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120 changes: 120 additions & 0 deletions .gitignore
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# Byte-compiled / optimized / DLL files
__pycache__/
*.py[cod]
*$py.class

# C extensions
*.so

# Distribution / packaging
.Python
build/
develop-eggs/
dist/
downloads/
eggs/
.eggs/
lib/
lib64/
parts/
sdist/
var/
wheels/
*.egg-info/
.installed.cfg
*.egg
MANIFEST

# PyInstaller
# Usually these files are written by a python script from a template
# before PyInstaller builds the exe, so as to inject date/other infos into it.
*.manifest
*.spec

# Installer logs
pip-log.txt
pip-delete-this-directory.txt

# Unit test / coverage reports
htmlcov/
.tox/
.coverage
.coverage.*
.cache
nosetests.xml
coverage.xml
*.cover
.hypothesis/
.pytest_cache/

# Translations
*.mo
*.pot

# Django stuff:
*.log
local_settings.py
db.sqlite3

# Flask stuff:
instance/
.webassets-cache

# Scrapy stuff:
.scrapy

# Sphinx documentation
docs/_build/

# PyBuilder
target/

# Jupyter Notebook
.ipynb_checkpoints

# pyenv
.python-version

# celery beat schedule file
celerybeat-schedule

# SageMath parsed files
*.sage.py

# Environments
.env
.venv
env/
venv/
ENV/
env.bak/
venv.bak/

# Spyder project settings
.spyderproject
.spyproject

# Rope project settings
.ropeproject

# mkdocs documentation
/site

# mypy
.mypy_cache/

__pycache__
.vscode
.DS_Store

# MFA
montreal-forced-aligner/

# data, checkpoint, and models
raw_data/
output/
*.npy
TextGrid/
hifigan/*.pth.tar
*.out
deepspeaker/pretrained_models/*
11 changes: 11 additions & 0 deletions CITATION.cff
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cff-version: 1.0.0
message: "If you use this software, please cite it as below."
authors:
- family-names: "Lee"
given-names: "Keon"
orcid: "https://orcid.org/0000-0001-9028-1018"
title: "Comprehensive-Transformer-TTS"
version: 0.1.0
doi: ___
date-released: 2021-08-25
url: "https://github.com/keonlee9420/Comprehensive-Transformer-TTS"
44 changes: 44 additions & 0 deletions Dockerfile
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FROM nvcr.io/nvidia/cuda:11.1.1-cudnn8-devel-ubuntu18.04
ARG UID
ARG USER_NAME

WORKDIR /workspace

RUN apt-get update && apt-get install -y --no-install-recommends \
apt-utils \
build-essential \
ca-certificates \
curl \
cmake \
ffmpeg \
git \
python3-pip \
python3-setuptools \
python3-dev \
sudo \
ssh \
unzip \
vim \
wget && rm -rf /var/lib/apt/lists/*

# RUN curl -o /tmp/miniconda.sh -sSL http://repo.continuum.io/miniconda/Miniconda3-py37_4.9.2-Linux-x86_64.sh && \
# bash /tmp/miniconda.sh -bfp /usr/local && \
# rm -rf /tmp/miniconda.sh
# RUN conda update -y conda

COPY requirements.txt requirements.txt

RUN pip3 install --upgrade pip setuptools wheel
RUN pip3 install -r requirements.txt

RUN adduser $USER_NAME --u $UID --quiet --gecos "" --disabled-password && \
echo "$USER_NAME ALL=(root) NOPASSWD:ALL" > /etc/sudoers.d/$USER_NAME && \
chmod 0440 /etc/sudoers.d/$USER_NAME

RUN echo "PasswordAuthentication yes" >> /etc/ssh/sshd_config
RUN echo "PermitEmptyPasswords yes" >> /etc/ssh/sshd_config
RUN echo "UsePAM no" >> /etc/ssh/sshd_config

USER $USER_NAME

EXPOSE 6006 6007 6008 6009
21 changes: 21 additions & 0 deletions LICENSE
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MIT License

Copyright (c) 2021 Keon Lee

Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to deal
in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:

The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.

THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
SOFTWARE.
163 changes: 163 additions & 0 deletions README.md
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# Comprehensive-Transformer-TTS - PyTorch Implementation

A Non-Autoregressive Transformer based TTS, supporting a family of SOTA transformers with supervised and unsupervised duration modelings. This project grows with the research community, **aiming to achieve the ultimate TTS**.

