Automatic Speech Recognition
Transformers
Safetensors
English
asr_model
asr
speech-recognition
speech-to-text
audio
speech-llm
word-timestamps
speaker-diarization
qwen
granite-speech
lora
custom_code
Eval Results (legacy)
Instructions to use mazesmazes/tiny-audio with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use mazesmazes/tiny-audio with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="mazesmazes/tiny-audio", trust_remote_code=True)# pip install -U transformers accelerate # Load model directly from transformers import AutoModelForSpeechSeq2Seq model = AutoModelForSpeechSeq2Seq.from_pretrained("mazesmazes/tiny-audio", trust_remote_code=True, device_map="auto") - Notebooks
- Google Colab
- Kaggle
Update custom model files, README, and requirements
Browse files- README.md +4 -3
- asr_pipeline.py +100 -19
- diarization.py +142 -38
README.md
CHANGED
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@@ -127,7 +127,8 @@ for word in result["words"]:
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Pass `return_speakers=True` to label every word with who said it. It also turns on word timestamps.
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It works on recordings of any length, keeps speaker labels consistent across hour-long meetings, and
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handles
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Diarization needs transformers installed from `main` until the next release:
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```
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Speakers are numbered by when they first speak. The number of speakers is detected automatically. If
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you know it, pass `num_speakers` (exact) or `max_speakers` (an upper bound);
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```python
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result = pipe("call.wav", return_speakers=True, num_speakers=2)
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Pass `return_speakers=True` to label every word with who said it. It also turns on word timestamps.
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It works on recordings of any length, keeps speaker labels consistent across hour-long meetings, and
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handles up to 8 speakers. Each speaker is transcribed separately on audio where everyone else is
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silenced, so overlapping speech is only partly handled.
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Diarization needs transformers installed from `main` until the next release:
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```
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Speakers are numbered by when they first speak. The number of speakers is detected automatically. If
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you know it, pass `num_speakers` (exact) or `max_speakers` (an upper bound); voices beyond the cap
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are not transcribed:
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```python
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result = pipe("call.wav", return_speakers=True, num_speakers=2)
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asr_pipeline.py
CHANGED
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from .alignment import QwenForcedAligner
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from .asr_modeling import ASRModel
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from .asr_processing import prepend_lead_in
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from .diarization import NemotronDiarizer
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else:
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try:
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from .alignment import QwenForcedAligner
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from .asr_modeling import ASRModel
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from .asr_processing import prepend_lead_in
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from .diarization import NemotronDiarizer
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except ImportError: # flat layout on the Hub: sibling modules, no package
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from alignment import QwenForcedAligner
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from asr_modeling import ASRModel
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from asr_processing import prepend_lead_in
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from diarization import NemotronDiarizer
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# Re-export for backwards compatibility
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__all__ = [
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return bounds
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# Below this RMS (-100 dBFS) a chunk is digital silence: exact zeros, as in
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# edited or remixed recordings. Given one, the model answers with a memorized
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# training sentence ("The film was directed by the same director who directed
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Args:
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inputs: Audio input (file path, dict with array/sampling_rate, etc.)
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return_timestamps: If True, return word-level timestamps (Qwen3-ForcedAligner)
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return_speakers: If True,
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user_prompt: Custom transcription prompt (default: "Transcribe: ")
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num_speakers: Exact number of speakers, if known
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max_speakers: Upper bound on the number of speakers
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diarization_params: _DiarizationParams,
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**kwargs: Any,
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) -> dict[str, Any]:
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"""Transcribe in chunks
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Chunks of 8-18 s cut at quiet points are transcribed one by one and each
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is aligned against its own transcript (batched), then offset onto the
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recording's timeline -- so audio of any length gets every word timed,
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and the model never sees a clip longer than it trained on. Speakers
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from
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recording-level: chunking never touches speaker identity.
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"""
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audio = self._extract_audio(inputs)
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if audio is None:
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raise ValueError(msg)
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array = np.asarray(audio["array"], dtype=np.float32)
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sr = audio.get("sampling_rate", 16000)
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bounds = chunk_bounds(array, sr)
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texts: list[str] = [
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self._transcribe_chunk(array[s:e], sr, **kwargs)["text"] for s, e in bounds
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except Exception as e:
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result["words"] = []
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result["timestamp_error"] = str(e)
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)
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return result
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def _extract_audio(self, inputs: object) -> _RawAudio | None:
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"""Extract audio array from various input formats using HF utilities."""
