VoiceFlow: Efficient Text-to-Speech with Rectified Flow Matching

This is the official implementation of our ICASSP 2024 paper VoiceFlow.

Environment Setup

This repo is tested on python 3.9 on Linux. You can set up the environment with conda

# Install required packages
conda create -n vflow python==3.9  # or any name you like
conda activate vflow
pip install -r requirements.txt

# Then, set PATH
source path.sh  # change the env name in it if you don't use "vflow"

# Install monotonic_align for MAS
cd model/monotonic_align
python setup.py build_ext --inplace

Note that to avoid the trouble of installing torchdyn, we directly copy the torchdyn 1.0.6 version here locally at torchdyn/.

The following process may also need bash and perl commands in your environment.

Data Preparation

This repo relies on Kaldi-style data organization. All data description files should be put in subdirectories in data/. See data/ljspeech/example for a basic example. In this example, the following plain text files are necessary:

1. wav.scp: organized as utt /path/to/wav.
2. utts.list: every line specifies an utterance. This can be obtained by cut -d ' ' -f 1 wav.scp > utts.list.
3. utt2spk: organized as utt spk_name.
4. text and phn_duration: specifies the phoneme sequence and the corresponding integer durations (in frames). Also, there is a data/ljspeech/phones.txt file to specify all the phones together with their indexes in dictionary.

For LJSpeech, we provide the processed file online. You can download it and unzip to data/ljspeech/{train,val}. If you want to train on your own dataset, you might have to create these files yourself (or change the data loading strategy).

After having these manifest files, please do the following to extract mel-spectrogram for training:

bash extract_fbank.sh --stage 0 --stop_stage 2 --nj 16
# nj: number of parallel jobs. 
# Have a look into the script if you need to change something
# Bash variables before "parse_options.sh" can be passed by CLI, e.g. "--key value".

Note that we default to use 16kHz data here. This will create feats/fbank and feats/normed_fbank, where Kaldi-style scp and ark files store the mel-spectrogram data. The normed features will be used for training.

If you want to use speaker-IDs (like LJSpeech, instead of using pretrained speaker embeddings such as xvectors) for training, please run:

make_utt2spk_id.py data/ljspeech/train/utt2spk data/ljspeech/val/utt2spk
# You can add more files in CLI. Will write utt2num_frames in the same directory to these files.

Training

Configurations for training is stored as yaml file in configs/. Data manifests and features for training and validation set will be specified in those yaml files. You will need to change double-quoted file paths there if you need to train on your own data.

Then, training is performed by

python train.py -c configs/${your_yaml} -m ${model_name}
# e.g. python train.py -c configs/lj_16k_gt_dur.yaml -m lj_16k_gt_dur

It will create logs/${model_name} for logging and checkpointing.

Several notes:

• By default, the program performs EMA to average weights. Weights with or without EMA will both be saved.
• By default, the program will try to find the latest checkpoint for resuming. EMA checkpoints are prior to non-EMA checkpoints.
• You can set use_gt_dur to false to turn on MAS algorithm. In this setting, it is better to set add_blank to true.

Generate Data for ReFlow and Perform Reflow

After training the model to some degree, it can be ready for flow rectification process. Flow rectification requires to generate data using the trained model and use the (noise, data) pair to train the model again. As this process should always involve the whole training dataset, it is recommended to run on multiple GPUs for parallel decoding. We provide a script to do this:

# Set CUDA_VISIBLE_DEVICES, or the program will use all available GPUs.
python generate_for_reflow.py -c configs/${your_yaml} -m ${model_name} \
                              --EMA --max-utt-num 100000000 \
                              --dataset train \
                              --solver euler -t 10 \
                              --gt-dur
# --EMA specifies to load EMA checkpoint (latest)
# --max-utt-num sets the number of utterances to decode (in this case, arbitrarily high)
# --solver euler -t 10 specifies the solver and timesteps. Could be adaptive solvers like dopri5.
# --gt-dur forces the model to use ground truth duration for decoding.

