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281 lines
10 KiB
Python
281 lines
10 KiB
Python
# Copyright (c) 2022, NVIDIA CORPORATION & AFFILIATES. All rights reserved.
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# http://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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"""
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This script is used to preprocess audio before TTS model training.
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It can be configured to do several processing steps such as silence trimming, volume normalization,
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and duration filtering.
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These can be done separately through multiple executions of the script, or all at once to avoid saving
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too many copies of the same audio.
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Most of these can also be done by the TTS data loader at training time, but doing them ahead of time
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lets us implement more complex processing, validate the correctness of the output, and save on compute time.
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$ python <nemo_root_path>/scripts/dataset_processing/tts/preprocess_audio.py \
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--input_manifest="<data_root_path>/manifest.json" \
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--output_manifest="<data_root_path>/manifest_processed.json" \
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--input_audio_dir="<data_root_path>/audio" \
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--output_audio_dir="<data_root_path>/audio_processed" \
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--num_workers=1 \
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--trim_config_path="<nemo_root_path>/examples/tts/conf/trim/energy.yaml" \
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--output_sample_rate=22050 \
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--output_format=flac \
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--volume_level=0.95 \
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--min_duration=0.5 \
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--max_duration=20.0 \
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--filter_file="filtered.txt"
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"""
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import argparse
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import os
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from pathlib import Path
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from typing import Tuple
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import librosa
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import soundfile as sf
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from joblib import Parallel, delayed
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from omegaconf import OmegaConf
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from tqdm import tqdm
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from nemo.collections.asr.parts.utils.manifest_utils import read_manifest, write_manifest
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from nemo.collections.tts.parts.preprocessing.audio_trimming import AudioTrimmer
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from nemo.collections.tts.parts.utils.tts_dataset_utils import get_abs_rel_paths, normalize_volume
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from nemo.core.classes.common import safe_instantiate
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from nemo.utils import logging
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def get_args():
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parser = argparse.ArgumentParser(
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formatter_class=argparse.ArgumentDefaultsHelpFormatter,
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description="Compute speaker level pitch statistics.",
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)
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parser.add_argument(
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"--input_manifest",
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required=True,
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type=Path,
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help="Path to input training manifest.",
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)
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parser.add_argument(
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"--input_audio_dir",
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required=True,
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type=Path,
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help="Path to base directory with audio files.",
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)
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parser.add_argument(
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"--output_manifest",
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required=True,
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type=Path,
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help="Path to output training manifest with processed audio.",
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)
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parser.add_argument(
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"--output_audio_dir",
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required=True,
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type=Path,
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help="Path to output directory for audio files.",
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)
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parser.add_argument(
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"--overwrite_audio",
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action=argparse.BooleanOptionalAction,
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help="Whether to reprocess and overwrite existing audio files in output_audio_dir.",
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)
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parser.add_argument(
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"--overwrite_manifest",
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action=argparse.BooleanOptionalAction,
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help="Whether to overwrite the output manifest file if it exists.",
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)
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parser.add_argument(
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"--num_workers", default=1, type=int, help="Number of parallel threads to use. If -1 all CPUs are used."
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)
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parser.add_argument(
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"--trim_config_path",
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required=False,
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type=Path,
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help="Path to config file for nemo.collections.tts.data.audio_trimming.AudioTrimmer",
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)
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parser.add_argument(
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"--max_entries", default=0, type=int, help="If provided, maximum number of entries in the manifest to process."
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)
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parser.add_argument(
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"--output_sample_rate", default=0, type=int, help="If provided, rate to resample the audio to."
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)
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parser.add_argument(
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"--output_format",
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default="wav",
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type=str,
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help="If provided, format output audio will be saved as. If not provided, will keep original format.",
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)
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parser.add_argument(
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"--volume_level", default=0.0, type=float, help="If provided, peak volume to normalize audio to."
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)
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parser.add_argument(
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"--min_duration", default=0.0, type=float, help="If provided, filter out utterances shorter than min_duration."
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)
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parser.add_argument(
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"--max_duration", default=0.0, type=float, help="If provided, filter out utterances longer than max_duration."
