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PaddleSpeech/parakeet/exps/voice_cloning/tacotron2_ge2e/voice_cloning.py

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# Copyright (c) 2021 PaddlePaddle Authors. 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 argparse
import os
from pathlib import Path
import numpy as np
import paddle
import soundfile as sf
from matplotlib import pyplot as plt
from parakeet.exps.ge2e.audio_processor import SpeakerVerificationPreprocessor
from parakeet.exps.voice_cloning.tacotron2_ge2e.aishell3 import voc_phones
from parakeet.exps.voice_cloning.tacotron2_ge2e.aishell3 import voc_tones
from parakeet.exps.voice_cloning.tacotron2_ge2e.chinese_g2p import convert_sentence
from parakeet.models.lstm_speaker_encoder import LSTMSpeakerEncoder
from parakeet.models.tacotron2 import Tacotron2
from parakeet.models.waveflow import ConditionalWaveFlow
from parakeet.utils import display
def voice_cloning(args):
# speaker encoder
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!")
synthesizer = Tacotron2(
vocab_size=68,
n_tones=10,
d_mels=80,
d_encoder=512,
encoder_conv_layers=3,
encoder_kernel_size=5,
d_prenet=256,
d_attention_rnn=1024,
d_decoder_rnn=1024,
attention_filters=32,
attention_kernel_size=31,
d_attention=128,
d_postnet=512,
postnet_kernel_size=5,
postnet_conv_layers=5,
reduction_factor=1,
p_encoder_dropout=0.5,
p_prenet_dropout=0.5,
p_attention_dropout=0.1,
p_decoder_dropout=0.1,
p_postnet_dropout=0.5,
d_global_condition=256,
use_stop_token=False, )
synthesizer.set_state_dict(paddle.load(args.tacotron2_params_path))
synthesizer.eval()
print("Tacotron2 Done!")
# vocoder
vocoder = ConditionalWaveFlow(
upsample_factors=[16, 16],
n_flows=8,
n_layers=8,
n_group=16,
channels=128,
n_mels=80,
kernel_size=[3, 3])
vocoder.set_state_dict(paddle.load(args.waveflow_params_path))
vocoder.eval()
print("WaveFlow Done!")
output_dir = Path(args.output_dir)
output_dir.mkdir(parents=True, exist_ok=True)
input_dir = Path(args.input_dir)
# 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。
# 值得的注意的是,句子的有效字符集仅包含汉字和 %, $, 因此输入的句子只能包含这些字符。
sentence = "每当你觉得%想要批评什么人的时候$你切要记着%这个世界上的人%并非都具备你禀有的条件$"
phones, tones = convert_sentence(sentence)
phones = np.array(
[voc_phones.lookup(item) for item in phones], dtype=np.int64)
tones = np.array([voc_tones.lookup(item) for item in tones], dtype=np.int64)
phones = paddle.to_tensor(phones).unsqueeze(0)
tones = paddle.to_tensor(tones).unsqueeze(0)
for name in os.listdir(input_dir):
utt_id = name.split(".")[0]
ref_audio_path = input_dir / name
mel_sequences = p.extract_mel_partials(p.preprocess_wav(ref_audio_path))
print("mel_sequences: ", mel_sequences.shape)
with paddle.no_grad():
embed = speaker_encoder.embed_utterance(
paddle.to_tensor(mel_sequences))
print("embed shape: ", embed.shape)
utterance_embeds = paddle.unsqueeze(embed, 0)
outputs = synthesizer.infer(
phones, tones=tones, global_condition=utterance_embeds)
mel_input = paddle.transpose(outputs["mel_outputs_postnet"], [0, 2, 1])
alignment = outputs["alignments"][0].numpy().T
display.plot_alignment(alignment)
plt.savefig(str(output_dir / (utt_id + ".png")))
with paddle.no_grad():
wav = vocoder.infer(mel_input)
wav = wav.numpy()[0]
sf.write(str(output_dir / (utt_id + ".wav")), wav, samplerate=22050)
def main():
# parse args and config and redirect to train_sp
parser = argparse.ArgumentParser(description="")
parser.add_argument(
"--ge2e_params_path", type=str, help="ge2e params path.")
parser.add_argument(
"--tacotron2_params_path", type=str, help="tacotron2 params path.")
parser.add_argument(
"--waveflow_params_path", type=str, help="waveflow params path.")
parser.add_argument(
"--device", type=str, default="gpu", help="device type to use.")
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()
paddle.set_device(args.device)
voice_cloning(args)
if __name__ == "__main__":
main()