metadata

license: apache-2.0
language:
  - en
metrics:
  - wer
base_model:
  - facebook/wav2vec2-base-960h
tags:
  - pytorch
  - Transformers
  - speech
  - audio

Model Description

This model is a fine-tuned version of facebook/wav2vec2-base-960h for automatic speech recognition (ASR). It has been trained using the LibriSpeech dataset and is designed to improve transcription accuracy over the base model.

The fine-tuning process involved:

Selecting a subset of speakers from the dev-clean and test-clean datasets.
Preprocessing audio files and their corresponding transcriptions.
Training with gradient accumulation, mixed precision (if available), and periodic evaluation.
Saving the fine-tuned model for inference.

GitHub Authors: Lucas Tramonte, Kiyoshi Araki

Usage

To transcribe audio files, the model can be used as follows:

from transformers import AutoProcessor, AutoModelForCTC
import torch
import librosa

# Load model and processor
processor = AutoProcessor.from_pretrained("deepl-project/conformer-finetunning")
model = AutoModelForCTC.from_pretrained("deepl-project/conformer-finetunning")

# Load and preprocess an audio file
file_path = "path/to/audio/file.wav"
speech, sr = librosa.load(file_path, sr=16000)
inputs = processor(speech, sampling_rate=sr, return_tensors="pt", padding=True)

# Perform inference
with torch.no_grad():
    logits = model(**inputs).logits

# Decode transcription
predicted_ids = torch.argmax(logits, dim=-1)
transcription = processor.batch_decode(predicted_ids)

print("Transcription:", transcription[0])

deepl-project
/

conformer-finetunning

Model Description

Usage

References