fixed for now, time-length of waterfall
 
         // Initialize before use!
 
         QByteArray empty((int)spectWidth, '\x01');
         spectrumPeaks = QByteArray( (int)spectWidth, '\x01' );
-        for(quint16 i=0; iclearData();
+        colorMap->data()->clear();
 
         colorMap->data()->setValueRange(QCPRange(0, wfLength-1));
         colorMap->data()->setKeyRange(QCPRange(0, spectWidth-1));
         colorMap->setDataRange(QCPRange(0, rigCaps.spectAmpMax));
         colorMap->setGradient(QCPColorGradient::gpJet); // TODO: Add preference
-        colorMapData = new QCPColorMapData(spectWidth, wfLength, QCPRange(0, spectWidth-1), QCPRange(0, wfLength-1));
+
+        if(colorMapData == Q_NULLPTR)
+        {
+            colorMapData = new QCPColorMapData(spectWidth, wfLength, QCPRange(0, spectWidth-1), QCPRange(0, wfLength-1));
+        } else {
+            //delete colorMapData; // TODO: Figure out why it crashes if we delete first.
+            colorMapData = new QCPColorMapData(spectWidth, wfLength, QCPRange(0, spectWidth-1), QCPRange(0, wfLength-1));
+        }
         colorMap->setData(colorMapData);
-        spectRowCurrent = 0;
+
         wf->yAxis->setRangeReversed(true);
         wf->xAxis->setVisible(false);
         rigName->setText(rigCaps.modelName);
+
+        spectrumDrawLock = false;
     } else {
         qInfo(logSystem()) << "Cannot prepare WF view without rigCaps. Waiting on this.";
         return;
@@ -2571,42 +2603,44 @@ void wfmain::receiveSpectrumData(QByteArray spectrum, double startFreq, double e
 
     }
 
-    //ui->qcp->addGraph();
-    plot->graph(0)->setData(x,y);
-    if((freq.MHzDouble < endFreq) && (freq.MHzDouble > startFreq))
+    if(!spectrumDrawLock)
     {
-        freqIndicatorLine->start->setCoords(freq.MHzDouble,0);
-        freqIndicatorLine->end->setCoords(freq.MHzDouble,160);
-    }
-    if(drawPeaks)
-    {
-        plot->graph(1)->setData(x,y2); // peaks
-    }
-    plot->yAxis->setRange(0, rigCaps.spectAmpMax+1);
-    plot->xAxis->setRange(startFreq, endFreq);
-    plot->replot();
-
-    if(specLen == spectWidth)
-    {
-        wfimage.prepend(spectrum);
-        if(wfimage.length() >  wfLength)
+        //ui->qcp->addGraph();
+        plot->graph(0)->setData(x,y);
+        if((freq.MHzDouble < endFreq) && (freq.MHzDouble > startFreq))
         {
-            wfimage.remove(wfLength);
+            freqIndicatorLine->start->setCoords(freq.MHzDouble,0);
+            freqIndicatorLine->end->setCoords(freq.MHzDouble,160);
         }
-
-        // Waterfall:
-        for(int row = 0; row < wfLength; row++)
+        if(drawPeaks)
         {
-            for(int col = 0; col < spectWidth; col++)
+            plot->graph(1)->setData(x,y2); // peaks
+        }
+        plot->yAxis->setRange(0, rigCaps.spectAmpMax+1);
+        plot->xAxis->setRange(startFreq, endFreq);
+        plot->replot();
+
+        if(specLen == spectWidth)
+        {
+            wfimage.prepend(spectrum);
+            if(wfimage.length() >  wfLength)
             {
-                colorMap->data()->setCell( col, row, wfimage.at(row).at(col));
+                wfimage.remove(wfLength);
             }
-        }
 
-        wf->yAxis->setRange(0,wfLength - 1);
-        wf->xAxis->setRange(0, spectWidth-1);
-        wf->replot();
-        spectRowCurrent = (spectRowCurrent + 1) % wfLength;
+            // Waterfall:
+            for(int row = 0; row < wfLength; row++)
+            {
+                for(int col = 0; col < spectWidth; col++)
+                {
+                    colorMap->data()->setCell( col, row, wfimage.at(row).at(col));
+                }
+            }
+
+           wf->yAxis->setRange(0,wfLength - 1);
+           wf->xAxis->setRange(0, spectWidth-1);
+           wf->replot();
+        }
     }
 }
 
