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167 lines
5.9 KiB
167 lines
5.9 KiB
# Copyright (c) 2021 PaddlePaddle Authors. 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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import argparse
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import os
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from pathlib import Path
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import numpy as np
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import paddle
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import soundfile as sf
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from matplotlib import pyplot as plt
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from paddlespeech.t2s.exps.voice_cloning.tacotron2_ge2e.aishell3 import voc_phones
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from paddlespeech.t2s.exps.voice_cloning.tacotron2_ge2e.aishell3 import voc_tones
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from paddlespeech.t2s.exps.voice_cloning.tacotron2_ge2e.chinese_g2p import convert_sentence
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from paddlespeech.t2s.models.tacotron2 import Tacotron2
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from paddlespeech.t2s.models.waveflow import ConditionalWaveFlow
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from paddlespeech.t2s.utils import display
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from paddlespeech.vector.exps.ge2e.audio_processor import SpeakerVerificationPreprocessor
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from paddlespeech.vector.models.lstm_speaker_encoder import LSTMSpeakerEncoder
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def voice_cloning(args):
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# speaker encoder
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p = SpeakerVerificationPreprocessor(
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sampling_rate=16000,
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audio_norm_target_dBFS=-30,
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vad_window_length=30,
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vad_moving_average_width=8,
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vad_max_silence_length=6,
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mel_window_length=25,
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mel_window_step=10,
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n_mels=40,
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partial_n_frames=160,
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min_pad_coverage=0.75,
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partial_overlap_ratio=0.5)
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print("Audio Processor Done!")
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speaker_encoder = LSTMSpeakerEncoder(
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n_mels=40, num_layers=3, hidden_size=256, output_size=256)
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speaker_encoder.set_state_dict(paddle.load(args.ge2e_params_path))
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speaker_encoder.eval()
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print("GE2E Done!")
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synthesizer = Tacotron2(
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vocab_size=68,
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n_tones=10,
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d_mels=80,
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d_encoder=512,
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encoder_conv_layers=3,
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encoder_kernel_size=5,
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d_prenet=256,
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d_attention_rnn=1024,
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d_decoder_rnn=1024,
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attention_filters=32,
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attention_kernel_size=31,
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d_attention=128,
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d_postnet=512,
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postnet_kernel_size=5,
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postnet_conv_layers=5,
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reduction_factor=1,
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p_encoder_dropout=0.5,
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p_prenet_dropout=0.5,
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p_attention_dropout=0.1,
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p_decoder_dropout=0.1,
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p_postnet_dropout=0.5,
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d_global_condition=256,
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use_stop_token=False, )
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synthesizer.set_state_dict(paddle.load(args.tacotron2_params_path))
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synthesizer.eval()
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print("Tacotron2 Done!")
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# vocoder
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vocoder = ConditionalWaveFlow(
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upsample_factors=[16, 16],
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n_flows=8,
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n_layers=8,
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n_group=16,
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channels=128,
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n_mels=80,
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kernel_size=[3, 3])
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vocoder.set_state_dict(paddle.load(args.waveflow_params_path))
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vocoder.eval()
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print("WaveFlow Done!")
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output_dir = Path(args.output_dir)
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output_dir.mkdir(parents=True, exist_ok=True)
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input_dir = Path(args.input_dir)
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# 因为 AISHELL-3 数据集中使用 % 和 $ 表示韵律词和韵律短语的边界,它们大约对应着较短和较长的停顿,在文本中可以使用 % 和 $ 来调节韵律。
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# 值得的注意的是,句子的有效字符集仅包含汉字和 %, $, 因此输入的句子只能包含这些字符。
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sentence = "每当你觉得%想要批评什么人的时候$你切要记着%这个世界上的人%并非都具备你禀有的条件$"
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phones, tones = convert_sentence(sentence)
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phones = np.array(
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[voc_phones.lookup(item) for item in phones], dtype=np.int64)
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tones = np.array([voc_tones.lookup(item) for item in tones], dtype=np.int64)
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phones = paddle.to_tensor(phones).unsqueeze(0)
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tones = paddle.to_tensor(tones).unsqueeze(0)
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for name in os.listdir(input_dir):
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utt_id = name.split(".")[0]
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ref_audio_path = input_dir / name
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mel_sequences = p.extract_mel_partials(p.preprocess_wav(ref_audio_path))
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print("mel_sequences: ", mel_sequences.shape)
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with paddle.no_grad():
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embed = speaker_encoder.embed_utterance(
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paddle.to_tensor(mel_sequences))
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print("embed shape: ", embed.shape)
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utterance_embeds = paddle.unsqueeze(embed, 0)
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outputs = synthesizer.infer(
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phones, tones=tones, global_condition=utterance_embeds)
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mel_input = paddle.transpose(outputs["mel_outputs_postnet"], [0, 2, 1])
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alignment = outputs["alignments"][0].numpy().T
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display.plot_alignment(alignment)
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plt.savefig(str(output_dir / (utt_id + ".png")))
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with paddle.no_grad():
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wav = vocoder.infer(mel_input)
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wav = wav.numpy()[0]
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sf.write(str(output_dir / (utt_id + ".wav")), wav, samplerate=22050)
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def main():
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# parse args and config and redirect to train_sp
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parser = argparse.ArgumentParser(description="")
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parser.add_argument(
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"--ge2e_params_path", type=str, help="ge2e params path.")
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parser.add_argument(
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"--tacotron2_params_path", type=str, help="tacotron2 params path.")
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parser.add_argument(
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"--waveflow_params_path", type=str, help="waveflow params path.")
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parser.add_argument(
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"--ngpu", type=int, default=1, help="if ngpu=0, use cpu.")
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parser.add_argument(
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"--input-dir",
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type=str,
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help="input dir of *.wav, the sample rate will be resample to 16k.")
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parser.add_argument("--output-dir", type=str, help="output dir.")
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args = parser.parse_args()
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if args.ngpu == 0:
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paddle.set_device("cpu")
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elif args.ngpu > 0:
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paddle.set_device("gpu")
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else:
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print("ngpu should >= 0 !")
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voice_cloning(args)
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if __name__ == "__main__":
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main()
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