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提交 11317999 编写于 作者: H Hui Zhang
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Merge branch 'develop' into audio_dev_merge

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.github/ISSUE_TEMPLATE/bug_report.md → .github/ISSUE_TEMPLATE/bug-report-s2t.md
浏览文件 @ 11317999
--- ---
name: Bug report name: "\U0001F41B S2T Bug Report"
about: Create a report to help us improve about: Create a report to help us improve
title: '' title: "[S2T]XXXX"
labels: '' labels: Bug, S2T
assignees: '' assignees: zh794390558
--- ---
...@@ -27,7 +27,7 @@ A clear and concise description of what you expected to happen. ...@@ -27,7 +27,7 @@ A clear and concise description of what you expected to happen.
**Screenshots** **Screenshots**
If applicable, add screenshots to help explain your problem. If applicable, add screenshots to help explain your problem.
** Environment (please complete the following information):** **Environment (please complete the following information):**
- OS: [e.g. Ubuntu] - OS: [e.g. Ubuntu]
- GCC/G++ Version [e.g. 8.3] - GCC/G++ Version [e.g. 8.3]
- Python Version [e.g. 3.7] - Python Version [e.g. 3.7]
......
.github/ISSUE_TEMPLATE/bug-report-tts.md 0 → 100644
浏览文件 @ 11317999
---
name: "\U0001F41B TTS Bug Report"
about: Create a report to help us improve
title: "[TTS]XXXX"
labels: Bug, T2S
assignees: yt605155624
---
For support and discussions, please use our [Discourse forums](https://github.com/PaddlePaddle/DeepSpeech/discussions).
If you've found a bug then please create an issue with the following information:
**Describe the bug**
A clear and concise description of what the bug is.
**To Reproduce**
Steps to reproduce the behavior:
1. Go to '...'
2. Click on '....'
3. Scroll down to '....'
4. See error
**Expected behavior**
A clear and concise description of what you expected to happen.
**Screenshots**
If applicable, add screenshots to help explain your problem.
**Environment (please complete the following information):**
- OS: [e.g. Ubuntu]
- GCC/G++ Version [e.g. 8.3]
- Python Version [e.g. 3.7]
- PaddlePaddle Version [e.g. 2.0.0]
- Model Version [e.g. 2.0.0]
- GPU/DRIVER Informationo [e.g. Tesla V100-SXM2-32GB/440.64.00]
- CUDA/CUDNN Version [e.g. cuda-10.2]
- MKL Version
- TensorRT Version
**Additional context**
Add any other context about the problem here.
.github/ISSUE_TEMPLATE/feature-request.md 0 → 100644
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---
name: "\U0001F680 Feature Request"
about: As a user, I want to request a New Feature on the product.
title: ''
labels: feature request
assignees: D-DanielYang, iftaken
---
## Feature Request
**Is your feature request related to a problem? Please describe:**
<!-- A clear and concise description of what the problem is. Ex. I'm always frustrated when [...] -->
**Describe the feature you'd like:**
<!-- A clear and concise description of what you want to happen. -->
**Describe alternatives you've considered:**
<!-- A clear and concise description of any alternative solutions or features you've considered. -->
.github/ISSUE_TEMPLATE/others.md 0 → 100644
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---
name: "\U0001F9E9 Others"
about: Report any other non-support related issues.
title: ''
labels: ''
assignees: ''
---
## Others
<!--
你可以在这里提出任何前面几类模板不适用的问题，包括但不限于：优化性建议、框架使用体验反馈、版本兼容性问题、报错信息不清楚等。
You can report any issues that are not applicable to the previous types of templates, including but not limited to: enhancement suggestions, feedback on the use of the framework, version compatibility issues, unclear error information, etc.
-->
.github/ISSUE_TEMPLATE/question.md 0 → 100644
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---
name: "\U0001F914 Ask a Question"
about: I want to ask a question.
title: ''
labels: Question
assignees: ''
---
## General Question
<!--
Before asking a question, make sure you have:
- Baidu/Google your question.
- Searched open and closed [GitHub issues](https://github.com/PaddlePaddle/PaddleSpeech/issues?q=is%3Aissue)
- Read the documentation:
- [Readme](https://github.com/PaddlePaddle/PaddleSpeech)
- [Doc](https://paddlespeech.readthedocs.io/)
-->
CHANGELOG.md 已删除 100644 → 0
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# Changelog
Date: 2022-3-22, Author: yt605155624.
Add features to: CLI:
- Support aishell3_hifigan、vctk_hifigan
- PRLink: https://github.com/PaddlePaddle/PaddleSpeech/pull/1587
Date: 2022-3-09, Author: yt605155624.
Add features to: T2S:
- Add ljspeech hifigan egs.
- PRLink: https://github.com/PaddlePaddle/PaddleSpeech/pull/1549
Date: 2022-3-08, Author: yt605155624.
Add features to: T2S:
- Add aishell3 hifigan egs.
- PRLink: https://github.com/PaddlePaddle/PaddleSpeech/pull/1545
Date: 2022-3-08, Author: yt605155624.
Add features to: T2S:
- Add vctk hifigan egs.
- PRLink: https://github.com/PaddlePaddle/PaddleSpeech/pull/1544
Date: 2022-1-29, Author: yt605155624.
Add features to: T2S:
- Update aishell3 vc0 with new Tacotron2.
- PRLink: https://github.com/PaddlePaddle/PaddleSpeech/pull/1419
Date: 2022-1-29, Author: yt605155624.
Add features to: T2S:
- Add ljspeech Tacotron2.
- PRLink: https://github.com/PaddlePaddle/PaddleSpeech/pull/1416
Date: 2022-1-24, Author: yt605155624.
Add features to: T2S:
- Add csmsc WaveRNN.
- PRLink: https://github.com/PaddlePaddle/PaddleSpeech/pull/1379
Date: 2022-1-19, Author: yt605155624.
Add features to: T2S:
- Add csmsc Tacotron2.
- PRLink: https://github.com/PaddlePaddle/PaddleSpeech/pull/1314
Date: 2022-1-10, Author: Jackwaterveg.
Add features to: CLI:
- Support English (librispeech/asr1/transformer).
- Support choosing `decode_method` for conformer and transformer models.
- Refactor the config, using the unified config.
- PRLink: https://github.com/PaddlePaddle/PaddleSpeech/pull/1297
***
Date: 2022-1-17, Author: Jackwaterveg.
Add features to: CLI:
- Support deepspeech2 online/offline model(aishell).
- PRLink: https://github.com/PaddlePaddle/PaddleSpeech/pull/1356
***
Date: 2022-1-24, Author: Jackwaterveg.
Add features to: ctc_decoders:
- Support online ctc prefix-beam search decoder.
- Unified ctc online decoder and ctc offline decoder.
- PRLink: https://github.com/PaddlePaddle/PaddleSpeech/pull/821
***
README.md
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此差异已折叠。
README_cn.md
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此差异已折叠。
demos/audio_searching/README.md
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...@@ -226,6 +226,12 @@ recall and elapsed time statistics are shown in the following figure： ...@@ -226,6 +226,12 @@ recall and elapsed time statistics are shown in the following figure：
The retrieval framework based on Milvus takes about 2.9 milliseconds to retrieve on the premise of 90% recall rate, and it takes about 500 milliseconds for feature extraction (testing audio takes about 5 seconds), that is, a single audio test takes about 503 milliseconds in total, which can meet most application scenarios. The retrieval framework based on Milvus takes about 2.9 milliseconds to retrieve on the premise of 90% recall rate, and it takes about 500 milliseconds for feature extraction (testing audio takes about 5 seconds), that is, a single audio test takes about 503 milliseconds in total, which can meet most application scenarios.
* compute embeding takes 500 ms
* retrieval with cosine takes 2.9 ms
* total takes 503 ms
> test audio is 5 sec
### 6.Pretrained Models ### 6.Pretrained Models
Here is a list of pretrained models released by PaddleSpeech : Here is a list of pretrained models released by PaddleSpeech :
......
demos/audio_searching/src/operations/load.py
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...@@ -26,8 +26,9 @@ def get_audios(path): ...@@ -26,8 +26,9 @@ def get_audios(path):
""" """
supported_formats = [".wav", ".mp3", ".ogg", ".flac", ".m4a"] supported_formats = [".wav", ".mp3", ".ogg", ".flac", ".m4a"]
return [ return [
item for sublist in [[os.path.join(dir, file) for file in files] item
for dir, _, files in list(os.walk(path))] for sublist in [[os.path.join(dir, file) for file in files]
for dir, _, files in list(os.walk(path))]
for item in sublist if os.path.splitext(item)[1] in supported_formats for item in sublist if os.path.splitext(item)[1] in supported_formats
] ]
......
demos/metaverse/README.md
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([简体中文](./README_cn.md)|English)
# Metaverse # Metaverse
## Introduction ## Introduction
Metaverse is a new Internet application and social form integrating virtual reality produced by integrating a variety of new technologies. Metaverse is a new Internet application and social form integrating virtual reality produced by integrating a variety of new technologies.
......
demos/metaverse/README_cn.md 0 → 100644
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(简体中文|[English](./README.md))
# Metaverse
## 简介
Metaverse 是一种新的互联网应用和社交形式，融合了多种新技术，产生了虚拟现实。
这个演示是一个让图片中的名人“说话”的实现。通过 `PaddleSpeech` 的 `TTS` 模块和 `PaddleGAN` 的组合，我们集成了安装和特定模块到一个 shell 脚本中。
## 使用
您可以使用 `PaddleSpeech` 的 `TTS` 模块和 `PaddleGAN` 让您最喜欢的人说出指定的内容，并构建您的虚拟人。
运行 `run.sh` 完成所有基本程序，包括安装。
```bash
./run.sh
```
在 `run.sh`, 先会执行 `source path.sh` 来设置好环境变量。
如果您想尝试您的句子，请替换 `sentences.txt` 中的句子。
如果您想尝试图像，请将图像替换 shell 脚本中的 `download/Lamarr.png` 。
结果已显示在我们的 [notebook](https://github.com/PaddlePaddle/PaddleSpeech/blob/develop/docs/tutorial/tts/tts_tutorial.ipynb)。
demos/speaker_verification/README.md
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...@@ -19,6 +19,7 @@ The input of this cli demo should be a WAV file(`.wav`), and the sample rate mus ...@@ -19,6 +19,7 @@ The input of this cli demo should be a WAV file(`.wav`), and the sample rate mus
Here are sample files for this demo that can be downloaded: Here are sample files for this demo that can be downloaded:
```bash ```bash
wget -c https://paddlespeech.bj.bcebos.com/vector/audio/85236145389.wav wget -c https://paddlespeech.bj.bcebos.com/vector/audio/85236145389.wav
wget -c https://paddlespeech.bj.bcebos.com/vector/audio/123456789.wav
``` ```
### 3. Usage ### 3. Usage
......
demos/speaker_verification/README_cn.md
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...@@ -19,6 +19,7 @@ ...@@ -19,6 +19,7 @@
```bash ```bash
# 该音频的内容是数字串 85236145389 # 该音频的内容是数字串 85236145389
wget -c https://paddlespeech.bj.bcebos.com/vector/audio/85236145389.wav wget -c https://paddlespeech.bj.bcebos.com/vector/audio/85236145389.wav
wget -c https://paddlespeech.bj.bcebos.com/vector/audio/123456789.wav
``` ```
### 3. 使用方法 ### 3. 使用方法
- 命令行 (推荐使用) - 命令行 (推荐使用)
......
demos/speech_server/conf/application.yaml
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...@@ -61,7 +61,7 @@ tts_python: ...@@ -61,7 +61,7 @@ tts_python:
phones_dict: phones_dict:
tones_dict: tones_dict:
speaker_dict: speaker_dict:
spk_id: 0
# voc (vocoder) choices=['pwgan_csmsc', 'pwgan_ljspeech', 'pwgan_aishell3', # voc (vocoder) choices=['pwgan_csmsc', 'pwgan_ljspeech', 'pwgan_aishell3',
# 'pwgan_vctk', 'mb_melgan_csmsc', 'style_melgan_csmsc', # 'pwgan_vctk', 'mb_melgan_csmsc', 'style_melgan_csmsc',
...@@ -87,7 +87,7 @@ tts_inference: ...@@ -87,7 +87,7 @@ tts_inference:
phones_dict: phones_dict:
tones_dict: tones_dict:
speaker_dict: speaker_dict:
spk_id: 0
am_predictor_conf: am_predictor_conf:
device: # set 'gpu:id' or 'cpu' device: # set 'gpu:id' or 'cpu'
......
demos/speech_web/API.md
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...@@ -401,4 +401,4 @@ curl -X 'GET' \ ...@@ -401,4 +401,4 @@ curl -X 'GET' \
"code": 0, "code": 0,
"result":"AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA", "result":"AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA",
"message": "ok" "message": "ok"
``` ```
\ No newline at end of file
demos/speech_web/speech_server/main.py
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...@@ -3,48 +3,48 @@ ...@@ -3,48 +3,48 @@
# 2. 接收录音音频，返回识别结果 # 2. 接收录音音频，返回识别结果
# 3. 接收ASR识别结果，返回NLP对话结果 # 3. 接收ASR识别结果，返回NLP对话结果
# 4. 接收NLP对话结果，返回TTS音频 # 4. 接收NLP对话结果，返回TTS音频
import argparse
import base64 import base64
import yaml
import os
import json
import datetime import datetime
import json
import os
from typing import List
import aiofiles
import librosa import librosa
import soundfile as sf import soundfile as sf
import numpy as np
import argparse
import uvicorn import uvicorn
import aiofiles from fastapi import FastAPI
from typing import Optional, List from fastapi import File
from pydantic import BaseModel from fastapi import Form
from fastapi import FastAPI, Header, File, UploadFile, Form, Cookie, WebSocket, WebSocketDisconnect from fastapi import UploadFile
from fastapi import WebSocket
from fastapi import WebSocketDisconnect
from fastapi.responses import StreamingResponse from fastapi.responses import StreamingResponse
from starlette.responses import FileResponse from pydantic import BaseModel
from starlette.middleware.cors import CORSMiddleware
from starlette.requests import Request
from starlette.websockets import WebSocketState as WebSocketState
from src.AudioManeger import AudioMannger from src.AudioManeger import AudioMannger
from src.util import *
from src.robot import Robot from src.robot import Robot
from src.WebsocketManeger import ConnectionManager
from src.SpeechBase.vpr import VPR from src.SpeechBase.vpr import VPR
from src.util import *
from src.WebsocketManeger import ConnectionManager
from starlette.middleware.cors import CORSMiddleware
from starlette.requests import Request
from starlette.responses import FileResponse
from starlette.websockets import WebSocketState as WebSocketState
from paddlespeech.server.engine.asr.online.python.asr_engine import PaddleASRConnectionHanddler from paddlespeech.server.engine.asr.online.python.asr_engine import PaddleASRConnectionHanddler
from paddlespeech.server.utils.audio_process import float2pcm from paddlespeech.server.utils.audio_process import float2pcm
# 解析配置 # 解析配置
parser = argparse.ArgumentParser( parser = argparse.ArgumentParser(prog='PaddleSpeechDemo', add_help=True)
prog='PaddleSpeechDemo', add_help=True)
parser.add_argument( parser.add_argument(
"--port", "--port",
action="store", action="store",
type=int, type=int,
help="port of the app", help="port of the app",
default=8010, default=8010,
required=False) required=False)
args = parser.parse_args() args = parser.parse_args()
port = args.port port = args.port
...@@ -60,39 +60,41 @@ ie_model_path = "source/model" ...@@ -60,39 +60,41 @@ ie_model_path = "source/model"
UPLOAD_PATH = "source/vpr" UPLOAD_PATH = "source/vpr"
WAV_PATH = "source/wav" WAV_PATH = "source/wav"
base_sources = [UPLOAD_PATH, WAV_PATH]
base_sources = [
UPLOAD_PATH, WAV_PATH
]
for path in base_sources: for path in base_sources:
os.makedirs(path, exist_ok=True) os.makedirs(path, exist_ok=True)
# 初始化 # 初始化
app = FastAPI() app = FastAPI()
chatbot = Robot(asr_config, tts_config, asr_init_path, ie_model_path=ie_model_path) chatbot = Robot(
asr_config, tts_config, asr_init_path, ie_model_path=ie_model_path)
manager = ConnectionManager() manager = ConnectionManager()
aumanager = AudioMannger(chatbot) aumanager = AudioMannger(chatbot)
aumanager.init() aumanager.init()
vpr = VPR(db_path, dim = 192, top_k = 5) vpr = VPR(db_path, dim=192, top_k=5)
# 服务配置 # 服务配置
class NlpBase(BaseModel): class NlpBase(BaseModel):
chat: str chat: str
class TtsBase(BaseModel): class TtsBase(BaseModel):
text: str text: str
class Audios: class Audios:
def __init__(self) -> None: def __init__(self) -> None:
self.audios = b"" self.audios = b""
audios = Audios() audios = Audios()
###################################################################### ######################################################################
########################### ASR 服务 ################################# ########################### ASR 服务 #################################
##################################################################### #####################################################################
# 接收文件，返回ASR结果 # 接收文件，返回ASR结果
# 上传文件 # 上传文件
@app.post("/asr/offline") @app.post("/asr/offline")
...@@ -101,7 +103,8 @@ async def speech2textOffline(files: List[UploadFile]): ...@@ -101,7 +103,8 @@ async def speech2textOffline(files: List[UploadFile]):
asr_res = "" asr_res = ""
for file in files[:1]: for file in files[:1]:
# 生成时间戳 # 生成时间戳
now_name = "asr_offline_" + datetime.datetime.strftime(datetime.datetime.now(), '%Y%m%d%H%M%S') + randName() + ".wav" now_name = "asr_offline_" + datetime.datetime.strftime(
datetime.datetime.now(), '%Y%m%d%H%M%S') + randName() + ".wav"
out_file_path = os.path.join(WAV_PATH, now_name) out_file_path = os.path.join(WAV_PATH, now_name)
async with aiofiles.open(out_file_path, 'wb') as out_file: async with aiofiles.open(out_file_path, 'wb') as out_file:
content = await file.read() # async read content = await file.read() # async read
...@@ -110,10 +113,9 @@ async def speech2textOffline(files: List[UploadFile]): ...@@ -110,10 +113,9 @@ async def speech2textOffline(files: List[UploadFile]):
# 返回ASR识别结果 # 返回ASR识别结果
asr_res = chatbot.speech2text(out_file_path) asr_res = chatbot.speech2text(out_file_path)
return SuccessRequest(result=asr_res) return SuccessRequest(result=asr_res)
# else:
# return ErrorRequest(message="文件不是.wav格式")
return ErrorRequest(message="上传文件为空") return ErrorRequest(message="上传文件为空")
# 接收文件，同时将wav强制转成16k, int16类型 # 接收文件，同时将wav强制转成16k, int16类型
@app.post("/asr/offlinefile") @app.post("/asr/offlinefile")
async def speech2textOfflineFile(files: List[UploadFile]): async def speech2textOfflineFile(files: List[UploadFile]):
...@@ -121,7 +123,8 @@ async def speech2textOfflineFile(files: List[UploadFile]): ...@@ -121,7 +123,8 @@ async def speech2textOfflineFile(files: List[UploadFile]):
asr_res = "" asr_res = ""
for file in files[:1]: for file in files[:1]:
# 生成时间戳 # 生成时间戳
now_name = "asr_offline_" + datetime.datetime.strftime(datetime.datetime.now(), '%Y%m%d%H%M%S') + randName() + ".wav" now_name = "asr_offline_" + datetime.datetime.strftime(
datetime.datetime.now(), '%Y%m%d%H%M%S') + randName() + ".wav"
out_file_path = os.path.join(WAV_PATH, now_name) out_file_path = os.path.join(WAV_PATH, now_name)
async with aiofiles.open(out_file_path, 'wb') as out_file: async with aiofiles.open(out_file_path, 'wb') as out_file:
content = await file.read() # async read content = await file.read() # async read
...@@ -132,22 +135,18 @@ async def speech2textOfflineFile(files: List[UploadFile]): ...@@ -132,22 +135,18 @@ async def speech2textOfflineFile(files: List[UploadFile]):
wav = float2pcm(wav) # float32 to int16 wav = float2pcm(wav) # float32 to int16
wav_bytes = wav.tobytes() # to bytes wav_bytes = wav.tobytes() # to bytes
wav_base64 = base64.b64encode(wav_bytes).decode('utf8') wav_base64 = base64.b64encode(wav_bytes).decode('utf8')
# 将文件重新写入 # 将文件重新写入
now_name = now_name[:-4] + "_16k" + ".wav" now_name = now_name[:-4] + "_16k" + ".wav"
out_file_path = os.path.join(WAV_PATH, now_name) out_file_path = os.path.join(WAV_PATH, now_name)
sf.write(out_file_path,wav,16000) sf.write(out_file_path, wav, 16000)
# 返回ASR识别结果 # 返回ASR识别结果
asr_res = chatbot.speech2text(out_file_path) asr_res = chatbot.speech2text(out_file_path)
response_res = { response_res = {"asr_result": asr_res, "wav_base64": wav_base64}
"asr_result": asr_res,
"wav_base64": wav_base64
}
return SuccessRequest(result=response_res) return SuccessRequest(result=response_res)
return ErrorRequest(message="上传文件为空")
return ErrorRequest(message="上传文件为空")
# 流式接收测试 # 流式接收测试
...@@ -161,15 +160,17 @@ async def speech2textOnlineRecive(files: List[UploadFile]): ...@@ -161,15 +160,17 @@ async def speech2textOnlineRecive(files: List[UploadFile]):
print(f"audios长度变化: {len(audios.audios)}") print(f"audios长度变化: {len(audios.audios)}")
return SuccessRequest(message="接收成功") return SuccessRequest(message="接收成功")
# 采集环境噪音大小 # 采集环境噪音大小
@app.post("/asr/collectEnv") @app.post("/asr/collectEnv")
async def collectEnv(files: List[UploadFile]): async def collectEnv(files: List[UploadFile]):
for file in files[:1]: for file in files[:1]:
content = await file.read() # async read content = await file.read() # async read
# 初始化, wav 前44字节是头部信息 # 初始化, wav 前44字节是头部信息
aumanager.compute_env_volume(content[44:]) aumanager.compute_env_volume(content[44:])
vad_ = aumanager.vad_threshold vad_ = aumanager.vad_threshold
return SuccessRequest(result=vad_,message="采集环境噪音成功") return SuccessRequest(result=vad_, message="采集环境噪音成功")
# 停止录音 # 停止录音
@app.get("/asr/stopRecord") @app.get("/asr/stopRecord")
...@@ -179,6 +180,7 @@ async def stopRecord(): ...@@ -179,6 +180,7 @@ async def stopRecord():
print("Online录音暂停") print("Online录音暂停")
return SuccessRequest(message="停止成功") return SuccessRequest(message="停止成功")
# 恢复录音 # 恢复录音
@app.get("/asr/resumeRecord") @app.get("/asr/resumeRecord")
async def resumeRecord(): async def resumeRecord():
...@@ -187,7 +189,7 @@ async def resumeRecord(): ...@@ -187,7 +189,7 @@ async def resumeRecord():
return SuccessRequest(message="Online录音恢复") return SuccessRequest(message="Online录音恢复")
# 聊天用的ASR # 聊天用的 ASR
@app.websocket("/ws/asr/offlineStream") @app.websocket("/ws/asr/offlineStream")
async def websocket_endpoint(websocket: WebSocket): async def websocket_endpoint(websocket: WebSocket):
await manager.connect(websocket) await manager.connect(websocket)
...@@ -210,9 +212,9 @@ async def websocket_endpoint(websocket: WebSocket): ...@@ -210,9 +212,9 @@ async def websocket_endpoint(websocket: WebSocket):
# print(f"用户-{user}-离开") # print(f"用户-{user}-离开")
# Online识别的ASR # 流式识别的 ASR
@app.websocket('/ws/asr/onlineStream') @app.websocket('/ws/asr/onlineStream')
async def websocket_endpoint(websocket: WebSocket): async def websocket_endpoint_online(websocket: WebSocket):
"""PaddleSpeech Online ASR Server api """PaddleSpeech Online ASR Server api
Args: Args:
...@@ -298,12 +300,14 @@ async def websocket_endpoint(websocket: WebSocket): ...@@ -298,12 +300,14 @@ async def websocket_endpoint(websocket: WebSocket):
except WebSocketDisconnect: except WebSocketDisconnect:
pass pass
###################################################################### ######################################################################
########################### NLP 服务 ################################# ########################### NLP 服务 #################################
##################################################################### #####################################################################
@app.post("/nlp/chat") @app.post("/nlp/chat")
async def chatOffline(nlp_base:NlpBase): async def chatOffline(nlp_base: NlpBase):
chat = nlp_base.chat chat = nlp_base.chat
if not chat: if not chat:
return ErrorRequest(message="传入文本为空") return ErrorRequest(message="传入文本为空")
...@@ -311,8 +315,9 @@ async def chatOffline(nlp_base:NlpBase): ...@@ -311,8 +315,9 @@ async def chatOffline(nlp_base:NlpBase):
res = chatbot.chat(chat) res = chatbot.chat(chat)
return SuccessRequest(result=res) return SuccessRequest(result=res)
@app.post("/nlp/ie") @app.post("/nlp/ie")
async def ieOffline(nlp_base:NlpBase): async def ieOffline(nlp_base: NlpBase):
nlp_text = nlp_base.chat nlp_text = nlp_base.chat
if not nlp_text: if not nlp_text:
return ErrorRequest(message="传入文本为空") return ErrorRequest(message="传入文本为空")
...@@ -320,17 +325,20 @@ async def ieOffline(nlp_base:NlpBase): ...@@ -320,17 +325,20 @@ async def ieOffline(nlp_base:NlpBase):
res = chatbot.ie(nlp_text) res = chatbot.ie(nlp_text)
return SuccessRequest(result=res) return SuccessRequest(result=res)
###################################################################### ######################################################################
########################### TTS 服务 ################################# ########################### TTS 服务 #################################
##################################################################### #####################################################################
@app.post("/tts/offline") @app.post("/tts/offline")
async def text2speechOffline(tts_base:TtsBase): async def text2speechOffline(tts_base: TtsBase):
text = tts_base.text text = tts_base.text
if not text: if not text:
return ErrorRequest(message="文本为空") return ErrorRequest(message="文本为空")
else: else:
now_name = "tts_"+ datetime.datetime.strftime(datetime.datetime.now(), '%Y%m%d%H%M%S') + randName() + ".wav" now_name = "tts_" + datetime.datetime.strftime(
datetime.datetime.now(), '%Y%m%d%H%M%S') + randName() + ".wav"
out_file_path = os.path.join(WAV_PATH, now_name) out_file_path = os.path.join(WAV_PATH, now_name)
# 保存为文件，再转成base64传输 # 保存为文件，再转成base64传输
chatbot.text2speech(text, outpath=out_file_path) chatbot.text2speech(text, outpath=out_file_path)
...@@ -339,12 +347,14 @@ async def text2speechOffline(tts_base:TtsBase): ...@@ -339,12 +347,14 @@ async def text2speechOffline(tts_base:TtsBase):
base_str = base64.b64encode(data_bin) base_str = base64.b64encode(data_bin)
return SuccessRequest(result=base_str) return SuccessRequest(result=base_str)
# http流式TTS # http流式TTS
@app.post("/tts/online") @app.post("/tts/online")
async def stream_tts(request_body: TtsBase): async def stream_tts(request_body: TtsBase):
text = request_body.text text = request_body.text
return StreamingResponse(chatbot.text2speechStreamBytes(text=text)) return StreamingResponse(chatbot.text2speechStreamBytes(text=text))
# ws流式TTS # ws流式TTS
@app.websocket("/ws/tts/online") @app.websocket("/ws/tts/online")
async def stream_ttsWS(websocket: WebSocket): async def stream_ttsWS(websocket: WebSocket):
...@@ -356,17 +366,11 @@ async def stream_ttsWS(websocket: WebSocket): ...@@ -356,17 +366,11 @@ async def stream_ttsWS(websocket: WebSocket):
if text: if text:
for sub_wav in chatbot.text2speechStream(text=text): for sub_wav in chatbot.text2speechStream(text=text):
# print("发送sub wav: ", len(sub_wav)) # print("发送sub wav: ", len(sub_wav))
res = { res = {"wav": sub_wav, "done": False}
"wav": sub_wav,
"done": False
}
await websocket.send_json(res) await websocket.send_json(res)
# 输送结束 # 输送结束
res = { res = {"wav": sub_wav, "done": True}
"wav": sub_wav,
"done": True
}
await websocket.send_json(res) await websocket.send_json(res)
# manager.disconnect(websocket) # manager.disconnect(websocket)
...@@ -396,8 +400,9 @@ async def vpr_enroll(table_name: str=None, ...@@ -396,8 +400,9 @@ async def vpr_enroll(table_name: str=None,
return {'status': False, 'msg': "spk_id can not be None"} return {'status': False, 'msg': "spk_id can not be None"}
