Merge branch 'PaddlePaddle:develop' into develop

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liangym 3 years ago committed by GitHub
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5
.github/stale.yml vendored

@ -6,7 +6,8 @@ daysUntilClose: 30
exemptLabels:
- Roadmap
- Bug
- New Feature
- feature request
- Tips
# Label to use when marking an issue as stale
staleLabel: Stale
# Comment to post when marking an issue as stale. Set to `false` to disable
@ -17,4 +18,4 @@ markComment: >
unmarkComment: false
# Comment to post when closing a stale issue. Set to `false` to disable
closeComment: >
This issue is closed. Please re-open if needed.
This issue is closed. Please re-open if needed.

@ -97,26 +97,40 @@
</thead>
<tbody>
<tr>
<td >Life was like a box of chocolates, you never know what you're gonna get.</td>
<td>Life was like a box of chocolates, you never know what you're gonna get.</td>
<td align = "center">
<a href="https://paddlespeech.bj.bcebos.com/Parakeet/docs/demos/tacotron2_ljspeech_waveflow_samples_0.2/sentence_1.wav" rel="nofollow">
<img align="center" src="./docs/images/audio_icon.png" width="200" style="max-width: 100%;"></a><br>
</td>
</tr>
<tr>
<td >早上好今天是2020/10/29最低温度是-3°C。</td>
<td>早上好今天是2020/10/29最低温度是-3°C。</td>
<td align = "center">
<a href="https://paddlespeech.bj.bcebos.com/Parakeet/docs/demos/parakeet_espnet_fs2_pwg_demo/tn_g2p/parakeet/001.wav" rel="nofollow">
<img align="center" src="./docs/images/audio_icon.png" width="200" style="max-width: 100%;"></a><br>
</td>
</tr>
<tr>
<td >季姬寂,集鸡,鸡即棘鸡。棘鸡饥叽,季姬及箕稷济鸡。鸡既济,跻姬笈,季姬忌,急咭鸡,鸡急,继圾几,季姬急,即籍箕击鸡,箕疾击几伎,伎即齑,鸡叽集几基,季姬急极屐击鸡,鸡既殛,季姬激,即记《季姬击鸡记》。</td>
<td>季姬寂,集鸡,鸡即棘鸡。棘鸡饥叽,季姬及箕稷济鸡。鸡既济,跻姬笈,季姬忌,急咭鸡,鸡急,继圾几,季姬急,即籍箕击鸡,箕疾击几伎,伎即齑,鸡叽集几基,季姬急极屐击鸡,鸡既殛,季姬激,即记《季姬击鸡记》。</td>
<td align = "center">
<a href="https://paddlespeech.bj.bcebos.com/Parakeet/docs/demos/jijiji.wav" rel="nofollow">
<img align="center" src="./docs/images/audio_icon.png" width="200" style="max-width: 100%;"></a><br>
</td>
</tr>
<tr>
<td>大家好,我是 parrot 虚拟老师我们来读一首诗我与春风皆过客I and the spring breeze are passing by你携秋水揽星河you take the autumn water to take the galaxy。</td>
<td align = "center">
<a href="https://paddlespeech.bj.bcebos.com/Parakeet/docs/demos/labixiaoxin.wav" rel="nofollow">
<img align="center" src="./docs/images/audio_icon.png" width="200" style="max-width: 100%;"></a><br>
</td>
</tr>
<tr>
<td>宜家唔系事必要你讲,但系你所讲嘅说话将会变成呈堂证供。</td>
<td align = "center">
<a href="https://paddlespeech.bj.bcebos.com/Parakeet/docs/demos/chengtangzhenggong.wav" rel="nofollow">
<img align="center" src="./docs/images/audio_icon.png" width="200" style="max-width: 100%;"></a><br>
</td>
</tr>
</tbody>
</table>
@ -157,18 +171,19 @@ Via the easy-to-use, efficient, flexible and scalable implementation, our vision
- 🧩 *Cascaded models application*: as an extension of the typical traditional audio tasks, we combine the workflows of the aforementioned tasks with other fields like Natural language processing (NLP) and Computer Vision (CV).
### Recent Update
- 🔥 2022.01.10: Add [code-switch asr CLI and Demos](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/demos/speech_recognition).
- 👑 2022.01.06: Add [code-switch asr tal_cs recipe](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/tal_cs/asr1/).
- 🎉 2022.12.02: Add [end-to-end Prosody Prediction pipeline](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/csmsc/tts3_rhy) (including using prosody labels in Acoustic Model).
- 🎉 2022.11.30: Add [TTS Android Demo](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/demos/TTSAndroid).
- 🎉 2023.02.16: Add [Cantonese TTS](./examples/canton/tts3).
- 🔥 2023.01.10: Add [code-switch asr CLI and Demos](./demos/speech_recognition).
- 👑 2023.01.06: Add [code-switch asr tal_cs recipe](./examples/tal_cs/asr1/).
- 🎉 2022.12.02: Add [end-to-end Prosody Prediction pipeline](./examples/csmsc/tts3_rhy) (including using prosody labels in Acoustic Model).
- 🎉 2022.11.30: Add [TTS Android Demo](./demos/TTSAndroid).
- 🤗 2022.11.28: PP-TTS and PP-ASR demos are available in [AIStudio](https://aistudio.baidu.com/aistudio/modelsoverview) and [official website
of paddlepaddle](https://www.paddlepaddle.org.cn/models).
- 👑 2022.11.18: Add [Whisper CLI and Demos](https://github.com/PaddlePaddle/PaddleSpeech/pull/2640), support multi language recognition and translation.
- 🔥 2022.11.18: Add [Wav2vec2 CLI and Demos](https://github.com/PaddlePaddle/PaddleSpeech/blob/develop/demos/speech_ssl), Support ASR and Feature Extraction.
- 🔥 2022.11.18: Add [Wav2vec2 CLI and Demos](./demos/speech_ssl), Support ASR and Feature Extraction.
- 🎉 2022.11.17: Add [male voice for TTS](https://github.com/PaddlePaddle/PaddleSpeech/pull/2660).
- 🔥 2022.11.07: Add [U2/U2++ C++ High Performance Streaming ASR Deployment](https://github.com/PaddlePaddle/PaddleSpeech/blob/develop/speechx/examples/u2pp_ol/wenetspeech).
- 🔥 2022.11.07: Add [U2/U2++ C++ High Performance Streaming ASR Deployment](./speechx/examples/u2pp_ol/wenetspeech).
- 👑 2022.11.01: Add [Adversarial Loss](https://arxiv.org/pdf/1907.04448.pdf) for [Chinese English mixed TTS](./examples/zh_en_tts/tts3).
- 🔥 2022.10.26: Add [Prosody Prediction](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/other/rhy) for TTS.
- 🔥 2022.10.26: Add [Prosody Prediction](./examples/other/rhy) for TTS.
- 🎉 2022.10.21: Add [SSML](https://github.com/PaddlePaddle/PaddleSpeech/discussions/2538) for TTS Chinese Text Frontend.
- 👑 2022.10.11: Add [Wav2vec2ASR-en](./examples/librispeech/asr3), wav2vec2.0 fine-tuning for ASR on LibriSpeech.
- 🔥 2022.09.26: Add Voice Cloning, TTS finetune, and [ERNIE-SAT](https://arxiv.org/abs/2211.03545) in [PaddleSpeech Web Demo](./demos/speech_web).
@ -191,7 +206,7 @@ Via the easy-to-use, efficient, flexible and scalable implementation, our vision
- Scan the QR code below with your Wechat, you can access to official technical exchange group and get the bonus ( more than 20GB learning materials, such as papers, codes and videos ) and the live link of the lessons. Look forward to your participation.
<div align="center">
<img src="https://user-images.githubusercontent.com/30135920/196351517-19dece6b-d6ea-448e-a341-d6bfe5712ec1.jpg" width = "200" />
<img src="https://user-images.githubusercontent.com/30135920/212860467-9e943cc3-8be8-49a4-97fd-7c94aad8e979.jpg" width = "200" />
</div>
## Installation
@ -987,8 +1002,9 @@ You are warmly welcome to submit questions in [discussions](https://github.com/P
- Many thanks to [vpegasus](https://github.com/vpegasus)/[xuesebot](https://github.com/vpegasus/xuesebot) for developing a rasa chatbot,which is able to speak and listen thanks to PaddleSpeech.
- Many thanks to [chenkui164](https://github.com/chenkui164)/[FastASR](https://github.com/chenkui164/FastASR) for the C++ inference implementation of PaddleSpeech ASR.
- Many thanks to [heyudage](https://github.com/heyudage)/[VoiceTyping](https://github.com/heyudage/VoiceTyping) for the real-time voice typing tool implementation of PaddleSpeech ASR streaming services.
- Many thanks to [EscaticZheng](https://github.com/EscaticZheng)/[ps3.9wheel-install](https://github.com/EscaticZheng/ps3.9wheel-install) for the python3.9 prebuilt wheel for PaddleSpeech installation in Windows without Viusal Studio.
Besides, PaddleSpeech depends on a lot of open source repositories. See [references](./docs/source/reference.md) for more information.
- Many thanks to [chinobing](https://github.com/chinobing)/[FastAPI-PaddleSpeech-Audio-To-Text](https://github.com/chinobing/FastAPI-PaddleSpeech-Audio-To-Text) for converting audio to text based on FastAPI and PaddleSpeech.
<a name="License"></a>
## License

