Update custom model files, README, and requirements
Browse files- asr_pipeline.py +1 -222
asr_pipeline.py
CHANGED
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@@ -30,12 +30,6 @@ class ForcedAligner:
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_model = None
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_labels = None
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_dictionary = None
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_vad_model = None
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# VAD parameters
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VAD_HOP_SIZE = 256 # TEN-VAD frame size (16ms at 16kHz)
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VAD_THRESHOLD = 0.5 # Speech detection threshold
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VAD_MAX_GAP = 0.15 # Max gap to merge speech segments (seconds)
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@classmethod
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def get_instance(cls, device: str = "cuda"):
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@@ -57,135 +51,6 @@ class ForcedAligner:
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cls._dictionary = {c: i for i, c in enumerate(cls._labels)}
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return cls._model, cls._labels, cls._dictionary
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@classmethod
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def _get_vad_model(cls):
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"""Lazy-load TEN-VAD model (singleton)."""
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if cls._vad_model is None:
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from ten_vad import TenVad
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cls._vad_model = TenVad(hop_size=cls.VAD_HOP_SIZE, threshold=cls.VAD_THRESHOLD)
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return cls._vad_model
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@classmethod
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def _get_speech_regions(
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cls, audio: np.ndarray, sample_rate: int = 16000
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) -> list[tuple[float, float]]:
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"""Get speech regions using TEN-VAD.
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Args:
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audio: Audio waveform as numpy array
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sample_rate: Audio sample rate
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Returns:
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List of (start_time, end_time) tuples for speech regions
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"""
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vad_model = cls._get_vad_model()
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# Convert to int16 as required by TEN-VAD
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if audio.dtype != np.int16:
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audio_int16 = (np.clip(audio, -1.0, 1.0) * 32767).astype(np.int16)
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else:
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audio_int16 = audio
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# Process frame by frame
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hop_size = cls.VAD_HOP_SIZE
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frame_duration = hop_size / sample_rate
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speech_frames: list[bool] = []
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for i in range(0, len(audio_int16) - hop_size, hop_size):
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frame = audio_int16[i : i + hop_size]
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_, is_speech = vad_model.process(frame)
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speech_frames.append(is_speech)
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# Convert frame-level decisions to segments
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segments: list[tuple[float, float]] = []
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in_speech = False
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start_idx = 0
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for i, is_speech in enumerate(speech_frames):
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if is_speech and not in_speech:
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start_idx = i
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in_speech = True
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elif not is_speech and in_speech:
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start_time = start_idx * frame_duration
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end_time = i * frame_duration
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segments.append((start_time, end_time))
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in_speech = False
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# Handle trailing speech
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if in_speech:
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start_time = start_idx * frame_duration
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end_time = len(speech_frames) * frame_duration
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segments.append((start_time, end_time))
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# Merge segments with small gaps
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return cls._merge_speech_segments(segments)
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@classmethod
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def _merge_speech_segments(
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cls, segments: list[tuple[float, float]]
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) -> list[tuple[float, float]]:
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"""Merge speech segments with small gaps."""
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if not segments:
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return segments
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merged: list[tuple[float, float]] = [segments[0]]
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for start, end in segments[1:]:
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prev_start, prev_end = merged[-1]
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if start - prev_end <= cls.VAD_MAX_GAP:
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merged[-1] = (prev_start, end)
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else:
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merged.append((start, end))
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return merged
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@classmethod
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def _is_in_speech(cls, time: float, speech_regions: list[tuple[float, float]]) -> bool:
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"""Check if a timestamp falls within any speech region."""
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return any(start <= time <= end for start, end in speech_regions)
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@classmethod
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def _find_nearest_speech_boundary(
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cls, time: float, speech_regions: list[tuple[float, float]], direction: str = "any"
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) -> float:
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"""Find the nearest speech region boundary to a timestamp.
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Args:
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time: Timestamp to find boundary for
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speech_regions: List of (start, end) speech regions
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direction: "start" for word starts, "end" for word ends, "any" for closest
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Returns:
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Adjusted timestamp snapped to nearest speech boundary
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"""
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if not speech_regions:
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return time
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best_time = time
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min_dist = float("inf")
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for start, end in speech_regions:
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# If time is inside this region, return as-is
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if start <= time <= end:
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return time
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# Check distance to boundaries
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if direction in ("start", "any"):
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dist = abs(time - start)
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if dist < min_dist:
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min_dist = dist
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best_time = start
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if direction in ("end", "any"):
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dist = abs(time - end)
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if dist < min_dist:
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min_dist = dist
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best_time = end
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return best_time
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# Confidence threshold for alignment scores (log probability)
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MIN_CONFIDENCE = -5.0 # Tokens with scores below this are considered low-confidence
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@classmethod
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def align(
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cls,
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@@ -194,7 +59,6 @@ class ForcedAligner:
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sample_rate: int = 16000,
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_language: str = "eng",
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_batch_size: int = 16,
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use_vad: bool = True,
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) -> list[dict]:
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"""Align transcript to audio and return word-level timestamps.