# Save the upload data to server. # Save the upload data to server.
content = await audio.read() content = await audio.read()
now_name = "vpr_enroll_" + datetime.datetime.strftime(datetime.datetime.now(), '%Y%m%d%H%M%S') + randName() + ".wav" now_name = "vpr_enroll_" + datetime.datetime.strftime(
audio_path = os.path.join(UPLOAD_PATH, now_name) datetime.datetime.now(), '%Y%m%d%H%M%S') + randName() + ".wav"
audio_path = os.path.join(UPLOAD_PATH, now_name)
with open(audio_path, "wb+") as f: with open(audio_path, "wb+") as f:
f.write(content) f.write(content)
...@@ -413,20 +418,19 @@ async def vpr_recog(request: Request, ...@@ -413,20 +418,19 @@ async def vpr_recog(request: Request,
audio: UploadFile=File(...)): audio: UploadFile=File(...)):
# Voice print recognition online # Voice print recognition online
# try: # try:
# Save the upload data to server. # Save the upload data to server.
content = await audio.read() content = await audio.read()
now_name = "vpr_query_" + datetime.datetime.strftime(datetime.datetime.now(), '%Y%m%d%H%M%S') + randName() + ".wav" now_name = "vpr_query_" + datetime.datetime.strftime(
query_audio_path = os.path.join(UPLOAD_PATH, now_name) datetime.datetime.now(), '%Y%m%d%H%M%S') + randName() + ".wav"
query_audio_path = os.path.join(UPLOAD_PATH, now_name)
with open(query_audio_path, "wb+") as f: with open(query_audio_path, "wb+") as f:
f.write(content) f.write(content)
spk_ids, paths, scores = vpr.do_search_vpr(query_audio_path) spk_ids, paths, scores = vpr.do_search_vpr(query_audio_path)
res = dict(zip(spk_ids, zip(paths, scores))) res = dict(zip(spk_ids, zip(paths, scores)))
# Sort results by distance metric, closest distances first # Sort results by distance metric, closest distances first
res = sorted(res.items(), key=lambda item: item[1][1], reverse=True) res = sorted(res.items(), key=lambda item: item[1][1], reverse=True)
return res return res
# except Exception as e:
# return {'status': False, 'msg': e}, 400
@app.post('/vpr/del') @app.post('/vpr/del')
...@@ -460,17 +464,18 @@ async def vpr_database64(vprId: int): ...@@ -460,17 +464,18 @@ async def vpr_database64(vprId: int):
return {'status': False, 'msg': "vpr_id can not be None"} return {'status': False, 'msg': "vpr_id can not be None"}
audio_path = vpr.do_get_wav(vprId) audio_path = vpr.do_get_wav(vprId)
# 返回base64 # 返回base64
# 将文件转成16k, 16bit类型的wav文件 # 将文件转成16k, 16bit类型的wav文件
wav, sr = librosa.load(audio_path, sr=16000) wav, sr = librosa.load(audio_path, sr=16000)
wav = float2pcm(wav) # float32 to int16 wav = float2pcm(wav) # float32 to int16
wav_bytes = wav.tobytes() # to bytes wav_bytes = wav.tobytes() # to bytes
wav_base64 = base64.b64encode(wav_bytes).decode('utf8') wav_base64 = base64.b64encode(wav_bytes).decode('utf8')
return SuccessRequest(result=wav_base64) return SuccessRequest(result=wav_base64)
except Exception as e: except Exception as e:
return {'status': False, 'msg': e}, 400 return {'status': False, 'msg': e}, 400
@app.get('/vpr/data') @app.get('/vpr/data')
async def vpr_data(vprId: int): async def vpr_data(vprId: int):
# Get the audio file from path by spk_id in MySQL # Get the audio file from path by spk_id in MySQL
...@@ -482,11 +487,6 @@ async def vpr_data(vprId: int): ...@@ -482,11 +487,6 @@ async def vpr_data(vprId: int):
except Exception as e: except Exception as e:
return {'status': False, 'msg': e}, 400 return {'status': False, 'msg': e}, 400
if __name__ == '__main__': if __name__ == '__main__':
uvicorn.run(app=app, host='0.0.0.0', port=port) uvicorn.run(app=app, host='0.0.0.0', port=port)
demos/speech_web/speech_server/requirements.txt
浏览文件 @ 11317999
aiofiles aiofiles
faiss-cpu
fastapi fastapi
librosa librosa
numpy numpy
paddlenlp
paddlepaddle
paddlespeech
pydantic pydantic
scikit_learn python-multipartscikit_learn
SoundFile SoundFile
starlette starlette
uvicorn uvicorn
paddlepaddle
paddlespeech
paddlenlp
faiss-cpu
python-multipart
\ No newline at end of file
demos/speech_web/speech_server/src/AudioManeger.py
浏览文件 @ 11317999
import imp import datetime
from queue import Queue
import numpy as np
import os import os
import wave import wave
import random
import datetime import numpy as np
from .util import randName from .util import randName
class AudioMannger: class AudioMannger:
def __init__(self, robot, frame_length=160, frame=10, data_width=2, vad_default = 300): def __init__(self,
robot,
frame_length=160,
frame=10,
data_width=2,
vad_default=300):
# 二进制 pcm 流 # 二进制 pcm 流
self.audios = b'' self.audios = b''
self.asr_result = "" self.asr_result = ""
...@@ -20,8 +24,9 @@ class AudioMannger: ...@@ -20,8 +24,9 @@ class AudioMannger:
os.makedirs(self.file_dir, exist_ok=True) os.makedirs(self.file_dir, exist_ok=True)
self.vad_deafult = vad_default self.vad_deafult = vad_default
self.vad_threshold = vad_default self.vad_threshold = vad_default
self.vad_threshold_path = os.path.join(self.file_dir, "vad_threshold.npy") self.vad_threshold_path = os.path.join(self.file_dir,
"vad_threshold.npy")
# 10ms 一帧 # 10ms 一帧
self.frame_length = frame_length self.frame_length = frame_length
# 10帧，检测一次 vad # 10帧，检测一次 vad
...@@ -30,67 +35,64 @@ class AudioMannger: ...@@ -30,67 +35,64 @@ class AudioMannger:
self.data_width = data_width self.data_width = data_width
# window # window
self.window_length = frame_length * frame * data_width self.window_length = frame_length * frame * data_width
# 是否开始录音 # 是否开始录音
self.on_asr = False self.on_asr = False
self.silence_cnt = 0 self.silence_cnt = 0
self.max_silence_cnt = 4 self.max_silence_cnt = 4
self.is_pause = False # 录音暂停与恢复 self.is_pause = False # 录音暂停与恢复
def init(self): def init(self):
if os.path.exists(self.vad_threshold_path): if os.path.exists(self.vad_threshold_path):
# 平均响度文件存在 # 平均响度文件存在
self.vad_threshold = np.load(self.vad_threshold_path) self.vad_threshold = np.load(self.vad_threshold_path)
def clear_audio(self): def clear_audio(self):
# 清空 pcm 累积片段与 asr 识别结果 # 清空 pcm 累积片段与 asr 识别结果
self.audios = b'' self.audios = b''
def clear_asr(self): def clear_asr(self):
self.asr_result = "" self.asr_result = ""
def compute_chunk_volume(self, start_index, pcm_bins): def compute_chunk_volume(self, start_index, pcm_bins):
# 根据帧长计算能量平均值 # 根据帧长计算能量平均值
pcm_bin = pcm_bins[start_index: start_index + self.window_length] pcm_bin = pcm_bins[start_index:start_index + self.window_length]
# 转成 numpy # 转成 numpy
pcm_np = np.frombuffer(pcm_bin, np.int16) pcm_np = np.frombuffer(pcm_bin, np.int16)
# 归一化 + 计算响度 # 归一化 + 计算响度
x = pcm_np.astype(np.float32) x = pcm_np.astype(np.float32)
x = np.abs(x) x = np.abs(x)
return np.mean(x) return np.mean(x)
def is_speech(self, start_index, pcm_bins): def is_speech(self, start_index, pcm_bins):
# 检查是否没 # 检查是否没
if start_index > len(pcm_bins): if start_index > len(pcm_bins):
return False return False
# 检查从这个 start 开始是否为静音帧 # 检查从这个 start 开始是否为静音帧
energy = self.compute_chunk_volume(start_index=start_index, pcm_bins=pcm_bins) energy = self.compute_chunk_volume(
start_index=start_index, pcm_bins=pcm_bins)
# print(energy) # print(energy)
if energy > self.vad_threshold: if energy > self.vad_threshold:
return True return True
else: else:
return False return False
def compute_env_volume(self, pcm_bins): def compute_env_volume(self, pcm_bins):
max_energy = 0 max_energy = 0
start = 0 start = 0
while start < len(pcm_bins): while start < len(pcm_bins):
energy = self.compute_chunk_volume(start_index=start, pcm_bins=pcm_bins) energy = self.compute_chunk_volume(
start_index=start, pcm_bins=pcm_bins)
if energy > max_energy: if energy > max_energy:
max_energy = energy max_energy = energy
start += self.window_length start += self.window_length
self.vad_threshold = max_energy + 100 if max_energy > self.vad_deafult else self.vad_deafult self.vad_threshold = max_energy + 100 if max_energy > self.vad_deafult else self.vad_deafult
# 保存成文件 # 保存成文件
np.save(self.vad_threshold_path, self.vad_threshold) np.save(self.vad_threshold_path, self.vad_threshold)
print(f"vad 阈值大小: {self.vad_threshold}") print(f"vad 阈值大小: {self.vad_threshold}")
print(f"环境采样保存: {os.path.realpath(self.vad_threshold_path)}") print(f"环境采样保存: {os.path.realpath(self.vad_threshold_path)}")
def stream_asr(self, pcm_bin): def stream_asr(self, pcm_bin):
# 先把 pcm_bin 送进去做端点检测 # 先把 pcm_bin 送进去做端点检测
start = 0 start = 0
...@@ -99,7 +101,7 @@ class AudioMannger: ...@@ -99,7 +101,7 @@ class AudioMannger:
self.on_asr = True self.on_asr = True
self.silence_cnt = 0 self.silence_cnt = 0
print("录音中") print("录音中")
self.audios += pcm_bin[ start : start + self.window_length] self.audios += pcm_bin[start:start + self.window_length]
else: else:
if self.on_asr: if self.on_asr:
self.silence_cnt += 1 self.silence_cnt += 1
...@@ -110,41 +112,42 @@ class AudioMannger: ...@@ -110,41 +112,42 @@ class AudioMannger:
print("录音停止") print("录音停止")
# audios 保存为 wav, 送入 ASR # audios 保存为 wav, 送入 ASR
if len(self.audios) > 2 * 16000: if len(self.audios) > 2 * 16000:
file_path = os.path.join(self.file_dir, "asr_" + datetime.datetime.strftime(datetime.datetime.now(), '%Y%m%d%H%M%S') + randName() + ".wav") file_path = os.path.join(
self.file_dir,
"asr_" + datetime.datetime.strftime(
datetime.datetime.now(),
'%Y%m%d%H%M%S') + randName() + ".wav")
self.save_audio(file_path=file_path) self.save_audio(file_path=file_path)
self.asr_result = self.robot.speech2text(file_path) self.asr_result = self.robot.speech2text(file_path)
self.clear_audio() self.clear_audio()
return self.asr_result return self.asr_result
else: else:
# 正常接收 # 正常接收
print("录音中 静音") print("录音中 静音")
self.audios += pcm_bin[ start : start + self.window_length] self.audios += pcm_bin[start:start + self.window_length]
start += self.window_length start += self.window_length
return "" return ""
def save_audio(self, file_path): def save_audio(self, file_path):
print("保存音频") print("保存音频")
wf = wave.open(file_path, 'wb') # 创建一个音频文件，名字为“01.wav" wf = wave.open(file_path, 'wb') # 创建一个音频文件，名字为“01.wav"
wf.setnchannels(1) # 设置声道数为2 wf.setnchannels(1) # 设置声道数为2
wf.setsampwidth(2) # 设置采样深度为 wf.setsampwidth(2) # 设置采样深度为
wf.setframerate(16000) # 设置采样率为16000 wf.setframerate(16000) # 设置采样率为16000
# 将数据写入创建的音频文件 # 将数据写入创建的音频文件
wf.writeframes(self.audios) wf.writeframes(self.audios)
# 写完后将文件关闭 # 写完后将文件关闭
wf.close() wf.close()
def end(self): def end(self):
# audios 保存为 wav, 送入 ASR # audios 保存为 wav, 送入 ASR
file_path = os.path.join(self.file_dir, "asr.wav") file_path = os.path.join(self.file_dir, "asr.wav")
self.save_audio(file_path=file_path) self.save_audio(file_path=file_path)
return self.robot.speech2text(file_path) return self.robot.speech2text(file_path)
def stop(self): def stop(self):
self.is_pause = True self.is_pause = True
self.audios = b'' self.audios = b''
def resume(self): def resume(self):
self.is_pause = False self.is_pause = False
\ No newline at end of file
demos/speech_web/speech_server/src/SpeechBase/asr.py
浏览文件 @ 11317999
from re import sub
import numpy as np import numpy as np
import paddle
import librosa
import soundfile
from paddlespeech.server.engine.asr.online.python.asr_engine import ASREngine from paddlespeech.server.engine.asr.online.python.asr_engine import ASREngine
from paddlespeech.server.engine.asr.online.python.asr_engine import PaddleASRConnectionHanddler from paddlespeech.server.engine.asr.online.python.asr_engine import PaddleASRConnectionHanddler
from paddlespeech.server.utils.config import get_config from paddlespeech.server.utils.config import get_config
def readWave(samples): def readWave(samples):
x_len = len(samples) x_len = len(samples)
...@@ -31,20 +28,23 @@ def readWave(samples): ...@@ -31,20 +28,23 @@ def readWave(samples):
class ASR: class ASR:
def __init__(self, config_path, ) -> None: def __init__(
self,
config_path, ) -> None:
self.config = get_config(config_path)['asr_online'] self.config = get_config(config_path)['asr_online']
self.engine = ASREngine() self.engine = ASREngine()
self.engine.init(self.config) self.engine.init(self.config)
self.connection_handler = PaddleASRConnectionHanddler(self.engine) self.connection_handler = PaddleASRConnectionHanddler(self.engine)
def offlineASR(self, samples, sample_rate=16000): def offlineASR(self, samples, sample_rate=16000):
x_chunk, x_chunk_lens = self.engine.preprocess(samples=samples, sample_rate=sample_rate) x_chunk, x_chunk_lens = self.engine.preprocess(
samples=samples, sample_rate=sample_rate)
self.engine.run(x_chunk, x_chunk_lens) self.engine.run(x_chunk, x_chunk_lens)
result = self.engine.postprocess() result = self.engine.postprocess()
self.engine.reset() self.engine.reset()
return result return result
def onlineASR(self, samples:bytes=None, is_finished=False): def onlineASR(self, samples: bytes=None, is_finished=False):
if not is_finished: if not is_finished:
# 流式开始 # 流式开始
self.connection_handler.extract_feat(samples) self.connection_handler.extract_feat(samples)
...@@ -58,5 +58,3 @@ class ASR: ...@@ -58,5 +58,3 @@ class ASR:
asr_results = self.connection_handler.get_result() asr_results = self.connection_handler.get_result()
self.connection_handler.reset() self.connection_handler.reset()
return asr_results return asr_results
\ No newline at end of file
demos/speech_web/speech_server/src/SpeechBase/nlp.py
浏览文件 @ 11317999
from paddlenlp import Taskflow from paddlenlp import Taskflow
class NLP: class NLP:
def __init__(self, ie_model_path=None): def __init__(self, ie_model_path=None):
schema = ["时间", "出发地", "目的地", "费用"] schema = ["时间", "出发地", "目的地", "费用"]
if ie_model_path: if ie_model_path:
self.ie_model = Taskflow("information_extraction", self.ie_model = Taskflow(
schema=schema, task_path=ie_model_path) "information_extraction",
schema=schema,
task_path=ie_model_path)
else: else:
self.ie_model = Taskflow("information_extraction", self.ie_model = Taskflow("information_extraction", schema=schema)
schema=schema)
self.dialogue_model = Taskflow("dialogue") self.dialogue_model = Taskflow("dialogue")
def chat(self, text): def chat(self, text):
result = self.dialogue_model([text]) result = self.dialogue_model([text])
return result[0] return result[0]
def ie(self, text): def ie(self, text):
result = self.ie_model(text) result = self.ie_model(text)
return result return result
\ No newline at end of file
demos/speech_web/speech_server/src/SpeechBase/sql_helper.py
浏览文件 @ 11317999
import base64 import base64
import sqlite3
import os import os
import sqlite3
import numpy as np import numpy as np
from pkg_resources import resource_stream
def dict_factory(cursor, row): def dict_factory(cursor, row):
d = {} d = {}
for idx, col in enumerate(cursor.description): for idx, col in enumerate(cursor.description):
d[col[0]] = row[idx] d[col[0]] = row[idx]
return d return d
class DataBase(object): class DataBase(object):
def __init__(self, db_path:str): def __init__(self, db_path: str):
db_path = os.path.realpath(db_path) db_path = os.path.realpath(db_path)
if os.path.exists(db_path): if os.path.exists(db_path):
...@@ -21,12 +22,12 @@ class DataBase(object): ...@@ -21,12 +22,12 @@ class DataBase(object):
db_path_dir = os.path.dirname(db_path) db_path_dir = os.path.dirname(db_path)
os.makedirs(db_path_dir, exist_ok=True) os.makedirs(db_path_dir, exist_ok=True)
self.db_path = db_path self.db_path = db_path
self.conn = sqlite3.connect(self.db_path) self.conn = sqlite3.connect(self.db_path)
self.conn.row_factory = dict_factory self.conn.row_factory = dict_factory
self.cursor = self.conn.cursor() self.cursor = self.conn.cursor()
self.init_database() self.init_database()
def init_database(self): def init_database(self):
""" """
初始化数据库， 若表不存在则创建 初始化数据库， 若表不存在则创建
...@@ -41,20 +42,21 @@ class DataBase(object): ...@@ -41,20 +42,21 @@ class DataBase(object):
""" """
self.cursor.execute(sql) self.cursor.execute(sql)
self.conn.commit() self.conn.commit()
def execute_base(self, sql, data_dict): def execute_base(self, sql, data_dict):
self.cursor.execute(sql, data_dict) self.cursor.execute(sql, data_dict)
self.conn.commit() self.conn.commit()
def insert_one(self, username, vector_base64:str, wav_path): def insert_one(self, username, vector_base64: str, wav_path):
if not os.path.exists(wav_path): if not os.path.exists(wav_path):
return None, "wav not exists" return None, "wav not exists"
else: else:
sql = f""" sql = """
insert into insert into
vprtable (username, vector, wavpath) vprtable (username, vector, wavpath)
values (?, ?, ?) values (?, ?, ?)
