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Merge branch 'main' of github.com:alibaba-damo-academy/FunASR

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游雁 2023-03-17 22:59:52 +08:00
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egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/infer.py Normal file
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import torch
import torchaudio
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model='damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online',
model_revision='v1.0.2')
waveform, sample_rate = torchaudio.load("asr_example_zh.wav")
speech_length = waveform.shape[1]
speech = waveform[0]
cache_en = {"start_idx": 0, "pad_left": 0, "stride": 10, "pad_right": 5, "cif_hidden": None, "cif_alphas": None}
cache_de = {"decode_fsmn": None}
cache = {"encoder": cache_en, "decoder": cache_de}
param_dict = {}
param_dict["cache"] = cache
first_chunk = True
speech_buffer = speech
speech_cache = []
final_result = ""
while len(speech_buffer) > 0:
if first_chunk:
if len(speech_buffer) >= 14400:
rec_result = inference_pipeline(audio_in=speech_buffer[0:14400], param_dict=param_dict)
speech_buffer = speech_buffer[4800:]
else:
cache_en["stride"] = len(speech_buffer) // 960
cache_en["pad_right"] = 0
rec_result = inference_pipeline(audio_in=speech_buffer, param_dict=param_dict)
speech_buffer = []
cache_en["start_idx"] = -5
first_chunk = False
else:
cache_en["start_idx"] += 10
if len(speech_buffer) >= 4800:
cache_en["pad_left"] = 5
rec_result = inference_pipeline(audio_in=speech_buffer[:19200], param_dict=param_dict)
speech_buffer = speech_buffer[9600:]
else:
cache_en["stride"] = len(speech_buffer) // 960
cache_en["pad_right"] = 0
rec_result = inference_pipeline(audio_in=speech_buffer, param_dict=param_dict)
speech_buffer = []
if rec_result['text'] != "sil":
final_result += rec_result['text']
print(rec_result)
print(final_result)