Repository for fine-tuning Transformers ๐Ÿค— based seq2seq speech models in JAX/Flax.

Overview

Seq2Seq Speech in JAX

A JAX/Flax repository for combining a pre-trained speech encoder model (e.g. Wav2Vec2, HuBERT, WavLM) with a pre-trained text decoder model (e.g. GPT2, Bart) to yield a Speech Sequence-to-Sequence (Seq2Seq) model for automatic speech recognition.

The script run_flax_speech_recognition_seq2seq.py can be used to fine-tune a Speech Seq2Seq model on one of the official speech recognition datasets or a custom dataset. It makes use of the pmap JAX operator to provide model parallelism accross GPU/TPU devices.

The modelling files are based very heavily on those from Hugging Face Transformers ๐Ÿค— . This is a standalone repository to enable rapid prototyping and involvement with the community. The final modelling files and training script will be merged into Transformers ๐Ÿค— to be used with the rest of the open-source library. The final system weights will be made publicly available at huggingface.co ๐Ÿš€

Seq2SeqModel Figure 1: Speech-encoder text-decoder style Seq2Seq model.

Example Usage

To instantiate a Wav2Vec2-2-Bart model with the FlaxSpeechEncoderDecoderModel framework, run the following Python script inside the cloned repo:

from transformers import AutoFeatureExtractor, AutoTokenizer
from models.modeling_flax_speech_encoder_decoder import FlaxSpeechEncoderDecoderModel
import numpy as np

# checkpoints to leverage
encoder_id = "facebook/wav2vec2-large-lv60"
decoder_id = "facebook/bart-large"

model = FlaxSpeechEncoderDecoderModel.from_encoder_decoder_pretrained(
    encoder_id, decoder_id, encoder_add_adapter=True, decoder_from_pt=True)

model.config.decoder_start_token_id = model.config.decoder.bos_token_id
model.config.pad_token_id = model.config.decoder.pad_token_id
model.config.eos_token_id = model.config.decoder.eos_token_id
model.config.use_cache = False
model.config.processor_class = "Wav2Vec2Processor"

# check if generation works
out = model.generate(np.ones((1, 2000)))

model.save_pretrained("./")

feature_extractor = AutoFeatureExtractor.from_pretrained(encoder_id)
feature_extractor.save_pretrained("./")
tokenizer = AutoTokenizer.from_pretrained(decoder_id)
tokenizer.save_pretrained("./")

To train the model on Librispeech ASR in default precision, run the bash script provided below:

#!/usr/bin/env bash
python run_flax_speech_recognition_seq2seq.py \
        --dataset_name="librispeech_asr" \
        --model_name_or_path="./" \
        --dataset_config_name="clean" \
        --train_split_name="train.100" \
        --eval_split_name="validation" \
        --output_dir="./" \
        --preprocessing_num_workers="16" \
        --length_column_name="input_length" \
        --overwrite_output_dir \
        --num_train_epochs="5" \
        --per_device_train_batch_size="2" \
        --per_device_eval_batch_size="2" \
        --gradient_accumulation_steps="1" \
        --logging_steps="25" \
        --max_duration_in_seconds="15" \
        --max_target_length="128" \
        --generation_max_length="40" \
        --generation_num_beams="1" \
        --learning_rate="1e-4" \
        --warmup_steps="500" \
        --text_column_name="text" \
        --save_total_limit="1" \
        --freeze_feature_encoder \
        --predict_with_generate \
        --do_lower_case \
        --do_eval \
        --do_train
Owner
Sanchit Gandhi
Open-Source Speech @huggingface
Sanchit Gandhi
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