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metadata
language: hr
datasets:
  - parlaspeech-hr
tags:
  - audio
  - automatic-speech-recognition
  - parlaspeech
widget:
  - example_title: example 1
    src: >-
      https://huggingface.co/classla/wav2vec2-xls-r-parlaspeech-hr/raw/main/1800.m4a
  - example_title: example 2
    src: >-
      https://huggingface.co/classla/wav2vec2-xls-r-parlaspeech-hr/raw/main/00020578b.flac.wav
  - example_title: example 3
    src: >-
      https://huggingface.co/classla/wav2vec2-xls-r-parlaspeech-hr/raw/main/00020570a.flac.wav

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.

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. Accepted at ParlaCLARIN@LREC.

There are similarly performing models available, one that does not use a language model and another that is based on the XLS-R model.

Metrics

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

split CER WER
dev 0.0253 0.0556
test 0.0188 0.0430

Usage in transformers

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

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:

arg value
per_device_train_batch_size 16
gradient_accumulation_steps 4
num_train_epochs 8
learning_rate 3e-4
warmup_steps 500