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--- |
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language: |
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- en |
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tags: |
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- audio |
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- automatic-speech-recognition |
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- transformers.js |
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widget: |
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- example_title: LibriSpeech sample 1 |
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src: https://cdn-media.huggingface.co/speech_samples/sample1.flac |
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- example_title: LibriSpeech sample 2 |
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src: https://cdn-media.huggingface.co/speech_samples/sample2.flac |
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pipeline_tag: automatic-speech-recognition |
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license: mit |
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library_name: transformers |
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--- |
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# Distil-Whisper: distil-large-v2 |
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Distil-Whisper was proposed in the paper [Robust Knowledge Distillation via Large-Scale Pseudo Labelling](https://arxiv.org/abs/2311.00430). |
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It is a distilled version of the Whisper model that is **6 times faster**, 49% smaller, and performs |
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**within 1% WER** on out-of-distribution evaluation sets. This is the repository for distil-large-v2, |
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a distilled variant of [Whisper large-v2](https://huggingface.co/openai/whisper-large-v2). |
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| Model | Params / M | Rel. Latency ↑ | Short-Form WER ↓ | Long-Form WER ↓ | |
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|----------------------------------------------------------------------------|------------|----------------|------------------|-----------------| |
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| [large-v3](https://huggingface.co/openai/whisper-large-v3) | 1550 | 1.0 | **8.4** | 11.0 | |
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| [large-v2](https://huggingface.co/openai/whisper-large-v2) | 1550 | 1.0 | 9.1 | 11.7 | |
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| | | | | | |
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| [distil-large-v3](https://huggingface.co/distil-whisper/distil-large-v3) | 756 | 6.3 | 9.7 | **10.8** | |
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| [distil-large-v2](https://huggingface.co/distil-whisper/distil-large-v2) | 756 | 5.8 | 10.1 | 11.6 | |
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| [distil-medium.en](https://huggingface.co/distil-whisper/distil-medium.en) | 394 | **6.8** | 11.1 | 12.4 | |
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| [distil-small.en](https://huggingface.co/distil-whisper/distil-small.en) | **166** | 5.6 | 12.1 | 12.8 | |
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<div class="course-tip course-tip-orange bg-gradient-to-br dark:bg-gradient-to-r before:border-orange-500 dark:before:border-orange-800 from-orange-50 dark:from-gray-900 to-white dark:to-gray-950 border border-orange-50 text-orange-700 dark:text-gray-400"> |
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<p><b>Update:</b> following the release of OpenAI's Whisper large-v3, an updated <a href="ttps://huggingface.co/distil-whisper/distil-large-v3"> distil-large-v3</a> model was published. This <a href="ttps://huggingface.co/distil-whisper/distil-large-v3"> distil-large-v3</a> model surpasses the performance of the distil-large-v2 model, with no architecture changes and better support for sequential long-form generation. Thus, it is recommended that the <a href="ttps://huggingface.co/distil-whisper/distil-large-v3"> distil-large-v3</a> model is used in-place of the large-v2 model. </p> |
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</div> |
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**Note:** Distil-Whisper is currently only available for English speech recognition. We are working with the community |
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to distill Whisper on other languages. If you are interested in distilling Whisper in your language, check out the |
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provided [training code](https://github.com/huggingface/distil-whisper/tree/main/training). We will update the |
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[Distil-Whisper repository](https://github.com/huggingface/distil-whisper/) with multilingual checkpoints when ready! |
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## Usage |
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Distil-Whisper is supported in Hugging Face 🤗 Transformers from version 4.35 onwards. To run the model, first |
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install the latest version of the Transformers library. For this example, we'll also install 🤗 Datasets to load toy |
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audio dataset from the Hugging Face Hub: |
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```bash |
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pip install --upgrade pip |
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pip install --upgrade transformers accelerate datasets[audio] |
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``` |
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### Short-Form Transcription |
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The model can be used with the [`pipeline`](https://huggingface.co/docs/transformers/main_classes/pipelines#transformers.AutomaticSpeechRecognitionPipeline) |
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class to transcribe short-form audio files (< 30-seconds) as follows: |
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```python |
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import torch |
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from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline |
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from datasets import load_dataset |
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device = "cuda:0" if torch.cuda.is_available() else "cpu" |
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torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32 |
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model_id = "distil-whisper/distil-large-v2" |
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model = AutoModelForSpeechSeq2Seq.from_pretrained( |
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model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True |
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) |
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model.to(device) |
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processor = AutoProcessor.from_pretrained(model_id) |
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pipe = pipeline( |
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"automatic-speech-recognition", |
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model=model, |
