Add tts python example and change onnx to paddle (#420)

* add tts example

* update example

* update use fd engine

* add tts python example

* add readme

* fix comment

* change paddle model

* fix readme style

Co-authored-by: Jason <jiangjiajun@baidu.com>
This commit is contained in:
Thomas Young
2022-10-25 10:24:56 +08:00
committed by GitHub
parent 1d3a114b66
commit f2c09a87a6
7 changed files with 366 additions and 41 deletions

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# PaddleSpeech 流式语音合成
- 本文示例的实现来自[PaddleSpeech 流式语音合成](https://github.com/PaddlePaddle/PaddleSpeech/tree/r1.2).
## 详细部署文档
- [Python部署](python)
- [Serving部署](serving)

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([简体中文](./README_cn.md)|English)
# PP-TTS Streaming Text-to-Speech Python Example
## Introduction
This demo is an implementation of starting the streaming speech synthesis.
## Usage
### 1. Installation
```bash
apt-get install libsndfile1 wget zip
For Centos, yum install libsndfile-devel wget zip
python3 -m pip install --upgrade pip
pip3 install -U fastdeploy-python -f https://www.paddlepaddle.org.cn/whl/fastdeploy.html
pip3 install -U paddlespeech paddlepaddle soundfile matplotlib
```
### 2. Run the example
```bash
python3 stream_play_tts.py
```
### 3. Result
The complete voice synthesis audio is saved as `demo_stream.wav`.
User can install `pyaudio` on their own terminals to play the results of speech synthesis in real time. The relevant code is in `stream_play_tts.py` and you can debug and run it yourself.

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(简体中文|[English](./README.md))
# PP-TTS流式语音合成Python示例
## 介绍
本文介绍了使用FastDeploy运行流式语音合成的示例.
## 使用
### 1. 安装
```bash
apt-get install libsndfile1 wget zip
对于Centos系统,使用yum install libsndfile-devel wget zip
python3 -m pip install --upgrade pip
pip3 install -U fastdeploy-python -f https://www.paddlepaddle.org.cn/whl/fastdeploy.html
pip3 install -U paddlespeech paddlepaddle soundfile matplotlib
```
### 2. 运行示例
```bash
python3 stream_play_tts.py
```
### 3. 运行效果
完整的语音合成音频被保存为`demo_stream.wav`文件.
用户可以在自己的终端上安装pyaudio, 对语音合成的结果进行实时播放, 相关代码在stream_play_tts.py处于注释状态, 用户可自行调试运行.

