This is the experimental description of MnTTS2.
Kailin Liang, Bin Liu, Yifan Hu, Rui Liu *, Feilong Bao, Guanglai Gao.
This project uses conda
to manage all the dependencies, you should install anaconda if you have not done so.
# Clone the repo
git clone https://github.com/ssmlkl/MnTTS2.git
cd $PROJECT_ROOT_DIR
conda env create -f Environment/environment.yaml
conda activate mntts2
Prepare our MnTTS2 dataset in the following format.
|- mntts2/
| |- train.txt
| |- spk_01/
| |- file1.wav
| |- file1.txt
| |- ......
| |- spk_01_train.txt
| |- spk_02/
| |- file1.wav
| |- file1.txt
| |- ......
| |- spk_02_train.txt
| |- spk_03/
| |- file1.wav
| |- file1.txt
| |- ......
| |- spk_03_train.txt
Where spk_spkID_train.txt
has the following format: uttID|transcription
. This is a ljspeech-like format.
And train.txt
has the following format: spkID|uttID|transcription
.
The complete dataset is available from our multilingual corpus website.
Take speaker 01 for example.
The preprocessing has two steps:
- Preprocess audio features
- Convert characters to IDs
- Compute mel spectrograms
- Normalize mel spectrograms to [-1, 1] range
- Split the dataset into train and validation
- Compute the mean and standard deviation of multiple features from the training split
- Standardize mel spectrogram based on computed statistics
Please note that you will also need to check the mntts process(/home/anaconda3/envs/mntts2/lib/python3.8/site-packages/tensorflow_tts/processor/mntts.py) before you start. Since tacotron2 needs to read mntts2/spk_spkID/train.txt and fastspeech2 needs to read mntts2/train.txt. Here is an example modification: Before starting tacotron work, you need modify mntts.py to look like the following code block.
positions = {
"wave_file": 0,
"text": 1,
"text_norm": 1,
}
train_f_name: str = "spk_01_train.txt"
def create_items(self):
if self.data_dir:
with open(
os.path.join(self.data_dir, self.train_f_name), encoding="utf-8"
) as f:
self.items = [self.split_line(self.data_dir, line, "|") for line in f]
def split_line(self, data_dir, line, split):
parts = line.strip().split(split)
wave_file = parts[self.positions["wave_file"]]
text_norm = parts[self.positions["text_norm"]]
wav_path = os.path.join(data_dir, f"{wave_file}.wav")
speaker_name = "spk_01"
return text_norm, wav_path, speaker_name
After completing the above work, you are ready to start the tacotron2 work officially.
To reproduce the steps above:
CUDA_VISIBLE_DEVICES=0 tensorflow-tts-preprocess \
--rootdir ./MnTTS2/spk_01 \
--outdir ./tacotron2_dump/spk_01 \
--config preprocess/mntts2_preprocess.yaml \
--dataset mntts
CUDA_VISIBLE_DEVICES=0 tensorflow-tts-normalize \
--rootdir ./tacotron2_dump/spk_01 \
--outdir ./tacotron2_dump/spk_01 \
--config preprocess/mntts2_preprocess.yaml \
--dataset mntts
Based on the script train_tacotron2.py
.
This example code show you how to train Tactron-2 from scratch with Tensorflow 2 based on custom training loop and tf.function.
Take speaker 01 for example: First you need to refer to the instructions in 2) and modify the mntts.py file.
CUDA_VISIBLE_DEVICES=0 python examples/tacotron2/train_tacotron2.py \
--train-dir ./tacotron2_dump/spk_01/train/ \
--dev-dir ./tacotron2_dump/spk_01/valid/ \
--outdir ./examples/tacotron2/exp/train.tacotron2.v1.spk_01/ \
--config ./examples/tacotron2/conf/tacotron2.v1.yaml \
--use-norm 1 \
--mixed_precision 0 \
--resume ""
IF you want to use MultiGPU to training you can replace CUDA_VISIBLE_DEVICES=0
by CUDA_VISIBLE_DEVICES=0,1,2,3
for example. You also need to tune the batch_size
for each GPU (in config file) by yourself to maximize the performance. Note that MultiGPU now support for Training but not yet support for Decode.