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tokenizer=processor.tokenizer, |
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feature_extractor=processor.feature_extractor, |
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max_new_tokens=128, |
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torch_dtype=torch_dtype, |
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device=device, |
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) |
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dataset = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation") |
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sample = dataset[0]["audio"] |
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result = pipe(sample) |
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print(result["text"]) |
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``` |
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To transcribe a local audio file, simply pass the path to your audio file when you call the pipeline: |
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```diff |
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- result = pipe(sample) |
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+ result = pipe("audio.mp3") |
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``` |
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### Long-Form Transcription |
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Distil-Whisper uses a chunked algorithm to transcribe long-form audio files (> 30-seconds). In practice, this chunked long-form algorithm |
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is 9x faster than the sequential algorithm proposed by OpenAI in the Whisper paper (see Table 7 of the [Distil-Whisper paper](https://arxiv.org/abs/2311.00430)). |
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To enable chunking, pass the `chunk_length_s` parameter to the `pipeline`. For Distil-Whisper, a chunk length of 15-seconds |
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is optimal. To activate batching, pass the argument `batch_size`: |
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```python |
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import torch |
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from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline |
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from datasets import load_dataset |
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device = "cuda:0" if torch.cuda.is_available() else "cpu" |
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torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32 |
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model_id = "distil-whisper/distil-large-v2" |
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model = AutoModelForSpeechSeq2Seq.from_pretrained( |
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model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True |
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) |
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model.to(device) |
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processor = AutoProcessor.from_pretrained(model_id) |
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pipe = pipeline( |
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"automatic-speech-recognition", |
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model=model, |
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tokenizer=processor.tokenizer, |
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feature_extractor=processor.feature_extractor, |
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max_new_tokens=128, |
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chunk_length_s=15, |
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batch_size=16, |
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torch_dtype=torch_dtype, |
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device=device, |
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) |
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dataset = load_dataset("distil-whisper/librispeech_long", "clean", split="validation") |
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sample = dataset[0]["audio"] |
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result = pipe(sample) |
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print(result["text"]) |
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``` |
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<!--- |
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**Tip:** The pipeline can also be used to transcribe an audio file from a remote URL, for example: |
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```python |
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result = pipe("https://huggingface.co/datasets/sanchit-gandhi/librispeech_long/resolve/main/audio.wav") |
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``` |
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---> |
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|
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### Speculative Decoding |
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Distil-Whisper can be used as an assistant model to Whisper for [speculative decoding](https://huggingface.co/blog/whisper-speculative-decoding). |
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Speculative decoding mathematically ensures the exact same outputs as Whisper are obtained while being 2 times faster. |
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This makes it the perfect drop-in replacement for existing Whisper pipelines, since the same outputs are guaranteed. |
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In the following code-snippet, we load the assistant Distil-Whisper model standalone to the main Whisper pipeline. We then |
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specify it as the "assistant model" for generation: |
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```python |
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from transformers import pipeline, AutoModelForCausalLM, AutoModelForSpeechSeq2Seq, AutoProcessor |
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import torch |
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from datasets import load_dataset |
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device = "cuda:0" if torch.cuda.is_available() else "cpu" |
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torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32 |
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assistant_model_id = "distil-whisper/distil-large-v2" |
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assistant_model = AutoModelForCausalLM.from_pretrained( |
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assistant_model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True |
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) |
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assistant_model.to(device) |
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model_id = "openai/whisper-large-v2" |
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model = AutoModelForSpeechSeq2Seq.from_pretrained( |
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model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True |
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) |
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model.to(device) |
