text/init.py (#203)
* Create all_process.py * Create asr_transcript.py * Update config.py * Create extract_list.py * Create clean_list.py * Create custom.css * Create compress_model.py * Update all_process.py * Update resample.py * Update resample.py * configs/config.json copy utils * mirror: openi + token bert models optimize * text/__init__.py platform compatibility compress_model.py output * [pre-commit.ci] auto fixes from pre-commit.com hooks for more information, see https://pre-commit.ci --------- Co-authored-by: pre-commit-ci[bot] <66853113+pre-commit-ci[bot]@users.noreply.github.com>
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1378
all_process.py
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1378
all_process.py
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102
asr_transcript.py
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102
asr_transcript.py
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import argparse
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import concurrent.futures
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import os
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from loguru import logger
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from modelscope.pipelines import pipeline
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from modelscope.utils.constant import Tasks
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from tqdm import tqdm
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os.environ["MODELSCOPE_CACHE"] = "./"
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def transcribe_worker(file_path: str, inference_pipeline, language):
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"""
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Worker function for transcribing a segment of an audio file.
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"""
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rec_result = inference_pipeline(audio_in=file_path)
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text = str(rec_result.get("text", "")).strip()
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text_without_spaces = text.replace(" ", "")
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logger.info(file_path)
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if language != "EN":
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logger.info("text: " + text_without_spaces)
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return text_without_spaces
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else:
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logger.info("text: " + text)
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return text
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def transcribe_folder_parallel(folder_path, language, max_workers=4):
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"""
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Transcribe all .wav files in the given folder using ThreadPoolExecutor.
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"""
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logger.critical(f"parallel transcribe: {folder_path}|{language}|{max_workers}")
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if language == "JP":
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workers = [
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pipeline(
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task=Tasks.auto_speech_recognition,
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model="damo/speech_UniASR_asr_2pass-ja-16k-common-vocab93-tensorflow1-offline",
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)
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for _ in range(max_workers)
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]
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elif language == "ZH":
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workers = [
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pipeline(
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task=Tasks.auto_speech_recognition,
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model="damo/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
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model_revision="v1.2.4",
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)
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for _ in range(max_workers)
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]
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else:
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workers = [
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pipeline(
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task=Tasks.auto_speech_recognition,
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model="damo/speech_UniASR_asr_2pass-en-16k-common-vocab1080-tensorflow1-offline",
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)
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for _ in range(max_workers)
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]
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file_paths = []
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langs = []
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for root, _, files in os.walk(folder_path):
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for file in files:
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if file.lower().endswith(".wav"):
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file_path = os.path.join(root, file)
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lab_file_path = os.path.splitext(file_path)[0] + ".lab"
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file_paths.append(file_path)
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langs.append(language)
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all_workers = (
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workers * (len(file_paths) // max_workers)
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+ workers[: len(file_paths) % max_workers]
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)
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with concurrent.futures.ThreadPoolExecutor(max_workers=max_workers) as executor:
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for i in tqdm(range(0, len(file_paths), max_workers), desc="转写进度: "):
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l, r = i, min(i + max_workers, len(file_paths))
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transcriptions = list(
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executor.map(
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transcribe_worker, file_paths[l:r], all_workers[l:r], langs[l:r]
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)
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)
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for file_path, transcription in zip(file_paths[l:r], transcriptions):
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if transcription:
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lab_file_path = os.path.splitext(file_path)[0] + ".lab"
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with open(lab_file_path, "w", encoding="utf-8") as lab_file:
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lab_file.write(transcription)
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logger.critical("已经将wav文件转写为同名的.lab文件")
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if __name__ == "__main__":
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parser = argparse.ArgumentParser()
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parser.add_argument(
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"-f", "--filepath", default="./raw/lzy_zh", help="path of your model"
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)
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parser.add_argument("-l", "--language", default="ZH", help="language")
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parser.add_argument("-w", "--workers", default="1", help="trans workers")
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args = parser.parse_args()
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transcribe_folder_parallel(args.filepath, args.language, int(args.workers))
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print("转写结束!")
