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README.md
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This model is a fine-tuned version of [openai/whisper-medium](https://huggingface.co/openai/whisper-medium) on the Tamil data available from multiple publicly available ASR corpuses.
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It has been fine-tuned as a part of the Whisper fine-tuning sprint.
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The following hyperparameters were used during training:
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- learning_rate: 1e-05
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- mixed_precision_training: True
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## Acknowledgement
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This work was done at Speech Lab,
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This model is a fine-tuned version of [openai/whisper-medium](https://huggingface.co/openai/whisper-medium) on the Tamil data available from multiple publicly available ASR corpuses.
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It has been fine-tuned as a part of the Whisper fine-tuning sprint.
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**NOTE:** The code used to train this model is available for re-use in the [whisper-finetune](https://github.com/vasistalodagala/whisper-finetune) repository.
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## Usage
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In order to evaluate this model on an entire dataset, the evaluation codes available in the [whisper-finetune](https://github.com/vasistalodagala/whisper-finetune) repository can be used.
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The same repository also provides the scripts for faster inference using whisper-jax.
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In order to infer a single audio file using this model, the following code snippet can be used:
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```python
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>>> import torch
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>>> from transformers import pipeline
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>>> # path to the audio file to be transcribed
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>>> audio = "/path/to/audio.format"
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>>> device = "cuda:0" if torch.cuda.is_available() else "cpu"
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>>> transcribe = pipeline(task="automatic-speech-recognition", model="vasista22/whisper-tamil-medium", chunk_length_s=30, device=device)
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>>> transcribe.model.config.forced_decoder_ids = transcribe.tokenizer.get_decoder_prompt_ids(language="ta", task="transcribe")
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>>> print('Transcription: ', transcribe(audio)["text"])
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```
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For faster inference of whisper models, the [whisper-jax](https://github.com/sanchit-gandhi/whisper-jax) library can be used. Please follow the necessary installation steps as mentioned [here](https://github.com/vasistalodagala/whisper-finetune#faster-evaluation-with-whisper-jax), before using the following code snippet:
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```python
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>>> import jax.numpy as jnp
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>>> from whisper_jax import FlaxWhisperForConditionalGeneration, FlaxWhisperPipline
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>>> # path to the audio file to be transcribed
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>>> audio = "/path/to/audio.format"
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>>> transcribe = FlaxWhisperPipline("vasista22/whisper-tamil-medium", batch_size=16)
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>>> transcribe.model.config.forced_decoder_ids = transcribe.tokenizer.get_decoder_prompt_ids(language="ta", task="transcribe")
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>>> print('Transcription: ', transcribe(audio)["text"])
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```
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## Training and evaluation data
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Training Data:
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- [IISc-MILE Tamil ASR Corpus](https://www.openslr.org/127/)
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- [ULCA ASR Corpus](https://github.com/Open-Speech-EkStep/ULCA-asr-dataset-corpus#tamil-labelled--total-duration-is-116024-hours)
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- [Shrutilipi ASR Corpus](https://ai4bharat.org/shrutilipi)
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- [Microsoft Speech Corpus (Indian Languages)](https://msropendata.com/datasets/7230b4b1-912d-400e-be58-f84e0512985e)
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- [Google/Fleurs Train+Dev set](https://huggingface.co/datasets/google/fleurs)
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- Babel ASR Corpus
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Evaluation Data:
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- [Microsoft Speech Corpus (Indian Languages) Test Set](https://msropendata.com/datasets/7230b4b1-912d-400e-be58-f84e0512985e)
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- [Google/Fleurs Test Set](https://huggingface.co/datasets/google/fleurs)
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- [IISc-MILE Test Set](https://www.openslr.org/127/)
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- Babel Test Set
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## Training hyperparameters
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The following hyperparameters were used during training:
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- learning_rate: 1e-05
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- mixed_precision_training: True
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## Acknowledgement
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This work was done at [Speech Lab, IIT Madras](https://asr.iitm.ac.in/).
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The compute resources for this work were funded by "Bhashini: National Language translation Mission" project of the Ministry of Electronics and Information Technology (MeitY), Government of India.
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