""" """
try: try:
self.cursor.execute(sql, (username, vector_base64, wav_path)) self.cursor.execute(sql, (username, vector_base64, wav_path))
self.conn.commit() self.conn.commit()
...@@ -63,25 +65,27 @@ class DataBase(object): ...@@ -63,25 +65,27 @@ class DataBase(object):
except Exception as e: except Exception as e:
print(e) print(e)
return None, e return None, e
def select_all(self): def select_all(self):
sql = """ sql = """
SELECT * from vprtable SELECT * from vprtable
""" """
result = self.cursor.execute(sql).fetchall() result = self.cursor.execute(sql).fetchall()
return result return result
def select_by_id(self, vpr_id): def select_by_id(self, vpr_id):
sql = f""" sql = f"""
SELECT * from vprtable WHERE `id` = {vpr_id} SELECT * from vprtable WHERE `id` = {vpr_id}
""" """
result = self.cursor.execute(sql).fetchall() result = self.cursor.execute(sql).fetchall()
return result return result
def select_by_username(self, username): def select_by_username(self, username):
sql = f""" sql = f"""
SELECT * from vprtable WHERE `username` = '{username}' SELECT * from vprtable WHERE `username` = '{username}'
""" """
result = self.cursor.execute(sql).fetchall() result = self.cursor.execute(sql).fetchall()
return result return result
...@@ -89,28 +93,30 @@ class DataBase(object): ...@@ -89,28 +93,30 @@ class DataBase(object):
sql = f""" sql = f"""
DELETE from vprtable WHERE `username`='{username}' DELETE from vprtable WHERE `username`='{username}'
""" """
self.cursor.execute(sql) self.cursor.execute(sql)
self.conn.commit() self.conn.commit()
def drop_all(self): def drop_all(self):
sql = f""" sql = """
DELETE from vprtable DELETE from vprtable
""" """
self.cursor.execute(sql) self.cursor.execute(sql)
self.conn.commit() self.conn.commit()
def drop_table(self): def drop_table(self):
sql = f""" sql = """
DROP TABLE vprtable DROP TABLE vprtable
""" """
self.cursor.execute(sql) self.cursor.execute(sql)
self.conn.commit() self.conn.commit()
def encode_vector(self, vector:np.ndarray): def encode_vector(self, vector: np.ndarray):
return base64.b64encode(vector).decode('utf8') return base64.b64encode(vector).decode('utf8')
def decode_vector(self, vector_base64, dtype=np.float32): def decode_vector(self, vector_base64, dtype=np.float32):
b = base64.b64decode(vector_base64) b = base64.b64decode(vector_base64)
vc = np.frombuffer(b, dtype=dtype) vc = np.frombuffer(b, dtype=dtype)
return vc return vc
\ No newline at end of file
demos/speech_web/speech_server/src/SpeechBase/tts.py
浏览文件 @ 11317999
...@@ -5,18 +5,19 @@ ...@@ -5,18 +5,19 @@
# 2. 加载模型 # 2. 加载模型
# 3. 端到端推理 # 3. 端到端推理
# 4. 流式推理 # 4. 流式推理
import base64 import base64
import math
import logging import logging
import math
import numpy as np import numpy as np
from paddlespeech.server.utils.onnx_infer import get_sess
from paddlespeech.t2s.frontend.zh_frontend import Frontend from paddlespeech.server.engine.tts.online.onnx.tts_engine import TTSEngine
from paddlespeech.server.utils.util import denorm, get_chunks
from paddlespeech.server.utils.audio_process import float2pcm from paddlespeech.server.utils.audio_process import float2pcm
from paddlespeech.server.utils.config import get_config from paddlespeech.server.utils.config import get_config
from paddlespeech.server.utils.util import denorm
from paddlespeech.server.utils.util import get_chunks
from paddlespeech.t2s.frontend.zh_frontend import Frontend
from paddlespeech.server.engine.tts.online.onnx.tts_engine import TTSEngine
class TTS: class TTS:
def __init__(self, config_path): def __init__(self, config_path):
...@@ -26,12 +27,12 @@ class TTS: ...@@ -26,12 +27,12 @@ class TTS:
self.engine.init(self.config) self.engine.init(self.config)
self.executor = self.engine.executor self.executor = self.engine.executor
#self.engine.warm_up() #self.engine.warm_up()
# 前端初始化 # 前端初始化
self.frontend = Frontend( self.frontend = Frontend(
phone_vocab_path=self.engine.executor.phones_dict, phone_vocab_path=self.engine.executor.phones_dict,
tone_vocab_path=None) tone_vocab_path=None)
def depadding(self, data, chunk_num, chunk_id, block, pad, upsample): def depadding(self, data, chunk_num, chunk_id, block, pad, upsample):
""" """
Streaming inference removes the result of pad inference Streaming inference removes the result of pad inference
...@@ -48,39 +49,37 @@ class TTS: ...@@ -48,39 +49,37 @@ class TTS:
data = data[front_pad * upsample:(front_pad + block) * upsample] data = data[front_pad * upsample:(front_pad + block) * upsample]
return data return data
def offlineTTS(self, text): def offlineTTS(self, text):
get_tone_ids = False get_tone_ids = False
merge_sentences = False merge_sentences = False
input_ids = self.frontend.get_input_ids( input_ids = self.frontend.get_input_ids(
text, text, merge_sentences=merge_sentences, get_tone_ids=get_tone_ids)
merge_sentences=merge_sentences,
get_tone_ids=get_tone_ids)
phone_ids = input_ids["phone_ids"] phone_ids = input_ids["phone_ids"]
wav_list = [] wav_list = []
for i in range(len(phone_ids)): for i in range(len(phone_ids)):
orig_hs = self.engine.executor.am_encoder_infer_sess.run( orig_hs = self.engine.executor.am_encoder_infer_sess.run(
None, input_feed={'text': phone_ids[i].numpy()} None, input_feed={'text': phone_ids[i].numpy()})
)
hs = orig_hs[0] hs = orig_hs[0]
am_decoder_output = self.engine.executor.am_decoder_sess.run( am_decoder_output = self.engine.executor.am_decoder_sess.run(
None, input_feed={'xs': hs}) None, input_feed={'xs': hs})
am_postnet_output = self.engine.executor.am_postnet_sess.run( am_postnet_output = self.engine.executor.am_postnet_sess.run(
None, None,
input_feed={ input_feed={
'xs': np.transpose(am_decoder_output[0], (0, 2, 1)) '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]
mel = denorm(normalized_mel, self.engine.executor.am_mu, self.engine.executor.am_std) mel = denorm(normalized_mel, self.engine.executor.am_mu,
self.engine.executor.am_std)
wav = self.engine.executor.voc_sess.run( wav = self.engine.executor.voc_sess.run(
output_names=None, input_feed={'logmel': mel})[0] output_names=None, input_feed={'logmel': mel})[0]
wav_list.append(wav) wav_list.append(wav)
wavs = np.concatenate(wav_list) wavs = np.concatenate(wav_list)
return wavs return wavs
def streamTTS(self, text): def streamTTS(self, text):
get_tone_ids = False get_tone_ids = False
...@@ -88,9 +87,7 @@ class TTS: ...@@ -88,9 +87,7 @@ class TTS:
# front # front
input_ids = self.frontend.get_input_ids( input_ids = self.frontend.get_input_ids(
text, text, merge_sentences=merge_sentences, get_tone_ids=get_tone_ids)
merge_sentences=merge_sentences,
get_tone_ids=get_tone_ids)
phone_ids = input_ids["phone_ids"] phone_ids = input_ids["phone_ids"]
for i in range(len(phone_ids)): for i in range(len(phone_ids)):
...@@ -105,14 +102,15 @@ class TTS: ...@@ -105,14 +102,15 @@ class TTS:
mel = mel[0] mel = mel[0]
# voc streaming # voc streaming
mel_chunks = get_chunks(mel, self.config.voc_block, self.config.voc_pad, "voc") mel_chunks = get_chunks(mel, self.config.voc_block,
self.config.voc_pad, "voc")
voc_chunk_num = len(mel_chunks) voc_chunk_num = len(mel_chunks)
for i, mel_chunk in enumerate(mel_chunks): for i, mel_chunk in enumerate(mel_chunks):
sub_wav = self.executor.voc_sess.run( sub_wav = self.executor.voc_sess.run(
output_names=None, input_feed={'logmel': mel_chunk}) output_names=None, input_feed={'logmel': mel_chunk})
sub_wav = self.depadding(sub_wav[0], voc_chunk_num, i, sub_wav = self.depadding(
self.config.voc_block, self.config.voc_pad, sub_wav[0], voc_chunk_num, i, self.config.voc_block,
self.config.voc_upsample) self.config.voc_pad, self.config.voc_upsample)
yield self.after_process(sub_wav) yield self.after_process(sub_wav)
...@@ -130,7 +128,8 @@ class TTS: ...@@ -130,7 +128,8 @@ class TTS:
end = min(self.config.voc_block + self.config.voc_pad, mel_len) end = min(self.config.voc_block + self.config.voc_pad, mel_len)
# streaming am # streaming am
hss = get_chunks(orig_hs, self.config.am_block, self.config.am_pad, "am") hss = get_chunks(orig_hs, self.config.am_block,
self.config.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 = self.executor.am_decoder_sess.run( am_decoder_output = self.executor.am_decoder_sess.run(
...@@ -147,7 +146,8 @@ class TTS: ...@@ -147,7 +146,8 @@ class TTS:
sub_mel = denorm(normalized_mel, self.executor.am_mu, sub_mel = denorm(normalized_mel, self.executor.am_mu,
self.executor.am_std) self.executor.am_std)
sub_mel = self.depadding(sub_mel, am_chunk_num, i, sub_mel = self.depadding(sub_mel, am_chunk_num, i,
self.config.am_block, self.config.am_pad, 1) self.config.am_block,
self.config.am_pad, 1)
if i == 0: if i == 0:
mel_streaming = sub_mel mel_streaming = sub_mel
...@@ -165,23 +165,22 @@ class TTS: ...@@ -165,23 +165,22 @@ class TTS:
output_names=None, input_feed={'logmel': voc_chunk}) output_names=None, input_feed={'logmel': voc_chunk})
sub_wav = self.depadding( sub_wav = self.depadding(
sub_wav[0], voc_chunk_num, voc_chunk_id, sub_wav[0], voc_chunk_num, voc_chunk_id,
self.config.voc_block, self.config.voc_pad, self.config.voc_upsample) self.config.voc_block, self.config.voc_pad,
self.config.voc_upsample)
yield self.after_process(sub_wav) yield self.after_process(sub_wav)
voc_chunk_id += 1 voc_chunk_id += 1
start = max( start = max(0, voc_chunk_id * self.config.voc_block -
0, voc_chunk_id * self.config.voc_block - self.config.voc_pad) self.config.voc_pad)
end = min( end = min((voc_chunk_id + 1) * self.config.voc_block +
(voc_chunk_id + 1) * self.config.voc_block + self.config.voc_pad, self.config.voc_pad, mel_len)
mel_len)
else: else:
logging.error( logging.error(
"Only support fastspeech2_csmsc or fastspeech2_cnndecoder_csmsc on streaming tts." "Only support fastspeech2_csmsc or fastspeech2_cnndecoder_csmsc on streaming tts."
) )
def streamTTSBytes(self, text): def streamTTSBytes(self, text):
for wav in self.engine.executor.infer( for wav in self.engine.executor.infer(
text=text, text=text,
...@@ -191,19 +190,14 @@ class TTS: ...@@ -191,19 +190,14 @@ class TTS:
wav = float2pcm(wav) # float32 to int16 wav = float2pcm(wav) # float32 to int16
wav_bytes = wav.tobytes() # to bytes wav_bytes = wav.tobytes() # to bytes
yield wav_bytes yield wav_bytes
def after_process(self, wav): def after_process(self, wav):
# for tvm # for tvm
wav = float2pcm(wav) # float32 to int16 wav = float2pcm(wav) # float32 to int16
wav_bytes = wav.tobytes() # to bytes wav_bytes = wav.tobytes() # to bytes
wav_base64 = base64.b64encode(wav_bytes).decode('utf8') # to base64 wav_base64 = base64.b64encode(wav_bytes).decode('utf8') # to base64
return wav_base64 return wav_base64
def streamTTS_TVM(self, text): def streamTTS_TVM(self, text):
# 用 TVM 优化 # 用 TVM 优化
pass pass
\ No newline at end of file
demos/speech_web/speech_server/src/SpeechBase/vpr.py
浏览文件 @ 11317999
# vpr Demo 没有使用 mysql 与 muilvs, 仅用于docker演示 # vpr Demo 没有使用 mysql 与 muilvs, 仅用于docker演示
import logging import logging
import faiss import faiss
from matplotlib import use
import numpy as np import numpy as np
from .sql_helper import DataBase from .sql_helper import DataBase
from .vpr_encode import get_audio_embedding from .vpr_encode import get_audio_embedding
class VPR: class VPR:
def __init__(self, db_path, dim, top_k) -> None: def __init__(self, db_path, dim, top_k) -> None:
# 初始化 # 初始化
...@@ -14,15 +16,15 @@ class VPR: ...@@ -14,15 +16,15 @@ class VPR:
self.top_k = top_k self.top_k = top_k
self.dtype = np.float32 self.dtype = np.float32
self.vpr_idx = 0 self.vpr_idx = 0
# db 初始化 # db 初始化
self.db = DataBase(db_path) self.db = DataBase(db_path)
# faiss 初始化 # faiss 初始化
index_ip = faiss.IndexFlatIP(dim) index_ip = faiss.IndexFlatIP(dim)
self.index_ip = faiss.IndexIDMap(index_ip) self.index_ip = faiss.IndexIDMap(index_ip)
self.init() self.init()
def init(self): def init(self):
# demo 初始化，把 mysql中的向量注册到 faiss 中 # demo 初始化，把 mysql中的向量注册到 faiss 中
sql_dbs = self.db.select_all() sql_dbs = self.db.select_all()
...@@ -34,12 +36,13 @@ class VPR: ...@@ -34,12 +36,13 @@ class VPR:
if len(vc.shape) == 1: if len(vc.shape) == 1:
vc = np.expand_dims(vc, axis=0) vc = np.expand_dims(vc, axis=0)
# 构建数据库 # 构建数据库
self.index_ip.add_with_ids(vc, np.array((idx,)).astype('int64')) self.index_ip.add_with_ids(vc, np.array(
(idx, )).astype('int64'))
logging.info("faiss 构建完毕") logging.info("faiss 构建完毕")
def faiss_enroll(self, idx, vc): def faiss_enroll(self, idx, vc):
self.index_ip.add_with_ids(vc, np.array((idx,)).astype('int64')) self.index_ip.add_with_ids(vc, np.array((idx, )).astype('int64'))
def vpr_enroll(self, username, wav_path): def vpr_enroll(self, username, wav_path):
# 注册声纹 # 注册声纹
emb = get_audio_embedding(wav_path) emb = get_audio_embedding(wav_path)
...@@ -53,21 +56,22 @@ class VPR: ...@@ -53,21 +56,22 @@ class VPR:
else: else:
last_idx, mess = None last_idx, mess = None
return last_idx return last_idx
def vpr_recog(self, wav_path): def vpr_recog(self, wav_path):
# 识别声纹 # 识别声纹
emb_search = get_audio_embedding(wav_path) emb_search = get_audio_embedding(wav_path)
if emb_search is not None: if emb_search is not None:
emb_search = np.expand_dims(emb_search, axis=0) emb_search = np.expand_dims(emb_search, axis=0)
D, I = self.index_ip.search(emb_search, self.top_k) D, I = self.index_ip.search(emb_search, self.top_k)
D = D.tolist()[0] D = D.tolist()[0]
I = I.tolist()[0] I = I.tolist()[0]
return [(round(D[i] * 100, 2 ), I[i]) for i in range(len(D)) if I[i] != -1] return [(round(D[i] * 100, 2), I[i]) for i in range(len(D))
if I[i] != -1]
else: else:
logging.error("识别失败") logging.error("识别失败")
return None return None
def do_search_vpr(self, wav_path): def do_search_vpr(self, wav_path):
spk_ids, paths, scores = [], [], [] spk_ids, paths, scores = [], [], []
recog_result = self.vpr_recog(wav_path) recog_result = self.vpr_recog(wav_path)
...@@ -78,41 +82,39 @@ class VPR: ...@@ -78,41 +82,39 @@ class VPR:
scores.append(score) scores.append(score)
paths.append("") paths.append("")
return spk_ids, paths, scores return spk_ids, paths, scores
def vpr_del(self, username): def vpr_del(self, username):
# 根据用户username, 删除声纹 # 根据用户username, 删除声纹
# 查用户ID，删除对应向量 # 查用户ID，删除对应向量
res = self.db.select_by_username(username) res = self.db.select_by_username(username)
for r in res: for r in res:
idx = r['id'] idx = r['id']
self.index_ip.remove_ids(np.array((idx,)).astype('int64')) self.index_ip.remove_ids(np.array((idx, )).astype('int64'))
self.db.drop_by_username(username) self.db.drop_by_username(username)
def vpr_list(self): def vpr_list(self):
# 获取数据列表 # 获取数据列表
return self.db.select_all() return self.db.select_all()
def do_list(self): def do_list(self):
spk_ids, vpr_ids = [], [] spk_ids, vpr_ids = [], []
for res in self.db.select_all(): for res in self.db.select_all():
spk_ids.append(res['username']) spk_ids.append(res['username'])
vpr_ids.append(res['id']) vpr_ids.append(res['id'])
return spk_ids, vpr_ids return spk_ids, vpr_ids
def do_get_wav(self, vpr_idx): def do_get_wav(self, vpr_idx):
res = self.db.select_by_id(vpr_idx) res = self.db.select_by_id(vpr_idx)
return res[0]['wavpath'] return res[0]['wavpath']
def vpr_data(self, idx): def vpr_data(self, idx):
# 获取对应ID的数据 # 获取对应ID的数据
res = self.db.select_by_id(idx) res = self.db.select_by_id(idx)
return res return res
def vpr_droptable(self): def vpr_droptable(self):
# 删除表 # 删除表
self.db.drop_table() self.db.drop_table()
# 清空 faiss # 清空 faiss
self.index_ip.reset() self.index_ip.reset()
demos/speech_web/speech_server/src/SpeechBase/vpr_encode.py
浏览文件 @ 11317999
from paddlespeech.cli.vector import VectorExecutor
import numpy as np
import logging import logging
import numpy as np
from paddlespeech.cli.vector import VectorExecutor
vector_executor = VectorExecutor() vector_executor = VectorExecutor()
def get_audio_embedding(path): def get_audio_embedding(path):
""" """
Use vpr_inference to generate embedding of audio Use vpr_inference to generate embedding of audio
...@@ -16,5 +19,3 @@ def get_audio_embedding(path): ...@@ -16,5 +19,3 @@ def get_audio_embedding(path):
except Exception as e: except Exception as e:
logging.error(f"Error with embedding:{e}") logging.error(f"Error with embedding:{e}")
return None return None
\ No newline at end of file
demos/speech_web/speech_server/src/WebsocketManeger.py
浏览文件 @ 11317999
...@@ -2,6 +2,7 @@ from typing import List ...@@ -2,6 +2,7 @@ from typing import List
from fastapi import WebSocket from fastapi import WebSocket
class ConnectionManager: class ConnectionManager:
def __init__(self): def __init__(self):
# 存放激活的ws连接对象 # 存放激活的ws连接对象
...@@ -28,4 +29,4 @@ class ConnectionManager: ...@@ -28,4 +29,4 @@ class ConnectionManager:
await connection.send_text(message) await connection.send_text(message)
manager = ConnectionManager() manager = ConnectionManager()
\ No newline at end of file
demos/speech_web/speech_server/src/robot.py
浏览文件 @ 11317999
from paddlespeech.cli.asr.infer import ASRExecutor
import soundfile as sf
import os import os
import librosa
import soundfile as sf
from src.SpeechBase.asr import ASR from src.SpeechBase.asr import ASR
from src.SpeechBase.tts import TTS
from src.SpeechBase.nlp import NLP from src.SpeechBase.nlp import NLP
from src.SpeechBase.tts import TTS
from paddlespeech.cli.asr.infer import ASRExecutor
class Robot: class Robot:
def __init__(self, asr_config, tts_config,asr_init_path, def __init__(self,
asr_config,
tts_config,
asr_init_path,
ie_model_path=None) -> None: ie_model_path=None) -> None:
self.nlp = NLP(ie_model_path=ie_model_path) self.nlp = NLP(ie_model_path=ie_model_path)
self.asr = ASR(config_path=asr_config) self.asr = ASR(config_path=asr_config)
self.tts = TTS(config_path=tts_config) self.tts = TTS(config_path=tts_config)
self.tts_sample_rate = 24000 self.tts_sample_rate = 24000
self.asr_sample_rate = 16000 self.asr_sample_rate = 16000
# 流式识别效果不如端到端的模型，这里流式模型与端到端模型分开 # 流式识别效果不如端到端的模型，这里流式模型与端到端模型分开
self.asr_model = ASRExecutor() self.asr_model = ASRExecutor()
self.asr_name = "conformer_wenetspeech" self.asr_name = "conformer_wenetspeech"
self.warm_up_asrmodel(asr_init_path) self.warm_up_asrmodel(asr_init_path)
def warm_up_asrmodel(self, asr_init_path): def warm_up_asrmodel(self, asr_init_path):
if not os.path.exists(asr_init_path): if not os.path.exists(asr_init_path):
path_dir = os.path.dirname(asr_init_path) path_dir = os.path.dirname(asr_init_path)
if not os.path.exists(path_dir): if not os.path.exists(path_dir):
os.makedirs(path_dir, exist_ok=True) os.makedirs(path_dir, exist_ok=True)
# TTS生成，采样率24000 # TTS生成，采样率24000
text = "生成初始音频" text = "生成初始音频"
self.text2speech(text, asr_init_path) self.text2speech(text, asr_init_path)
# asr model初始化 # asr model初始化
self.asr_model(asr_init_path, model=self.asr_name,lang='zh', self.asr_model(
sample_rate=16000, force_yes=True) asr_init_path,
model=self.asr_name,
lang='zh',
sample_rate=16000,
force_yes=True)
def speech2text(self, audio_file): def speech2text(self, audio_file):
self.asr_model.preprocess(self.asr_name, audio_file) self.asr_model.preprocess(self.asr_name, audio_file)
self.asr_model.infer(self.asr_name) self.asr_model.infer(self.asr_name)
res = self.asr_model.postprocess() res = self.asr_model.postprocess()
return res return res
def text2speech(self, text, outpath): def text2speech(self, text, outpath):
wav = self.tts.offlineTTS(text) wav = self.tts.offlineTTS(text)
sf.write( sf.write(outpath, wav, samplerate=self.tts_sample_rate)
outpath, wav, samplerate=self.tts_sample_rate)
res = wav res = wav
return res return res
def text2speechStream(self, text): def text2speechStream(self, text):
for sub_wav_base64 in self.tts.streamTTS(text=text): for sub_wav_base64 in self.tts.streamTTS(text=text):
yield sub_wav_base64 yield sub_wav_base64
def text2speechStreamBytes(self, text): def text2speechStreamBytes(self, text):
for wav_bytes in self.tts.streamTTSBytes(text=text): for wav_bytes in self.tts.streamTTSBytes(text=text):
yield wav_bytes yield wav_bytes
...@@ -66,5 +70,3 @@ class Robot: ...@@ -66,5 +70,3 @@ class Robot:
def ie(self, text): def ie(self, text):
result = self.nlp.ie(text) result = self.nlp.ie(text)
return result return result
\ No newline at end of file
demos/speech_web/speech_server/src/util.py
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import random import random
def randName(n=5): def randName(n=5):
return "".join(random.sample('zyxwvutsrqponmlkjihgfedcba',n)) return "".join(random.sample('zyxwvutsrqponmlkjihgfedcba', n))
def SuccessRequest(result=None, message="ok"): def SuccessRequest(result=None, message="ok"):
return { return {"code": 0, "result": result, "message": message}
"code": 0,
"result":result,
"message": message
}
def ErrorRequest(result=None, message="error"): def ErrorRequest(result=None, message="error"):
return { return {"code": -1, "result": result, "message": message}
"code": -1,
"result":result,
"message": message
}
\ No newline at end of file
demos/story_talker/README.md
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([简体中文](./README_cn.md)|English)
# Story Talker # Story Talker
## Introduction ## Introduction
Storybooks are very important children's enlightenment books, but parents usually don't have enough time to read storybooks for their children. For very young children, they may not understand the Chinese characters in storybooks. Or sometimes, children just want to "listen" but don't want to "read". Storybooks are very important children's enlightenment books, but parents usually don't have enough time to read storybooks for their children. For very young children, they may not understand the Chinese characters in storybooks. Or sometimes, children just want to "listen" but don't want to "read".
......
demos/story_talker/README_cn.md 0 → 100644
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(简体中文|[English](./README.md))
# Story Talker
## 简介
故事书是非常重要的儿童启蒙书，但家长通常没有足够的时间为孩子读故事书。对于非常小的孩子，他们可能不理解故事书中的汉字。或有时，孩子们只是想“听”，而不想“读”。
您可以使用 `PaddleOCR` 获取故事书的文本，并通过 `PaddleSpeech` 的 `TTS` 模块进行阅读。
## 使用
运行以下命令行开始：
```
./run.sh
```
结果已显示在 [notebook](https://github.com/PaddlePaddle/PaddleSpeech/blob/develop/docs/tutorial/tts/tts_tutorial.ipynb)。
demos/streaming_asr_server/conf/application.yaml
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...@@ -28,6 +28,7 @@ asr_online: ...@@ -28,6 +28,7 @@ asr_online:
sample_rate: 16000 sample_rate: 16000
cfg_path: cfg_path:
decode_method: decode_method:
num_decoding_left_chunks: -1
force_yes: True force_yes: True
device: 'cpu' # cpu or gpu:id device: 'cpu' # cpu or gpu:id
decode_method: "attention_rescoring" decode_method: "attention_rescoring"
......
demos/streaming_asr_server/local/rtf_from_log.py
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...@@ -34,7 +34,7 @@ if __name__ == '__main__': ...@@ -34,7 +34,7 @@ if __name__ == '__main__':
n = 0 n = 0
for m in rtfs: for m in rtfs:
# not accurate, may have duplicate log # not accurate, may have duplicate log
n += 1 n += 1
T += m['T'] T += m['T']
P += m['P'] P += m['P']
......
demos/streaming_tts_server/conf/tts_online_application.yaml
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...@@ -29,7 +29,7 @@ tts_online: ...@@ -29,7 +29,7 @@ tts_online:
phones_dict: phones_dict:
tones_dict: tones_dict:
speaker_dict: speaker_dict:
spk_id: 0
# voc (vocoder) choices=['mb_melgan_csmsc, hifigan_csmsc'] # voc (vocoder) choices=['mb_melgan_csmsc, hifigan_csmsc']
# Both mb_melgan_csmsc and hifigan_csmsc support streaming voc inference # Both mb_melgan_csmsc and hifigan_csmsc support streaming voc inference
...@@ -70,7 +70,6 @@ tts_online-onnx: ...@@ -70,7 +70,6 @@ tts_online-onnx:
phones_dict: phones_dict:
tones_dict: tones_dict:
speaker_dict: speaker_dict:
spk_id: 0
am_sample_rate: 24000 am_sample_rate: 24000
am_sess_conf: am_sess_conf:
device: "cpu" # set 'gpu:id' or 'cpu' device: "cpu" # set 'gpu:id' or 'cpu'
......
demos/streaming_tts_server/conf/tts_online_ws_application.yaml
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...@@ -29,7 +29,7 @@ tts_online: ...@@ -29,7 +29,7 @@ tts_online:
phones_dict: phones_dict:
tones_dict: tones_dict:
speaker_dict: speaker_dict:
spk_id: 0
# voc (vocoder) choices=['mb_melgan_csmsc, hifigan_csmsc'] # voc (vocoder) choices=['mb_melgan_csmsc, hifigan_csmsc']
# Both mb_melgan_csmsc and hifigan_csmsc support streaming voc inference # Both mb_melgan_csmsc and hifigan_csmsc support streaming voc inference
...@@ -70,7 +70,6 @@ tts_online-onnx: ...@@ -70,7 +70,6 @@ tts_online-onnx:
phones_dict: phones_dict:
tones_dict: tones_dict:
speaker_dict: speaker_dict:
spk_id: 0
am_sample_rate: 24000 am_sample_rate: 24000
am_sess_conf: am_sess_conf:
device: "cpu" # set 'gpu:id' or 'cpu' device: "cpu" # set 'gpu:id' or 'cpu'
......
demos/style_fs2/README.md
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([简体中文](./README_cn.md)|English)
# Style FastSpeech2 # Style FastSpeech2
## Introduction ## Introduction
[FastSpeech2](https://arxiv.org/abs/2006.04558) is a classical acoustic model for Text-to-Speech synthesis, which introduces controllable speech input, including `phoneme duration`、 `energy` and `pitch`. [FastSpeech2](https://arxiv.org/abs/2006.04558) is a classical acoustic model for Text-to-Speech synthesis, which introduces controllable speech input, including `phoneme duration`、 `energy` and `pitch`.
