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https://github.com/m-bain/whisperX.git
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Merge pull request #210 from sorgfresser/v3
Update pyannote and torch version
This commit is contained in:
@ -1,6 +1,5 @@
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torch==1.11.0
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torch==2.0.0
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torchaudio==0.11.0
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torchaudio==2.0.1
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pyannote.audio
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faster-whisper
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faster-whisper
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transformers
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transformers
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ffmpeg-python==0.2.0
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ffmpeg-python==0.2.0
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2
setup.py
2
setup.py
@ -19,7 +19,7 @@ setup(
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for r in pkg_resources.parse_requirements(
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for r in pkg_resources.parse_requirements(
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open(os.path.join(os.path.dirname(__file__), "requirements.txt"))
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open(os.path.join(os.path.dirname(__file__), "requirements.txt"))
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)
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)
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],
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] + ["pyannote.audio @ git+https://github.com/pyannote/pyannote-audio@11b56a137a578db9335efc00298f6ec1932e6317"],
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entry_points = {
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entry_points = {
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'console_scripts': ['whisperx=whisperx.transcribe:cli'],
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'console_scripts': ['whisperx=whisperx.transcribe:cli'],
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},
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},
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@ -181,6 +181,9 @@ class FasterWhisperPipeline(Pipeline):
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def preprocess(self, audio):
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def preprocess(self, audio):
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audio = audio['inputs']
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audio = audio['inputs']
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if isinstance(audio, np.ndarray):
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audio = torch.from_numpy(audio)
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features = log_mel_spectrogram(audio, padding=N_SAMPLES - audio.shape[0])
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features = log_mel_spectrogram(audio, padding=N_SAMPLES - audio.shape[0])
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return {'inputs': features}
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return {'inputs': features}
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@ -253,7 +256,7 @@ class FasterWhisperPipeline(Pipeline):
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def detect_language(self, audio: np.ndarray):
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def detect_language(self, audio: np.ndarray):
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if audio.shape[0] < N_SAMPLES:
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if audio.shape[0] < N_SAMPLES:
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print("Warning: audio is shorter than 30s, language detection may be inaccurate.")
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print("Warning: audio is shorter than 30s, language detection may be inaccurate.")
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segment = log_mel_spectrogram(audio[: N_SAMPLES],
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segment = log_mel_spectrogram(torch.from_numpy(audio[:N_SAMPLES]),
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padding=0 if audio.shape[0] >= N_SAMPLES else N_SAMPLES - audio.shape[0])
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padding=0 if audio.shape[0] >= N_SAMPLES else N_SAMPLES - audio.shape[0])
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encoder_output = self.model.encode(segment)
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encoder_output = self.model.encode(segment)
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results = self.model.model.detect_language(encoder_output)
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results = self.model.model.detect_language(encoder_output)
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@ -22,6 +22,12 @@ N_SAMPLES_PER_TOKEN = HOP_LENGTH * 2 # the initial convolutions has stride 2
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FRAMES_PER_SECOND = exact_div(SAMPLE_RATE, HOP_LENGTH) # 10ms per audio frame
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FRAMES_PER_SECOND = exact_div(SAMPLE_RATE, HOP_LENGTH) # 10ms per audio frame
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TOKENS_PER_SECOND = exact_div(SAMPLE_RATE, N_SAMPLES_PER_TOKEN) # 20ms per audio token
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TOKENS_PER_SECOND = exact_div(SAMPLE_RATE, N_SAMPLES_PER_TOKEN) # 20ms per audio token
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with np.load(
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os.path.join(os.path.dirname(__file__), "assets", "mel_filters.npz")
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) as f:
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MEL_FILTERS = torch.from_numpy(f[f"mel_{80}"])
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def load_audio(file: str, sr: int = SAMPLE_RATE):
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def load_audio(file: str, sr: int = SAMPLE_RATE):
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"""
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"""
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@ -79,27 +85,9 @@ def pad_or_trim(array, length: int = N_SAMPLES, *, axis: int = -1):
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return array
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return array
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@lru_cache(maxsize=None)
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@torch.compile(fullgraph=True)
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def mel_filters(device, n_mels: int = N_MELS) -> torch.Tensor:
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"""
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load the mel filterbank matrix for projecting STFT into a Mel spectrogram.
