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https://github.com/m-bain/whisperX.git
synced 2025-07-01 18:17:27 -04:00
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v3.3.4
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036b5b0717
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3
.github/workflows/build-and-release.yml
vendored
3
.github/workflows/build-and-release.yml
vendored
@ -17,6 +17,9 @@ jobs:
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version: "0.5.14"
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python-version: "3.9"
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- name: Check if lockfile is up to date
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run: uv lock --check
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- name: Build package
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run: uv build
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3
.github/workflows/python-compatibility.yml
vendored
3
.github/workflows/python-compatibility.yml
vendored
@ -23,6 +23,9 @@ jobs:
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version: "0.5.14"
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python-version: ${{ matrix.python-version }}
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- name: Check if lockfile is up to date
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run: uv lock --check
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- name: Install the project
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run: uv sync --all-extras
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23
README.md
23
README.md
@ -97,6 +97,25 @@ uv sync --all-extras --dev
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You may also need to install ffmpeg, rust etc. Follow openAI instructions here https://github.com/openai/whisper#setup.
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### Common Issues & Troubleshooting 🔧
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#### libcudnn Dependencies (GPU Users)
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If you're using WhisperX with GPU support and encounter errors like:
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- `Could not load library libcudnn_ops_infer.so.8`
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- `Unable to load any of {libcudnn_cnn.so.9.1.0, libcudnn_cnn.so.9.1, libcudnn_cnn.so.9, libcudnn_cnn.so}`
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- `libcudnn_ops_infer.so.8: cannot open shared object file: No such file or directory`
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This means your system is missing the CUDA Deep Neural Network library (cuDNN). This library is needed for GPU acceleration but isn't always installed by default.
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**Install cuDNN (example for apt based systems):**
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```bash
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sudo apt update
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sudo apt install libcudnn8 libcudnn8-dev -y
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```
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### Speaker Diarization
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To **enable Speaker Diarization**, include your Hugging Face access token (read) that you can generate from [Here](https://huggingface.co/settings/tokens) after the `--hf_token` argument and accept the user agreement for the following models: [Segmentation](https://huggingface.co/pyannote/segmentation-3.0) and [Speaker-Diarization-3.1](https://huggingface.co/pyannote/speaker-diarization-3.1) (if you choose to use Speaker-Diarization 2.x, follow requirements [here](https://huggingface.co/pyannote/speaker-diarization) instead.)
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@ -170,7 +189,7 @@ result = model.transcribe(audio, batch_size=batch_size)
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print(result["segments"]) # before alignment
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# delete model if low on GPU resources
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# import gc; gc.collect(); torch.cuda.empty_cache(); del model
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# import gc; import torch; gc.collect(); torch.cuda.empty_cache(); del model
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# 2. Align whisper output
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model_a, metadata = whisperx.load_align_model(language_code=result["language"], device=device)
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@ -179,7 +198,7 @@ result = whisperx.align(result["segments"], model_a, metadata, audio, device, re
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print(result["segments"]) # after alignment
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# delete model if low on GPU resources
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# import gc; gc.collect(); torch.cuda.empty_cache(); del model_a
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# import gc; import torch; gc.collect(); torch.cuda.empty_cache(); del model_a
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# 3. Assign speaker labels
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diarize_model = whisperx.diarize.DiarizationPipeline(use_auth_token=YOUR_HF_TOKEN, device=device)
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2
uv.lock
generated
2
uv.lock
generated
@ -2787,7 +2787,7 @@ wheels = [
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[[package]]
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name = "whisperx"
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version = "3.3.3"
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version = "3.3.4"
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source = { editable = "." }
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dependencies = [
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{ name = "ctranslate2" },
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@ -43,6 +43,7 @@ def cli():
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parser.add_argument("--diarize", action="store_true", help="Apply diarization to assign speaker labels to each segment/word")
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parser.add_argument("--min_speakers", default=None, type=int, help="Minimum number of speakers to in audio file")
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parser.add_argument("--max_speakers", default=None, type=int, help="Maximum number of speakers to in audio file")
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parser.add_argument("--diarize_model", default="pyannote/speaker-diarization-3.1", type=str, help="Name of the speaker diarization model to use")
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parser.add_argument("--temperature", type=float, default=0, help="temperature to use for sampling")
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parser.add_argument("--best_of", type=optional_int, default=5, help="number of candidates when sampling with non-zero temperature")
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@ -11,13 +11,14 @@ from whisperx.types import TranscriptionResult, AlignedTranscriptionResult
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class DiarizationPipeline:
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def __init__(
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self,
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model_name="pyannote/speaker-diarization-3.1",
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model_name=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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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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model_config = model_name or "pyannote/speaker-diarization-3.1"
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self.model = Pipeline.from_pretrained(model_config, use_auth_token=use_auth_token).to(device)
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def __call__(
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self,
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@ -57,6 +57,7 @@ def transcribe_task(args: dict, parser: argparse.ArgumentParser):
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diarize: bool = args.pop("diarize")
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min_speakers: int = args.pop("min_speakers")
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max_speakers: int = args.pop("max_speakers")
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diarize_model_name: str = args.pop("diarize_model")
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print_progress: bool = args.pop("print_progress")
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if args["language"] is not None:
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@ -204,8 +205,9 @@ def transcribe_task(args: dict, parser: argparse.ArgumentParser):
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)
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tmp_results = results
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print(">>Performing diarization...")
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print(">>Using model:", diarize_model_name)
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results = []
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diarize_model = DiarizationPipeline(use_auth_token=hf_token, device=device)
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diarize_model = DiarizationPipeline(model_name=diarize_model_name, use_auth_token=hf_token, device=device)
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for result, input_audio_path in tmp_results:
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diarize_segments = diarize_model(
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input_audio_path, min_speakers=min_speakers, max_speakers=max_speakers
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@ -410,7 +410,7 @@ class WriteJSON(ResultWriter):
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def get_writer(
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output_format: str, output_dir: str
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) -> Callable[[dict, TextIO, dict], None]:
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) -> Callable[[dict, str, dict], None]:
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writers = {
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"txt": WriteTXT,
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"vtt": WriteVTT,
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@ -425,7 +425,7 @@ def get_writer(
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if output_format == "all":
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all_writers = [writer(output_dir) for writer in writers.values()]
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def write_all(result: dict, file: TextIO, options: dict):
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def write_all(result: dict, file: str, options: dict):
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for writer in all_writers:
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writer(result, file, options)
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