| New file |
| | |
| | | import os |
| | | import logging |
| | | import torch |
| | | import soundfile |
| | | |
| | | from modelscope.pipelines import pipeline |
| | | from modelscope.utils.constant import Tasks |
| | | from modelscope.utils.logger import get_logger |
| | | |
| | | logger = get_logger(log_level=logging.CRITICAL) |
| | | logger.setLevel(logging.CRITICAL) |
| | | |
| | | os.environ["MODELSCOPE_CACHE"] = "./" |
| | | inference_pipeline = pipeline( |
| | | task=Tasks.auto_speech_recognition, |
| | | model='damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online', |
| | | model_revision='v1.0.4' |
| | | ) |
| | | |
| | | model_dir = os.path.join(os.environ["MODELSCOPE_CACHE"], "damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online") |
| | | speech, sample_rate = soundfile.read(os.path.join(model_dir, "example/asr_example.wav")) |
| | | speech_length = speech.shape[0] |
| | | |
| | | sample_offset = 0 |
| | | chunk_size = [8, 8, 4] #[5, 10, 5] 600ms, [8, 8, 4] 480ms |
| | | stride_size = chunk_size[1] * 960 |
| | | param_dict = {"cache": dict(), "is_final": False, "chunk_size": chunk_size} |
| | | final_result = "" |
| | | |
| | | for sample_offset in range(0, speech_length, min(stride_size, speech_length - sample_offset)): |
| | | if sample_offset + stride_size >= speech_length - 1: |
| | | stride_size = speech_length - sample_offset |
| | | param_dict["is_final"] = True |
| | | rec_result = inference_pipeline(audio_in=speech[sample_offset: sample_offset + stride_size], |
| | | param_dict=param_dict) |
| | | if len(rec_result) != 0: |
| | | final_result += rec_result['text'] |
| | | print(rec_result) |
| | | print(final_result.strip()) |