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# Copyright (c) 2022 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 math
import os
import time
import fastdeploy as fd
import numpy as np
import soundfile as sf
from paddlespeech.server.utils.util import denorm
from paddlespeech.server.utils.util import get_chunks
from paddlespeech.t2s.frontend.zh_frontend import Frontend
model_name_fastspeech2 = "fastspeech2_cnndecoder_csmsc_streaming_static_1.0.0"
model_zip_fastspeech2 = model_name_fastspeech2 + ".zip"
model_url_fastspeech2 = "https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/" + model_zip_fastspeech2
model_name_mb_melgan = "mb_melgan_csmsc_static_0.1.1"
model_zip_mb_melgan = model_name_mb_melgan + ".zip"
model_url_mb_melgan = "https://paddlespeech.bj.bcebos.com/Parakeet/released_models/mb_melgan/" + model_zip_mb_melgan
dir_name = os.path.dirname(os.path.realpath(__file__)) + "/"
if not os.path.exists(model_name_fastspeech2):
if os.path.exists(model_zip_fastspeech2):
os.remove(model_zip_fastspeech2)
fd.download_and_decompress(model_url_fastspeech2, path=dir_name)
os.remove(model_zip_fastspeech2)
if not os.path.exists(model_name_mb_melgan):
if os.path.exists(model_zip_mb_melgan):
os.remove(model_zip_mb_melgan)
fd.download_and_decompress(model_url_mb_melgan, path=dir_name)
os.remove(model_zip_mb_melgan)
voc_block = 36
voc_pad = 14
am_block = 72
am_pad = 12
voc_upsample = 300
# 模型路径
phones_dict = dir_name + model_name_fastspeech2 + "/phone_id_map.txt"
am_stat_path = dir_name + model_name_fastspeech2 + "/speech_stats.npy"
am_encoder_model = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_encoder_infer.pdmodel"
am_decoder_model = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_decoder.pdmodel"
am_postnet_model = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_postnet.pdmodel"
voc_melgan_model = dir_name + model_name_mb_melgan + "/mb_melgan_csmsc.pdmodel"
am_encoder_para = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_encoder_infer.pdiparams"
am_decoder_para = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_decoder.pdiparams"
am_postnet_para = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_postnet.pdiparams"
voc_melgan_para = dir_name + model_name_mb_melgan + "/mb_melgan_csmsc.pdiparams"
frontend = Frontend(phone_vocab_path=phones_dict, tone_vocab_path=None)
am_mu, am_std = np.load(am_stat_path)
option_1 = fd.RuntimeOption()
option_1.set_model_path(am_encoder_model, am_encoder_para)
option_1.use_cpu()
option_1.use_ort_backend()
option_1.set_cpu_thread_num(12)
am_encoder_runtime = fd.Runtime(option_1)
option_2 = fd.RuntimeOption()
option_2.set_model_path(am_decoder_model, am_decoder_para)
option_2.use_cpu()
option_2.use_ort_backend()
option_2.set_cpu_thread_num(12)
am_decoder_runtime = fd.Runtime(option_2)
option_3 = fd.RuntimeOption()
option_3.set_model_path(am_postnet_model, am_postnet_para)
option_3.use_cpu()
option_3.use_ort_backend()
option_3.set_cpu_thread_num(12)
am_postnet_runtime = fd.Runtime(option_3)
option_4 = fd.RuntimeOption()
option_4.set_model_path(voc_melgan_model, voc_melgan_para)
option_4.use_cpu()
option_4.use_ort_backend()
option_4.set_cpu_thread_num(12)
voc_melgan_runtime = fd.Runtime(option_4)
def depadding(data, chunk_num, chunk_id, block, pad, upsample):
"""
Streaming inference removes the result of pad inference
"""
front_pad = min(chunk_id * block, pad)
# first chunk
if chunk_id == 0:
data = data[:block * upsample]
# last chunk
elif chunk_id == chunk_num - 1:
data = data[front_pad * upsample:]
# middle chunk
else:
data = data[front_pad * upsample:(front_pad + block) * upsample]
return data
def inference_stream(text):
input_ids = frontend.get_input_ids(
text, merge_sentences=False, get_tone_ids=False)
phone_ids = input_ids["phone_ids"]
for i in range(len(phone_ids)):
part_phone_ids = phone_ids[i].numpy()
voc_chunk_id = 0
orig_hs = am_encoder_runtime.infer({
'text':
part_phone_ids.astype("int64")
})
orig_hs = orig_hs[0]
# streaming voc chunk info
mel_len = orig_hs.shape[1]
voc_chunk_num = math.ceil(mel_len / voc_block)
start = 0
end = min(voc_block + voc_pad, mel_len)
# streaming am
hss = get_chunks(orig_hs, am_block, am_pad, "am")
am_chunk_num = len(hss)
for i, hs in enumerate(hss):
am_decoder_output = am_decoder_runtime.infer({
'xs':
hs.astype("float32")
})
am_postnet_output = am_postnet_runtime.infer({
'xs':
np.transpose(am_decoder_output[0], (0, 2, 1))
})
am_output_data = am_decoder_output + np.transpose(
am_postnet_output[0], (0, 2, 1))
normalized_mel = am_output_data[0][0]
sub_mel = denorm(normalized_mel, am_mu, am_std)
sub_mel = depadding(sub_mel, am_chunk_num, i, am_block, am_pad, 1)
if i == 0:
mel_streaming = sub_mel
else:
mel_streaming = np.concatenate((mel_streaming, sub_mel), axis=0)
# streaming voc
# 当流式AM推理的mel帧数大于流式voc推理的chunk size开始进行流式voc 推理
while (mel_streaming.shape[0] >= end and
voc_chunk_id < voc_chunk_num):
voc_chunk = mel_streaming[start:end, :]
sub_wav = voc_melgan_runtime.infer({
'logmel':
voc_chunk.astype("float32")
})
sub_wav = depadding(sub_wav[0], voc_chunk_num, voc_chunk_id,
voc_block, voc_pad, voc_upsample)
yield sub_wav
voc_chunk_id += 1
start = max(0, voc_chunk_id * voc_block - voc_pad)
end = min((voc_chunk_id + 1) * voc_block + voc_pad, mel_len)
if __name__ == '__main__':
text = "欢迎使用飞桨语音合成系统,测试一下合成效果。"
# warm up
# onnxruntime 第一次时间会长一些,建议先 warmup 一下
'''
# pyaudio 播放
p = pyaudio.PyAudio()
stream = p.open(
format=p.get_format_from_width(2), # int16
channels=1,
rate=24000,
output=True)
'''
# 计时
wavs = []
t1 = time.time()
for sub_wav in inference_stream(text):
print("响应时间:", time.time() - t1)
t1 = time.time()
wavs.append(sub_wav.flatten())
# float32 to int16
#wav = float2pcm(sub_wav)
# to bytes
#wav_bytes = wav.tobytes()
#stream.write(wav_bytes)
# 关闭 pyaudio 播放器
#stream.stop_stream()
#stream.close()
#p.terminate()
# 流式合成的结果导出
wav = np.concatenate(wavs)
sf.write("demo_stream.wav", data=wav, samplerate=24000)