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processor = AutoProcessor.from_pretrained(model_id) |
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pipe = pipeline( |
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"automatic-speech-recognition", |
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model=model, |
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tokenizer=processor.tokenizer, |
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feature_extractor=processor.feature_extractor, |
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max_new_tokens=128, |
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generate_kwargs={"assistant_model": assistant_model}, |
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torch_dtype=torch_dtype, |
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device=device, |
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) |
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dataset = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation") |
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sample = dataset[0]["audio"] |
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result = pipe(sample) |
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print(result["text"]) |
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``` |
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## Additional Speed & Memory Improvements |
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You can apply additional speed and memory improvements to Distil-Whisper which we cover in the following. |
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### Flash Attention |
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We recommend using [Flash-Attention 2](https://huggingface.co/docs/transformers/main/en/perf_infer_gpu_one#flashattention-2) if your GPU allows for it. |
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To do so, you first need to install [Flash Attention](https://github.com/Dao-AILab/flash-attention): |
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``` |
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pip install flash-attn --no-build-isolation |
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``` |
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and then all you have to do is to pass `use_flash_attention_2=True` to `from_pretrained`: |
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```diff |
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- model = AutoModelForSpeechSeq2Seq.from_pretrained(model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True) |
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+ model = AutoModelForSpeechSeq2Seq.from_pretrained(model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True, use_flash_attention_2=True) |
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``` |
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### Torch Scale-Product-Attention (SDPA) |
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If your GPU does not support Flash Attention, we recommend making use of [BetterTransformers](https://huggingface.co/docs/transformers/main/en/perf_infer_gpu_one#bettertransformer). |
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To do so, you first need to install optimum: |
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``` |
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pip install --upgrade optimum |
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``` |
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And then convert your model to a "BetterTransformer" model before using it: |
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```diff |
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model = AutoModelForSpeechSeq2Seq.from_pretrained(model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True) |
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+ model = model.to_bettertransformer() |
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``` |
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### Running Distil-Whisper in `openai-whisper` |
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To use the model in the original Whisper format, first ensure you have the [`openai-whisper`](https://pypi.org/project/openai-whisper/) package installed: |
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```bash |
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pip install --upgrade openai-whisper |
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``` |
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The following code-snippet demonstrates how to transcribe a sample file from the LibriSpeech dataset loaded using |
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🤗 Datasets: |
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```python |
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import torch |
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from datasets import load_dataset |
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from huggingface_hub import hf_hub_download |
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from whisper import load_model, transcribe |
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distil_large_v2 = hf_hub_download(repo_id="distil-whisper/distil-large-v2", filename="original-model.bin") |
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model = load_model(distil_large_v2) |
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dataset = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation") |
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sample = dataset[0]["audio"]["array"] |
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sample = torch.from_numpy(sample).float() |
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pred_out = transcribe(model, audio=sample) |
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print(pred_out["text"]) |
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``` |
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To transcribe a local audio file, simply pass the path to the audio file as the `audio` argument to transcribe: |
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```python |
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pred_out = transcribe(model, audio="audio.mp3") |
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``` |
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### Whisper.cpp |
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Distil-Whisper can be run from the [Whisper.cpp](https://github.com/ggerganov/whisper.cpp) repository with the original |
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sequential long-form transcription algorithm. In a [provisional benchmark](https://github.com/ggerganov/whisper.cpp/pull/1424#issuecomment-1793513399) |
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on Mac M1, `distil-large-v2` is 2x faster than `large-v2`, while performing to within 0.1% WER over long-form audio. |
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Note that future releases of Distil-Whisper will target faster CPU inference more! By distilling smaller encoders, we |
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aim to achieve similar speed-ups to what we obtain on GPU. |
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Steps for getting started: |
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1. Clone the Whisper.cpp repository: |
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``` |
