File size: 4,340 Bytes
8b4523b aab5371 876dff9 aab5371 1e2b39a aab5371 876dff9 aab5371 8b4523b aab5371 9603413 64771ba 55c34da 64771ba 52867a6 b466f87 64771ba f932e73 b96632b 64771ba b466f87 64771ba b466f87 64771ba aab5371 feb891f b2d9405 a585e5e b2d9405 4a36ad9 b2d9405 4a36ad9 f9ab6fe 4a36ad9 b2d9405 4a36ad9 feb891f 64771ba aab5371 9603413 aab5371 4a36ad9 828771f |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 |
---
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 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](https://github.com/kpu/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:
```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 use the code below:
```
*Input csv files format:*
| path| reference|
|---|---|
| path to audio files | corresponding transcription|
```
```python
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:
```bibtex
?
``` |