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chore: import upstream snapshot with attribution
2026-07-13 13:28:58 +08:00

214 行
8.3 KiB
Python

# Copyright (c) 2023, NVIDIA CORPORATION. All rights reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
import json
import os
from dataclasses import dataclass, is_dataclass
from typing import List, Optional, Union
import lightning.pytorch as pl
import torch
from omegaconf import OmegaConf
from nemo.collections.asr.modules.conformer_encoder import ConformerChangeConfig
from nemo.collections.asr.parts.utils.transcribe_utils import compute_output_filename, prepare_audio_data, setup_model
from nemo.core.config import hydra_runner
from nemo.utils import logging
"""
Translate audio file on a single CPU/GPU. Useful for translations of moderate amounts of audio data.
# Arguments
model_path: path to .nemo ST checkpoint
pretrained_name: name of pretrained ST model (from NGC registry)
audio_dir: path to directory with audio files
dataset_manifest: path to dataset JSON manifest file (in NeMo format)
output_filename: Output filename where the translations will be written
batch_size: batch size during inference
cuda: Optional int to enable or disable execution of model on certain CUDA device.
allow_mps: Bool to allow using MPS (Apple Silicon M-series GPU) device if available
amp: Bool to decide if Automatic Mixed Precision should be used during inference
audio_type: Str filetype of the audio. Supported = wav, flac, mp3
overwrite_translations: Bool which when set allows repeated translations to overwrite previous results.
# Usage
ST model can be specified by either "model_path" or "pretrained_name".
Data for translation can be defined with either "audio_dir" or "dataset_manifest".
Results are returned in a JSON manifest file.
python translate_speech.py \
model_path=null \
pretrained_name=null \
audio_dir="<remove or path to folder of audio files>" \
dataset_manifest="<remove or path to manifest>" \
output_filename="<remove or specify output filename>" \
batch_size=32 \
cuda=0 \
amp=True \
"""
@dataclass
class ModelChangeConfig:
"""
Sub-config for changes specific to the Conformer Encoder
"""
conformer: ConformerChangeConfig = ConformerChangeConfig()
@dataclass
class TranslationConfig:
"""
Translation Configuration for audio to text translation.
"""
# Required configs
model_path: Optional[str] = None # Path to a .nemo file
pretrained_name: Optional[str] = None # Name of a pretrained model
audio_dir: Optional[str] = None # Path to a directory which contains audio files
dataset_manifest: Optional[str] = None # Path to dataset's JSON manifest
audio_key: str = 'audio_filepath' # Used to override the default audio key in dataset_manifest
eval_config_yaml: Optional[str] = None # Path to a yaml file of config of evaluation
# General configs
output_filename: Optional[str] = None
batch_size: int = 32
random_seed: Optional[int] = None # seed number going to be used in seed_everything()
# Set `cuda` to int to define CUDA device. If 'None', will look for CUDA
# device anyway, and do inference on CPU only if CUDA device is not found.
# If `cuda` is a negative number, inference will be on CPU only.
cuda: Optional[int] = None
allow_mps: bool = False # allow to select MPS device (Apple Silicon M-series GPU)
amp: bool = False
audio_type: str = "wav"
# Recompute model translation, even if the output folder exists with scores.
overwrite_translations: bool = True
# can be set to True to return list of translations instead of the config
# if True, will also skip writing anything to the output file
return_translations: bool = False
presort_manifest: bool = False # sort manifest by duration before inference
pred_name_postfix: str = "translation" # postfix to add to the audio filename for the output
@hydra_runner(config_name="TranslationConfig", schema=TranslationConfig)
def main(cfg: TranslationConfig) -> Union[TranslationConfig, List[str]]:
"""
Main function to translate audio to text using a pretrained/finetuned model.
"""
logging.info(f'Hydra config: {OmegaConf.to_yaml(cfg)}')
for key in cfg:
cfg[key] = None if cfg[key] == 'None' else cfg[key]
if is_dataclass(cfg):
cfg = OmegaConf.structured(cfg)
if cfg.random_seed:
pl.seed_everything(cfg.random_seed)
if cfg.model_path is None and cfg.pretrained_name is None:
raise ValueError("Both cfg.model_path and cfg.pretrained_name cannot be None!")
if cfg.audio_dir is None and cfg.dataset_manifest is None:
raise ValueError("Both cfg.audio_dir and cfg.dataset_manifest cannot be None!")
# Load augmentor from exteranl yaml file which contains eval info, could be extend to other feature such VAD, P&C
augmentor = None
if cfg.eval_config_yaml:
eval_config = OmegaConf.load(cfg.eval_config_yaml)
augmentor = eval_config.test_ds.get("augmentor")
logging.info(f"Will apply on-the-fly augmentation on samples during translation: {augmentor} ")
# setup GPU
if cfg.cuda is None:
if torch.cuda.is_available():
device = [0] # use 0th CUDA device
accelerator = 'gpu'
map_location = torch.device('cuda:0')
elif cfg.allow_mps and hasattr(torch.backends, "mps") and torch.backends.mps.is_available():
logging.warning(
"MPS device (Apple Silicon M-series GPU) support is experimental."
" Env variable `PYTORCH_ENABLE_MPS_FALLBACK=1` should be set in most cases to avoid failures."
)
device = [0]
accelerator = 'mps'
map_location = torch.device('mps')
else:
device = 1
accelerator = 'cpu'
map_location = torch.device('cpu')
else:
device = [cfg.cuda]
accelerator = 'gpu'
map_location = torch.device(f'cuda:{cfg.cuda}')
logging.info(f"Inference will be done on device: {map_location}")
asr_model, model_name = setup_model(cfg, map_location)
trainer = pl.Trainer(devices=device, accelerator=accelerator)
asr_model.set_trainer(trainer)
asr_model = asr_model.eval()
# collect additional translation information
return_hypotheses = False
# prepare audio filepaths and decide wether it's partial audio
filepaths, partial_audio = prepare_audio_data(cfg)
# Compute output filename
cfg = compute_output_filename(cfg, model_name)
# if translations should not be overwritten, and already exists, skip re-translation step and return
if not cfg.return_translations and not cfg.overwrite_translations and os.path.exists(cfg.output_filename):
logging.info(
f"Previous translations found at {cfg.output_filename}, and flag `overwrite_translations`"
f"is {cfg.overwrite_translations}. Returning without re-translating text."
)
return cfg
# translate audio
with torch.amp.autocast(asr_model.device.type, enabled=cfg.amp):
with torch.no_grad():
translations = asr_model.transcribe(
audio=filepaths,
batch_size=cfg.batch_size,
return_hypotheses=return_hypotheses,
)
logging.info(f"Finished translating {len(filepaths)} files !")
logging.info(f"Writing translations into file: {cfg.output_filename}")
if cfg.return_translations:
return translations
# write audio translations
with open(cfg.output_filename, 'w', encoding='utf-8', newline='\n') as f:
for filepath, translation in zip(filepaths, translations):
item = {'audio_filepath': filepath, 'pred_translation': translation}
f.write(json.dumps(item, ensure_ascii=False) + "\n")
logging.info(f"Finished writing predictions to {cfg.output_filename}!")
return cfg
if __name__ == '__main__':
main() # noqa pylint: disable=no-value-for-parameter