@ -122,6 +122,20 @@
<img align="center" src="./docs/images/audio_icon.png" width="200" style="max-width: 100%;"></a><br>
</td>
</tr>
<tr>
<td>大家好,我是 parrot 虚拟老师我们来读一首诗我与春风皆过客I and the spring breeze are passing by你携秋水揽星河you take the autumn water to take the galaxy。</td>
<td align = "center">
<a href="https://paddlespeech.bj.bcebos.com/Parakeet/docs/demos/labixiaoxin.wav" rel="nofollow">
<img align="center" src="./docs/images/audio_icon.png" width="200" style="max-width: 100%;"></a><br>
</td>
</tr>
<tr>
<td>宜家唔系事必要你讲,但系你所讲嘅说话将会变成呈堂证供。</td>
<td align = "center">
<a href="https://paddlespeech.bj.bcebos.com/Parakeet/docs/demos/chengtangzhenggong.wav" rel="nofollow">
<img align="center" src="./docs/images/audio_icon.png" width="200" style="max-width: 100%;"></a><br>
</td>
</tr>
</tbody>
</table>
@ -161,20 +175,19 @@
- 🔬 主流模型及数据集: 本工具包实现了参与整条语音任务流水线的各个模块,并且采用了主流数据集如 LibriSpeech、LJSpeech、AIShell、CSMSC详情请见 [模型列表](#model-list)。
- 🧩 级联模型应用: 作为传统语音任务的扩展,我们结合了自然语言处理、计算机视觉等任务,实现更接近实际需求的产业级应用。
### 近期更新
- 🔥 2022.01.10: 新增 [中英混合 ASR CLI 和 Demos](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/demos/speech_recognition).
- 👑 2022.01.06: 新增 [ASR中英混合 tal_cs 训练推理流程](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/tal_cs/asr1/).
- 🎉 2022.12.02: 新增 [端到端韵律预测全流程](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/csmsc/tts3_rhy) (包含在声学模型中使用韵律标签)。
- 🎉 2022.11.30: 新增 [TTS Android 部署示例](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/demos/TTSAndroid)。
- 🎉 2023.02.16: 新增[粤语语音合成](./examples/canton/tts3)。
- 🔥 2023.01.10: 新增[中英混合 ASR CLI 和 Demos](./demos/speech_recognition)。
- 👑 2023.01.06: 新增 [ASR 中英混合 tal_cs 训练推理流程](./examples/tal_cs/asr1/)。
- 🎉 2022.12.02: 新增[端到端韵律预测全流程](./examples/csmsc/tts3_rhy) (包含在声学模型中使用韵律标签)。
- 🎉 2022.11.30: 新增 [TTS Android 部署示例](./demos/TTSAndroid)。
- 🤗 2022.11.28: PP-TTS and PP-ASR 示例可在 [AIStudio](https://aistudio.baidu.com/aistudio/modelsoverview) 和[飞桨官网](https://www.paddlepaddle.org.cn/models)体验!
- 👑 2022.11.18: 新增 [Whisper CLI 和 Demos](https://github.com/PaddlePaddle/PaddleSpeech/pull/2640), 支持多种语言的识别与翻译。
- 🔥 2022.11.18: 新增 [Wav2vec2 CLI 和 Demos](https://github.com/PaddlePaddle/PaddleSpeech/blob/develop/demos/speech_ssl), 支持 ASR 和 特征提取.
- 🔥 2022.11.18: 新增 [Wav2vec2 CLI 和 Demos](./demos/speech_ssl), 支持 ASR 和特征提取。
- 🎉 2022.11.17: TTS 新增[高质量男性音色](https://github.com/PaddlePaddle/PaddleSpeech/pull/2660)。
- 🔥 2022.11.07: 新增 [U2/U2++ 高性能流式 ASR C++ 部署](https://github.com/PaddlePaddle/PaddleSpeech/blob/develop/speechx/examples/u2pp_ol/wenetspeech)。
- 🔥 2022.11.07: 新增 [U2/U2++ 高性能流式 ASR C++ 部署](./speechx/examples/u2pp_ol/wenetspeech)。
- 👑 2022.11.01: [中英文混合 TTS](./examples/zh_en_tts/tts3) 新增 [Adversarial Loss](https://arxiv.org/pdf/1907.04448.pdf) 模块。
- 🔥 2022.10.26: TTS 新增[韵律预测](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/other/rhy)功能。
- 🔥 2022.10.26: TTS 新增[韵律预测](./develop/examples/other/rhy)功能。
- 🎉 2022.10.21: TTS 中文文本前端新增 [SSML](https://github.com/PaddlePaddle/PaddleSpeech/discussions/2538) 功能。
- 👑 2022.10.11: 新增 [Wav2vec2ASR-en](./examples/librispeech/asr3), 在 LibriSpeech 上针对 ASR 任务对 wav2vec2.0 的 finetuning。
- 🔥 2022.09.26: 新增 Voice Cloning, TTS finetune 和 [ERNIE-SAT](https://arxiv.org/abs/2211.03545) 到 [PaddleSpeech 网页应用](./demos/speech_web)。
@ -202,7 +215,7 @@
微信扫描二维码关注公众号,点击“马上报名”填写问卷加入官方交流群,获得更高效的问题答疑,与各行各业开发者充分交流,期待您的加入。
<div align="center">
<img src="https://user-images.githubusercontent.com/30135920/196351517-19dece6b-d6ea-448e-a341-d6bfe5712ec1.jpg" width = "200" />
<img src="https://user-images.githubusercontent.com/30135920/212860467-9e943cc3-8be8-49a4-97fd-7c94aad8e979.jpg" width = "200" />
</div>
<a name="安装"></a>
@ -988,10 +1001,11 @@ PaddleSpeech 的 **语音合成** 主要包含三个模块:文本前端、声
- 非常感谢 [awmmmm](https://github.com/awmmmm) 提供 fastspeech2 aishell3 conformer 预训练模型。
- 非常感谢 [phecda-xu](https://github.com/phecda-xu)/[PaddleDubbing](https://github.com/phecda-xu/PaddleDubbing) 基于 PaddleSpeech 的 TTS 模型搭建带 GUI 操作界面的配音工具。
- 非常感谢 [jerryuhoo](https://github.com/jerryuhoo)/[VTuberTalk](https://github.com/jerryuhoo/VTuberTalk) 基于 PaddleSpeech 的 TTS GUI 界面和基于 ASR 制作数据集的相关代码。
- 非常感谢 [vpegasus](https://github.com/vpegasus)/[xuesebot](https://github.com/vpegasus/xuesebot) 基于 PaddleSpeech 的 ASR 与 TTS 设计的可听、说对话机器人。
- 非常感谢 [chenkui164](https://github.com/chenkui164)/[FastASR](https://github.com/chenkui164/FastASR) 对 PaddleSpeech 的 ASR 进行 C++ 推理实现。
- 非常感谢 [heyudage](https://github.com/heyudage)/[VoiceTyping](https://github.com/heyudage/VoiceTyping) 基于 PaddleSpeech 的 ASR 流式服务实现的实时语音输入法工具。
- 非常感谢 [EscaticZheng](https://github.com/EscaticZheng)/[ps3.9wheel-install](https://github.com/EscaticZheng/ps3.9wheel-install) 对PaddleSpeech在Windows下的安装提供了无需Visua Studio基于python3.9的预编译依赖安装包。
- 非常感谢 [chinobing](https://github.com/chinobing)/[FastAPI-PaddleSpeech-Audio-To-Text](https://github.com/chinobing/FastAPI-PaddleSpeech-Audio-To-Text) 利用 FastAPI 实现 PaddleSpeech 语音转文字,文件上传、分割、转换进度显示、后台更新任务并以 csv 格式输出。
此外PaddleSpeech 依赖于许多开源存储库。有关更多信息,请参阅 [references](./docs/source/reference.md)。

@ -40,14 +40,9 @@ COMMITID = 'none'
base = [
"kaldiio",
"librosa==0.8.1",
"scipy>=1.0.0",
"soundfile~=0.10",
"colorlog",
"pathos == 0.2.8",
"pathos",
"pybind11",
"parameterized",
"tqdm",
"scikit-learn"
]
requirements = {
@ -273,7 +268,7 @@ def main():
},
# Package info
packages=find_packages(include=('paddleaudio*')),
packages=find_packages(include=['paddleaudio*']),
package_data=lib_package_data,
ext_modules=setup_helpers.get_ext_modules(),
zip_safe=True,

@ -1,8 +1,6 @@
aiofiles
faiss-cpu
praatio==5.0.0
praatio>=5.0.0
pydantic
python-multipart
scikit_learn
starlette
uvicorn

@ -1,12 +1,9 @@
braceexpand
colorlog
editdistance
fastapi
g2p_en
g2pM
h5py
inflect
jieba
jsonlines
kaldiio
keyboard
@ -16,7 +13,7 @@ matplotlib
myst-parser
nara_wpe
numpydoc
onnxruntime==1.10.0
onnxruntime>=1.11.0
opencc
paddlenlp
# use paddlepaddle == 2.3.* according to: https://github.com/PaddlePaddle/Paddle/issues/48243
@ -24,32 +21,25 @@ paddlepaddle>=2.2.2,<2.4.0
paddlespeech_ctcdecoders
paddlespeech_feat
pandas
pathos==0.2.8
pattern_singleton
Pillow>=9.0.0
ppdiffusers>=0.9.0
praatio==5.0.0
praatio>=5.0.0
prettytable
pypinyin-dict
pypinyin<=0.44.0
python-dateutil
pyworld==0.2.12
pyworld>=0.2.12
recommonmark>=0.5.0
resampy==0.2.2
resampy
sacrebleu
scipy
sentencepiece~=0.1.96
soundfile~=0.10
sphinx
sphinx-autobuild
sphinx-markdown-tables
sphinx_rtd_theme
textgrid
timer
tqdm
ToJyutping
typeguard
uvicorn
visualdl
webrtcvad
websockets
yacs~=0.1.8

