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import time |
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from datasets import Dataset |
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import warnings |
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import argparse |
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import os |
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from helpers import * |
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from faster_whisper import WhisperModel |
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import whisperx |
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import torch |
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from pydub import AudioSegment |
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from nemo.collections.asr.models.msdd_models import NeuralDiarizer |
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from deepmultilingualpunctuation import PunctuationModel |
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import re |
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import logging |
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mtypes = {"cpu": "int8", "cuda": "float16"} |
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parser = argparse.ArgumentParser() |
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parser.add_argument( |
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"-a", "--audio", help="name of the target audio file", required=True |
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) |
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parser.add_argument( |
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"--no-stem", |
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action="store_false", |
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dest="stemming", |
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default=True, |
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help="Disables source separation." |
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"This helps with long files that don't contain a lot of music.", |
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) |
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parser.add_argument( |
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"--suppress_numerals", |
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action="store_true", |
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dest="suppress_numerals", |
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default=False, |
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help="Suppresses Numerical Digits." |
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"This helps the diarization accuracy but converts all digits into written text.", |
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) |
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parser.add_argument( |
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"--whisper-model", |
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dest="model_name", |
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default="medium.en", |
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help="name of the Whisper model to use", |
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) |
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parser.add_argument( |
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"--batch-size", |
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type=int, |
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dest="batch_size", |
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default=8, |
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help="Batch size for batched inference, reduce if you run out of memory, set to 0 for non-batched inference", |
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) |
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parser.add_argument( |
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"--language", |
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type=str, |
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default=None, |
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choices=whisper_langs, |
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help="Language spoken in the audio, specify None to perform language detection", |
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) |
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parser.add_argument( |
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"--device", |
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dest="device", |
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default="cuda" if torch.cuda.is_available() else "cpu", |
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help="if you have a GPU use 'cuda', otherwise 'cpu'", |
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) |
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args = parser.parse_args() |
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if args.stemming: |
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return_code = os.system( |
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f'python3 -m demucs.separate -n htdemucs --two-stems=vocals "{args.audio}" -o "temp_outputs"' |
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) |
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if return_code != 0: |
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logging.warning( |
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"Source splitting failed, using original audio file. Use --no-stem argument to disable it." |
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) |
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vocal_target = args.audio |
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else: |
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vocal_target = os.path.join( |
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"temp_outputs", |
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"htdemucs", |
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os.path.splitext(os.path.basename(args.audio))[0], |
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"vocals.wav", |
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) |
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else: |
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vocal_target = args.audio |
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if args.batch_size != 0: |
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from transcription_helpers import transcribe_batched |
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whisper_results, language = transcribe_batched( |
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vocal_target, |
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args.language, |
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args.batch_size, |
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args.model_name, |
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mtypes[args.device], |
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args.suppress_numerals, |
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args.device, |
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) |
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else: |
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from transcription_helpers import transcribe |
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whisper_results, language = transcribe( |
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vocal_target, |
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args.language, |
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args.model_name, |
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mtypes[args.device], |
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args.suppress_numerals, |
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args.device, |
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) |
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if language in wav2vec2_langs: |
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alignment_model, metadata = whisperx.load_align_model( |
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language_code=language, device=args.device |
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) |
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result_aligned = whisperx.align( |
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whisper_results, alignment_model, metadata, vocal_target, args.device |
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) |
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word_timestamps = filter_missing_timestamps( |
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result_aligned["word_segments"], |
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initial_timestamp=whisper_results[0].get("start"), |
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final_timestamp=whisper_results[-1].get("end"), |
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) |
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del alignment_model |
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torch.cuda.empty_cache() |
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else: |
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assert ( |
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args.batch_size == 0 |
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), ( |
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f"Unsupported language: {language}, use --batch_size to 0" |
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" to generate word timestamps using whisper directly and fix this error." |
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) |
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word_timestamps = [] |
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for segment in whisper_results: |
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for word in segment["words"]: |
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word_timestamps.append({"word": word[2], "start": word[0], "end": word[1]}) |
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sound = AudioSegment.from_file(vocal_target).set_channels(1) |
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ROOT = os.getcwd() |
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temp_path = os.path.join(ROOT, "temp_outputs") |
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os.makedirs(temp_path, exist_ok=True) |
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sound.export(os.path.join(temp_path, "mono_file.wav"), format="wav") |
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msdd_model = NeuralDiarizer(cfg=create_config(temp_path)).to(args.device) |
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msdd_model.diarize() |
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del msdd_model |
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torch.cuda.empty_cache() |
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speaker_ts = [] |
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with open(os.path.join(temp_path, "pred_rttms", "mono_file.rttm"), "r") as f: |
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lines = f.readlines() |
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for line in lines: |
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line_list = line.split(" ") |
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s = int(float(line_list[5]) * 1000) |
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e = s + int(float(line_list[8]) * 1000) |
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speaker_ts.append([s, e, int(line_list[11].split("_")[-1])]) |
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wsm = get_words_speaker_mapping(word_timestamps, speaker_ts, "start") |
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if language in punct_model_langs: |
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punct_model = PunctuationModel(model="kredor/punctuate-all") |
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words_list = list(map(lambda x: x["word"], wsm)) |
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while True: |
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try: |
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labled_words = punct_model.pipe(words_list) |
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break |
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except ValueError as e: |
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if str(e) == "Queue is full! Please try again.": |
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print("Queue is full. Retrying in 1 second...") |
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time.sleep(1) |
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else: |
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raise e |
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ending_puncts = ".?!" |
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model_puncts = ".,;:!?" |
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is_acronym = lambda x: re.fullmatch(r"\b(?:[a-zA-Z]\.){2,}", x) |
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for i, labeled_tuple in enumerate(labled_words): |
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word = wsm[i]["word"] |
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if ( |
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word |
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and labeled_tuple |
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and "entity" in labeled_tuple[0] |
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and labeled_tuple[0]["entity"] in ending_puncts |
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and (word[-1] not in model_puncts or is_acronym(word)) |
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): |
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word += labeled_tuple[0]["entity"] |
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if word.endswith(".."): |
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word = word.rstrip(".") |
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wsm[i]["word"] = word |
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else: |
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logging.warning( |
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f"Punctuation restoration is not available for {language} language. Using the original punctuation." |
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) |
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wsm = get_realigned_ws_mapping_with_punctuation(wsm) |
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ssm = get_sentences_speaker_mapping(wsm, speaker_ts) |
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with open(f"{os.path.splitext(args.audio)[0]}.txt", "w", encoding="utf-8-sig") as f: |
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get_speaker_aware_transcript(ssm, f) |
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with open(f"{os.path.splitext(args.audio)[0]}.srt", "w", encoding="utf-8-sig") as srt: |
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write_srt(ssm, srt) |
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cleanup(temp_path) |
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