language: fa
datasets:
- common_voice_6_1
tags:
- audio
- automatic-speech-recognition
license: mit
widget:
- example_title: Common Voice Sample 1
src: >-
https://datasets-server.huggingface.co/assets/common_voice/--/fa/train/0/audio/audio.mp3
- example_title: Common Voice Sample 2
src: >-
https://datasets-server.huggingface.co/assets/common_voice/--/fa/train/1/audio/audio.mp3
model-index:
- name: Sharif-wav2vec2
results:
- task:
name: Automatic Speech Recognition
type: automatic-speech-recognition
dataset:
name: Common Voice Corpus 6.1 (clean)
type: common_voice_6_1
config: clean
split: test
args:
language: fa
metrics:
- name: Test WER
type: wer
value: 6
Sharif-wav2vec2
This is the fine-tuned version of Sharif Wav2vec2 for Farsi. The base model was fine-tuned on 108 hours of Commonvoice's Farsi samples with a sampling rate equal to 16kHz. Afterward, we trained a 5gram using kenlm toolkit and used it in the processor which increased our accuracy on online ASR.
Usage
When using the model make sure that your speech input is sampled at 16Khz. Prior to the usage, you may need to install the below dependencies:
pip install pyctcdecode
pip install pypi-kenlm
For testing you can use the hosted inference API at the hugging face (There are provided examples from common voice) it may take a while to transcribe the given voice. Or you can use the bellow code for a local run:
import tensorflow
import torchaudio
import torch
import numpy as np
from transformers import AutoProcessor, AutoModelForCTC
processor = AutoProcessor.from_pretrained("SLPL/Sharif-wav2vec2")
model = AutoModelForCTC.from_pretrained("SLPL/Sharif-wav2vec2")
speech_array, sampling_rate = torchaudio.load("path/to/your.wav")
speech_array = speech_array.squeeze().numpy()
features = processor(
speech_array,
sampling_rate=processor.feature_extractor.sampling_rate,
return_tensors="pt",
padding=True)
with torch.no_grad():
logits = model(
features.input_values,
attention_mask=features.attention_mask).logits
prediction = processor.batch_decode(logits.numpy()).text
print(prediction[0])
# تست
Evaluation
For the evaluation use the code below:
*Input csv files format:*
| path| reference|
|---|---|
| path to audio files | corresponding transcription|
import torch
import torchaudio
import librosa
from datasets import load_dataset,load_metric
import numpy as np
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
from transformers import Wav2Vec2ProcessorWithLM
model = Wav2Vec2ForCTC.from_pretrained("SLPL/Sharif-wav2vec2")
processor = Wav2Vec2ProcessorWithLM.from_pretrained("SLPL/Sharif-wav2vec2")
def speech_file_to_array_fn(batch):
speech_array, sampling_rate = torchaudio.load(batch["path"])
speech_array = speech_array.squeeze().numpy()
speech_array = librosa.resample(np.asarray(speech_array), sampling_rate, processor.feature_extractor.sampling_rate)
batch["speech"] = speech_array
return batch
def predict(batch):
features = processor(
batch["speech"],
sampling_rate=processor.feature_extractor.sampling_rate,
return_tensors="pt",
padding=True
)
input_values = features.input_values
attention_mask = features.attention_mask
with torch.no_grad():
logits = model(input_values, attention_mask=attention_mask).logits #when we are trying to load model with LM we have to use logits instead of argmax(logits)
batch["prediction"] = processor.batch_decode(logits.numpy()).text
return batch
dataset = load_dataset("csv", data_files={"test":"path/to/your.csv"}, delimiter=",")["test"]
dataset = dataset.map(speech_file_to_array_fn)
result = dataset.map(predict, batched=True, batch_size=4)
wer = load_metric("wer")
cer = load_metric("cer")
print("WER: {:.2f}".format(100 * wer.compute(predictions=result["prediction"], references=result["reference"])))
print("CER: {:.2f}".format(100 * cer.compute(predictions=result["prediction"], references=result["reference"])))
Result (WER):
clean | other |
---|---|
6.0 | 16.4 |
Citation
If you want to cite this model you can use this:
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