......
demos/style_fs2/README_cn.md 0 → 100644
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(简体中文|[English](./README.md))
# Style FastSpeech2
## 简介
[FastSpeech2](https://arxiv.org/abs/2006.04558) 是用于语音合成的经典声学模型，它引入了可控语音输入，包括 `phoneme duration` 、 `energy` 和 `pitch` 。
在预测阶段，您可以更改这些变量以获得一些有趣的结果。
例如:
1. `FastSpeech2` 中的 `duration` 可以控制音频的速度 ，并保持 `pitch` 。（在某些语音工具中，增加速度将增加音调，反之亦然。）
2. 当我们将一个句子的 `pitch` 设置为平均值并将音素的 `tones` 设置为 `1` 时，我们将获得 `robot-style` 的音色。
3. 当我们提高成年女性的 `pitch` （比例固定）时，我们会得到 `child-style` 的音色。
句子中不同音素的 `duration` 和 `pitch` 可以具有不同的比例。您可以设置不同的音阶比例来强调或削弱某些音素的发音。
## 运行
运行以下命令行开始：
```
./run.sh
```
在 `run.sh`, 会首先执行 `source path.sh` 去设置好环境变量。
如果您想尝试您的句子，请替换 `sentences.txt`中的句子。
更多的细节，请查看 `style_syn.py`。
语音样例可以在 [style-control-in-fastspeech2](https://paddlespeech.readthedocs.io/en/latest/tts/demo.html#style-control-in-fastspeech2) 查看。
demos/text_to_speech/README.md
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...@@ -16,8 +16,8 @@ You can choose one way from easy, meduim and hard to install paddlespeech. ...@@ -16,8 +16,8 @@ You can choose one way from easy, meduim and hard to install paddlespeech.
The input of this demo should be a text of the specific language that can be passed via argument. The input of this demo should be a text of the specific language that can be passed via argument.
### 3. Usage ### 3. Usage
- Command Line (Recommended) - Command Line (Recommended)
The default acoustic model is `Fastspeech2`, and the default vocoder is `HiFiGAN`, the default inference method is dygraph inference.
- Chinese - Chinese
The default acoustic model is `Fastspeech2`, and the default vocoder is `Parallel WaveGAN`.
```bash ```bash
paddlespeech tts --input "你好，欢迎使用百度飞桨深度学习框架！" paddlespeech tts --input "你好，欢迎使用百度飞桨深度学习框架！"
``` ```
...@@ -45,7 +45,33 @@ The input of this demo should be a text of the specific language that can be pas ...@@ -45,7 +45,33 @@ The input of this demo should be a text of the specific language that can be pas
You can change `spk_id` here. You can change `spk_id` here.
```bash ```bash
paddlespeech tts --am fastspeech2_vctk --voc pwgan_vctk --input "hello, boys" --lang en --spk_id 0 paddlespeech tts --am fastspeech2_vctk --voc pwgan_vctk --input "hello, boys" --lang en --spk_id 0
``` ```
- Chinese English Mixed, multi-speaker
You can change `spk_id` here.
```bash
# The `am` must be `fastspeech2_mix`!
# The `lang` must be `mix`!
# The voc must be chinese datasets' voc now!
# spk 174 is csmcc, spk 175 is ljspeech
paddlespeech tts --am fastspeech2_mix --voc hifigan_csmsc --lang mix --input "热烈欢迎您在 Discussions 中提交问题，并在 Issues 中指出发现的 bug。此外，我们非常希望您参与到 Paddle Speech 的开发中！" --spk_id 174 --output mix_spk174.wav
paddlespeech tts --am fastspeech2_mix --voc hifigan_aishell3 --lang mix --input "热烈欢迎您在 Discussions 中提交问题，并在 Issues 中指出发现的 bug。此外，我们非常希望您参与到 Paddle Speech 的开发中！" --spk_id 174 --output mix_spk174_aishell3.wav
paddlespeech tts --am fastspeech2_mix --voc pwgan_csmsc --lang mix --input "我们的声学模型使用了 Fast Speech Two, 声码器使用了 Parallel Wave GAN and Hifi GAN." --spk_id 175 --output mix_spk175_pwgan.wav
paddlespeech tts --am fastspeech2_mix --voc hifigan_csmsc --lang mix --input "我们的声学模型使用了 Fast Speech Two, 声码器使用了 Parallel Wave GAN and Hifi GAN." --spk_id 175 --output mix_spk175.wav
```
- Use ONNXRuntime infer：
```bash
paddlespeech tts --input "你好，欢迎使用百度飞桨深度学习框架！" --output default.wav --use_onnx True
paddlespeech tts --am speedyspeech_csmsc --input "你好，欢迎使用百度飞桨深度学习框架！" --output ss.wav --use_onnx True
paddlespeech tts --voc mb_melgan_csmsc --input "你好，欢迎使用百度飞桨深度学习框架！" --output mb.wav --use_onnx True
paddlespeech tts --voc pwgan_csmsc --input "你好，欢迎使用百度飞桨深度学习框架！" --output pwgan.wav --use_onnx True
paddlespeech tts --am fastspeech2_aishell3 --voc pwgan_aishell3 --input "你好，欢迎使用百度飞桨深度学习框架！" --spk_id 0 --output aishell3_fs2_pwgan.wav --use_onnx True
paddlespeech tts --am fastspeech2_aishell3 --voc hifigan_aishell3 --input "你好，欢迎使用百度飞桨深度学习框架！" --spk_id 0 --output aishell3_fs2_hifigan.wav --use_onnx True
paddlespeech tts --am fastspeech2_ljspeech --voc pwgan_ljspeech --lang en --input "Life was like a box of chocolates, you never know what you're gonna get." --output lj_fs2_pwgan.wav --use_onnx True
paddlespeech tts --am fastspeech2_ljspeech --voc hifigan_ljspeech --lang en --input "Life was like a box of chocolates, you never know what you're gonna get." --output lj_fs2_hifigan.wav --use_onnx True
paddlespeech tts --am fastspeech2_vctk --voc pwgan_vctk --input "Life was like a box of chocolates, you never know what you're gonna get." --lang en --spk_id 0 --output vctk_fs2_pwgan.wav --use_onnx True
paddlespeech tts --am fastspeech2_vctk --voc hifigan_vctk --input "Life was like a box of chocolates, you never know what you're gonna get." --lang en --spk_id 0 --output vctk_fs2_hifigan.wav --use_onnx True
```
Usage: Usage:
```bash ```bash
...@@ -68,6 +94,8 @@ The input of this demo should be a text of the specific language that can be pas ...@@ -68,6 +94,8 @@ The input of this demo should be a text of the specific language that can be pas
- `lang`: Language of tts task. Default: `zh`. - `lang`: Language of tts task. Default: `zh`.
- `device`: Choose device to execute model inference. Default: default device of paddlepaddle in current environment. - `device`: Choose device to execute model inference. Default: default device of paddlepaddle in current environment.
- `output`: Output wave filepath. Default: `output.wav`. - `output`: Output wave filepath. Default: `output.wav`.
- `use_onnx`: whether to usen ONNXRuntime inference.
- `fs`: sample rate for ONNX models when use specified model files.
Output: Output:
```bash ```bash
...@@ -75,54 +103,76 @@ The input of this demo should be a text of the specific language that can be pas ...@@ -75,54 +103,76 @@ The input of this demo should be a text of the specific language that can be pas
``` ```
- Python API - Python API
```python - Dygraph infer:
import paddle ```python
from paddlespeech.cli.tts import TTSExecutor import paddle
from paddlespeech.cli.tts import TTSExecutor
tts_executor = TTSExecutor() tts_executor = TTSExecutor()
wav_file = tts_executor( wav_file = tts_executor(
text='今天的天气不错啊', text='今天的天气不错啊',
output='output.wav', output='output.wav',
am='fastspeech2_csmsc', am='fastspeech2_csmsc',
am_config=None, am_config=None,
am_ckpt=None, am_ckpt=None,
am_stat=None, am_stat=None,
spk_id=0, spk_id=0,
phones_dict=None, phones_dict=None,
tones_dict=None, tones_dict=None,
speaker_dict=None, speaker_dict=None,
voc='pwgan_csmsc', voc='pwgan_csmsc',
voc_config=None, voc_config=None,
voc_ckpt=None, voc_ckpt=None,
voc_stat=None, voc_stat=None,
lang='zh', lang='zh',
device=paddle.get_device()) device=paddle.get_device())
print('Wave file has been generated: {}'.format(wav_file)) print('Wave file has been generated: {}'.format(wav_file))
``` ```
- ONNXRuntime infer:
```python
from paddlespeech.cli.tts import TTSExecutor
tts_executor = TTSExecutor()
wav_file = tts_executor(
text='对数据集进行预处理',
output='output.wav',
am='fastspeech2_csmsc',
voc='hifigan_csmsc',
lang='zh',
use_onnx=True,
cpu_threads=2)
```
Output: Output:
```bash ```bash
Wave file has been generated: output.wav Wave file has been generated: output.wav
``` ```
### 4. Pretrained Models ### 4. Pretrained Models
Here is a list of pretrained models released by PaddleSpeech that can be used by command and python API: Here is a list of pretrained models released by PaddleSpeech that can be used by command and python API:
- Acoustic model - Acoustic model
| Model | Language | Model | Language |
| :--- | :---: | | :--- | :---: |
| speedyspeech_csmsc| zh | speedyspeech_csmsc | zh |
| fastspeech2_csmsc| zh | fastspeech2_csmsc | zh |
| fastspeech2_aishell3| zh | fastspeech2_ljspeech | en |
| fastspeech2_ljspeech| en | fastspeech2_aishell3 | zh |
| fastspeech2_vctk| en | fastspeech2_vctk | en |
| fastspeech2_cnndecoder_csmsc | zh |
| fastspeech2_mix | mix |
| tacotron2_csmsc | zh |
| tacotron2_ljspeech | en |
- Vocoder - Vocoder
| Model | Language | Model | Language |
| :--- | :---: | | :--- | :---: |
| pwgan_csmsc| zh | pwgan_csmsc | zh |
| pwgan_aishell3| zh | pwgan_ljspeech | en |
| pwgan_ljspeech| en | pwgan_aishell3 | zh |
| pwgan_vctk| en | pwgan_vctk | en |
| mb_melgan_csmsc| zh | mb_melgan_csmsc | zh |
| style_melgan_csmsc | zh |
| hifigan_csmsc | zh |
| hifigan_ljspeech | en |
| hifigan_aishell3 | zh |
| hifigan_vctk | en |
| wavernn_csmsc | zh |
demos/text_to_speech/README_cn.md
浏览文件 @ 11317999
(简体中文|[English](./README.md)) (简体中文|[English](./README.md))
# 语音合成 # 语音合成
## 介绍 ## 介绍
语音合成是一种自然语言建模过程，其将文本转换为语音以进行音频演示。 语音合成是一种自然语言建模过程，其将文本转换为语音以进行音频演示。
这个 demo 是一个从给定文本生成音频的实现，它可以通过使用 `PaddleSpeech` 的单个命令或 python 中的几行代码来实现。 这个 demo 是一个从给定文本生成音频的实现，它可以通过使用 `PaddleSpeech` 的单个命令或 python 中的几行代码来实现。
## 使用方法 ## 使用方法
### 1. 安装 ### 1. 安装
请看[安装文档](https://github.com/PaddlePaddle/PaddleSpeech/blob/develop/docs/source/install_cn.md)。 请看[安装文档](https://github.com/PaddlePaddle/PaddleSpeech/blob/develop/docs/source/install_cn.md)。
你可以从 easy，medium，hard 三中方式中选择一种方式安装。 你可以从 easy，medium，hard 三种方式中选择一种方式安装。
### 2. 准备输入 ### 2. 准备输入
这个 demo 的输入是通过参数传递的特定语言的文本。 这个 demo 的输入是通过参数传递的特定语言的文本。
### 3. 使用方法 ### 3. 使用方法
- 命令行 (推荐使用) - 命令行 (推荐使用)
默认的声学模型是 `Fastspeech2`，默认的声码器是 `HiFiGAN`，默认推理方式是动态图推理。
- 中文 - 中文
默认的声学模型是 `Fastspeech2`，默认的声码器是 `Parallel WaveGAN`.
```bash ```bash
paddlespeech tts --input "你好，欢迎使用百度飞桨深度学习框架！" paddlespeech tts --input "你好，欢迎使用百度飞桨深度学习框架！"
``` ```
...@@ -34,7 +31,7 @@ ...@@ -34,7 +31,7 @@
``` ```
- 中文， 多说话人 - 中文， 多说话人
你可以改变 `spk_id` 。 你可以改变 `spk_id`。
```bash ```bash
paddlespeech tts --am fastspeech2_aishell3 --voc pwgan_aishell3 --input "你好，欢迎使用百度飞桨深度学习框架！" --spk_id 0 paddlespeech tts --am fastspeech2_aishell3 --voc pwgan_aishell3 --input "你好，欢迎使用百度飞桨深度学习框架！" --spk_id 0
``` ```
...@@ -45,10 +42,36 @@ ...@@ -45,10 +42,36 @@
``` ```
- 英文，多说话人 - 英文，多说话人
你可以改变 `spk_id` 。 你可以改变 `spk_id`。
```bash ```bash
paddlespeech tts --am fastspeech2_vctk --voc pwgan_vctk --input "hello, boys" --lang en --spk_id 0 paddlespeech tts --am fastspeech2_vctk --voc pwgan_vctk --input "hello, boys" --lang en --spk_id 0
``` ```
- 中英文混合，多说话人
你可以改变 `spk_id`。
```bash
# The `am` must be `fastspeech2_mix`!
# The `lang` must be `mix`!
# The voc must be chinese datasets' voc now!
# spk 174 is csmcc, spk 175 is ljspeech
paddlespeech tts --am fastspeech2_mix --voc hifigan_csmsc --lang mix --input "热烈欢迎您在 Discussions 中提交问题，并在 Issues 中指出发现的 bug。此外，我们非常希望您参与到 Paddle Speech 的开发中！" --spk_id 174 --output mix_spk174.wav
paddlespeech tts --am fastspeech2_mix --voc hifigan_aishell3 --lang mix --input "热烈欢迎您在 Discussions 中提交问题，并在 Issues 中指出发现的 bug。此外，我们非常希望您参与到 Paddle Speech 的开发中！" --spk_id 174 --output mix_spk174_aishell3.wav
paddlespeech tts --am fastspeech2_mix --voc pwgan_csmsc --lang mix --input "我们的声学模型使用了 Fast Speech Two, 声码器使用了 Parallel Wave GAN and Hifi GAN." --spk_id 175 --output mix_spk175_pwgan.wav
paddlespeech tts --am fastspeech2_mix --voc hifigan_csmsc --lang mix --input "我们的声学模型使用了 Fast Speech Two, 声码器使用了 Parallel Wave GAN and Hifi GAN." --spk_id 175 --output mix_spk175.wav
```
- 使用 ONNXRuntime 推理：
```bash
paddlespeech tts --input "你好，欢迎使用百度飞桨深度学习框架！" --output default.wav --use_onnx True
paddlespeech tts --am speedyspeech_csmsc --input "你好，欢迎使用百度飞桨深度学习框架！" --output ss.wav --use_onnx True
paddlespeech tts --voc mb_melgan_csmsc --input "你好，欢迎使用百度飞桨深度学习框架！" --output mb.wav --use_onnx True
paddlespeech tts --voc pwgan_csmsc --input "你好，欢迎使用百度飞桨深度学习框架！" --output pwgan.wav --use_onnx True
paddlespeech tts --am fastspeech2_aishell3 --voc pwgan_aishell3 --input "你好，欢迎使用百度飞桨深度学习框架！" --spk_id 0 --output aishell3_fs2_pwgan.wav --use_onnx True
paddlespeech tts --am fastspeech2_aishell3 --voc hifigan_aishell3 --input "你好，欢迎使用百度飞桨深度学习框架！" --spk_id 0 --output aishell3_fs2_hifigan.wav --use_onnx True
paddlespeech tts --am fastspeech2_ljspeech --voc pwgan_ljspeech --lang en --input "Life was like a box of chocolates, you never know what you're gonna get." --output lj_fs2_pwgan.wav --use_onnx True
paddlespeech tts --am fastspeech2_ljspeech --voc hifigan_ljspeech --lang en --input "Life was like a box of chocolates, you never know what you're gonna get." --output lj_fs2_hifigan.wav --use_onnx True
paddlespeech tts --am fastspeech2_vctk --voc pwgan_vctk --input "Life was like a box of chocolates, you never know what you're gonna get." --lang en --spk_id 0 --output vctk_fs2_pwgan.wav --use_onnx True
paddlespeech tts --am fastspeech2_vctk --voc hifigan_vctk --input "Life was like a box of chocolates, you never know what you're gonna get." --lang en --spk_id 0 --output vctk_fs2_hifigan.wav --use_onnx True
```
使用方法： 使用方法：
```bash ```bash
...@@ -71,6 +94,8 @@ ...@@ -71,6 +94,8 @@
- `lang`：TTS 任务的语言， 默认值：`zh`。 - `lang`：TTS 任务的语言， 默认值：`zh`。
- `device`：执行预测的设备， 默认值：当前系统下 paddlepaddle 的默认 device。 - `device`：执行预测的设备， 默认值：当前系统下 paddlepaddle 的默认 device。
- `output`：输出音频的路径， 默认值：`output.wav`。 - `output`：输出音频的路径， 默认值：`output.wav`。
- `use_onnx`: 是否使用 ONNXRuntime 进行推理。
- `fs`: 使用特定 ONNX 模型时的采样率。
输出： 输出：
```bash ```bash
...@@ -78,31 +103,44 @@ ...@@ -78,31 +103,44 @@
``` ```
- Python API - Python API
```python - 动态图推理:
import paddle ```python
from paddlespeech.cli.tts import TTSExecutor import paddle
from paddlespeech.cli.tts import TTSExecutor
tts_executor = TTSExecutor() tts_executor = TTSExecutor()
wav_file = tts_executor( wav_file = tts_executor(
text='今天的天气不错啊', text='今天的天气不错啊',
output='output.wav', output='output.wav',
am='fastspeech2_csmsc', am='fastspeech2_csmsc',
am_config=None, am_config=None,
am_ckpt=None, am_ckpt=None,
am_stat=None, am_stat=None,
spk_id=0, spk_id=0,
phones_dict=None, phones_dict=None,
tones_dict=None, tones_dict=None,
speaker_dict=None, speaker_dict=None,
voc='pwgan_csmsc', voc='pwgan_csmsc',
voc_config=None, voc_config=None,
voc_ckpt=None, voc_ckpt=None,
voc_stat=None, voc_stat=None,
lang='zh', lang='zh',
device=paddle.get_device()) device=paddle.get_device())
print('Wave file has been generated: {}'.format(wav_file)) print('Wave file has been generated: {}'.format(wav_file))
``` ```
- ONNXRuntime 推理:
```python
from paddlespeech.cli.tts import TTSExecutor
tts_executor = TTSExecutor()
wav_file = tts_executor(
text='对数据集进行预处理',
output='output.wav',
am='fastspeech2_csmsc',
voc='hifigan_csmsc',
lang='zh',
use_onnx=True,
cpu_threads=2)
```
输出： 输出：
```bash ```bash
Wave file has been generated: output.wav Wave file has been generated: output.wav
...@@ -112,19 +150,29 @@ ...@@ -112,19 +150,29 @@
以下是 PaddleSpeech 提供的可以被命令行和 python API 使用的预训练模型列表： 以下是 PaddleSpeech 提供的可以被命令行和 python API 使用的预训练模型列表：
- 声学模型 - 声学模型
| 模型 | 语言 | 模型 | 语言 |
| :--- | :---: | | :--- | :---: |
| speedyspeech_csmsc| zh | speedyspeech_csmsc | zh |
| fastspeech2_csmsc| zh | fastspeech2_csmsc | zh |
| fastspeech2_aishell3| zh | fastspeech2_ljspeech | en |
| fastspeech2_ljspeech| en | fastspeech2_aishell3 | zh |
| fastspeech2_vctk| en | fastspeech2_vctk | en |
| fastspeech2_cnndecoder_csmsc | zh |
| fastspeech2_mix | mix |
| tacotron2_csmsc | zh |
| tacotron2_ljspeech | en |
- 声码器 - 声码器
| 模型 | 语言 | 模型 | 语言 |
| :--- | :---: | | :--- | :---: |
| pwgan_csmsc| zh | pwgan_csmsc | zh |
| pwgan_aishell3| zh | pwgan_ljspeech | en |
| pwgan_ljspeech| en | pwgan_aishell3 | zh |
| pwgan_vctk| en | pwgan_vctk | en |
| mb_melgan_csmsc| zh | mb_melgan_csmsc | zh |
| style_melgan_csmsc | zh |
| hifigan_csmsc | zh |
| hifigan_ljspeech | en |
| hifigan_aishell3 | zh |
| hifigan_vctk | en |
| wavernn_csmsc | zh |
docs/requirements.txt
浏览文件 @ 11317999
myst-parser braceexpand
numpydoc colorlog
recommonmark>=0.5.0
sphinx
sphinx-autobuild
sphinx-markdown-tables
sphinx_rtd_theme
paddlepaddle>=2.2.2
editdistance editdistance
fastapi
g2p_en g2p_en
g2pM g2pM
h5py h5py
...@@ -14,40 +9,45 @@ inflect ...@@ -14,40 +9,45 @@ inflect
jieba jieba
jsonlines jsonlines
kaldiio kaldiio
keyboard
librosa==0.8.1 librosa==0.8.1
loguru loguru
matplotlib matplotlib
myst-parser
nara_wpe nara_wpe
numpydoc
onnxruntime==1.10.0 onnxruntime==1.10.0
opencc opencc
pandas
paddlenlp paddlenlp
paddlepaddle>=2.2.2
paddlespeech_feat paddlespeech_feat
pandas
pathos == 0.2.8
pattern_singleton
Pillow>=9.0.0 Pillow>=9.0.0
praatio==5.0.0 praatio==5.0.0
pypinyin prettytable
pypinyin<=0.44.0
pypinyin-dict pypinyin-dict
python-dateutil python-dateutil
pyworld==0.2.12 pyworld==0.2.12
recommonmark>=0.5.0
resampy==0.2.2 resampy==0.2.2
sacrebleu sacrebleu
scipy scipy
sentencepiece~=0.1.96 sentencepiece~=0.1.96
soundfile~=0.10 soundfile~=0.10
sphinx
sphinx-autobuild
sphinx-markdown-tables
sphinx_rtd_theme
textgrid textgrid
timer timer
tqdm tqdm
typeguard typeguard
uvicorn
visualdl visualdl
webrtcvad webrtcvad
websockets
yacs~=0.1.8 yacs~=0.1.8
prettytable
zhon zhon
colorlog
pathos == 0.2.8
fastapi
websockets
keyboard
uvicorn
pattern_singleton
braceexpand
\ No newline at end of file
docs/source/conf.py
浏览文件 @ 11317999
...@@ -20,10 +20,11 @@ ...@@ -20,10 +20,11 @@
# If extensions (or modules to document with autodoc) are in another directory, # If extensions (or modules to document with autodoc) are in another directory,
# add these directories to sys.path here. If the directory is relative to the # add these directories to sys.path here. If the directory is relative to the
# documentation root, use os.path.abspath to make it absolute, like shown here. # documentation root, use os.path.abspath to make it absolute, like shown here.
import os
import sys
import recommonmark.parser import recommonmark.parser
import sphinx_rtd_theme import sphinx_rtd_theme
import sys
import os
sys.path.insert(0, os.path.abspath('../..')) sys.path.insert(0, os.path.abspath('../..'))
autodoc_mock_imports = ["soundfile", "librosa"] autodoc_mock_imports = ["soundfile", "librosa"]
......
