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classla/wav2vec2-large-slavic-parlaspeech-hr-lm

sourceHugging Faceupdated 1y agoView on Hugging Face
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wav2vec2-large-slavic-parlaspeech-hr-lm

This model for Croatian ASR is based on the facebook/wav2vec2-large-slavic-voxpopuli-v2 model and was fine-tuned with 300 hours of recordings and transcripts from the ASR Croatian parliament dataset ParlaSpeech-HR v1.0 and enhanced with a 5-gram language model based on the ParlaMint dataset.

<div style="border: 5px solid #ff6700; padding: 10px; margin: 10px 0;"> <strong>Notice:</strong> ParlaSpeech corpora are currently in the process of enrichment with new features. Follow our progress here: <a href="http://clarinsi.github.io/parlaspeech">http://clarinsi.github.io/parlaspeech</a> </div>

If you use this model, please cite the following paper:

Nikola Ljubešić, Danijel Koržinek, Peter Rupnik, Ivo-Pavao Jazbec. ParlaSpeech-HR -- a freely available ASR dataset for Croatian bootstrapped from the ParlaMint corpus. http://www.lrec-conf.org/proceedings/lrec2022/workshops/ParlaCLARINIII/pdf/2022.parlaclariniii-1.16.pdf

Metrics

Evaluation is performed on the dev and test portions of the ParlaSpeech-HR v1.0 dataset.

splitCERWER
dev0.02530.0556
test0.01880.0430

Usage in transformers

Tested with transformers==4.18.0, torch==1.11.0, and SoundFile==0.10.3.post1.

python
from transformers import Wav2Vec2ProcessorWithLM, Wav2Vec2ForCTC
import soundfile as sf
import torch
import os
device = torch.device("cuda:0" if torch.cuda.is_available() else "cpu")
# load model and tokenizer
processor = Wav2Vec2ProcessorWithLM.from_pretrained(
    "classla/wav2vec2-large-slavic-parlaspeech-hr-lm")
model = Wav2Vec2ForCTC.from_pretrained("classla/wav2vec2-large-slavic-parlaspeech-hr-lm")
# download the example wav files:
os.system("wget https://huggingface.co/classla/wav2vec2-large-slavic-parlaspeech-hr-lm/raw/main/00020570a.flac.wav")
# read the wav file 
speech, sample_rate = sf.read("00020570a.flac.wav")
input_values = processor(speech, sampling_rate=sample_rate, return_tensors="pt").input_values.cuda()
inputs = processor(speech, sampling_rate=sample_rate, return_tensors="pt")
with torch.no_grad():
    logits = model(**inputs).logits
transcription = processor.batch_decode(logits.numpy()).text[0]

# remove the raw wav file
os.system("rm 00020570a.flac.wav")

transcription # 'velik broj poslovnih subjekata poslao je sa minusom velik dio'

Training hyperparameters

In fine-tuning, the following arguments were used:

argvalue
per_device_train_batch_size16
gradient_accumulation_steps4
num_train_epochs8
learning_rate3e-4
warmup_steps500