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@@ -22,13 +22,16 @@ docker exec -it -u root fastdeploy bash
#### 1.2 Installation (inside the docker) #### 1.2 Installation (inside the docker)
```bash ```bash
apt-get install build-essential python3-dev libssl-dev libffi-dev libxml2 libxml2-dev libxslt1-dev zlib1g-dev libsndfile1 language-pack-zh-hans wget zip apt-get install build-essential python3-dev libssl-dev libffi-dev libxml2 libxml2-dev libxslt1-dev zlib1g-dev libsndfile1 language-pack-zh-hans wget zip
pip3 install paddlespeech python3 -m pip install --upgrade pip
pip3 install -U fastdeploy-python -f https://www.paddlepaddle.org.cn/whl/fastdeploy.html
pip3 install -U paddlespeech paddlepaddle
export LC_ALL="zh_CN.UTF-8" export LC_ALL="zh_CN.UTF-8"
export LANG="zh_CN.UTF-8" export LANG="zh_CN.UTF-8"
export LANGUAGE="zh_CN:zh:en_US:en" export LANGUAGE="zh_CN:zh:en_US:en"
``` ```
#### 1.3 Download models (inside the docker) #### 1.3 Download models (inside the docker, skippable)
The model file will be downloaded and decompressed automatically. If you want to download manually, please use the following command.
```bash ```bash
cd /models/streaming_pp_tts/1 cd /models/streaming_pp_tts/1
wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0.zip wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0.zip

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# PP-TTS流式语音合成服务化部署 # PP-TTS流式语音合成服务化部署
## 介绍 ## 介绍
本文介绍了使用FastDeploy搭建流式语音合成服务的方法 本文介绍了使用FastDeploy搭建流式语音合成服务的方法.
服务端必须在docker内启动,而客户端不是必须在docker容器内. 服务端必须在docker内启动,而客户端不是必须在docker容器内.
**本文所在路径($PWD)下的streaming_pp_tts里包含模型的配置和代码(服务端会加载模型和代码以启动服务),需要将其映射到docker中使用** **本文所在路径($PWD)下的streaming_pp_tts里包含模型的配置和代码(服务端会加载模型和代码以启动服务), 需要将其映射到docker中使用.**
## 使用 ## 使用
### 1. 服务端 ### 1. 服务端
@@ -21,13 +21,18 @@ docker exec -it -u root fastdeploy bash
#### 1.2 安装(在docker内) #### 1.2 安装(在docker内)
```bash ```bash
apt-get install build-essential python3-dev libssl-dev libffi-dev libxml2 libxml2-dev libxslt1-dev zlib1g-dev libsndfile1 language-pack-zh-hans wget zip apt-get install build-essential python3-dev libssl-dev libffi-dev libxml2 libxml2-dev libxslt1-dev zlib1g-dev libsndfile1 language-pack-zh-hans wget zip
pip3 install paddlespeech python3 -m pip install --upgrade pip
pip3 install -U fastdeploy-python -f https://www.paddlepaddle.org.cn/whl/fastdeploy.html
pip3 install -U paddlespeech paddlepaddle
export LC_ALL="zh_CN.UTF-8" export LC_ALL="zh_CN.UTF-8"
export LANG="zh_CN.UTF-8" export LANG="zh_CN.UTF-8"
export LANGUAGE="zh_CN:zh:en_US:en" export LANGUAGE="zh_CN:zh:en_US:en"
``` ```
#### 1.3 下载模型(在docker内) #### 1.3 下载模型(在docker内,可跳过)
模型文件会自动下载并解压缩, 如果您想要手动下载, 请使用下面的命令.
```bash ```bash
cd /models/streaming_pp_tts/1 cd /models/streaming_pp_tts/1
wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0.zip wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0.zip
@@ -35,7 +40,7 @@ wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/mb_melgan/mb_me
unzip fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0.zip unzip fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0.zip
unzip mb_melgan_csmsc_onnx_0.2.0.zip unzip mb_melgan_csmsc_onnx_0.2.0.zip
``` ```
**为了方便用户使用我们推荐用户使用1.1中的`docker -v`命令将$PWD(streaming_pp_tts及里面包含的模型的配置和代码)映射到了docker内的`/models`路径,用户也可以使用其他办法,但无论使用哪种方法,最终在docker内的模型目录及结构如下图所示** **为了方便用户使用, 我们推荐用户使用1.1中的`docker -v`命令将$PWD(streaming_pp_tts及里面包含的模型的配置和代码)映射到了docker内的`/models`路径, 用户也可以使用其他办法, 但无论使用哪种方法, 最终在docker内的模型目录及结构如下图所示.**
``` ```
/models /models

