144 lines
5.4 KiB
Python
144 lines
5.4 KiB
Python
import sys
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from unittest.mock import MagicMock
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# Comprehensive workaround for missing _lzma in some Python builds
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try:
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import lzma
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except ImportError:
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mock_lzma = MagicMock()
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mock_lzma.FORMAT_XZ = 1
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mock_lzma.FORMAT_ALONE = 2
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mock_lzma.FORMAT_RAW = 3
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mock_lzma.CHECK_NONE = 0
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mock_lzma.CHECK_CRC32 = 1
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mock_lzma.CHECK_CRC64 = 4
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mock_lzma.CHECK_SHA256 = 10
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sys.modules["_lzma"] = MagicMock()
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sys.modules["lzma"] = mock_lzma
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import sys
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import time
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import numpy as np
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import sounddevice as sd
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import torch
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import mlx_whisper
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from silero_vad import load_silero_vad, get_speech_timestamps
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import queue
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import argparse
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# Audio Constants
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SAMPLERATE = 16000
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BLOCK_SIZE = 512
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VAD_THRESHOLD = 0.5
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def transcribe_with_controls(audio, transcribe_kwargs):
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"""Call mlx_whisper.transcribe with graceful fallback for unsupported args."""
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try:
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return mlx_whisper.transcribe(audio, **transcribe_kwargs)
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except Exception as exc:
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# Fallback logic for beam_size or other specific MLX implementation gaps
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if "beam_size" in transcribe_kwargs:
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transcribe_kwargs.pop("beam_size")
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return mlx_whisper.transcribe(audio, **transcribe_kwargs)
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raise exc
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def audio_callback(indata, frames, time, status, audio_queue):
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if status:
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print(status, file=sys.stderr)
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audio_queue.put(indata.copy())
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def run_transcription(out_queue, args):
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device = "mps" if torch.backends.mps.is_available() else "cpu"
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print(f"[Transcribe] Loading Whisper model '{args.model}' on {device}...")
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vad_model = load_silero_vad()
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audio_queue = queue.Queue()
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# Selection of device
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device_index = args.device
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def callback_wrapper(indata, frames, time, status):
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audio_callback(indata, frames, time, status, audio_queue)
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audio_buffer = []
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speech_started = False
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buffer_limit = SAMPLERATE * args.max_buffer
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print(f"[Transcribe] Starting audio stream on device {device_index}...")
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try:
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with sd.InputStream(samplerate=SAMPLERATE, channels=args.channels, callback=callback_wrapper, blocksize=BLOCK_SIZE, device=device_index):
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while True:
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while not audio_queue.empty():
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data = audio_queue.get()
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if args.channels > 1:
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data = np.mean(data, axis=1) # Mix to mono
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audio_buffer.append(data.flatten())
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if audio_buffer:
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current_audio = np.concatenate(audio_buffer)
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audio_tensor = torch.from_numpy(current_audio)
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buffer_duration = len(current_audio) / SAMPLERATE
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speech_timestamps = get_speech_timestamps(
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audio_tensor,
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vad_model,
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sampling_rate=SAMPLERATE,
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threshold=VAD_THRESHOLD,
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min_silence_duration_ms=args.silence
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)
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if speech_timestamps:
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speech_started = True
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last_end = speech_timestamps[-1]['end']
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buffer_len_samples = len(current_audio)
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if (buffer_len_samples - last_end) > (SAMPLERATE * args.silence / 1000) or buffer_len_samples > buffer_limit:
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# 1. Transcribe
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transcribe_kwargs = {
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"path_or_hf_repo": args.model,
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"temperature": (0.0, 0.2, 0.4, 0.6, 0.8, 1.0),
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}
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if args.lang:
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transcribe_kwargs["language"] = args.lang
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result = transcribe_with_controls(current_audio, transcribe_kwargs)
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text = result['text'].strip()
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detected_lang = result.get('language', 'en')
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if text:
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# Send to Translation Process
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out_queue.put({
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"original": text,
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"detected_lang": detected_lang,
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"ts": time.time()
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})
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print(f"[Transcribe] {detected_lang.upper()}: {text}")
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audio_buffer = []
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speech_started = False
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elif not speech_started and len(current_audio) > SAMPLERATE * 2:
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audio_buffer = []
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except KeyboardInterrupt:
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print("[Transcribe] Stopped.")
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except Exception as e:
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print(f"[Transcribe] Error: {e}")
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if __name__ == "__main__":
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parser = argparse.ArgumentParser()
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parser.add_argument("--model", type=str, default="mlx-community/whisper-base-mlx")
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parser.add_argument("--device", type=int, default=None)
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parser.add_argument("--lang", type=str, default=None)
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parser.add_argument("--silence", type=int, default=1000)
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parser.add_argument("--max-buffer", type=int, default=20)
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parser.add_argument("--channels", type=int, default=1)
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args = parser.parse_args()
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# This part would normally be called by the main coordinator
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# For testing, we can use a dummy queue
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import multiprocessing
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q = multiprocessing.Queue()
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run_transcription(q, args)
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