diff --git a/README.md b/README.md
index 3558195..b89b413 100644
--- a/README.md
+++ b/README.md
@@ -2,6 +2,13 @@
 
 Generates samples using [Piper](https://github.com/rhasspy/piper/) for training a wake word system like [openWakeWord](https://github.com/dscripka/openWakeWord).
 
+Available models:
+
+* [English](https://github.com/rhasspy/piper-sample-generator/releases/download/v2.0.0/en_US-libritts_r-medium.pt)
+* [French](https://github.com/rhasspy/piper-sample-generator/releases/download/v2.0.0/fr_FR-mls-medium.pt)
+* [German](https://github.com/rhasspy/piper-sample-generator/releases/download/v2.0.0/de_DE-mls-medium.pt)
+* [Dutch](https://github.com/rhasspy/piper-sample-generator/releases/download/v2.0.0/nl_NL-mls-medium.pt)
+
 
 ## Install
 
@@ -23,6 +30,7 @@ Download the LibriTTS-R generator (exported from [checkpoint](https://huggingfac
 wget -O models/en-us-libritts-high.pt 'https://github.com/rhasspy/piper-sample-generator/releases/download/v2.0.0/en_US-libritts_r-medium.pt'
 ```
 
+See links above for models for other languages.
 
 ## Run
 
@@ -72,3 +80,28 @@ This will do several things to each sample:
     * Change the acoustics of the sample to sound like the speaker was in a room with echo or using a poor quality microphone
 3. Resample to 16Khz for training (e.g., [openWakeWord](https://github.com/dscripka/openWakeWord))
 
+
+## Short Phrases
+
+Models that were trained on audio books tend to perform poorly when speaking short phrases or single words.
+The French, German, and Dutch models trained from the [MLS](http://openslr.org/94/) have this problem.
+
+The problem can be mitigated by repeating the phrase over and over, and then clipping out a single sample.
+To do this automatically, follow these steps:
+
+1. Ensure your short phrase ends with a comma (`<phrase>,`)
+2. Lower the noise settings with `--noise-scales 0.333` and `--noise-scale-ws 0.333`
+3. Use `--min-phoneme-count 300` (the value 300 was determined empirically and may be less for some models)
+
+For example:
+
+``` sh
+python3 generate_samples.py \
+    'framboise,' \
+    --model models/fr_FR-mls-medium.pt \
+    --noise-scales 0.333 \
+    --noise-scale-ws 0.333 \
+    --min-phoneme-count 300 
+    --max-samples 1 \
+    --output-dir . 
+```
diff --git a/generate_samples.py b/generate_samples.py
index 349ebde..d0a6e85 100755
--- a/generate_samples.py
+++ b/generate_samples.py
@@ -7,7 +7,7 @@ import logging
 import os
 import wave
 from pathlib import Path
-from typing import List, Union
+from typing import Any, Dict, List, Optional, Tuple, Union
 
 import numpy as np
 import torch
@@ -24,21 +24,20 @@ logging.basicConfig(level=logging.DEBUG)
 
 # Main generation function
 def generate_samples(
-    text: Union[List, str],
-    output_dir: str,
-    max_samples: int = None,
-    file_names: List[str] = [],
-    model: str = os.path.join(
-        Path(__file__).parent, "models", "en_US-libritts_r-medium.pt"
-    ),
+    text: Union[List[str], str],
+    output_dir: Union[str, Path],
+    max_samples: Optional[int] = None,
+    file_names: Optional[List[str]] = None,
+    model: Union[str, Path] = _DIR / "models" / "en_US-libritts_r-medium.pt",
     batch_size: int = 1,
-    slerp_weights: List[float] = [0.5],
-    length_scales: List[float] = [0.75, 1, 1.25],
-    noise_scales: List[float] = [0.667],
-    noise_scale_ws: List[float] = [0.8],
-    max_speakers: float = None,
+    slerp_weights: Tuple[float, ...] = (0.5,),
+    length_scales: Tuple[float, ...] = (0.75, 1, 1.25),
+    noise_scales: Tuple[float, ...] = (0.667,),
+    noise_scale_ws: Tuple[float, ...] = (0.8,),
+    max_speakers: Optional[float] = None,
     verbose: bool = False,
     auto_reduce_batch_size: bool = False,
+    min_phoneme_count: Optional[int] = None,
     **kwargs,
 ) -> None:
     """
@@ -61,6 +60,8 @@ def generate_samples(
         verbose (bool): Enable or disable more detailed logging messages (default: False).
         auto_reduce_batch_size (bool): Automatically and temporarily reduce the batch size
                                        if CUDA OOM errors are detected, and try to resume generation.
+        min_phoneme_count (int): If set, ensure this number of phonemes is always sent to the model.