docs/source/released_model.md
浏览文件 @ 11317999
...@@ -42,9 +42,11 @@ SpeedySpeech| CSMSC | [speedyspeech-csmsc](https://github.com/PaddlePaddle/Paddl ...@@ -42,9 +42,11 @@ SpeedySpeech| CSMSC | [speedyspeech-csmsc](https://github.com/PaddlePaddle/Paddl
FastSpeech2| CSMSC |[fastspeech2-csmsc](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/csmsc/tts3)|[fastspeech2_nosil_baker_ckpt_0.4.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_baker_ckpt_0.4.zip)|[fastspeech2_csmsc_static_0.2.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_csmsc_static_0.2.0.zip) </br> [fastspeech2_csmsc_onnx_0.2.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_csmsc_onnx_0.2.0.zip)|157MB| FastSpeech2| CSMSC |[fastspeech2-csmsc](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/csmsc/tts3)|[fastspeech2_nosil_baker_ckpt_0.4.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_baker_ckpt_0.4.zip)|[fastspeech2_csmsc_static_0.2.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_csmsc_static_0.2.0.zip) </br> [fastspeech2_csmsc_onnx_0.2.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_csmsc_onnx_0.2.0.zip)|157MB|
FastSpeech2-Conformer| CSMSC |[fastspeech2-csmsc](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/csmsc/tts3)|[fastspeech2_conformer_baker_ckpt_0.5.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_conformer_baker_ckpt_0.5.zip)||| FastSpeech2-Conformer| CSMSC |[fastspeech2-csmsc](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/csmsc/tts3)|[fastspeech2_conformer_baker_ckpt_0.5.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_conformer_baker_ckpt_0.5.zip)|||
FastSpeech2-CNNDecoder| CSMSC| [fastspeech2-csmsc](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/csmsc/tts3)| [fastspeech2_cnndecoder_csmsc_ckpt_1.0.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_ckpt_1.0.0.zip) | [fastspeech2_cnndecoder_csmsc_static_1.0.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_static_1.0.0.zip) </br>[fastspeech2_cnndecoder_csmsc_streaming_static_1.0.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_streaming_static_1.0.0.zip) </br>[fastspeech2_cnndecoder_csmsc_onnx_1.0.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_onnx_1.0.0.zip) </br>[fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0.zip) | 84MB| FastSpeech2-CNNDecoder| CSMSC| [fastspeech2-csmsc](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/csmsc/tts3)| [fastspeech2_cnndecoder_csmsc_ckpt_1.0.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_ckpt_1.0.0.zip) | [fastspeech2_cnndecoder_csmsc_static_1.0.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_static_1.0.0.zip) </br>[fastspeech2_cnndecoder_csmsc_streaming_static_1.0.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_streaming_static_1.0.0.zip) </br>[fastspeech2_cnndecoder_csmsc_onnx_1.0.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_onnx_1.0.0.zip) </br>[fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_cnndecoder_csmsc_streaming_onnx_1.0.0.zip) | 84MB|
FastSpeech2| AISHELL-3 |[fastspeech2-aishell3](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/tts3)|[fastspeech2_nosil_aishell3_ckpt_0.4.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_aishell3_ckpt_0.4.zip)|[fastspeech2_aishell3_static_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_aishell3_static_1.1.0.zip) </br> [fastspeech2_aishell3_onnx_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_aishell3_onnx_1.1.0.zip)|147MB| FastSpeech2| AISHELL-3 |[fastspeech2-aishell3](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/tts3)|[fastspeech2_aishell3_ckpt_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_aishell3_ckpt_1.1.0.zip)|[fastspeech2_aishell3_static_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_aishell3_static_1.1.0.zip) </br> [fastspeech2_aishell3_onnx_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_aishell3_onnx_1.1.0.zip)|147MB|
FastSpeech2| LJSpeech |[fastspeech2-ljspeech](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/ljspeech/tts3)|[fastspeech2_nosil_ljspeech_ckpt_0.5.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_ljspeech_ckpt_0.5.zip)|[fastspeech2_ljspeech_static_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_ljspeech_static_1.1.0.zip) </br> [fastspeech2_ljspeech_onnx_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_ljspeech_onnx_1.1.0.zip)|145MB| FastSpeech2| LJSpeech |[fastspeech2-ljspeech](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/ljspeech/tts3)|[fastspeech2_nosil_ljspeech_ckpt_0.5.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_ljspeech_ckpt_0.5.zip)|[fastspeech2_ljspeech_static_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_ljspeech_static_1.1.0.zip) </br> [fastspeech2_ljspeech_onnx_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_ljspeech_onnx_1.1.0.zip)|145MB|
FastSpeech2| VCTK |[fastspeech2-vctk](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/vctk/tts3)|[fastspeech2_nosil_vctk_ckpt_0.5.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_vctk_ckpt_0.5.zip)|[fastspeech2_vctk_static_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_vctk_static_1.1.0.zip) </br> [fastspeech2_vctk_onnx_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_vctk_onnx_1.1.0.zip) | 145MB| FastSpeech2| VCTK |[fastspeech2-vctk](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/vctk/tts3)|[fastspeech2_vctk_ckpt_1.2.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_vctk_ckpt_1.2.0.zip)|[fastspeech2_vctk_static_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_vctk_static_1.1.0.zip) </br> [fastspeech2_vctk_onnx_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_vctk_onnx_1.1.0.zip) | 145MB|
FastSpeech2| ZH_EN |[fastspeech2-zh_en](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/zh_en_tts/tts3)|[fastspeech2_mix_ckpt_1.2.0.zip](https://paddlespeech.bj.bcebos.com/t2s/chinse_english_mixed/models/fastspeech2_mix_ckpt_1.2.0.zip)|[fastspeech2_mix_static_0.2.0.zip](https://paddlespeech.bj.bcebos.com/t2s/chinse_english_mixed/models/fastspeech2_mix_static_0.2.0.zip) </br> [fastspeech2_mix_onnx_0.2.0.zip](https://paddlespeech.bj.bcebos.com/t2s/chinse_english_mixed/models/fastspeech2_mix_onnx_0.2.0.zip) | 145MB|
### Vocoders ### Vocoders
Model Type | Dataset| Example Link | Pretrained Models| Static/ONNX Models|Size (static) Model Type | Dataset| Example Link | Pretrained Models| Static/ONNX Models|Size (static)
...@@ -67,7 +69,7 @@ WaveRNN | CSMSC |[WaveRNN-csmsc](https://github.com/PaddlePaddle/PaddleSpeech/tr ...@@ -67,7 +69,7 @@ WaveRNN | CSMSC |[WaveRNN-csmsc](https://github.com/PaddlePaddle/PaddleSpeech/tr
Model Type | Dataset| Example Link | Pretrained Models Model Type | Dataset| Example Link | Pretrained Models
:-------------:| :------------:| :-----: | :-----: | :-------------:| :------------:| :-----: | :-----: |
GE2E| AISHELL-3, etc. |[ge2e](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/other/ge2e)|[ge2e_ckpt_0.3.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/ge2e/ge2e_ckpt_0.3.zip) GE2E| AISHELL-3, etc. |[ge2e](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/other/ge2e)|[ge2e_ckpt_0.3.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/ge2e/ge2e_ckpt_0.3.zip)
GE2E + Tactron2| AISHELL-3 |[ge2e-tactron2-aishell3](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/vc0)|[tacotron2_aishell3_ckpt_vc0_0.2.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/tacotron2/tacotron2_aishell3_ckpt_vc0_0.2.0.zip) GE2E + Tacotron2| AISHELL-3 |[ge2e-Tacotron2-aishell3](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/vc0)|[tacotron2_aishell3_ckpt_vc0_0.2.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/tacotron2/tacotron2_aishell3_ckpt_vc0_0.2.0.zip)
GE2E + FastSpeech2 | AISHELL-3 |[ge2e-fastspeech2-aishell3](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/vc1)|[fastspeech2_nosil_aishell3_vc1_ckpt_0.5.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_aishell3_vc1_ckpt_0.5.zip) GE2E + FastSpeech2 | AISHELL-3 |[ge2e-fastspeech2-aishell3](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/vc1)|[fastspeech2_nosil_aishell3_vc1_ckpt_0.5.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_aishell3_vc1_ckpt_0.5.zip)
......
docs/source/tts/quick_start.md
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...@@ -7,7 +7,7 @@ The examples in PaddleSpeech are mainly classified by datasets, the TTS datasets ...@@ -7,7 +7,7 @@ The examples in PaddleSpeech are mainly classified by datasets, the TTS datasets
* VCTK (English multiple speakers) * VCTK (English multiple speakers)
The models in PaddleSpeech TTS have the following mapping relationship: The models in PaddleSpeech TTS have the following mapping relationship:
* tts0 - Tactron2 * tts0 - Tacotron2
* tts1 - TransformerTTS * tts1 - TransformerTTS
* tts2 - SpeedySpeech * tts2 - SpeedySpeech
* tts3 - FastSpeech2 * tts3 - FastSpeech2
...@@ -17,7 +17,7 @@ The models in PaddleSpeech TTS have the following mapping relationship: ...@@ -17,7 +17,7 @@ The models in PaddleSpeech TTS have the following mapping relationship:
* voc3 - MultiBand MelGAN * voc3 - MultiBand MelGAN
* voc4 - Style MelGAN * voc4 - Style MelGAN
* voc5 - HiFiGAN * voc5 - HiFiGAN
* vc0 - Tactron2 Voice Clone with GE2E * vc0 - Tacotron2 Voice Clone with GE2E
* vc1 - FastSpeech2 Voice Clone with GE2E * vc1 - FastSpeech2 Voice Clone with GE2E
## Quick Start ## Quick Start
......
docs/source/tts/quick_start_cn.md
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...@@ -9,7 +9,7 @@ ...@@ -9,7 +9,7 @@
PaddleSpeech 的 TTS 模型具有以下映射关系： PaddleSpeech 的 TTS 模型具有以下映射关系：
* tts0 - Tactron2 * tts0 - Tacotron2
* tts1 - TransformerTTS * tts1 - TransformerTTS
* tts2 - SpeedySpeech * tts2 - SpeedySpeech
* tts3 - FastSpeech2 * tts3 - FastSpeech2
...@@ -19,7 +19,7 @@ PaddleSpeech 的 TTS 模型具有以下映射关系： ...@@ -19,7 +19,7 @@ PaddleSpeech 的 TTS 模型具有以下映射关系：
* voc3 - MultiBand MelGAN * voc3 - MultiBand MelGAN
* voc4 - Style MelGAN * voc4 - Style MelGAN
* voc5 - HiFiGAN * voc5 - HiFiGAN
* vc0 - Tactron2 Voice Clone with GE2E * vc0 - Tacotron2 Voice Clone with GE2E
* vc1 - FastSpeech2 Voice Clone with GE2E * vc1 - FastSpeech2 Voice Clone with GE2E
## 快速开始 ## 快速开始
......
docs/source/tts/tts_papers.md
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...@@ -5,6 +5,7 @@ ...@@ -5,6 +5,7 @@
- [Disambiguation of Chinese Polyphones in an End-to-End Framework with Semantic Features Extracted by Pre-trained BERT](https://www1.se.cuhk.edu.hk/~hccl/publications/pub/201909_INTERSPEECH_DongyangDAI.pdf) - [Disambiguation of Chinese Polyphones in an End-to-End Framework with Semantic Features Extracted by Pre-trained BERT](https://www1.se.cuhk.edu.hk/~hccl/publications/pub/201909_INTERSPEECH_DongyangDAI.pdf)
- [Polyphone Disambiguation in Mandarin Chinese with Semi-Supervised Learning](https://www.isca-speech.org/archive/pdfs/interspeech_2021/shi21d_interspeech.pdf) - [Polyphone Disambiguation in Mandarin Chinese with Semi-Supervised Learning](https://www.isca-speech.org/archive/pdfs/interspeech_2021/shi21d_interspeech.pdf)
* github: https://github.com/PaperMechanica/SemiPPL * github: https://github.com/PaperMechanica/SemiPPL
- [WikipediaHomographData](https://github.com/google-research-datasets/WikipediaHomographData)
### Text Normalization ### Text Normalization
#### English #### English
- [applenob/text_normalization](https://github.com/applenob/text_normalization) - [applenob/text_normalization](https://github.com/applenob/text_normalization)
......
docs/tutorial/tts/tts_tutorial.ipynb
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...@@ -769,7 +769,7 @@ ...@@ -769,7 +769,7 @@
"```\n", "```\n",
"我们在每个数据集的 README.md 介绍了子目录和模型的对应关系, 在 TTS 中有如下对应关系:\n", "我们在每个数据集的 README.md 介绍了子目录和模型的对应关系, 在 TTS 中有如下对应关系:\n",
"```text\n", "```text\n",
"tts0 - Tactron2\n", "tts0 - Tacotron2\n",
"tts1 - TransformerTTS\n", "tts1 - TransformerTTS\n",
"tts2 - SpeedySpeech\n", "tts2 - SpeedySpeech\n",
"tts3 - FastSpeech2\n", "tts3 - FastSpeech2\n",
......
examples/aishell/asr0/README.md
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...@@ -197,7 +197,7 @@ In some situations, you want to use the trained model to do the inference for th ...@@ -197,7 +197,7 @@ In some situations, you want to use the trained model to do the inference for th
```bash ```bash
if [ ${stage} -le 6 ] && [ ${stop_stage} -ge 6 ]; then if [ ${stage} -le 6 ] && [ ${stop_stage} -ge 6 ]; then
# test a single .wav file # test a single .wav file
CUDA_VISIBLE_DEVICES=0 ./local/test_wav.sh ${conf_path} exp/${ckpt}/checkpoints/${avg_ckpt} ${model_type} ${audio_file} CUDA_VISIBLE_DEVICES=0 ./local/test_wav.sh ${conf_path} ${decode_conf_path} exp/${ckpt}/checkpoints/${avg_ckpt} ${model_type} ${audio_file}
fi fi
``` ```
you can train the model by yourself, or you can download the pretrained model by the script below: you can train the model by yourself, or you can download the pretrained model by the script below:
...@@ -211,5 +211,5 @@ wget -nc https://paddlespeech.bj.bcebos.com/datasets/single_wav/zh/demo_01_03.wa ...@@ -211,5 +211,5 @@ wget -nc https://paddlespeech.bj.bcebos.com/datasets/single_wav/zh/demo_01_03.wa
``` ```
You need to prepare an audio file or use the audio demo above, please confirm the sample rate of the audio is 16K. You can get the result of the audio demo by running the script below. You need to prepare an audio file or use the audio demo above, please confirm the sample rate of the audio is 16K. You can get the result of the audio demo by running the script below.
```bash ```bash
CUDA_VISIBLE_DEVICES= ./local/test_wav.sh conf/deepspeech2.yaml exp/deepspeech2/checkpoints/avg_1 data/demo_01_03.wav CUDA_VISIBLE_DEVICES= ./local/test_wav.sh conf/deepspeech2.yaml conf/tuning/decode.yaml exp/deepspeech2/checkpoints/avg_1 data/demo_01_03.wav
``` ```
examples/aishell3/README.md
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# Aishell3 # Aishell3
* tts0 - Tactron2 * tts0 - Tacotron2
* tts1 - TransformerTTS * tts1 - TransformerTTS
* tts2 - SpeedySpeech * tts2 - SpeedySpeech
* tts3 - FastSpeech2 * tts3 - FastSpeech2
...@@ -8,5 +8,7 @@ ...@@ -8,5 +8,7 @@
* voc1 - Parallel WaveGAN * voc1 - Parallel WaveGAN
* voc2 - MelGAN * voc2 - MelGAN
* voc3 - MultiBand MelGAN * voc3 - MultiBand MelGAN
* vc0 - Tactron2 Voice Cloning with GE2E * vc0 - Tacotron2 Voice Cloning with GE2E
* vc1 - FastSpeech2 Voice Cloning with GE2E * vc1 - FastSpeech2 Voice Cloning with GE2E
* vc2 - FastSpeech2 Voice Cloning with ECAPA-TDNN
* ernie_sat - ERNIE-SAT
examples/aishell3/ernie_sat/README.md
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# ERNIE SAT with AISHELL3 dataset # ERNIE-SAT with VCTK dataset
ERNIE-SAT speech-text joint pretraining framework, which achieves SOTA results in cross-lingual multi-speaker speech synthesis and cross-lingual speech editing tasks, It can be applied to a series of scenarios such as Speech Editing, personalized Speech Synthesis, and Voice Cloning.
## Model Framework
In ERNIE-SAT, we propose two innovations:
- In the pretraining process, the phonemes corresponding to Chinese and English are used as input to achieve cross-language and personalized soft phoneme mapping
- The joint mask learning of speech and text is used to realize the alignment of speech and text
<p align="center">
<img src="https://user-images.githubusercontent.com/24568452/186110814-1b9c6618-a0ab-4c0c-bb3d-3d860b0e8cc2.png" />
</p>
## Dataset
### Download and Extract
Download AISHELL-3 from it's [Official Website](http://www.aishelltech.com/aishell_3) and extract it to `~/datasets`. Then the dataset is in the directory `~/datasets/data_aishell3`.
### Get MFA Result and Extract
We use [MFA2.x](https://github.com/MontrealCorpusTools/Montreal-Forced-Aligner) to get durations for aishell3_fastspeech2.
You can download from here [aishell3_alignment_tone.tar.gz](https://paddlespeech.bj.bcebos.com/MFA/AISHELL-3/with_tone/aishell3_alignment_tone.tar.gz), or train your MFA model reference to [mfa example](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/other/mfa) (use MFA1.x now) of our repo.
## Get Started
Assume the path to the dataset is `~/datasets/data_aishell3`.
Assume the path to the MFA result of AISHELL-3 is `./aishell3_alignment_tone`.
Run the command below to
1. **source path**.
2. preprocess the dataset.
3. train the model.
4. synthesize wavs.
- synthesize waveform from `metadata.jsonl`.
- synthesize waveform from text file.
```bash
./run.sh
```
You can choose a range of stages you want to run, or set `stage` equal to `stop-stage` to use only one stage, for example, running the following command will only preprocess the dataset.
```bash
./run.sh --stage 0 --stop-stage 0
```
### Data Preprocessing
```bash
./local/preprocess.sh ${conf_path}
```
When it is done. A `dump` folder is created in the current directory. The structure of the dump folder is listed below.
```text
dump
├── dev
│ ├── norm
│ └── raw
├── phone_id_map.txt
├── speaker_id_map.txt
├── test
│ ├── norm
│ └── raw
└── train
├── norm
├── raw
└── speech_stats.npy
```
The dataset is split into 3 parts, namely `train`, `dev`, and` test`, each of which contains a `norm` and `raw` subfolder. The raw folder contains speech features of each utterance, while the norm folder contains normalized ones. The statistics used to normalize features are computed from the training set, which is located in `dump/train/*_stats.npy`.
Also, there is a `metadata.jsonl` in each subfolder. It is a table-like file that contains phones, text_lengths, speech_lengths, durations, the path of speech features, speaker, and id of each utterance.
### Model Training
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/train.sh ${conf_path} ${train_output_path}
```
`./local/train.sh` calls `${BIN_DIR}/train.py`.
### Synthesizing
We use [HiFiGAN](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/voc5) as the neural vocoder.
Download pretrained HiFiGAN model from [hifigan_aishell3_ckpt_0.2.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/hifigan/hifigan_aishell3_ckpt_0.2.0.zip) and unzip it.
```bash
unzip hifigan_aishell3_ckpt_0.2.0.zip
```
HiFiGAN checkpoint contains files listed below.
```text
hifigan_aishell3_ckpt_0.2.0
├── default.yaml # default config used to train HiFiGAN
├── feats_stats.npy # statistics used to normalize spectrogram when training HiFiGAN
└── snapshot_iter_2500000.pdz # generator parameters of HiFiGAN
```
`./local/synthesize.sh` calls `${BIN_DIR}/../synthesize.py`, which can synthesize waveform from `metadata.jsonl`.
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_path} ${ckpt_name}
```
## Speech Synthesis and Speech Editing
### Prepare
**prepare aligner**
```bash
mkdir -p tools/aligner
cd tools
# download MFA
wget https://github.com/MontrealCorpusTools/Montreal-Forced-Aligner/releases/download/v1.0.1/montreal-forced-aligner_linux.tar.gz
# extract MFA
tar xvf montreal-forced-aligner_linux.tar.gz
# fix .so of MFA
cd montreal-forced-aligner/lib
ln -snf libpython3.6m.so.1.0 libpython3.6m.so
cd -
# download align models and dicts
cd aligner
wget https://paddlespeech.bj.bcebos.com/MFA/ernie_sat/aishell3_model.zip
wget https://paddlespeech.bj.bcebos.com/MFA/AISHELL-3/with_tone/simple.lexicon
wget https://paddlespeech.bj.bcebos.com/MFA/ernie_sat/vctk_model.zip
wget https://paddlespeech.bj.bcebos.com/MFA/LJSpeech-1.1/cmudict-0.7b
cd ../../
```
**prepare pretrained FastSpeech2 models**
ERNIE-SAT use FastSpeech2 as phoneme duration predictor:
```bash
mkdir download
cd download
wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_conformer_baker_ckpt_0.5.zip
wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_ljspeech_ckpt_0.5.zip
unzip fastspeech2_conformer_baker_ckpt_0.5.zip
unzip fastspeech2_nosil_ljspeech_ckpt_0.5.zip
cd ../
```
**prepare source data**
```bash
mkdir source
cd source
wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/ernie_sat/source/SSB03540307.wav
wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/ernie_sat/source/SSB03540428.wav
wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/ernie_sat/source/LJ050-0278.wav
wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/ernie_sat/source/p243_313.wav
wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/ernie_sat/source/p299_096.wav
wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/ernie_sat/source/this_was_not_the_show_for_me.wav
wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/ernie_sat/source/README.md
cd ../
```
You can check the text of downloaded wavs in `source/README.md`.
### Speech Synthesis and Speech Editing
```bash
./run.sh --stage 3 --stop-stage 3 --gpus 0
```
`stage 3` of `run.sh` calls `local/synthesize_e2e.sh`, `stage 0` of it is **Speech Synthesis** and `stage 1` of it is **Speech Editing**.
You can modify `--wav_path`、`--old_str` and `--new_str` yourself, `--old_str` should be the text corresponding to the audio of `--wav_path`, `--new_str` should be designed according to `--task_name`, both `--source_lang` and `--target_lang` should be `zh` for model trained with AISHELL3 dataset.
## Pretrained Model
Pretrained ErnieSAT model:
- [erniesat_aishell3_ckpt_1.2.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/ernie_sat/erniesat_aishell3_ckpt_1.2.0.zip)
Model | Step | eval/mlm_loss | eval/loss
:-------------:| :------------:| :-----: | :-----:
default| 8(gpu) x 289500|51.723782|51.723782
examples/aishell3/ernie_sat/conf/default.yaml
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# This configuration tested on 8 GPUs (A100) with 80GB GPU memory.
# It takes around 3 days to finish the training,You can adjust
# batch_size、num_workers here and ngpu in local/train.sh for your machine
########################################################### ###########################################################
# FEATURE EXTRACTION SETTING # # FEATURE EXTRACTION SETTING #
########################################################### ###########################################################
...@@ -21,8 +24,8 @@ mlm_prob: 0.8 ...@@ -21,8 +24,8 @@ mlm_prob: 0.8
########################################################### ###########################################################
# DATA SETTING # # DATA SETTING #
########################################################### ###########################################################
batch_size: 20 batch_size: 40
num_workers: 2 num_workers: 8
########################################################### ###########################################################
# MODEL SETTING # # MODEL SETTING #
...@@ -280,4 +283,4 @@ token_list: ...@@ -280,4 +283,4 @@ token_list:
- o3 - o3
- iang5 - iang5
- ei5 - ei5
- <sos/eos> - <sos/eos>
\ No newline at end of file
examples/aishell3/ernie_sat/local/synthesize.sh
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...@@ -4,28 +4,11 @@ config_path=$1 ...@@ -4,28 +4,11 @@ config_path=$1
train_output_path=$2 train_output_path=$2
ckpt_name=$3 ckpt_name=$3
stage=1 stage=0
stop_stage=1 stop_stage=0
# pwgan
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize.py \
--erniesat_config=${config_path} \
--erniesat_ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--erniesat_stat=dump/train/speech_stats.npy \
--voc=pwgan_aishell3 \
--voc_config=pwg_aishell3_ckpt_0.5/default.yaml \
--voc_ckpt=pwg_aishell3_ckpt_0.5/snapshot_iter_1000000.pdz \
--voc_stat=pwg_aishell3_ckpt_0.5/feats_stats.npy \
--test_metadata=dump/test/norm/metadata.jsonl \
--output_dir=${train_output_path}/test \
--phones_dict=dump/phone_id_map.txt
fi
# hifigan # hifigan
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
FLAGS_allocator_strategy=naive_best_fit \ FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \ FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize.py \ python3 ${BIN_DIR}/synthesize.py \
......
examples/aishell3/ernie_sat/local/synthesize_e2e.sh 0 → 100755
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#!/bin/bash
config_path=$1
train_output_path=$2
ckpt_name=$3
stage=0
stop_stage=1
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
echo 'speech synthesize !'
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize_e2e.py \
--task_name=synthesize \
--wav_path=source/SSB03540307.wav \
--old_str='请播放歌曲小苹果' \
--new_str='歌曲真好听' \
--source_lang=zh \
--target_lang=zh \
--erniesat_config=${config_path} \
--phones_dict=dump/phone_id_map.txt \
--erniesat_ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--erniesat_stat=dump/train/speech_stats.npy \
--voc=hifigan_aishell3 \
--voc_config=hifigan_aishell3_ckpt_0.2.0/default.yaml \
--voc_ckpt=hifigan_aishell3_ckpt_0.2.0/snapshot_iter_2500000.pdz \
--voc_stat=hifigan_aishell3_ckpt_0.2.0/feats_stats.npy \
--output_name=exp/pred_gen.wav
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
echo 'speech edit !'