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git clone https://github.com/ggerganov/whisper.cpp.git |
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cd whisper.cpp |
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``` |
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2. Download the ggml weights for `distil-medium.en` from the Hugging Face Hub: |
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```bash |
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python -c "from huggingface_hub import hf_hub_download; hf_hub_download(repo_id='distil-whisper/distil-large-v2', filename='ggml-large-32-2.en.bin', local_dir='./models')" |
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``` |
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Note that if you do not have the `huggingface_hub` package installed, you can also download the weights with `wget`: |
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```bash |
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wget https://huggingface.co/distil-whisper/distil-large-v2/resolve/main/ggml-large-32-2.en.bin -P ./models |
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``` |
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3. Run inference using the provided sample audio: |
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```bash |
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make -j && ./main -m models/ggml-large-32-2.en.bin -f samples/jfk.wav |
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``` |
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### Transformers.js |
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```js |
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import { pipeline } from '@xenova/transformers'; |
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let transcriber = await pipeline('automatic-speech-recognition', 'distil-whisper/distil-large-v2'); |
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let url = 'https://huggingface.co/datasets/Xenova/transformers.js-docs/resolve/main/jfk.wav'; |
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let output = await transcriber(url); |
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// { text: " And so, my fellow Americans, ask not what your country can do for you. Ask what you can do for your country." } |
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``` |
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See the [docs](https://huggingface.co/docs/transformers.js/api/pipelines#module_pipelines.AutomaticSpeechRecognitionPipeline) for more information. |
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*Note:* Due to the large model size, we recommend running this model server-side with [Node.js](https://huggingface.co/docs/transformers.js/guides/node-audio-processing) (instead of in-browser). |
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### Candle |
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Through an integration with Hugging Face [Candle](https://github.com/huggingface/candle/tree/main) 🕯️, Distil-Whisper is |
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now available in the Rust library 🦀 |
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Benefit from: |
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* Optimised CPU backend with optional MKL support for x86 and Accelerate for Macs |
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* CUDA backend for efficiently running on GPUs, multiple GPU distribution via NCCL |
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* WASM support: run Distil-Whisper in a browser |
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Steps for getting started: |
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1. Install [`candle-core`](https://github.com/huggingface/candle/tree/main/candle-core) as explained [here](https://huggingface.github.io/candle/guide/installation.html) |
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2. Clone the `candle` repository locally: |
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``` |
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git clone https://github.com/huggingface/candle.git |
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``` |
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3. Enter the example directory for [Whisper](https://github.com/huggingface/candle/tree/main/candle-examples/examples/whisper): |
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``` |
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cd candle/candle-examples/examples/whisper |
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``` |
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4. Run an example: |
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``` |
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cargo run --example whisper --release -- --model distil-large-v2 |
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``` |
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5. To specify your own audio file, add the `--input` flag: |
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``` |
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cargo run --example whisper --release -- --model distil-large-v2 --input audio.wav |
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``` |
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### 8bit & 4bit Quantization |
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Coming soon ... |
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### Whisper.cpp |
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Coming soon ... |
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## Model Details |
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Distil-Whisper inherits the encoder-decoder architecture from Whisper. The encoder maps a sequence of speech vector |
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inputs to a sequence of hidden-state vectors. The decoder auto-regressively predicts text tokens, conditional on all |
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previous tokens and the encoder hidden-states. Consequently, the encoder is only run forward once, whereas the decoder |
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is run as many times as the number of tokens generated. In practice, this means the decoder accounts for over 90% of |
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total inference time. Thus, to optimise for latency, the focus should be on minimising the inference time of the decoder. |
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|
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To distill the Whisper model, we reduce the number of decoder layers while keeping the encoder fixed. |
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The encoder (shown in green) is entirely copied from the teacher to the student and frozen during training. |
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The student's decoder consists of only two decoder layers, which are initialised from the first and last decoder layer of |
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the teacher (shown in red). All other decoder layers of the teacher are discarded. The model is then trained on a weighted sum |
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of the KL divergence and pseudo-label loss terms. |
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<p align="center"> |
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<img src="https://huggingface.co/datasets/distil-whisper/figures/resolve/main/architecture.png?raw=true" width="600"/> |
