Skip to content

Latest commit

 

History

History
91 lines (78 loc) · 10.7 KB

README.md

File metadata and controls

91 lines (78 loc) · 10.7 KB

Low Resource Automatic Speech Recognition (ASR)

Introduction

With the advances in Automatic Speech Recognition (ASR) for high-resource languages, this paper aims to open the door to understanding the magnitude of data needed for speech recognition tasks.

In this paper, we compare Wav2Vec2XLSR (56K hours of data) against Wav2Vec2IPA, a fine-tuned Wav2Vec2-based architecture model trained on <5 hours of data from the TIMIT speech corpus (0.5% of baseline training data). From our evaluations, we find that the baseline model outperforms the smaller data model by 8.3 percentage points with respect to Phoneme Error Rate.

In addition, to address a gap in the availability of the TIMIT speech corpus, we also introduce 2 new datasets on HuggingFace Hub:

Additionally, this research on smaller models intends to address current issues in Natural Language Processing (NLP) surrounding environmental sustainability and democratizing access to AI with the advent of small machine learning models.

Paper

Click here for a full writeup on the methods, motivation, and evaluation.

Wav2Vec2IPA Pipeline

Train_Eval_Pipeline The 3 step (1) Preprocess, (2) Fine Tune, and (3) Evaluate pipeline above are implemented through notebooks, datasets, and models.

Notebooks

  • Each of the 3 phases of the pipeline are encapsulated into notebook files
Pipeline Step Notebook
Data Preprocessing preprocessing/split_data.ipynb
Training training/fine_tune_w2v2.ipynb
Evaluation Baseline Evaluation
evaluation/eval_model_xlsr.ipynb

Wav2Vec2IPA
evaluation/eval_model_w2v2ipa.ipynb

Datasets

  • We expand on the existing HuggingFace timit-asr/timit_asr and contribute two HuggingFace datasets in this project:
Name Description
kylelovesllms/timit_asr Implementation of TIMIT dataset with Test and Train split
kylelovesllms/timit_asr_ipa Implementation of kylelovesllms/timit_asr with Validation Split and IPA Transcriptions

Models

  • Similar to datasets, the models used/trained in this project live in HuggingFace Hub
Models Description
facebook/XLSR-Wav2Vec2 Baseline Evaluation Model
facebook/wav2vec2-base Pretrained Wav2Vec2 Model
Wav2Vec2IPA and Wav2Vec2IpaTokenizer
 Fine Tuned Model Wav2Vec2-Base model trained in this repository

Training Methodology

Audio Pipeline

Implementation

Preprocess Transcriptions (Data Preprocessing)

Challenges Using Existing HuggingFace Dataset

Although HuggingFace has an implementation of the TIMIT database, timit-asr/timit_asr, there are problems that prevent us from using the HuggingFace Hub dataset directly:

  1. TIMIT transcriptions are phonetic but not IPA (TIMIT has its own transcription system closely closely related to IPA)
  2. The original TIMIT dataset contains only Train / Test dataset splits but no Validation dataset to tune hyper-parameters
  3. HuggingFace timit-asr/timit_asr uses only 1/5 of the entire speech corpus (<1 hour)
    1. Since the dataset is <5 hours in total, using 1/5 of the dataset drastically impacts model performance
  4. HuggingFace timit-asr/timit_asr uses a deprecated dataset API, requiring users to download the audio files via 3rd party zip

Creating Custom HuggingFace Dataset

To address these issues, we contribute two datasets to HuggingFaceHub:

  1. kylelovesllms/timit_asr
    1. Reimplementation of HuggingFace timit_asr with the full Train / Test dataset (5 hours total)
    2. Stores TIMIT database in native HuggingFace Hub Dataset API (Parquet format)
  2. kylelovesllms/timit_asr_ipa
    1. Builds off of timit_asr and adds:
      1. IPA transcriptions used for Wav2Vec2 Base fine-tuning
      2. Stratified Validation Dataset
        1. The validation dataset splits the Test Dataset in half following the TIMIT recommendation of having unique speakers between training and validation/test dataset
        2. Stratifies speaker population based on speaker sex and speaker dialect region

Preprocessing Scripts

Dependency Description
timit_metadata_extractors.py Handles metadata extraction (speaker sex, dialect region, audio duration) and extracts transcriptions at the sentence, word, and phoneme level
timit_ipa_translation.py Helper to timit_metadata_extractors for handling TIMIT transcription to IPA mapping
timit_dataset_splitter.py Helper to timit_metadata_extractors to stratify the Test dataset into Test and Validation

Training

  • To train the wav2vec2 IPA base model into run the evaluation/fine_tune_w2v2.ipynb
    • Note: to save the model, make sure to edit the HF_ID to your HuggingFace UserID
Dependency Description
vocab_manual.json Learnable output tokens for final softmax layer in Wav2Vec2 Architecture.

Also contains Connectionist Temporal Classification (CTC) Loss token settings (UNKnown token, PADding token, and space token)

Evaluation

  • To evaluate model performance, select one of the notebooks eval_model_w2v2ipa.ipynb to evaluate the fine tuned model and eval_model_xlsr.ipynb to evaluate the baseline mode.
  • At the time of writing, we implement vanilla Character Error Rate (CER) since the fine tuned model produces less tokens than the ground truth transcription (making CER effectively normalized CER)
Dependency Description
eval_helpers.py Removes extra symbols from baseline model to avoid mis-penalization during evaluation