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize_e2e.py \
--task_name=edit \
--wav_path=source/SSB03540428.wav \
--old_str='今天天气很好' \
--new_str='今天心情很好' \
--source_lang=zh \
--target_lang=zh \
--erniesat_config=${config_path} \
--phones_dict=dump/phone_id_map.txt \
--erniesat_ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--erniesat_stat=dump/train/speech_stats.npy \
--voc=hifigan_aishell3 \
--voc_config=hifigan_aishell3_ckpt_0.2.0/default.yaml \
--voc_ckpt=hifigan_aishell3_ckpt_0.2.0/snapshot_iter_2500000.pdz \
--voc_stat=hifigan_aishell3_ckpt_0.2.0/feats_stats.npy \
--output_name=exp/pred_edit.wav
fi
examples/aishell3/ernie_sat/local/train.sh
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...@@ -8,5 +8,5 @@ python3 ${BIN_DIR}/train.py \ ...@@ -8,5 +8,5 @@ python3 ${BIN_DIR}/train.py \
--dev-metadata=dump/dev/norm/metadata.jsonl \ --dev-metadata=dump/dev/norm/metadata.jsonl \
--config=${config_path} \ --config=${config_path} \
--output-dir=${train_output_path} \ --output-dir=${train_output_path} \
--ngpu=2 \ --ngpu=8 \
--phones-dict=dump/phone_id_map.txt --phones-dict=dump/phone_id_map.txt
\ No newline at end of file
examples/aishell3/ernie_sat/run.sh
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...@@ -3,13 +3,13 @@ ...@@ -3,13 +3,13 @@
set -e set -e
source path.sh source path.sh
gpus=0,1 gpus=0,1,2,3,4,5,6,7
stage=0 stage=0
stop_stage=100 stop_stage=100
conf_path=conf/default.yaml conf_path=conf/default.yaml
train_output_path=exp/default train_output_path=exp/default
ckpt_name=snapshot_iter_153.pdz ckpt_name=snapshot_iter_289500.pdz
# with the following command, you can choose the stage range you want to run # with the following command, you can choose the stage range you want to run
# such as `./run.sh --stage 0 --stop-stage 0` # such as `./run.sh --stage 0 --stop-stage 0`
...@@ -30,3 +30,7 @@ if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then ...@@ -30,3 +30,7 @@ if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
# synthesize, vocoder is pwgan # synthesize, vocoder is pwgan
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_path} ${ckpt_name} || exit -1 CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_path} ${ckpt_name} || exit -1
fi fi
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize_e2e.sh ${conf_path} ${train_output_path} ${ckpt_name} || exit -1
fi
examples/aishell3/tts3/README.md
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...@@ -217,7 +217,7 @@ optional arguments: ...@@ -217,7 +217,7 @@ optional arguments:
## Pretrained Model ## Pretrained Model
Pretrained FastSpeech2 model with no silence in the edge of audios: Pretrained FastSpeech2 model with no silence in the edge of audios:
- [fastspeech2_nosil_aishell3_ckpt_0.4.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_aishell3_ckpt_0.4.zip) - [fastspeech2_aishell3_ckpt_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_aishell3_ckpt_1.1.0.zip)
- [fastspeech2_conformer_aishell3_ckpt_0.2.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_conformer_aishell3_ckpt_0.2.0.zip) (Thanks for [@awmmmm](https://github.com/awmmmm)'s contribution) - [fastspeech2_conformer_aishell3_ckpt_0.2.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_conformer_aishell3_ckpt_0.2.0.zip) (Thanks for [@awmmmm](https://github.com/awmmmm)'s contribution)
The static model can be downloaded here: The static model can be downloaded here:
...@@ -229,9 +229,11 @@ The ONNX model can be downloaded here: ...@@ -229,9 +229,11 @@ The ONNX model can be downloaded here:
FastSpeech2 checkpoint contains files listed below. FastSpeech2 checkpoint contains files listed below.
```text ```text
fastspeech2_nosil_aishell3_ckpt_0.4 fastspeech2_aishell3_ckpt_1.1.0
├── default.yaml # default config used to train fastspeech2 ├── default.yaml # default config used to train fastspeech2
├── energy_stats.npy # statistics used to normalize energy when training fastspeech2
├── phone_id_map.txt # phone vocabulary file when training fastspeech2 ├── phone_id_map.txt # phone vocabulary file when training fastspeech2
├── pitch_stats.npy # statistics used to normalize pitch when training fastspeech2
├── snapshot_iter_96400.pdz # model parameters and optimizer states ├── snapshot_iter_96400.pdz # model parameters and optimizer states
├── speaker_id_map.txt # speaker id map file when training a multi-speaker fastspeech2 ├── speaker_id_map.txt # speaker id map file when training a multi-speaker fastspeech2
└── speech_stats.npy # statistics used to normalize spectrogram when training fastspeech2 └── speech_stats.npy # statistics used to normalize spectrogram when training fastspeech2
...@@ -244,9 +246,9 @@ FLAGS_allocator_strategy=naive_best_fit \ ...@@ -244,9 +246,9 @@ FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \ FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/../synthesize_e2e.py \ python3 ${BIN_DIR}/../synthesize_e2e.py \
--am=fastspeech2_aishell3 \ --am=fastspeech2_aishell3 \
--am_config=fastspeech2_nosil_aishell3_ckpt_0.4/default.yaml \ --am_config=fastspeech2_aishell3_ckpt_1.1.0/default.yaml \
--am_ckpt=fastspeech2_nosil_aishell3_ckpt_0.4/snapshot_iter_96400.pdz \ --am_ckpt=fastspeech2_aishell3_ckpt_1.1.0/snapshot_iter_96400.pdz \
--am_stat=fastspeech2_nosil_aishell3_ckpt_0.4/speech_stats.npy \ --am_stat=fastspeech2_aishell3_ckpt_1.1.0/speech_stats.npy \
--voc=pwgan_aishell3 \ --voc=pwgan_aishell3 \
--voc_config=pwg_aishell3_ckpt_0.5/default.yaml \ --voc_config=pwg_aishell3_ckpt_0.5/default.yaml \
--voc_ckpt=pwg_aishell3_ckpt_0.5/snapshot_iter_1000000.pdz \ --voc_ckpt=pwg_aishell3_ckpt_0.5/snapshot_iter_1000000.pdz \
...@@ -254,9 +256,8 @@ python3 ${BIN_DIR}/../synthesize_e2e.py \ ...@@ -254,9 +256,8 @@ python3 ${BIN_DIR}/../synthesize_e2e.py \
--lang=zh \ --lang=zh \
--text=${BIN_DIR}/../sentences.txt \ --text=${BIN_DIR}/../sentences.txt \
--output_dir=exp/default/test_e2e \ --output_dir=exp/default/test_e2e \
--phones_dict=fastspeech2_nosil_aishell3_ckpt_0.4/phone_id_map.txt \ --phones_dict=fastspeech2_aishell3_ckpt_1.1.0/phone_id_map.txt \
--speaker_dict=fastspeech2_nosil_aishell3_ckpt_0.4/speaker_id_map.txt \ --speaker_dict=fastspeech2_aishell3_ckpt_1.1.0/speaker_id_map.txt \
--spk_id=0 \ --spk_id=0 \
--inference_dir=exp/default/inference --inference_dir=exp/default/inference
``` ```
examples/aishell3/tts3/local/synthesize_e2e.sh
浏览文件 @ 11317999
...@@ -38,7 +38,7 @@ if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then ...@@ -38,7 +38,7 @@ if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
--am=fastspeech2_aishell3 \ --am=fastspeech2_aishell3 \
--am_config=${config_path} \ --am_config=${config_path} \
--am_ckpt=${train_output_path}/checkpoints/${ckpt_name} \ --am_ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--am_stat=fastspeech2_nosil_aishell3_ckpt_0.4/speech_stats.npy \ --am_stat=dump/train/speech_stats.npy \
--voc=hifigan_aishell3 \ --voc=hifigan_aishell3 \
--voc_config=hifigan_aishell3_ckpt_0.2.0/default.yaml \ --voc_config=hifigan_aishell3_ckpt_0.2.0/default.yaml \
--voc_ckpt=hifigan_aishell3_ckpt_0.2.0/snapshot_iter_2500000.pdz \ --voc_ckpt=hifigan_aishell3_ckpt_0.2.0/snapshot_iter_2500000.pdz \
...@@ -46,8 +46,8 @@ if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then ...@@ -46,8 +46,8 @@ if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
--lang=zh \ --lang=zh \
--text=${BIN_DIR}/../sentences.txt \ --text=${BIN_DIR}/../sentences.txt \
--output_dir=${train_output_path}/test_e2e \ --output_dir=${train_output_path}/test_e2e \
--phones_dict=fastspeech2_nosil_aishell3_ckpt_0.4/phone_id_map.txt \ --phones_dict=dump/phone_id_map.txt \
--speaker_dict=fastspeech2_nosil_aishell3_ckpt_0.4/speaker_id_map.txt \ --speaker_dict=dump/speaker_id_map.txt \
--spk_id=0 \ --spk_id=0 \
--inference_dir=${train_output_path}/inference --inference_dir=${train_output_path}/inference
fi fi
examples/aishell3/tts3/run.sh
浏览文件 @ 11317999
...@@ -44,8 +44,8 @@ fi ...@@ -44,8 +44,8 @@ fi
if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
# install paddle2onnx # install paddle2onnx
version=$(echo `pip list |grep "paddle2onnx"` |awk -F" " '{print $2}') version=$(echo `pip list |grep "paddle2onnx"` |awk -F" " '{print $2}')
if [[ -z "$version" || ${version} != '0.9.8' ]]; then if [[ -z "$version" || ${version} != '1.0.0' ]]; then
pip install paddle2onnx==0.9.8 pip install paddle2onnx==1.0.0
fi fi
./local/paddle2onnx.sh ${train_output_path} inference inference_onnx fastspeech2_aishell3 ./local/paddle2onnx.sh ${train_output_path} inference inference_onnx fastspeech2_aishell3
# considering the balance between speed and quality, we recommend that you use hifigan as vocoder # considering the balance between speed and quality, we recommend that you use hifigan as vocoder
......
examples/aishell3/vc1/README.md
浏览文件 @ 11317999
...@@ -99,7 +99,7 @@ CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_p ...@@ -99,7 +99,7 @@ CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_p
The synthesizing step is very similar to that one of [tts3](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/tts3), but we should set `--voice-cloning=True` when calling `${BIN_DIR}/../synthesize.py`. The synthesizing step is very similar to that one of [tts3](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/tts3), but we should set `--voice-cloning=True` when calling `${BIN_DIR}/../synthesize.py`.
### Voice Cloning ### Voice Cloning
Assume there are some reference audios in `./ref_audio` Assume there are some reference audios in `./ref_audio`
```text ```text
ref_audio ref_audio
├── 001238.wav ├── 001238.wav
...@@ -116,7 +116,7 @@ CUDA_VISIBLE_DEVICES=${gpus} ./local/voice_cloning.sh ${conf_path} ${train_outpu ...@@ -116,7 +116,7 @@ CUDA_VISIBLE_DEVICES=${gpus} ./local/voice_cloning.sh ${conf_path} ${train_outpu
Model | Step | eval/loss | eval/l1_loss | eval/duration_loss | eval/pitch_loss| eval/energy_loss Model | Step | eval/loss | eval/l1_loss | eval/duration_loss | eval/pitch_loss| eval/energy_loss
:-------------:| :------------:| :-----: | :-----: | :--------: |:--------:|:---------: :-------------:| :------------:| :-----: | :-----: | :--------: |:--------:|:---------:
default|2(gpu) x 96400|0.99699|0.62013|0.53057|0.11954| 0.20426| default|2(gpu) x 96400|0.99699|0.62013|0.053057|0.11954| 0.20426|
FastSpeech2 checkpoint contains files listed below. FastSpeech2 checkpoint contains files listed below.
(There is no need for `speaker_id_map.txt` here ) (There is no need for `speaker_id_map.txt` here )
......
examples/aishell3/vc2/README.md 0 → 100644
浏览文件 @ 11317999
# FastSpeech2 + AISHELL-3 Voice Cloning (ECAPA-TDNN)
This example contains code used to train a [FastSpeech2](https://arxiv.org/abs/2006.04558) model with [AISHELL-3](http://www.aishelltech.com/aishell_3). The trained model can be used in Voice Cloning Task, We refer to the model structure of [Transfer Learning from Speaker Verification to Multispeaker Text-To-Speech Synthesis](https://arxiv.org/pdf/1806.04558.pdf). The general steps are as follows:
1. Speaker Encoder: We use Speaker Verification to train a speaker encoder. Datasets used in this task are different from those used in `FastSpeech2` because the transcriptions are not needed, we use more datasets, refer to [ECAPA-TDNN](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/voxceleb/sv0).
2. Synthesizer: We use the trained speaker encoder to generate speaker embedding for each sentence in AISHELL-3. This embedding is an extra input of `FastSpeech2` which will be concated with encoder outputs.
3. Vocoder: We use [Parallel Wave GAN](http://arxiv.org/abs/1910.11480) as the neural Vocoder, refer to [voc1](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/voc1).
## Dataset
### Download and Extract
Download AISHELL-3 from it's [Official Website](http://www.aishelltech.com/aishell_3) and extract it to `~/datasets`. Then the dataset is in the directory `~/datasets/data_aishell3`.
### Get MFA Result and Extract
We use [MFA2.x](https://github.com/MontrealCorpusTools/Montreal-Forced-Aligner) to get durations for aishell3_fastspeech2.
You can download from here [aishell3_alignment_tone.tar.gz](https://paddlespeech.bj.bcebos.com/MFA/AISHELL-3/with_tone/aishell3_alignment_tone.tar.gz), or train your MFA model reference to [mfa example](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/other/mfa) (use MFA1.x now) of our repo.
## Get Started
Assume the path to the dataset is `~/datasets/data_aishell3`.
Assume the path to the MFA result of AISHELL-3 is `./aishell3_alignment_tone`.
Run the command below to
1. **source path**.
2. preprocess the dataset.
3. train the model.
4. synthesize waveform from `metadata.jsonl`.
5. start a voice cloning inference.
```bash
./run.sh
```
You can choose a range of stages you want to run, or set `stage` equal to `stop-stage` to use only one stage, for example, running the following command will only preprocess the dataset.
```bash
./run.sh --stage 0 --stop-stage 0
```
### Data Preprocessing
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/preprocess.sh ${conf_path}
```
When it is done. A `dump` folder is created in the current directory. The structure of the dump folder is listed below.
```text
dump
├── dev
│ ├── norm
│ └── raw
├── embed
│ ├── SSB0005
│ ├── SSB0009
│ ├── ...
│ └── ...
├── phone_id_map.txt
├── speaker_id_map.txt
├── test
│ ├── norm
│ └── raw
└── train
├── energy_stats.npy
├── norm
├── pitch_stats.npy
├── raw
└── speech_stats.npy
```
The `embed` contains the generated speaker embedding for each sentence in AISHELL-3, which has the same file structure with wav files and the format is `.npy`.
The computing time of utterance embedding can be x hours.
The dataset is split into 3 parts, namely `train`, `dev`, and` test`, each of which contains a `norm` and `raw` subfolder. The raw folder contains speech、pitch and energy features of each utterance, while the norm folder contains normalized ones. The statistics used to normalize features are computed from the training set, which is located in `dump/train/*_stats.npy`.
Also, there is a `metadata.jsonl` in each subfolder. It is a table-like file that contains phones, text_lengths, speech_lengths, durations, the path of speech features, the path of pitch features, the path of energy features, speaker, and id of each utterance.
The preprocessing step is very similar to that one of [tts3](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/tts3), but there is one more `ECAPA-TDNN/inference` step here.
### Model Training
`./local/train.sh` calls `${BIN_DIR}/train.py`.
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/train.sh ${conf_path} ${train_output_path}
```
The training step is very similar to that one of [tts3](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/tts3), but we should set `--voice-cloning=True` when calling `${BIN_DIR}/train.py`.
### Synthesizing
We use [parallel wavegan](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/voc1) as the neural vocoder.
Download pretrained parallel wavegan model from [pwg_aishell3_ckpt_0.5.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/pwgan/pwg_aishell3_ckpt_0.5.zip) and unzip it.
```bash
unzip pwg_aishell3_ckpt_0.5.zip
```
Parallel WaveGAN checkpoint contains files listed below.
```text
pwg_aishell3_ckpt_0.5
├── default.yaml # default config used to train parallel wavegan
├── feats_stats.npy # statistics used to normalize spectrogram when training parallel wavegan
└── snapshot_iter_1000000.pdz # generator parameters of parallel wavegan
```
`./local/synthesize.sh` calls `${BIN_DIR}/../synthesize.py`, which can synthesize waveform from `metadata.jsonl`.
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_path} ${ckpt_name}
```
The synthesizing step is very similar to that one of [tts3](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/tts3), but we should set `--voice-cloning=True` when calling `${BIN_DIR}/../synthesize.py`.
### Voice Cloning
Assume there are some reference audios in `./ref_audio` (the format must be wav here)
```text
ref_audio
├── 001238.wav
├── LJ015-0254.wav
└── audio_self_test.wav
```
`./local/voice_cloning.sh` calls `${BIN_DIR}/../voice_cloning.py`
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/voice_cloning.sh ${conf_path} ${train_output_path} ${ckpt_name} ${ref_audio_dir}
```
## Pretrained Model
- [fastspeech2_aishell3_ckpt_vc2_1.2.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_aishell3_ckpt_vc2_1.2.0.zip)
Model | Step | eval/loss | eval/l1_loss | eval/duration_loss | eval/pitch_loss| eval/energy_loss
:-------------:| :------------:| :-----: | :-----: | :--------: |:--------:|:---------:
default|2(gpu) x 96400|0.991855|0.599517|0.052142|0.094877| 0.245318|
FastSpeech2 checkpoint contains files listed below.
(There is no need for `speaker_id_map.txt` here )
```text
fastspeech2_aishell3_ckpt_vc2_1.2.0
├── default.yaml # default config used to train fastspeech2
├── energy_stats.npy # statistics used to normalize energy when training fastspeech2
├── phone_id_map.txt # phone vocabulary file when training fastspeech2
├── pitch_stats.npy # statistics used to normalize pitch when training fastspeech2
├── snapshot_iter_96400.pdz # model parameters and optimizer states
└── speech_stats.npy # statistics used to normalize spectrogram when training fastspeech2
```
examples/aishell3/vc2/conf/default.yaml 0 → 100644
浏览文件 @ 11317999
###########################################################
# FEATURE EXTRACTION SETTING #
###########################################################
fs: 24000 # sr
n_fft: 2048 # FFT size (samples).
n_shift: 300 # Hop size (samples). 12.5ms
win_length: 1200 # Window length (samples). 50ms
# If set to null, it will be the same as fft_size.
window: "hann" # Window function.
# Only used for feats_type != raw
fmin: 80 # Minimum frequency of Mel basis.
fmax: 7600 # Maximum frequency of Mel basis.
n_mels: 80 # The number of mel basis.
# Only used for the model using pitch features (e.g. FastSpeech2)
f0min: 80 # Minimum f0 for pitch extraction.
f0max: 400 # Maximum f0 for pitch extraction.
###########################################################
# DATA SETTING #
###########################################################
batch_size: 64
num_workers: 2
###########################################################
# MODEL SETTING #
###########################################################
model:
adim: 384 # attention dimension
aheads: 2 # number of attention heads
elayers: 4 # number of encoder layers
eunits: 1536 # number of encoder ff units
dlayers: 4 # number of decoder layers
dunits: 1536 # number of decoder ff units
positionwise_layer_type: conv1d # type of position-wise layer
positionwise_conv_kernel_size: 3 # kernel size of position wise conv layer
duration_predictor_layers: 2 # number of layers of duration predictor
duration_predictor_chans: 256 # number of channels of duration predictor
duration_predictor_kernel_size: 3 # filter size of duration predictor
postnet_layers: 5 # number of layers of postnset
postnet_filts: 5 # filter size of conv layers in postnet
postnet_chans: 256 # number of channels of conv layers in postnet
use_scaled_pos_enc: True # whether to use scaled positional encoding
encoder_normalize_before: True # whether to perform layer normalization before the input
decoder_normalize_before: True # whether to perform layer normalization before the input
reduction_factor: 1 # reduction factor
init_type: xavier_uniform # initialization type
init_enc_alpha: 1.0 # initial value of alpha of encoder scaled position encoding
init_dec_alpha: 1.0 # initial value of alpha of decoder scaled position encoding
transformer_enc_dropout_rate: 0.2 # dropout rate for transformer encoder layer
transformer_enc_positional_dropout_rate: 0.2 # dropout rate for transformer encoder positional encoding
transformer_enc_attn_dropout_rate: 0.2 # dropout rate for transformer encoder attention layer
transformer_dec_dropout_rate: 0.2 # dropout rate for transformer decoder layer
transformer_dec_positional_dropout_rate: 0.2 # dropout rate for transformer decoder positional encoding
transformer_dec_attn_dropout_rate: 0.2 # dropout rate for transformer decoder attention layer
pitch_predictor_layers: 5 # number of conv layers in pitch predictor
pitch_predictor_chans: 256 # number of channels of conv layers in pitch predictor
pitch_predictor_kernel_size: 5 # kernel size of conv leyers in pitch predictor
pitch_predictor_dropout: 0.5 # dropout rate in pitch predictor
pitch_embed_kernel_size: 1 # kernel size of conv embedding layer for pitch
pitch_embed_dropout: 0.0 # dropout rate after conv embedding layer for pitch
stop_gradient_from_pitch_predictor: True # whether to stop the gradient from pitch predictor to encoder
energy_predictor_layers: 2 # number of conv layers in energy predictor
energy_predictor_chans: 256 # number of channels of conv layers in energy predictor
energy_predictor_kernel_size: 3 # kernel size of conv leyers in energy predictor
energy_predictor_dropout: 0.5 # dropout rate in energy predictor
energy_embed_kernel_size: 1 # kernel size of conv embedding layer for energy
energy_embed_dropout: 0.0 # dropout rate after conv embedding layer for energy
stop_gradient_from_energy_predictor: False # whether to stop the gradient from energy predictor to encoder
spk_embed_dim: 192 # speaker embedding dimension
spk_embed_integration_type: concat # speaker embedding integration type
###########################################################
# UPDATER SETTING #
###########################################################
updater:
use_masking: True # whether to apply masking for padded part in loss calculation
###########################################################
# OPTIMIZER SETTING #
###########################################################
optimizer:
optim: adam # optimizer type
learning_rate: 0.001 # learning rate
###########################################################
# TRAINING SETTING #
###########################################################
max_epoch: 200
num_snapshots: 5
###########################################################
# OTHER SETTING #
###########################################################
seed: 10086
examples/aishell3/vc2/local/preprocess.sh 0 → 100755
浏览文件 @ 11317999
#!/bin/bash
stage=0
stop_stage=100
config_path=$1
# gen speaker embedding
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
python3 ${BIN_DIR}/vc2_infer.py \
--input=~/datasets/data_aishell3/train/wav/ \
--output=dump/embed \
--num-cpu=20
fi
# copy from tts3/preprocess
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
# get durations from MFA's result
echo "Generate durations.txt from MFA results ..."
python3 ${MAIN_ROOT}/utils/gen_duration_from_textgrid.py \
--inputdir=./aishell3_alignment_tone \
--output durations.txt \
--config=${config_path}
fi
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
# extract features
echo "Extract features ..."
python3 ${BIN_DIR}/preprocess.py \
--dataset=aishell3 \
--rootdir=~/datasets/data_aishell3/ \
--dumpdir=dump \
--dur-file=durations.txt \
--config=${config_path} \
--num-cpu=20 \
--cut-sil=True \
--spk_emb_dir=dump/embed
fi
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
# get features' stats(mean and std)
echo "Get features' stats ..."
python3 ${MAIN_ROOT}/utils/compute_statistics.py \
--metadata=dump/train/raw/metadata.jsonl \
--field-name="speech"
python3 ${MAIN_ROOT}/utils/compute_statistics.py \
--metadata=dump/train/raw/metadata.jsonl \
--field-name="pitch"
python3 ${MAIN_ROOT}/utils/compute_statistics.py \
--metadata=dump/train/raw/metadata.jsonl \
--field-name="energy"
fi
if [ ${stage} -le 4 ] && [ ${stop_stage} -ge 4 ]; then
# normalize and covert phone/speaker to id, dev and test should use train's stats
echo "Normalize ..."