This will create synthetic_wav/${model_name}/generate_for_reflow/train for storage. noise.scp together with feats.scp will be stored. After decoding the training set, you can also decode validation set by --dataset val.

Then, specify the paths to these feats.scp and noise.scp in a new configuration yaml, like in the lj_16k_gt_dur_reflow.yaml:

perform_reflow: true
...
data:
    train:
        feats_scp: "synthetic_wav/lj_16k_gt_dur/train/feats.scp"
        noise_scp: "synthetic_wav/lj_16k_gt_dur/train/noise.scp"
...

Now it is ready for training again in ReFlow, with the same script in training but new yaml config files. Feel free to copy a trained model to the new log dir for resuming. Also, it is possible to change the model structure and train from scratch on the reflow data.

Inference

Similar to "generate data for reflow", model inference can be done by

python inference_dataset.py -c configs/${your_yaml} -m ${model_name} --EMA \
                          --solver euler -t 10

This will synthesize mel-spectrograms for the validation set in your config, storing them at synthetic_wav/${model_name}/tts_gt_spk/feats.scp. Speaker, speed and temperature can be specified; see tools.get_hparams_decode() function for complete set of options.

Inference can then be done in the hifigan/ directory. Please refer to the README there.

Acknowledgement

During the development, the following repositories were referred to:

• Kaldi and UniCATS-CTX-vec2wav for most utility scripts in utils/.
• GradTTS, where most of the model architecture and training pipelines are adopted.
• VITS, whose distributed bucket sampler is used.
• CFM, for the ODE samplers.

💡Easter Eggs & Citation

This repository also contains some experimental functionalities. ⚠️Warning: not guaranteed to be correct!

• Voice conversion. As GlowTTS can perform voice conversion via the disentangling property of normalizing flows, it is reasonable that flow matching can also perform it. Method model.tts.GradTTS.voice_conversion gives a preliminary try.

• Likelihood estimation. Differential equation-based generative models have the ability to estimate data likelihoods by the instantaneous change-of-variable formula

$$\log p_0(\boldsymbol x(0)) = \log p_1(\boldsymbol x(1)) + \int _0^1 \nabla_{\boldsymbol x} \cdot {\boldsymbol v}(\boldsymbol x(t), t)\mathrm d t$$

In practice, integral is replaced by summation, and divergence is replaced by the Skilling-Hutchinson trace estimator. See the Appendix D.2 in Song, et. al for theoretical details. I implemented this in model.tts.GradTTS.compute_likelihood.

• Optimal transport. The conditional flow matching used in this paper is not a marginally optimal transport path but only a conditionally optimal path. For the marginal optimal transport, Tong et. al introduces to sample $x_0,x_1$ together from the joint optimal transport distribution $\pi(x_0,x_1)$. I tried this in model.cfm.OTCFM, though it doe not work very well for now.
• Different estimator architectures. You can specify an estimator besides the GradLogPEstimator2d by the model.fm_net_type configuration. Currently the DiffSinger's estimator architecture is also supported. You can add more, e.g. that introduced in Matcha-TTS.
• Better alignment learning. This repo supports supervised duration modeling together with monotonic alignment search as that in GradTTS. However, there might be a better way for MAS in flow-matching TTS. model.tts.GradTTS.forward now supports beta binomial prior for alignment maps; and if you want, you can change the variable MAS_target to something else, e.g. flow-transformed noise!

Feel free to cite this work if it helps 😄

@INPROCEEDINGS{guo2024voiceflow,
  author={Guo, Yiwei and Du, Chenpeng and Ma, Ziyang and Chen, Xie and Yu, Kai},
  booktitle={ICASSP 2024 - 2024 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)}, 
  title={{VoiceFlow}: Efficient Text-To-Speech with Rectified Flow Matching}, 
  year={2024},
  volume={},
  number={},
  pages={11121-11125},
  keywords={Signal processing algorithms;Signal processing;Acoustics;Mathematical models;Vectors;Trajectory;Speech processing;Text-to-speech;flow matching;rectified flow;efficiency;speed-quality tradeoff},
  doi={10.1109/ICASSP48485.2024.10445948}
}