⚡ FINETUNE WAV2VEC 2.0 FOR SPEECH RECOGNITION

Table of contents

1. Documentation
2. Available Features
3. Installation
4. Train
5. Inference
6. Logs and Visualization
7. Citation
8. Vietnamese

Documentation

Suppose you need a simple way to fine-tune the Wav2vec 2.0 model for the task of Speech Recognition on your datasets, then you came to the right place.
All documents related to this repo can be found here:

• Wav2vec2ForCTC
• Tutorial

Available Features

• Multi-GPU training
• Automatic Mix Precision
• Push to Huggingface Hub

Installation

pip install -r requirements.txt

Train

1. Prepare your dataset
   • Your dataset can be in .txt or .csv format.
   • path and transcript columns are compulsory. The path column contains the paths to your stored audio files, depending on your dataset location, it can be either absolute paths or relative paths. The transcript column contains the corresponding transcripts to the audio paths.
   • Check out our example files for more information.
   • Important: Ignoring these following notes is still OK but can hurt the performance.
     • Make sure that your transcript contains words only. Numbers should be converted into words and special characters such as r'[,?.!\-;:"“%\'�]' are removed by default, but you can change them in the base_dataset.py if your transcript is not clean enough.
     • If your transcript contains special tokens like bos_token, eos_token, unk_token (eg: <unk>, [unk],...) or pad_token (eg: <pad>, [pad],...)). Please specify it in the config.toml otherwise the Tokenizer can't recognize them.
2. Configure the config.toml file: Pay attention to the pretrained_path argument, it loads "facebook/wav2vec2-base" pre-trained model from Facebook by default. If you wish to pre-train wav2vec2 on your dataset, check out this REPO.
3. Run
   • Start training from scratch:

     python train.py -c config.toml
     

   • Resume:

     python train.py -c config.toml -r
     

   • Load specific model and start training:

     python train.py -c config.toml -p path/to/your/model.tar
     

Inference

We provide an inference script that can transcribe a given audio file or even a list of audio files. Please take a look at the arguments below, especially the -f TEST_FILEPATH and the -s HUGGINGFACE_FOLDER arguments:

usage: inference.py [-h] -f TEST_FILEPATH [-s HUGGINGFACE_FOLDER] [-m MODEL_PATH] [-d DEVICE_ID]

ASR INFERENCE ARGS

optional arguments:
  -h, --help            show this help message and exit
  -f TEST_FILEPATH, --test_filepath TEST_FILEPATH
                        It can be either the path to your audio file (.wav, .mp3) or a text file (.txt) containing a list of audio file paths.
  -s HUGGINGFACE_FOLDER, --huggingface_folder HUGGINGFACE_FOLDER
                        The folder where you stored the huggingface files. Check the <local_dir> argument of [huggingface.args] in config.toml. Default
                        value: "huggingface-hub".
  -m MODEL_PATH, --model_path MODEL_PATH
                        Path to the model (.tar file) in saved/<project_name>/checkpoints. If not provided, default uses the pytorch_model.bin in the
                        <HUGGINGFACE_FOLDER>
  -d DEVICE_ID, --device_id DEVICE_ID
                        The device you want to test your model on if CUDA is available. Otherwise, CPU is used. Default value: 0

Transcribe an audio file:

python inference.py \
    -f path/to/your/audio/file.wav(.mp3) \
    -s huggingface-hub

# output example:
>>> transcript: Hello World 

Transcribe a list of audio files. Check the input file test.txt and the output file transcript_test.txt (which will be stored in the same folder as the input file):

python inference.py \
    -f path/to/your/test.txt \
    -s huggingface-hub

Logs and Visualization

The logs during the training will be stored, and you can visualize it using TensorBoard by running this command:

# specify the <project_name> in config.json
tensorboard --logdir ~/saved/<project_name>

# specify a port 8080
tensorboard --logdir ~/saved/<project_name> --port 8080

Citation

@software{Duy_Khanh_Finetune_Wav2vec_2_0_2022,
    author = {Duy Khanh, Le},
    doi = {10.5281/zenodo.6540979},
    month = {5},
    title = {{Finetune Wav2vec 2.0 For Speech Recognition}},
    url = {https://github.com/khanld/ASR-Wa2vec-Finetune},
    year = {2022}
}

Vietnamese

Please take a look here for Vietnamese people who want to train on public datasets like VIOS, COMMON VOICE, FOSD, and VLSP.