In case you want to resume the training progress, please following below example command line:
--resume ./examples/tacotron2/exp/train.tacotron2.v1.spk_01/checkpoints/ckpt-100000
If you want to finetune a model, use --pretrained
like this with your model filename:
--pretrained pretrained.h5
Extract duration from alignments for FastSpeech
You may need to extract durations for student models like fastspeech. Here we use teacher forcing with window masking trick to extract durations from alignment maps.
Extract for valid set:
CUDA_VISIBLE_DEVICES=0 python examples/tacotron2/extract_duration.py \
--rootdir ./tacotron2_dump/spk_01/valid/ \
--outdir ./tacotron2_dump/spk_01/valid/durations/ \
--checkpoint ./examples/tacotron2/exp/train.tacotron2.v1.spk_01/checkpoints/model-100000.h5 \
--use-norm 1 \
--config ./examples/tacotron2/conf/tacotron2.v1.yaml \
--batch-size 32 \
--win-front 3 \
--win-back 3
Extract for training set:
CUDA_VISIBLE_DEVICES=0 python examples/tacotron2/extract_duration.py \
--rootdir ./tacotron2_dump/spk_01/train/ \
--outdir ./tacotron2_dump/spk_01/train/durations/ \
--checkpoint ./examples/tacotron2/exp/train.tacotron2.v1.spk_01/checkpoints/model-100000.h5 \
--use-norm 1 \
--config ./examples/tacotron2/conf/tacotron2.v1.yaml \
--batch-size 32 \
--win-front 3 \
--win-back 3
To extract postnets for training vocoder, follow above steps but with extract_postnets.py
After completing the extraction of the durations of the three speakers, the durations in the training and test sets of each speaker are collated together.
|- mntts2/
| |- durations
| |- 01_1_uttID-durations.npy
| |- ......
| |- 02_1_uttID-durations.npy
| |- ......
| |- 03_1_uttID-durations.npy
| |- ......
| |- train.txt
| |- spk_01/
| |- file1.wav
| |- file1.txt
| |- ......
| |- spk_01_train.txt
| |- spk_02/
| |- file1.wav
| |- file1.txt
| |- ......
| |- spk_02_train.txt
| |- spk_03/
| |- file1.wav
| |- file1.txt
| |- ......
| |- spk_03_train.txt
Based on the script train_fastspeech2.py
.
First you need to refer to the instructions in 2) and modify the mntts.py file. Before you are ready to start fastspeech2, you should modify mntts.py in MNTTSProcessor class to look like the following code block.
positions = {
"speaker_name": 0,
"wave_file": 1,
"text": 2,
"text_norm": 2,
}
train_f_name: str = "train.txt"
def create_items(self):
if self.data_dir:
with open(
os.path.join(self.data_dir, self.train_f_name), encoding="utf-8"
) as f:
self.items = [self.split_line(self.data_dir, line, "|") for line in f]
def split_line(self, data_dir, line, split):
parts = line.strip().split(split)
wave_file = parts[self.positions["wave_file"]]
text_norm = parts[self.positions["text_norm"]]
speaker_name = parts[self.positions["speaker_name"]]
wav_path = os.path.join(data_dir,speaker_name,f"{wave_file}.wav")
return text_norm, wav_path, speaker_name
After completing the above work, you are ready to start the FastSpeech2 work officially.
CUDA_VISIBLE_DEVICES=0 tensorflow-tts-preprocess \
--rootdir ./mntts2 \
--outdir ./fastspeech2_dump \
--config ./preprocess/mntts_preprocess.yaml \
--dataset mntts
CUDA_VISIBLE_DEVICES=0 tensorflow-tts-normalize \
--rootdir ./fastspeech2_dump \
--outdir ./fastspeech2_dump \
--config ./preprocess/mntts_preprocess.yaml \
--dataset mntts
Run fix mismatch to fix few frames difference in audio and duration files.