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)
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parser.add_argument(
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"--filter_file",
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required=False,
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type=Path,
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help="If provided, output filter_file will contain list of " "utterances filtered out.",
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)
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args = parser.parse_args()
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return args
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def _process_entry(
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entry: dict,
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input_audio_dir: Path,
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output_audio_dir: Path,
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overwrite_audio: bool,
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audio_trimmer: AudioTrimmer,
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output_sample_rate: int,
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output_format: str,
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volume_level: float,
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) -> Tuple[dict, float, float]:
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audio_filepath = Path(entry["audio_filepath"])
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audio_path, audio_path_rel = get_abs_rel_paths(input_path=audio_filepath, base_path=input_audio_dir)
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if not output_format:
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output_format = audio_path.suffix
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output_path = output_audio_dir / audio_path_rel
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output_path = output_path.with_suffix(output_format)
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output_path.parent.mkdir(exist_ok=True, parents=True)
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if output_path.exists() and not overwrite_audio:
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original_duration = librosa.get_duration(path=audio_path)
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output_duration = librosa.get_duration(path=output_path)
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else:
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audio, sample_rate = librosa.load(audio_path, sr=None)
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original_duration = librosa.get_duration(y=audio, sr=sample_rate)
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if audio_trimmer is not None:
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audio, start_i, end_i = audio_trimmer.trim_audio(
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audio=audio, sample_rate=int(sample_rate), audio_id=str(audio_path)
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)
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if output_sample_rate:
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audio = librosa.resample(y=audio, orig_sr=sample_rate, target_sr=output_sample_rate)
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sample_rate = output_sample_rate
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if volume_level:
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audio = normalize_volume(audio, volume_level=volume_level)
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if audio.size > 0:
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sf.write(file=output_path, data=audio, samplerate=sample_rate)
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output_duration = librosa.get_duration(y=audio, sr=sample_rate)
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else:
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output_duration = 0.0
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entry["duration"] = round(output_duration, 2)
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if os.path.isabs(audio_filepath):
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entry["audio_filepath"] = str(output_path)
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else:
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output_filepath = audio_path_rel.with_suffix(output_format)
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entry["audio_filepath"] = str(output_filepath)
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return entry, original_duration, output_duration
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def main():
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args = get_args()
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input_manifest_path = args.input_manifest
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output_manifest_path = args.output_manifest
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input_audio_dir = args.input_audio_dir
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output_audio_dir = args.output_audio_dir
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overwrite_audio = args.overwrite_audio
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overwrite_manifest = args.overwrite_manifest
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num_workers = args.num_workers
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max_entries = args.max_entries
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output_sample_rate = args.output_sample_rate
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output_format = args.output_format
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volume_level = args.volume_level
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min_duration = args.min_duration
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max_duration = args.max_duration
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filter_file = args.filter_file
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if output_manifest_path.exists():
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if overwrite_manifest:
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print(f"Will overwrite existing manifest path: {output_manifest_path}")
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else:
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raise ValueError(f"Manifest path already exists: {output_manifest_path}")
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if args.trim_config_path:
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audio_trimmer_config = OmegaConf.load(args.trim_config_path)
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audio_trimmer = safe_instantiate(audio_trimmer_config)
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else:
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audio_trimmer = None
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if output_format:
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if output_format.upper() not in sf.available_formats():
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raise ValueError(f"Unsupported output audio format: {output_format}")
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output_format = f".{output_format}"
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output_audio_dir.mkdir(exist_ok=True, parents=True)
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entries = read_manifest(input_manifest_path)
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if max_entries:
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entries = entries[:max_entries]
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# 'threading' backend is required when parallelizing torch models.
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job_outputs = Parallel(n_jobs=num_workers, backend='threading')(
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delayed(_process_entry)(
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entry=entry,
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input_audio_dir=input_audio_dir,
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output_audio_dir=output_audio_dir,
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overwrite_audio=overwrite_audio,
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audio_trimmer=audio_trimmer,
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output_sample_rate=output_sample_rate,
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output_format=output_format,
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volume_level=volume_level,
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)
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for entry in tqdm(entries)
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)
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output_entries = []
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filtered_entries = []
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original_durations = 0.0
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output_durations = 0.0
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for output_entry, original_duration, output_duration in job_outputs:
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original_durations += original_duration
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if (
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output_duration == 0.0
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or (min_duration and output_duration < min_duration)
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or (max_duration and output_duration > max_duration)
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):
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if output_duration != original_duration:
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output_entry["original_duration"] = original_duration
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filtered_entries.append(output_entry)
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continue
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output_durations += output_duration
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output_entries.append(output_entry)
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write_manifest(output_path=output_manifest_path, target_manifest=output_entries, ensure_ascii=False)
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if filter_file:
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write_manifest(output_path=str(filter_file), target_manifest=filtered_entries, ensure_ascii=False)
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logging.info(f"Duration of original audio: {original_durations / 3600:.2f} hours")
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logging.info(f"Duration of processed audio: {output_durations / 3600:.2f} hours")
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if __name__ == "__main__":
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main()
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