@@ -3552,7 +3586,7 @@ void wfmain::on_serialEnableBtn_clicked(bool checked)
     prefs.enableLAN = !checked;
     ui->serialDeviceListCombo->setEnabled(checked);
 
-    ui->connectBtn->setEnabled(!checked);
+    ui->connectBtn->setEnabled(true);
     ui->ipAddressTxt->setEnabled(!checked);
     ui->controlPortTxt->setEnabled(!checked);
     ui->usernameTxt->setEnabled(!checked);
@@ -3572,7 +3606,7 @@ void wfmain::on_serialEnableBtn_clicked(bool checked)
 void wfmain::on_lanEnableBtn_clicked(bool checked)
 {
     prefs.enableLAN = checked;
-    ui->connectBtn->setEnabled(checked);
+    ui->connectBtn->setEnabled(true);
     ui->ipAddressTxt->setEnabled(checked);
     ui->controlPortTxt->setEnabled(checked);
     ui->usernameTxt->setEnabled(checked);
@@ -4447,10 +4481,15 @@ void wfmain::on_baudRateCombo_activated(int index)
     (void)index;
 }
 
+void wfmain::on_wfLengthSlider_valueChanged(int value)
+{
+    prepareWf(value);
+}
+
 // --- DEBUG FUNCTION ---
 void wfmain::on_debugBtn_clicked()
 {
     qInfo(logSystem()) << "Debug button pressed.";
-    qInfo(logSystem()) << "getting mode.";
-    emit getRigID();
+    prepareWf(160);
+
 }
diff --git a/wfmain.h b/wfmain.h
index e0a55688..71f1fabb 100644
--- a/wfmain.h
+++ b/wfmain.h
@@ -45,7 +45,7 @@ public:
 
 signals:
     // Basic to rig:
-    void setCIVAddr(unsigned char);
+    void setCIVAddr(unsigned char newRigCIVAddr);
 
     // Power
     void sendPowerOn();
@@ -446,6 +446,8 @@ private slots:
 
     void on_baudRateCombo_activated(int);
 
+    void on_wfLengthSlider_valueChanged(int value);
+
 private:
     Ui::wfmain *ui;
     void closeEvent(QCloseEvent *event);
@@ -459,6 +461,7 @@ private:
     void setAppTheme(bool isCustom);
     void setPlotTheme(QCustomPlot *plot, bool isDark);
     void prepareWf();
+    void prepareWf(unsigned int wfLength);
     void getInitialRigState();
     void setBandButtons();
     void showButton(QPushButton *btn);
@@ -543,8 +546,7 @@ private:
 
     quint16 spectWidth;
     quint16 wfLength;
-
-    quint16 spectRowCurrent;
+    bool spectrumDrawLock;
 
     QByteArray spectrumPeaks;
 
diff --git a/wfmain.ui b/wfmain.ui
index 33cf6f9a..d2a6adea 100644
--- a/wfmain.ui
+++ b/wfmain.ui
@@ -18,7 +18,7 @@
     
      
       
-       3
+       0
       
       
        
@@ -767,6 +767,60 @@
             
            
           
+          
+           
+            
+             0
+            
+            
+             
+              
+               
+                0
+                70
+               
+              
+              
+               
+                16777215
+                80
+               
+              
+              
+               Waterfall Length
+              
+              
+               Waterfall Length
+              
+              
+               100
+              
+              
+               1024
+              
+              
+               160
+              
+              
+               Qt::Vertical
+              
+             
+            
+            
+             
+              
+               
+                16777215
+                15
+               
+              
+              
+               Len
+              
+             
+            
+           
+          
           
            
             

From 4279d0d2b29a866b49d80a416d10033b43cd69a7 Mon Sep 17 00:00:00 2001
From: Elliott Liggett 
Date: Sun, 30 May 2021 23:35:30 -0700
Subject: [PATCH 42/47] Removing my own uninformed sidenote.

