vall-e/vall_e/emb/process.py

327 lines
10 KiB
Python

"""
# Handles processing audio provided through --input-audio of adequately annotated transcriptions provided through --input-metadata (through transcribe.py)
# Outputs NumPy objects containing quantized audio and adequate metadata for use of loading in the trainer through --output-dataset
"""
import os
import json
import argparse
import torch
import torchaudio
import numpy as np
import logging
_logger = logging.getLogger(__name__)
from tqdm.auto import tqdm
from pathlib import Path
from ..config import cfg
# need to validate if this is safe to import before modifying the config
from .g2p import encode as phonemize
from .qnt import encode as quantize
def pad(num, zeroes):
return str(num).zfill(zeroes+1)
def load_audio( path, device=None ):
waveform, sr = torchaudio.load( path )
if waveform.shape[0] > 1:
# mix channels
waveform = torch.mean(waveform, dim=0, keepdim=True)
if device is not None:
waveform = waveform.to(device)
return waveform, sr
def process_items( items, stride=0, stride_offset=0 ):
items = sorted( items )
return items if stride == 0 else [ item for i, item in enumerate( items ) if (i+stride_offset) % stride == 0 ]
def process_job( outpath, waveform, sample_rate, text=None, language="en", device="cuda" ):
# encodec requires this to be on CPU for resampling
qnt = quantize(waveform, sr=sample_rate, device=device)
if cfg.audio_backend == "dac":
state_dict = {
"codes": qnt.codes.cpu().numpy().astype(np.uint16),
"metadata": {
"original_length": qnt.original_length,
"sample_rate": qnt.sample_rate,
"input_db": qnt.input_db.cpu().numpy().astype(np.float32),
"chunk_length": qnt.chunk_length,
"channels": qnt.channels,
"padding": qnt.padding,
"dac_version": "1.0.0",
},
}
else:
state_dict = {
"codes": qnt.cpu().numpy().astype(np.uint16),
"metadata": {
"original_length": waveform.shape[-1],
"sample_rate": sample_rate,
},
}
if text:
text = text.strip()
state_dict['metadata'] |= {
"text": text,
"phonemes": phonemize(text, language=language),
"language": language,
}
np.save(open(outpath, "wb"), state_dict)
def process_jobs( jobs, speaker_id="", device=None, raise_exceptions=True ):
if not jobs:
return
for job in tqdm(jobs, desc=f"Quantizing: {speaker_id}"):
outpath, waveform, sample_rate, text, language = job
try:
process_job( outpath, waveform, sample_rate, text, language, device )
except Exception as e:
_logger.error(f"Failed to quantize: {outpath}: {str(e)}")
if raise_exceptions:
raise e
continue
def process(
audio_backend="encodec",
input_audio="voices",
input_voice=None,
input_metadata="metadata",
output_dataset="training",
raise_exceptions=False,
stride=0,
stride_offset=0,
slice="auto",
low_memory=False,
device="cuda",
dtype="float16",
amp=False,
):
# prepare from args
cfg.device = device
cfg.set_audio_backend(audio_backend)
audio_extension = cfg.audio_backend_extension
cfg.inference.weight_dtype = dtype # "bfloat16"
cfg.inference.amp = amp # False
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"
# to-do: make this also prepared from args
language_map = {} # k = group, v = language
ignore_groups = [] # skip these groups
ignore_speakers = [] # skip these speakers
only_groups = [] # only process these groups
only_speakers = [] # only process these speakers
always_slice_groups = ["Audiobooks", "LibriVox"] # always slice from this group
audio_only = ["Noise"] # special pathway for processing audio only (without a transcription)
missing = {
"transcription": [],
"audio": []
}
dataset = []
if input_voice is not None:
only_speakers = [input_voice]
for group_name in sorted(os.listdir(f'./{input_audio}/')):
if not os.path.isdir(f'./{input_audio}/{group_name}/'):
_logger.warning(f'Is not dir:" /{input_audio}/{group_name}/')
continue
if group_name in ignore_groups:
continue
if only_groups and group_name not in only_groups:
continue
for speaker_id in tqdm(process_items(os.listdir(f'./{input_audio}/{group_name}/'), stride=stride, stride_offset=stride_offset), desc=f"Processing speaker in {group_name}"):
if not os.path.isdir(f'./{input_audio}/{group_name}/{speaker_id}'):
_logger.warning(f'Is not dir: ./{input_audio}/{group_name}/{speaker_id}')
continue
if speaker_id in ignore_speakers:
continue
if only_speakers and speaker_id not in only_speakers:
continue
os.makedirs(f'./{output_dataset}/{group_name}/{speaker_id}/', exist_ok=True)
if speaker_id in audio_only:
for filename in sorted(os.listdir(f'./{input_audio}/{group_name}/{speaker_id}/')):
inpath = Path(f'./{input_audio}/{group_name}/{speaker_id}/{filename}')
outpath = Path(f'./{output_dataset}/{group_name}/{speaker_id}/{filename}').with_suffix(audio_extension)
if outpath.exists():
continue
waveform, sample_rate = load_audio( inpath )
qnt = quantize(waveform, sr=sample_rate, device=device)