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if isinstance(inputs, dict):
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from .alignment import QwenForcedAligner
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from .asr_modeling import ASRModel
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from .asr_processing import prepend_lead_in
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from .diarization import NemotronDiarizer, StreamChunk, masked_audio, pack_spans
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else:
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try:
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from .alignment import QwenForcedAligner
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from .asr_modeling import ASRModel
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from .asr_processing import prepend_lead_in
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from .diarization import NemotronDiarizer, StreamChunk, masked_audio, pack_spans
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except ImportError: # flat layout on the Hub: sibling modules, no package
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from alignment import QwenForcedAligner
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from asr_modeling import ASRModel
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from asr_processing import prepend_lead_in
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from diarization import NemotronDiarizer, StreamChunk, masked_audio, pack_spans
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# Re-export for backwards compatibility
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__all__ = [
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return bounds
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def stream_chunks(
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audio: npt.NDArray[np.float32],
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spans: list[tuple[int, int]],
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sample_rate: int,
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max_s: float = CHUNK_MAX_S,
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) -> list[tuple[int, int]]:
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"""Ranges of at most `max_s` over one speaker's `spans`; long turns cut by `chunk_bounds`."""
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pieces: list[tuple[int, int]] = []
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for s, e in spans:
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pieces.extend((s + a, s + b) for a, b in chunk_bounds(audio[s:e], sample_rate, max_s))
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return pack_spans(pieces, int(max_s * sample_rate))
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# Below this RMS (-100 dBFS) a chunk is digital silence: exact zeros, as in
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# edited or remixed recordings. Given one, the model answers with a memorized
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# training sentence ("The film was directed by the same director who directed
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Args:
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inputs: Audio input (file path, dict with array/sampling_rate, etc.)
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return_timestamps: If True, return word-level timestamps (Qwen3-ForcedAligner)
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return_speakers: If True, transcribe each Nemotron-3-Diarization speaker on
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audio with the others silenced and label words by speaker (see
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`_transcribe_streams`). Implies return_timestamps.
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user_prompt: Custom transcription prompt (default: "Transcribe: ")
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num_speakers: Exact number of speakers, if known
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max_speakers: Upper bound on the number of speakers
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diarization_params: _DiarizationParams,
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**kwargs: Any,
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) -> dict[str, Any]:
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"""Transcribe in chunks and time every word; per-speaker streams if requested.
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Chunks of 8-18 s cut at quiet points are transcribed one by one and each
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is aligned against its own transcript (batched), then offset onto the
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recording's timeline -- so audio of any length gets every word timed,
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and the model never sees a clip longer than it trained on. Speakers
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come from `_transcribe_streams`.
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"""
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audio = self._extract_audio(inputs)
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if audio is None:
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raise ValueError(msg)
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array = np.asarray(audio["array"], dtype=np.float32)
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sr = audio.get("sampling_rate", 16000)
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if return_speakers:
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return self._transcribe_streams(array, sr, diarization_params, **kwargs)
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return self._transcribe_aligned(array, sr, **kwargs)
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def _transcribe_aligned(
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self, array: npt.NDArray[np.float32], sr: int, **kwargs: Any
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) -> dict[str, Any]:
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"""`{"text", "words"}` for the whole recording: chunk, transcribe, align, offset."""
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bounds = chunk_bounds(array, sr)
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texts: list[str] = [
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self._transcribe_chunk(array[s:e], sr, **kwargs)["text"] for s, e in bounds
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except Exception as e:
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result["words"] = []
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result["timestamp_error"] = str(e)
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return result
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def _transcribe_streams(
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self,
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array: npt.NDArray[np.float32],
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sr: int,
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diarization_params: _DiarizationParams,
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**kwargs: Any,
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) -> dict[str, Any]:
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"""Per-speaker masked ASR, a zero-shot port of NeMo's `masked_asr`.
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Nemotron runs first, one pass over the whole recording, so speaker ids
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are recording-level. Each speaker is transcribed only where they talk
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(`stream_chunks`), with everyone else zeroed (`masked_audio`); words take
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their stream's speaker, and a word two streams both heard at once is
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kept once (`dedupe_words`). Crosstalk under a speaker's mask is heard.