---
 wfmain.cpp | 3 ---
 1 file changed, 3 deletions(-)

diff --git a/wfmain.cpp b/wfmain.cpp
index 2e35fdc7..75d3ea6e 100644
--- a/wfmain.cpp
+++ b/wfmain.cpp
@@ -1450,9 +1450,6 @@ void wfmain::prepareWf(unsigned int wfLength)
         {
             unsigned int i=0;
             unsigned int oldSize = wfimage.size();
-            // Note: apparently .append() in this case does not add to the initial size.
-            // So if the initial size is 100 and you append 50 times,
-            // the end size is 50.
             for(i=oldSize; i<(wfLength); i++)
             {
                 wfimage.append(empty);

From 774685037edaaa2974584aa2463594b33b4c1d72 Mon Sep 17 00:00:00 2001
From: Elliott Liggett 
Date: Sun, 30 May 2021 23:42:45 -0700
Subject: [PATCH 43/47] Keep the theme during resize. TODO: preference for wf
 theme

---
 wfmain.cpp | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/wfmain.cpp b/wfmain.cpp
index 75d3ea6e..7197333a 100644
--- a/wfmain.cpp
+++ b/wfmain.cpp
@@ -1464,7 +1464,7 @@ void wfmain::prepareWf(unsigned int wfLength)
         colorMap->data()->setValueRange(QCPRange(0, wfLength-1));
         colorMap->data()->setKeyRange(QCPRange(0, spectWidth-1));
         colorMap->setDataRange(QCPRange(0, rigCaps.spectAmpMax));
-        colorMap->setGradient(QCPColorGradient::gpJet); // TODO: Add preference
+        colorMap->setGradient(static_cast(ui->wfthemeCombo->currentData().toInt()));
 
         if(colorMapData == Q_NULLPTR)
         {

From dec4ebe4afab4ffc8bb6a77ae39907027155e5ba Mon Sep 17 00:00:00 2001
From: Phil Taylor 
Date: Mon, 31 May 2021 10:18:37 +0100
Subject: [PATCH 44/47] Fix compile issue after merge

---
 wfmain.cpp | 7 ++++---
 1 file changed, 4 insertions(+), 3 deletions(-)

diff --git a/wfmain.cpp b/wfmain.cpp
index c06a3fa0..011b636a 100644
--- a/wfmain.cpp
+++ b/wfmain.cpp
@@ -769,6 +769,7 @@ void wfmain::setServerToPrefs()
             serverConfig.baudRate = prefs.serialPortBaud;
             serverConfig.audioInput = udpPrefs.audioInput;
             serverConfig.audioOutput = udpPrefs.audioOutput;
+        }
 
         udp = new udpServer(serverConfig);
 
@@ -1204,7 +1205,7 @@ void wfmain::loadSettings()
     udpPrefs.audioRXSampleRate = settings->value("AudioRXSampleRate", udpDefPrefs.audioRXSampleRate).toInt();
     udpPrefs.audioTXSampleRate = settings->value("AudioTXSampleRate",udpDefPrefs.audioTXSampleRate).toInt();
     ui->audioSampleRateCombo->setEnabled(ui->lanEnableBtn->isChecked());
-    int audioSampleRateIndex = ui->audioSampleRateCombo->findText(QString::number(udpDefPrefs.audioRXSampleRate));
+    int audioSampleRateIndex = ui->audioSampleRateCombo->findText(QString::number(udpPrefs.audioRXSampleRate));
     if (audioSampleRateIndex != -1) {
         ui->audioSampleRateCombo->setCurrentIndex(audioSampleRateIndex);
     }
@@ -1358,7 +1359,7 @@ void wfmain::saveSettings()
     settings->setValue("AudioTXLatency", udpPrefs.audioTXLatency);
     settings->setValue("AudioRXSampleRate", udpPrefs.audioRXSampleRate);
     settings->setValue("AudioRXCodec", udpPrefs.audioRXCodec);
-    settings->setValue("AudioTXSampleRate", udpPrefs.audioRXSampleRate);
+    settings->setValue("AudioTXSampleRate", udpPrefs.audioTXSampleRate);
     settings->setValue("AudioTXCodec", udpPrefs.audioTXCodec);
     settings->setValue("AudioOutput", udpPrefs.audioOutputName);
     settings->setValue("AudioInput", udpPrefs.audioInputName);
@@ -1486,7 +1487,7 @@ void wfmain::prepareWf(unsigned int wfLength)
             wfimage.remove(wfLength, wfimage.size()-wfLength);
         }
         wfimage.squeeze();
-        colorMap->clearData();
+        //colorMap->clearData();
         colorMap->data()->clear();
 
         colorMap->data()->setValueRange(QCPRange(0, wfLength-1));

