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from faster_whisper import WhisperModel | |
model = None | |
model_size = None | |
def load_model(_model_size): | |
global model_size, model | |
if _model_size and model_size != _model_size: | |
model_size = _model_size | |
try: | |
model = WhisperModel(model_size, device="cuda", compute_type="float16") | |
except: | |
model = WhisperModel(model_size, device="cpu", compute_type="int8") | |
def speech_to_text(audio_file, _model_size = None): | |
global model_size, model | |
load_model(_model_size) | |
segments, info = model.transcribe( | |
audio_file, | |
language='ja', | |
beam_size=5, | |
vad_filter=True, | |
without_timestamps=False, | |
) | |
text_only = '' | |
text_with_timestamps = '' | |
for segment in segments: | |
text_only += f"{segment.text}\n" | |
text_with_timestamps += f"{segment.start:.2f}\t{segment.end:.2f}\t{segment.text}\n" | |
return text_only, text_with_timestamps | |