moved transcribe and process dataset scripts to vall_e/emb within the module itself, argparse-ified transcription script
This commit is contained in:
parent
7cdfa3dc0c
commit
597441e48b
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@ -1,103 +0,0 @@
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import os
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import json
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import torch
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import torchaudio
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import whisperx
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from tqdm.auto import tqdm
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from pathlib import Path
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# to-do: use argparser
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batch_size = 16
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device = "cuda"
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dtype = "float16"
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model_name = "large-v3"
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input_audio = "voices"
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output_dataset = "training/metadata"
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skip_existing = True
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diarize = False
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#
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model = whisperx.load_model(model_name, device, compute_type=dtype)
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align_model, align_model_metadata, align_model_language = (None, None, None)
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if diarize:
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diarize_model = whisperx.DiarizationPipeline(device=device)
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else:
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diarize_model = None
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def pad(num, zeroes):
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return str(num).zfill(zeroes+1)
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for dataset_name in os.listdir(f'./{input_audio}/'):
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if not os.path.isdir(f'./{input_audio}/{dataset_name}/'):
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continue
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for speaker_id in tqdm(os.listdir(f'./{input_audio}/{dataset_name}/'), desc="Processing speaker"):
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if not os.path.isdir(f'./{input_audio}/{dataset_name}/{speaker_id}'):
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continue
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outpath = Path(f'./{output_dataset}/{dataset_name}/{speaker_id}/whisper.json')
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if outpath.exists():
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metadata = json.loads(open(outpath, 'r', encoding='utf-8').read())
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else:
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os.makedirs(f'./{output_dataset}/{dataset_name}/{speaker_id}/', exist_ok=True)
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metadata = {}
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for filename in tqdm(os.listdir(f'./{input_audio}/{dataset_name}/{speaker_id}/'), desc=f"Processing speaker: {speaker_id}"):
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if skip_existing and filename in metadata:
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continue
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if ".json" in filename:
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continue
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inpath = f'./{input_audio}/{dataset_name}/{speaker_id}/{filename}'
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if os.path.isdir(inpath):
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continue
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metadata[filename] = {
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"segments": [],
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"language": "",
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"text": "",
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"start": 0,
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"end": 0,
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}
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audio = whisperx.load_audio(inpath)
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result = model.transcribe(audio, batch_size=batch_size)
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language = result["language"]
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if language[:2] not in ["ja"]:
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language = "en"
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if align_model_language != language:
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tqdm.write(f'Loading language: {language}')
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align_model, align_model_metadata = whisperx.load_align_model(language_code=language, device=device)
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align_model_language = language
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result = whisperx.align(result["segments"], align_model, align_model_metadata, audio, device, return_char_alignments=False)
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metadata[filename]["segments"] = result["segments"]
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metadata[filename]["language"] = language
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if diarize_model is not None:
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diarize_segments = diarize_model(audio)
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result = whisperx.assign_word_speakers(diarize_segments, result)
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text = []
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start = 0
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end = 0
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for segment in result["segments"]:
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text.append( segment["text"] )
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start = min( start, segment["start"] )
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end = max( end, segment["end"] )
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metadata[filename]["text"] = " ".join(text).strip()
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metadata[filename]["start"] = start
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metadata[filename]["end"] = end
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open(outpath, 'w', encoding='utf-8').write(json.dumps(metadata))
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@ -1,3 +1,8 @@
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"""
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# Handles processing audio provided through --input-audio of adequately annotated transcriptions provided through --input-metadata (through transcribe.py)
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# Outputs NumPy objects containing quantized audio and adequate metadata for use of loading in the trainer through --output-dataset
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"""
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import os
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import json
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import argparse
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@ -7,8 +12,8 @@ import numpy as np
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from tqdm.auto import tqdm
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from pathlib import Path
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from vall_e.config import cfg
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from ..config import cfg
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def pad(num, zeroes):
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return str(num).zfill(zeroes+1)
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@ -17,14 +22,26 @@ def process_items( items, stride=0 ):
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items = sorted( items )
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return items if stride == 0 else [ item for i, item in enumerate( items ) if i % stride == 0 ]
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def process_dataset( args ):
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def process(
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audio_backend="encodec",
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input_audio="voices",
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input_metadata="metadata",
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output_dataset="training",
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raise_exceptions=False,
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stride=0,
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slice="auto",
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device="cuda",
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dtype="float16",
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amp=False,
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):
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# encodec / vocos
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if args.audio_backend in ["encodec", "vocos"]:
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if audio_backend in ["encodec", "vocos"]:
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audio_extension = ".enc"
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cfg.sample_rate = 24_000
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cfg.model.resp_levels = 8
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elif args.audio_backend == "dac":
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elif audio_backend == "dac":
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audio_extension = ".dac"
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cfg.sample_rate = 44_100
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cfg.model.resp_levels = 9
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@ -33,24 +50,18 @@ def process_dataset( args ):
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audio_extension = ".dec"
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cfg.model.resp_levels = 8 # ?