python3 ${BIN_DIR}/normalize.py \
--metadata=dump/train/raw/metadata.jsonl \
--dumpdir=dump/train/norm \
--speech-stats=dump/train/speech_stats.npy \
--pitch-stats=dump/train/pitch_stats.npy \
--energy-stats=dump/train/energy_stats.npy \
--phones-dict=dump/phone_id_map.txt \
--speaker-dict=dump/speaker_id_map.txt
python3 ${BIN_DIR}/normalize.py \
--metadata=dump/dev/raw/metadata.jsonl \
--dumpdir=dump/dev/norm \
--speech-stats=dump/train/speech_stats.npy \
--pitch-stats=dump/train/pitch_stats.npy \
--energy-stats=dump/train/energy_stats.npy \
--phones-dict=dump/phone_id_map.txt \
--speaker-dict=dump/speaker_id_map.txt
python3 ${BIN_DIR}/normalize.py \
--metadata=dump/test/raw/metadata.jsonl \
--dumpdir=dump/test/norm \
--speech-stats=dump/train/speech_stats.npy \
--pitch-stats=dump/train/pitch_stats.npy \
--energy-stats=dump/train/energy_stats.npy \
--phones-dict=dump/phone_id_map.txt \
--speaker-dict=dump/speaker_id_map.txt
fi
examples/aishell3/vc2/local/synthesize.sh 0 → 100755
浏览文件 @ 11317999
#!/bin/bash
config_path=$1
train_output_path=$2
ckpt_name=$3
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/../synthesize.py \
--am=fastspeech2_aishell3 \
--am_config=${config_path} \
--am_ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--am_stat=dump/train/speech_stats.npy \
--voc=pwgan_aishell3 \
--voc_config=pwg_aishell3_ckpt_0.5/default.yaml \
--voc_ckpt=pwg_aishell3_ckpt_0.5/snapshot_iter_1000000.pdz \
--voc_stat=pwg_aishell3_ckpt_0.5/feats_stats.npy \
--test_metadata=dump/test/norm/metadata.jsonl \
--output_dir=${train_output_path}/test \
--phones_dict=dump/phone_id_map.txt \
--speaker_dict=dump/speaker_id_map.txt \
--voice-cloning=True
examples/aishell3/vc2/local/train.sh 0 → 100755
浏览文件 @ 11317999
#!/bin/bash
config_path=$1
train_output_path=$2
python3 ${BIN_DIR}/train.py \
--train-metadata=dump/train/norm/metadata.jsonl \
--dev-metadata=dump/dev/norm/metadata.jsonl \
--config=${config_path} \
--output-dir=${train_output_path} \
--ngpu=2 \
--phones-dict=dump/phone_id_map.txt \
--voice-cloning=True
\ No newline at end of file
examples/aishell3/vc2/local/voice_cloning.sh 0 → 100755
浏览文件 @ 11317999
#!/bin/bash
config_path=$1
train_output_path=$2
ckpt_name=$3
ref_audio_dir=$4
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/../voice_cloning.py \
--am=fastspeech2_aishell3 \
--am_config=${config_path} \
--am_ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--am_stat=dump/train/speech_stats.npy \
--voc=pwgan_aishell3 \
--voc_config=pwg_aishell3_ckpt_0.5/default.yaml \
--voc_ckpt=pwg_aishell3_ckpt_0.5/snapshot_iter_1000000.pdz \
--voc_stat=pwg_aishell3_ckpt_0.5/feats_stats.npy \
--text="凯莫瑞安联合体的经济崩溃迫在眉睫。" \
--input-dir=${ref_audio_dir} \
--output-dir=${train_output_path}/vc_syn \
--phones-dict=dump/phone_id_map.txt \
--use_ecapa=True
examples/aishell3/vc2/path.sh 0 → 100755
浏览文件 @ 11317999
#!/bin/bash
export MAIN_ROOT=`realpath ${PWD}/../../../`
export PATH=${MAIN_ROOT}:${MAIN_ROOT}/utils:${PATH}
export LC_ALL=C
export PYTHONDONTWRITEBYTECODE=1
# Use UTF-8 in Python to avoid UnicodeDecodeError when LC_ALL=C
export PYTHONIOENCODING=UTF-8
export PYTHONPATH=${MAIN_ROOT}:${PYTHONPATH}
MODEL=fastspeech2
export BIN_DIR=${MAIN_ROOT}/paddlespeech/t2s/exps/${MODEL}
examples/aishell3/vc2/run.sh 0 → 100755
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#!/bin/bash
set -e
source path.sh
gpus=0,1
stage=0
stop_stage=100
conf_path=conf/default.yaml
train_output_path=exp/default
ckpt_name=snapshot_iter_96400.pdz
ref_audio_dir=ref_audio
# with the following command, you can choose the stage range you want to run
# such as `./run.sh --stage 0 --stop-stage 0`
# this can not be mixed use with `$1`, `$2` ...
source ${MAIN_ROOT}/utils/parse_options.sh || exit 1
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
# prepare data
CUDA_VISIBLE_DEVICES=${gpus} ./local/preprocess.sh ${conf_path} || exit -1
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
# train model, all `ckpt` under `train_output_path/checkpoints/` dir
CUDA_VISIBLE_DEVICES=${gpus} ./local/train.sh ${conf_path} ${train_output_path} || exit -1
fi
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
# synthesize, vocoder is pwgan
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_path} ${ckpt_name} || exit -1
fi
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
# synthesize, vocoder is pwgan
CUDA_VISIBLE_DEVICES=${gpus} ./local/voice_cloning.sh ${conf_path} ${train_output_path} ${ckpt_name} ${ref_audio_dir} || exit -1
fi
examples/aishell3/vits-vc/README.md 0 → 100644
浏览文件 @ 11317999
# VITS with AISHELL-3
This example contains code used to train a [VITS](https://arxiv.org/abs/2106.06103) model with [AISHELL-3](http://www.aishelltech.com/aishell_3). The trained model can be used in Voice Cloning Task, We refer to the model structure of [Transfer Learning from Speaker Verification to Multispeaker Text-To-Speech Synthesis](https://arxiv.org/pdf/1806.04558.pdf). The general steps are as follows:
1. Speaker Encoder: We use Speaker Verification to train a speaker encoder. Datasets used in this task are different from those used in `VITS` because the transcriptions are not needed, we use more datasets, refer to [ge2e](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/other/ge2e).
2. Synthesizer and Vocoder: We use the trained speaker encoder to generate speaker embedding for each sentence in AISHELL-3. This embedding is an extra input of `VITS` which will be concated with encoder outputs. The vocoder is part of `VITS` due to its special structure.
## Dataset
### Download and Extract
Download AISHELL-3 from it's [Official Website](http://www.aishelltech.com/aishell_3) and extract it to `~/datasets`. Then the dataset is in the directory `~/datasets/data_aishell3`.
### Get MFA Result and Extract
We use [MFA2.x](https://github.com/MontrealCorpusTools/Montreal-Forced-Aligner) to get phonemes for VITS, the durations of MFA are not needed here.
You can download from here [aishell3_alignment_tone.tar.gz](https://paddlespeech.bj.bcebos.com/MFA/AISHELL-3/with_tone/aishell3_alignment_tone.tar.gz), or train your MFA model reference to [mfa example](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/other/mfa) (use MFA1.x now) of our repo.
## Pretrained GE2E Model
We use pretrained GE2E model to generate speaker embedding for each sentence.
Download pretrained GE2E model from here [ge2e_ckpt_0.3.zip](https://bj.bcebos.com/paddlespeech/Parakeet/released_models/ge2e/ge2e_ckpt_0.3.zip), and `unzip` it.
## Get Started
Assume the path to the dataset is `~/datasets/data_aishell3`.
Assume the path to the MFA result of AISHELL-3 is `./aishell3_alignment_tone`.
Assume the path to the pretrained ge2e model is `./ge2e_ckpt_0.3`.
Run the command below to
1. **source path**.
2. preprocess the dataset.
3. train the model.
4. synthesize waveform from `metadata.jsonl`.
5. start a voice cloning inference.
```bash
./run.sh
```
You can choose a range of stages you want to run, or set `stage` equal to `stop-stage` to use only one stage, for example, running the following command will only preprocess the dataset.
```bash
./run.sh --stage 0 --stop-stage 0
```
### Data Preprocessing
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/preprocess.sh ${conf_path} ${ge2e_ckpt_path}
```
When it is done. A `dump` folder is created in the current directory. The structure of the dump folder is listed below.
```text
dump
├── dev
│   ├── norm
│   └── raw
├── embed
│ ├── SSB0005
│ ├── SSB0009
│ ├── ...
│ └── ...
├── phone_id_map.txt
├── speaker_id_map.txt
├── test
│   ├── norm
│   └── raw
└── train
├── feats_stats.npy
├── norm
└── raw
```
The `embed` contains the generated speaker embedding for each sentence in AISHELL-3, which has the same file structure with wav files and the format is `.npy`.
The computing time of utterance embedding can be x hours.
The dataset is split into 3 parts, namely `train`, `dev`, and` test`, each of which contains a `norm` and `raw` subfolder. The raw folder contains wave and linear spectrogram of each utterance, while the norm folder contains normalized ones. The statistics used to normalize features are computed from the training set, which is located in `dump/train/feats_stats.npy`.
Also, there is a `metadata.jsonl` in each subfolder. It is a table-like file that contains phones, text_lengths, feats, feats_lengths, the path of linear spectrogram features, the path of raw waves, speaker, and the id of each utterance.
The preprocessing step is very similar to that one of [vits](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/vits), but there is one more `ge2e/inference` step here.
### Model Training
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/train.sh ${conf_path} ${train_output_path}
```
The training step is very similar to that one of [vits](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/vits), but we should set `--voice-cloning=True` when calling `${BIN_DIR}/train.py`.
### Synthesizing
`./local/synthesize.sh` calls `${BIN_DIR}/synthesize.py`, which can synthesize waveform from `metadata.jsonl`.
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_path} ${ckpt_name}
```
```text
usage: synthesize.py [-h] [--config CONFIG] [--ckpt CKPT]
[--phones_dict PHONES_DICT] [--speaker_dict SPEAKER_DICT]
[--voice-cloning VOICE_CLONING] [--ngpu NGPU]
[--test_metadata TEST_METADATA] [--output_dir OUTPUT_DIR]
Synthesize with VITS
optional arguments:
-h, --help show this help message and exit
--config CONFIG Config of VITS.
--ckpt CKPT Checkpoint file of VITS.
--phones_dict PHONES_DICT
phone vocabulary file.
--speaker_dict SPEAKER_DICT
speaker id map file.
--voice-cloning VOICE_CLONING
whether training voice cloning model.
--ngpu NGPU if ngpu == 0, use cpu.
--test_metadata TEST_METADATA
test metadata.
--output_dir OUTPUT_DIR
output dir.
```
The synthesizing step is very similar to that one of [vits](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/vits), but we should set `--voice-cloning=True` when calling `${BIN_DIR}/../synthesize.py`.
### Voice Cloning
Assume there are some reference audios in `./ref_audio`
```text
ref_audio
├── 001238.wav
├── LJ015-0254.wav
└── audio_self_test.mp3
```
`./local/voice_cloning.sh` calls `${BIN_DIR}/voice_cloning.py`
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/voice_cloning.sh ${conf_path} ${train_output_path} ${ckpt_name} ${ge2e_params_path} ${add_blank} ${ref_audio_dir}
```
If you want to convert a speaker audio file to refered speaker, run:
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/voice_cloning.sh ${conf_path} ${train_output_path} ${ckpt_name} ${ge2e_params_path} ${add_blank} ${ref_audio_dir} ${src_audio_path}
```
<!-- TODO display these after we trained the model -->
<!--
## Pretrained Model
The pretrained model can be downloaded here:
- [vits_vc_aishell3_ckpt_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/vits/vits_vc_aishell3_ckpt_1.1.0.zip) (add_blank=true)
VITS checkpoint contains files listed below.
(There is no need for `speaker_id_map.txt` here )
```text
vits_vc_aishell3_ckpt_1.1.0
├── default.yaml # default config used to train vitx
├── phone_id_map.txt # phone vocabulary file when training vits
└── snapshot_iter_333000.pdz # model parameters and optimizer states
```
ps: This ckpt is not good enough, a better result is training
-->
examples/aishell3/vits-vc/conf/default.yaml 0 → 100644
浏览文件 @ 11317999
# This configuration tested on 4 GPUs (V100) with 32GB GPU
# memory. It takes around 2 weeks to finish the training
# but 100k iters model should generate reasonable results.
###########################################################
# FEATURE EXTRACTION SETTING #
###########################################################
fs: 22050 # sr
n_fft: 1024 # FFT size (samples).
n_shift: 256 # Hop size (samples). 12.5ms
win_length: null # Window length (samples). 50ms
# If set to null, it will be the same as fft_size.
window: "hann" # Window function.
##########################################################
# TTS MODEL SETTING #
##########################################################
model:
# generator related
generator_type: vits_generator
generator_params:
hidden_channels: 192
spk_embed_dim: 256
global_channels: 256
segment_size: 32
text_encoder_attention_heads: 2
text_encoder_ffn_expand: 4
text_encoder_blocks: 6
text_encoder_positionwise_layer_type: "conv1d"
text_encoder_positionwise_conv_kernel_size: 3
text_encoder_positional_encoding_layer_type: "rel_pos"
text_encoder_self_attention_layer_type: "rel_selfattn"
text_encoder_activation_type: "swish"
text_encoder_normalize_before: True
text_encoder_dropout_rate: 0.1
text_encoder_positional_dropout_rate: 0.0
text_encoder_attention_dropout_rate: 0.1
use_macaron_style_in_text_encoder: True
use_conformer_conv_in_text_encoder: False
text_encoder_conformer_kernel_size: -1
decoder_kernel_size: 7
decoder_channels: 512
decoder_upsample_scales: [8, 8, 2, 2]
decoder_upsample_kernel_sizes: [16, 16, 4, 4]
decoder_resblock_kernel_sizes: [3, 7, 11]
decoder_resblock_dilations: [[1, 3, 5], [1, 3, 5], [1, 3, 5]]
use_weight_norm_in_decoder: True
posterior_encoder_kernel_size: 5
posterior_encoder_layers: 16
posterior_encoder_stacks: 1
posterior_encoder_base_dilation: 1
posterior_encoder_dropout_rate: 0.0
use_weight_norm_in_posterior_encoder: True
flow_flows: 4
flow_kernel_size: 5
flow_base_dilation: 1
flow_layers: 4
flow_dropout_rate: 0.0
use_weight_norm_in_flow: True
use_only_mean_in_flow: True
stochastic_duration_predictor_kernel_size: 3
stochastic_duration_predictor_dropout_rate: 0.5
stochastic_duration_predictor_flows: 4
stochastic_duration_predictor_dds_conv_layers: 3
# discriminator related
discriminator_type: hifigan_multi_scale_multi_period_discriminator
discriminator_params:
scales: 1
scale_downsample_pooling: "AvgPool1D"
scale_downsample_pooling_params:
kernel_size: 4
stride: 2
padding: 2
scale_discriminator_params:
in_channels: 1
out_channels: 1
kernel_sizes: [15, 41, 5, 3]
channels: 128
max_downsample_channels: 1024
max_groups: 16
bias: True
downsample_scales: [2, 2, 4, 4, 1]
nonlinear_activation: "leakyrelu"
nonlinear_activation_params:
negative_slope: 0.1
use_weight_norm: True
use_spectral_norm: False
follow_official_norm: False
periods: [2, 3, 5, 7, 11]
period_discriminator_params:
in_channels: 1
out_channels: 1
kernel_sizes: [5, 3]
channels: 32
downsample_scales: [3, 3, 3, 3, 1]
max_downsample_channels: 1024
bias: True
nonlinear_activation: "leakyrelu"
nonlinear_activation_params:
negative_slope: 0.1
use_weight_norm: True
use_spectral_norm: False
# others
sampling_rate: 22050 # needed in the inference for saving wav
cache_generator_outputs: True # whether to cache generator outputs in the training
###########################################################
# LOSS SETTING #
###########################################################
# loss function related
generator_adv_loss_params:
average_by_discriminators: False # whether to average loss value by #discriminators
loss_type: mse # loss type, "mse" or "hinge"
discriminator_adv_loss_params:
average_by_discriminators: False # whether to average loss value by #discriminators
loss_type: mse # loss type, "mse" or "hinge"
feat_match_loss_params:
average_by_discriminators: False # whether to average loss value by #discriminators
average_by_layers: False # whether to average loss value by #layers of each discriminator
include_final_outputs: True # whether to include final outputs for loss calculation
mel_loss_params:
fs: 22050 # must be the same as the training data
fft_size: 1024 # fft points
hop_size: 256 # hop size
win_length: null # window length
window: hann # window type
num_mels: 80 # number of Mel basis
fmin: 0 # minimum frequency for Mel basis
fmax: null # maximum frequency for Mel basis
log_base: null # null represent natural log
###########################################################
# ADVERSARIAL LOSS SETTING #
###########################################################
lambda_adv: 1.0 # loss scaling coefficient for adversarial loss
lambda_mel: 45.0 # loss scaling coefficient for Mel loss
lambda_feat_match: 2.0 # loss scaling coefficient for feat match loss
lambda_dur: 1.0 # loss scaling coefficient for duration loss
lambda_kl: 1.0 # loss scaling coefficient for KL divergence loss
# others
sampling_rate: 22050 # needed in the inference for saving wav
cache_generator_outputs: True # whether to cache generator outputs in the training
###########################################################
# DATA LOADER SETTING #
###########################################################
batch_size: 50 # Batch size.
num_workers: 4 # Number of workers in DataLoader.
##########################################################
# OPTIMIZER & SCHEDULER SETTING #
##########################################################
# optimizer setting for generator
generator_optimizer_params:
beta1: 0.8
beta2: 0.99
epsilon: 1.0e-9
weight_decay: 0.0
generator_scheduler: exponential_decay
generator_scheduler_params:
learning_rate: 2.0e-4
gamma: 0.999875
# optimizer setting for discriminator
discriminator_optimizer_params:
beta1: 0.8
beta2: 0.99
epsilon: 1.0e-9
weight_decay: 0.0
discriminator_scheduler: exponential_decay
discriminator_scheduler_params:
learning_rate: 2.0e-4
gamma: 0.999875
generator_first: False # whether to start updating generator first
##########################################################
# OTHER TRAINING SETTING #
##########################################################
num_snapshots: 10 # max number of snapshots to keep while training
train_max_steps: 350000 # Number of training steps. == total_iters / ngpus, total_iters = 1000000
save_interval_steps: 1000 # Interval steps to save checkpoint.
eval_interval_steps: 250 # Interval steps to evaluate the network.
seed: 777 # random seed number
examples/aishell3/vits-vc/local/preprocess.sh 0 → 100755
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#!/bin/bash
stage=0
stop_stage=100
config_path=$1
add_blank=$2
ge2e_ckpt_path=$3
# gen speaker embedding
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
python3 ${MAIN_ROOT}/paddlespeech/vector/exps/ge2e/inference.py \
--input=~/datasets/data_aishell3/train/wav/ \
--output=dump/embed \
--checkpoint_path=${ge2e_ckpt_path}
fi
# copy from tts3/preprocess
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
# get durations from MFA's result
echo "Generate durations.txt from MFA results ..."
python3 ${MAIN_ROOT}/utils/gen_duration_from_textgrid.py \
--inputdir=./aishell3_alignment_tone \
--output durations.txt \
--config=${config_path}
fi
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
# extract features
echo "Extract features ..."
python3 ${BIN_DIR}/preprocess.py \
--dataset=aishell3 \
--rootdir=~/datasets/data_aishell3/ \
--dumpdir=dump \
--dur-file=durations.txt \
--config=${config_path} \
--num-cpu=20 \
--cut-sil=True \
--spk_emb_dir=dump/embed
fi
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
# get features' stats(mean and std)
echo "Get features' stats ..."
python3 ${MAIN_ROOT}/utils/compute_statistics.py \
--metadata=dump/train/raw/metadata.jsonl \
--field-name="feats"
fi
if [ ${stage} -le 4 ] && [ ${stop_stage} -ge 4 ]; then
# normalize and covert phone/speaker to id, dev and test should use train's stats
echo "Normalize ..."
python3 ${BIN_DIR}/normalize.py \
--metadata=dump/train/raw/metadata.jsonl \
--dumpdir=dump/train/norm \
--feats-stats=dump/train/feats_stats.npy \
--phones-dict=dump/phone_id_map.txt \
--speaker-dict=dump/speaker_id_map.txt \
--add-blank=${add_blank} \
--skip-wav-copy
python3 ${BIN_DIR}/normalize.py \
--metadata=dump/dev/raw/metadata.jsonl \
--dumpdir=dump/dev/norm \
--feats-stats=dump/train/feats_stats.npy \
--phones-dict=dump/phone_id_map.txt \
--speaker-dict=dump/speaker_id_map.txt \
--add-blank=${add_blank} \
--skip-wav-copy
python3 ${BIN_DIR}/normalize.py \
--metadata=dump/test/raw/metadata.jsonl \
--dumpdir=dump/test/norm \
--feats-stats=dump/train/feats_stats.npy \
--phones-dict=dump/phone_id_map.txt \
--speaker-dict=dump/speaker_id_map.txt \
--add-blank=${add_blank} \
--skip-wav-copy
fi
examples/aishell3/vits-vc/local/synthesize.sh 0 → 100755
浏览文件 @ 11317999
#!/bin/bash
config_path=$1
train_output_path=$2
ckpt_name=$3
stage=0
stop_stage=0
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize.py \
--config=${config_path} \
--ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--phones_dict=dump/phone_id_map.txt \
--test_metadata=dump/test/norm/metadata.jsonl \
--output_dir=${train_output_path}/test \
--voice-cloning=True
fi
examples/aishell3/vits-vc/local/train.sh 0 → 100755
浏览文件 @ 11317999
#!/bin/bash
config_path=$1
train_output_path=$2
# install monotonic_align
cd ${MAIN_ROOT}/paddlespeech/t2s/models/vits/monotonic_align
python3 setup.py build_ext --inplace
cd -
python3 ${BIN_DIR}/train.py \
--train-metadata=dump/train/norm/metadata.jsonl \
--dev-metadata=dump/dev/norm/metadata.jsonl \
--config=${config_path} \
--output-dir=${train_output_path} \
--ngpu=4 \
--phones-dict=dump/phone_id_map.txt \
--voice-cloning=True
examples/aishell3/vits-vc/local/voice_cloning.sh 0 → 100755
浏览文件 @ 11317999
#!/bin/bash
config_path=$1
train_output_path=$2
ckpt_name=$3
ge2e_params_path=$4
add_blank=$5
ref_audio_dir=$6
src_audio_path=$7
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/voice_cloning.py \
--config=${config_path} \
--ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--ge2e_params_path=${ge2e_params_path} \
--phones_dict=dump/phone_id_map.txt \
--text="凯莫瑞安联合体的经济崩溃迫在眉睫。" \
--audio-path=${src_audio_path} \
--input-dir=${ref_audio_dir} \
--output-dir=${train_output_path}/vc_syn \
--add-blank=${add_blank}
examples/ernie_sat/path.sh → examples/aishell3/vits-vc/path.sh
浏览文件 @ 11317999
#!/bin/bash #!/bin/bash
export MAIN_ROOT=`realpath ${PWD}/../../` export MAIN_ROOT=`realpath ${PWD}/../../../`
export PATH=${MAIN_ROOT}:${MAIN_ROOT}/utils:${PATH} export PATH=${MAIN_ROOT}:${MAIN_ROOT}/utils:${PATH}
export LC_ALL=C export LC_ALL=C
...@@ -9,5 +9,5 @@ export PYTHONDONTWRITEBYTECODE=1 ...@@ -9,5 +9,5 @@ export PYTHONDONTWRITEBYTECODE=1
export PYTHONIOENCODING=UTF-8 export PYTHONIOENCODING=UTF-8
export PYTHONPATH=${MAIN_ROOT}:${PYTHONPATH} export PYTHONPATH=${MAIN_ROOT}:${PYTHONPATH}
MODEL=ernie_sat MODEL=vits
export BIN_DIR=${MAIN_ROOT}/paddlespeech/t2s/exps/${MODEL} export BIN_DIR=${MAIN_ROOT}/paddlespeech/t2s/exps/${MODEL}
\ No newline at end of file
examples/aishell3/vits-vc/run.sh 0 → 100755
浏览文件 @ 11317999
#!/bin/bash
set -e
source path.sh
gpus=0,1,2,3
stage=0
stop_stage=100
conf_path=conf/default.yaml
train_output_path=exp/default
ckpt_name=snapshot_iter_153.pdz
add_blank=true
ref_audio_dir=ref_audio
src_audio_path=''
# not include ".pdparams" here
ge2e_ckpt_path=./ge2e_ckpt_0.3/step-3000000
# include ".pdparams" here
ge2e_params_path=${ge2e_ckpt_path}.pdparams
# with the following command, you can choose the stage range you want to run
# such as `./run.sh --stage 0 --stop-stage 0`
# this can not be mixed use with `$1`, `$2` ...
source ${MAIN_ROOT}/utils/parse_options.sh || exit 1
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
# prepare data
CUDA_VISIBLE_DEVICES=${gpus} ./local/preprocess.sh ${conf_path} ${add_blank} ${ge2e_ckpt_path} || exit -1
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
# train model, all `ckpt` under `train_output_path/checkpoints/` dir
CUDA_VISIBLE_DEVICES=${gpus} ./local/train.sh ${conf_path} ${train_output_path} || exit -1
fi
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_path} ${ckpt_name} || exit -1
fi
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
CUDA_VISIBLE_DEVICES=${gpus} ./local/voice_cloning.sh ${conf_path} ${train_output_path} ${ckpt_name} \
${ge2e_params_path} ${add_blank} ${ref_audio_dir} ${src_audio_path} || exit -1
fi
examples/aishell3/vits/README.md 0 → 100644
浏览文件 @ 11317999
# VITS with AISHELL-3
This example contains code used to train a [VITS](https://arxiv.org/abs/2106.06103) model with [AISHELL-3](http://www.aishelltech.com/aishell_3).
AISHELL-3 is a large-scale and high-fidelity multi-speaker Mandarin speech corpus that could be used to train multi-speaker Text-to-Speech (TTS) systems.
We use AISHELL-3 to train a multi-speaker VITS model here.
## Dataset
### Download and Extract
Download AISHELL-3 from it's [Official Website](http://www.aishelltech.com/aishell_3) and extract it to `~/datasets`. Then the dataset is in the directory `~/datasets/data_aishell3`.
### Get MFA Result and Extract
We use [MFA2.x](https://github.com/MontrealCorpusTools/Montreal-Forced-Aligner) to get phonemes for VITS, the durations of MFA are not needed here.
You can download from here [aishell3_alignment_tone.tar.gz](https://paddlespeech.bj.bcebos.com/MFA/AISHELL-3/with_tone/aishell3_alignment_tone.tar.gz), or train your MFA model reference to [mfa example](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/other/mfa) (use MFA1.x now) of our repo.