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@@ -11,7 +11,6 @@
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and # See the License for the specific language governing permissions and
# limitations under the License. # limitations under the License.
import codecs import codecs
import json import json
import math import math
@@ -19,14 +18,34 @@ import sys
import threading import threading
import time import time
import fastdeploy as fd
import numpy as np import numpy as np
import onnxruntime as ort
import triton_python_backend_utils as pb_utils import triton_python_backend_utils as pb_utils
from paddlespeech.server.utils.util import denorm from paddlespeech.server.utils.util import denorm
from paddlespeech.server.utils.util import get_chunks from paddlespeech.server.utils.util import get_chunks
from paddlespeech.t2s.frontend.zh_frontend import Frontend from paddlespeech.t2s.frontend.zh_frontend import Frontend
model_name_fastspeech2 = "fastspeech2_cnndecoder_csmsc_streaming_static_1.0.0"
model_zip_fastspeech2 = model_name_fastspeech2 + ".zip"
model_url_fastspeech2 = "https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/" + model_zip_fastspeech2
model_name_mb_melgan = "mb_melgan_csmsc_static_0.1.1"
model_zip_mb_melgan = model_name_mb_melgan + ".zip"
model_url_mb_melgan = "https://paddlespeech.bj.bcebos.com/Parakeet/released_models/mb_melgan/" + model_zip_mb_melgan
dir_name = os.path.dirname(os.path.realpath(__file__)) + "/"
if not os.path.exists(model_name_fastspeech2):
if os.path.exists(model_zip_fastspeech2):
os.remove(model_zip_fastspeech2)
fd.download_and_decompress(model_url_fastspeech2, path=dir_name)
os.remove(model_zip_fastspeech2)
if not os.path.exists(model_name_mb_melgan):
if os.path.exists(model_zip_mb_melgan):
os.remove(model_zip_mb_melgan)
fd.download_and_decompress(model_url_mb_melgan, path=dir_name)
os.remove(model_zip_mb_melgan)
voc_block = 36 voc_block = 36
voc_pad = 14 voc_pad = 14
am_block = 72 am_block = 72
@@ -34,33 +53,49 @@ am_pad = 12
voc_upsample = 300 voc_upsample = 300
# 模型路径 # 模型路径
dir_name = "/models/streaming_tts_serving/1/" phones_dict = dir_name + model_name_fastspeech2 + "/phone_id_map.txt"
phones_dict = dir_name + "fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0/phone_id_map.txt" am_stat_path = dir_name + model_name_fastspeech2 + "/speech_stats.npy"
am_stat_path = dir_name + "fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0/speech_stats.npy"
onnx_am_encoder = dir_name + "fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0/fastspeech2_csmsc_am_encoder_infer.onnx" am_encoder_model = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_encoder_infer.pdmodel"
onnx_am_decoder = dir_name + "fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0/fastspeech2_csmsc_am_decoder.onnx" am_decoder_model = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_decoder.pdmodel"
onnx_am_postnet = dir_name + "fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0/fastspeech2_csmsc_am_postnet.onnx" am_postnet_model = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_postnet.pdmodel"
onnx_voc_melgan = dir_name + "mb_melgan_csmsc_onnx_0.2.0/mb_melgan_csmsc.onnx" voc_melgan_model = dir_name + model_name_mb_melgan + "/mb_melgan_csmsc.pdmodel"
am_encoder_para = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_encoder_infer.pdiparams"
am_decoder_para = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_decoder.pdiparams"
am_postnet_para = dir_name + model_name_fastspeech2 + "/fastspeech2_csmsc_am_postnet.pdiparams"
voc_melgan_para = dir_name + model_name_mb_melgan + "/mb_melgan_csmsc.pdiparams"