### Transformers
- [x] [Fastformer: Additive Attention Can Be All You Need](https://arxiv.org/abs/2108.09084) (Wu et al., 2021)
- [ ] [Long-Short Transformer: Efficient Transformers for Language and Vision](https://arxiv.org/abs/2107.02192) (Zhu et al., 2021)
- [x] [Conformer: Convolution-augmented Transformer for Speech Recognition](https://arxiv.org/abs/2005.08100) (Gulati et al., 2020)
- [ ] [Reformer: The Efficient Transformer](https://arxiv.org/abs/2001.04451) (Kitaev et al., 2020)
- [x] [Attention Is All You Need](https://arxiv.org/abs/1706.03762) (Vaswani et al., 2017)

### Supervised Duration Modelings
- [x] [FastSpeech 2: Fast and High-Quality End-to-End Text to Speech](https://arxiv.org/abs/2006.04558) (Ren et al., 2020)

### Unsupervised Duration Modelings
- [x] [One TTS Alignment To Rule Them All](https://arxiv.org/abs/2108.10447) (Badlani et al., 2021)

### Transformer Performance Comparison on LJSpeech (1 TITAN RTX 24G / 16 batch size)
| Model | Memory Usage | Training Time (1K steps) |
| --- | ----------- | ----- |
|Fastformer (lucidrains')|10531MiB / 24220MiB|4m 25s
|Fastformer (wuch15's)|10515MiB / 24220MiB|4m 45s
|Long-Short Transformer|-|-
|Conformer|18903MiB / 24220MiB|7m 4s
|Reformer|-|-
|Transformer|7909MiB / 24220MiB|4m 51s

Toggle type of building blocks by
```yaml
# In the model.yaml
block_type: "transformer" # ["transformer", "fastformer", "conformer"]
```
Toggle type of duration modelings by
```yaml
# In the model.yaml
duration_modeling:
learn_alignment: True # for unsupervised modeling, False for supervised modeling
```
# Quickstart
***DATASET*** refers to the names of datasets such as `LJSpeech` and `VCTK` in the following documents.

