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SPEECH-CONDITIONED FACE GENERATION USING GENERATIVE ADVERSARIAL NETWORKS

Introduction

Note: This repository holds my final year and dissertation project during my time at the Image Processing Group (Universitat Politècnica de Catalunya). To obtain a more updated code, you can visit the official repo here.

Speech is a rich biometric signal that contains information about the identity, gender and emotional state of the speaker. In this work, we explore its potential to generate face images of a speaker by conditioning a Generative Adversarial Network (GAN) with raw speech input. We propose a deep neural network that is trained from scratch in an end-to-end fashion, generating a face directly from the raw speech waveform without any additional identity information (e.g reference image or one-hot encoding). Our model is trained in a self-supervised fashion by exploiting the audio and visual signals naturally aligned in videos. With the purpose of training from video data, we present a novel dataset collected for this work, with high-quality videos of ten youtubers with notable expressiveness in both the speech and visual signals.

We used this project as baseline.

Dependencies

  • Python 2.7
  • PyTorch

This implementation only supports running with GPUs.

Data

Although having initially trained with 10 different identities, we can only publish the dataset in relation to those who have answered our request for working with their images and voice.

If you want to create your own dataset, it works as follows:

  • Run get_data.py. It requires the path of a .csv file with the following information: the youtube url Channel-URL, the name of the youtuber Name and the gender Gender. The output dataset will consist on a folder for each different youtuber specified in the .csv file as well as two folders (video/audio) in each of them containing the full video/audio downloaded.

  • Run preprocess_audio_frames.py for audio preprocessing purposes.

  • Remove any file whose name does not finish like preprocessing_wav in each audio folder.

  • Run face_detector.py specifying the dataset path, a confidence threshold needed to compute the bounding boxes and the path of the haarcascade pretrained classifier (included in this repo in /assets)

  • The useful audio snippets will be stored in DATASET-PATH/audio/NAME-OF-THE-DOWNLOADED-VIDEO_frames. The useful youtuber faces will be stored in DATASET-PATH/video/NAME-OF-THE-DOWNLOADED-VIDEO_cropped_frames

Once the dataset is build, you can store the paths for all the images/audio frames in a pickle file with:

scripts/generate_pickle.py

Include both pickle files (train/test datasets) in:

config.yaml

Training

python runtime.py

Arguments:

  • lr_D : The learning rate of the disciminator. default = 0.0004
  • lr_G : The learning rate of the generator. default = 0.0001
  • vis_screen : The visdom env name for visualization. default = gan
  • save_path : Name of the directory (inside checkpoints) where the parameters of them odel will be stored.
  • l1_coef : L1 loss coefficient in the generator loss fucntion. default=50
  • l2_coef : Feature matching coefficient in the generator loss fucntion. default=100
  • pre_trained_disc : Discriminator pre-tranined model path used for intializing training.
  • pre_trained_gen : Generator pre-tranined model path used for intializing training.
  • batch_size : Batch size. default= 64
  • num_workers: Number of dataloader workers used for fetching data. default = 8
  • epochs : Number of training epochs. default=200
  • softmax_coef: Paramete for the scale of the loss of the classifier on top of the embedding
  • image_size : Number of pixels per dimension. They are assumed to be squared. Two possible values: 64 | 128. default = 64
  • inference : Boolean for choosing whether train or test. default = False

References

If the code of this repository was useful for your research, please cite our work:

@inproceedings{wav2pix2019icassp,
  title={Wav2Pix: Speech-conditioned Face Generation 
          using Generative Adversarial Networks},
  author={Amanda Duarte, Francisco Roldan, Miquel Tubau, Janna Escur, 
          Santiago Pascual, Amaia Salvador, Eva Mohedano, Kevin McGuinness, 
           Jordi Torres and Xavier Giro-i-Nieto},
  booktitle={2019 IEEE International Conference on Acoustics, Speech 
            and Signal Processing (ICASSP)},
  year={2019},
  organization={IEEE}
}