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app.py
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@@ -15,18 +15,24 @@ from transformers import Wav2Vec2FeatureExtractor, WavLMForXVector
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os.environ.setdefault("HF_XET_HIGH_PERFORMANCE", "1")
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DATASETS_SERVER = "https://datasets-server.huggingface.co"
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MODEL_ID = "microsoft/wavlm-base-plus-sv"
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TARGET_SR = 16000
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_feature_extractor = None
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_model = None
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-
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def _load_model():
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global _feature_extractor, _model
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with
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if _model is None:
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_feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained(MODEL_ID)
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_model = WavLMForXVector.from_pretrained(MODEL_ID)
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@@ -35,6 +41,7 @@ def _load_model():
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def _embed(audio_array: np.ndarray, sr: int, max_sec: int) -> np.ndarray:
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fe, mdl = _load_model()
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waveform = torch.tensor(audio_array, dtype=torch.float32)
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if waveform.ndim == 2:
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@@ -43,9 +50,8 @@ def _embed(audio_array: np.ndarray, sr: int, max_sec: int) -> np.ndarray:
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waveform = torchaudio.functional.resample(waveform, sr, TARGET_SR)
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waveform = waveform[: max_sec * TARGET_SR]
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inputs = fe(waveform.numpy(), sampling_rate=TARGET_SR, return_tensors="pt")
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with
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out = mdl(**inputs)
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return out.embeddings.squeeze().numpy()
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@@ -186,8 +192,8 @@ def identify_speakers(
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errors: list[str] = []
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done = 0
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# Process
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with concurrent.futures.ThreadPoolExecutor(max_workers=
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future_to_repo = {
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ex.submit(_process_repo, repo, int(samples_per_book), int(audio_sec), token): repo
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for repo in repos
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os.environ.setdefault("HF_XET_HIGH_PERFORMANCE", "1")
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# 1 PyTorch thread per worker — lets N_CPUS threads run inference in parallel
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# instead of one wide inference that blocks everyone else.
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torch.set_num_threads(1)
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N_CPUS = os.cpu_count() or 2
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DATASETS_SERVER = "https://datasets-server.huggingface.co"
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MODEL_ID = "microsoft/wavlm-base-plus-sv"
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TARGET_SR = 16000
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_feature_extractor = None
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_model = None
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_init_lock = threading.Lock() # only for one-time model init
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def _load_model():
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global _feature_extractor, _model
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with _init_lock:
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if _model is None:
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_feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained(MODEL_ID)
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_model = WavLMForXVector.from_pretrained(MODEL_ID)
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def _embed(audio_array: np.ndarray, sr: int, max_sec: int) -> np.ndarray:
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# Thread-safe: eval() + no_grad() — weights are read-only, GIL released in C++ ops
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fe, mdl = _load_model()
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waveform = torch.tensor(audio_array, dtype=torch.float32)
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if waveform.ndim == 2:
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waveform = torchaudio.functional.resample(waveform, sr, TARGET_SR)
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waveform = waveform[: max_sec * TARGET_SR]
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inputs = fe(waveform.numpy(), sampling_rate=TARGET_SR, return_tensors="pt")
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with torch.no_grad():
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out = mdl(**inputs)
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return out.embeddings.squeeze().numpy()
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errors: list[str] = []
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done = 0
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# Process repos in parallel — capped at N_CPUS since embed is the bottleneck
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with concurrent.futures.ThreadPoolExecutor(max_workers=N_CPUS) as ex:
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future_to_repo = {
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ex.submit(_process_repo, repo, int(samples_per_book), int(audio_sec), token): repo
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for repo in repos
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