process_job(outpath, waveform, sample_rate)
continue
metadata_path = Path(f'./{input_metadata}/{group_name}/{speaker_id}/whisper.json')
if not metadata_path.exists():
missing["transcription"].append(str(metadata_path))
_logger.warning(f'Missing transcription metadata: ./{input_audio}/{group_name}/{speaker_id}/whisper.json')
continue
try:
metadata = json.loads(open(metadata_path, "r", encoding="utf-8").read())
except Exception as e:
missing["transcription"].append(str(metadata_path))
_logger.warning(f'Failed to open transcription metadata: ./{input_audio}/{group_name}/{speaker_id}/whisper.json: {e}')
continue
if f'{group_name}/{speaker_id}' not in dataset:
dataset.append(f'{group_name}/{speaker_id}')
jobs = []
use_slices = slice == True or (slice == "auto" and len(metadata.keys()) == 1) or group_name in always_slice_groups
for filename in sorted(metadata.keys()):
inpath = Path(f'./{input_audio}/{group_name}/{speaker_id}/{filename}')
"""
if not inpath.exists():
missing["audio"].append(str(inpath))
continue
"""
extension = os.path.splitext(filename)[-1][1:]
fname = filename.replace(f'.{extension}', "")
waveform, sample_rate = None, None
language = language_map[group_name] if group_name in language_map else (metadata[filename]["language"] if "language" in metadata[filename] else "en")
if len(metadata[filename]["segments"]) == 0 or not use_slices:
outpath = Path(f'./{output_dataset}/{group_name}/{speaker_id}/{fname}.{extension}').with_suffix(audio_extension)
text = metadata[filename]["text"]
if len(text) == 0 or outpath.exists():
continue
# audio not already loaded, load it
if waveform is None:
waveform, sample_rate = load_audio( inpath )
jobs.append(( outpath, waveform, sample_rate, text, language ))
else:
i = 0
presliced = not inpath.exists()
for segment in metadata[filename]["segments"]:
id = pad(i, 4)
i = i + 1
if presliced:
inpath = Path(f'./{input_audio}/{group_name}/{speaker_id}/{fname}_{id}.{extension}')
if not inpath.exists():
missing["audio"].append(str(inpath))
continue
outpath = Path(f'./{output_dataset}/{group_name}/{speaker_id}/{fname}_{id}.{extension}').with_suffix(audio_extension)
text = segment["text"]
if len(text) == 0 or outpath.exists():
continue
# audio not already loaded, load it
if waveform is None:
waveform, sample_rate = load_audio( inpath )
start = int((segment['start']-0.05) * sample_rate)
end = int((segment['end']+0.5) * sample_rate)
if not presliced:
if start < 0:
start = 0
if end >= waveform.shape[-1]:
end = waveform.shape[-1] - 1
if end - start < 0:
continue
jobs.append(( outpath, waveform if presliced else waveform[:, start:end], sample_rate, text, language ))
# processes audio files one at a time
if low_memory:
process_jobs( jobs, device=device, speaker_id=f'{speaker_id}/{filename}', raise_exceptions=raise_exceptions )
jobs = []
# processes all audio files for a given speaker
if not low_memory:
process_jobs( jobs, device=device, speaker_id=speaker_id, raise_exceptions=raise_exceptions )
jobs = []
open(f"./{output_dataset}/missing.json", 'w', encoding='utf-8').write(json.dumps(missing))
open(f"./{output_dataset}/dataset.json", 'w', encoding='utf-8').write(json.dumps(dataset))
def main():
parser = argparse.ArgumentParser()
parser.add_argument("--audio-backend", type=str, default="encodec")
parser.add_argument("--input-audio", type=str, default="voices")
parser.add_argument("--input-voice", type=str, default=None)
parser.add_argument("--input-metadata", type=str, default="training/metadata")
parser.add_argument("--output-dataset", type=str, default="training/dataset")
parser.add_argument("--raise-exceptions", action="store_true")
parser.add_argument("--low-memory", action="store_true")
parser.add_argument("--stride", type=int, default=0)
parser.add_argument("--stride-offset", type=int, default=0)
parser.add_argument("--slice", type=str, default="auto")
parser.add_argument("--device", type=str, default="cuda")
parser.add_argument("--dtype", type=str, default="bfloat16")
parser.add_argument("--amp", action="store_true")
args = parser.parse_args()
# do some assumption magic
# to-do: find a nice way to spawn multiple processes where tqdm plays nicely
if args.device.isnumeric():
args.stride = torch.cuda.device_count()
args.stride_offset = int(args.device)
args.device = f'cuda:{args.device}'
if args.slice == "true":
args.slice = True
elif args.slice == "false":
args.slice = False
process(
audio_backend=args.audio_backend,
input_audio=args.input_audio,
input_voice=args.input_voice,
input_metadata=args.input_metadata,
output_dataset=args.output_dataset,
raise_exceptions=args.raise_exceptions,
stride=args.stride,
stride_offset=args.stride_offset,
slice=args.slice,
low_memory=args.low_memory,
device=args.device,
dtype=args.dtype,
amp=args.amp,
)
if __name__ == "__main__":
main()