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else:
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raise Exception(f"Unknown audio backend: {args.audio_backend}")
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raise Exception(f"Unknown audio backend: {audio_backend}")
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# prepare from args
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cfg.audio_backend = args.audio_backend # "encodec"
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cfg.inference.weight_dtype = args.dtype # "bfloat16"
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cfg.inference.amp = args.amp # False
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cfg.audio_backend = audio_backend # "encodec"
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cfg.inference.weight_dtype = dtype # "bfloat16"
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cfg.inference.amp = amp # False
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# import after because we've overriden the config above
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from vall_e.emb.g2p import encode as valle_phonemize
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from vall_e.emb.qnt import encode as valle_quantize, _replace_file_extension
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from .g2p import encode as phonemize
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from .qnt import encode as quantize, _replace_file_extension
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input_audio = args.input_audio # "voice""
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input_metadata = args.input_metadata # "metadata"
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output_group = f"{args.output_group}-{'2' if cfg.sample_rate == 24_000 else '4'}{'8' if cfg.sample_rate == 48_000 else '4'}KHz-{cfg.audio_backend}" # "training"
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device = args.device # "cuda"
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raise_exceptions = args.raise_exceptions # False
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stride = args.stride # 0
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slice = args.slice # "auto"
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output_dataset = f"{output_dataset}/{'2' if cfg.sample_rate == 24_000 else '4'}{'8' if cfg.sample_rate == 48_000 else '4'}KHz-{cfg.audio_backend}" # "training"
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language_map = {} # k = group, v = language
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@ -88,18 +99,18 @@ def process_dataset( args ):
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if only_speakers and speaker_id not in only_speakers:
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continue
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os.makedirs(f'./{output_group}/{group_name}/{speaker_id}/', exist_ok=True)
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os.makedirs(f'./{output_dataset}/{group_name}/{speaker_id}/', exist_ok=True)
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if speaker_id == "Noise":
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for filename in sorted(os.listdir(f'./{input_audio}/{group_name}/{speaker_id}/')):
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inpath = Path(f'./{input_audio}/{group_name}/{speaker_id}/{filename}')
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outpath = Path(f'./{output_group}/{group_name}/{speaker_id}/{filename}')
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outpath = Path(f'./{output_dataset}/{group_name}/{speaker_id}/{filename}')
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if _replace_file_extension(outpath, audio_extension).exists():
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continue
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waveform, sample_rate = torchaudio.load(inpath)
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qnt = valle_quantize(waveform, sr=sample_rate, device=device)
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qnt = quantize(waveform, sr=sample_rate, device=device)
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if cfg.audio_backend == "dac":
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np.save(open(_replace_file_extension(outpath, audio_extension), "wb"), {
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@ -158,7 +169,7 @@ def process_dataset( args ):
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language = language_map[group_name] if group_name in language_map else (metadata[filename]["language"] if "language" in metadata[filename] else "en")
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if len(metadata[filename]["segments"]) == 0 or not use_slices:
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outpath = Path(f'./{output_group}/{group_name}/{speaker_id}/{fname}.{extension}')
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outpath = Path(f'./{output_dataset}/{group_name}/{speaker_id}/{fname}.{extension}')
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text = metadata[filename]["text"]
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if len(text) == 0:
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@ -185,7 +196,7 @@ def process_dataset( args ):
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id = pad(i, 4)
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i = i + 1
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outpath = Path(f'./{output_group}/{group_name}/{speaker_id}/{fname}_{id}.{extension}')
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outpath = Path(f'./{output_dataset}/{group_name}/{speaker_id}/{fname}_{id}.{extension}')
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text = segment["text"]
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if len(text) == 0:
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try:
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outpath, text, language, waveform, sample_rate = job
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phones = valle_phonemize(text, language=language)
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qnt = valle_quantize(waveform, sr=sample_rate, device=device)
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phones = phonemize(text, language=language)
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qnt = quantize(waveform, sr=sample_rate, device=device)
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if cfg.audio_backend == "dac":
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@ -273,8 +284,8 @@ def main():
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parser.add_argument("--dtype", type=str, default="bfloat16")
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parser.add_argument("--amp", action="store_true")
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parser.add_argument("--input-audio", type=str, default="voices")
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parser.add_argument("--input-metadata", type=str, default="metadata")
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parser.add_argument("--output_group", type=str, default="training")
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parser.add_argument("--input-metadata", type=str, default="training/metadata")
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parser.add_argument("--output-dataset", type=str, default="training/dataset")
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parser.add_argument("--device", type=str, default="cuda")
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parser.add_argument("--raise-exceptions", action="store_true")
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parser.add_argument("--stride", type=int, default=0)
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args = parser.parse_args()
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process_dataset( args )
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process(
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audio_backend=args.audio_backend,
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input_audio=args.input_audio,
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input_metadata=args.input_metadata,
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output_dataset=args.output_dataset,
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raise_exceptions=args.raise_exceptions,
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stride=args.stride,
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slice=args.slice,
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device=args.device,
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dtype=args.dtype,
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amp=args.amp,
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)
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if __name__ == "__main__":
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main()
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147
vall_e/emb/transcribe.py
Normal file
147
vall_e/emb/transcribe.py
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"""
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# Handles transcribing audio provided through --input-audio
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"""