@ -0,0 +1,198 @@
# Wav2vec2ASR with Aishell
This example contains code used to finetune [wav2vec2.0](https://https://arxiv.org/pdf/2006.11477.pdf) model with [Aishell dataset](http://www.openslr.org/resources/33)
## Overview
All the scripts you need are in `run.sh`. There are several stages in `run.sh`, and each stage has its function.
| Stage | Function |
|:---- |:----------------------------------------------------------- |
| 0 | Process data. It includes: <br> (1) Download the dataset <br> (2) Calculate the CMVN of the train dataset <br> (3) Get the vocabulary file <br> (4) Get the manifest files of the train, development and test dataset<br> (5) Download the pretrained wav2vec2 model |
| 1 | Train the model |
| 2 | Get the final model by averaging the top-k models, set k = 1 means to choose the best model |
| 3 | Test the final model performance |
| 4 | Infer the single audio file |
You can choose to run a range of stages by setting `stage` and `stop_stage `.
For example, if you want to execute the code in stage 2 and stage 3, you can run this script:
```bash
bash run.sh --stage 2 --stop_stage 3
```
Or you can set `stage` equal to `stop-stage` to only run one stage.
For example, if you only want to run `stage 0`, you can use the script below:
```bash
bash run.sh --stage 0 --stop_stage 0
```
The document below will describe the scripts in `run.sh` in detail.
## The Environment Variables
The path.sh contains the environment variables.
```bash
. ./path.sh
. ./cmd.sh
```
This script needs to be run first. And another script is also needed:
```bash
source ${MAIN_ROOT}/utils/parse_options.sh
```
It will support the way of using `--variable value` in the shell scripts.
## The Local Variables
Some local variables are set in `run.sh`.
`gpus` denotes the GPU number you want to use. If you set `gpus=`, it means you only use CPU.
`stage` denotes the number of stages you want to start from in the experiments.
`stop stage` denotes the number of the stage you want to end at in the experiments.
`conf_path` denotes the config path of the model.
`avg_num` denotes the number K of top-K models you want to average to get the final model.
`audio file` denotes the file path of the single file you want to infer in stage 5
`ckpt` denotes the checkpoint prefix of the model, e.g. "wav2vec2ASR"
You can set the local variables (except `ckpt`) when you use `run.sh`
For example, you can set the `gpus` and `avg_num` when you use the command line:
```bash
bash run.sh --gpus 0,1 --avg_num 20
```
## Stage 0: Data Processing
To use this example, you need to process data firstly and you can use stage 0 in `run.sh` to do this. The code is shown below:
```bash
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
# prepare data
bash ./local/data.sh || exit -1
fi
```
Stage 0 is for processing the data.
If you only want to process the data. You can run
```bash
bash run.sh --stage 0 --stop_stage 0
```
You can also just run these scripts in your command line.
```bash
. ./path.sh
. ./cmd.sh
bash ./local/data.sh
```
After processing the data, the `data` directory will look like this:
```bash
data/
|-- dev.meta
|-- lang_char
| `-- vocab.txt
|-- manifest.dev
|-- manifest.dev.raw
|-- manifest.test
|-- manifest.test.raw
|-- manifest.train
|-- manifest.train.raw
|-- mean_std.json
|-- test.meta
|-- train.meta
|-- train.csv
|-- dev.csv
|-- test.csv
```
Stage 0 also downloads the Chinese pre-trained [wav2vec2](https://paddlespeech.bj.bcebos.com/wav2vec/chinese-wav2vec2-large.pdparams) model.
```bash
mkdir -p exp/wav2vec2
wget -P exp/wav2vec2 https://paddlespeech.bj.bcebos.com/wav2vec/chinese-wav2vec2-large.pdparams
```
## Stage 1: Model Training
If you want to train the model. you can use stage 1 in `run.sh`. The code is shown below.
```bash
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
# train model, all `ckpt` under `exp` dir
CUDA_VISIBLE_DEVICES=${gpus} ./local/train.sh ${conf_path} ${ckpt}
fi
```
If you want to train the model, you can use the script below to execute stage 0 and stage 1:
```bash
bash run.sh --stage 0 --stop_stage 1
```
or you can run these scripts in the command line (only use CPU).
```bash
. ./path.sh
. ./cmd.sh
bash ./local/data.sh
CUDA_VISIBLE_DEVICES= ./local/train.sh conf/wav2vec2ASR.yaml wav2vec2ASR
```
## Stage 2: Top-k Models Averaging
After training the model, we need to get the final model for testing and inference. In every epoch, the model checkpoint is saved, so we can choose the best model from them based on the validation loss or we can sort them and average the parameters of the top-k models to get the final model. We can use stage 2 to do this, and the code is shown below. Note: We only train one epoch for wav2vec2ASR, thus the `avg_num` is set to 1.
```bash
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
# avg n best model
avg.sh best exp/${ckpt}/checkpoints ${avg_num}
fi
```
The `avg.sh` is in the `../../../utils/` which is define in the `path.sh`.
If you want to get the final model, you can use the script below to execute stage 0, stage 1, and stage 2:
```bash
bash run.sh --stage 0 --stop_stage 2
```
or you can run these scripts in the command line (only use CPU).
```bash
. ./path.sh
. ./cmd.sh
bash ./local/data.sh
CUDA_VISIBLE_DEVICES= ./local/train.sh conf/wav2vec2ASR.yaml wav2vec2ASR
avg.sh best exp/wav2vec2ASR/checkpoints 1
```
## Stage 3: Model Testing
The test stage is to evaluate the model performance. The code of test stage is shown below:
```bash
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
# test ckpt avg_n
CUDA_VISIBLE_DEVICES=0 ./local/test.sh ${conf_path} ${decode_conf_path} exp/${ckpt}/checkpoints/${avg_ckpt} || exit -1
fi
```
If you want to train a model and test it, you can use the script below to execute stage 0, stage 1, stage 2, and stage 3 :
```bash
bash run.sh --stage 0 --stop_stage 3
```
or you can run these scripts in the command line (only use CPU).
```bash
. ./path.sh
. ./cmd.sh
bash ./local/data.sh
CUDA_VISIBLE_DEVICES= ./local/train.sh conf/wav2vec2ASR.yaml wav2vec2ASR
avg.sh best exp/wav2vec2ASR/checkpoints 1
CUDA_VISIBLE_DEVICES= ./local/test.sh conf/wav2vec2ASR.yaml conf/tuning/decode.yaml exp/wav2vec2ASR/checkpoints/avg_1
```
## Pretrained Model
You can get the pretrained wav2vec2ASR from [this](../../../docs/source/released_model.md).
using the `tar` scripts to unpack the model and then you can use the script to test the model.
For example:
```bash
wget https://paddlespeech.bj.bcebos.com/s2t/aishell/asr3/wav2vec2ASR-large-aishell1_ckpt_1.3.0.model.tar.gz
tar xzvf wav2vec2ASR-large-aishell1_ckpt_1.3.0.model.tar.gz
source path.sh
# If you have process the data and get the manifest file you can skip the following 2 steps
bash local/data.sh --stage -1 --stop_stage -1
bash local/data.sh --stage 2 --stop_stage 2
CUDA_VISIBLE_DEVICES= ./local/test.sh conf/wav2vec2ASR.yaml conf/tuning/decode.yaml exp/wav2vec2ASR/checkpoints/avg_1
```
The performance of the released models are shown in [here](./RESULTS.md).
## Stage 4: Single Audio File Inference
In some situations, you want to use the trained model to do the inference for the single audio file. You can use stage 5. The code is shown below
```bash
if [ ${stage} -le 4 ] && [ ${stop_stage} -ge 4 ]; then
# test a single .wav file
CUDA_VISIBLE_DEVICES=0 ./local/test_wav.sh ${conf_path} ${decode_conf_path} exp/${ckpt}/checkpoints/${avg_ckpt} ${audio_file} || exit -1
fi
```
you can train the model by yourself using ```bash run.sh --stage 0 --stop_stage 3```, or you can download the pretrained model through the script below:
```bash
wget https://paddlespeech.bj.bcebos.com/s2t/aishell/asr3/wav2vec2ASR-large-aishell1_ckpt_1.3.0.model.tar.gz
tar xzvf wav2vec2ASR-large-aishell1_ckpt_1.3.0.model.tar.gz
```
You can download the audio demo:
```bash
wget -nc https://paddlespeech.bj.bcebos.com/datasets/single_wav/en/demo_002_en.wav -P data/
```
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
CUDA_VISIBLE_DEVICES= ./local/test_wav.sh conf/wav2vec2ASR.yaml conf/tuning/decode.yaml exp/wav2vec2ASR/checkpoints/avg_1 data/demo_002_en.wav
```

@ -0,0 +1,89 @@
# ====== About run.pl, queue.pl, slurm.pl, and ssh.pl ======
# Usage: <cmd>.pl [options] JOB=1:<nj> <log> <command...>
# e.g.
# run.pl --mem 4G JOB=1:10 echo.JOB.log echo JOB
#
# Options:
# --time <time>: Limit the maximum time to execute.
# --mem <mem>: Limit the maximum memory usage.
# -max-jobs-run <njob>: Limit the number parallel jobs. This is ignored for non-array jobs.
# --num-threads <ngpu>: Specify the number of CPU core.
# --gpu <ngpu>: Specify the number of GPU devices.
# --config: Change the configuration file from default.
#
# "JOB=1:10" is used for "array jobs" and it can control the number of parallel jobs.
# The left string of "=", i.e. "JOB", is replaced by <N>(Nth job) in the command and the log file name,
# e.g. "echo JOB" is changed to "echo 3" for the 3rd job and "echo 8" for 8th job respectively.
# Note that the number must start with a positive number, so you can't use "JOB=0:10" for example.
#
# run.pl, queue.pl, slurm.pl, and ssh.pl have unified interface, not depending on its backend.
# These options are mapping to specific options for each backend and
# it is configured by "conf/queue.conf" and "conf/slurm.conf" by default.
# If jobs failed, your configuration might be wrong for your environment.
#
#
# The official documentation for run.pl, queue.pl, slurm.pl, and ssh.pl:
# "Parallelization in Kaldi": http://kaldi-asr.org/doc/queue.html
# =========================================================~
# Select the backend used by run.sh from "local", "sge", "slurm", or "ssh"
cmd_backend='local'