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48
clean_list.py
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48
clean_list.py
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import argparse
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import shutil
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from tempfile import NamedTemporaryFile
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from loguru import logger
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def remove_chars_from_file(chars_to_remove, input_file, output_file):
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rm_cnt = 0
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with open(input_file, "r", encoding="utf-8") as f_in, NamedTemporaryFile(
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"w", delete=False, encoding="utf-8"
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) as f_tmp:
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for line in f_in:
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if any(char in line for char in chars_to_remove):
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logger.info(f"删除了这一行:\n {line.strip()}")
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rm_cnt += 1
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else:
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f_tmp.write(line)
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shutil.move(f_tmp.name, output_file)
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logger.critical(f"总计移除了: {rm_cnt} 行")
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if __name__ == "__main__":
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parser = argparse.ArgumentParser(
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description="Remove lines from a file containing specified characters."
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)
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parser.add_argument(
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"-c",
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"--chars",
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type=str,
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required=True,
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help="String of characters. If a line contains any of these characters, it will be removed.",
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)
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parser.add_argument(
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"-i", "--input", type=str, required=True, help="Path to the input file."
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)
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parser.add_argument(
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"-o", "--output", type=str, required=True, help="Path to the output file."
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)
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args = parser.parse_args()
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# Setting up basic logging configuration for loguru
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logger.add("removed_lines.log", rotation="1 MB")
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remove_chars_from_file(args.chars, args.input, args.output)
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88
compress_model.py
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88
compress_model.py
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from collections import OrderedDict
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from text.symbols import symbols
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import torch
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from tools.log import logger
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import utils
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from models import SynthesizerTrn
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import os
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def copyStateDict(state_dict):
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if list(state_dict.keys())[0].startswith("module"):
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start_idx = 1
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else:
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start_idx = 0
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new_state_dict = OrderedDict()
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for k, v in state_dict.items():
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name = ",".join(k.split(".")[start_idx:])
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new_state_dict[name] = v
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return new_state_dict
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def removeOptimizer(config: str, input_model: str, ishalf: bool, output_model: str):
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hps = utils.get_hparams_from_file(config)
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net_g = SynthesizerTrn(
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len(symbols),
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hps.data.filter_length // 2 + 1,
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hps.train.segment_size // hps.data.hop_length,
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n_speakers=hps.data.n_speakers,
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**hps.model,
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)
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optim_g = torch.optim.AdamW(
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net_g.parameters(),
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hps.train.learning_rate,
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betas=hps.train.betas,
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eps=hps.train.eps,
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)
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state_dict_g = torch.load(input_model, map_location="cpu")
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new_dict_g = copyStateDict(state_dict_g)
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keys = []
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for k, v in new_dict_g["model"].items():
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if "enc_q" in k:
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continue # noqa: E701
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keys.append(k)
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new_dict_g = (
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{k: new_dict_g["model"][k].half() for k in keys}
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if ishalf
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else {k: new_dict_g["model"][k] for k in keys}
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)
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torch.save(
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{
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"model": new_dict_g,
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"iteration": 0,
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"optimizer": optim_g.state_dict(),
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"learning_rate": 0.0001,
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},
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output_model,
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)
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if __name__ == "__main__":
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import argparse
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parser = argparse.ArgumentParser()
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parser.add_argument("-c", "--config", type=str, default="configs/config.json")
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parser.add_argument("-i", "--input", type=str)
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parser.add_argument("-o", "--output", type=str, default=None)
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parser.add_argument(
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"-hf", "--half", action="store_true", default=False, help="Save as FP16"
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)
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args = parser.parse_args()
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output = args.output
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if output is None:
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import os.path
