Model reference · open weights
wav2vec2-xls-r-english is an open-weight audio or speech model from jonatasgrosman. AxForge deploys and operates it for you on dedicated EU-owned hardware — with the licence handled where one is required.
Available as managed deployment — configured and operated for you on dedicated EU hardware, quoted per deployment.
What it is
| Released by | jonatasgrosman |
|---|---|
| Type | Audio & music |
| Task | Speech→text |
| Runs with | transformers |
| Released | 2022-03-02 |
| Popularity | 844 downloads / month |
| Licence | Open weights |
About
Fine-tuned facebook/wav2vec2-xls-r-1b on English using the train and validation splits of Common Voice 8.0, Multilingual LibriSpeech, TED-LIUMv3, and Voxpopuli. When using this model, make sure that your speech input is sampled at 16kHz.
This model has been fine-tuned by the HuggingSound tool, and thanks to the GPU credits generously given by the OVHcloud :)
Using the HuggingSound library:
from huggingsound import SpeechRecognitionModel
model = SpeechRecognitionModel("jonatasgrosman/wav2vec2-xls-r-1b-english")
audio_paths = ["/path/to/file.mp3", "/path/to/another_file.wav"]
transcriptions = model.transcribe(audio_paths)
Writing your own inference script:
import torch
import librosa
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
LANG_ID = "en"
MODEL_ID = "jonatasgrosman/wav2vec2-xls-r-1b-english"
SAMPLES = 10
test_dataset = load_dataset("common_voice", LANG_ID, split=f"test[:{SAMPLES}]")
processor = Wav2Vec2Processor.from_pretrained(MODEL_ID)
model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
# Preprocessing the datasets.
# We need to read the audio files as arrays
def speech_file_to_array_fn(batch):
speech_array, sampling_rate = librosa.load(batch["path"], sr=16_000)
batch["speech"] = speech_array
batch["sentence"] = batch["sentence"].upper()
return batch
test_dataset = test_dataset.map(speech_file_to_array_fn)
inputs = processor(test_dataset["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
with torch.no_grad():
logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits
predicted_ids = torch.argmax(logits, dim=-1)
predicted_sentences = processor.batch_decode(predicted_ids)
mozilla-foundation/common_voice_8_0 with split testpython eval.py --model_id jonatasgrosman/wav2vec2-xls-r-1b-english --dataset mozilla-foundation/common_voice_8_0 --config en --split test
speech-recognition-community-v2/dev_datapython eval.py --model_id jonatasgrosman/wav2vec2-xls-r-1b-english --dataset speech-recognition-community-v2/dev_data --config en --split validation --chunk_length_s 5.0 --stride_length_s 1.0
If you want to cite this model you can use this:
@misc{grosman2021xlsr-1b-english,
title={Fine-tuned {XLS-R} 1{B} model for speech recognition in {E}nglish},
author={Grosman, Jonatas},
howpublished={\url{https://huggingface.co/jonatasgrosman/wav2vec2-xls-r-1b-english}},
year={2022}
}
From the published model card. Full card on the HuggingFace links in the sidebar.
Benchmarks
As published on the model card — the maker's own numbers, not measured by AxForge.
| Task | Dataset | Metric | Score |
|---|---|---|---|
| Automatic Speech Recognition | Common Voice 8 | Test WER | 21.050 |
| Automatic Speech Recognition | Common Voice 8 | Test CER | 8.440 |
| Automatic Speech Recognition | Common Voice 8 | Test WER (+LM) | 17.310 |
| Automatic Speech Recognition | Common Voice 8 | Test CER (+LM) | 7.770 |
| Automatic Speech Recognition | Robust Speech Event - Dev Data | Dev WER | 20.530 |
| Automatic Speech Recognition | Robust Speech Event - Dev Data | Dev CER | 9.310 |
| Automatic Speech Recognition | Robust Speech Event - Dev Data | Dev WER (+LM) | 17.700 |
| Automatic Speech Recognition | Robust Speech Event - Dev Data | Dev CER (+LM) | 8.930 |
| Automatic Speech Recognition | Robust Speech Event - Test Data | Test WER | 17.880 |
Using it via the API
Once AxForge deploys wav2vec2-xls-r-english for you, it answers on the OpenAI-compatible API — the same base URL and keys as every other model. (wav2vec2-xls-r-english below is illustrative; you get the exact model name on deployment.)
$ curl -sS https://api.axforge.ai/v1/audio/transcriptions \ -H "Authorization: Bearer $AXFORGE_API_KEY" \ -F model="wav2vec2-xls-r-english" -F file=@audio.mp3
Create an account — your API key is available in the console. 3M free tokens every 30 days with every new account.