+                                 Clip audio to extract original phrase.
 
     Returns:
         None
@@ -71,12 +72,13 @@ def generate_samples(
 
     _LOGGER.debug("Loading %s", model)
     model_path = Path(model)
-    model = torch.load(model_path)
-    model.eval()
+
+    torch_model = torch.load(model_path)
+    torch_model.eval()
     _LOGGER.info("Successfully loaded the model")
 
     if torch.cuda.is_available():
-        model.cuda()
+        torch_model.cuda()
         _LOGGER.debug("CUDA available, using GPU")
 
     output_dir = Path(output_dir)
@@ -147,10 +149,14 @@ def generate_samples(
             speaker_1 = torch.LongTensor([s[0] for s in speakers_batch])
             speaker_2 = torch.LongTensor([s[1] for s in speakers_batch])
 
-            phoneme_ids = [
-                get_phonemes(voice, config, next(texts), verbose)
-                for i in range(batch_size)
-            ]
+            phoneme_ids_by_batch = []
+            clip_indexes_by_batch = []
+            for i in range(batch_size):
+                phoneme_ids, clip_phoneme_index = get_phonemes(
+                    voice, config, next(texts), verbose, min_phoneme_count
+                )
+                phoneme_ids_by_batch.append(phoneme_ids)
+                clip_indexes_by_batch.append(clip_phoneme_index)
 
             def right_pad_lists(lists):
                 max_length = max(len(lst) for lst in lists)
@@ -162,18 +168,18 @@ def generate_samples(
                     padded_lists.append(padded_l)
                 return padded_lists
 
-            phoneme_ids = right_pad_lists(phoneme_ids)
+            phoneme_ids_by_batch = right_pad_lists(phoneme_ids_by_batch)
 
             if auto_reduce_batch_size:
                 oom_error = True
                 counter = 1
                 while oom_error is True:
                     try:
-                        audio = generate_audio(
-                            model,
+                        audio, phoneme_samples = generate_audio(
+                            torch_model,
                             speaker_1[0 : batch_size // counter],
                             speaker_2[0 : batch_size // counter],
-                            phoneme_ids[0 : batch_size // counter],
+                            phoneme_ids_by_batch[0 : batch_size // counter],
                             slerp_weight,
                             noise_scale,
                             noise_scale_w,
@@ -186,11 +192,11 @@ def generate_samples(
                         gc.collect()
                         counter += 1  # reduce batch size to avoid OOM errors
             else:
-                audio = generate_audio(
-                    model,
+                audio, phoneme_samples = generate_audio(
+                    torch_model,
                     speaker_1,
                     speaker_2,
-                    phoneme_ids,
+                    phoneme_ids_by_batch,
                     slerp_weight,
                     noise_scale,
                     noise_scale_w,
@@ -198,19 +204,34 @@ def generate_samples(
                     max_len,
                 )
 
+            # Clip audio when using min_phoneme_count
+            for i, clip_phoneme_index in enumerate(clip_indexes_by_batch):
+                if clip_phoneme_index is not None:
+                    last_sample_idx = int(
+                        phoneme_samples[i].flatten()[clip_phoneme_index:].sum().item()
+                    )
+
+                    # Fill remainder of audio with silence.
+                    # It will be removed in the next stage.
+                    audio[i, 0, :-last_sample_idx] = 0
+
             # Resample audio
             audio = resampler(audio.cpu()).numpy()
 
             audio_int16 = audio_float_to_int16(audio)
             for audio_idx in range(audio_int16.shape[0]):
                 # Use webrtcvad to trip silence from the clips
-                audio_data = remove_silence(audio_int16[audio_idx].flatten())[None,]
+                audio_data = remove_silence(audio_int16[audio_idx].flatten())[
+                    None,
+                ]
 
                 if isinstance(file_names, it.cycle):
                     wav_path = output_dir / next(file_names)
                 else:
                     wav_path = output_dir / f"{sample_idx}.wav"
-                with wave.open(str(wav_path), "wb") as wav_file:
+
+                wav_file: wave.Wave_write = wave.open(str(wav_path), "wb")
+                with wav_file:
