Model reference · open weights
wav2vec2-large-ru-golos is an open-weight audio or speech model from bond005. 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 | bond005 |
|---|---|
| Type | Audio & music |
| Task | Speech→text |
| Runs with | transformers |
| Released | 2022-06-21 |
| Popularity | 62k downloads / month |
| Licence | Open weights |
About
This model is a component of the Pisets speech-to-text system, presented in the paper Pisets: A Robust Speech Recognition System for Lectures and Interviews.
The source code for the Pisets system is available on GitHub: bond005/pisets.
The Wav2Vec2 model is based on facebook/wav2vec2-large-xlsr-53, fine-tuned in Russian using Sberdevices Golos with audio augmentations like as pitch shift, acceleration/deceleration of sound, reverberation etc.
When using this model, make sure that your speech input is sampled at 16kHz.
To transcribe audio files the model can be used as a standalone acoustic model as follows:
from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
from datasets import load_dataset
import torch
# load model and tokenizer
processor = Wav2Vec2Processor.from_pretrained("bond005/wav2vec2-large-ru-golos")
model = Wav2Vec2ForCTC.from_pretrained("bond005/wav2vec2-large-ru-golos")
# load the test part of Golos dataset and read first soundfile
ds = load_dataset("bond005/sberdevices_golos_10h_crowd", split="test")
# tokenize
processed = processor(ds[0]["audio"]["array"], return_tensors="pt", padding="longest") # Batch size 1
# retrieve logits
logits = model(processed.input_values, attention_mask=processed.attention_mask).logits
# take argmax and decode
predicted_ids = torch.argmax(logits, dim=-1)
transcription = processor.batch_decode(predicted_ids)[0]
print(transcription)
This code snippet shows how to evaluate bond005/wav2vec2-large-ru-golos on Golos dataset's "crowd" and "farfield" test data.
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import torch
from jiwer import wer, cer # we need word error rate (WER) and character error rate (CER)
# load the test part of Golos Crowd and remove samples with empty "true" transcriptions
golos_crowd_test = load_dataset("bond005/sberdevices_golos_10h_crowd", split="test")
golos_crowd_test = golos_crowd_test.filter(
lambda it1: (it1["transcription"] is not None) and (len(it1["transcription"].strip()) > 0)
)
# load the test part of Golos Farfield and remove sampels with empty "true" transcriptions
golos_farfield_test = load_dataset("bond005/sberdevices_golos_100h_farfield", split="test")
golos_farfield_test = golos_farfield_test.filter(
lambda it2: (it2["transcription"] is not None) and (len(it2["transcription"].strip()) > 0)
)
# load model and tokenizer
model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-base-960h").to("cuda")
processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-base-960h")
# recognize one sound
def map_to_pred(batch):
# tokenize and vectorize
processed = processor(
batch["audio"]["array"], sampling_rate=batch["audio"]["sampling_rate"],
return_tensors="pt", padding="longest"
)
input_values = processed.input_values.to("cuda")
attention_mask = processed.attention_mask.to("cuda")
# recognize
with torch.no_grad():
logits = model(input_values, attention_mask=attention_mask).logits
predicted_ids = torch.argmax(logits, dim=-1)
# decode
transcription = processor.batch_decode(predicted_ids)
batch["text"] = transcription[0]
return batch
# calculate WER and CER on the crowd domain
crowd_result = golos_crowd_test.map(map_to_pred, remove_columns=["audio"])
crowd_wer = wer(crowd_result["transcription"], crowd_result["text"])
crowd_cer = cer(crowd_result["transcription"], crowd_result["text"])
print("Word error rate on the Crowd domain:", crowd_wer)
print("Character error rate on the Crowd domain:", crowd_cer)
# calculate WER and CER on the farfield domain
farfield_result = golos_farfield_test.map(map_to_pred, remove_columns=["audio"])
farfield_wer = wer(farfield_result["transcription"], farfield_result["text"])
farfield_cer = cer(farfield_result["transcription"], farfield_result["text"])
print("Word error rate on the Farfield domain:", farfield_wer)
print("Character error rate on the Farfield domain:", farfield_cer)
Result (WER, %):
| "crowd" | "farfield" |
|---|---|
| 10.144 | 20.353 |
Result (CER, %):
| "crowd" | "farfield" |
|---|---|
| 2.168 | 6.030 |
If you want to cite this model you can use this:
@misc{bondarenko2022wav2vec2-large-ru-golos,
title={XLSR Wav2Vec2 Russian by Ivan Bondarenko},
author={Bondarenko, Ivan},
publisher={Hugging Face},
journal={Hugging Face Hub},
howpublished={\url{https://huggingface.co/bond005/wav2vec2-large-ru-golos}},
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 |
|---|---|---|---|
| Speech Recognition | Sberdevices Golos (crowd) | Test WER | 10.144 |
| Speech Recognition | Sberdevices Golos (crowd) | Test CER | 2.168 |
| Speech Recognition | Sberdevices Golos (crowd) | Test WER | 20.353 |
| Speech Recognition | Sberdevices Golos (crowd) | Test CER | 6.030 |
| Automatic Speech Recognition | Common Voice ru | Test WER | 18.548 |
| Automatic Speech Recognition | Common Voice ru | Test CER | 4 |
| Automatic Speech Recognition | Sova RuDevices | Test WER | 25.410 |
| Automatic Speech Recognition | Sova RuDevices | Test CER | 7.965 |
| Automatic Speech Recognition | Russian Librispeech | Test WER | 21.872 |
| Automatic Speech Recognition | Russian Librispeech | Test CER | 4.469 |
| Automatic Speech Recognition | Voxforge Ru | Test WER | 27.084 |
| Automatic Speech Recognition | Voxforge Ru | Test CER | 6.986 |
Using it via the API
Once AxForge deploys wav2vec2-large-ru-golos for you, it answers on the OpenAI-compatible API — the same base URL and keys as every other model. (wav2vec2-large-ru-golos 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-large-ru-golos" -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.