[Update] 機能部分をmodelsフォルダに移動
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misygauziya
91
models/transcription/transcription_languages.py
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91
models/transcription/transcription_languages.py
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transcription_lang = {
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"Japanese (Japan)":"ja-JP",
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"English (United States)":"en-US",
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"English (United Kingdom)":"en-GB",
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"Afrikaans (South Africa)":"af-ZA",
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"Arabic (Algeria)":"ar-DZ",
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"Arabic (Bahrain)":"ar-BH",
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"Arabic (Egypt)":"ar-EG",
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"Arabic (Israel)":"ar-IL",
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"Arabic (Iraq)":"ar-IQ",
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"Arabic (Jordan)":"ar-JO",
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"Arabic (Kuwait)":"ar-KW",
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"Arabic (Lebanon)":"ar-LB",
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"Arabic (Morocco)":"ar-MA",
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"Arabic (Oman)":"ar-OM",
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"Arabic (State of Palestine)":"ar-PS",
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"Arabic (Qatar)":"ar-QA",
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"Arabic (Saudi Arabia)":"ar-SA",
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"Arabic (Tunisia)":"ar-TN",
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"Arabic (United Arab Emirates)":"ar-AE",
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"Basque (Spain)":"eu-ES",
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"Bulgarian (Bulgaria)":"bg-BG",
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"Catalan (Spain)":"ca-ES",
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"Chinese, Mandarin (Simplified, China)":"cmn-Hans-CN",
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"Chinese, Mandarin (Simplified, Hong Kong)":"cmn-Hans-HK",
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"Chinese, Mandarin (Traditional, Taiwan)":"cmn-Hant-TW",
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"Chinese, Cantonese (Traditional Hong Kong)":"yue-Hant-HK",
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"Croatian (Croatia)":"hr-HR",
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"Czech (Czech Republic)":"cs-CZ",
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"Danish (Denmark)":"da-DK",
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"English (Australia)":"en-AU",
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"English (Canada)":"en-CA",
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"English (India)":"en-IN",
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"English (Ireland)":"en-IE",
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"English (New Zealand)":"en-NZ",
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"English (Philippines)":"en-PH",
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"English (South Africa)":"en-ZA",
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"Persian (Iran)":"fa-IR",
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"French (France)":"fr-FR",
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"Filipino (Philippines)":"fil-PH",
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"Galician (Spain)":"gl-ES",
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"German (Germany)":"de-DE",
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"Greek (Greece)":"el-GR",
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"Finnish (Finland)":"fi-FI",
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"Hebrew (Israel)":"he-IL",
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"Hindi (India)":"hi-IN",
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"Hungarian (Hungary)":"hu-HU",
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"Indonesian (Indonesia)":"id-ID",
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"Icelandic (Iceland)":"is-IS",
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"Italian (Italy)":"it-IT",
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"Italian (Switzerland)":"it-CH",
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"Korean (South Korea)":"ko-KR",
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"Lithuanian (Lithuania)":"lt-LT",
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"Malay (Malaysia)":"ms-MY",
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"Dutch (Netherlands)":"nl-NL",
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"Norwegian Bokmål (Norway)":"nb-NO",
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"Polish (Poland)":"pl-PL",
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"Portuguese (Brazil)":"pt-BR",
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"Portuguese (Portugal)":"pt-PT",
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"Romanian (Romania)":"ro-RO",
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"Russian (Russia)":"ru-RU",
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"Serbian (Serbia)":"sr-RS",
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"Slovak (Slovakia)":"sk-SK",
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"Slovenian (Slovenia)":"sl-SI",
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"Spanish (Argentina)":"es-AR",
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"Spanish (Bolivia)":"es-BO",
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"Spanish (Chile)":"es-CL",
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"Spanish (Colombia)":"es-CO",
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"Spanish (Costa Rica)":"es-CR",
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"Spanish (Dominican Republic)":"es-DO",
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"Spanish (Ecuador)":"es-EC",
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"Spanish (El Salvador)":"es-SV",
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"Spanish (Guatemala)":"es-GT",
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"Spanish (Honduras)":"es-HN",
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"Spanish (Mexico)":"es-MX",
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"Spanish (Nicaragua)":"es-NI",
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"Spanish (Panama)":"es-PA",
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"Spanish (Paraguay)":"es-PY",
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"Spanish (Peru)":"es-PE",
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"Spanish (Puerto Rico)":"es-PR",
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"Spanish (Spain)":"es-ES",
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"Spanish (Uruguay)":"es-UY",
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"Spanish (United States)":"es-US",
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"Spanish (Venezuela)":"es-VE",
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"Swedish (Sweden)":"sv-SE",
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"Thai (Thailand)":"th-TH",