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@@ -204,10 +68,9 @@ class ForcedAligner:
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sample_rate: Audio sample rate (default 16000)
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_language: ISO-639-3 language code (default "eng" for English, unused)
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_batch_size: Batch size for alignment model (unused)
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use_vad: If True, use VAD to refine word boundaries (default True)
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Returns:
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List of dicts with 'word', 'start', 'end'
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"""
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import torchaudio
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from torchaudio.functional import forced_align, merge_tokens
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device = _get_device()
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model, labels, dictionary = cls.get_instance(device)
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# Step 1: Get speech regions using VAD (before any processing)
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speech_regions = []
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if use_vad:
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speech_regions = cls._get_speech_regions(audio, sample_rate)
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# Convert audio to tensor (copy to ensure array is writable)
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if isinstance(audio, np.ndarray):
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waveform = torch.from_numpy(audio.copy()).float()
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@@ -272,122 +130,43 @@ class ForcedAligner:
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frame_duration = 320 / cls._bundle.sample_rate
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# Group token spans into words based on pipe separator
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# Track confidence scores per word
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words = text.split()
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word_timestamps = []
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current_word_start = None
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current_word_end = None
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current_word_scores: list[float] = []
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word_idx = 0
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for span in token_spans:
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token_char = labels[span.token]
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if token_char == "|": # Word separator
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if current_word_start is not None and word_idx < len(words):
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# Calculate word confidence as mean of token scores
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confidence = (
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sum(current_word_scores) / len(current_word_scores)
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if current_word_scores
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else 0.0
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)
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word_timestamps.append(
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{
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"word": words[word_idx],
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"start": current_word_start * frame_duration,
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"end": current_word_end * frame_duration,
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"confidence": confidence,
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}
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)
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word_idx += 1
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current_word_start = None
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current_word_end = None
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current_word_scores = []
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else:
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if current_word_start is None:
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current_word_start = span.start
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current_word_end = span.end
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current_word_scores.append(span.score)
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# Don't forget the last word
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if current_word_start is not None and word_idx < len(words):
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confidence = (
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sum(current_word_scores) / len(current_word_scores) if current_word_scores else 0.0
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)
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word_timestamps.append(
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{
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"word": words[word_idx],
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"start": current_word_start * frame_duration,
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"end": current_word_end * frame_duration,
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"confidence": confidence,
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}
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)
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# Step 2: Refine timestamps using VAD
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if use_vad and speech_regions:
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word_timestamps = cls._refine_with_vad(word_timestamps, speech_regions)
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return word_timestamps
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@classmethod
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def _refine_with_vad(
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cls, word_timestamps: list[dict], speech_regions: list[tuple[float, float]]
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) -> list[dict]:
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"""Refine word timestamps using VAD speech regions.
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- Words with low confidence that fall outside speech regions are flagged
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- Word boundaries are snapped to speech region boundaries when close
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Args:
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word_timestamps: List of word dicts with 'start', 'end', 'confidence'
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speech_regions: List of (start, end) speech regions
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Returns:
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Refined word timestamps
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"""
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if not word_timestamps or not speech_regions:
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return word_timestamps
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refined = []
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for word in word_timestamps:
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start = word["start"]
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end = word["end"]
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confidence = word.get("confidence", 0.0)
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# Check if word midpoint is in a speech region
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midpoint = (start + end) / 2
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in_speech = cls._is_in_speech(midpoint, speech_regions)
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# For low-confidence words outside speech, snap to nearest speech boundary
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if not in_speech and confidence < cls.MIN_CONFIDENCE:
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# Find the nearest speech region and snap boundaries
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start = cls._find_nearest_speech_boundary(start, speech_regions, "start")
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end = cls._find_nearest_speech_boundary(end, speech_regions, "end")
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# Ensure start < end
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if start >= end:
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end = start + 0.01
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# For words near speech boundaries, snap to the boundary
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# This helps align word edges with actual speech onset/offset
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snap_threshold = 0.05 # 50ms
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for region_start, region_end in speech_regions:
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# Snap start to speech region start if close
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if abs(start - region_start) < snap_threshold:
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start = region_start
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# Snap end to speech region end if close
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if abs(end - region_end) < snap_threshold:
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end = region_end
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refined.append(
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{
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"word": word["word"],
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"start": start,
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"end": end,
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"confidence": confidence,
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}
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)
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return refined
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try:
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from .diarization import SpeakerDiarizer
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_model = None
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_labels = None
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_dictionary = None
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@classmethod
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def get_instance(cls, device: str = "cuda"):
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cls._dictionary = {c: i for i, c in enumerate(cls._labels)}
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return cls._model, cls._labels, cls._dictionary
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@classmethod
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def align(
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cls,
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sample_rate: int = 16000,
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_language: str = "eng",
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_batch_size: int = 16,
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) -> list[dict]:
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"""Align transcript to audio and return word-level timestamps.
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sample_rate: Audio sample rate (default 16000)
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_language: ISO-639-3 language code (default "eng" for English, unused)
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_batch_size: Batch size for alignment model (unused)
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Returns:
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+
List of dicts with 'word', 'start', 'end' keys
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"""
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import torchaudio
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from torchaudio.functional import forced_align, merge_tokens
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device = _get_device()
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model, labels, dictionary = cls.get_instance(device)
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# Convert audio to tensor (copy to ensure array is writable)
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if isinstance(audio, np.ndarray):
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waveform = torch.from_numpy(audio.copy()).float()
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frame_duration = 320 / cls._bundle.sample_rate
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# Group token spans into words based on pipe separator
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words = text.split()
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word_timestamps = []
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current_word_start = None
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current_word_end = None
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word_idx = 0
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for span in token_spans:
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token_char = labels[span.token]
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if token_char == "|": # Word separator
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if current_word_start is not None and word_idx < len(words):
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word_timestamps.append(
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{
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"word": words[word_idx],
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"start": current_word_start * frame_duration,
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"end": current_word_end * frame_duration,
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}
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)
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word_idx += 1
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current_word_start = None
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current_word_end = None
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else:
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if current_word_start is None:
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current_word_start = span.start
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current_word_end = span.end
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# Don't forget the last word
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if current_word_start is not None and word_idx < len(words):
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word_timestamps.append(
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{
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"word": words[word_idx],
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"start": current_word_start * frame_duration,
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"end": current_word_end * frame_duration,
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}
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)
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return word_timestamps
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try:
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from .diarization import SpeakerDiarizer
|