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@@ -43,15 +43,58 @@ should probably proofread and complete it, then remove this comment. -->
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  This model is a fine-tuned version of [openai/whisper-medium](https://huggingface.co/openai/whisper-medium) on the Hindi 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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- ## Training and evaluation data at Speech Lab, IITM
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- Training Data: GramVaani ASR Corpus, ULCA ASR Corpus, Shrutilipi ASR Corpus, Google/Fleurs (Train+Dev) set.
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- Evaluation Data: GramVaani ASR Corpus Test, Google/Fleurs Test set.
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- ## Training procedure
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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, IITM. 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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  This model is a fine-tuned version of [openai/whisper-medium](https://huggingface.co/openai/whisper-medium) on the Hindi 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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+
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+ ```python
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+ >>> import torch
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+ >>> from transformers import pipeline
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+
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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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+
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+ >>> transcribe = pipeline(task="automatic-speech-recognition", model="vasista22/whisper-hindi-medium", chunk_length_s=30, device=device)
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+ >>> transcribe.model.config.forced_decoder_ids = transcribe.tokenizer.get_decoder_prompt_ids(language="hi", task="transcribe")
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+
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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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+
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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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+
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+ >>> transcribe = FlaxWhisperPipline("vasista22/whisper-hindi-medium", batch_size=16)
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+ >>> transcribe.model.config.forced_decoder_ids = transcribe.tokenizer.get_decoder_prompt_ids(language="hi", task="transcribe")
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+
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+ >>> print('Transcription: ', transcribe(audio)["text"])
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+ ```
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+
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+ ## Training and evaluation data
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+ Training Data:
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+ - [GramVaani ASR Corpus](https://sites.google.com/view/gramvaaniasrchallenge/dataset?authuser=0)
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+ - [ULCA ASR Corpus](https://github.com/Open-Speech-EkStep/ULCA-asr-dataset-corpus#hindi-labelled--total-duration-is-239876-hours)
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+ - [Shrutilipi ASR Corpus](https://ai4bharat.org/shrutilipi)
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+ - [Google/Fleurs Train+Dev set](https://huggingface.co/datasets/google/fleurs)
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+ Evaluation Data:
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+ - [GramVaani ASR Corpus Test Set](https://sites.google.com/view/gramvaaniasrchallenge/dataset?authuser=0)
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+ - [Google/Fleurs Test Set](https://huggingface.co/datasets/google/fleurs)
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+
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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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+
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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.