# # 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. # from datasets import load_dataset from bigdl.llm.transformers import AutoModelForSpeechSeq2Seq from transformers import WhisperProcessor import torch from evaluate import load import time import argparse import pandas as pd import os import csv from datetime import date current_dir = os.path.dirname(os.path.realpath(__file__)) def get_args(): parser = argparse.ArgumentParser(description="Evaluate Whisper performance and accuracy") parser.add_argument('--model_path', required=True, help='pretrained model path') parser.add_argument('--data_type', required=True, help='clean, other') parser.add_argument('--device', required=False, help='cpu, xpu') parser.add_argument('--load_in_low_bit', default='sym_int4', help='Specify whether to load data in low bit format (e.g., 4-bit)') parser.add_argument('--save_result', action='store_true', help='Save the results to a CSV file') args = parser.parse_args() return args if __name__ == '__main__': args = get_args() if args.device == "": args.device = "cpu" speech_dataset = load_dataset('./librispeech_asr.py', name=args.data_type, split='test').select(range(500)) processor = WhisperProcessor.from_pretrained(args.model_path) forced_decoder_ids = processor.get_decoder_prompt_ids(language='en', task='transcribe') model = AutoModelForSpeechSeq2Seq.from_pretrained(args.model_path, load_in_low_bit=args.load_in_low_bit, optimize_model=True).eval().to(args.device) model.config.forced_decoder_ids = None def map_to_pred(batch): audio = batch["audio"] start_time = time.time() input_features = processor(audio["array"], sampling_rate=audio["sampling_rate"], return_tensors="pt").input_features batch["reference"] = processor.tokenizer._normalize(batch['text']) with torch.no_grad(): predicted_ids = model.generate(input_features.to(args.device), forced_decoder_ids=forced_decoder_ids, use_cache=True)[0] if args.device == "xpu": torch.xpu.synchronize() infer_time = time.time() - start_time transcription = processor.decode(predicted_ids) batch["prediction"] = processor.tokenizer._normalize(transcription) batch["length"] = len(audio["array"])/audio["sampling_rate"] batch["time"] = infer_time print(batch["reference"]) print(batch["prediction"]) return batch result = speech_dataset.map(map_to_pred, keep_in_memory=True) wer = load("./wer") speech_length = sum(result["length"][1:]) prc_time = sum(result["time"][1:]) MODEL = args.model_path.split('/')[-2] RTF = prc_time/speech_length RTX = speech_length/prc_time WER = 100 * wer.compute(references=result["reference"], predictions=result["prediction"]) today = date.today() if args.save_result: csv_name = f'{current_dir}/results/{MODEL}-{args.data_type}-{args.device}-{args.load_in_low_bit}-{today}.csv' os.makedirs(os.path.dirname(csv_name), exist_ok=True) with open(csv_name, mode='a', newline='') as file: csv_writer = csv.writer(file) file.seek(0, os.SEEK_END) if file.tell() == 0: csv_writer.writerow(["models","precision","WER","RTF"]) csv_writer.writerow([MODEL, args.load_in_low_bit, WER, RTF]) print(f'Results saved to {csv_name}') print("Realtime Factor(RTF) is : %.4f" % RTF) print("Realtime X(RTX) is : %.2f" % RTX) print(f'WER is {WER}')