                     wav_file.setframerate(resample_rate)
                     wav_file.setsampwidth(2)
                     wav_file.setnchannels(1)
@@ -224,17 +245,22 @@ def generate_samples(
             # print(f"Batch {batch_idx +1}/{max_samples//batch_size} complete", " "*200, end='\r')
 
         # Next batch
-        _LOGGER.debug(f"Batch {batch_idx +1}/{max_samples//batch_size} complete")
+        _LOGGER.debug("Batch %s/%s complete", batch_idx + 1, max_samples // batch_size)
         speakers_batch = list(it.islice(speakers_iter, 0, batch_size))
         batch_idx += 1
 
     _LOGGER.info("Done")
 
 
-def remove_silence(x, frame_duration=0.030, sample_rate=16000, min_start=2000):
+def remove_silence(
+    x: np.ndarray,
+    frame_duration: float = 0.030,
+    sample_rate: int = 16000,
+    min_start: int = 2000,
+) -> np.ndarray:
     """Uses webrtc voice activity detection to remove silence from the clips"""
     vad = webrtcvad.Vad(0)
-    if x.dtype == np.float32 or x.dtype == np.float64:
+    if x.dtype in (np.float32, np.float64):
         x = (x * 32767).astype(np.int16)
     x_new = x[0:min_start].tolist()
     step_size = int(sample_rate * frame_duration)
@@ -295,10 +321,18 @@ def generate_audio(
     o = model.dec((z * y_mask)[:, :, :max_len], g=g)
 
     audio = o
-    return audio
+    phoneme_samples = w_ceil * 256  # hop length
+
+    return audio, phoneme_samples
 
 
-def get_phonemes(voice, config, text, verbose):
+def get_phonemes(
+    voice: str,
+    config: Dict[str, Any],
+    text: str,
+    verbose: bool = False,
+    min_phoneme_count: Optional[int] = None,
+) -> Tuple[List[int], Optional[int]]:
     # Combine all sentences
     phonemes = [
         p
@@ -309,18 +343,41 @@ def get_phonemes(voice, config, text, verbose):
         _LOGGER.debug("Phonemes: %s", phonemes)
 
     id_map = config["phoneme_id_map"]
+
+    # Beginning of utterance
     phoneme_ids = list(id_map["^"])
+
+    # Phoneme ids for just the text
+    text_phoneme_ids = []
+
     for phoneme in phonemes:
         p_ids = id_map.get(phoneme)
         if p_ids is not None:
             phoneme_ids.extend(p_ids)
+            text_phoneme_ids.extend(p_ids)
             phoneme_ids.extend(id_map["_"])
+            text_phoneme_ids.extend(id_map["_"])
 
+    # Index where audio should be clipped at.
+    # When None, all of the audio will be used.
+    clip_phoneme_index: Optional[int] = None
+
+    if min_phoneme_count is not None:
+        # Repeat phrase until minimum phoneme count is met.
+        # NOTE: It is critical that the ^ and $ phonemes are not repeated here.
+        while (len(phoneme_ids) - 1) < min_phoneme_count:
+            # We will clip audio at the beginning of the last phrase
+            clip_phoneme_index = len(phoneme_ids) - 1
+
+            phoneme_ids.extend(text_phoneme_ids)
+
+    # End of utterance
     phoneme_ids.extend(id_map["$"])
-    return phoneme_ids
+
+    return phoneme_ids, clip_phoneme_index
 
 
-def slerp(v1, v2, t, DOT_THR=0.9995, zdim=-1):
+def slerp(v1, v2, t: float, DOT_THR: float = 0.9995, zdim: int = -1):
     """SLERP for pytorch tensors interpolating `v1` to `v2` with scale of `t`.
 
     `DOT_THR` determines when the vectors are too close to parallel.
@@ -375,7 +432,9 @@ def audio_float_to_int16(
     return audio_norm
 
 
-if __name__ == "__main__":
+def main() -> None:
+    """Main entry point."""
+
     # Get command line arguments
     parser = argparse.ArgumentParser()
     parser.add_argument("text")
@@ -401,7 +460,13 @@ if __name__ == "__main__":
         type=int,
         help="Maximum number of speakers to use (default: all)",
     )
+    parser.add_argument("--min-phoneme-count", type=int)
+    parser.add_argument("--verbose", action="store_true")
     args = parser.parse_args().__dict__
 
     # Generate speech
     generate_samples(**args)
+
+
+if __name__ == "__main__":
+    main()
diff --git a/pylintrc b/pylintrc
index 22a70d0..561b2f1 100644
--- a/pylintrc
+++ b/pylintrc
@@ -35,3 +35,10 @@ disable=
 
 [FORMAT]
 expected-line-ending-format=LF
+
+[TYPECHECK]
+
+# List of members which are set dynamically and missed by pylint inference
+# system, and so shouldn't trigger E1101 when accessed. Python regular
+# expressions are accepted.
+generated-members=numpy.*,torch.*
diff --git a/requirements.txt b/requirements.txt
index c6ac75a..8443d84 100644
--- a/requirements.txt
+++ b/requirements.txt
@@ -1,6 +1,6 @@
 audiomentations==0.33.0
 piper-phonemize==1.1.0
 numpy<2
-torch
+torch<2
 torchaudio
 webrtcvad