Refactor tool-speechtotext: extract sttlib shared library and add tests
Extract duplicated code (Whisper loading, audio recording, transcription, VAD processing) into reusable sttlib/ package. Rewrite all 3 scripts as thin wrappers. Add 24 unit tests with mocked hardware. Fix GPU fallback bug in assistant.py and args.system assignment bug.
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58
python/tool-speechtotext/sttlib/vad.py
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58
python/tool-speechtotext/sttlib/vad.py
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import sys
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import queue
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import collections
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import webrtcvad
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SAMPLE_RATE = 16000
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CHANNELS = 1
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FRAME_DURATION_MS = 30
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FRAME_SIZE = int(SAMPLE_RATE * FRAME_DURATION_MS / 1000) # 480 samples
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MIN_UTTERANCE_FRAMES = 10 # ~300ms minimum to filter coughs/clicks
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audio_queue = queue.Queue()
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def audio_callback(indata, frames, time_info, status):
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"""sounddevice callback that pushes raw bytes to the audio queue."""
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if status:
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print(status, file=sys.stderr)
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audio_queue.put(bytes(indata))
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class VADProcessor:
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def __init__(self, aggressiveness, silence_threshold):
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self.vad = webrtcvad.Vad(aggressiveness)
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self.silence_threshold = silence_threshold
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self.reset()
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def reset(self):
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self.triggered = False
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self.utterance_frames = []
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self.silence_duration = 0.0
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self.pre_buffer = collections.deque(maxlen=10) # ~300ms pre-roll
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def process_frame(self, frame_bytes):
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"""Process one 30ms frame. Returns utterance bytes when complete, else None."""
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is_speech = self.vad.is_speech(frame_bytes, SAMPLE_RATE)
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if not self.triggered:
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self.pre_buffer.append(frame_bytes)
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if is_speech:
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self.triggered = True
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self.silence_duration = 0.0
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self.utterance_frames = list(self.pre_buffer)
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self.utterance_frames.append(frame_bytes)
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else:
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self.utterance_frames.append(frame_bytes)
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if is_speech:
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self.silence_duration = 0.0
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else:
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self.silence_duration += FRAME_DURATION_MS / 1000.0
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if self.silence_duration >= self.silence_threshold:
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if len(self.utterance_frames) < MIN_UTTERANCE_FRAMES:
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self.reset()
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return None
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result = b"".join(self.utterance_frames)
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self.reset()
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return result
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return None
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