# Local machine, without any Job scheduling system
if [ "${cmd_backend}" = local ]; then
# The other usage
export train_cmd="run.pl"
# Used for "*_train.py": "--gpu" is appended optionally by run.sh
export cuda_cmd="run.pl"
# Used for "*_recog.py"
export decode_cmd="run.pl"
# "qsub" (SGE, Torque, PBS, etc.)
elif [ "${cmd_backend}" = sge ]; then
# The default setting is written in conf/queue.conf.
# You must change "-q g.q" for the "queue" for your environment.
# To know the "queue" names, type "qhost -q"
# Note that to use "--gpu *", you have to setup "complex_value" for the system scheduler.
export train_cmd="queue.pl"
export cuda_cmd="queue.pl"
export decode_cmd="queue.pl"
# "sbatch" (Slurm)
elif [ "${cmd_backend}" = slurm ]; then
# The default setting is written in conf/slurm.conf.
# You must change "-p cpu" and "-p gpu" for the "partion" for your environment.
# To know the "partion" names, type "sinfo".
# You can use "--gpu * " by default for slurm and it is interpreted as "--gres gpu:*"
# The devices are allocated exclusively using "${CUDA_VISIBLE_DEVICES}".
export train_cmd="slurm.pl"
export cuda_cmd="slurm.pl"
export decode_cmd="slurm.pl"
elif [ "${cmd_backend}" = ssh ]; then
# You have to create ".queue/machines" to specify the host to execute jobs.
# e.g. .queue/machines
# host1
# host2
# host3
# Assuming you can login them without any password, i.e. You have to set ssh keys.
export train_cmd="ssh.pl"
export cuda_cmd="ssh.pl"
export decode_cmd="ssh.pl"
# This is an example of specifying several unique options in the JHU CLSP cluster setup.
# Users can modify/add their own command options according to their cluster environments.
elif [ "${cmd_backend}" = jhu ]; then
export train_cmd="queue.pl --mem 2G"
export cuda_cmd="queue-freegpu.pl --mem 2G --gpu 1 --config conf/gpu.conf"
export decode_cmd="queue.pl --mem 4G"
else
echo "$0: Error: Unknown cmd_backend=${cmd_backend}" 1>&2
return 1
fi

@ -0,0 +1,3 @@
process:
# use raw audio
- type: wav_process

@ -0,0 +1,101 @@
# Copyright (c) 2023 speechbrain Authors. All Rights Reserved.
# Copyright (c) 2023 PaddlePaddle Authors. All Rights Reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
#
# Modified from speechbrain 2023 (https://github.com/speechbrain/speechbrain/blob/develop/recipes/AISHELL-1/ASR/CTC/hparams/train_with_wav2vec.yaml)
# ############################################################################
# Model: CTC-wav2vec2
# Encoder: wav2vec2
# Decoder: -
# Tokens: Char
# losses: CTC
# Training: AISHELL-1
# Authors: Yingzhi WANG 2022
# ############################################################################
output_folder: !ref data
cer_file: !ref <output_folder>/cer.txt
save_folder: !ref <output_folder>/save
train_log: !ref <output_folder>/train_log.txt
# Data files
data_folder: data/aishell # e,g./path/to/aishell
skip_prep: False
ckpt_interval_minutes: 15 # save checkpoint every N min
train_data: !ref <output_folder>/train.csv
valid_data: !ref <output_folder>/dev.csv
test_data: !ref <output_folder>/test.csv
wav2vec2_hub: TencentGameMate/chinese-wav2vec2-large
# Training parameters
number_of_epochs: 80
lr: 1.0
lr_wav2vec: 0.0001
sorting: ascending
auto_mix_prec: False
sample_rate: 16000
# With data_parallel batch_size is split into N jobs
# With DDP batch_size is multiplied by N jobs
# Must be 8 per GPU to fit 32GB of VRAM
batch_size: 5
test_batch_size: 1 # need set to 1 when decoding
dynamic_batching: False
dynamic_batch_sampler:
feats_hop_size: 0.01
max_batch_len: 15 # in terms of "duration" in annotations by default, second here
left_bucket_len: 200 # old implementation attributs
multiplier: 1.1 # old implementation attributs
shuffle_ex: False # if true re-creates batches at each epoch shuffling examples.
num_buckets: 10 # floor(log(max_batch_len/left_bucket_len, multiplier)) + 1
batch_ordering: ascending
num_workers: 6
# Dataloader options
train_dataloader_opts:
batch_size: !ref <batch_size>
num_workers: !ref <num_workers>
valid_dataloader_opts:
batch_size: !ref <test_batch_size>
num_workers: !ref <num_workers>
test_dataloader_opts:
batch_size: !ref <test_batch_size>
num_workers: !ref <num_workers>
wav2vec_output_dim: 1024
dnn_neurons: 1024
freeze_wav2vec: False
dropout: 0.15
tokenizer: !apply:transformers.BertTokenizer.from_pretrained
pretrained_model_name_or_path: bert-base-chinese
# bert-base-chinese tokens length
output_neurons: 21128
# Decoding parameters
# Be sure that the bos and eos index match with the BPEs ones
blank_index: 0
# AISHELL-1 has spaces between words in the transcripts,
# which Chinese writing normally does not do.
# If remove_spaces, spaces are removed
# from the transcript before computing CER.
# (e.g., 祝 可爱 的 你 —> 祝可爱的你)
remove_spaces: True
split_tokens: !apply:operator.not_ [!ref <remove_spaces>]

@ -0,0 +1,4 @@
decode_batch_size: 1
error_rate_type: cer
decoding_method: ctc_greedy_search # 'ctc_greedy_search', 'ctc_prefix_beam_search'
beam_size: 10

@ -0,0 +1,167 @@
############################################
# Network Architecture #
############################################
freeze_wav2vec2: False
normalize_wav: True
output_norm: True
init_type: 'kaiming_uniform' # !Warning: need to convergence
enc:
input_shape: 1024
dnn_blocks: 3
dnn_neurons: 1024
activation: True
normalization: True
dropout_rate: [0.15, 0.15, 0.0]
ctc:
enc_n_units: 1024
blank_id: 0
dropout_rate: 0.0
audio_augment:
speeds: [90, 100, 110]
spec_augment:
time_warp: True
time_warp_window: 5
time_warp_mode: bicubic
freq_mask: True
n_freq_mask: 2
time_mask: True
n_time_mask: 2
replace_with_zero: False
freq_mask_width: 30
time_mask_width: 40
wav2vec2_params_path: exp/wav2vec2/chinese-wav2vec2-large.pdparams
############################################
# Wav2Vec2.0 #
############################################
# vocab_size: 1000000
hidden_size: 1024
num_hidden_layers: 24
num_attention_heads: 16
intermediate_size: 4096
hidden_act: gelu
hidden_dropout: 0.1
activation_dropout: 0.0
attention_dropout: 0.1
feat_proj_dropout: 0.1
feat_quantizer_dropout: 0.0
final_dropout: 0.0
layerdrop: 0.1
initializer_range: 0.02
layer_norm_eps: 1e-5
feat_extract_norm: layer
feat_extract_activation: gelu
conv_dim: [512, 512, 512, 512, 512, 512, 512]
conv_stride: [5, 2, 2, 2, 2, 2, 2]
conv_kernel: [10, 3, 3, 3, 3, 2, 2]
conv_bias: True
num_conv_pos_embeddings: 128
num_conv_pos_embedding_groups: 16
do_stable_layer_norm: True
apply_spec_augment: False
mask_channel_length: 10
mask_channel_min_space: 1
mask_channel_other: 0.0
mask_channel_prob: 0.0
mask_channel_selection: static
mask_feature_length: 10
mask_feature_min_masks: 0
mask_feature_prob: 0.0
mask_time_length: 10
mask_time_min_masks: 2
mask_time_min_space: 1
mask_time_other: 0.0
mask_time_prob: 0.075
mask_time_selection: static
num_codevectors_per_group: 320
num_codevector_groups: 2
contrastive_logits_temperature: 0.1
num_negatives: 100
codevector_dim: 256
proj_codevector_dim: 256
diversity_loss_weight: 0.1
use_weighted_layer_sum: False
# pad_token_id: 0
# bos_token_id: 1
# eos_token_id: 2
add_adapter: False
adapter_kernel_size: 3
adapter_stride: 2
num_adapter_layers: 3
output_hidden_size: None
###########################################
# Data #
###########################################
train_manifest: data/manifest.train
dev_manifest: data/manifest.dev
test_manifest: data/manifest.test
vocab_filepath: data/lang_char/vocab.txt
###########################################
# Dataloader #
###########################################
unit_type: 'char'
mean_std_filepath:
preprocess_config: conf/preprocess.yaml
sortagrad: -1 # Feed samples from shortest to longest ; -1: enabled for all epochs, 0: disabled, other: enabled for 'other' epochs
batch_size: 5 # Different batch_size may cause large differences in results
maxlen_in: 51200000000 # if input length > maxlen-in batchsize is automatically reduced
maxlen_out: 1500000 # if output length > maxlen-out batchsize is automatically reduced
minibatches: 0 # for debug
batch_count: auto
batch_bins: 0
batch_frames_in: 0
batch_frames_out: 0
batch_frames_inout: 0
num_workers: 6
subsampling_factor: 1
num_encs: 1
dist_sampler: True
shortest_first: True
return_lens_rate: True
###########################################
# use speechbrain dataloader #
###########################################
use_sb_pipeline: True # whether use speechbrain pipeline. Default is True.
sb_pipeline_conf: conf/train_with_wav2vec.yaml
###########################################
# Training #
###########################################
n_epoch: 80
accum_grad: 1
global_grad_clip: 5.0
model_optim: adadelta
model_optim_conf:
lr: 1.0
weight_decay: 0.0
rho: 0.95
epsilon: 1.0e-8
wav2vec2_optim: adam
wav2vec2_optim_conf:
lr: 0.0001
weight_decay: 0.0
model_scheduler: newbobscheduler
model_scheduler_conf:
improvement_threshold: 0.0025
annealing_factor: 0.8
patient: 0
wav2vec2_scheduler: newbobscheduler
wav2vec2_scheduler_conf:
improvement_threshold: 0.0025
annealing_factor: 0.9
patient: 0
log_interval: 1
checkpoint:
kbest_n: 50
latest_n: 5

@ -0,0 +1,129 @@
# Copyright (c) 2023 speechbrain Authors. All Rights Reserved.
# Copyright (c) 2023 PaddlePaddle Authors. All Rights Reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
#
# Modified from speechbrain 2023
# (https://github.com/speechbrain/speechbrain/blob/develop/recipes/AISHELL-1/aishell_prepare.py)
import argparse
import csv
import glob
import logging
import os
from paddlespeech.s2t.models.wav2vec2.io.dataio import read_audio
logger = logging.getLogger(__name__)
DATA_HOME = os.path.expanduser('~/.cache/paddle/dataset/speech')
parser = argparse.ArgumentParser(description=__doc__)
