mirror of
https://github.com/QuentinFuxa/WhisperLiveKit.git
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Merge branch 'whisper-mlx'
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@@ -156,6 +156,63 @@ class FasterWhisperASR(ASRBase):
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def set_translate_task(self):
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self.transcribe_kargs["task"] = "translate"
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class MLXWhisper(ASRBase):
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"""
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Uses MPX Whisper library as the backend, optimized for Apple Silicon.
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Models available: https://huggingface.co/collections/mlx-community/whisper-663256f9964fbb1177db93dc
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Significantly faster than faster-whisper (without CUDA) on Apple M1. Model used by default: mlx-community/whisper-large-v3-mlx
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"""
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sep = " "
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def load_model(self, modelsize=None, model_dir=None):
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from mlx_whisper import transcribe
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if model_dir is not None:
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logger.debug(f"Loading whisper model from model_dir {model_dir}. modelsize parameter is not used.")
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model_size_or_path = model_dir
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elif modelsize is not None:
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logger.debug(f"Loading whisper model {modelsize}. You use mlx whisper, so make sure you use a mlx-compatible model.")
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model_size_or_path = modelsize
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elif modelsize == None:
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logger.debug("No model size or path specified. Using mlx-community/whisper-large-v3-mlx.")
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model_size_or_path = "mlx-community/whisper-large-v3-mlx"
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self.model_size_or_path = model_size_or_path
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return transcribe
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def transcribe(self, audio, init_prompt=""):
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segments = self.model(
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audio,
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language=self.original_language,
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initial_prompt=init_prompt,
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word_timestamps=True,
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condition_on_previous_text=True,
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path_or_hf_repo=self.model_size_or_path,
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**self.transcribe_kargs
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)
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return segments.get("segments", [])
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def ts_words(self, segments):
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"""
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Extract timestamped words from transcription segments and skips words with high no-speech probability.
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"""
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return [
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(word["start"], word["end"], word["word"])
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for segment in segments
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for word in segment.get("words", [])
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if segment.get("no_speech_prob", 0) <= 0.9
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]
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def segments_end_ts(self, res):
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return [s['end'] for s in res]
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def use_vad(self):
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self.transcribe_kargs["vad_filter"] = True
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def set_translate_task(self):
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self.transcribe_kargs["task"] = "translate"
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class OpenaiApiASR(ASRBase):
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"""Uses OpenAI's Whisper API for audio transcription."""
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@@ -660,7 +717,7 @@ def add_shared_args(parser):
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parser.add_argument('--model_dir', type=str, default=None, help="Dir where Whisper model.bin and other files are saved. This option overrides --model and --model_cache_dir parameter.")
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parser.add_argument('--lan', '--language', type=str, default='auto', help="Source language code, e.g. en,de,cs, or 'auto' for language detection.")
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parser.add_argument('--task', type=str, default='transcribe', choices=["transcribe","translate"],help="Transcribe or translate.")
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parser.add_argument('--backend', type=str, default="faster-whisper", choices=["faster-whisper", "whisper_timestamped", "openai-api"],help='Load only this backend for Whisper processing.')
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parser.add_argument('--backend', type=str, default="faster-whisper", choices=["faster-whisper", "whisper_timestamped", "mlx-whisper", "openai-api"],help='Load only this backend for Whisper processing.')
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parser.add_argument('--vac', action="store_true", default=False, help='Use VAC = voice activity controller. Recommended. Requires torch.')
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parser.add_argument('--vac-chunk-size', type=float, default=0.04, help='VAC sample size in seconds.')
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parser.add_argument('--vad', action="store_true", default=False, help='Use VAD = voice activity detection, with the default parameters.')
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@@ -679,6 +736,8 @@ def asr_factory(args, logfile=sys.stderr):
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else:
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if backend == "faster-whisper":
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asr_cls = FasterWhisperASR
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elif backend == "mlx-whisper":
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asr_cls = MLXWhisper
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else:
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asr_cls = WhisperTimestampedASR
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