Automatic speech recognition using Distil-Whisper and OpenVINO

This Jupyter notebook can be launched after a local installation only.

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Distil-Whisper is a distilled variant of the Whisper model by OpenAI. The Distil-Whisper is proposed in the paper Robust Knowledge Distillation via Large-Scale Pseudo Labelling. According to authors, compared to Whisper, Distil-Whisper runs in several times faster with 50% fewer parameters, while performing to within 1% word error rate (WER) on out-of-distribution evaluation data.

Whisper is a Transformer based encoder-decoder model, also referred to as a sequence-to-sequence model. It maps a sequence of audio spectrogram features to a sequence of text tokens. First, the raw audio inputs are converted to a log-Mel spectrogram by action of the feature extractor. Then, the Transformer encoder encodes the spectrogram to form a sequence of encoder hidden states. Finally, the decoder autoregressively predicts text tokens, conditional on both the previous tokens and the encoder hidden states.

You can see the model architecture in the diagram below:

whisper_architecture.svg

whisper_architecture.svg

In this tutorial, we consider how to run Distil-Whisper using OpenVINO. We will use the pre-trained model from the Hugging Face Transformers library. To simplify the user experience, the Hugging Face Optimum library is used to convert the model to OpenVINO™ IR format. To further improve OpenVINO Distil-Whisper model performance INT8 post-training quantization from NNCF is applied.

Table of contents:

Prerequisites

%pip install -q "transformers>=4.35" "torch>=2.1" onnx "git+https://github.com/huggingface/optimum-intel.git" "peft==0.6.2" --extra-index-url https://download.pytorch.org/whl/cpu
%pip install -q "openvino>=2023.2.0" datasets  "gradio>=4.0" "librosa" "soundfile"
%pip install -q "nncf>=2.6.0" "jiwer"

Load PyTorch model

The AutoModelForSpeechSeq2Seq.from_pretrained method is used for the initialization of PyTorch Whisper model using the transformers library. By default, we will use the distil-whisper/distil-large-v2 model as an example in this tutorial. The model will be downloaded once during first run and this process may require some time.

You may also choose other models from Distil-Whisper hugging face collection such as distil-whisper/distil-medium.en or distil-whisper/distil-small.en. Models of the original Whisper architecture are also available, more on them here.

Preprocessing and post-processing are important in this model use. AutoProcessor class used for initialization WhisperProcessor is responsible for preparing audio input data for the model, converting it to Mel-spectrogram and decoding predicted output token_ids into string using tokenizer.

import ipywidgets as widgets

model_ids = {
    "Distil-Whisper": [
        "distil-whisper/distil-large-v2",
        "distil-whisper/distil-medium.en",
        "distil-whisper/distil-small.en",
    ],
    "Whisper": [
        "openai/whisper-large-v3",
        "openai/whisper-large-v2",
        "openai/whisper-large",
        "openai/whisper-medium",
        "openai/whisper-small",
        "openai/whisper-base",
        "openai/whisper-tiny",
        "openai/whisper-medium.en",
        "openai/whisper-small.en",
        "openai/whisper-base.en",
        "openai/whisper-tiny.en",
    ],
}

model_type = widgets.Dropdown(
    options=model_ids.keys(),
    value="Distil-Whisper",
    description="Model type:",
    disabled=False,
)

model_type
model_id = widgets.Dropdown(
    options=model_ids[model_type.value],
    value=model_ids[model_type.value][0],
    description="Model:",
    disabled=False,
)

model_id
from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq

processor = AutoProcessor.from_pretrained(model_id.value)

pt_model = AutoModelForSpeechSeq2Seq.from_pretrained(model_id.value)
pt_model.eval();
Special tokens have been added in the vocabulary, make sure the associated word embeddings are fine-tuned or trained.

