import subprocess import random import os from pathlib import Path import librosa from scipy.io import wavfile import numpy as np import torch import csv import whisper import gradio as gr import soundfile as sf os.system("pip install --upgrade Cython==0.29.35") os.system("pip install pysptk --no-build-isolation") os.system("pip install kantts -f https://modelscope.oss-cn-beijing.aliyuncs.com/releases/repo.html") os.system("pip install tts-autolabel -f https://modelscope.oss-cn-beijing.aliyuncs.com/releases/repo.html") import sox def split_long_audio(model, filepaths, save_dir="data_dir", out_sr=44100): if isinstance(filepaths, str): filepaths = [filepaths] for file_idx, filepath in enumerate(filepaths): save_path = Path(save_dir) save_path.mkdir(exist_ok=True, parents=True) print(f"Transcribing file {file_idx}: '{filepath}' to segments...") result = model.transcribe(filepath, word_timestamps=True, task="transcribe", beam_size=5, best_of=5) segments = result['segments'] wav, sr = librosa.load(filepath, sr=None, offset=0, duration=None, mono=True) wav, _ = librosa.effects.trim(wav, top_db=20) peak = np.abs(wav).max() if peak > 1.0: wav = 0.98 * wav / peak wav2 = librosa.resample(wav, orig_sr=sr, target_sr=out_sr) wav2 /= max(wav2.max(), -wav2.min()) for i, seg in enumerate(segments): start_time = seg['start'] end_time = seg['end'] wav_seg = wav2[int(start_time * out_sr):int(end_time * out_sr)] wav_seg_name = f"{file_idx}_{i}.wav" out_fpath = save_path / wav_seg_name wavfile.write(out_fpath, rate=out_sr, data=(wav_seg * np.iinfo(np.int16).max).astype(np.int16)) device = 'cuda' if torch.cuda.is_available() else 'cpu' whisper_size = "medium" whisper_model = whisper.load_model(whisper_size).to(device) from modelscope.tools import run_auto_label from modelscope.models.audio.tts import SambertHifigan from modelscope.pipelines import pipeline from modelscope.utils.constant import Tasks from modelscope.metainfo import Trainers from modelscope.trainers import build_trainer from modelscope.utils.audio.audio_utils import TtsTrainType pretrained_model_id = 'damo/speech_personal_sambert-hifigan_nsf_tts_zh-cn_pretrain_16k' dataset_id = "/home/user/app/output_training_data/" pretrain_work_dir = "/home/user/app/pretrain_work_dir/" def auto_label(Voicetoclone, VoiceMicrophone): if VoiceMicrophone is not None: audio = VoiceMicrophone else: audio = Voicetoclone try: split_long_audio(whisper_model, audio, "/home/user/app/test_wavs/") input_wav = "/home/user/app/test_wavs/" output_data = "/home/user/app/output_training_data/" ret, report = run_auto_label(input_wav=input_wav, work_dir=output_data, resource_revision="v1.0.7") except Exception as e: print(e) return "标注成功" def train(train_step): try: train_info = { TtsTrainType.TRAIN_TYPE_SAMBERT: { # 配置训练AM(sambert)模型 'train_steps': int(train_step / 20) * 20 + 2, # 训练多少个step 'save_interval_steps': int(train_step / 20) * 20, # 每训练多少个step保存一次checkpoint 'log_interval': int(train_step / 20) * 20 # 每训练多少个step打印一次训练日志 } } kwargs = dict( model=pretrained_model_id, # 指定要finetune的模型 model_revision = "v1.0.6", work_dir=pretrain_work_dir, # 指定临时工作目录 train_dataset=dataset_id, # 指定数据集id train_type=train_info # 指定要训练类型及参数 ) trainer = build_trainer(Trainers.speech_kantts_trainer, default_args=kwargs) trainer.train() except Exception as e: print(e) return "训练完成" # 保存模型 import shutil import datetime def save_model(worked_dir,dest_dir): worked_dir = "/home/user/app/pretrain_work_dir" dest_dir = "/home/user/app/trained_model" if os.listdir(worked_dir): now = datetime.datetime.now() date_str = now.strftime("%Y%m%d%H%M%S") dest_folder = os.path.join(dest_dir, date_str) shutil.copytree(worked_dir, dest_folder) # List of files and directories to delete files_to_delete = [ "tmp_voc", "tmp_am/ckpt/checkpoint_2400000.pth", "orig_model/description", "orig_model/.mdl", "orig_model/.msc", "orig_model/README.md", "orig_model/resource", "orig_model/description", "orig_model/basemodel_16k/sambert", "orig_model/basemodel_16k/speaker_embedding", "data/duration", "data/energy", "data/f0", "data/frame_energy", "data/frame_f0", "data/frame_uv", "data/mel", "data/raw_duration", "data/wav", "data/am_train.lst", "data/am_valid.lst", "data/badlist.txt", "data/raw_metafile.txt", "data/Script.xml", "data/train.lst", "data/valid.lst", "data/se/0_*" ] for item in files_to_delete: item_path = os.path.join(dest_folder, item) if os.path.exists(item_path): if os.path.isdir(item_path): shutil.rmtree(item_path) else: os.remove(item_path) shutil.rmtree("/home/user/app/output_training_data") shutil.rmtree("/home/user/app/pretrain_work_dir") shutil.rmtree("/home/user/app/test_wavs") os.mkdir("/home/user/app/output_training_data") os.mkdir("/home/user/app/pretrain_work_dir") os.mkdir("/home/user/app/test_wavs") return f"模型已成功保存为 {date_str}" else: return "保存失败,模型已保存或已被清除" import random def infer(text): model_dir = "/home/user/app/pretrain_work_dir/" test_infer_abs = { 'voice_name': 'F7', 'am_ckpt': os.path.join(model_dir, 'tmp_am', 'ckpt'), 'am_config': os.path.join(model_dir, 'tmp_am', 'config.yaml'), 'voc_ckpt': os.path.join(model_dir, 'orig_model', 'basemodel_16k', 'hifigan', 'ckpt'), 'voc_config': os.path.join(model_dir, 'orig_model', 'basemodel_16k', 'hifigan', 'config.yaml'), 'audio_config': os.path.join(model_dir, 'data', 'audio_config.yaml'), 'se_file': os.path.join(model_dir, 'data', 'se', 'se.npy') } kwargs = {'custom_ckpt': test_infer_abs} model_id = SambertHifigan(os.path.join(model_dir, "orig_model"), **kwargs) inference = pipeline(task=Tasks.text_to_speech, model=model_id) output = inference(input=text) now = datetime.datetime.now() date_str = now.strftime("%Y%m%d%H%M%S") rand_num = random.randint(1000, 9999) filename = date_str + str(rand_num) with open(filename + "0.wav", mode='bx') as f: f.write(output["output_wav"]) y, sr = librosa.load(filename + "0.wav") S = librosa.stft(y) noise = S[np.abs(S) < np.percentile(S, 95)] noise_mean, noise_std = np.mean(noise), np.std(noise) filter_ = np.ones_like(S) filter_[np.abs(S) < noise_mean + 2 * noise_std] = 0 filtered_S = filter_ * S filtered_y = librosa.istft(filtered_S) sf.write(filename + "testfile.wav", filtered_y, sr) os.remove(filename + "0.wav") return filename + "testfile.wav" def infer_custom(model_name, text, noise_level): custom_model_dir = os.path.join("/home/user/app/trained_model/", model_name) custom_infer_abs = { 'voice_name': 'F7', 'am_ckpt': os.path.join(custom_model_dir, 'tmp_am', 'ckpt'), 'am_config': os.path.join(custom_model_dir, 