parser.add_argument(
"--data_folder",
default=DATA_HOME + "/Aishell",
type=str,
help="Directory to save the dataset. (default: %(default)s)")
parser.add_argument(
"--save_folder",
default="data/",
type=str,
help="Filepath prefix for output manifests. (default: %(default)s)")
parser.add_argument(
"--skip_prep",
default=False,
type=bool,
help="If True, skip data preparation. (default: %(default)s)")
args = parser.parse_args()
def prepare_aishell(data_folder, save_folder, skip_prep=False):
"""
This function prepares the AISHELL-1 dataset.
If the folder does not exist, the zip file will be extracted. If the zip file does not exist, it will be downloaded.
data_folder : path to AISHELL-1 dataset.
save_folder: path where to store the manifest csv files.
skip_prep: If True, skip data preparation.
"""
if skip_prep:
return
# Create filename-to-transcript dictionary
filename2transcript = {}
with open(
os.path.join(data_folder,
"data_aishell/transcript/aishell_transcript_v0.8.txt"),
"r", ) as f:
lines = f.readlines()
for line in lines:
key = line.split()[0]
value = " ".join(line.split()[1:])
filename2transcript[key] = value
splits = [
"train",
"dev",
"test",
]
ID_start = 0 # needed to have a unique ID for each audio
for split in splits:
new_filename = os.path.join(save_folder, split) + ".csv"
if os.path.exists(new_filename):
continue
logger.info("Preparing %s..." % new_filename)
csv_output = [["ID", "duration", "wav", "transcript"]]
entry = []
all_wavs = glob.glob(
os.path.join(data_folder, "data_aishell/wav") + "/" + split +
"/*/*.wav")
for i in range(len(all_wavs)):
filename = all_wavs[i].split("/")[-1].split(".wav")[0]
if filename not in filename2transcript:
continue
signal = read_audio(all_wavs[i])
duration = signal.shape[0] / 16000
transcript_ = filename2transcript[filename]
csv_line = [
ID_start + i,
str(duration),
all_wavs[i],
transcript_,
]
entry.append(csv_line)
csv_output = csv_output + entry
with open(new_filename, mode="w") as csv_f:
csv_writer = csv.writer(
csv_f, delimiter=",", quotechar='"', quoting=csv.QUOTE_MINIMAL)
for line in csv_output:
csv_writer.writerow(line)
msg = "\t%s successfully created!" % (new_filename)
logger.info(msg)
ID_start += len(all_wavs)
def main():
if args.data_folder.startswith('~'):
args.data_folder = os.path.expanduser(args.data_folder)
prepare_aishell(args.data_folder, args.save_folder, skip_prep=False)
print("Data csv prepare done!")
if __name__ == '__main__':
main()

@ -0,0 +1,101 @@
#!/bin/bash
stage=-1
stop_stage=-1
dict_dir=data/lang_char
. ${MAIN_ROOT}/utils/parse_options.sh || exit -1;
mkdir -p data
mkdir -p ${dict_dir}
TARGET_DIR=${MAIN_ROOT}/dataset
mkdir -p ${TARGET_DIR}
if [ ${stage} -le -1 ] && [ ${stop_stage} -ge -1 ]; then
# download data, generate manifests
python3 ${TARGET_DIR}/aishell/aishell.py \
--manifest_prefix="data/manifest" \
--target_dir="${TARGET_DIR}/aishell"
#generate csv file for speechbrain dataloader
python3 local/aishell_prepare.py \
--data_folder="${TARGET_DIR}/aishell" \
--save_folder="data/"
if [ $? -ne 0 ]; then
echo "Prepare Aishell failed. Terminated."
exit 1
fi
for dataset in train dev test; do
mv data/manifest.${dataset} data/manifest.${dataset}.raw
done
fi
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
# compute mean and stddev for normalizer
num_workers=$(nproc)
python3 ${MAIN_ROOT}/utils/compute_mean_std.py \
--manifest_path="data/manifest.train.raw" \
--spectrum_type="fbank" \
--feat_dim=80 \
--delta_delta=false \
--stride_ms=10 \
--window_ms=25 \
--sample_rate=16000 \
--use_dB_normalization=False \
--num_samples=-1 \
--num_workers=${num_workers} \
--output_path="data/mean_std.json"
if [ $? -ne 0 ]; then
echo "Compute mean and stddev failed. Terminated."
exit 1
fi
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
# download data, generate manifests
# build vocabulary
python3 ${MAIN_ROOT}/utils/build_vocab.py \
--unit_type="char" \
--count_threshold=0 \
--vocab_path="${dict_dir}/vocab.txt" \
--manifest_paths "data/manifest.train.raw"
if [ $? -ne 0 ]; then
echo "Build vocabulary failed. Terminated."
exit 1
fi
fi
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
# format manifest with tokenids, vocab size
for dataset in train dev test; do
{
python3 ${MAIN_ROOT}/utils/format_data.py \
--cmvn_path "data/mean_std.json" \
--unit_type "char" \
--vocab_path="${dict_dir}/vocab.txt" \
--manifest_path="data/manifest.${dataset}.raw" \
--output_path="data/manifest.${dataset}"
if [ $? -ne 0 ]; then
echo "Formt mnaifest failed. Terminated."
exit 1
fi
} &
done
wait
fi
echo "Aishell data preparation done."
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
mkdir -p exp/wav2vec2
echo "Pretrained wav2vec2 model download"
wget -P exp/wav2vec2 https://paddlespeech.bj.bcebos.com/wav2vec/chinese-wav2vec2-large.pdparams
fi
exit 0

@ -0,0 +1,84 @@
#!/bin/bash
set -e
ngpu=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
echo "using $ngpu gpus..."
expdir=exp
datadir=data
train_set=train_960
recog_set="test-clean test-other dev-clean dev-other"
recog_set="test-clean"
config_path=$1
decode_config_path=$2
ckpt_prefix=$3
source ${MAIN_ROOT}/utils/parse_options.sh || exit 1;
# download language model
#bash local/download_lm_en.sh
#if [ $? -ne 0 ]; then
# exit 1
#fi
python3 utils/format_rsl.py \
--origin_ref data/manifest.test.raw \
--trans_ref data/manifest.test.text
for type in ctc_greedy_search; do
echo "decoding ${type}"
batch_size=1
python3 -u ${BIN_DIR}/test.py \
--ngpu ${ngpu} \
--config ${config_path} \
--decode_cfg ${decode_config_path} \
--result_file ${ckpt_prefix}.${type}.rsl \
--checkpoint_path ${ckpt_prefix} \
--opts decode.decoding_method ${type} \
--opts decode.decode_batch_size ${batch_size}
if [ $? -ne 0 ]; then
echo "Failed in evaluation!"
exit 1
fi
python3 utils/format_rsl.py \
--origin_hyp ${ckpt_prefix}.${type}.rsl \
--trans_hyp ${ckpt_prefix}.${type}.rsl.text
python3 utils/compute-wer.py --char=1 --v=1 \
data/manifest.test.text ${ckpt_prefix}.${type}.rsl.text > ${ckpt_prefix}.${type}.error
echo "decoding ${type} done."
done
for type in ctc_prefix_beam_search; do
echo "decoding ${type}"
batch_size=1
python3 -u ${BIN_DIR}/test.py \
--ngpu ${ngpu} \
--config ${config_path} \
--decode_cfg ${decode_config_path} \
--result_file ${ckpt_prefix}.${type}.rsl \
--checkpoint_path ${ckpt_prefix} \
--opts decode.decoding_method ${type} \
--opts decode.decode_batch_size ${batch_size}
if [ $? -ne 0 ]; then
echo "Failed in evaluation!"
exit 1
fi
python3 utils/format_rsl.py \
--origin_hyp ${ckpt_prefix}.${type}.rsl \
--trans_hyp ${ckpt_prefix}.${type}.rsl.text
python3 utils/compute-wer.py --char=1 --v=1 \
data/manifest.test-clean.text ${ckpt_prefix}.${type}.rsl.text > ${ckpt_prefix}.${type}.error
echo "decoding ${type} done."
done
echo "Finished"
exit 0

@ -0,0 +1,58 @@
#!/bin/bash
if [ $# != 4 ];then
echo "usage: ${0} config_path decode_config_path ckpt_path_prefix audio_file"
exit -1
fi
ngpu=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
echo "using $ngpu gpus..."
config_path=$1
decode_config_path=$2
ckpt_prefix=$3
audio_file=$4
mkdir -p data
wget -nc https://paddlespeech.bj.bcebos.com/datasets/single_wav/en/demo_002_en.wav -P data/
if [ $? -ne 0 ]; then
exit 1
fi
if [ ! -f ${audio_file} ]; then
echo "Plase input the right audio_file path"
exit 1
fi
chunk_mode=false
if [[ ${config_path} =~ ^.*chunk_.*yaml$ ]];then
chunk_mode=true
fi
# download language model
#bash local/download_lm_ch.sh
#if [ $? -ne 0 ]; then
# exit 1
#fi
for type in ctc_greedy_search; do
echo "decoding ${type}"
batch_size=1
output_dir=${ckpt_prefix}
mkdir -p ${output_dir}
python3 -u ${BIN_DIR}/test_wav.py \
--ngpu ${ngpu} \
--config ${config_path} \
--decode_cfg ${decode_config_path} \
--result_file ${output_dir}/${type}.rsl \
--checkpoint_path ${ckpt_prefix} \
--opts decode.decoding_method ${type} \
--opts decode.decode_batch_size ${batch_size} \
--audio_file ${audio_file}
if [ $? -ne 0 ]; then
echo "Failed in evaluation!"
exit 1
fi
done
exit 0

@ -0,0 +1,59 @@
#!/bin/bash
if [ $# -lt 2 ] && [ $# -gt 3 ];then
echo "usage: CUDA_VISIBLE_DEVICES=0 ${0} config_path ckpt_name ips(optional)"
exit -1
fi
ngpu=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
echo "using $ngpu gpus..."
config_path=$1
ckpt_name=$2
resume=$3
ips=$4
if [ ! $ips ];then
ips_config=
else
ips_config="--ips="${ips}
fi
mkdir -p exp
# seed may break model convergence
seed=2
if [ ${seed} != 0 ]; then
export FLAGS_cudnn_deterministic=True
fi
# export FLAGS_cudnn_exhaustive_search=true
# export FLAGS_conv_workspace_size_limit=4000
# export FLAGS_allocator_strategy=naive_best_fit
if [ ${ngpu} == 0 ]; then
python3 -u ${BIN_DIR}/train.py \
--ngpu ${ngpu} \
--config ${config_path} \
--output exp/${ckpt_name} \
--seed ${seed} \
--resume ${resume}
else
python3 -m paddle.distributed.launch --log_dir=${ckpt_name} --gpus=${CUDA_VISIBLE_DEVICES} ${ips_config} ${BIN_DIR}/train.py \
--ngpu ${ngpu} \
--config ${config_path} \
--output exp/${ckpt_name} \
--seed ${seed} \
--resume ${resume}
fi
if [ ${seed} != 0 ]; then
unset FLAGS_cudnn_deterministic
fi
if [ $? -ne 0 ]; then
echo "Failed in training!"
exit 1
fi
exit 0

@ -0,0 +1,15 @@
export MAIN_ROOT=`realpath ${PWD}/../../../`
export PATH=${MAIN_ROOT}:${MAIN_ROOT}/tools/sctk/bin:${PWD}/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}:${PYTHONPATH}
export LD_LIBRARY_PATH=${LD_LIBRARY_PATH}:/usr/local/lib/
MODEL=wav2vec2
export BIN_DIR=${MAIN_ROOT}/paddlespeech/s2t/exps/${MODEL}/bin

@ -0,0 +1,48 @@
#!/bin/bash
set -e
. ./path.sh || exit 1;
. ./cmd.sh || exit 1;
gpus=0,1,2,3
stage=0
stop_stage=4
conf_path=conf/wav2vec2ASR.yaml
ips= #xx.xx.xx.xx,xx.xx.xx.xx
