ipex-llm/python/llm/test/benchmark/gpu/whisper.py
xingyuan li 704a896e90 [LLM] Add perf test on xpu for bigdl-llm (#8866)
* add xpu latency job
* update install way
* remove duplicated workflow
* add perf upload
2023-09-05 17:36:24 +09:00

75 lines
2.7 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 argparse
import intel_extension_for_pytorch as ipex
from transformers import WhisperProcessor
import torch
import time
from benchmark_util import BenchmarkWrapper
from bigdl.llm.transformers import AutoModelForSpeechSeq2Seq
from datasets import load_dataset, load_from_disk
if __name__ == '__main__':
parser = argparse.ArgumentParser('OPT generation script', add_help=False)
parser.add_argument('-m', '--model-dir',
default="/mnt/disk1/models/whisper-medium", type=str)
args = parser.parse_args()
print(args)
model_path = args.model_dir
dataset_path = "hf-internal-testing/librispeech_asr_dummy"
# load model and processor
ds = load_dataset(dataset_path, "clean", split="validation")
print("pass")
processor = WhisperProcessor.from_pretrained(model_path)
print("model loaded")
# load dummy dataset and read audio files
sample = ds[0]["audio"]
# for transformer == 4.30.2
input_features = processor(
sample["array"], sampling_rate=sample["sampling_rate"], return_tensors="pt").input_features
input_features = input_features.half().contiguous()
input_features = input_features.to('xpu')
print(input_features.shape)
print(input_features.is_contiguous())
# generate token ids
whisper = AutoModelForSpeechSeq2Seq.from_pretrained(
model_path, load_in_4bit=True, optimize_model=False)
whisper.config.forced_decoder_ids = None
whisper = whisper.half().to('xpu')
whisper = BenchmarkWrapper(whisper)
with torch.inference_mode():
e2e_time = []
for i in range(10):
torch.xpu.synchronize()
st = time.time()
predicted_ids = whisper.generate(input_features)
# print(len(predicted_ids[0]))
torch.xpu.synchronize()
output_str = processor.batch_decode(
predicted_ids, skip_special_tokens=True)
end = time.time()
e2e_time.append(end-st)
print(f'Inference time: {end-st} s')
print('Output:', output_str)
print(f'Inference time: {end-st} s')
print(e2e_time)