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With at most one speaker active nothing is masked (today's exact text);
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if diarization fails, the unmasked transcript gets `diarization_error`.
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"""
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try:
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activity = NemotronDiarizer.activity(array, sample_rate=sr)
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keep = NemotronDiarizer.top_speakers(activity, **diarization_params)
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except Exception as e:
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result = self._transcribe_aligned(array, sr, **kwargs)
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result["speaker_segments"] = []
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result["diarization_error"] = str(e)
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return result
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masks = {c: activity[:, c] > NemotronDiarizer.SEGMENT_THRESHOLD for c in keep}
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talking = [c for c in keep if masks[c].any()]
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if len(talking) <= 1:
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result = self._transcribe_aligned(array, sr, **kwargs)
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name = f"SPEAKER_{keep.index(talking[0]) if talking else 0}"
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result["words"] = [{**w, "speaker": name} for w in result["words"]]
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else:
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chunks: list[StreamChunk] = []
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for c in talking:
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spans = NemotronDiarizer.sample_spans(masks[c], sr, len(array))
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chunks.extend(
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{"col": c, "start": s, "audio": masked_audio(array, spans, s, e)}
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for s, e in stream_chunks(array, spans, sr)
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)
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texts = [self._transcribe_chunk(ch["audio"], sr, **kwargs)["text"] for ch in chunks]
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result = self._align_streams(chunks, texts, sr, masks, keep)
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if result["words"]:
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result["words"] = NemotronDiarizer.dedupe_words(result["words"], activity, keep)
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result["text"] = " ".join(w["word"] for w in result["words"])
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result["speaker_segments"] = NemotronDiarizer.segments(activity, keep)
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return result
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@staticmethod
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def _align_streams(
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chunks: list[StreamChunk],
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texts: list[str],
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sr: int,
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masks: dict[int, npt.NDArray[np.bool_]],
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keep: list[int],
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) -> dict[str, Any]:
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"""Align every stream chunk in one batch; `{"text", "words"}` merged by time.
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If alignment fails the text is the chunk transcripts by start time and
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`timestamp_error` is set.
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"""
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try:
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aligned = QwenForcedAligner.align_chunks(
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[(ch["audio"], text) for ch, text in zip(chunks, texts, strict=True)],
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sample_rate=sr,
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)
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except Exception as e:
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order = sorted(range(len(chunks)), key=lambda k: chunks[k]["start"])
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text = " ".join(texts[k] for k in order if texts[k])
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return {"text": text, "words": [], "timestamp_error": str(e)}
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words = NemotronDiarizer.stream_words(chunks, aligned, sr, masks, keep)
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return {"text": " ".join(w["word"] for w in words), "words": words}
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+
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def _extract_audio(self, inputs: object) -> _RawAudio | None:
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"""Extract audio array from various input formats using HF utilities."""
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if isinstance(inputs, dict):
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diarization.py
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"""Speaker diarization with NVIDIA Nemotron-3-Diarization."""
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import importlib.util
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from typing import TYPE_CHECKING, Any, TypedDict
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import numpy as np
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Nemotron3DiarizationProcessor,
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)
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class SpeakerSegment(TypedDict):
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"""One speaker turn, in seconds."""
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return [(labels[s].item(), int(s), int(e)) for s, e in zip(starts, ends, strict=True)]
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class NemotronDiarizer:
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"""Speaker activity from NVIDIA Nemotron-3-Diarization, per 10 ms frame.
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>>> activity = NemotronDiarizer.activity(audio) # (frames, 8) probabilities
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>>> keep = NemotronDiarizer.top_speakers(activity, max_speakers=2)
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>>> segments = NemotronDiarizer.segments(activity, keep)
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-
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"""
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MODEL_ID = "nvidia/Nemotron-3-Diarization"
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FRAME_S = 0.01
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SEGMENT_THRESHOLD = 0.5 # a speaker's segment = frames above this
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-
MIN_WORD_ACTIVITY = 0.1 # below this nobody is heard: the word inherits its neighbour
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_model: "Nemotron3DiarizationForAudioFrameClassification | None" = None
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_processor: "Nemotron3DiarizationProcessor | None" = None
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@@ -130,8 +185,7 @@ class NemotronDiarizer:
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Every column that is ever above SEGMENT_THRESHOLD, capped by
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`num_speakers` (exact, when known) or `max_speakers` to the columns
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-
with the most speech.
