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README.md
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@@ -351,8 +351,8 @@ This code snippet shows how to evaluate Whisper Medium on [LibriSpeech test-clea
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The Whisper model is intrinsically designed to work on audio samples of up to 30s in duration. However, by using a chunking
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algorithm, it can be used to transcribe audio samples of up to arbitrary length. This is possible through Transformers
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[`pipeline`](https://huggingface.co/docs/transformers/main_classes/pipelines#transformers.AutomaticSpeechRecognitionPipeline)
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method. Chunking is enabled by setting `chunk_length_s=30` when instantiating the pipeline.
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predict
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```python
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>>> import torch
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>>> pipe = pipeline(
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>>> "automatic-speech-recognition",
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>>> model="openai/whisper-
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>>> chunk_length_s=30,
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>>> device=device,
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>>> )
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>>> ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
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>>> sample = ds[0]["audio"]
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>>> prediction = pipe(sample.copy())["text"]
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" Mr. Quilter is the apostle of the middle classes, and we are glad to welcome his gospel."
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>>> # we can also return timestamps for the predictions
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>>> prediction = pipe(sample, return_timestamps=True)["chunks"]
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[{'text': ' Mr. Quilter is the apostle of the middle classes and we are glad to welcome his gospel.',
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'timestamp': (0.0, 5.44)}]
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```
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## Fine-Tuning
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The pre-trained Whisper model demonstrates a strong ability to generalise to different datasets and domains. However,
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The Whisper model is intrinsically designed to work on audio samples of up to 30s in duration. However, by using a chunking
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algorithm, it can be used to transcribe audio samples of up to arbitrary length. This is possible through Transformers
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[`pipeline`](https://huggingface.co/docs/transformers/main_classes/pipelines#transformers.AutomaticSpeechRecognitionPipeline)
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method. Chunking is enabled by setting `chunk_length_s=30` when instantiating the pipeline. With chunking enabled, the pipeline
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can be run with batched inference. It can also be extended to predict sequence level timestamps by passing `return_timestamps=True`:
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```python
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>>> import torch
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>>> pipe = pipeline(
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>>> "automatic-speech-recognition",
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>>> model="openai/whisper-large-v2",
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>>> chunk_length_s=30,
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>>> device=device,
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>>> )
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>>> ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
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>>> sample = ds[0]["audio"]
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>>> prediction = pipe(sample.copy(), batch_size=8)["text"]
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" Mr. Quilter is the apostle of the middle classes, and we are glad to welcome his gospel."
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>>> # we can also return timestamps for the predictions
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>>> prediction = pipe(sample.copy(), batch_size=8, return_timestamps=True)["chunks"]
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[{'text': ' Mr. Quilter is the apostle of the middle classes and we are glad to welcome his gospel.',
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'timestamp': (0.0, 5.44)}]
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```
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Refer to the blog post [ASR Chunking](https://huggingface.co/blog/asr-chunking) for more details on the chunking algorithm.
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## Fine-Tuning
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The pre-trained Whisper model demonstrates a strong ability to generalise to different datasets and domains. However,
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