PyTorch implementation of DiffGAN-TTS: High-Fidelity and Efficient Text-to-Speech with Denoising Diffusion GANs
Audio samples are available at /demo.
DATASET refers to the names of datasets such as LJSpeech
and VCTK
in the following documents.
MODEL refers to the types of model (choose from 'naive', 'aux', 'shallow').
You can install the Python dependencies with
pip3 install -r requirements.txt
You have to download the pretrained models and put them in
output/ckpt/DATASET_naive/
for 'naive' model.output/ckpt/DATASET_shallow/
for 'shallow' model. Please note that the checkpoint of the 'shallow' model contains both 'shallow' and 'aux' models, and these two models will share all directories except results throughout the whole process.For a single-speaker TTS, run
python3 synthesize.py --text "YOUR_DESIRED_TEXT" --model MODEL --restore_step RESTORE_STEP --mode single --dataset DATASET
For a multi-speaker TTS, run
python3 synthesize.py --text "YOUR_DESIRED_TEXT" --model MODEL --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 is also supported, try
python3 synthesize.py --source preprocessed_data/DATASET/val.txt --model MODEL --restore_step RESTORE_STEP --mode batch --dataset DATASET
to synthesize all utterances in preprocessed_data/DATASET/val.txt
.
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" --model MODEL --restore_step RESTORE_STEP --mode single --dataset DATASET --duration_control 0.8 --energy_control 0.8
Please note that the controllability is originated from FastSpeech2 and not a vital interest of DiffGAN-TTS.
The supported datasets are
LJSpeech: 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: 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.
./deepspeaker/pretrained_models/
.Run
python3 prepare_align.py --dataset DATASET
for some preparations.
For the forced alignment, Montreal Forced Aligner (MFA) is used to obtain the alignments between the utterances and the phoneme sequences.
Pre-extracted alignments for the datasets are provided here.
You have to unzip the files in preprocessed_data/DATASET/TextGrid/
. Alternately, you can run the aligner by yourself.
After that, run the preprocessing script by
python3 preprocess.py --dataset DATASET
You can train three types of model: 'naive', 'aux', and 'shallow'.
Training Naive Version ('naive'):
Train the naive version with
python3 train.py --model naive --dataset DATASET
Training Basic Acoustic Model for Shallow Version ('aux'):
To train the shallow version, we need a pre-trained FastSpeech2. The below command will let you train the FastSpeech2 modules, including Auxiliary (Mel) Decoder.
python3 train.py --model aux --dataset DATASET
Training Shallow Version ('shallow'):
To leverage pre-trained FastSpeech2, including Auxiliary (Mel) Decoder, you must pass --restore_step
with the final step of auxiliary FastSpeech2 training as the following command.
python3 train.py --model shallow --restore_step RESTORE_STEP --dataset DATASET
For example, if the last checkpoint is saved at 200000 steps during the auxiliary training, you have to set --restore_step
with 200000
. Then it will load and freeze the aux model and then continue the training under the active shallow diffusion mechanism.
Use
tensorboard --logdir output/log/DATASET
to serve TensorBoard on your localhost. The loss curves, synthesized mel-spectrograms, and audios are shown.
lambda_fm
is fixed to a scala value since the dynamically scaled scalar computed as L_recon/L_fm makes the model explode.'none'
and 'DeepSpeaker'
).
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