'tmp_am', 'config.yaml'), 'voc_ckpt': os.path.join(custom_model_dir, 'orig_model', 'basemodel_16k', 'hifigan', 'ckpt'), 'voc_config': os.path.join(custom_model_dir, 'orig_model', 'basemodel_16k', 'hifigan', 'config.yaml'), 'audio_config': os.path.join(custom_model_dir, 'data', 'audio_config.yaml'), 'se_file': os.path.join(custom_model_dir, 'data', 'se', 'se.npy') } kwargs = {'custom_ckpt': custom_infer_abs} model_id = SambertHifigan(os.path.join(custom_model_dir, "orig_model"), **kwargs) inference = pipeline(task=Tasks.text_to_speech, model=model_id) output = inference(input=text) now = datetime.datetime.now() date_str = now.strftime("%Y%m%d%H%M%S") rand_num = random.randint(1000, 9999) filename = date_str + str(rand_num) with open(filename + ".wav", mode='bx') as f: f.write(output["output_wav"]) y, sr = librosa.load(filename + ".wav") S = librosa.stft(y) noise = S[np.abs(S) < np.percentile(S, 95)] noise_mean, noise_std = np.mean(noise), np.std(noise) filter_ = np.ones_like(S) filter_[np.abs(S) < noise_mean + noise_level * noise_std] = 0 filtered_S = filter_ * S filtered_y = librosa.istft(filtered_S) sf.write(filename + "customfile.wav", filtered_y, sr) os.remove(filename + ".wav") return filename + "customfile.wav" trained_model = "/home/user/app/trained_model/" def update_model_dropdown(inp3): model_list = os.listdir(trained_model) return gr.Dropdown(choices=model_list, value=inp3) def rename_model(old_name, new_name): if not os.path.isdir(os.path.join(trained_model, old_name)): return "模型名称不存在,请重新输入!" else: try: os.rename(os.path.join(trained_model, old_name), os.path.join(trained_model, new_name)) return "模型重命名成功!" except OSError: return "新名称已经存在,请重新输入!" # 清除训练缓存 def clear_cache(a): shutil.rmtree("/home/user/app/output_training_data") shutil.rmtree("/home/user/app/pretrain_work_dir") shutil.rmtree("/home/user/app/test_wavs") os.mkdir("/home/user/app/output_training_data") os.mkdir("/home/user/app/pretrain_work_dir") os.mkdir("/home/user/app/test_wavs") return "已清除缓存,请返回训练页面重新训练" from textwrap import dedent def FRCRN_De_Noise(noise_wav, noisemic_wav): if noisemic_wav is not None: noise_audio = noisemic_wav else: noise_audio = noise_wav ans = pipeline( Tasks.acoustic_noise_suppression, model='damo/speech_frcrn_ans_cirm_16k') now = datetime.datetime.now() date_str = now.strftime("%Y%m%d%H%M%S") rand_num = random.randint(1000, 9999) filename = date_str + str(rand_num) result = ans( noise_audio, output_path= filename + "AIdenoise.wav" ) return filename + "AIdenoise.wav" def Normal_De_Noise(noise_wav, noisemic_wav, noise_level): if noisemic_wav is not None: noise_audio = noisemic_wav else: noise_audio = noise_wav now = datetime.datetime.now() date_str = now.strftime("%Y%m%d%H%M%S") rand_num = random.randint(1000, 9999) filename = date_str + str(rand_num) y, sr = librosa.load(noise_audio) S = librosa.stft(y) noise = S[np.abs(S) < np.percentile(S, 95)] noise_mean, noise_std = np.mean(noise), np.std(noise) filter_ = np.ones_like(S) filter_[np.abs(S) < noise_mean + noise_level * noise_std] = 0 filtered_S = filter_ * S filtered_y = librosa.istft(filtered_S) sf.write(filename + "denoise.wav", filtered_y, sr) return filename + "denoise.wav" app = gr.Blocks() with app: gr.Markdown("#