decode_conf_path=conf/tuning/decode.yaml
avg_num=1
resume= # xx e.g. 30
export FLAGS_cudnn_deterministic=1
. ${MAIN_ROOT}/utils/parse_options.sh || exit 1;
audio_file=data/demo_002_en.wav
avg_ckpt=avg_${avg_num}
ckpt=$(basename ${conf_path} | awk -F'.' '{print $1}')
echo "checkpoint name ${ckpt}"git revert -v
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
# prepare data
bash ./local/data.sh || exit -1
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
# train model, all `ckpt` under `exp` dir
CUDA_VISIBLE_DEVICES=${gpus} ./local/train.sh ${conf_path} ${ckpt} ${resume} ${ips}
fi
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
# avg n best model
avg.sh last exp/${ckpt}/checkpoints ${avg_num}
fi
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
# greedy search decoder
CUDA_VISIBLE_DEVICES=0 ./local/test.sh ${conf_path} ${decode_conf_path} exp/${ckpt}/checkpoints/${avg_ckpt} || exit -1
fi
if [ ${stage} -le 4 ] && [ ${stop_stage} -ge 4 ]; then
# test a single .wav file
CUDA_VISIBLE_DEVICES=0 ./local/test_wav.sh ${conf_path} ${decode_conf_path} exp/${ckpt}/checkpoints/${avg_ckpt} ${audio_file} || exit -1
fi

@ -0,0 +1 @@
../../../utils

@ -1,13 +0,0 @@
#!/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}

@ -0,0 +1 @@
../../csmsc/tts3/path.sh

@ -1,13 +0,0 @@
#!/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=tacotron2
export BIN_DIR=${MAIN_ROOT}/paddlespeech/t2s/exps/${MODEL}

@ -0,0 +1 @@
../../csmsc/tts0/path.sh

@ -1,13 +0,0 @@
#!/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

@ -0,0 +1 @@
../../vc0/local/train.sh

@ -1,13 +0,0 @@
#!/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}

@ -0,0 +1 @@
../../csmsc/tts3/path.sh

@ -1,20 +0,0 @@
#!/bin/bash
config_path=$1
train_output_path=$2
ckpt_name=$3
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

@ -0,0 +1 @@
../../vc1/local/synthesize.sh

@ -1,13 +0,0 @@
#!/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

@ -0,0 +1 @@
../../vc0/local/train.sh

@ -1,13 +0,0 @@
#!/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}

@ -0,0 +1 @@
../../csmsc/tts3/path.sh

@ -42,14 +42,18 @@ if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/train/raw/metadata.jsonl \
--dumpdir=dump/train/norm \
--stats=dump/train/feats_stats.npy
--stats=dump/train/feats_stats.npy \
--skip-wav-copy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/dev/raw/metadata.jsonl \
--dumpdir=dump/dev/norm \
--stats=dump/train/feats_stats.npy
--stats=dump/train/feats_stats.npy \
--skip-wav-copy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/test/raw/metadata.jsonl \
--dumpdir=dump/test/norm \
--stats=dump/train/feats_stats.npy
--stats=dump/train/feats_stats.npy \
--skip-wav-copy
fi

@ -1,14 +0,0 @@
#!/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 ${BIN_DIR}/../synthesize.py \
--config=${config_path} \
--checkpoint=${train_output_path}/checkpoints/${ckpt_name} \
--test-metadata=dump/test/norm/metadata.jsonl \
--output-dir=${train_output_path}/test \
--generator-type=pwgan

@ -0,0 +1 @@
../../../csmsc/voc1/local/synthesize.sh

@ -1,13 +0,0 @@
#!/bin/bash
config_path=$1
train_output_path=$2
FLAGS_cudnn_exhaustive_search=true \
FLAGS_conv_workspace_size_limit=4000 \
python ${BIN_DIR}/train.py \
--train-metadata=dump/train/norm/metadata.jsonl \
--dev-metadata=dump/dev/norm/metadata.jsonl \
--config=${config_path} \
--output-dir=${train_output_path} \
--ngpu=1

@ -0,0 +1 @@
../../../csmsc/voc1/local/train.sh

@ -1,13 +0,0 @@
#!/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=parallelwave_gan
export BIN_DIR=${MAIN_ROOT}/paddlespeech/t2s/exps/gan_vocoder/${MODEL}

@ -0,0 +1 @@
../../csmsc/voc1/path.sh

@ -1,55 +0,0 @@
#!/bin/bash
stage=0
stop_stage=100
config_path=$1
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 ${BIN_DIR}/../preprocess.py \
--rootdir=~/datasets/data_aishell3/ \
--dataset=aishell3 \
--dumpdir=dump \
--dur-file=durations.txt \
--config=${config_path} \
--cut-sil=True \
--num-cpu=20
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, 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 \
--stats=dump/train/feats_stats.npy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/dev/raw/metadata.jsonl \
--dumpdir=dump/dev/norm \
--stats=dump/train/feats_stats.npy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/test/raw/metadata.jsonl \
--dumpdir=dump/test/norm \
--stats=dump/train/feats_stats.npy
fi

@ -0,0 +1 @@
../../voc1/local/preprocess.sh

@ -1,14 +0,0 @@
#!/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 ${BIN_DIR}/../synthesize.py \
--config=${config_path} \
--checkpoint=${train_output_path}/checkpoints/${ckpt_name} \
--test-metadata=dump/test/norm/metadata.jsonl \
--output-dir=${train_output_path}/test \
--generator-type=hifigan

@ -0,0 +1 @@
../../../csmsc/voc5/local/synthesize.sh

@ -1,13 +0,0 @@
#!/bin/bash
config_path=$1
train_output_path=$2
FLAGS_cudnn_exhaustive_search=true \
FLAGS_conv_workspace_size_limit=4000 \
python ${BIN_DIR}/train.py \
--train-metadata=dump/train/norm/metadata.jsonl \
--dev-metadata=dump/dev/norm/metadata.jsonl \
--config=${config_path} \
--output-dir=${train_output_path} \
--ngpu=1

@ -0,0 +1 @@
../../../csmsc/voc1/local/train.sh

@ -1,13 +0,0 @@
#!/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=hifigan
export BIN_DIR=${MAIN_ROOT}/paddlespeech/t2s/exps/gan_vocoder/${MODEL}

@ -0,0 +1 @@
../../csmsc/voc5/path.sh

@ -1,25 +0,0 @@
#!/bin/bash
config_path=$1
train_output_path=$2
ckpt_name=$3
stage=0
stop_stage=0
# hifigan
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=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 \
--test_metadata=dump/test/norm/metadata.jsonl \
--output_dir=${train_output_path}/test \
--phones_dict=dump/phone_id_map.txt
fi

@ -0,0 +1 @@
../../../aishell3/ernie_sat/local/synthesize.sh

@ -1,12 +0,0 @@
#!/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=8 \
--phones-dict=dump/phone_id_map.txt

@ -0,0 +1 @@
../../../aishell3/ernie_sat/local/train.sh

@ -1,13 +0,0 @@
#!/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=ernie_sat
export BIN_DIR=${MAIN_ROOT}/paddlespeech/t2s/exps/${MODEL}

@ -0,0 +1 @@
../../aishell3/ernie_sat/path.sh

@ -0,0 +1,121 @@
# FastSpeech2 with Cantonese language
## Dataset
### Download and Extract
If you don't have the Cantonese datasets mentioned above, please download and unzip [Guangzhou_Cantonese_Scripted_Speech_Corpus_Daily_Use_Sentence](https://magichub.com/datasets/guangzhou-cantonese-scripted-speech-corpus-daily-use-sentence/) and [Guangzhou_Cantonese_Scripted_Speech_Corpus_in_Vehicle](https://magichub.com/datasets/guangzhou-cantonese-scripted-speech-corpus-in-the-vehicle/) under `~/datasets/`.
To obtain better performance, please combine these two datasets together as follows:
```bash
mkdir -p ~/datasets/canton_all/WAV
cp -r ~/datasets/Guangzhou_Cantonese_Scripted_Speech_Corpus_Daily_Use_Sentence/WAV/* ~/datasets/canton_all/WAV
cp -r ~/datasets/Guangzhou_Cantonese_Scripted_Speech_Corpus_in_Vehicle/WAV/* ~/datasets/canton_all/WAV
```
After that, it should be look like:
```
~/datasets/canton_all
│ └── WAV
│ └──G0001
│ └──G0002
│ ...
│ └──G0071
│ └──G0072
```
### Get MFA Result and Extract
We use [MFA1.x](https://github.com/MontrealCorpusTools/Montreal-Forced-Aligner) to get durations for canton_fastspeech2.
You can train your MFA model reference to [canton_mfa example](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/other/mfa) (use MFA1.x now) of our repo.
We here provide the MFA results of these two datasets. [canton_alignment.zip](https://paddlespeech.bj.bcebos.com/MFA/Canton/canton_alignment.zip)
## Get Started
Assume the path to the Cantonese MFA result of the two datsets mentioned above is `./canton_alignment`.
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
├── energy_stats.npy
├── norm
├── pitch_stats.npy
├── 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、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, a path of energy features, speaker, and id of each utterance.
### Training details can refer to the script of [examples/aishell3/tts3](../../aishell3/tts3).
## Pretrained Model
Pretrained FastSpeech2 model with no silence in the edge of audios:
- [fastspeech2_canton_ckpt_1.4.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_canton_ckpt_1.4.0.zip)
FastSpeech2 checkpoint contains files listed below.
```text
fastspeech2_canton_ckpt_1.4.0
├── 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_140000.pdz # model parameters and optimizer states
├── 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
```
We use [parallel wavegan](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/voc1) as the neural vocoder.
Download the 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
```
You can use the following scripts to synthesize for `${BIN_DIR}/../sentences_canton.txt` using pretrained fastspeech2 and parallel wavegan models.
```bash
source path.sh
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/../synthesize_e2e.py \
--am=fastspeech2_aishell3 \
--am_config=fastspeech2_canton_ckpt_1.4.0/default.yaml \
--am_ckpt=fastspeech2_canton_ckpt_1.4.0/snapshot_iter_140000.pdz \
--am_stat=fastspeech2_canton_ckpt_1.4.0/speech_stats.npy \
--voc=pwgan_aishell3 \
--voc_config=pwg_aishell3_ckpt_0.5/default.yaml \
--voc_ckpt=pwg_aishell3_ckpt_0.5/snapshot_iter_1000000.pdz \
--voc_stat=pwg_aishell3_ckpt_0.5/feats_stats.npy \
--lang=canton \
--text=${BIN_DIR}/../sentences_canton.txt \
--output_dir=exp/default/test_e2e \
--phones_dict=fastspeech2_canton_ckpt_1.4.0/phone_id_map.txt \
--speaker_dict=fastspeech2_canton_ckpt_1.4.0/speaker_id_map.txt \
--spk_id=0 \
--inference_dir=exp/default/inference
```

