Source code for paddlespeech.t2s.exps.voice_cloning

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# you may not use this file except in compliance with the License.
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#     http://www.apache.org/licenses/LICENSE-2.0
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import argparse
import os
from pathlib import Path

import numpy as np
import paddle
import soundfile as sf
import yaml
from yacs.config import CfgNode

from paddlespeech.cli.vector import VectorExecutor
from paddlespeech.t2s.exps.syn_utils import get_am_inference
from paddlespeech.t2s.exps.syn_utils import get_voc_inference
from paddlespeech.t2s.frontend.zh_frontend import Frontend
from paddlespeech.t2s.utils import str2bool
from paddlespeech.vector.exps.ge2e.audio_processor import SpeakerVerificationPreprocessor
from paddlespeech.vector.models.lstm_speaker_encoder import LSTMSpeakerEncoder


[docs]def gen_random_embed(use_ecapa: bool=False): if use_ecapa: # Randomly generate numbers of -25 ~ 25, 192 is the dim of spk_emb random_spk_emb = (-1 + 2 * np.random.rand(192)) * 25 # GE2E else: # Randomly generate numbers of 0 ~ 0.2, 256 is the dim of spk_emb random_spk_emb = np.random.rand(256) * 0.2 random_spk_emb = paddle.to_tensor(random_spk_emb, dtype='float32') return random_spk_emb
[docs]def voice_cloning(args): # Init body. with open(args.am_config) as f: am_config = CfgNode(yaml.safe_load(f)) with open(args.voc_config) as f: voc_config = CfgNode(yaml.safe_load(f)) print("========Args========") print(yaml.safe_dump(vars(args))) print("========Config========") print(am_config) print(voc_config) output_dir = Path(args.output_dir) output_dir.mkdir(parents=True, exist_ok=True) input_dir = Path(args.input_dir) # speaker encoder if args.use_ecapa: vec_executor = VectorExecutor() # warm up vec_executor( audio_file=input_dir / os.listdir(input_dir)[0], force_yes=True) print("ECAPA-TDNN Done!") # use GE2E else: p = SpeakerVerificationPreprocessor( sampling_rate=16000, audio_norm_target_dBFS=-30, vad_window_length=30, vad_moving_average_width=8, vad_max_silence_length=6, mel_window_length=25, mel_window_step=10, n_mels=40, partial_n_frames=160, min_pad_coverage=0.75, partial_overlap_ratio=0.5) print("Audio Processor Done!") speaker_encoder = LSTMSpeakerEncoder( n_mels=40, num_layers=3, hidden_size=256, output_size=256) speaker_encoder.set_state_dict(paddle.load(args.ge2e_params_path)) speaker_encoder.eval() print("GE2E Done!") frontend = Frontend(phone_vocab_path=args.phones_dict) print("frontend done!") sentence = args.text input_ids = frontend.get_input_ids(sentence, merge_sentences=True) phone_ids = input_ids["phone_ids"][0] # acoustic model am_inference = get_am_inference( am=args.am, am_config=am_config, am_ckpt=args.am_ckpt, am_stat=args.am_stat, phones_dict=args.phones_dict) # vocoder voc_inference = get_voc_inference( voc=args.voc, voc_config=voc_config, voc_ckpt=args.voc_ckpt, voc_stat=args.voc_stat) for name in os.listdir(input_dir): utt_id = name.split(".")[0] ref_audio_path = input_dir / name if args.use_ecapa: spk_emb = vec_executor(audio_file=ref_audio_path, force_yes=True) spk_emb = paddle.to_tensor(spk_emb) # GE2E else: mel_sequences = p.extract_mel_partials( p.preprocess_wav(ref_audio_path)) with paddle.no_grad(): spk_emb = speaker_encoder.embed_utterance( paddle.to_tensor(mel_sequences)) with paddle.no_grad(): wav = voc_inference(am_inference(phone_ids, spk_emb=spk_emb)) sf.write( str(output_dir / (utt_id + ".wav")), wav.numpy(), samplerate=am_config.fs) print(f"{utt_id} done!")
# generate 5 random_spk_emb # for i in range(5): # random_spk_emb = gen_random_embed(args.use_ecapa) # utt_id = "random_spk_emb" # with paddle.no_grad(): # wav = voc_inference(am_inference(phone_ids, spk_emb=random_spk_emb)) # sf.write( # str(output_dir / (utt_id + "_" + str(i) + ".wav")), # wav.numpy(), # samplerate=am_config.fs) # print(f"{utt_id} done!")
[docs]def parse_args(): # parse args and config parser = argparse.ArgumentParser(description="") parser.add_argument( '--am', type=str, default='fastspeech2_csmsc', choices=['fastspeech2_aishell3', 'tacotron2_aishell3'], help='Choose acoustic model type of tts task.') parser.add_argument( '--am_config', type=str, default=None, help='Config of acoustic model.') parser.add_argument( '--am_ckpt', type=str, default=None, help='Checkpoint file of acoustic model.') parser.add_argument( "--am_stat", type=str, default=None, help="mean and standard deviation used to normalize spectrogram when training acoustic model." ) parser.add_argument( "--phones-dict", type=str, default="phone_id_map.txt", help="phone vocabulary file.") # vocoder parser.add_argument( '--voc', type=str, default='pwgan_csmsc', choices=['pwgan_aishell3'], help='Choose vocoder type of tts task.') parser.add_argument( '--voc_config', type=str, default=None, help='Config of voc.') parser.add_argument( '--voc_ckpt', type=str, default=None, help='Checkpoint file of voc.') parser.add_argument( "--voc_stat", type=str, default=None, help="mean and standard deviation used to normalize spectrogram when training voc." ) parser.add_argument( "--text", type=str, default="每当你觉得,想要批评什么人的时候,你切要记着,这个世界上的人,并非都具备你禀有的条件。", help="text to synthesize, a line") parser.add_argument( "--ge2e_params_path", type=str, help="ge2e params path.") parser.add_argument( "--use_ecapa", type=str2bool, default=False, help="whether to use ECAPA-TDNN as speaker encoder.") parser.add_argument( "--ngpu", type=int, default=1, help="if ngpu=0, use cpu.") parser.add_argument( "--input-dir", type=str, help="input dir of *.wav, the sample rate will be resample to 16k.") parser.add_argument("--output-dir", type=str, help="output dir.") args = parser.parse_args() return args
[docs]def main(): args = parse_args() if args.ngpu == 0: paddle.set_device("cpu") elif args.ngpu > 0: paddle.set_device("gpu") else: print("ngpu should >= 0 !") voice_cloning(args)
if __name__ == "__main__": main()