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Allows decoupling librosa dependency; saved using:
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np.savez_compressed(
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"mel_filters.npz",
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mel_80=librosa.filters.mel(sr=16000, n_fft=400, n_mels=80),
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)
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"""
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assert n_mels == 80, f"Unsupported n_mels: {n_mels}"
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with np.load(
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os.path.join(os.path.dirname(__file__), "assets", "mel_filters.npz")
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) as f:
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return torch.from_numpy(f[f"mel_{n_mels}"]).to(device)
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def log_mel_spectrogram(
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def log_mel_spectrogram(
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audio: Union[str, np.ndarray, torch.Tensor],
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audio: torch.Tensor,
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n_mels: int = N_MELS,
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padding: int = 0,
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padding: int = 0,
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device: Optional[Union[str, torch.device]] = None,
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device: Optional[Union[str, torch.device]] = None,
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):
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):
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@ -108,7 +96,7 @@ def log_mel_spectrogram(
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Parameters
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Parameters
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----------
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----------
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audio: Union[str, np.ndarray, torch.Tensor], shape = (*)
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audio: torch.Tensor, shape = (*)
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The path to audio or either a NumPy array or Tensor containing the audio waveform in 16 kHz
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The path to audio or either a NumPy array or Tensor containing the audio waveform in 16 kHz
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n_mels: int
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n_mels: int
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@ -125,21 +113,19 @@ def log_mel_spectrogram(
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torch.Tensor, shape = (80, n_frames)
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torch.Tensor, shape = (80, n_frames)
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A Tensor that contains the Mel spectrogram
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A Tensor that contains the Mel spectrogram
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"""
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"""
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if not torch.is_tensor(audio):
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global MEL_FILTERS
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if isinstance(audio, str):
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audio = load_audio(audio)
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audio = torch.from_numpy(audio)
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if device is not None:
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if device is not None:
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audio = audio.to(device)
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audio = audio.to(device)
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if padding > 0:
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if padding > 0:
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audio = F.pad(audio, (0, padding))
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audio = F.pad(audio, (0, padding))
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window = torch.hann_window(N_FFT).to(audio.device)
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window = torch.hann_window(N_FFT).to(audio.device)
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stft = torch.stft(audio, N_FFT, HOP_LENGTH, window=window, return_complex=True)
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stft = torch.stft(audio, N_FFT, HOP_LENGTH, window=window, return_complex=False)
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magnitudes = stft[..., :-1].abs() ** 2
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# Square the real and imaginary components and sum them together, similar to torch.abs() on complex tensors
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magnitudes = (stft[:, :-1, :] ** 2).sum(dim=-1)
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filters = mel_filters(audio.device, n_mels)
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MEL_FILTERS = MEL_FILTERS.to(audio.device)
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mel_spec = filters @ magnitudes
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mel_spec = MEL_FILTERS @ magnitudes
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log_spec = torch.clamp(mel_spec, min=1e-10).log10()
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log_spec = torch.clamp(mel_spec, min=1e-10).log10()
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log_spec = torch.maximum(log_spec, log_spec.max() - 8.0)
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log_spec = torch.maximum(log_spec, log_spec.max() - 8.0)
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@ -1,14 +1,19 @@
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import numpy as np
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import numpy as np
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import pandas as pd
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import pandas as pd
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from pyannote.audio import Pipeline
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from pyannote.audio import Pipeline
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from typing import Optional, Union
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import torch
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class DiarizationPipeline:
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class DiarizationPipeline:
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def __init__(
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def __init__(
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self,
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self,
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model_name="pyannote/speaker-diarization@2.1",
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model_name="pyannote/speaker-diarization@2.1",
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use_auth_token=None,
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use_auth_token=None,
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device: Optional[Union[str, torch.device]] = "cpu",
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):
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):
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self.model = Pipeline.from_pretrained(model_name, use_auth_token=use_auth_token)
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if isinstance(device, str):
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device = torch.device(device)
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self.model = Pipeline.from_pretrained(model_name, use_auth_token=use_auth_token).to(device)
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def __call__(self, audio, min_speakers=None, max_speakers=None):
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def __call__(self, audio, min_speakers=None, max_speakers=None):
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segments = self.model(audio, min_speakers=min_speakers, max_speakers=max_speakers)
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segments = self.model(audio, min_speakers=min_speakers, max_speakers=max_speakers)
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@ -193,8 +193,9 @@ def cli():
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if hf_token is None:
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if hf_token is None:
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print("Warning, no --hf_token used, needs to be saved in environment variable, otherwise will throw error loading diarization model...")
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print("Warning, no --hf_token used, needs to be saved in environment variable, otherwise will throw error loading diarization model...")
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tmp_results = results
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tmp_results = results
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print(">>Performing diarization...")
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results = []
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results = []
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diarize_model = DiarizationPipeline(use_auth_token=hf_token)
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diarize_model = DiarizationPipeline(use_auth_token=hf_token, device=device)
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for result, input_audio_path in tmp_results:
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for result, input_audio_path in tmp_results:
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diarize_segments = diarize_model(input_audio_path, min_speakers=min_speakers, max_speakers=max_speakers)
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diarize_segments = diarize_model(input_audio_path, min_speakers=min_speakers, max_speakers=max_speakers)
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results_segments, word_segments = assign_word_speakers(diarize_segments, result["segments"])
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results_segments, word_segments = assign_word_speakers(diarize_segments, result["segments"])
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