wav2vec2-large-xlsr-53-polish

jonatasgrosman/wav2vec2-large-xlsr-53-polish

作者
jonatasgrosman
任务
语音识别
下载量
2,925,696
点赞数
26
趋势分
0
许可证
apache-2.0
收录时间
2026/11/4

language: pl license: apache-2.0 datasets:

  • common_voice
  • mozilla-foundation/common_voice_6_0 metrics:
  • wer
  • cer tags:
  • audio
  • automatic-speech-recognition
  • hf-asr-leaderboard
  • mozilla-foundation/common_voice_6_0
  • pl
  • robust-speech-event
  • speech
  • xlsr-fine-tuning-week model-index:
  • name: XLSR Wav2Vec2 Polish by Jonatas Grosman results:
    • task: name: Automatic Speech Recognition type: automatic-speech-recognition dataset: name: Common Voice pl type: common_voice args: pl metrics:
      • name: Test WER type: wer value: 14.21
      • name: Test CER type: cer value: 3.49
      • name: Test WER (+LM) type: wer value: 10.98
      • name: Test CER (+LM) type: cer value: 2.93
    • task: name: Automatic Speech Recognition type: automatic-speech-recognition dataset: name: Robust Speech Event - Dev Data type: speech-recognition-community-v2/dev_data args: pl metrics:
      • name: Dev WER type: wer value: 33.18
      • name: Dev CER type: cer value: 15.92
      • name: Dev WER (+LM) type: wer value: 29.31
      • name: Dev CER (+LM) type: cer value: 15.17

Fine-tuned XLSR-53 large model for speech recognition in Polish

Fine-tuned facebook/wav2vec2-large-xlsr-53 on Polish using the train and validation splits of Common Voice 6.1. When using this model, make sure that your speech input is sampled at 16kHz.

This model has been fine-tuned thanks to the GPU credits generously given by the OVHcloud :)

The script used for training can be found here: https://github.com/jonatasgrosman/wav2vec2-sprint

Usage

The model can be used directly (without a language model) as follows...

Using the HuggingSound library:

from huggingsound import SpeechRecognitionModel

model = SpeechRecognitionModel("jonatasgrosman/wav2vec2-large-xlsr-53-polish")
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 = "pl"
MODEL_ID = "jonatasgrosman/wav2vec2-large-xlsr-53-polish"
SAMPLES = 5

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)

for i, predicted_sentence in enumerate(predicted_sentences):
    print("-" * 100)
    print("Reference:", test_dataset[i]["sentence"])
    print("Prediction:", predicted_sentence)
ReferencePrediction
"""CZY DRZWI BYŁY ZAMKNIĘTE?"""PRZY DRZWI BYŁY ZAMKNIĘTE
GDZIEŻ TU POWÓD DO WYRZUTÓW?WGDZIEŻ TO POM DO WYRYDÓ
"""O TEM JEDNAK NIE BYŁO MOWY."""O TEM JEDNAK NIE BYŁO MOWY
LUBIĘ GO.LUBIĄ GO
— TO MI NIE POMAGA.TO MNIE NIE POMAGA
WCIĄŻ LUDZIE WYSIADAJĄ PRZED ZAMKIEM, Z MIASTA, Z PRAGI.WCIĄŻ LUDZIE WYSIADAJĄ PRZED ZAMKIEM Z MIASTA Z PRAGI
ALE ON WCALE INACZEJ NIE MYŚLAŁ.ONY MONITCENIE PONACZUŁA NA MASU
A WY, CO TAK STOICIE?A WY CO TAK STOICIE
A TEN PRZYRZĄD DO CZEGO SŁUŻY?A TEN PRZYRZĄD DO CZEGO SŁUŻY
NA JUTRZEJSZYM KOLOKWIUM BĘDZIE PIĘĆ PYTAŃ OTWARTYCH I TEST WIELOKROTNEGO WYBORU.NAJUTRZEJSZYM KOLOKWIUM BĘDZIE PIĘĆ PYTAŃ OTWARTYCH I TEST WIELOKROTNEGO WYBORU

Evaluation

  1. To evaluate on mozilla-foundation/common_voice_6_0 with split test
python eval.py --model_id jonatasgrosman/wav2vec2-large-xlsr-53-polish --dataset mozilla-foundation/common_voice_6_0 --config pl --split test
  1. To evaluate on speech-recognition-community-v2/dev_data
python eval.py --model_id jonatasgrosman/wav2vec2-large-xlsr-53-polish --dataset speech-recognition-community-v2/dev_data --config pl --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{grosman2021xlsr53-large-polish,
  title={Fine-tuned {XLSR}-53 large model for speech recognition in {P}olish},
  author={Grosman, Jonatas},
  howpublished={\url{https://huggingface.co/jonatasgrosman/wav2vec2-large-xlsr-53-polish}},
  year={2021}
}

内容来源:Hugging Face 模型卡

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