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license: apache-2.0 |
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This is a d-Matrix functional reference of the whisper-medium model. |
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The reference provides the following functional *configurations*: |
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Configuration | Explanation |
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:-- | :-- |
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**`BASELINE`** | a reference functionally equivalent to the original model |
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**`BASIC`** | all linear algebraic operands quantized to `MXINT8-64` |
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### Usage |
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Install d-Matrix [Dmx_Compressor](https://github.com/d-matrix-ai/dmx-compressor) first. |
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```sh |
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pip install dmx_compressor |
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``` |
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The following is an example model and its evaluation. |
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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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from dmx.compressor.modeling import DmxModel |
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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 = "d-matrix/whisper-medium" |
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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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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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shorter_audio = sample["array"][:1000] |
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pipe.model = DmxModel.from_torch(pipe.model) |
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result = pipe(shorter_audio) |
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print(result["text"]) |
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``` |