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---
language:
- fr
license: apache-2.0
tags:
- automatic-speech-recognition
- mozilla-foundation/common_voice_9_0
- generated_from_trainer
- hf-asr-leaderboard
- robust-speech-event
datasets:
- common_voice
- mozilla-foundation/common_voice_9_0
model-index:
- name: Fine-tuned Wav2Vec2 XLS-R 1B model for ASR in French
results:
- task:
name: Automatic Speech Recognition
type: automatic-speech-recognition
dataset:
name: Common Voice 9
type: mozilla-foundation/common_voice_9_0
args: fr
metrics:
- name: Test WER
type: wer
value: 12.72
- name: Test WER (+LM)
type: wer
value: 10.60
- task:
name: Automatic Speech Recognition
type: automatic-speech-recognition
dataset:
name: Robust Speech Event - Dev Data
type: speech-recognition-community-v2/dev_data
args: fr
metrics:
- name: Test WER
type: wer
value: 24.28
- name: Test WER (+LM)
type: wer
value: 20.85
---
# Fine-tuned Wav2Vec2 XLS-R 1B model for ASR in French
This model is a fine-tuned version of [facebook/wav2vec2-xls-r-1b](https://huggingface.co/facebook/wav2vec2-xls-r-1b) on the MOZILLA-FOUNDATION/COMMON_VOICE_9_0 - FR dataset.
## Usage
1. To use on a local audio file without the language model
```python
import torch
import torchaudio
from transformers import AutoModelForCTC, Wav2Vec2Processor
processor = Wav2Vec2Processor.from_pretrained("bhuang/wav2vec2-xls-r-1b-cv9-fr")
model = AutoModelForCTC.from_pretrained("bhuang/wav2vec2-xls-r-1b-cv9-fr").cuda()
# path to your audio file
wav_path = "example.wav"
waveform, sample_rate = torchaudio.load(wav_path)
waveform = waveform.squeeze(axis=0) # mono
# resample
if sample_rate != 16_000:
resampler = torchaudio.transforms.Resample(sample_rate, 16_000)
waveform = resampler(waveform)
# normalize
input_dict = processor(waveform, sampling_rate=16_000, return_tensors="pt")
with torch.inference_mode():
logits = model(input_dict.input_values.to("cuda")).logits
# decode
predicted_ids = torch.argmax(logits, dim=-1)
predicted_sentence = processor.batch_decode(predicted_ids)[0]
```
2. To use on a local audio file with the language model
```python
import torch
import torchaudio
from transformers import AutoModelForCTC, Wav2Vec2ProcessorWithLM
processor_with_lm = Wav2Vec2ProcessorWithLM.from_pretrained("bhuang/wav2vec2-xls-r-1b-cv9-fr")
model = AutoModelForCTC.from_pretrained("bhuang/wav2vec2-xls-r-1b-cv9-fr").cuda()
model_sampling_rate = processor_with_lm.feature_extractor.sampling_rate
# path to your audio file
wav_path = "example.wav"
waveform, sample_rate = torchaudio.load(wav_path)
waveform = waveform.squeeze(axis=0) # mono
# resample
if sample_rate != 16_000:
resampler = torchaudio.transforms.Resample(sample_rate, 16_000)
waveform = resampler(waveform)
# normalize
input_dict = processor_with_lm(waveform, sampling_rate=16_000, return_tensors="pt")
with torch.inference_mode():
logits = model(input_dict.input_values.to("cuda")).logits
predicted_sentence = processor_with_lm.batch_decode(logits.cpu().numpy()).text[0]
```
## Evaluation
1. To evaluate on `mozilla-foundation/common_voice_9_0`
```bash
python eval.py \
--model_id "bhuang/wav2vec2-xls-r-1b-cv9-fr" \
--dataset "mozilla-foundation/common_voice_9_0" \
--config "fr" \
--split "test" \
--log_outputs \
--outdir "outputs/results_mozilla-foundatio_common_voice_9_0_with_lm"
```
2. To evaluate on `speech-recognition-community-v2/dev_data`
```bash
python eval.py \
--model_id "bhuang/wav2vec2-xls-r-1b-cv9-fr" \
--dataset "speech-recognition-community-v2/dev_data" \
--config "fr" \
--split "validation" \
--chunk_length_s 5.0 \
--stride_length_s 1.0 \
--log_outputs \
--outdir "outputs/results_speech-recognition-community-v2_dev_data_with_lm"
```
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