xsway

xsway/wav2vec2-large-xlsr-georgian

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Model Documentation

Wav2Vec2-Large-XLSR-53-Georgian



Fine-tuned facebook/wav2vec2-large-xlsr-53 on Georgian using the Common Voice. When using this model, make sure that your speech input is sampled at 16kHz.

Usage



The model can be used directly (without a language model) as follows:

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

test_dataset = load_dataset("common_voice", "ka", split="test[:2%]")

processor = Wav2Vec2Processor.from_pretrained("xsway/wav2vec2-large-xlsr-georgian") model = Wav2Vec2ForCTC.from_pretrained("xsway/wav2vec2-large-xlsr-georgian")

resampler = lambda sampling_rate, y: librosa.resample(y.numpy().squeeze(), sampling_rate, 16_000)

Preprocessing the datasets.

We need to read the audio files as arrays

def speech_file_to_array_fn(batch): \\\\tspeech_array, sampling_rate = torchaudio.load(batch["path"]) \\\\tbatch["speech"] = resampler(sampling_rate, speech_array).squeeze() \\\\treturn batch

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

with torch.no_grad(): \\\\tlogits = 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_dataset["sentence"][:2])


Evaluation



The model can be evaluated as follows on the Georgian test data of Common Voice.

python
import torch
import torchaudio
from datasets import load_dataset, load_metric
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import re
import librosa

test_dataset = load_dataset("common_voice", "ka", split="test") wer = load_metric("wer")

processor = Wav2Vec2Processor.from_pretrained("xsway/wav2vec2-large-xlsr-georgian") model = Wav2Vec2ForCTC.from_pretrained("xsway/wav2vec2-large-xlsr-georgian") model.to("cuda")

chars_to_ignore_regex = '[\\\\\\\\,\\\\\\\\?\\\\\\\\.\\\\\\\\!\\\\\\\\-\\\\\\\\;\\\\\\\\:\\\\\\\\"\\\\\\\\“]' resampler = lambda sampling_rate, y: librosa.resample(y.numpy().squeeze(), sampling_rate, 16_000)

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"]).lower() speech_array, sampling_rate = torchaudio.load(batch["path"]) batch["speech"] = resampler(sampling_rate, speech_array).squeeze() return batch

test_dataset = test_dataset.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) batch["pred_strings"] = processor.batch_decode(pred_ids) return batch

result = test_dataset.map(evaluate, batched=True, batch_size=8)

print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"], references=result["sentence"])))


Test Result: 45.28 %

Training



The Common Voice train, validation datasets were used for training.

The script used for training can be found here

Files & Weights

FilenameSizeAction
pytorch_model.bin 1.18 GB