126 lines
4.6 KiB
Python
126 lines
4.6 KiB
Python
from io import BytesIO
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from threading import Event
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import wave
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from speech_recognition import Recognizer, AudioData, AudioFile
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from datetime import timedelta
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from pyaudiowpatch import get_sample_size, paInt16
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from .transcription_languages import transcription_lang
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import torch
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import numpy as np
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from faster_whisper import WhisperModel
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PHRASE_TIMEOUT = 3
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MAX_PHRASES = 10
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class AudioTranscriber:
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def __init__(self, speaker, source, phrase_timeout, max_phrases):
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self.speaker = speaker
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self.phrase_timeout = phrase_timeout
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self.max_phrases = max_phrases
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self.transcript_data = []
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self.transcript_changed_event = Event()
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self.audio_recognizer = Recognizer()
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self.audio_sources = {
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"sample_rate": source.SAMPLE_RATE,
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"sample_width": source.SAMPLE_WIDTH,
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"channels": source.channels,
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"last_sample": bytes(),
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"last_spoken": None,
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"new_phrase": True,
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"process_data_func": self.processSpeakerData if speaker else self.processSpeakerData
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}
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self.whisper_model = WhisperModel("base", device="cpu", device_index=0, compute_type="int8", cpu_threads=4, num_workers=1)
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def transcribeAudioQueue(self, audio_queue, language, country):
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# while True:
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audio, time_spoken = audio_queue.get()
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self.updateLastSampleAndPhraseStatus(audio, time_spoken)
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text = ''
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try:
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# fd, path = tempfile.mkstemp(suffix=".wav")
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# os.close(fd)
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audio_data = self.audio_sources["process_data_func"]()
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text = self.audio_recognizer.recognize_google(audio_data, language=transcription_lang[language][country])
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audio_data = np.frombuffer(audio_data.get_raw_data(convert_rate=16000, convert_width=2), np.int16).flatten().astype(np.float32) / 32768.0
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if isinstance(audio_data, torch.Tensor):
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audio_data = audio_data.detach().numpy()
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segments, _ = self.whisper_model.transcribe(
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audio_data,
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beam_size=5,
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temperature=0.0,
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log_prob_threshold=-0.8,
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no_speech_threshold=0.6,
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language="ja",
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word_timestamps=False,
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without_timestamps=True,
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task="transcribe",
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vad_filter=False,
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)
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_text = ""
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for s in segments:
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if s.avg_logprob < -0.8 or s.no_speech_prob > 0.6:
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continue
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_text += s.text
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print(_text)
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except Exception:
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pass
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finally:
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pass
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# os.unlink(path)
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if text != '':
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self.updateTranscript(text)
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def updateLastSampleAndPhraseStatus(self, data, time_spoken):
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source_info = self.audio_sources
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if source_info["last_spoken"] and time_spoken - source_info["last_spoken"] > timedelta(seconds=self.phrase_timeout):
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source_info["last_sample"] = bytes()
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source_info["new_phrase"] = True
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else:
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source_info["new_phrase"] = False
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source_info["last_sample"] += data
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source_info["last_spoken"] = time_spoken
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def processMicData(self):
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audio_data = AudioData(self.audio_sources["last_sample"], self.audio_sources["sample_rate"], self.audio_sources["sample_width"])
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return audio_data
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def processSpeakerData(self):
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temp_file = BytesIO()
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with wave.open(temp_file, 'wb') as wf:
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wf.setnchannels(self.audio_sources["channels"])
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wf.setsampwidth(get_sample_size(paInt16))
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wf.setframerate(self.audio_sources["sample_rate"])
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wf.writeframes(self.audio_sources["last_sample"])
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temp_file.seek(0)
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with AudioFile(temp_file) as source:
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audio = self.audio_recognizer.record(source)
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return audio
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def updateTranscript(self, text):
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source_info = self.audio_sources
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transcript = self.transcript_data
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if source_info["new_phrase"] or len(transcript) == 0:
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if len(transcript) > self.max_phrases:
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transcript.pop(-1)
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transcript.insert(0, text)
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else:
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transcript[0] = text
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def getTranscript(self):
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if len(self.transcript_data) > 0:
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text = self.transcript_data.pop(-1)
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else:
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text = ""
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return text
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def clearTranscriptData(self):
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self.transcript_data.clear()
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self.audio_sources["last_sample"] = bytes()
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self.audio_sources["new_phrase"] = True |