metadata
language:
- en
license: apache-2.0
tags:
- automatic-speech-recognition
- pytorch
- transformers
- en
- generated_from_trainer
model-index:
- name: wav2vec2-xls-r-300m-phoneme
results:
- task:
name: Speech Recognition
type: automatic-speech-recognition
dataset:
name: DARPA TIMIT
type: timit
args: en
metrics:
- name: Test CER
type: cer
value: 7.996
Model
This model is a fine-tuned version of facebook/wav2vec2-xls-r-300m on the Timit dataset. Check this notebook for training detail.
Usage
Approach 1: Using HuggingFace's pipeline, this will cover everything end-to-end from raw audio input to text output.
from transformers import pipeline
# Load the model
pipe = pipeline(model="vitouphy/wav2vec2-xls-r-300m-timit-phoneme")
# Process raw audio
output = pipe("audio_file.wav", chunk_length_s=10, stride_length_s=(4, 2))
Approach 2: More custom way to predict phonemes.
from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
from datasets import load_dataset
import torch
import soundfile as sf
# load model and processor
processor = Wav2Vec2Processor.from_pretrained("vitouphy/wav2vec2-xls-r-300m-timit-phoneme")
model = Wav2Vec2ForCTC.from_pretrained("vitouphy/wav2vec2-xls-r-300m-timit-phoneme")
# Read and process the input
audio_input, sample_rate = sf.read("audio_file.wav")
inputs = processor(audio_input, sampling_rate=16_000, return_tensors="pt", padding=True)
with torch.no_grad():
logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits
# Decode id into string
predicted_ids = torch.argmax(logits, axis=-1)
predicted_sentences = processor.batch_decode(predicted_ids)
print(predicted_sentences)
Training and evaluation data
We use DARPA TIMIT dataset for this model.
- We split into 80/10/10 for training, validation, and testing respectively.
- That roughly corresponds to about 137/17/17 minutes.
- The model obtained 7.996% on this test set.
Training procedure
Training hyperparameters
The following hyperparameters were used during training:
- learning_rate: 3e-05
- train_batch_size: 8
- eval_batch_size: 8
- seed: 42
- gradient_accumulation_steps: 4
- total_train_batch_size: 32
- optimizer: Adam with betas=(0.9,0.999) and epsilon=1e-08
- lr_scheduler_type: linear
- lr_scheduler_warmup_steps: 2000
- training_steps: 10000
- mixed_precision_training: Native AMP
Framework versions
- Transformers 4.17.0.dev0
- Pytorch 1.10.2+cu102
- Datasets 1.18.2.dev0
- Tokenizers 0.11.0
Citation
@software{phy22-phoneme,
author = {Phy, Vitou},
title = {{Automatic Phoneme Recognition on TIMIT Dataset with Wav2Vec 2.0}},
year = 2022,
note = {{If you use this model, please cite it using these metadata.}},
publisher = {Hugging Face},
version = {1.0},
doi = {10.57967/hf/0125},
url = {https://huggingface.co/vitouphy/wav2vec2-xls-r-300m-timit-phoneme}
}