## Dependencies
You can install the Python dependencies with
```
pip3 install -r requirements.txt
```
Also, `Dockerfile` is provided for `Docker` users.
## Inference
You have to download the [pretrained models](https://drive.google.com/drive/folders/1xEOVbv3PLfGX8EgEkzg1014c9h8QMxQ-?usp=sharing) and put them in `output/ckpt/DATASET/`. The model is trained on LJSpeech with unsupervised duration modeling under transformer building blocks.
For a **single-speaker TTS**, run
```
python3 synthesize.py --text "YOUR_DESIRED_TEXT" --restore_step RESTORE_STEP --mode single --dataset DATASET
```
For a **multi-speaker TTS**, run
```
python3 synthesize.py --text "YOUR_DESIRED_TEXT" --speaker_id SPEAKER_ID --restore_step RESTORE_STEP --mode single --dataset DATASET
```
The dictionary of learned speakers can be found at `preprocessed_data/DATASET/speakers.json`, and the generated utterances will be put in `output/result/`.
## Batch Inference
Batch inference is also supported, try
```
python3 synthesize.py --source preprocessed_data/DATASET/val.txt --restore_step RESTORE_STEP --mode batch --dataset DATASET
```
to synthesize all utterances in `preprocessed_data/DATASET/val.txt`.
## Controllability
The pitch/volume/speaking rate of the synthesized utterances can be controlled by specifying the desired pitch/energy/duration ratios.
For example, one can increase the speaking rate by 20 % and decrease the volume by 20 % by
```
python3 synthesize.py --text "YOUR_DESIRED_TEXT" --restore_step RESTORE_STEP --mode single --dataset DATASET --duration_control 0.8 --energy_control 0.8
```
Add ***--speaker_id SPEAKER_ID*** for a multi-speaker TTS.
# Training
## Datasets
The supported datasets are
- [LJSpeech](https://keithito.com/LJ-Speech-Dataset/): a **single-speaker** English dataset consists of 13100 short audio clips of a female speaker reading passages from 7 non-fiction books, approximately 24 hours in total.
- [VCTK](https://datashare.ed.ac.uk/handle/10283/3443): The CSTR VCTK Corpus includes speech data uttered by 110 English speakers (**multi-speaker TTS**) with various accents. Each speaker reads out about 400 sentences, which were selected from a newspaper, the rainbow passage and an elicitation paragraph used for the speech accent archive.
Any of both **single-speaker TTS** dataset (e.g., [Blizzard Challenge 2013](https://www.synsig.org/index.php/Blizzard_Challenge_2013)) and **multi-speaker TTS** dataset (e.g., [LibriTTS](https://openslr.org/60/)) can be added following LJSpeech and VCTK, respectively. Moreover, **your own language and dataset** can be adapted following [here](https://github.com/keonlee9420/Expressive-FastSpeech2).
## Preprocessing
- For a **multi-speaker TTS** with external speaker embedder, download [ResCNN Softmax+Triplet pretrained model](https://drive.google.com/file/d/1F9NvdrarWZNktdX9KlRYWWHDwRkip_aP) of [philipperemy's DeepSpeaker](https://github.com/philipperemy/deep-speaker) for the speaker embedding and locate it in `./deepspeaker/pretrained_models/`.
- Run
```
python3 prepare_align.py --dataset DATASET
```
for some preparations.
For the forced alignment, [Montreal Forced Aligner](https://montreal-forced-aligner.readthedocs.io/en/latest/) (MFA) is used to obtain the alignments between the utterances and the phoneme sequences.
Pre-extracted alignments for the datasets are provided [here](https://drive.google.com/drive/folders/1fizpyOiQ1lG2UDaMlXnT3Ll4_j6Xwg7K?usp=sharing).
You have to unzip the files in `preprocessed_data/DATASET/TextGrid/`. Alternately, you can [run the aligner by yourself](https://montreal-forced-aligner.readthedocs.io/en/latest/aligning.html).
After that, run the preprocessing script by
```
python3 preprocess.py --dataset DATASET
```
## Training
Train your model with
```
python3 train.py --dataset DATASET
```
Useful options:
- Support [Automatic Mixed Precision](https://pytorch.org/tutorials/recipes/recipes/amp_recipe.html) by `--use_amp` argument.
- Assume single-node multi-GPU training. To use a specific GPU, specify `CUDA_VISIBLE_DEVICES=<GPU_ID>` at the beginning of the above command.
# TensorBoard
Use
```
tensorboard --logdir output/log
```
to serve TensorBoard on your localhost.
<!-- The loss curves, synthesized mel-spectrograms, and audios are shown.
![](./img/tensorboard_loss.png)
![](./img/tensorboard_spec.png)
![](./img/tensorboard_audio.png) -->
# Notes
- Both phoneme-level and frame-level variance are supported in both supervised and unsupervised duration modeling.
- Note that there are no pre-extracted phoneme-level variance features in unsupervised duration modeling.
- Convolutional embedding is used as [StyleSpeech](https://github.com/keonlee9420/StyleSpeech) for phoneme-level variance in unsupervised duration modeling. Otherwise, bucket-based embedding is used as [FastSpeech2](https://github.com/ming024/FastSpeech2).
- Unsupervised duration modeling in phoneme-level will take longer time than frame-level since the additional computation of phoneme-level variance is activated at runtime.
- Two options for embedding for the **multi-speaker TTS** setting: training speaker embedder from scratch or using a pre-trained [philipperemy's DeepSpeaker](https://github.com/philipperemy/deep-speaker) model (as [STYLER](https://github.com/keonlee9420/STYLER) did). You can toggle it by setting the config (between `'none'` and `'DeepSpeaker'`).
- DeepSpeaker on VCTK dataset shows clear identification among speakers. The following figure shows the T-SNE plot of extracted speaker embedding.
<p align="center">
<img src="./preprocessed_data/VCTK/spker_embed_tsne.png" width="40%">
</p>
- For vocoder, **HiFi-GAN** and **MelGAN** are supported.
# Citation
Please cite this repository by the "[Cite this repository](https://github.blog/2021-08-19-enhanced-support-citations-github/)" of **About** section (top right of the main page).
# References
- [ming024's FastSpeech2](https://github.com/ming024/FastSpeech2)
- [wuch15's Fastformer](https://github.com/wuch15/Fastformer)
- [lucidrains' fast-transformer-pytorch](https://github.com/lucidrains/fast-transformer-pytorch)
- [sooftware's conformer](https://github.com/sooftware/conformer)
- [NVIDIA' NeMo](https://github.com/NVIDIA/NeMo): special thanks to [Onur Babacan](https://github.com/babua) and [Rafael Valle](https://github.com/rafaelvalle) for unsupervised duration modeling.
3 changes: 3 additions & 0 deletions audio/__init__.py
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import audio.tools
import audio.stft
import audio.audio_processing
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