* Rename bigdl/llm to ipex_llm * rm python/llm/src/bigdl * from bigdl.llm to from ipex_llm
		
			
				
	
	
		
			69 lines
		
	
	
	
		
			3 KiB
		
	
	
	
		
			Python
		
	
	
	
	
	
			
		
		
	
	
			69 lines
		
	
	
	
		
			3 KiB
		
	
	
	
		
			Python
		
	
	
	
	
	
#
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# Copyright 2016 The BigDL Authors.
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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#     http://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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#
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import time
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import argparse
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from ipex_llm import optimize_model
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from datasets import load_dataset
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from transformers import AutoModelForSpeechSeq2Seq, pipeline
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from transformers.models.whisper import WhisperFeatureExtractor, WhisperTokenizer
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if __name__ == '__main__':
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     parser = argparse.ArgumentParser(description='Recognize Long Segment using `generate()` API for Distil-Whisper model')
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     parser.add_argument('--repo-id-or-model-path', type=str, default="distil-whisper/distil-large-v2",
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                         help='The huggingface repo id for the Distil-Whisper model to be downloaded'
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                              ', or the path to the huggingface checkpoint folder')
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     parser.add_argument('--repo-id-or-data-path', type=str,
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                         default="distil-whisper/librispeech_long",
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                         help='The huggingface repo id for the audio dataset to be downloaded'
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                              ', or the path to the huggingface dataset folder')
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     parser.add_argument('--language', type=str, default="english",
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                         help='language to be transcribed')
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     parser.add_argument('--batch-size', type=int, default=16,
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                         help='The batch_size of pipeline inference, '
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                              'it usually equals of length of the audio divided by chunk-length.')
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     parser.add_argument('--chunk-length', type=int, default=15,
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                         help="The maximum time lengths of chuncks of sampling_rate samples used to trim"
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                              "and pad longer or shorter audio sequences. Default to be 30s.")
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     args = parser.parse_args()
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     model_path = args.repo_id_or_model_path
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     dataset_path = args.repo_id_or_data_path
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     # Load dummy dataset and read audio files
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     dataset = load_dataset(dataset_path, "clean", split="validation")
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     audio = dataset[0]["audio"]
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     model = AutoModelForSpeechSeq2Seq.from_pretrained(model_path)
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     model = optimize_model(model)
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     model.config.forced_decoder_ids = None
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     pipe = pipeline(
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          "automatic-speech-recognition",
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          model=model,
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          feature_extractor=WhisperFeatureExtractor.from_pretrained(model_path),
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          tokenizer= WhisperTokenizer.from_pretrained(model_path, language=args.language),
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          chunk_length_s=args.chunk_length,
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     )
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     start = time.time()
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     prediction = pipe(audio, batch_size=args.batch_size)["text"]
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     print(f"inference time is {time.time()-start}")
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     print(prediction)
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