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-
a dropped column's words go to the most active remaining speaker.
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"""
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active = [
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c for c in range(activity.shape[1]) if (activity[:, c] > cls.SEGMENT_THRESHOLD).any()
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@@ -156,45 +210,95 @@ class NemotronDiarizer:
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return sorted(out, key=lambda seg: (seg["start"], seg["end"]))
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@classmethod
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-
def
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cls,
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-
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-
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keep: list[int],
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-
) -> list[
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-
"""
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-
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-
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a pause), takes the previous word's speaker -- the next word's at the
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start of the recording.
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"""
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-
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for
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-
return out
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| 191 |
@classmethod
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-
def
|
| 193 |
cls, words: list[dict[str, Any]], activity: npt.NDArray[np.float32], keep: list[int]
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| 194 |
) -> list[dict[str, Any]]:
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-
"""
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-
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-
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-
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-
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| 1 |
"""Speaker diarization with NVIDIA Nemotron-3-Diarization."""
|
| 2 |
|
| 3 |
import importlib.util
|
| 4 |
+
import re
|
| 5 |
from typing import TYPE_CHECKING, Any, TypedDict
|
| 6 |
|
| 7 |
import numpy as np
|
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|
|
| 18 |
Nemotron3DiarizationProcessor,
|
| 19 |
)
|
| 20 |
|
| 21 |
+
from .alignment import AlignedWord
|
| 22 |
+
|
| 23 |
|
| 24 |
class SpeakerSegment(TypedDict):
|
| 25 |
"""One speaker turn, in seconds."""
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|
| 55 |
return [(labels[s].item(), int(s), int(e)) for s, e in zip(starts, ends, strict=True)]
|
| 56 |
|
| 57 |
|
| 58 |
+
def masked_audio(
|
| 59 |
+
audio: npt.NDArray[np.float32], spans: list[tuple[int, int]], start: int, end: int
|
| 60 |
+
) -> npt.NDArray[np.float32]:
|
| 61 |
+
"""`audio[start:end]` with every sample outside `spans` set to 0.0.
|
| 62 |
+
|
| 63 |
+
`spans` are absolute `(start, end)` sample ranges, end exclusive. Only the
|
| 64 |
+
slice is copied, so a speaker stream of a long recording never holds a
|
| 65 |
+
full-length masked copy. Zeros are digital silence: the encoder sees its
|
| 66 |
+
silence floor there, and `is_silent` skips a chunk that is nothing else.
|
| 67 |
+
"""
|
| 68 |
+
out = np.zeros(end - start, dtype=np.float32)
|
| 69 |
+
for s, e in spans:
|
| 70 |
+
lo, hi = max(s, start), min(e, end)
|
| 71 |
+
if lo < hi:
|
| 72 |
+
out[lo - start : hi - start] = audio[lo:hi]
|
| 73 |
+
return out
|
| 74 |
+
|
| 75 |
+
|
| 76 |
+
def pack_spans(spans: list[tuple[int, int]], limit: int) -> list[tuple[int, int]]:
|
| 77 |
+
"""Merge consecutive sorted `(start, end)` spans into ranges at most `limit` long.
|
| 78 |
+
|
| 79 |
+
Each range starts and ends on a span, so a speaker's chunk never begins or
|
| 80 |
+
ends in audio where they are silent. A span already over `limit` stays alone.
|
| 81 |
+
"""
|
| 82 |
+
ranges: list[tuple[int, int]] = []
|
| 83 |
+
for s, e in spans:
|
| 84 |
+
if ranges and e - ranges[-1][0] <= limit:
|
| 85 |
+
ranges[-1] = (ranges[-1][0], e)
|
| 86 |
+
else:
|
| 87 |
+
ranges.append((s, e))
|
| 88 |
+
return ranges
|
| 89 |
+
|
| 90 |
+
|
| 91 |
+
class StreamChunk(TypedDict):
|
| 92 |
+
"""One stretch of one speaker's masked audio: kept column, absolute start, samples."""
|
| 93 |
+
|
| 94 |
+
col: int
|
| 95 |
+
start: int
|
| 96 |
+
audio: npt.NDArray[np.float32]
|
| 97 |
+
|
| 98 |
+
|
| 99 |
+
_NON_WORD_RE = re.compile(r"[^\w']+")
|
| 100 |
+
|
| 101 |
+
|
| 102 |
+
def _norm_word(word: str) -> str:
|
| 103 |
+
"""Lowercase `word` without punctuation, for matching one word heard in two streams."""
