Update custom model files, README, and requirements
Browse files- asr_pipeline.py +133 -22
asr_pipeline.py
CHANGED
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@@ -24,7 +24,10 @@ def _get_device() -> str:
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class ForcedAligner:
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"""Lazy-loaded forced aligner for word-level timestamps using torchaudio wav2vec2.
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_bundle = None
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_model = None
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@@ -51,6 +54,107 @@ 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 align(
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cls,
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@@ -59,21 +163,26 @@ 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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) -> list[dict]:
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"""Align transcript to audio and return word-level timestamps.
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Args:
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audio: Audio waveform as numpy array
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text: Transcript text to align
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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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# Normalize text: uppercase, keep only valid characters
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transcript = text.upper()
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tokens = []
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for char in transcript:
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if char in dictionary:
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@@ -116,35 +226,34 @@ class ForcedAligner:
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if not tokens:
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return []
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-
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-
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# Note: forced_align is deprecated in torchaudio 2.6+ and will be removed in 2.9 (late 2025)
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# No official replacement announced yet. See https://github.com/pytorch/audio/issues/3902
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aligned_tokens, scores = forced_align(emission.unsqueeze(0), targets, blank=0)
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# Use torchaudio's merge_tokens to get token spans (removes blanks and merges repeats)
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token_spans = merge_tokens(aligned_tokens[0], scores[0])
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# Convert frame indices to time (model stride is 320 samples at 16kHz = 20ms)
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frame_duration = 320 / cls._bundle.sample_rate
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#
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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
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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":
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"end":
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}
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)
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word_idx += 1
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@@ -152,16 +261,18 @@ class ForcedAligner:
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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 =
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current_word_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":
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"end":
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}
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)
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class ForcedAligner:
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"""Lazy-loaded forced aligner for word-level timestamps using torchaudio wav2vec2.
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Uses Viterbi trellis algorithm for optimal alignment path finding.
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"""
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_bundle = None
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_model = None
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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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@staticmethod
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def _get_trellis(emission: torch.Tensor, tokens: list[int], blank_id: int = 0) -> torch.Tensor:
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"""Build Viterbi trellis for forced alignment.
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The trellis is a 2D matrix where trellis[t, j] represents the log probability
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of the most likely path that has emitted j tokens at time t.
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Args:
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emission: Log-softmax emission matrix of shape (num_frames, num_classes)
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tokens: List of target token indices
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blank_id: Index of the blank/CTC token (default 0)
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Returns:
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Trellis matrix of shape (num_frames + 1, num_tokens + 1)
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"""
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num_frames = emission.size(0)
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num_tokens = len(tokens)
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# Initialize trellis with -inf (impossible paths)
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trellis = torch.full((num_frames + 1, num_tokens + 1), -float("inf"))
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trellis[0, 0] = 0 # Start state has probability 1
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for t in range(num_frames):
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for j in range(num_tokens + 1):
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# Stay in current state (emit blank)
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if j < num_tokens + 1:
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stay_prob = trellis[t, j] + emission[t, blank_id]
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else:
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stay_prob = -float("inf")
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# Move to next state (emit token)
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if j > 0:
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move_prob = trellis[t, j - 1] + emission[t, tokens[j - 1]]
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else:
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move_prob = -float("inf")
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trellis[t + 1, j] = max(stay_prob, move_prob)
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return trellis
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@staticmethod
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def _backtrack(
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trellis: torch.Tensor, emission: torch.Tensor, tokens: list[int], blank_id: int = 0
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) -> list[tuple[int, int, int]]:
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"""Backtrack through trellis to find optimal alignment path.
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Args:
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trellis: Trellis matrix from _get_trellis
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emission: Log-softmax emission matrix
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tokens: List of target token indices
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blank_id: Index of the blank/CTC token
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Returns:
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List of (token_idx, start_frame, end_frame) tuples
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"""
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num_frames = emission.size(0)
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num_tokens = len(tokens)
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# Start from the end
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t = num_frames
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j = num_tokens
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path = []
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# Backtrack to find where each token was emitted
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while j > 0:
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# Find the frame where token j-1 was first emitted
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token_end = t
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# Walk back while staying in state j (emitting blanks)
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while t > 0:
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stay_prob = trellis[t - 1, j] + emission[t - 1, blank_id]
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if j > 0:
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move_prob = trellis[t - 1, j - 1] + emission[t - 1, tokens[j - 1]]
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else:
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move_prob = -float("inf")
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# Check if we moved into this state or stayed
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if move_prob > stay_prob:
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# We moved into state j at time t-1
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token_start = t - 1
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path.append((tokens[j - 1], token_start, token_end))
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j -= 1
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t -= 1
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break
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else:
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# We stayed in state j
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t -= 1
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if t == 0 and j > 0:
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# Handle edge case: remaining tokens at the start
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path.append((tokens[j - 1], 0, token_end))
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j -= 1
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# Reverse to get chronological order
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path.reverse()
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return path
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# Sub-frame offset to compensate for Wav2Vec2 convolutional look-ahead (in seconds)
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# This makes timestamps feel more "natural" by shifting them earlier
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OFFSET_COMPENSATION = 0.04 # 40ms
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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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offset_compensation: float | None = None,
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) -> list[dict]:
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"""Align transcript to audio and return word-level timestamps.
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Uses Viterbi trellis algorithm for optimal forced alignment.
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Args:
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audio: Audio waveform as numpy array
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text: Transcript text to align
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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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offset_compensation: Time offset in seconds to subtract from timestamps
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to compensate for Wav2Vec2 look-ahead (default: 0.04s / 40ms).
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Set to 0 to disable.
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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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device = _get_device()
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model, labels, dictionary = cls.get_instance(device)
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# Normalize text: uppercase, keep only valid characters
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transcript = text.upper()
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# Build tokens from transcript (including word separators)
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tokens = []
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for char in transcript:
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if char in dictionary:
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if not tokens:
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return []
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# Build Viterbi trellis and backtrack for optimal path
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trellis = cls._get_trellis(emission, tokens, blank_id=0)
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alignment_path = cls._backtrack(trellis, emission, tokens, blank_id=0)
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# Convert frame indices to time (model stride is 320 samples at 16kHz = 20ms)
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frame_duration = 320 / cls._bundle.sample_rate
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# Apply offset compensation for Wav2Vec2 look-ahead
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offset = offset_compensation if offset_compensation is not None else cls.OFFSET_COMPENSATION
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# Group aligned tokens 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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separator_id = dictionary.get("|", dictionary.get(" ", 0))
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for token_id, start_frame, end_frame in alignment_path:
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if token_id == separator_id: # Word separator
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if current_word_start is not None and word_idx < len(words):
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start_time = max(0.0, current_word_start * frame_duration - offset)
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end_time = max(0.0, current_word_end * frame_duration - offset)
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word_timestamps.append(
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{
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"word": words[word_idx],
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"start": start_time,
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"end": end_time,
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}
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)
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word_idx += 1
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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 = start_frame
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current_word_end = end_frame
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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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start_time = max(0.0, current_word_start * frame_duration - offset)
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end_time = max(0.0, current_word_end * frame_duration - offset)
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word_timestamps.append(
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{
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"word": words[word_idx],
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"start": start_time,
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"end": end_time,
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}
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)
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