mirror of
https://github.com/m-bain/whisperX.git
synced 2025-07-01 18:17:27 -04:00
215 lines
13 KiB
Python
215 lines
13 KiB
Python
import argparse
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import gc
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import os
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import warnings
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import numpy as np
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import torch
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from .alignment import align, load_align_model
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from .asr import load_model
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from .audio import load_audio
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from .diarize import DiarizationPipeline, assign_word_speakers
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from .utils import (LANGUAGES, TO_LANGUAGE_CODE, get_writer, optional_float,
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optional_int, str2bool)
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def cli():
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# fmt: off
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parser = argparse.ArgumentParser(formatter_class=argparse.ArgumentDefaultsHelpFormatter)
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parser.add_argument("audio", nargs="+", type=str, help="audio file(s) to transcribe")
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parser.add_argument("--model", default="small", help="name of the Whisper model to use")
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parser.add_argument("--model_dir", type=str, default=None, help="the path to save model files; uses ~/.cache/whisper by default")
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parser.add_argument("--device", default="cuda" if torch.cuda.is_available() else "cpu", help="device to use for PyTorch inference")
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parser.add_argument("--batch_size", default=8, type=int, help="device to use for PyTorch inference")
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parser.add_argument("--compute_type", default="float16", type=str, choices=["float16", "float32", "int8"], help="compute type for computation")
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parser.add_argument("--output_dir", "-o", type=str, default=".", help="directory to save the outputs")
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parser.add_argument("--output_format", "-f", type=str, default="all", choices=["all", "srt", "vtt", "txt", "tsv", "json"], help="format of the output file; if not specified, all available formats will be produced")
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parser.add_argument("--verbose", type=str2bool, default=True, help="whether to print out the progress and debug messages")
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parser.add_argument("--task", type=str, default="transcribe", choices=["transcribe", "translate"], help="whether to perform X->X speech recognition ('transcribe') or X->English translation ('translate')")
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parser.add_argument("--language", type=str, default=None, choices=sorted(LANGUAGES.keys()) + sorted([k.title() for k in TO_LANGUAGE_CODE.keys()]), help="language spoken in the audio, specify None to perform language detection")
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# alignment params
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parser.add_argument("--align_model", default=None, help="Name of phoneme-level ASR model to do alignment")
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parser.add_argument("--interpolate_method", default="nearest", choices=["nearest", "linear", "ignore"], help="For word .srt, method to assign timestamps to non-aligned words, or merge them into neighbouring.")
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parser.add_argument("--no_align", action='store_true', help="Do not perform phoneme alignment")
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# vad params
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parser.add_argument("--vad_onset", type=float, default=0.500, help="Onset threshold for VAD (see pyannote.audio), reduce this if speech is not being detected")
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parser.add_argument("--vad_offset", type=float, default=0.363, help="Offset threshold for VAD (see pyannote.audio), reduce this if speech is not being detected.")
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# diarization params
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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)
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parser.add_argument("--max_speakers", default=None, type=int)
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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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parser.add_argument("--beam_size", type=optional_int, default=5, help="number of beams in beam search, only applicable when temperature is zero")
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parser.add_argument("--patience", type=float, default=None, help="optional patience value to use in beam decoding, as in https://arxiv.org/abs/2204.05424, the default (1.0) is equivalent to conventional beam search")
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parser.add_argument("--length_penalty", type=float, default=None, help="optional token length penalty coefficient (alpha) as in https://arxiv.org/abs/1609.08144, uses simple length normalization by default")
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parser.add_argument("--suppress_tokens", type=str, default="-1", help="comma-separated list of token ids to suppress during sampling; '-1' will suppress most special characters except common punctuations")
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parser.add_argument("--initial_prompt", type=str, default=None, help="optional text to provide as a prompt for the first window.")
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parser.add_argument("--condition_on_previous_text", type=str2bool, default=False, help="if True, provide the previous output of the model as a prompt for the next window; disabling may make the text inconsistent across windows, but the model becomes less prone to getting stuck in a failure loop")
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parser.add_argument("--fp16", type=str2bool, default=True, help="whether to perform inference in fp16; True by default")
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parser.add_argument("--temperature_increment_on_fallback", type=optional_float, default=0.2, help="temperature to increase when falling back when the decoding fails to meet either of the thresholds below")
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parser.add_argument("--compression_ratio_threshold", type=optional_float, default=2.4, help="if the gzip compression ratio is higher than this value, treat the decoding as failed")
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parser.add_argument("--logprob_threshold", type=optional_float, default=-1.0, help="if the average log probability is lower than this value, treat the decoding as failed")
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parser.add_argument("--no_speech_threshold", type=optional_float, default=0.6, help="if the probability of the <|nospeech|> token is higher than this value AND the decoding has failed due to `logprob_threshold`, consider the segment as silence")
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parser.add_argument("--max_line_width", type=optional_int, default=None, help="(not possible with --no_align) the maximum number of characters in a line before breaking the line")
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parser.add_argument("--max_line_count", type=optional_int, default=None, help="(requires --no_align) the maximum number of lines in a segment")
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parser.add_argument("--highlight_words", type=str2bool, default=False, help="(requires --word_timestamps True) underline each word as it is spoken in srt and vtt")
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# parser.add_argument("--word_timestamps", type=str2bool, default=False, help="(experimental) extract word-level timestamps and refine the results based on them")
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# parser.add_argument("--prepend_punctuations", type=str, default="\"\'“¿([{-", help="if word_timestamps is True, merge these punctuation symbols with the next word")
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# parser.add_argument("--append_punctuations", type=str, default="\"\'.。,，!！?？:：”)]}、", help="if word_timestamps is True, merge these punctuation symbols with the previous word")
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parser.add_argument("--threads", type=optional_int, default=0, help="number of threads used by torch for CPU inference; supercedes MKL_NUM_THREADS/OMP_NUM_THREADS")
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parser.add_argument("--hf_token", type=str, default=None, help="Hugging Face Access Token to access PyAnnote gated models")
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# parser.add_argument("--model_flush", action="store_true", help="Flush memory from each model after use, reduces GPU requirement but slower processing >1 audio file.")