|
| 104 |
+
return _NON_WORD_RE.sub("", word.lower())
|
| 105 |
+
|
| 106 |
+
|
| 107 |
class NemotronDiarizer:
|
| 108 |
"""Speaker activity from NVIDIA Nemotron-3-Diarization, per 10 ms frame.
|
| 109 |
|
|
|
|
| 120 |
>>> activity = NemotronDiarizer.activity(audio) # (frames, 8) probabilities
|
| 121 |
>>> keep = NemotronDiarizer.top_speakers(activity, max_speakers=2)
|
| 122 |
>>> segments = NemotronDiarizer.segments(activity, keep)
|
| 123 |
+
|
| 124 |
+
`ASRPipeline(..., return_speakers=True)` turns this into speaker streams:
|
| 125 |
+
`sample_spans` gives each kept speaker a mask, the ASR runs once per
|
| 126 |
+
speaker on audio with everyone else silenced, and `span_is_active` /
|
| 127 |
+
`dedupe_words` clean up the merge.
|
| 128 |
"""
|
| 129 |
|
| 130 |
MODEL_ID = "nvidia/Nemotron-3-Diarization"
|
| 131 |
FRAME_S = 0.01
|
| 132 |
SEGMENT_THRESHOLD = 0.5 # a speaker's segment = frames above this
|
|
|
|
| 133 |
_model: "Nemotron3DiarizationForAudioFrameClassification | None" = None
|
| 134 |
_processor: "Nemotron3DiarizationProcessor | None" = None
|
| 135 |
|
|
|
|
| 185 |
|
| 186 |
Every column that is ever above SEGMENT_THRESHOLD, capped by
|
| 187 |
`num_speakers` (exact, when known) or `max_speakers` to the columns
|
| 188 |
+
with the most speech. A dropped column gets no stream of its own.
|
|
|
|
| 189 |
"""
|
| 190 |
active = [
|
| 191 |
c for c in range(activity.shape[1]) if (activity[:, c] > cls.SEGMENT_THRESHOLD).any()
|
|
|
|
| 210 |
return sorted(out, key=lambda seg: (seg["start"], seg["end"]))
|
| 211 |
|
| 212 |
@classmethod
|
| 213 |
+
def _frame_range(cls, start: float, end: float, n_frames: int) -> tuple[int, int]:
|
| 214 |
+
"""Frames `[lo, hi)` under the span `start..end` s: at least one, clamped to the array."""
|
| 215 |
+
lo = min(int(start / cls.FRAME_S), n_frames - 1)
|
| 216 |
+
hi = max(lo + 1, min(int(np.ceil(end / cls.FRAME_S)), n_frames))
|
| 217 |
+
return lo, hi
|
| 218 |
+
|
| 219 |
+
@classmethod
|
| 220 |
+
def sample_spans(
|
| 221 |
+
cls, frames: npt.NDArray[np.bool_], sample_rate: int, n_samples: int
|
| 222 |
+
) -> list[tuple[int, int]]:
|
| 223 |
+
"""The active runs of a frame mask as `(start, end)` sample ranges, end exclusive.
|
| 224 |
+
|
| 225 |
+
One frame covers `FRAME_S * sample_rate` samples. A run that reaches the
|
| 226 |
+
last frame runs to `n_samples`: the model's frame count can fall a few
|
| 227 |
+
samples short of the audio, and that tail belongs to the run.
|
| 228 |
+
"""
|
| 229 |
+
hop = cls.FRAME_S * sample_rate
|
| 230 |
+
spans: list[tuple[int, int]] = []
|
| 231 |
+
for on, s, e in _label_runs(frames):
|
| 232 |
+
start = min(round(s * hop), n_samples)
|
| 233 |
+
end = n_samples if e == len(frames) else min(round(e * hop), n_samples)
|
| 234 |
+
if on and start < end:
|
| 235 |
+
spans.append((start, end))
|
| 236 |
+
return spans
|
| 237 |
+
|
| 238 |
+
@classmethod
|
| 239 |
+
def span_is_active(cls, frames: npt.NDArray[np.bool_], start: float, end: float) -> bool:
|
| 240 |
+
"""True if any frame of the span `start..end` s is in the mask `frames`."""