@ -0,0 +1,107 @@
###########################################################
# 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)
# The canton datasets we use are different from others like Databaker or LJSpeech,
# we set it to 110 to avoid too many zero-pitch problem.
# Reference: https://github.com/JeremyCCHsu/Python-Wrapper-for-World-Vocoder/issues/38
f0min: 110 # Minimum f0 for pitch extraction.
f0max: 400 # Maximum f0 for pitch extraction.
###########################################################
# DATA SETTING #
###########################################################
batch_size: 32
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: 256 # 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: 1000
num_snapshots: 5
###########################################################
# OTHER SETTING #
###########################################################
seed: 10086

@ -0,0 +1,75 @@
#!/bin/bash
stage=0
stop_stage=100
config_path=$1
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
# get durations from MFA's result
echo "Generate durations.txt from MFA results ..."
python3 ${MAIN_ROOT}/utils/gen_duration_from_textgrid.py \
--inputdir=./canton_alignment \
--output durations.txt \
--config=${config_path}
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
# extract features
echo "Extract features ..."
python3 ${BIN_DIR}/preprocess.py \
--dataset=canton \
--rootdir=~/datasets/canton_all \
--dumpdir=dump \
--dur-file=durations.txt \
--config=${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="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 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 \
--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

@ -0,0 +1 @@
../../../aishell3/tts3/local/synthesize.sh

@ -0,0 +1,53 @@
#!/bin/bash
config_path=$1
train_output_path=$2
ckpt_name=$3
stage=0
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 ${BIN_DIR}/../synthesize_e2e.py \
--am=fastspeech2_canton \
--am_config=${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 \
--lang=canton \
--text=${BIN_DIR}/../sentences_canton.txt \
--output_dir=${train_output_path}/test_e2e \
--phones_dict=dump/phone_id_map.txt \
--speaker_dict=dump/speaker_id_map.txt \
--spk_id=0 \
--inference_dir=${train_output_path}/inference
fi
# hifigan
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
echo "in hifigan syn_e2e"
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/../synthesize_e2e.py \
--am=fastspeech2_canton \
--am_config=${config_path} \
--am_ckpt=${train_output_path}/checkpoints/${ckpt_name} \
--am_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 \
--lang=canton \
--text=${BIN_DIR}/../sentences_canton.txt \
--output_dir=${train_output_path}/test_e2e \
--phones_dict=dump/phone_id_map.txt \
--speaker_dict=dump/speaker_id_map.txt \
--spk_id=0 \
--inference_dir=${train_output_path}/inference
fi

@ -0,0 +1 @@
../../../aishell3/tts3/local/train.sh

@ -0,0 +1 @@
../../csmsc/tts3/path.sh

@ -0,0 +1,38 @@
#!/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_140000.pdz
# 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} || 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 ] && [ ${stop_stage} -ge 2 ]; then
# synthesize, vocoder is pwgan by default
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize.sh ${conf_path} ${train_output_path} ${ckpt_name} || exit -1
fi
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
# synthesize_e2e, vocoder is pwgan by default
CUDA_VISIBLE_DEVICES=${gpus} ./local/synthesize_e2e.sh ${conf_path} ${train_output_path} ${ckpt_name} || exit -1
fi

@ -5,4 +5,4 @@ python3 ${BIN_DIR}/../PTQ_static.py \
--dev-metadata=dump/dev/norm/metadata.jsonl \
--inference_dir ${train_output_path}/inference \
--model_name ${model_name} \
--onnx_forma=True
--onnx_format=True

@ -147,14 +147,14 @@ optional arguments:
The pretrained model can be downloaded here:
- [vits_csmsc_ckpt_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/vits/vits_csmsc_ckpt_1.1.0.zip) (add_blank=true)
- [vits_csmsc_ckpt_1.4.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/vits/vits_csmsc_ckpt_1.4.0.zip) (add_blank=true)
VITS checkpoint contains files listed below.
```text
vits_csmsc_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
vits_csmsc_ckpt_1.4.0
├── default.yaml # default config used to train vitx
├── phone_id_map.txt # phone vocabulary file when training vits
└── snapshot_iter_150000.pdz # model parameters and optimizer states
```
ps: This ckpt is not good enough, a better result is training
@ -168,9 +168,9 @@ add_blank=true
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize_e2e.py \
--config=vits_csmsc_ckpt_1.1.0/default.yaml \
--ckpt=vits_csmsc_ckpt_1.1.0/snapshot_iter_333000.pdz \
--phones_dict=vits_csmsc_ckpt_1.1.0/phone_id_map.txt \
--config=vits_csmsc_ckpt_1.4.0/default.yaml \
--ckpt=vits_csmsc_ckpt_1.4.0/snapshot_iter_150000.pdz \
--phones_dict=vits_csmsc_ckpt_1.4.0/phone_id_map.txt \
--output_dir=exp/default/test_e2e \
--text=${BIN_DIR}/../sentences.txt \
--add-blank=${add_blank}

@ -2,7 +2,7 @@ train_output_path=$1
model_name=$2
python3 ${BIN_DIR}/../../PTQ_static.py \
--dev-metadata=dump/dev/norm/metadata.jsonl \
--dev-metadata=dump/dev/raw/metadata.jsonl \
--inference_dir ${train_output_path}/inference \
--model_name ${model_name} \
--onnx_format=True

@ -42,14 +42,18 @@ if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/train/raw/metadata.jsonl \
--dumpdir=dump/train/norm \
--stats=dump/train/feats_stats.npy
--stats=dump/train/feats_stats.npy \
--skip-wav-copy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/dev/raw/metadata.jsonl \
--dumpdir=dump/dev/norm \
--stats=dump/train/feats_stats.npy
--stats=dump/train/feats_stats.npy \
--skip-wav-copy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/test/raw/metadata.jsonl \
--dumpdir=dump/test/norm \
--stats=dump/train/feats_stats.npy
--stats=dump/train/feats_stats.npy \
--skip-wav-copy
fi

@ -1,64 +0,0 @@
#!/bin/bash
source path.sh
gpus=0
stage=0
stop_stage=100
source ${MAIN_ROOT}/utils/parse_options.sh || exit 1
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
python3 ${MAIN_ROOT}/paddlespeech/t2s/exps/fastspeech2/gen_gta_mel.py \
--fastspeech2-config=fastspeech2_nosil_baker_ckpt_0.4/default.yaml \
--fastspeech2-checkpoint=fastspeech2_nosil_baker_ckpt_0.4/snapshot_iter_76000.pdz \
--fastspeech2-stat=fastspeech2_nosil_baker_ckpt_0.4/speech_stats.npy \
--dur-file=durations.txt \
--output-dir=dump_finetune \
--phones-dict=fastspeech2_nosil_baker_ckpt_0.4/phone_id_map.txt \
--dataset=baker \
--rootdir=~/datasets/BZNSYP/
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
python3 ${MAIN_ROOT}/utils/link_wav.py \
--old-dump-dir=dump \
--dump-dir=dump_finetune
fi
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
# get features' stats(mean and std)
echo "Get features' stats ..."
cp dump/train/feats_stats.npy dump_finetune/train/
fi
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
# normalize, dev and test should use train's stats
echo "Normalize ..."
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump_finetune/train/raw/metadata.jsonl \
--dumpdir=dump_finetune/train/norm \
--stats=dump_finetune/train/feats_stats.npy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump_finetune/dev/raw/metadata.jsonl \
--dumpdir=dump_finetune/dev/norm \
--stats=dump_finetune/train/feats_stats.npy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump_finetune/test/raw/metadata.jsonl \
--dumpdir=dump_finetune/test/norm \
--stats=dump_finetune/train/feats_stats.npy
fi
if [ ${stage} -le 4 ] && [ ${stop_stage} -ge 4 ]; then
CUDA_VISIBLE_DEVICES=${gpus} \
FLAGS_cudnn_exhaustive_search=true \
FLAGS_conv_workspace_size_limit=4000 \
python ${BIN_DIR}/train.py \
--train-metadata=dump_finetune/train/norm/metadata.jsonl \
--dev-metadata=dump_finetune/dev/norm/metadata.jsonl \
--config=conf/finetune.yaml \
--output-dir=exp/finetune \
--ngpu=1
fi