frontend = Frontend(phone_vocab_path=phones_dict, tone_vocab_path=None) frontend = Frontend(phone_vocab_path=phones_dict, tone_vocab_path=None)
am_mu, am_std = np.load(am_stat_path) am_mu, am_std = np.load(am_stat_path)
# 用CPU推理 option_1 = fd.RuntimeOption()
providers = ['CPUExecutionProvider'] option_1.set_model_path(am_encoder_model, am_encoder_para)
option_1.use_cpu()
option_1.use_ort_backend()
option_1.set_cpu_thread_num(12)
am_encoder_runtime = fd.Runtime(option_1)
# 配置ort session option_2 = fd.RuntimeOption()
sess_options = ort.SessionOptions() option_2.set_model_path(am_decoder_model, am_decoder_para)
option_2.use_cpu()
option_2.use_ort_backend()
option_2.set_cpu_thread_num(12)
am_decoder_runtime = fd.Runtime(option_2)
# 创建session option_3 = fd.RuntimeOption()
am_encoder_infer_sess = ort.InferenceSession( option_3.set_model_path(am_postnet_model, am_postnet_para)
onnx_am_encoder, providers=providers, sess_options=sess_options) option_3.use_cpu()
am_decoder_sess = ort.InferenceSession( option_3.use_ort_backend()
onnx_am_decoder, providers=providers, sess_options=sess_options) option_3.set_cpu_thread_num(12)
am_postnet_sess = ort.InferenceSession( am_postnet_runtime = fd.Runtime(option_3)
onnx_am_postnet, providers=providers, sess_options=sess_options)
voc_melgan_sess = ort.InferenceSession( option_4 = fd.RuntimeOption()
onnx_voc_melgan, providers=providers, sess_options=sess_options) option_4.set_model_path(voc_melgan_model, voc_melgan_para)
option_4.use_cpu()
option_4.use_ort_backend()
option_4.set_cpu_thread_num(12)
voc_melgan_runtime = fd.Runtime(option_4)
def depadding(data, chunk_num, chunk_id, block, pad, upsample): def depadding(data, chunk_num, chunk_id, block, pad, upsample):
@@ -199,8 +234,10 @@ class TritonPythonModel:
part_phone_ids = phone_ids[i].numpy() part_phone_ids = phone_ids[i].numpy()
voc_chunk_id = 0 voc_chunk_id = 0
orig_hs = am_encoder_infer_sess.run( orig_hs = am_encoder_runtime.infer({
None, input_feed={'text': part_phone_ids}) 'text':
part_phone_ids.astype("int64")
})
orig_hs = orig_hs[0] orig_hs = orig_hs[0]
# streaming voc chunk info # streaming voc chunk info
@@ -213,13 +250,16 @@ class TritonPythonModel:
hss = get_chunks(orig_hs, am_block, am_pad, "am") hss = get_chunks(orig_hs, am_block, am_pad, "am")
am_chunk_num = len(hss) am_chunk_num = len(hss)
for i, hs in enumerate(hss): for i, hs in enumerate(hss):
am_decoder_output = am_decoder_sess.run(
None, input_feed={'xs': hs}) am_decoder_output = am_decoder_runtime.infer({
am_postnet_output = am_postnet_sess.run( 'xs':
None, hs.astype("float32")
input_feed={ })
'xs': np.transpose(am_decoder_output[0], (0, 2, 1))
}) am_postnet_output = am_postnet_runtime.infer({
'xs':
np.transpose(am_decoder_output[0], (0, 2, 1))
})
am_output_data = am_decoder_output + np.transpose( am_output_data = am_decoder_output + np.transpose(
am_postnet_output[0], (0, 2, 1)) am_postnet_output[0], (0, 2, 1))
normalized_mel = am_output_data[0][0] normalized_mel = am_output_data[0][0]
@@ -239,9 +279,10 @@ class TritonPythonModel:
while (mel_streaming.shape[0] >= end and while (mel_streaming.shape[0] >= end and
voc_chunk_id < voc_chunk_num): voc_chunk_id < voc_chunk_num):
voc_chunk = mel_streaming[start:end, :] voc_chunk = mel_streaming[start:end, :]
sub_wav = voc_melgan_runtime.infer({
sub_wav = voc_melgan_sess.run( 'logmel':
output_names=None, input_feed={'logmel': voc_chunk}) voc_chunk.astype("float32")
})
sub_wav = depadding(sub_wav[0], voc_chunk_num, voc_chunk_id, sub_wav = depadding(sub_wav[0], voc_chunk_num, voc_chunk_id,
voc_block, voc_pad, voc_upsample) voc_block, voc_pad, voc_upsample)