## Get Started
Assume the path to the dataset is `~/datasets/data_aishell3`.
Assume the path to the MFA result of AISHELL-3 is `./aishell3_alignment_tone`.
Run the command below to
1. **source path**.
2. preprocess the dataset.
3. train the model.
4. synthesize wavs.
- synthesize waveform from `metadata.jsonl`.
- synthesize waveform from a text file.
```bash
./run.sh
```
You can choose a range of stages you want to run, or set `stage` equal to `stop-stage` to use only one stage, for example, running the following command will only preprocess the dataset.
```bash
./run.sh --stage 0 --stop-stage 0
```
### Data Preprocessing
```bash
./local/preprocess.sh ${conf_path}
```
When it is done. A `dump` folder is created in the current directory. The structure of the dump folder is listed below.
```text
dump
├── dev
│   ├── norm
│   └── raw
├── phone_id_map.txt
├── speaker_id_map.txt
├── test
│   ├── norm
│   └── raw
└── train
├── feats_stats.npy
├── norm
└── raw
```
The dataset is split into 3 parts, namely `train`, `dev`, and` test`, each of which contains a `norm` and `raw` subfolder. The raw folder contains wave and linear spectrogram of each utterance, while the norm folder contains normalized ones. The statistics used to normalize features are computed from the training set, which is located in `dump/train/feats_stats.npy`.
Also, there is a `metadata.jsonl` in each subfolder. It is a table-like file that contains phones, text_lengths, feats, feats_lengths, the path of linear spectrogram features, the path of raw waves, speaker, and the id of each utterance.
### Model Training
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/train.sh ${conf_path} ${train_output_path}
```
`./local/train.sh` calls `${BIN_DIR}/train.py`.
Here's the complete help message.
```text
usage: train.py [-h] [--config CONFIG] [--train-metadata TRAIN_METADATA]
[--dev-metadata DEV_METADATA] [--output-dir OUTPUT_DIR]
[--ngpu NGPU] [--phones-dict PHONES_DICT]
[--speaker-dict SPEAKER_DICT] [--voice-cloning VOICE_CLONING]
Train a VITS model.
optional arguments:
-h, --help show this help message and exit
--config CONFIG config file to overwrite default config.
--train-metadata TRAIN_METADATA
training data.
--dev-metadata DEV_METADATA
dev data.
--output-dir OUTPUT_DIR
output dir.
--ngpu NGPU if ngpu == 0, use cpu.
--phones-dict PHONES_DICT
phone vocabulary file.
--speaker-dict SPEAKER_DICT
speaker id map file for multiple speaker model.
--voice-cloning VOICE_CLONING
whether training voice cloning model.
```
1. `--config` is a config file in yaml format to overwrite the default config, which can be found at `conf/default.yaml`.
2. `--train-metadata` and `--dev-metadata` should be the metadata file in the normalized subfolder of `train` and `dev` in the `dump` folder.
3. `--output-dir` is the directory to save the results of the experiment. Checkpoints are saved in `checkpoints/` inside this directory.
4. `--ngpu` is the number of gpus to use, if ngpu == 0, use cpu.
5. `--phones-dict` is the path of the phone vocabulary file.
6. `--speaker-dict` is the path of the speaker id map file when training a multi-speaker VITS.
### Synthesizing
`./local/synthesize.sh` calls `${BIN_DIR}/synthesize.py`, which can synthesize waveform from `metadata.jsonl`.
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_path} ${ckpt_name}
```
```text
usage: synthesize.py [-h] [--config CONFIG] [--ckpt CKPT]
[--phones_dict PHONES_DICT] [--speaker_dict SPEAKER_DICT]
[--voice-cloning VOICE_CLONING] [--ngpu NGPU]
[--test_metadata TEST_METADATA] [--output_dir OUTPUT_DIR]
Synthesize with VITS
optional arguments:
-h, --help show this help message and exit
--config CONFIG Config of VITS.
--ckpt CKPT Checkpoint file of VITS.
--phones_dict PHONES_DICT
phone vocabulary file.
--speaker_dict SPEAKER_DICT
speaker id map file.
--voice-cloning VOICE_CLONING
whether training voice cloning model.
--ngpu NGPU if ngpu == 0, use cpu.
--test_metadata TEST_METADATA
test metadata.
--output_dir OUTPUT_DIR
output dir.
```
`./local/synthesize_e2e.sh` calls `${BIN_DIR}/synthesize_e2e.py`, which can synthesize waveform from text file.
```bash
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize_e2e.sh ${conf_path} ${train_output_path} ${ckpt_name}
```
```text
usage: synthesize_e2e.py [-h] [--config CONFIG] [--ckpt CKPT]
[--phones_dict PHONES_DICT]
[--speaker_dict SPEAKER_DICT] [--spk_id SPK_ID]
[--lang LANG]
[--inference_dir INFERENCE_DIR] [--ngpu NGPU]
[--text TEXT] [--output_dir OUTPUT_DIR]
Synthesize with VITS
optional arguments:
-h, --help show this help message and exit
--config CONFIG Config of VITS.
--ckpt CKPT Checkpoint file of VITS.
--phones_dict PHONES_DICT
phone vocabulary file.
--speaker_dict SPEAKER_DICT
speaker id map file.
--spk_id SPK_ID spk id for multi speaker acoustic model
--lang LANG Choose model language. zh or en
--inference_dir INFERENCE_DIR
dir to save inference models
--ngpu NGPU if ngpu == 0, use cpu.
--text TEXT text to synthesize, a 'utt_id sentence' pair per line.
--output_dir OUTPUT_DIR
output dir.
```
1. `--config`, `--ckpt`, `--phones_dict` and `--speaker_dict` are arguments for acoustic model, which correspond to the 3 files in the VITS pretrained model.
2. `--lang` is the model language, which can be `zh` or `en`.
3. `--test_metadata` should be the metadata file in the normalized subfolder of `test` in the `dump` folder.
4. `--text` is the text file, which contains sentences to synthesize.
5. `--output_dir` is the directory to save synthesized audio files.
6. `--ngpu` is the number of gpus to use, if ngpu == 0, use cpu.
<!-- TODO display these after we trained the model -->
<!--
## Pretrained Model
The pretrained model can be downloaded here:
- [vits_aishell3_ckpt_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/vits/vits_aishell3_ckpt_1.1.0.zip) (add_blank=true)
VITS checkpoint contains files listed below.
```text
vits_aishell3_ckpt_1.1.0
├── default.yaml # default config used to train vitx
├── phone_id_map.txt # phone vocabulary file when training vits
├── speaker_id_map.txt # speaker id map file when training a multi-speaker vits
└── snapshot_iter_333000.pdz # model parameters and optimizer states
```
ps: This ckpt is not good enough, a better result is training
You can use the following scripts to synthesize for `${BIN_DIR}/../sentences.txt` using pretrained VITS.
```bash
source path.sh
add_blank=true
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize_e2e.py \
--config=vits_aishell3_ckpt_1.1.0/default.yaml \
--ckpt=vits_aishell3_ckpt_1.1.0/snapshot_iter_333000.pdz \
--phones_dict=vits_aishell3_ckpt_1.1.0/phone_id_map.txt \
--speaker_dict=vits_aishell3_ckpt_1.1.0/speaker_id_map.txt \
--output_dir=exp/default/test_e2e \
--text=${BIN_DIR}/../sentences.txt \
--add-blank=${add_blank}
```
-->
examples/aishell3/vits/conf/default.yaml 0 → 100644
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# This configuration tested on 4 GPUs (V100) with 32GB GPU
# memory. It takes around 2 weeks to finish the training
# but 100k iters model should generate reasonable results.
###########################################################
# FEATURE EXTRACTION SETTING #
###########################################################
fs: 22050 # sr
n_fft: 1024 # FFT size (samples).
n_shift: 256 # Hop size (samples). 12.5ms
win_length: null # Window length (samples). 50ms
# If set to null, it will be the same as fft_size.
window: "hann" # Window function.
##########################################################
# TTS MODEL SETTING #
##########################################################
model:
# generator related
generator_type: vits_generator
generator_params:
hidden_channels: 192
global_channels: 256
segment_size: 32
text_encoder_attention_heads: 2
text_encoder_ffn_expand: 4
text_encoder_blocks: 6
text_encoder_positionwise_layer_type: "conv1d"
text_encoder_positionwise_conv_kernel_size: 3
text_encoder_positional_encoding_layer_type: "rel_pos"
text_encoder_self_attention_layer_type: "rel_selfattn"
text_encoder_activation_type: "swish"
text_encoder_normalize_before: True
text_encoder_dropout_rate: 0.1
text_encoder_positional_dropout_rate: 0.0
text_encoder_attention_dropout_rate: 0.1
use_macaron_style_in_text_encoder: True
use_conformer_conv_in_text_encoder: False
text_encoder_conformer_kernel_size: -1
decoder_kernel_size: 7
decoder_channels: 512
decoder_upsample_scales: [8, 8, 2, 2]
decoder_upsample_kernel_sizes: [16, 16, 4, 4]
decoder_resblock_kernel_sizes: [3, 7, 11]
decoder_resblock_dilations: [[1, 3, 5], [1, 3, 5], [1, 3, 5]]
use_weight_norm_in_decoder: True
posterior_encoder_kernel_size: 5
posterior_encoder_layers: 16
posterior_encoder_stacks: 1
posterior_encoder_base_dilation: 1
posterior_encoder_dropout_rate: 0.0
use_weight_norm_in_posterior_encoder: True
flow_flows: 4
flow_kernel_size: 5
flow_base_dilation: 1
flow_layers: 4
flow_dropout_rate: 0.0
use_weight_norm_in_flow: True
use_only_mean_in_flow: True
stochastic_duration_predictor_kernel_size: 3
stochastic_duration_predictor_dropout_rate: 0.5
stochastic_duration_predictor_flows: 4
stochastic_duration_predictor_dds_conv_layers: 3
# discriminator related
discriminator_type: hifigan_multi_scale_multi_period_discriminator
discriminator_params:
scales: 1
scale_downsample_pooling: "AvgPool1D"
scale_downsample_pooling_params:
kernel_size: 4
stride: 2
padding: 2
scale_discriminator_params:
in_channels: 1
out_channels: 1
kernel_sizes: [15, 41, 5, 3]
channels: 128
max_downsample_channels: 1024
max_groups: 16
bias: True
downsample_scales: [2, 2, 4, 4, 1]
nonlinear_activation: "leakyrelu"
nonlinear_activation_params:
negative_slope: 0.1
use_weight_norm: True
use_spectral_norm: False
follow_official_norm: False
periods: [2, 3, 5, 7, 11]
period_discriminator_params:
in_channels: 1
out_channels: 1
kernel_sizes: [5, 3]
channels: 32
downsample_scales: [3, 3, 3, 3, 1]
max_downsample_channels: 1024
bias: True
nonlinear_activation: "leakyrelu"
nonlinear_activation_params:
negative_slope: 0.1
use_weight_norm: True
use_spectral_norm: False
# others
sampling_rate: 22050 # needed in the inference for saving wav
cache_generator_outputs: True # whether to cache generator outputs in the training
###########################################################
# LOSS SETTING #
###########################################################
# loss function related
generator_adv_loss_params:
average_by_discriminators: False # whether to average loss value by #discriminators
loss_type: mse # loss type, "mse" or "hinge"
discriminator_adv_loss_params:
average_by_discriminators: False # whether to average loss value by #discriminators
loss_type: mse # loss type, "mse" or "hinge"
feat_match_loss_params:
average_by_discriminators: False # whether to average loss value by #discriminators
average_by_layers: False # whether to average loss value by #layers of each discriminator
include_final_outputs: True # whether to include final outputs for loss calculation
mel_loss_params:
fs: 22050 # must be the same as the training data
fft_size: 1024 # fft points
hop_size: 256 # hop size
win_length: null # window length
window: hann # window type
num_mels: 80 # number of Mel basis
fmin: 0 # minimum frequency for Mel basis
fmax: null # maximum frequency for Mel basis
log_base: null # null represent natural log
###########################################################
# ADVERSARIAL LOSS SETTING #
###########################################################
lambda_adv: 1.0 # loss scaling coefficient for adversarial loss
lambda_mel: 45.0 # loss scaling coefficient for Mel loss
lambda_feat_match: 2.0 # loss scaling coefficient for feat match loss
lambda_dur: 1.0 # loss scaling coefficient for duration loss
lambda_kl: 1.0 # loss scaling coefficient for KL divergence loss
# others
sampling_rate: 22050 # needed in the inference for saving wav
cache_generator_outputs: True # whether to cache generator outputs in the training
###########################################################
# DATA LOADER SETTING #
###########################################################
batch_size: 50 # Batch size.
num_workers: 4 # Number of workers in DataLoader.
##########################################################
# OPTIMIZER & SCHEDULER SETTING #
##########################################################
# optimizer setting for generator
generator_optimizer_params:
beta1: 0.8
beta2: 0.99
epsilon: 1.0e-9
weight_decay: 0.0
generator_scheduler: exponential_decay
generator_scheduler_params:
learning_rate: 2.0e-4
gamma: 0.999875
# optimizer setting for discriminator
discriminator_optimizer_params:
beta1: 0.8
beta2: 0.99
epsilon: 1.0e-9
weight_decay: 0.0
discriminator_scheduler: exponential_decay
discriminator_scheduler_params:
learning_rate: 2.0e-4
gamma: 0.999875
generator_first: False # whether to start updating generator first
##########################################################
# OTHER TRAINING SETTING #
##########################################################
num_snapshots: 10 # max number of snapshots to keep while training
train_max_steps: 350000 # Number of training steps. == total_iters / ngpus, total_iters = 1000000
save_interval_steps: 1000 # Interval steps to save checkpoint.
eval_interval_steps: 250 # Interval steps to evaluate the network.
seed: 777 # random seed number
examples/aishell3/vits/local/preprocess.sh 0 → 100755
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#!/bin/bash
stage=0
stop_stage=100
config_path=$1
add_blank=$2
# copy from tts3/preprocess
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
# get durations from MFA's result
echo "Generate durations.txt from MFA results ..."
python3 ${MAIN_ROOT}/utils/gen_duration_from_textgrid.py \
--inputdir=./aishell3_alignment_tone \
--output durations.txt \
--config=${config_path}
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
# extract features
echo "Extract features ..."
python3 ${BIN_DIR}/preprocess.py \
--dataset=aishell3 \
--rootdir=~/datasets/data_aishell3/ \
--dumpdir=dump \
--dur-file=durations.txt \
--config=${config_path} \
--num-cpu=20 \
--cut-sil=True
fi
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
# get features' stats(mean and std)
echo "Get features' stats ..."
python3 ${MAIN_ROOT}/utils/compute_statistics.py \
--metadata=dump/train/raw/metadata.jsonl \
--field-name="feats"
fi
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
# normalize and covert phone/speaker to id, dev and test should use train's stats
echo "Normalize ..."
python3 ${BIN_DIR}/normalize.py \
--metadata=dump/train/raw/metadata.jsonl \
--dumpdir=dump/train/norm \
--feats-stats=dump/train/feats_stats.npy \
--phones-dict=dump/phone_id_map.txt \
--speaker-dict=dump/speaker_id_map.txt \
--add-blank=${add_blank} \
--skip-wav-copy
python3 ${BIN_DIR}/normalize.py \
--metadata=dump/dev/raw/metadata.jsonl \
--dumpdir=dump/dev/norm \
--feats-stats=dump/train/feats_stats.npy \
--phones-dict=dump/phone_id_map.txt \
--speaker-dict=dump/speaker_id_map.txt \
--add-blank=${add_blank} \
--skip-wav-copy
python3 ${BIN_DIR}/normalize.py \
--metadata=dump/test/raw/metadata.jsonl \
--dumpdir=dump/test/norm \
--feats-stats=dump/train/feats_stats.npy \
--phones-dict=dump/phone_id_map.txt \
--speaker-dict=dump/speaker_id_map.txt \
--add-blank=${add_blank} \
--skip-wav-copy
fi
examples/aishell3/vits/local/synthesize.sh 0 → 100755
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#!/bin/bash
config_path=$1
train_output_path=$2
ckpt_name=$3
stage=0
stop_stage=0
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize.py \
--config=${config_path} \
--ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--phones_dict=dump/phone_id_map.txt \
--speaker_dict=dump/speaker_id_map.txt \
--test_metadata=dump/test/norm/metadata.jsonl \
--output_dir=${train_output_path}/test
fi
examples/aishell3/vits/local/synthesize_e2e.sh 0 → 100755
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#!/bin/bash
config_path=$1
train_output_path=$2
ckpt_name=$3
add_blank=$4
stage=0
stop_stage=0
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize_e2e.py \
--config=${config_path} \
--ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--phones_dict=dump/phone_id_map.txt \
--speaker_dict=dump/speaker_id_map.txt \
--spk_id=0 \
--output_dir=${train_output_path}/test_e2e \
--text=${BIN_DIR}/../sentences.txt \
--add-blank=${add_blank}
fi
examples/aishell3/vits/local/train.sh 0 → 100755
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#!/bin/bash
config_path=$1
train_output_path=$2
# install monotonic_align
cd ${MAIN_ROOT}/paddlespeech/t2s/models/vits/monotonic_align
python3 setup.py build_ext --inplace
cd -
python3 ${BIN_DIR}/train.py \
--train-metadata=dump/train/norm/metadata.jsonl \
--dev-metadata=dump/dev/norm/metadata.jsonl \
--config=${config_path} \
--output-dir=${train_output_path} \
--ngpu=4 \
--phones-dict=dump/phone_id_map.txt \
--speaker-dict=dump/speaker_id_map.txt
examples/aishell3/vits/path.sh 0 → 100755
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#!/bin/bash
export MAIN_ROOT=`realpath ${PWD}/../../../`
export PATH=${MAIN_ROOT}:${MAIN_ROOT}/utils:${PATH}
export LC_ALL=C
export PYTHONDONTWRITEBYTECODE=1
# Use UTF-8 in Python to avoid UnicodeDecodeError when LC_ALL=C
export PYTHONIOENCODING=UTF-8
export PYTHONPATH=${MAIN_ROOT}:${PYTHONPATH}
MODEL=vits
export BIN_DIR=${MAIN_ROOT}/paddlespeech/t2s/exps/${MODEL}
\ No newline at end of file
examples/aishell3/vits/run.sh 0 → 100755
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#!/bin/bash
set -e
source path.sh
gpus=0,1,2,3
stage=0
stop_stage=100
conf_path=conf/default.yaml
train_output_path=exp/default
ckpt_name=snapshot_iter_153.pdz
add_blank=true
# with the following command, you can choose the stage range you want to run
# such as `./run.sh --stage 0 --stop-stage 0`
# this can not be mixed use with `$1`, `$2` ...
source ${MAIN_ROOT}/utils/parse_options.sh || exit 1
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
# prepare data
./local/preprocess.sh ${conf_path} ${add_blank}|| exit -1
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
# train model, all `ckpt` under `train_output_path/checkpoints/` dir
CUDA_VISIBLE_DEVICES=${gpus} ./local/train.sh ${conf_path} ${train_output_path} || exit -1
fi
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_path} ${ckpt_name} || exit -1
fi
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize_e2e.sh ${conf_path} ${train_output_path} ${ckpt_name} ${add_blank}|| exit -1
fi
examples/aishell3_vctk/README.md
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# Mixed Chinese and English TTS with AISHELL3 and VCTK datasets # Mixed Chinese and English TTS with AISHELL3 and VCTK datasets
* ernie_sat - ERNIE-SAT
examples/aishell3_vctk/ernie_sat/conf/default.yaml
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# This configuration tested on 8 GPUs (A100) with 80GB GPU memory.
# It takes around 4 days to finish the training,You can adjust
# batch_size、num_workers here and ngpu in local/train.sh for your machine
########################################################### ###########################################################
# FEATURE EXTRACTION SETTING # # FEATURE EXTRACTION SETTING #
########################################################### ###########################################################
...@@ -21,8 +24,8 @@ mlm_prob: 0.8 ...@@ -21,8 +24,8 @@ mlm_prob: 0.8
########################################################### ###########################################################
# DATA SETTING # # DATA SETTING #
########################################################### ###########################################################
batch_size: 20 batch_size: 40
num_workers: 2 num_workers: 8
########################################################### ###########################################################
# MODEL SETTING # # MODEL SETTING #
...@@ -79,7 +82,7 @@ grad_clip: 1.0 ...@@ -79,7 +82,7 @@ grad_clip: 1.0
########################################################### ###########################################################
# TRAINING SETTING # # TRAINING SETTING #
########################################################### ###########################################################
max_epoch: 700 max_epoch: 1500
num_snapshots: 50 num_snapshots: 50
########################################################### ###########################################################
......
examples/aishell3_vctk/ernie_sat/local/synthesize.sh
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...@@ -4,28 +4,11 @@ config_path=$1 ...@@ -4,28 +4,11 @@ config_path=$1
train_output_path=$2 train_output_path=$2
ckpt_name=$3 ckpt_name=$3
stage=1 stage=0
stop_stage=1 stop_stage=0
# pwgan
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize.py \
--erniesat_config=${config_path} \
--erniesat_ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--erniesat_stat=dump/train/speech_stats.npy \
--voc=pwgan_aishell3 \
--voc_config=pwg_aishell3_ckpt_0.5/default.yaml \
--voc_ckpt=pwg_aishell3_ckpt_0.5/snapshot_iter_1000000.pdz \
--voc_stat=pwg_aishell3_ckpt_0.5/feats_stats.npy \
--test_metadata=dump/test/norm/metadata.jsonl \
--output_dir=${train_output_path}/test \
--phones_dict=dump/phone_id_map.txt
fi
# hifigan # hifigan
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
FLAGS_allocator_strategy=naive_best_fit \ FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \ FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize.py \ python3 ${BIN_DIR}/synthesize.py \
......
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# not ready yet
#!/bin/bash
config_path=$1
train_output_path=$2
ckpt_name=$3
stage=0
stop_stage=1
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
echo 'speech cross language from en to zh !'
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize_e2e.py \
--task_name=synthesize \
--wav_path=source/p243_313.wav \
--old_str='For that reason cover should not be given' \
--new_str='今天天气很好' \
--source_lang=en \
--target_lang=zh \
--erniesat_config=${config_path} \
--phones_dict=dump/phone_id_map.txt \
--erniesat_ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--erniesat_stat=dump/train/speech_stats.npy \
--voc=hifigan_aishell3 \
--voc_config=hifigan_aishell3_ckpt_0.2.0/default.yaml \
--voc_ckpt=hifigan_aishell3_ckpt_0.2.0/snapshot_iter_2500000.pdz \
--voc_stat=hifigan_aishell3_ckpt_0.2.0/feats_stats.npy \
--output_name=exp/pred_clone_en_zh.wav
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
echo 'speech cross language from zh to en !'
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize_e2e.py \
--task_name=synthesize \
--wav_path=source/SSB03540307.wav \
--old_str='请播放歌曲小苹果' \
--new_str="Thank you" \
--source_lang=zh \
--target_lang=en \
--erniesat_config=${config_path} \
--phones_dict=dump/phone_id_map.txt \
--erniesat_ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--erniesat_stat=dump/train/speech_stats.npy \
--voc=hifigan_aishell3 \
--voc_config=hifigan_aishell3_ckpt_0.2.0/default.yaml \
--voc_ckpt=hifigan_aishell3_ckpt_0.2.0/snapshot_iter_2500000.pdz \
--voc_stat=hifigan_aishell3_ckpt_0.2.0/feats_stats.npy \
--output_name=exp/pred_clone_zh_en.wav
fi
examples/aishell3_vctk/ernie_sat/local/train.sh
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...@@ -8,5 +8,5 @@ python3 ${BIN_DIR}/train.py \ ...@@ -8,5 +8,5 @@ python3 ${BIN_DIR}/train.py \
--dev-metadata=dump/dev/norm/metadata.jsonl \ --dev-metadata=dump/dev/norm/metadata.jsonl \
--config=${config_path} \ --config=${config_path} \
--output-dir=${train_output_path} \ --output-dir=${train_output_path} \
--ngpu=2 \ --ngpu=8 \
--phones-dict=dump/phone_id_map.txt --phones-dict=dump/phone_id_map.txt
\ No newline at end of file
examples/aishell3_vctk/ernie_sat/run.sh
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...@@ -3,13 +3,13 @@ ...@@ -3,13 +3,13 @@
set -e set -e
source path.sh source path.sh
gpus=0,1 gpus=0,1,2,3,4,5,6,7
stage=0 stage=0
stop_stage=100 stop_stage=100
conf_path=conf/default.yaml conf_path=conf/default.yaml
train_output_path=exp/default train_output_path=exp/default
ckpt_name=snapshot_iter_153.pdz ckpt_name=snapshot_iter_489000.pdz
# with the following command, you can choose the stage range you want to run # with the following command, you can choose the stage range you want to run
# such as `./run.sh --stage 0 --stop-stage 0` # such as `./run.sh --stage 0 --stop-stage 0`
...@@ -30,3 +30,7 @@ if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then ...@@ -30,3 +30,7 @@ if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
# synthesize, vocoder is pwgan # synthesize, vocoder is pwgan
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_path} ${ckpt_name} || exit -1 CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_path} ${ckpt_name} || exit -1
fi fi
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize_e2e.sh ${conf_path} ${train_output_path} ${ckpt_name} || exit -1
fi
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