license: apache-2.0
datasets:
- librispeech_asr
metrics:
- wer
pipeline_tag: automatic-speech-recognition
tags:
- automatic-speech-recognition
- ONNX
- Intel® Neural Compressor
- neural-compressor
library_name: transformers
INT4 Whisper medium ONNX Model
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 is the repository of INT4 weight only quantization for the Whisper medium model in ONNX format, powered by Intel® Neural Compressor and Intel® Extension for Transformers.
This INT4 ONNX model is generated by Intel® Neural Compressor's weight-only quantization method.
Model Detail | Description |
---|---|
Model Authors - Company | Intel |
Date | October 8, 2023 |
Version | 1 |
Type | Speech Recognition |
Paper or Other Resources | - |
License | Apache 2.0 |
Questions or Comments | Community Tab |
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. |
Export to ONNX Model
The FP32 model is exported with openai/whisper-medium:
optimum-cli export onnx --model openai/whisper-medium whisper-medium-with-past/ --task automatic-speech-recognition-with-past --opset 13
Install ONNX Runtime
Install onnxruntime>=1.16.0
to support MatMulFpQ4
operator.
Run Quantization
Build Intel® Neural Compressor from master branch and run INT4 weight-only quantization.
The weight-only quantization cofiguration is as below:
dtype | group_size | scheme | algorithm |
---|---|---|---|
INT4 | 32 | sym | RTN |
We provide the key code below. For the complete script, please refer to whisper example.
from neural_compressor import quantization, PostTrainingQuantConfig
from neural_compressor.utils.constant import FP32
model_list = ['encoder_model.onnx', 'decoder_model.onnx', 'decoder_with_past_model.onnx']
for model in model_list:
config = PostTrainingQuantConfig(
approach="weight_only",
calibration_sampling_size=[8],
op_type_dict={".*": {"weight": {"bits": 4,
"algorithm": ["RTN"],
"scheme": ["sym"],
"group_size": 32}}},)
q_model = quantization.fit(
os.path.join("/path/to/whisper-medium-with-past", model), # FP32 model path
config,
calib_dataloader=dataloader)
q_model.save(os.path.join("/path/to/whisper-medium-onnx-int4", model)) # INT4 model path
Evaluation
Operator Statistics
Below shows the operator statistics in the INT4 ONNX model:
Model | Op Type | Total | INT4 weight | FP32 weight |
---|---|---|---|---|
encoder_model | MatMul | 192 | 144 | 48 |
decoder_model | MatMul | 337 | 241 | 96 |
decoder_with_past_model | MatMul | 289 | 193 | 96 |
Evaluation of wer
Evaluate the model on librispeech_asr
dataset with below code:
import os
from evaluate import load
from datasets import load_dataset
from transformers import WhisperForConditionalGeneration, WhisperProcessor, AutoConfig
model_name = 'openai/whisper-medium'
model_path = 'whisper-medium-onnx-int4'
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}")
Metrics (Model Performance):
Model | Model Size (GB) | wer |
---|---|---|
FP32 | 4.9 | 2.88 |
INT4 | 1.1 | 2.98 |