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111 lines
4.6 KiB
Python
111 lines
4.6 KiB
Python
import whisper
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import torch
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import wave
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import os
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import threading
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from tempfile import NamedTemporaryFile
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import speech_recognition as sr
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import io
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from datetime import timedelta
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from time import sleep
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import pyaudiowpatch as pyaudio
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from AudioRecorder import DefaultMicRecorder, DefaultSpeakerRecorder
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from heapq import merge
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PHRASE_TIMEOUT = 4
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class AudioTranscriber:
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def __init__(self, default_mic : DefaultMicRecorder, default_speaker : DefaultSpeakerRecorder):
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self.mic_transcript_data = []
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self.speaker_transcript_data = []
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self.transcript_changed_event = threading.Event()
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self.audio_model = whisper.load_model(os.getcwd() + r'\tiny.en' + '.pt')
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self.mic_sample_rate = default_mic.source.SAMPLE_RATE
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self.mic_sample_width = default_mic.source.SAMPLE_WIDTH
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self.mic_channels = default_mic.num_channels
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self.speaker_sample_rate = default_speaker.source.SAMPLE_RATE
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self.speaker_sample_rate = default_speaker.source.SAMPLE_RATE
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self.speaker_channels = default_speaker.num_channels
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def create_transcription_from_queue(self, audio_queue):
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mic_last_sample = bytes()
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speaker_last_sample = bytes()
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mic_last_spoken = None
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speaker_last_spoken = None
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mic_start_new_phrase = True
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speaker_start_new_phrase = True
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while True:
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top_of_queue = audio_queue.get()
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who_spoke = top_of_queue[0]
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data = top_of_queue[1]
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time_spoken = top_of_queue[2]
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if who_spoke == "You":
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if mic_last_spoken and time_spoken - mic_last_spoken > timedelta(seconds=PHRASE_TIMEOUT):
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mic_last_sample = bytes()
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mic_start_new_phrase = True
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else:
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mic_start_new_phrase = False
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mic_last_sample += data
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mic_last_spoken = time_spoken
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mic_temp_file = NamedTemporaryFile().name
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audio_data = sr.AudioData(mic_last_sample, self.mic_sample_rate, self.mic_sample_width)
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wav_data = io.BytesIO(audio_data.get_wav_data())
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with open(mic_temp_file, 'w+b') as f:
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f.write(wav_data.read())
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result = self.audio_model.transcribe(mic_temp_file, fp16=torch.cuda.is_available())
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text = result['text'].strip()
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if text != '' and text.lower() != 'you':
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if mic_start_new_phrase or len(self.mic_transcript_data) == 0:
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self.mic_transcript_data = [(who_spoke + ": [" + text + ']\n\n', time_spoken)] + self.mic_transcript_data
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self.transcript_changed_event.set()
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else:
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self.mic_transcript_data[0] = (who_spoke + ": [" + text + ']\n\n',
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time_spoken)
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self.transcript_changed_event.set()
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else:
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if speaker_last_spoken and time_spoken - speaker_last_spoken > timedelta(seconds=PHRASE_TIMEOUT):
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speaker_last_sample = bytes()
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speaker_start_new_phrase = True
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else:
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speaker_start_new_phrase = False
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speaker_last_sample += data
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speaker_last_spoken = time_spoken
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speaker_temp_file = NamedTemporaryFile().name
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with wave.open(speaker_temp_file, 'wb') as wf:
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wf.setnchannels(self.speaker_channels)
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p = pyaudio.PyAudio()
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wf.setsampwidth(p.get_sample_size(pyaudio.paInt16))
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wf.setframerate(self.speaker_sample_rate)
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wf.writeframes(speaker_last_sample)
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result = self.audio_model.transcribe(speaker_temp_file, fp16=torch.cuda.is_available())
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text = result['text'].strip()
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if text != '' and text.lower() != 'you':
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if speaker_start_new_phrase or len(self.speaker_transcript_data) == 0:
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self.speaker_transcript_data = [(who_spoke + ": [" + text + ']\n\n', time_spoken)] + self.speaker_transcript_data
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self.transcript_changed_event.set()
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else:
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self.speaker_transcript_data[0] = (who_spoke + ": [" + text + ']\n\n',
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time_spoken)
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self.transcript_changed_event.set()
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def get_transcript(self):
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key = lambda x : x[1]
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transcript_tuple = list(merge(self.mic_transcript_data, self.speaker_transcript_data, key=key, reverse=True))
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return "".join([t[0] for t in transcript_tuple])
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