Prepare input sample

The processor expects audio data in numpy array format and information about the audio sampling rate and returns the input_features tensor for making predictions. Conversion of audio to numpy format is handled by Hugging Face datasets implementation.

from datasets import load_dataset


def extract_input_features(sample):
    input_features = processor(
        sample["audio"]["array"],
        sampling_rate=sample["audio"]["sampling_rate"],
        return_tensors="pt",
    ).input_features
    return input_features


dataset = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
sample = dataset[0]
input_features = extract_input_features(sample)

Run model inference

To perform speech recognition, one can use generate interface of the model. After generation is finished processor.batch_decode can be used for decoding predicted token_ids into text transcription.

import IPython.display as ipd

predicted_ids = pt_model.generate(input_features)
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)

display(ipd.Audio(sample["audio"]["array"], rate=sample["audio"]["sampling_rate"]))
print(f"Reference: {sample['text']}")
print(f"Result: {transcription[0]}")
Reference: MISTER QUILTER IS THE APOSTLE OF THE MIDDLE CLASSES AND WE ARE GLAD TO WELCOME HIS GOSPEL
Result:  Mr. Quilter is the apostle of the middle classes, and we are glad to welcome his gospel.

Load OpenVINO model using Optimum library

The Hugging Face Optimum API is a high-level API that enables us to convert and quantize models from the Hugging Face Transformers library to the OpenVINO™ IR format. For more details, refer to the Hugging Face Optimum documentation.

Optimum Intel can be used to load optimized models from the Hugging Face Hub and create pipelines to run an inference with OpenVINO Runtime using Hugging Face APIs. The Optimum Inference models are API compatible with Hugging Face Transformers models. This means we just need to replace the AutoModelForXxx class with the corresponding OVModelForXxx class.

Below is an example of the distil-whisper model

-from transformers import AutoModelForSpeechSeq2Seq
+from optimum.intel.openvino import OVModelForSpeechSeq2Seq
from transformers import AutoTokenizer, pipeline

model_id = "distil-whisper/distil-large-v2"
-model = AutoModelForSpeechSeq2Seq.from_pretrained(model_id)
+model = OVModelForSpeechSeq2Seq.from_pretrained(model_id, export=True)

Model class initialization starts with calling the from_pretrained method. When downloading and converting the Transformers model, the parameter export=True should be added. We can save the converted model for the next usage with the save_pretrained method. Tokenizers and Processors are distributed with models also compatible with the OpenVINO model. It means that we can reuse initialized early processor.

from pathlib import Path
from optimum.intel.openvino import OVModelForSpeechSeq2Seq

model_path = Path(model_id.value.replace("/", "_"))
ov_config = {"CACHE_DIR": ""}

if not model_path.exists():
    ov_model = OVModelForSpeechSeq2Seq.from_pretrained(
        model_id.value,
        ov_config=ov_config,
        export=True,
        compile=False,
        load_in_8bit=False,
    )
    ov_model.half()
    ov_model.save_pretrained(model_path)
else:
    ov_model = OVModelForSpeechSeq2Seq.from_pretrained(model_path, ov_config=ov_config, compile=False)
INFO:nncf:NNCF initialized successfully. Supported frameworks detected: torch, onnx, openvino

Select Inference device

import openvino as ov
import ipywidgets as widgets

core = ov.Core()

device = widgets.Dropdown(
    options=core.available_devices + ["AUTO"],
    value="AUTO",
    description="Device:",
    disabled=False,
)

device
Dropdown(description='Device:', index=4, options=('CPU', 'GPU.0', 'GPU.1', 'GPU.2', 'AUTO'), value='AUTO')

Compile OpenVINO model

ov_model.to(device.value)
ov_model.compile()
Compiling the encoder to AUTO ...
Compiling the decoder to AUTO ...
Compiling the decoder to AUTO ...

Run OpenVINO model inference

predicted_ids = ov_model.generate(input_features)
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)

display(ipd.Audio(sample["audio"]["array"], rate=sample["audio"]["sampling_rate"]))
print(f"Reference: {sample['text']}")
print(f"Result: {transcription[0]}")
/home/nsavel/venvs/ov_notebooks_tmp/lib/python3.8/site-packages/optimum/intel/openvino/modeling_seq2seq.py:457: FutureWarning: shared_memory is deprecated and will be removed in 2024.0. Value of shared_memory is going to override share_inputs value. Please use only share_inputs explicitly.
  last_hidden_state = torch.from_numpy(self.request(inputs, shared_memory=True)["last_hidden_state"]).to(
/home/nsavel/venvs/ov_notebooks_tmp/lib/python3.8/site-packages/optimum/intel/openvino/modeling_seq2seq.py:538: FutureWarning: shared_memory is deprecated and will be removed in 2024.0. Value of shared_memory is going to override share_inputs value. Please use only share_inputs explicitly.
  self.request.start_async(inputs, shared_memory=True)