Hui Zhang
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README.md
English | 简体中文
PaddleSpeech
PaddleSpeech is an open-source toolkit on PaddlePaddle platform for a variety of critical tasks in speech, with state-of-art and influential models.
Via the easy-to-use, efficient, flexible and scalable implementation, our vision is to empower both industrial application and academic research, including training, inference & testing modules, and deployment process. To be more specific, this toolkit features at:
- Fast and Light-weight: we provide high-speed and ultra-lightweight models that are convenient for industrial deployment.
- Rule-based Chinese frontend: our frontend contains Text Normalization (TN) and Grapheme-to-Phoneme (G2P, including Polyphone and Tone Sandhi). Moreover, we use self-defined linguistic rules to adapt Chinese context.
- Varieties of Functions that Vitalize both Industrial and Academia:
- Implementation of critical audio tasks: this toolkit contains audio functions like Speech Translation (ST), Automatic Speech Recognition (ASR), Text-To-Speech Synthesis (TTS), Voice Cloning(VC), Punctuation Restoration, etc.
- Integration of mainstream models and datasets: the toolkit implements modules that participate in the whole pipeline of the speech tasks, and uses mainstream datasets like LibriSpeech, LJSpeech, AIShell, CSMSC, etc. See also model lists for more details.
- Cross-domain application: as an extension of the application of traditional audio tasks, we combine the aforementioned tasks with other fields like NLP.
Let's install PaddleSpeech with only a few lines of code!
Note: The official name is still deepspeech. 2021/10/26
If you are using Ubuntu, PaddleSpeech can be set up with pip installation (with root privilege).
git clone https://github.com/PaddlePaddle/DeepSpeech.git
cd DeepSpeech
pip install -e .
Table of Contents
The contents of this README is as follow:
- Alternative Installation
- Quick Start
- Models List
- Tutorials
- FAQ and Contributing
- License
- Acknowledgement
Alternative Installation
The base environment in this page is
- Ubuntu 16.04
- python>=3.7
- paddlepaddle==2.1.2
If you want to set up PaddleSpeech in other environment, please see the ASR installation and TTS installation documents for all the alternatives.
Quick Start
Note: the current links to
English ASR
andEnglish TTS
are not valid.
Just a quick test of our functions: English ASR and English TTS by typing message or upload your own audio file.
Developers can have a try of our model with only a few lines of code.
A tiny ASR DeepSpeech2 model training on toy set of LibriSpeech:
cd examples/tiny/s0/
# source the environment
source path.sh
# prepare librispeech dataset
bash local/data.sh
# evaluate your ckptfile model file
bash local/test.sh conf/deepspeech2.yaml ckptfile offline
For TTS, try pretrained FastSpeech2 + Parallel WaveGAN on CSMSC:
cd examples/csmsc/tts3
# download the pretrained models and unaip them
wget https://paddlespeech.bj.bcebos.com/Parakeet/pwg_baker_ckpt_0.4.zip
unzip pwg_baker_ckpt_0.4.zip
wget https://paddlespeech.bj.bcebos.com/Parakeet/fastspeech2_nosil_baker_ckpt_0.4.zip
unzip fastspeech2_nosil_baker_ckpt_0.4.zip
# source the environment
source path.sh
# run end-to-end synthesize
FLAGS_allocator_strategy=naive_best_fit \
FLAGS_fraction_of_gpu_memory_to_use=0.01 \
python3 ${BIN_DIR}/synthesize_e2e.py \
--fastspeech2-config=fastspeech2_nosil_baker_ckpt_0.4/default.yaml \
--fastspeech2-checkpoint=fastspeech2_nosil_baker_ckpt_0.4/snapshot_iter_76000.pdz \
--fastspeech2-stat=fastspeech2_nosil_baker_ckpt_0.4/speech_stats.npy \
--pwg-config=pwg_baker_ckpt_0.4/pwg_default.yaml \
--pwg-checkpoint=pwg_baker_ckpt_0.4/pwg_snapshot_iter_400000.pdz \
--pwg-stat=pwg_baker_ckpt_0.4/pwg_stats.npy \
--text=${BIN_DIR}/../sentences.txt \
--output-dir=exp/default/test_e2e \
--inference-dir=exp/default/inference \
--device="gpu" \
--phones-dict=fastspeech2_nosil_baker_ckpt_0.4/phone_id_map.txt
If you want to try more functions like training and tuning, please see ASR getting started and TTS Basic Use.
Models List
PaddleSpeech supports a series of most popular models, summarized in released models with available pretrained models.
ASR module contains Acoustic Model and Language Model, with the following details:
Note: The
Link
should be code path rather than download links.
ASR Module Type | Dataset | Model Type | Link |
---|---|---|---|
Acoustic Model | Aishell | 2 Conv + 5 LSTM layers with only forward direction | Ds2 Online Aishell Model |
2 Conv + 3 bidirectional GRU layers | Ds2 Offline Aishell Model | ||
Encoder:Conformer, Decoder:Transformer, Decoding method: Attention + CTC | Conformer Offline Aishell Model | ||
Encoder:Conformer, Decoder:Transformer, Decoding method: Attention | Conformer Librispeech Model | ||
Librispeech | Encoder:Conformer, Decoder:Transformer, Decoding method: Attention | Conformer Librispeech Model | |
Encoder:Transformer, Decoder:Transformer, Decoding method: Attention | Transformer Librispeech Model | ||
Language Model | CommonCrawl(en.00) | English Language Model | English Language Model |
Baidu Internal Corpus | Mandarin Language Model Small | Mandarin Language Model Small | |
Mandarin Language Model Large | Mandarin Language Model Large |
PaddleSpeech TTS mainly contains three modules: Text Frontend, Acoustic Model and Vocoder. Acoustic Model and Vocoder models are listed as follow:
TTS Module Type | Model Type | Dataset | Link |
---|---|---|---|
Text Frontend | chinese-fronted | ||
Acoustic Model | Tacotron2 | LJSpeech | tacotron2-vctk |
TransformerTTS | transformer-ljspeech | ||
SpeedySpeech | CSMSC | speedyspeech-csmsc | |
FastSpeech2 | AISHELL-3 | fastspeech2-aishell3 | |
VCTK | fastspeech2-vctk | ||
LJSpeech | fastspeech2-ljspeech | ||
CSMSC | fastspeech2-csmsc | ||
Vocoder | WaveFlow | LJSpeech | waveflow-ljspeech |
Parallel WaveGAN | LJSpeech | PWGAN-ljspeech | |
VCTK | PWGAN-vctk | ||
CSMSC | PWGAN-csmsc | ||
Voice Cloning | GE2E | AISHELL-3, etc. | ge2e |
GE2E + Tactron2 | AISHELL-3 | ge2e-tactron2-aishell3 |
Tutorials
Normally, Speech SoTA gives you an overview of the hot academic topics in speech. If you want to focus on the two tasks in PaddleSpeech, you will find the following guidelines are helpful to grasp the core ideas.
The original ASR module is based on Baidu's DeepSpeech which is an independent product named DeepSpeech. However, the toolkit aligns almost all the SoTA modules in the pipeline. Specifically, these modules are
The TTS module is originally called Parakeet, and now merged with DeepSpeech. If you are interested in academic research about this function, please see TTS research overview. Also, this document is a good guideline for the pipeline components.
FAQ and Contributing
You are warmly welcome to submit questions in discussions and bug reports in issues! Also, we highly appreciate if you would like to contribute to this project!
License
PaddleSpeech is provided under the Apache-2.0 License.
Acknowledgement
PaddleSpeech depends on a lot of open source repos. See references for more information.