Improve model card: add link to code and example usage
Browse filesThis PR improves the model card by adding:
- A code snippet for example usage
    	
        README.md
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            ---
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            license: apache-2.0
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            library_name: transformers
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            base_model: openai/whisper-large-v3
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              - automatic-speech-recognition
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              - whisper
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              - hf-asr-leaderboard
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            pipeline_tag: automatic-speech-recognition
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            ---
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            # Model Card for Lite-Whisper large-v3
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            Lite-Whisper is a compressed version of OpenAI Whisper with LiteASR. See our [GitHub repository](https://github.com/efeslab/LiteASR) and [paper](https://arxiv.org/abs/2502.20583) for details.
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            ## Benchmark Results
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            Following is the average word error rate (WER) evaluated on the [ESB datasets](https://huggingface.co/datasets/hf-audio/esb-datasets-test-only-sorted):
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            ---
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            base_model: openai/whisper-large-v3
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            library_name: transformers
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            license: apache-2.0
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            pipeline_tag: automatic-speech-recognition
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            tags:
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            - audio
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            - automatic-speech-recognition
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            - whisper
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            - hf-asr-leaderboard
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            ---
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            # Model Card for Lite-Whisper large-v3
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            Lite-Whisper is a compressed version of OpenAI Whisper with LiteASR. See our [GitHub repository](https://github.com/efeslab/LiteASR) and [paper](https://arxiv.org/abs/2502.20583) for details.
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            Here's a code snippet to get started:
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            ```python
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            import librosa 
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            import torch
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            from transformers import AutoProcessor, AutoModel
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            device = "cuda:0"
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            dtype = torch.float16
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            # load the compressed Whisper model
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            model = AutoModel.from_pretrained(
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                "efficient-speech/lite-whisper-large-v3-turbo", 
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                trust_remote_code=True, 
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            )
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            model.to(dtype).to(device)
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            # we use the same processor as the original model
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            processor = AutoProcessor.from_pretrained("openai/whisper-large-v3")
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            # set the path to your audio file
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            path = "path/to/audio.wav"
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            audio, _ = librosa.load(path, sr=16000)
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            input_features = processor(audio, sampling_rate=16000, return_tensors="pt").input_features
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            input_features = input_features.to(dtype).to(device)
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            predicted_ids = model.generate(input_features)
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            transcription = processor.batch_decode(
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                predicted_ids, 
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                skip_special_tokens=True
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            )[0]
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            print(transcription)
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            ```
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            ## Benchmark Results
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            Following is the average word error rate (WER) evaluated on the [ESB datasets](https://huggingface.co/datasets/hf-audio/esb-datasets-test-only-sorted):
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