Files
VRCT/audio_transcriber.py
misyaguziya 55e7fab281 rename files
2023-06-30 22:02:48 +09:00

97 lines
3.4 KiB
Python

import io
import threading
import wave
import custom_speech_recognition as sr
from datetime import timedelta
import pyaudiowpatch as pyaudio
PHRASE_TIMEOUT = 3
MAX_PHRASES = 10
class AudioTranscriber:
def __init__(self, speaker, source, language):
self.speaker = speaker
self.language = language
self.transcript_data = []
self.transcript_changed_event = threading.Event()
self.audio_recognizer = sr.Recognizer()
self.audio_sources = {
"sample_rate": source.SAMPLE_RATE,
"sample_width": source.SAMPLE_WIDTH,
"channels": source.channels,
"last_sample": bytes(),
"last_spoken": None,
"new_phrase": True,
"process_data_func": self.process_speaker_data if speaker else self.process_speaker_data
}
def transcribe_audio_queue(self, audio_queue):
# while True:
audio, time_spoken = audio_queue.get()
self.update_last_sample_and_phrase_status(audio, time_spoken)
text = ''
try:
# fd, path = tempfile.mkstemp(suffix=".wav")
# os.close(fd)
audio_data = self.audio_sources["process_data_func"]()
text = self.audio_recognizer.recognize_google(audio_data, language=self.language)
except Exception as e:
pass
finally:
pass
# os.unlink(path)
if text != '':
self.update_transcript(text)
def update_last_sample_and_phrase_status(self, data, time_spoken):
source_info = self.audio_sources
if source_info["last_spoken"] and time_spoken - source_info["last_spoken"] > timedelta(seconds=PHRASE_TIMEOUT):
source_info["last_sample"] = bytes()
source_info["new_phrase"] = True
else:
source_info["new_phrase"] = False
source_info["last_sample"] += data
source_info["last_spoken"] = time_spoken
def process_mic_data(self):
audio_data = sr.AudioData(self.audio_sources["last_sample"], self.audio_sources["sample_rate"], self.audio_sources["sample_width"])
return audio_data
def process_speaker_data(self):
temp_file = io.BytesIO()
with wave.open(temp_file, 'wb') as wf:
wf.setnchannels(self.audio_sources["channels"])
p = pyaudio.PyAudio()
wf.setsampwidth(p.get_sample_size(pyaudio.paInt16))
wf.setframerate(self.audio_sources["sample_rate"])
wf.writeframes(self.audio_sources["last_sample"])
temp_file.seek(0)
with sr.AudioFile(temp_file) as source:
audio = self.audio_recognizer.record(source)
return audio
def update_transcript(self, text):
source_info = self.audio_sources
transcript = self.transcript_data
if source_info["new_phrase"] or len(transcript) == 0:
if len(transcript) > MAX_PHRASES:
transcript.pop(-1)
transcript.insert(0, text)
else:
transcript[0] = text
def get_transcript(self):
if len(self.transcript_data) > 0:
text = self.transcript_data.pop(-1)
else:
text = ""
return text
def clear_transcript_data(self):
self.transcript_data.clear()
self.audio_sources["last_sample"] = bytes()
self.audio_sources["new_phrase"] = True