Model reference · open weights

wav2vec2-xls-r-english

Available as managed deployment Audio jonatasgrosman · community Speech→text 1 variants 844 dl/mo

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 byjonatasgrosman
TypeAudio & music
TaskSpeech→text
Runs withtransformers
Released2022-03-02
Popularity844 downloads / month
LicenceOpen weights

About

What wav2vec2-xls-r-english is

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 :)

Read the full model card

Usage

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)

Evaluation Commands

  1. To evaluate on mozilla-foundation/common_voice_8_0 with split test
python eval.py --model_id jonatasgrosman/wav2vec2-xls-r-1b-english --dataset mozilla-foundation/common_voice_8_0 --config en --split test
  1. To evaluate on speech-recognition-community-v2/dev_data
python 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

Citation

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

Reported results

As published on the model card — the maker's own numbers, not measured by AxForge.

TaskDatasetMetricScore
Automatic Speech RecognitionCommon Voice 8Test WER21.050
Automatic Speech RecognitionCommon Voice 8Test CER8.440
Automatic Speech RecognitionCommon Voice 8Test WER (+LM)17.310
Automatic Speech RecognitionCommon Voice 8Test CER (+LM)7.770
Automatic Speech RecognitionRobust Speech Event - Dev DataDev WER20.530
Automatic Speech RecognitionRobust Speech Event - Dev DataDev CER9.310
Automatic Speech RecognitionRobust Speech Event - Dev DataDev WER (+LM)17.700
Automatic Speech RecognitionRobust Speech Event - Dev DataDev CER (+LM)8.930
Automatic Speech RecognitionRobust Speech Event - Test DataTest WER17.880

Using it via the API

Call it like any OpenAI endpoint

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

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