--- license: apache-2.0 datasets: - librispeech_asr metrics: - wer pipeline_tag: automatic-speech-recognition tags: - automatic-speech-recognition - int8 - ONNX - PostTrainingDynamic - IntelĀ® Neural Compressor - neural-compressor library_name: transformers --- ## Model Details: INT8 Whisper large Whisper is a pre-trained model for automatic speech recognition (ASR) and speech translation. Trained on 680k hours of labelled data, Whisper models demonstrate a strong ability to generalise to many datasets and domains without the need for fine-tuning. This int8 ONNX model is generated by [neural-compressor](https://github.com/intel/neural-compressor) and the fp32 model can be exported with below command: ```shell optimum-cli export onnx --model openai/whisper-large whisper-large-with-past/ --task automatic-speech-recognition-with-past --opset 13 ``` | Model Detail | Description | | ----------- | ----------- | | Model Authors - Company | Intel | | Date | May 15, 2022 | | Version | 1 | | Type | Speech Recognition | | Paper or Other Resources | - | | License | Apache 2.0 | | Questions or Comments | [Community Tab](https://huggingface.co/Intel/whisper-large-int8-dynamic/discussions)| | Intended Use | Description | | ----------- | ----------- | | Primary intended uses | You can use the raw model for automatic speech recognition inference | | Primary intended users | Anyone doing automatic speech recognition inference | | Out-of-scope uses | This model in most cases will need to be fine-tuned for your particular task. The model should not be used to intentionally create hostile or alienating environments for people.| ### How to use Download the model by cloning the repository: ```shell git clone https://huggingface.co/Intel/whisper-large-int8-dynamic ``` Evaluate the model with below code: ```python import os from evaluate import load from datasets import load_dataset from transformers import WhisperForConditionalGeneration, WhisperProcessor, AutoConfig model_name = 'openai/whisper-large' model_path = 'whisper-large-int8-dynamic' processor = WhisperProcessor.from_pretrained(model_name) model = WhisperForConditionalGeneration.from_pretrained(model_name) config = AutoConfig.from_pretrained(model_name) wer = load("wer") librispeech_test_clean = load_dataset("librispeech_asr", "clean", split="test") from optimum.onnxruntime import ORTModelForSpeechSeq2Seq from transformers import PretrainedConfig model_config = PretrainedConfig.from_pretrained(model_name) predictions = [] references = [] sessions = ORTModelForSpeechSeq2Seq.load_model( os.path.join(model_path, 'encoder_model.onnx'), os.path.join(model_path, 'decoder_model.onnx'), os.path.join(model_path, 'decoder_with_past_model.onnx')) model = ORTModelForSpeechSeq2Seq(sessions[0], sessions[1], model_config, model_path, sessions[2]) for idx, batch in enumerate(librispeech_test_clean): audio = batch["audio"] input_features = processor(audio["array"], sampling_rate=audio["sampling_rate"], return_tensors="pt").input_features reference = processor.tokenizer._normalize(batch['text']) references.append(reference) predicted_ids = model.generate(input_features)[0] transcription = processor.decode(predicted_ids) prediction = processor.tokenizer._normalize(transcription) predictions.append(prediction) wer_result = wer.compute(references=references, predictions=predictions) print(f"Result wer: {wer_result * 100}") accuracy = 1 - wer_result print("Accuracy: %.5f" % accuracy) ``` ## Metrics (Model Performance): | Model | Model Size (GB) | wer | |---|:---:|:---:| | FP32 |9.4|3.04| | INT8 |2.4|2.89|