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</p> |
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## Evaluation |
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The following code-snippets demonstrates how to evaluate the Distil-Whisper model on the LibriSpeech validation.clean |
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dataset with [streaming mode](https://huggingface.co/blog/audio-datasets#streaming-mode-the-silver-bullet), meaning no |
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audio data has to be downloaded to your local device. |
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First, we need to install the required packages, including 🤗 Datasets to stream and load the audio data, and 🤗 Evaluate to |
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perform the WER calculation: |
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```bash |
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pip install --upgrade pip |
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pip install --upgrade transformers datasets[audio] evaluate jiwer |
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``` |
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Evaluation can then be run end-to-end with the following example: |
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|
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```python |
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from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor |
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from transformers.models.whisper.english_normalizer import EnglishTextNormalizer |
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from datasets import load_dataset |
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from evaluate import load |
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import torch |
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from tqdm import tqdm |
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|
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# define our torch configuration |
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device = "cuda:0" if torch.cuda.is_available() else "cpu" |
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torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32 |
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model_id = "distil-whisper/distil-large-v2" |
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# load the model + processor |
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model = AutoModelForSpeechSeq2Seq.from_pretrained(model_id, torch_dtype=torch_dtype, use_safetensors=True, low_cpu_mem_usage=True) |
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model = model.to(device) |
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processor = AutoProcessor.from_pretrained(model_id) |
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# load the dataset with streaming mode |
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dataset = load_dataset("librispeech_asr", "clean", split="validation", streaming=True) |
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# define the evaluation metric |
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wer_metric = load("wer") |
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normalizer = EnglishTextNormalizer(processor.tokenizer.english_spelling_normalizer) |
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|
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def inference(batch): |
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# 1. Pre-process the audio data to log-mel spectrogram inputs |
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audio = [sample["array"] for sample in batch["audio"]] |
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input_features = processor(audio, sampling_rate=batch["audio"][0]["sampling_rate"], return_tensors="pt").input_features |
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input_features = input_features.to(device, dtype=torch_dtype) |
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|
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# 2. Auto-regressively generate the predicted token ids |
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pred_ids = model.generate(input_features, max_new_tokens=128, language="en", task="transcribe") |
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|
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# 3. Decode the token ids to the final transcription |
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batch["transcription"] = processor.batch_decode(pred_ids, skip_special_tokens=True) |
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batch["reference"] = batch["text"] |
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return batch |
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dataset = dataset.map(function=inference, batched=True, batch_size=16) |
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|
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all_transcriptions = [] |
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all_references = [] |
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|
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# iterate over the dataset and run inference |
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for i, result in tqdm(enumerate(dataset), desc="Evaluating..."): |
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all_transcriptions.append(result["transcription"]) |
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all_references.append(result["reference"]) |
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|
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# normalize predictions and references |
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all_transcriptions = [normalizer(transcription) for transcription in all_transcriptions] |
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all_references = [normalizer(reference) for reference in all_references] |
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|
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# compute the WER metric |
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wer = 100 * wer_metric.compute(predictions=all_transcriptions, references=all_references) |
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print(wer) |
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|
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``` |
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**Print Output:** |
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``` |
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2.983685535968466 |
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``` |
|
|
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## Intended Use |
|
|
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Distil-Whisper is intended to be a drop-in replacement for Whisper on English speech recognition. In particular, it |
|
achieves comparable WER results over out-of-distribution test data, while being 6x faster over both short and long-form |
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audio. |
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## Data |
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Distil-Whisper is trained on 22,000 hours of audio data from 9 open-source, permissively licensed speech datasets on the |
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Hugging Face Hub: |