From aa7a57f89c54d529ef4852ad7dd4ba769a560e71 Mon Sep 17 00:00:00 2001
From: Phil Taylor 
Date: Mon, 31 May 2021 17:55:01 +0100
Subject: [PATCH 45/47] Fix for crashing pty on mac

---
 audiohandler.cpp |  1 +
 pttyhandler.cpp  | 19 ++++++++++++++-----
 udphandler.cpp   | 10 +++++++---
 3 files changed, 22 insertions(+), 8 deletions(-)

diff --git a/audiohandler.cpp b/audiohandler.cpp
index 5441db32..1b3e0016 100644
--- a/audiohandler.cpp
+++ b/audiohandler.cpp
@@ -145,6 +145,7 @@ bool audioHandler::init(const quint8 bits, const quint8 radioChan, const quint16
 	wf_resampler_get_ratio(resampler, &ratioNum, &ratioDen);
 	qInfo(logAudio()) << (isInput ? "Input" : "Output") << "wf_resampler_init() returned: " << resample_error << " ratioNum" << ratioNum << " ratioDen" << ratioDen;
 
+    qInfo(logAudio()) << (isInput ? "Input" : "Output") << "thread id" << QThread::currentThreadId();
 	return isInitialized;
 }
 
diff --git a/pttyhandler.cpp b/pttyhandler.cpp
index d24f5f0a..f39f9043 100644
--- a/pttyhandler.cpp
+++ b/pttyhandler.cpp
@@ -15,7 +15,7 @@
 pttyHandler::pttyHandler(QString pty)
 {
     //constructor
-    if (pty == "" || pty == "None")
+    if (pty == "" || pty.toLower() == "none")
     {
         // Just return if pty is not configured.
         return;
@@ -94,7 +94,7 @@ void pttyHandler::openPort()
 #ifndef Q_OS_WIN
     ptDevSlave = QString::fromLocal8Bit(ptsname(ptfd));
 
-    if (portName != "" && portName != "None")
+    if (portName != "" && portName.toLower() != "none")
     {
         if (!QFile::link(ptDevSlave, portName))
         {
@@ -115,7 +115,16 @@ pttyHandler::~pttyHandler()
 
 void pttyHandler::receiveDataFromRigToPtty(const QByteArray& data)
 {
-    if (isConnected && (unsigned char)data[2] != (unsigned char)0xE1 && (unsigned char)data[3] != (unsigned char)0xE1)
+    int fePos=data.lastIndexOf((char)0xfe);
+    if (fePos>0)
+        fePos=fePos-1;
+    else
+    {
+        qDebug(logSerial()) << "Invalid command";
+        printHex(data,false,true);
+    }
+
+    if (isConnected && (unsigned char)data.mid(fePos)[2] != (unsigned char)0xE1 && (unsigned char)data.mid(fePos)[3] != (unsigned char)0xE1)
     {
         // send to the pseudo port as well
         // index 2 is dest, 0xE1 is wfview, 0xE0 is assumed to be the other device.
@@ -124,7 +133,7 @@ void pttyHandler::receiveDataFromRigToPtty(const QByteArray& data)
         // 0xE0 = pseudo-term host
         // 0x00 = broadcast to all
         //qInfo(logSerial()) << "Sending data from radio to pseudo-terminal";
-        sendDataOut(data);
+        sendDataOut(data.mid(fePos));
     }
 }
 
