README.md

metadata

language: mr
datasets:
  - openslr
  - interspeech_2021_asr
metrics:
  - wer
tags:
  - audio
  - automatic-speech-recognition
  - speech
  - xlsr-fine-tuning-week
  - hindi
  - marathi
license: apache-2.0
model-index:
  - name: XLSR Wav2Vec2 Large 53 Hindi-Marathi by Tanmay Laud
    results:
      - task:
          name: Speech Recognition
          type: automatic-speech-recognition
        dataset:
          name: OpenSLR hi, OpenSLR mr
          type: openslr, interspeech_2021_asr
        metrics:
          - name: Test WER
            type: wer
            value: 23.736641

Wav2Vec2-Large-XLSR-53-Hindi-Marathi

Fine-tuned facebook/wav2vec2-large-xlsr-53 on Hindi and Marathi using the OpenSLR SLR64 datasets. When using this model, make sure that your speech input is sampled at 16kHz.

Installation

pip install git+https://github.com/huggingface/transformers.git datasets librosa torch==1.7.0 torchaudio==0.7.0 jiwer

Eval dataset:

wget https://www.openslr.org/resources/103/Marathi_test.zip  -P data/marathi
unzip -P "K3[2?do9" data/marathi/Marathi_test.zip -d data/marathi/.
tar -xzf data/marathi/Marathi_test.tar.gz -C data/marathi/.
wget https://www.openslr.org/resources/103/Hindi_test.zip  -P data/hindi
unzip -P "w9I2{3B*" data/hindi/Hindi_test.zip -d data/hindi/.
tar -xzf data/hindi/Hindi_test.tar.gz -C data/hindi/.
wget  -O test.csv 'https://filebin.net/snrz6bt13usv8w2e/test_large.csv?t=ps3n99ho'
#If download does not work, paste this link in browser: https://filebin.net/snrz6bt13usv8w2e/test_large.csv

Usage

The model can be used directly (without a language model) as follows, assuming you have a dataset with Marathi text and path fields:

import torch
import torchaudio
import librosa
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor

from datasets import load_metric, Dataset
from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC

wer = load_metric("wer")
processor = Wav2Vec2Processor.from_pretrained('tanmaylaud/wav2vec2-large-xlsr-hindi-marathi')
model = Wav2Vec2ForCTC.from_pretrained('tanmaylaud/wav2vec2-large-xlsr-hindi-marathi').to("cuda")

# Preprocessing the datasets.
# We need to read the audio files as arrays
def speech_file_to_array_fn(batch):
    batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"])
    speech_array, sampling_rate = torchaudio.load(batch["path"])
    batch["speech"] = speech_array[0].numpy()
    batch["sampling_rate"] = sampling_rate
    batch["target_text"] = batch["sentence"]
    batch["speech"] = librosa.resample(np.asarray(batch["speech"]), sampling_rate, 16_000)
    batch["sampling_rate"] = 16_000
    return batch

test_data= test_data.map(speech_file_to_array_fn)
inputs = processor(test_data["speech"][:2], sampling_rate=16_000, return_tensors="pt", padding=True)

with torch.no_grad():
    logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits

predicted_ids = torch.argmax(logits, dim=-1)

print("Prediction:", processor.batch_decode(predicted_ids))
print("Reference:", test_data["text"][:2])

Code For Evaluation on OpenSLR (Hindi + Marathi : https://filebin.net/snrz6bt13usv8w2e/test_large.csv)

import torchaudio
import torch
import librosa
import numpy as np
import re

test = Dataset.from_csv('test.csv')


chars_to_ignore_regex = '[\\\\\\\\\\\\\\\\,\\\\\\\\\\\\\\\\?\\\\\\\\\\\\\\\\.\\\\\\\\\\\\\\\\!\\\\\\\\\\\\\\\\-\\\\\\\\\\\\\\\\;\\\\\\\\\\\\\\\\:\\\\\\\\\\\\\\\\"\\\\\\\\\\\\\\\\“\\\\\\\\\\\\\\\\%\\\\\\\\\\\\\\\\‘\\\\\\\\\\\\\\\\”\\\\\\\\\\\\\\\\�\\\\\\\\\\\\\\\\।]'