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| Dataset | Size / h | Speakers | Domain | Licence | |
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|-----------------------------------------------------------------------------------------|----------|----------|-----------------------------|-----------------| |
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| [People's Speech](https://huggingface.co/datasets/MLCommons/peoples_speech) | 12,000 | unknown | Internet Archive | CC-BY-SA-4.0 | |
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| [Common Voice 13](https://huggingface.co/datasets/mozilla-foundation/common_voice_13_0) | 3,000 | unknown | Narrated Wikipedia | CC0-1.0 | |
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| [GigaSpeech](https://huggingface.co/datasets/speechcolab/gigaspeech) | 2,500 | unknown | Audiobook, podcast, YouTube | apache-2.0 | |
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| Fisher | 1,960 | 11,900 | Telephone conversations | LDC | |
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| [LibriSpeech](https://huggingface.co/datasets/librispeech_asr) | 960 | 2,480 | Audiobooks | CC-BY-4.0 | |
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| [VoxPopuli](https://huggingface.co/datasets/facebook/voxpopuli) | 540 | 1,310 | European Parliament | CC0 | |
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| [TED-LIUM](https://huggingface.co/datasets/LIUM/tedlium) | 450 | 2,030 | TED talks | CC-BY-NC-ND 3.0 | |
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| SwitchBoard | 260 | 540 | Telephone conversations | LDC | |
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| [AMI](https://huggingface.co/datasets/edinburghcstr/ami) | 100 | unknown | Meetings | CC-BY-4.0 | |
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| **Total** | 21,770 | 18,260+ | | | |
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The combined dataset spans 10 distinct domains and over 50k speakers. The diversity of this dataset is crucial to ensuring |
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the distilled model is robust to audio distributions and noise. |
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The audio data is then pseudo-labelled using the Whisper large-v2 model: we use Whisper to generate predictions for all |
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the audio in our training set and use these as the target labels during training. Using pseudo-labels ensures that the |
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transcriptions are consistently formatted across datasets and provides sequence-level distillation signal during training. |
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## WER Filter |
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The Whisper pseudo-label predictions are subject to mis-transcriptions and hallucinations. To ensure we only train on |
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accurate pseudo-labels, we employ a simple WER heuristic during training. First, we normalise the Whisper pseudo-labels |
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and the ground truth labels provided by each dataset. We then compute the WER between these labels. If the WER exceeds |
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a specified threshold, we discard the training example. Otherwise, we keep it for training. |
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Section 9.2 of the [Distil-Whisper paper](https://arxiv.org/abs/2311.00430) demonstrates the effectiveness of this filter for improving downstream performance |
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of the distilled model. We also partially attribute Distil-Whisper's robustness to hallucinations to this filter. |
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## Training |
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The model was trained for 80,000 optimisation steps (or eight epochs). The Tensorboard training logs can be found under: https://huggingface.co/distil-whisper/distil-large-v2/tensorboard?params=scalars#frame |
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## Results |
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The distilled model performs to within 1% WER of Whisper on out-of-distribution (OOD) short-form audio, and outperforms Whisper |
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by 0.1% on OOD long-form audio. This performance gain is attributed to lower hallucinations. |
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For a detailed per-dataset breakdown of the evaluation results, refer to Tables 16 and 17 of the [Distil-Whisper paper](https://arxiv.org/abs/2311.00430) |
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Distil-Whisper is also evaluated on the [ESB benchmark](https://arxiv.org/abs/2210.13352) datasets as part of the [OpenASR leaderboard](https://huggingface.co/spaces/hf-audio/open_asr_leaderboard), |
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where it performs to within 0.2% WER of Whisper. |
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## Reproducing Distil-Whisper |
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Training and evaluation code to reproduce Distil-Whisper is available under the Distil-Whisper repository: https://github.com/huggingface/distil-whisper/tree/main/training |
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## License |
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Distil-Whisper inherits the [MIT license](https://github.com/huggingface/distil-whisper/blob/main/LICENSE) from OpenAI's Whisper model. |
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## Citation |
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If you use this model, please consider citing the [Distil-Whisper paper](https://arxiv.org/abs/2311.00430): |
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``` |
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@misc{gandhi2023distilwhisper, |
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title={Distil-Whisper: Robust Knowledge Distillation via Large-Scale Pseudo Labelling}, |
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author={Sanchit Gandhi and Patrick von Platen and Alexander M. Rush}, |
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year={2023}, |
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eprint={2311.00430}, |
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archivePrefix={arXiv}, |
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primaryClass={cs.CL} |
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} |
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``` |
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## Acknowledgements |
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* OpenAI for the Whisper [model](https://huggingface.co/openai/whisper-large-v2) and [original codebase](https://github.com/openai/whisper) |
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* Hugging Face 🤗 [Transformers](https://github.com/huggingface/transformers) for the model integration |
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* Google's [TPU Research Cloud (TRC)](https://sites.research.google/trc/about/) programme for Cloud TPU v4s |
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* [`@rsonavane`](https://huggingface.co/rsonavane/distil-whisper-large-v2-8-ls) for releasing an early iteration of Distil-Whisper on the LibriSpeech dataset |
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