@@ -258,7 +267,7 @@ void pttyHandler::closePort()
         delete port;
     }
 #else
-    if (isConnected && portName != "" && portName != "None")
+    if (isConnected && portName != "" && portName.toLower() != "none")
     {
         QFile::remove(portName);
     }
diff --git a/udphandler.cpp b/udphandler.cpp
index 701dafa1..d2fde850 100644
--- a/udphandler.cpp
+++ b/udphandler.cpp
@@ -590,7 +590,7 @@ void udpCivData::watchdog()
 
 void udpCivData::send(QByteArray d)
 {
-    // qInfo(logUdp()) << "Sending: (" << d.length() << ") " << d;
+    //qInfo(logUdp()) << "Sending: (" << d.length() << ") " << d;
     data_packet p;
     memset(p.packet, 0x0, sizeof(p)); // We can't be sure it is initialized with 0x00!
     p.len = sizeof(p)+d.length();
@@ -642,6 +642,7 @@ void udpCivData::dataReceived()
         //qInfo(logUdp()) << "Received: " << datagram.data();
         QByteArray r = datagram.data();
 
+
         switch (r.length())
         {
             case (CONTROL_SIZE): // Control packet
@@ -678,7 +679,9 @@ void udpCivData::dataReceived()
                         lastReceived = QTime::currentTime();
                         if (quint16(in->datalen + 0x15) == (quint16)in->len)
                         {
-                            emit receive(r.mid(0x15));
+                            //if (r.mid(0x15).length() != 157)
+                               emit receive(r.mid(0x15));
+                            //qDebug(logUdp()) << "Got incoming CIV datagram" << r.mid(0x15).length();
                         }
 
                     }
@@ -687,6 +690,7 @@ void udpCivData::dataReceived()
             }
         }
         udpBase::dataReceived(r); // Call parent function to process the rest.
+
         r.clear();
         datagram.clear();
 
@@ -852,7 +856,7 @@ void udpAudio::watchdog()
             /* Just log it at the moment, maybe try signalling the control channel that it needs to 
                 try requesting civ/audio again? */
 
-            qInfo(logUdp()) << " Audio Watchdog: no audio data received for 2s, restart required";
+            qInfo(logUdp()) << " Audio Watchdog: no audio data received for 2s, restart required?";
             alerted = true;
         }
     }

From c778e1839c5b76964430ea955cc9d26e9103e51e Mon Sep 17 00:00:00 2001
From: Phil Taylor 
Date: Mon, 31 May 2021 18:11:24 +0100
Subject: [PATCH 46/47] Open pty non-blocking

---
 pttyhandler.cpp | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

diff --git a/pttyhandler.cpp b/pttyhandler.cpp
index f39f9043..61551e56 100644
--- a/pttyhandler.cpp
+++ b/pttyhandler.cpp
@@ -53,7 +53,7 @@ void pttyHandler::openPort()
     }
 #else
     // Generic method in Linux/MacOS to find a pty
-    ptfd = ::posix_openpt(O_RDWR | O_NOCTTY);
+    ptfd = ::posix_openpt(O_RDWR | O_NONBLOCK);
 
     if (ptfd >=0)
     {
@@ -115,6 +115,7 @@ pttyHandler::~pttyHandler()
 
 void pttyHandler::receiveDataFromRigToPtty(const QByteArray& data)
 {
+
     int fePos=data.lastIndexOf((char)0xfe);
     if (fePos>0)
         fePos=fePos-1;
@@ -133,7 +134,7 @@ void pttyHandler::receiveDataFromRigToPtty(const QByteArray& data)
         // 0xE0 = pseudo-term host
         // 0x00 = broadcast to all
         //qInfo(logSerial()) << "Sending data from radio to pseudo-terminal";
-        sendDataOut(data.mid(fePos));
+        sendDataOut(data);
     }
 }
 

From 28c84d1765779fef91dbad09d92e0044e7681136 Mon Sep 17 00:00:00 2001
From: Phil Taylor 
Date: Tue, 1 Jun 2021 00:25:20 +0100
Subject: [PATCH 47/47] Fix for crash when remote requests tone.

---
 rigcommander.cpp | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/rigcommander.cpp b/rigcommander.cpp
index a96378ad..2539f396 100644
--- a/rigcommander.cpp
+++ b/rigcommander.cpp
@@ -840,6 +840,9 @@ quint16 rigCommander::decodeTone(QByteArray eTone, bool &tinv, bool &rinv)
     // CTCSS:  1B 01  00 12 73 = PL 127.3, decode as 1273
     // D(T)CS: 1B 01  TR 01 23 = T/R Invert bits + DCS code 123
 
+    if (eTone.length() < 5) {
+        return 0;
+    }
     tinv = false; rinv = false;
     quint16 result = 0;