ipex-llm/python/llm/example/GPU/HF-Transformers-AutoModels/Model/whisper/recognize.py
Qiyuan Gong 0284801fbd [LLM] IPEX auto importer turn on by default for XPU (#9730)
* Set BIGDL_IMPORT_IPEX default to true, i.e., auto import IPEX for XPU.
* Remove import intel_extension_for_pytorch as ipex from GPU example.
* Add support for bigdl-core-xe-21.
2023-12-22 16:20:32 +08:00

77 lines
3.4 KiB
Python

#
# Copyright 2016 The BigDL Authors.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
#
import torch
import time
import argparse
from bigdl.llm.transformers import AutoModelForSpeechSeq2Seq
from transformers import WhisperProcessor
from datasets import load_dataset
if __name__ == '__main__':
parser = argparse.ArgumentParser(description='Recognize Tokens using `generate()` API for Whisper model')
parser.add_argument('--repo-id-or-model-path', type=str, default="openai/whisper-tiny",
help='The huggingface repo id for the Whisper model to be downloaded'
', or the path to the huggingface checkpoint folder')
parser.add_argument('--repo-id-or-data-path', type=str,
default="hf-internal-testing/librispeech_asr_dummy",
help='The huggingface repo id for the audio dataset to be downloaded'
', or the path to the huggingface dataset folder')
parser.add_argument('--language', type=str, default="english",
help='language to be transcribed')
args = parser.parse_args()
model_path = args.repo_id_or_model_path
dataset_path = args.repo_id_or_data_path
language = args.language
# Load model in 4 bit,
# which convert the relevant layers in the model into INT4 format
model = AutoModelForSpeechSeq2Seq.from_pretrained(model_path,
load_in_4bit=True,
optimize_model=False,
use_cache=True)
model.to('xpu')
model.config.forced_decoder_ids = None
# Load processor
processor = WhisperProcessor.from_pretrained(model_path)
forced_decoder_ids = processor.get_decoder_prompt_ids(language=language, task="transcribe")
# Load dummy dataset and read audio files
ds = load_dataset(dataset_path, "clean", split="validation")
# Generate predicted tokens
with torch.inference_mode():
sample = ds[0]["audio"]
input_features = processor(sample["array"],
sampling_rate=sample["sampling_rate"],
return_tensors="pt").input_features.to('xpu')
st = time.time()
# if your selected model is capable of utilizing previous key/value attentions
# to enhance decoding speed, but has `"use_cache": false` in its model config,
# it is important to set `use_cache=True` explicitly in the `generate` function
# to obtain optimal performance with BigDL-LLM INT4 optimizations
predicted_ids = model.generate(input_features,
forced_decoder_ids=forced_decoder_ids)
end = time.time()
output_str = processor.batch_decode(predicted_ids, skip_special_tokens=True)
print(f'Inference time: {end-st} s')
print('-'*20, 'Output', '-'*20)
print(output_str)