@ -0,0 +1 @@
../voc5/finetune.sh

@ -1,55 +0,0 @@
#!/bin/bash
stage=0
stop_stage=100
config_path=$1
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
# get durations from MFA's result
echo "Generate durations.txt from MFA results ..."
python3 ${MAIN_ROOT}/utils/gen_duration_from_textgrid.py \
--inputdir=./baker_alignment_tone \
--output=durations.txt \
--config=${config_path}
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
# extract features
echo "Extract features ..."
python3 ${BIN_DIR}/../preprocess.py \
--rootdir=~/datasets/BZNSYP/ \
--dataset=baker \
--dumpdir=dump \
--dur-file=durations.txt \
--config=${config_path} \
--cut-sil=True \
--num-cpu=20
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, 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 \
--stats=dump/train/feats_stats.npy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/dev/raw/metadata.jsonl \
--dumpdir=dump/dev/norm \
--stats=dump/train/feats_stats.npy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/test/raw/metadata.jsonl \
--dumpdir=dump/test/norm \
--stats=dump/train/feats_stats.npy
fi

@ -0,0 +1 @@
../../voc1/local/preprocess.sh

@ -1,13 +0,0 @@
#!/bin/bash
config_path=$1
train_output_path=$2
FLAGS_cudnn_exhaustive_search=true \
FLAGS_conv_workspace_size_limit=4000 \
python ${BIN_DIR}/train.py \
--train-metadata=dump/train/norm/metadata.jsonl \
--dev-metadata=dump/dev/norm/metadata.jsonl \
--config=${config_path} \
--output-dir=${train_output_path} \
--ngpu=1

@ -0,0 +1 @@
../../voc1/local/train.sh

@ -1,55 +0,0 @@
#!/bin/bash
stage=0
stop_stage=100
config_path=$1
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
# get durations from MFA's result
echo "Generate durations.txt from MFA results ..."
python3 ${MAIN_ROOT}/utils/gen_duration_from_textgrid.py \
--inputdir=./baker_alignment_tone \
--output=durations.txt \
--config=${config_path}
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
# extract features
echo "Extract features ..."
python3 ${BIN_DIR}/../preprocess.py \
--rootdir=~/datasets/BZNSYP/ \
--dataset=baker \
--dumpdir=dump \
--dur-file=durations.txt \
--config=${config_path} \
--cut-sil=True \
--num-cpu=20
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, 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 \
--stats=dump/train/feats_stats.npy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/dev/raw/metadata.jsonl \
--dumpdir=dump/dev/norm \
--stats=dump/train/feats_stats.npy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/test/raw/metadata.jsonl \
--dumpdir=dump/test/norm \
--stats=dump/train/feats_stats.npy
fi

@ -0,0 +1 @@
../../voc1/local/preprocess.sh

@ -1,13 +0,0 @@
#!/bin/bash
config_path=$1
train_output_path=$2
FLAGS_cudnn_exhaustive_search=true \
FLAGS_conv_workspace_size_limit=4000 \
python ${BIN_DIR}/train.py \
--train-metadata=dump/train/norm/metadata.jsonl \
--dev-metadata=dump/dev/norm/metadata.jsonl \
--config=${config_path} \
--output-dir=${train_output_path} \
--ngpu=1

@ -0,0 +1 @@
../../voc1/local/train.sh

@ -39,16 +39,19 @@ if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump_finetune/train/raw/metadata.jsonl \
--dumpdir=dump_finetune/train/norm \
--stats=dump_finetune/train/feats_stats.npy
--stats=dump_finetune/train/feats_stats.npy \
--skip-wav-copy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump_finetune/dev/raw/metadata.jsonl \
--dumpdir=dump_finetune/dev/norm \
--stats=dump_finetune/train/feats_stats.npy
--stats=dump_finetune/train/feats_stats.npy \
--skip-wav-copy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump_finetune/test/raw/metadata.jsonl \
--dumpdir=dump_finetune/test/norm \
--stats=dump_finetune/train/feats_stats.npy
--stats=dump_finetune/train/feats_stats.npy \
--skip-wav-copy
fi
if [ ${stage} -le 4 ] && [ ${stop_stage} -ge 4 ]; then

@ -1,55 +0,0 @@
#!/bin/bash
stage=0
stop_stage=100
config_path=$1
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
# get durations from MFA's result
echo "Generate durations.txt from MFA results ..."
python3 ${MAIN_ROOT}/utils/gen_duration_from_textgrid.py \
--inputdir=./baker_alignment_tone \
--output=durations.txt \
--config=${config_path}
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
# extract features
echo "Extract features ..."
python3 ${BIN_DIR}/../preprocess.py \
--rootdir=~/datasets/BZNSYP/ \
--dataset=baker \
--dumpdir=dump \
--dur-file=durations.txt \
--config=${config_path} \
--cut-sil=True \
--num-cpu=20
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, 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 \
--stats=dump/train/feats_stats.npy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/dev/raw/metadata.jsonl \
--dumpdir=dump/dev/norm \
--stats=dump/train/feats_stats.npy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/test/raw/metadata.jsonl \
--dumpdir=dump/test/norm \
--stats=dump/train/feats_stats.npy
fi

@ -0,0 +1 @@
../../voc1/local/preprocess.sh

@ -1,13 +0,0 @@
#!/bin/bash
config_path=$1
train_output_path=$2
FLAGS_cudnn_exhaustive_search=true \
FLAGS_conv_workspace_size_limit=4000 \
python ${BIN_DIR}/train.py \
--train-metadata=dump/train/norm/metadata.jsonl \
--dev-metadata=dump/dev/norm/metadata.jsonl \
--config=${config_path} \
--output-dir=${train_output_path} \
--ngpu=1

@ -0,0 +1 @@
../../voc1/local/train.sh

@ -42,14 +42,18 @@ if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
python3 ${BIN_DIR}/../gan_vocoder/normalize.py \
--metadata=dump/train/raw/metadata.jsonl \
--dumpdir=dump/train/norm \
--stats=dump/train/feats_stats.npy
--stats=dump/train/feats_stats.npy \
--skip-wav-copy
python3 ${BIN_DIR}/../gan_vocoder/normalize.py \
--metadata=dump/dev/raw/metadata.jsonl \
--dumpdir=dump/dev/norm \
--stats=dump/train/feats_stats.npy
--stats=dump/train/feats_stats.npy \
--skip-wav-copy
python3 ${BIN_DIR}/../gan_vocoder/normalize.py \
--metadata=dump/test/raw/metadata.jsonl \
--dumpdir=dump/test/norm \
--stats=dump/train/feats_stats.npy
--stats=dump/train/feats_stats.npy \
--skip-wav-copy
fi

@ -1,13 +0,0 @@
#!/bin/bash
config_path=$1
train_output_path=$2
FLAGS_cudnn_exhaustive_search=true \
FLAGS_conv_workspace_size_limit=4000 \
python ${BIN_DIR}/train.py \
--train-metadata=dump/train/norm/metadata.jsonl \
--dev-metadata=dump/dev/norm/metadata.jsonl \
--config=${config_path} \
--output-dir=${train_output_path} \
--ngpu=1

@ -0,0 +1 @@
../../voc1/local/train.sh

@ -1,12 +0,0 @@
#!/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=1 \
--phones-dict=dump/phone_id_map.txt

@ -0,0 +1 @@
../../../csmsc/tts0/local/train.sh

@ -1,13 +0,0 @@
#!/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=tacotron2
export BIN_DIR=${MAIN_ROOT}/paddlespeech/t2s/exps/${MODEL}

@ -0,0 +1 @@
../../csmsc/tts0/path.sh

@ -1,12 +0,0 @@
#!/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=1 \
--phones-dict=dump/phone_id_map.txt

@ -0,0 +1 @@
../../../csmsc/tts3/local/train.sh

@ -1,13 +0,0 @@
#!/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}

@ -0,0 +1 @@
../../csmsc/tts3/path.sh

@ -42,14 +42,18 @@ if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/train/raw/metadata.jsonl \
--dumpdir=dump/train/norm \
--stats=dump/train/feats_stats.npy
--stats=dump/train/feats_stats.npy \
--skip-wav-copy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/dev/raw/metadata.jsonl \
--dumpdir=dump/dev/norm \
--stats=dump/train/feats_stats.npy
--stats=dump/train/feats_stats.npy \
--skip-wav-copy
python3 ${BIN_DIR}/../normalize.py \
--metadata=dump/test/raw/metadata.jsonl \
--dumpdir=dump/test/norm \
--stats=dump/train/feats_stats.npy
--stats=dump/train/feats_stats.npy \
--skip-wav-copy
fi

@ -1,14 +0,0 @@
#!/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 ${BIN_DIR}/../synthesize.py \
--config=${config_path} \
--checkpoint=${train_output_path}/checkpoints/${ckpt_name} \
--test-metadata=dump/test/norm/metadata.jsonl \
--output-dir=${train_output_path}/test \
--generator-type=pwgan

@ -0,0 +1 @@
../../../csmsc/voc1/local/synthesize.sh

@ -1,13 +0,0 @@
#!/bin/bash
config_path=$1
train_output_path=$2
FLAGS_cudnn_exhaustive_search=true \
FLAGS_conv_workspace_size_limit=4000 \
python ${BIN_DIR}/train.py \
--train-metadata=dump/train/norm/metadata.jsonl \
--dev-metadata=dump/dev/norm/metadata.jsonl \
--config=${config_path} \
--output-dir=${train_output_path} \
--ngpu=1

@ -0,0 +1 @@
../../../csmsc/voc1/local/train.sh

@ -1,13 +0,0 @@
#!/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=parallelwave_gan
export BIN_DIR=${MAIN_ROOT}/paddlespeech/t2s/exps/gan_vocoder/${MODEL}

@ -0,0 +1 @@
../../csmsc/voc1/path.sh

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