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import os
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import json
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import argparse
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import torch
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import torchaudio
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import whisperx
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from tqdm.auto import tqdm
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from pathlib import Path
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def pad(num, zeroes):
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return str(num).zfill(zeroes+1)
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def transcribe(
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input_audio = "voices",
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output_metadata = "training/metadata",
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model_name = "large-v3",
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skip_existing = True,
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diarize = False,
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batch_size = 16,
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device = "cuda",
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dtype = "float16",
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):
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#
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model = whisperx.load_model(model_name, device, compute_type=dtype)
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align_model, align_model_metadata, align_model_language = (None, None, None)
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if diarize:
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diarize_model = whisperx.DiarizationPipeline(device=device)
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else:
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diarize_model = None
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for dataset_name in os.listdir(f'./{input_audio}/'):
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if not os.path.isdir(f'./{input_audio}/{dataset_name}/'):
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continue
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for speaker_id in tqdm(os.listdir(f'./{input_audio}/{dataset_name}/'), desc="Processing speaker"):
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if not os.path.isdir(f'./{input_audio}/{dataset_name}/{speaker_id}'):
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continue
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outpath = Path(f'./{output_metadata}/{dataset_name}/{speaker_id}/whisper.json')
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if outpath.exists():
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metadata = json.loads(open(outpath, 'r', encoding='utf-8').read())
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else:
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os.makedirs(f'./{output_metadata}/{dataset_name}/{speaker_id}/', exist_ok=True)
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metadata = {}
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for filename in tqdm(os.listdir(f'./{input_audio}/{dataset_name}/{speaker_id}/'), desc=f"Processing speaker: {speaker_id}"):
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if skip_existing and filename in metadata:
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continue
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if ".json" in filename:
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continue
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inpath = f'./{input_audio}/{dataset_name}/{speaker_id}/{filename}'
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if os.path.isdir(inpath):
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continue
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metadata[filename] = {
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"segments": [],
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"language": "",
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"text": "",
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"start": 0,
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"end": 0,
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}
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audio = whisperx.load_audio(inpath)
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result = model.transcribe(audio, batch_size=batch_size)
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language = result["language"]
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"""
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if language[:2] not in ["ja"]:
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language = "en"
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"""
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if align_model_language != language:
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tqdm.write(f'Loading language: {language}')
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align_model, align_model_metadata = whisperx.load_align_model(language_code=language, device=device)
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align_model_language = language
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result = whisperx.align(result["segments"], align_model, align_model_metadata, audio, device, return_char_alignments=False)
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metadata[filename]["segments"] = result["segments"]
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metadata[filename]["language"] = language
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if diarize_model is not None:
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diarize_segments = diarize_model(audio)
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result = whisperx.assign_word_speakers(diarize_segments, result)
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text = []
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start = 0
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end = 0
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for segment in result["segments"]:
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text.append( segment["text"] )
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start = min( start, segment["start"] )
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end = max( end, segment["end"] )
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metadata[filename]["text"] = " ".join(text).strip()
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metadata[filename]["start"] = start
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metadata[filename]["end"] = end
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open(outpath, 'w', encoding='utf-8').write(json.dumps(metadata))
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def main():
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parser = argparse.ArgumentParser()
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parser.add_argument("--input-audio", type=str, default="voices")
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parser.add_argument("--output-metadata", type=str, default="training/metadata")
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parser.add_argument("--model-name", type=str, default="large-v3")
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parser.add_argument("--skip-existing", action="store_true")
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parser.add_argument("--diarize", action="store_true")
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parser.add_argument("--batch-size", type=int, default=16)
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parser.add_argument("--device", type=str, default="cuda")
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parser.add_argument("--dtype", type=str, default="bfloat16")
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parser.add_argument("--amp", action="store_true")
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# parser.add_argument("--raise-exceptions", action="store_true")
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args = parser.parse_args()
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transcribe(
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input_audio = args.input_audio,
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output_metadata = args.output_metadata,
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model_name = args.model_name,
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skip_existing = args.skip_existing,
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diarize = args.diarize,
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batch_size = args.batch_size,
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device = args.device,
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dtype = args.dtype,
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)
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if __name__ == "__main__":
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main()
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