Model reference · open weights

wav2vec2-xls-r-russian

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

wav2vec2-xls-r-russian 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
Popularity1k downloads / month
LicenceOpen weights

About

What wav2vec2-xls-r-russian is

Fine-tuned facebook/wav2vec2-xls-r-1b on Russian using the train and validation splits of Common Voice 8.0, Golos, and Multilingual TEDx. 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-russian")
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 = "ru"
MODEL_ID = "jonatasgrosman/wav2vec2-xls-r-1b-russian"
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-russian --dataset mozilla-foundation/common_voice_8_0 --config ru --split test
  1. To evaluate on speech-recognition-community-v2/dev_data
python eval.py --model_id jonatasgrosman/wav2vec2-xls-r-1b-russian --dataset speech-recognition-community-v2/dev_data --config ru --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-russian,
  title={Fine-tuned {XLS-R} 1{B} model for speech recognition in {R}ussian},
  author={Grosman, Jonatas},
  howpublished={\url{https://huggingface.co/jonatasgrosman/wav2vec2-xls-r-1b-russian}},
  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 WER9.820
Automatic Speech RecognitionCommon Voice 8Test CER2.300
Automatic Speech RecognitionCommon Voice 8Test WER (+LM)7.080
Automatic Speech RecognitionCommon Voice 8Test CER (+LM)1.870
Automatic Speech RecognitionRobust Speech Event - Dev DataDev WER23.960
Automatic Speech RecognitionRobust Speech Event - Dev DataDev CER8.880
Automatic Speech RecognitionRobust Speech Event - Dev DataDev WER (+LM)15.880
Automatic Speech RecognitionRobust Speech Event - Dev DataDev CER (+LM)7.420
Automatic Speech RecognitionRobust Speech Event - Test DataTest WER14.230

Using it via the API

Call it like any OpenAI endpoint

Once AxForge deploys wav2vec2-xls-r-russian for you, it answers on the OpenAI-compatible API — the same base URL and keys as every other model. (wav2vec2-xls-r-russian 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-russian" -F file=@audio.mp3

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