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filename, ext = os.path.splitext(args.input)
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half = "_half" if args.half else ""
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output = filename + "_release" + half + ext
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removeOptimizer(args.config, args.input, args.half, output)
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logger.info(f"压缩模型成功, 输出模型: {os.path.abspath(output)}")
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@@ -241,3 +241,4 @@ parser = argparse.ArgumentParser()
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parser.add_argument("-y", "--yml_config", type=str, default="config.yml")
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args, _ = parser.parse_known_args()
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config = Config(args.yml_config)
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yml_config = args.yml_config
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18
css/custom.css
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css/custom.css
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#yml_code {
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height: 600px;
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flex-grow: inherit;
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overflow-y: auto;
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}
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#json_code {
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height: 600px;
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flex-grow: inherit;
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overflow-y: auto;
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}
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#gpu_code {
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height: 300px;
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flex-grow: inherit;
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overflow-y: auto;
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}
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53
extract_list.py
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53
extract_list.py
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import argparse
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import os
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from loguru import logger
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def extract_list(folder_path, language, name, transcript_txt_file):
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logger.info(f"extracting list: {folder_path}|{name}|{language}")
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current_dir = os.getcwd()
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relative_path = os.path.relpath(folder_path, current_dir)
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print(relative_path)
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os.makedirs(os.path.dirname(transcript_txt_file), exist_ok=True)
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with open(transcript_txt_file, "w", encoding="utf-8") as f:
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# 遍历 raw 文件夹下的所有子文件夹
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for root, _, files in os.walk(relative_path):
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for file in files:
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if file.endswith(".lab"):
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lab_file_path = os.path.join(root, file)
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# 读取转写文本
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with open(lab_file_path, "r", encoding="utf-8") as lab_file:
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transcription = lab_file.read().strip()
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if len(transcription) == 0:
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continue
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# 获取对应的 WAV 文件路径
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# ./Data/宵宫/audios/raw
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# ./Data/宵宫/audios/wavs
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wav_file_path = os.path.splitext(lab_file_path)[0] + ".wav"
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if os.path.isfile(wav_file_path):
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wav_file_path = wav_file_path.replace("\\", "/").replace(
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"/raw", "/wavs"
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)
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# 写入数据到总的转写文本文件
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line = f"{wav_file_path}|{name}|{language}|{transcription}\n"
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f.write(line)
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else:
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print("not exists!")
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return f"转写文本 {transcript_txt_file} 生成完成"
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if __name__ == "__main__":
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parser = argparse.ArgumentParser()
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parser.add_argument(
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"-f",
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"--filepath",
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required=True,
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help="path of your rawaudios, e.g. ./Data/xxx/audios/raw",
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)
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parser.add_argument("-l", "--language", default="ZH", help="language")
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parser.add_argument("-n", "--name", required=True, help="name of the character")
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parser.add_argument("-o", "--outfile", required=True, help="outfile")
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args = parser.parse_args()
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status_str = extract_list(args.filepath, args.language, args.name, args.outfile)
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logger.critical(status_str)
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@@ -12,7 +12,7 @@ from config import config
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def process(item):
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spkdir, wav_name, args = item
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wav_path = os.path.join(args.in_dir, spkdir, wav_name)
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if os.path.exists(wav_path) and ".wav" in wav_path:
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if os.path.exists(wav_path) and wav_path.lower().endswith(".wav"):
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wav, sr = librosa.load(wav_path, sr=args.sr)
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soundfile.write(os.path.join(args.out_dir, spkdir, wav_name), wav, sr)
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@@ -60,7 +60,7 @@ if __name__ == "__main__":
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if not os.path.isdir(spk_dir_out):
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os.makedirs(spk_dir_out, exist_ok=True)
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for filename in filenames:
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if filename.endswith(".wav"):
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if filename.lower().endswith(".wav"):
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twople = (spk_dir, filename, args)
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tasks.append(twople)
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@@ -49,9 +49,12 @@ def check_bert_models():
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def init_openjtalk():
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import pyopenjtalk
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import platform
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pyopenjtalk.g2p("こんにちは,世界。")
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if platform.platform() == "Linux":
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import pyopenjtalk
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pyopenjtalk.g2p("こんにちは,世界。")
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init_openjtalk()
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