|
| 241 |
+
if len(frames) == 0:
|
| 242 |
+
return False
|
| 243 |
+
lo, hi = cls._frame_range(start, end, len(frames))
|
| 244 |
+
return bool(frames[lo:hi].any())
|
| 245 |
+
|
| 246 |
+
@classmethod
|
| 247 |
+
def stream_words(
|
| 248 |
cls,
|
| 249 |
+
chunks: list[StreamChunk],
|
| 250 |
+
aligned: "list[list[AlignedWord]]",
|
| 251 |
+
sample_rate: int,
|
| 252 |
+
masks: dict[int, npt.NDArray[np.bool_]],
|
| 253 |
keep: list[int],
|
| 254 |
+
) -> list[dict[str, Any]]:
|
| 255 |
+
"""Every stream's aligned words on the recording timeline, labelled, by start.
|
| 256 |
|
| 257 |
+
A word placed wholly in its own speaker's silenced audio (`masks`) is a
|
| 258 |
+
hallucination on silence and is dropped.
|
|
|
|
|
|
|
| 259 |
"""
|
| 260 |
+
names = {c: f"SPEAKER_{i}" for i, c in enumerate(keep)}
|
| 261 |
+
words: list[dict[str, Any]] = []
|
| 262 |
+
for ch, chunk_words in zip(chunks, aligned, strict=True):
|
| 263 |
+
offset = ch["start"] / sample_rate
|
| 264 |
+
for w in chunk_words:
|
| 265 |
+
start, end = w["start"] + offset, w["end"] + offset
|
| 266 |
+
if cls.span_is_active(masks[ch["col"]], start, end):
|
| 267 |
+
words.append({**w, "start": start, "end": end, "speaker": names[ch["col"]]})
|
| 268 |
+
return sorted(words, key=lambda w: (w["start"], w["end"]))
|
| 269 |
+
|
| 270 |
+
@classmethod
|
| 271 |
+
def _span_activity(
|
| 272 |
+
cls, activity: npt.NDArray[np.float32], col: int, start: float, end: float
|
| 273 |
+
) -> float:
|
| 274 |
+
"""Mean activity of column `col` over the span `start..end` s."""
|
| 275 |
+
lo, hi = cls._frame_range(start, end, len(activity))
|
| 276 |
+
return float(activity[lo:hi, col].mean())
|
|
|
|
| 277 |
|
| 278 |
@classmethod
|
| 279 |
+
def dedupe_words(
|
| 280 |
cls, words: list[dict[str, Any]], activity: npt.NDArray[np.float32], keep: list[int]
|
| 281 |
) -> list[dict[str, Any]]:
|
| 282 |
+
"""Drop one copy of a word two speaker streams both heard.
|
| 283 |
+
|
| 284 |
+
`words` are sorted by start and labelled `SPEAKER_i` (position in
|
| 285 |
+
`keep`). Two words from different speakers with the same text (case
|
| 286 |
+
and punctuation ignored) and overlapping spans are one word that
|
| 287 |
+
leaked into both masks: the copy whose speaker is more active over its
|
| 288 |
+
span stays, the earlier word on a tie.
|
| 289 |
+
"""
|
| 290 |
+
cols = {f"SPEAKER_{i}": c for i, c in enumerate(keep)}
|
| 291 |
+
dropped: set[int] = set()
|
| 292 |
+
for i, a in enumerate(words):
|
| 293 |
+
for j in range(i + 1, len(words)):
|
| 294 |
+
b = words[j]
|
| 295 |
+
if b["start"] >= a["end"]:
|
| 296 |
+
break
|
| 297 |
+
if i in dropped or j in dropped or a["speaker"] == b["speaker"]:
|
| 298 |
+
continue
|
| 299 |
+
if not _norm_word(a["word"]) or _norm_word(a["word"]) != _norm_word(b["word"]):
|
| 300 |
+
continue
|
| 301 |
+
score_a = cls._span_activity(activity, cols[a["speaker"]], a["start"], a["end"])
|
| 302 |
+
score_b = cls._span_activity(activity, cols[b["speaker"]], b["start"], b["end"])
|
| 303 |
+
dropped.add(j if score_a >= score_b else i)
|
| 304 |
+
return [w for k, w in enumerate(words) if k not in dropped]
|