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---
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: 6.0
---
# Sharif-wav2vec2
This is a fine-tuned version of Sharif Wav2vec2 for Farsi. The base model went through a fine-tuning process in which 108 hours of Commonvoice's Farsi samples with a sampling rate equal to 16kHz. Afterward, we trained a 5gram using [kenlm](https://github.com/kpu/kenlm) toolkit and used it in the processor which increased our accuracy on online ASR.
## Usage
When using the model, ensure that your speech input is sampled at 16Khz. Prior to the usage, you may need to install the below dependencies:
```shell
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:
```python
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, you can use the code below. Ensure your dataset to be in following form in order to avoid any further conflict:
| path | reference|
|:----:|:--------:|
| path/to/audio_file.wav | "TRANSCRIPTION" |
also, make sure you have installed `pip install jiwer` prior to running.
```python
import tensorflow
import torchaudio
import torch
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
)
with torch.no_grad():
logits = model(
features.input_values,
attention_mask=features.attention_mask).logits
batch["prediction"] = processor.batch_decode(logits.numpy()).text
return batch
dataset = load_dataset(
"csv",
data_files={"test":"dataset.eval.csv"},
delimiter=",")["test"]
dataset = dataset.map(speech_file_to_array_fn)
result = dataset.map(predict, batched=True, batch_size=4)
wer = load_metric("wer")
print("WER: {:.2f}".format(wer.compute(
predictions=result["prediction"],
references=result["reference"])))
```
*Result (WER) on common-voice 6.1*:
| cleaned | other |
|:---:|:---:|
| 0.06 | 0.16 |
## Citation
If you want to cite this model you can use this:
```bibtex
?
``` |