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parser.add_argument("--tmp_dir", default=None, help="Temporary directory to write audio file if input if not .wav format (only for VAD).")
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# fmt: on
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args = parser.parse_args().__dict__
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model_name: str = args.pop("model")
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batch_size: int = args.pop("batch_size")
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output_dir: str = args.pop("output_dir")
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output_format: str = args.pop("output_format")
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device: str = args.pop("device")
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compute_type: str = args.pop("compute_type")
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# model_flush: bool = args.pop("model_flush")
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os.makedirs(output_dir, exist_ok=True)
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tmp_dir: str = args.pop("tmp_dir")
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if tmp_dir is not None:
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os.makedirs(tmp_dir, exist_ok=True)
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align_model: str = args.pop("align_model")
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interpolate_method: str = args.pop("interpolate_method")
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no_align: bool = args.pop("no_align")
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hf_token: str = args.pop("hf_token")
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vad_onset: float = args.pop("vad_onset")
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vad_offset: float = args.pop("vad_offset")
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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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# TODO: check model loading works.
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if model_name.endswith(".en") and args["language"] not in {"en", "English"}:
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if args["language"] is not None:
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warnings.warn(
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f"{model_name} is an English-only model but receipted '{args['language']}'; using English instead."
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)
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args["language"] = "en"
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temperature = args.pop("temperature")
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if (increment := args.pop("temperature_increment_on_fallback")) is not None:
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temperature = tuple(np.arange(temperature, 1.0 + 1e-6, increment))
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else:
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temperature = [temperature]
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if (threads := args.pop("threads")) > 0:
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torch.set_num_threads(threads)
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asr_options = {
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"beam_size": args.pop("beam_size"),
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"patience": args.pop("patience"),
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"length_penalty": args.pop("length_penalty"),
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"temperatures": temperature,
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"compression_ratio_threshold": args.pop("compression_ratio_threshold"),
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"log_prob_threshold": args.pop("logprob_threshold"),
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"no_speech_threshold": args.pop("no_speech_threshold"),
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"condition_on_previous_text": False,
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"initial_prompt": args.pop("initial_prompt"),
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}
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writer = get_writer(output_format, output_dir)
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word_options = ["highlight_words", "max_line_count", "max_line_width"]
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if no_align:
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for option in word_options:
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if args[option]:
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parser.error(f"--{option} requires --word_timestamps True")
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if args["max_line_count"] and not args["max_line_width"]:
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warnings.warn("--max_line_count has no effect without --max_line_width")
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writer_args = {arg: args.pop(arg) for arg in word_options}
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# Part 1: VAD & ASR Loop
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results = []
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tmp_results = []
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# model = load_model(model_name, device=device, download_root=model_dir)
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model = load_model(model_name, device=device, compute_type=compute_type, language=args['language'], asr_options=asr_options, vad_options={"vad_onset": vad_onset, "vad_offset": vad_offset},)
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for audio_path in args.pop("audio"):
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audio = load_audio(audio_path)
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# >> VAD & ASR
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print(">>Performing transcription...")
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result = model.transcribe(audio, batch_size=batch_size)
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results.append((result, audio_path))
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# Unload Whisper and VAD
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del model
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gc.collect()
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torch.cuda.empty_cache()
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# Part 2: Align Loop
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if not no_align:
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tmp_results = results
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results = []
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align_language = args["language"] if args["language"] is not None else "en" # default to loading english if not specified
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align_model, align_metadata = load_align_model(align_language, device, model_name=align_model)
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for result, audio_path in tmp_results:
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# >> Align
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if len(tmp_results) > 1:
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input_audio = audio_path
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else:
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# lazily load audio from part 1
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input_audio = audio
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if align_model is not None and len(result["segments"]) > 0:
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if result.get("language", "en") != align_metadata["language"]:
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# load new language
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print(f"New language found ({result['language']})! Previous was ({align_metadata['language']}), loading new alignment model for new language...")
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align_model, align_metadata = load_align_model(result["language"], device)
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print(">>Performing alignment...")
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result = align(result["segments"], align_model, align_metadata, input_audio, device, interpolate_method=interpolate_method)
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results.append((result, audio_path))
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# Unload align model
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del align_model
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gc.collect()
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torch.cuda.empty_cache()
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# >> Diarize
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if diarize:
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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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tmp_results = results
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results = []
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diarize_model = DiarizationPipeline(use_auth_token=hf_token)
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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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results_segments, word_segments = assign_word_speakers(diarize_segments, result["segments"])
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result = {"segments": results_segments, "word_segments": word_segments}
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results.append((result, input_audio_path))
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# >> Write
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for result, audio_path in results:
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# Remove pandas dataframes from result so that
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# we can serialize the result with json
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for seg in result["segments"]:
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seg.pop("word-segments", None)
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seg.pop("char-segments", None)
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writer(result, audio_path, writer_args)
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if __name__ == "__main__":
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cli() |