135 lines
4.3 KiB
Markdown
135 lines
4.3 KiB
Markdown
---
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language: ta
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#datasets:
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#- Interspeech 2021
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metrics:
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- wer
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tags:
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- audio
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- automatic-speech-recognition
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- speech
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license: mit
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model-index:
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- name: Wav2Vec2 Vakyansh Tamil Model by Harveen Chadha
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results:
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- task:
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name: Speech Recognition
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type: automatic-speech-recognition
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dataset:
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name: Common Voice ta
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type: common_voice
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args: ta
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metrics:
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- name: Test WER
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type: wer
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value: 53.64
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---
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## Pretrained Model
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Fine-tuned on Multilingual Pretrained Model [CLSRIL-23](https://arxiv.org/abs/2107.07402). The original fairseq checkpoint is present [here](https://github.com/Open-Speech-EkStep/vakyansh-models). When using this model, make sure that your speech input is sampled at 16kHz.
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**Note: The result from this model is without a language model so you may witness a higher WER in some cases.**
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## Dataset
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This model was trained on 4200 hours of Hindi Labelled Data. The labelled data is not present in public domain as of now.
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## Training Script
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Models were trained using experimental platform setup by Vakyansh team at Ekstep. Here is the [training repository](https://github.com/Open-Speech-EkStep/vakyansh-wav2vec2-experimentation).
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In case you want to explore training logs on wandb they are [here](https://wandb.ai/harveenchadha/tamil-finetuning-multilingual).
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## [Colab Demo](https://github.com/harveenchadha/bol/blob/main/demos/hf/tamil/hf_tamil_tnm_4200_demo.ipynb)
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## Usage
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The model can be used directly (without a language model) as follows:
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```python
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import soundfile as sf
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import torch
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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import argparse
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def parse_transcription(wav_file):
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# load pretrained model
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processor = Wav2Vec2Processor.from_pretrained("Harveenchadha/vakyansh-wav2vec2-tamil-tam-250")
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model = Wav2Vec2ForCTC.from_pretrained("Harveenchadha/vakyansh-wav2vec2-tamil-tam-250")
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# load audio
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audio_input, sample_rate = sf.read(wav_file)
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# pad input values and return pt tensor
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input_values = processor(audio_input, sampling_rate=sample_rate, return_tensors="pt").input_values
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# INFERENCE
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# retrieve logits & take argmax
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logits = model(input_values).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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# transcribe
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transcription = processor.decode(predicted_ids[0], skip_special_tokens=True)
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print(transcription)
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```
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## Evaluation
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The model can be evaluated as follows on the hindi test data of Common Voice.
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```python
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import torch
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import torchaudio
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from datasets import load_dataset, load_metric
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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import re
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test_dataset = load_dataset("common_voice", "ta", split="test")
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wer = load_metric("wer")
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processor = Wav2Vec2Processor.from_pretrained("Harveenchadha/vakyansh-wav2vec2-tamil-tam-250")
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model = Wav2Vec2ForCTC.from_pretrained("Harveenchadha/vakyansh-wav2vec2-tamil-tam-250")
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model.to("cuda")
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resampler = torchaudio.transforms.Resample(48_000, 16_000)
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chars_to_ignore_regex = '[\,\?\.\!\-\;\:\"\“]'
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# Preprocessing the datasets.
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# We need to read the aduio files as arrays
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def speech_file_to_array_fn(batch):
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batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower()
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speech_array, sampling_rate = torchaudio.load(batch["path"])
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batch["speech"] = resampler(speech_array).squeeze().numpy()
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return batch
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test_dataset = test_dataset.map(speech_file_to_array_fn)
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# Preprocessing the datasets.
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# We need to read the aduio files as arrays
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def evaluate(batch):
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inputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
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with torch.no_grad():
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logits = model(inputs.input_values.to("cuda")).logits
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pred_ids = torch.argmax(logits, dim=-1)
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batch["pred_strings"] = processor.batch_decode(pred_ids, skip_special_tokens=True)
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return batch
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result = test_dataset.map(evaluate, batched=True, batch_size=8)
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print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"], references=result["sentence"])))
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```
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**Test Result**: 53.64 %
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[**Colab Evaluation**](https://github.com/harveenchadha/bol/blob/main/demos/hf/tamil/hf_vakyansh_tamil_tnm_4200_evaluation_common_voice.ipynb)
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## Credits
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Thanks to Ekstep Foundation for making this possible. The vakyansh team will be open sourcing speech models in all the Indic Languages. |