# Preprocessing the datasets.
# We need to read the audio files as arrays
def speech_file_to_array_fn(batch):
    batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"])
    speech_array, sampling_rate = torchaudio.load(batch["path"])
    batch["speech"] = speech_array[0].numpy()
    batch["sampling_rate"] = sampling_rate
    batch["target_text"] = batch["sentence"]
    batch["speech"] = librosa.resample(np.asarray(batch["speech"]), sampling_rate, 16_000)
    batch["sampling_rate"] = 16_000
    return batch

test= test.map(speech_file_to_array_fn)

# Preprocessing the datasets.
# We need to read the audio files as arrays
def evaluate(batch):
    inputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
    with torch.no_grad():
        logits = model(inputs.input_values.to("cuda"), attention_mask=inputs.attention_mask.to("cuda")).logits
        pred_ids = torch.argmax(logits, dim=-1)
        # we do not want to group tokens when computing the metrics
        batch["pred_strings"] = processor.batch_decode(pred_ids)
        return batch

test = test.map(evaluate, batched=True, batch_size=32)
print("WER: {:2f}".format(100 * wer.compute(predictions=test["pred_strings"], references=test["sentence"])))

Code for Evaluation on Common Voice Hindi (Common voice does not have Marathi yet)

import torchaudio
import torch
import librosa
import numpy as np
import re
from datasets import load_dataset

chars_to_ignore_regex = '[\\\\\\\\\\\\\\\\,\\\\\\\\\\\\\\\\?\\\\\\\\\\\\\\\\.\\\\\\\\\\\\\\\\!\\\\\\\\\\\\\\\\-\\\\\\\\\\\\\\\\;\\\\\\\\\\\\\\\\:\\\\\\\\\\\\\\\\"\\\\\\\\\\\\\\\\“\\\\\\\\\\\\\\\\%\\\\\\\\\\\\\\\\‘\\\\\\\\\\\\\\\\”\\\\\\\\\\\\\\\\�\\\\\\\\\\\\\\\\।]'

# Preprocessing the datasets.
# We need to read the audio files as arrays
def speech_file_to_array_fn(batch):
    batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"])
    speech_array, sampling_rate = torchaudio.load(batch["path"])
    batch["speech"] = speech_array[0].numpy()
    batch["sampling_rate"] = sampling_rate
    batch["target_text"] = batch["sentence"]
    batch["speech"] = librosa.resample(np.asarray(batch["speech"]), sampling_rate, 16_000)
    batch["sampling_rate"] = 16_000
    return batch
  
#Run prediction on batch  
def evaluate(batch):
    inputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
    with torch.no_grad():
        logits = model(inputs.input_values.to("cuda"), attention_mask=inputs.attention_mask.to("cuda")).logits
        pred_ids = torch.argmax(logits, dim=-1)
        # we do not want to group tokens when computing the metrics
        batch["pred_strings"] = processor.batch_decode(pred_ids)
        return batch


test_data = load_dataset("common_voice", "hi", split="test")
test_data = test_data.map(speech_file_to_array_fn)
test_data = test_data.map(evaluate, batched=True, batch_size=32)
print("WER: {:2f}".format(100 * wer.compute(predictions=test_data["pred_strings"],
                                            references=test_data["sentence"])))

Link to eval notebook : https://colab.research.google.com/drive/1nZRTgKfxCD9cvy90wikTHkg2il3zgcqW#scrollTo=cXWFbhb0d7DT

WER : 23.736641% (OpenSLR Hindi+Marathi Test set : https://filebin.net/snrz6bt13usv8w2e/test_large.csv)

WER: 44.083527% (Common Voice Hindi Test Split)