CUDA_VISIBLE_DEVICES=0 python examples/mfa_extraction/fix_mismatch.py \
--base_path ./fastspeech2_dump \
--trimmed_dur_path ./mntts2/trimmed-durations \
--dur_path ./mntts2/durations
Checking the dimensional alignment of ids and durations.
python toolkit/fix.py
Change below example command line to match your dataset and run.
CUDA_VISIBLE_DEVICES=0 python examples/fastspeech2_mntts2/train_fastspeech2.py \
--train-dir ./fastspeech2_dump/train/ \
--dev-dir ./fastspeech2_dump/valid/ \
--outdir ./examples/fastspeech2/exp/train.fastspeech2.v1/ \
--config ./examples/fastspeech2/conf/fastspeech2.v1.yaml \
--use-norm 1 \
--f0-stat ./fastspeech2_dump/stats_f0.npy \
--energy-stat ./fastspeech2_dump/stats_energy.npy \
--mixed_precision 1 \
--dataset_config ./preprocess/mntts_preprocess.yaml \
--dataset_stats ./fastspeech2_dump/stats.npy \
--dataset_mapping ./fastspeech2_dump/mntts_mapper.json
Take speaker 01 for example.
First, you need training generator with only stft loss:
CUDA_VISIBLE_DEVICES=0 python examples/hifigan/train_hifigan.py \
--train-dir ./tacotron2_dump/spk_01/train/ \
--dev-dir ./tacotron2_dump/spk_01/valid/ \
--outdir ./examples/hifigan/exp/train.hifigan.v1.spk_01/ \
--config ./examples/hifigan/conf/hifigan.v1.yaml \
--use-norm 1 \
--generator_mixed_precision 1 \
--resume ""
Then resume and start training generator + discriminator:
CUDA_VISIBLE_DEVICES=0 python examples/hifigan/train_hifigan.py \
--train-dir ./tacotron2_dump/spk_01/train/ \
--dev-dir ./tacotron2_dump/spk_01/valid/ \
--outdir ./examples/hifigan/exp/train.hifigan.v1.spk_01/ \
--config ./examples/hifigan/conf/hifigan.v1.yaml \
--use-norm 1 \
--resume ./examples/hifigan/exp/train.hifigan.v1.spk_01/checkpoints/ckpt-100000
You can follow below example command line to generate synthesized speech for a given text in 'prediction/spk_01/inference.txt' using Griffin-Lim and trained HiFi-GAN vocoder, take speaker 01 for example:
CUDA_VISIBLE_DEVICES=0 python examples/fastspeech2_mntts2/mntts2_inference_fastspeech2.py \
--outdir prediction/spk_01/MnTTS_inference \
--infile prediction/spk_01/inference.txt \
--tts_ckpt examples/fastspeech2/exp/train.fastspeech2.v1/checkpoints/model-200000.h5 \
--vocoder_ckpt examples/hifigan/exp/train.hifigan.v1.spk_01/checkpoints/generator-200000.h5 \
--stats_path fastspeech2_dump/stats.npy \
--dataset_config preprocess/mntts_preprocess.yaml \
--tts_config examples/fastspeech2/conf/fastspeech2.v1.yaml \
--vocoder_config examples/hifigan/conf/hifigan.v1.yaml \
--lan_json fastspeech2_dump/mntts_mapper.json \
--speaker_id 0
20 audio tracks synthesised for three speakers. Griffin-Lim vocoder, HiFi-GAN vocoder and Ground-Tuth are included, with corresponding texts given for reference: Inference Audios
Please contact the Author to access the Pre-trained Models. liangkailin98@foxmail.com
Tensorflow-TTS: https://github.com/TensorSpeech/TensorFlowTTS