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"Turkish (Turkey)":"tr-TR",
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"Ukrainian (Ukraine)":"uk-UA",
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"Vietnamese (Vietnam)":"vi-VN",
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"Zulu (South Africa)":"zu-ZA"
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}
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91
models/transcription/transcription_recorder.py
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models/transcription/transcription_recorder.py
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from speech_recognition import Recognizer, Microphone
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from pyaudiowpatch import get_sample_size, paInt16
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from datetime import datetime
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class BaseRecorder:
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def __init__(self, source, energy_threshold, dynamic_energy_threshold, record_timeout):
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self.recorder = Recognizer()
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self.recorder.energy_threshold = energy_threshold
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self.recorder.dynamic_energy_threshold = dynamic_energy_threshold
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self.record_timeout = record_timeout
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self.stop = None
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if source is None:
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raise ValueError("audio source can't be None")
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self.source = source
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def adjust_for_noise(self):
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with self.source:
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self.recorder.adjust_for_ambient_noise(self.source)
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def record_into_queue(self, audio_queue):
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def record_callback(_, audio):
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audio_queue.put((audio.get_raw_data(), datetime.now()))
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self.stop = self.recorder.listen_in_background(self.source, record_callback, phrase_time_limit=self.record_timeout)
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class SelectedMicRecorder(BaseRecorder):
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def __init__(self, device, energy_threshold, dynamic_energy_threshold, record_timeout):
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source=Microphone(
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device_index=device['index'],
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sample_rate=int(device["defaultSampleRate"]),
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)
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super().__init__(source=source, energy_threshold=energy_threshold, dynamic_energy_threshold=dynamic_energy_threshold, record_timeout=record_timeout)
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# self.adjust_for_noise()
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class SelectedSpeakerRecorder(BaseRecorder):
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def __init__(self, device, energy_threshold, dynamic_energy_threshold, record_timeout):
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source = Microphone(speaker=True,
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device_index= device["index"],
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sample_rate=int(device["defaultSampleRate"]),
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chunk_size=get_sample_size(paInt16),
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channels=device["maxInputChannels"]
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)
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super().__init__(source=source, energy_threshold=energy_threshold, dynamic_energy_threshold=dynamic_energy_threshold, record_timeout=record_timeout)
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# self.adjust_for_noise()
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class BaseEnergyRecorder:
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def __init__(self, source):
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self.recorder = Recognizer()
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self.recorder.energy_threshold = 0
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self.recorder.dynamic_energy_threshold = False
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self.record_timeout = 0
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self.stop = None
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if source is None:
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raise ValueError("audio source can't be None")
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self.source = source
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def adjust_for_noise(self):
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with self.source:
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self.recorder.adjust_for_ambient_noise(self.source)
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def record_into_queue(self, energy_queue):
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def record_callback(_, energy):
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energy_queue.put(energy)
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self.stop = self.recorder.listen_energy_in_background(self.source, record_callback)
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class SelectedMicEnergyRecorder(BaseEnergyRecorder):
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def __init__(self, device):
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source=Microphone(
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device_index=device['index'],
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sample_rate=int(device["defaultSampleRate"]),
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)
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super().__init__(source=source)
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# self.adjust_for_noise()
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class SelectedSpeakeEnergyRecorder(BaseEnergyRecorder):
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def __init__(self, device):
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source = Microphone(speaker=True,
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device_index= device["index"],
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sample_rate=int(device["defaultSampleRate"]),
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chunk_size=get_sample_size(paInt16),
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channels=device["maxInputChannels"]
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)
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super().__init__(source=source)
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# self.adjust_for_noise()
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98
models/transcription/transcription_transcriber.py
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98
models/transcription/transcription_transcriber.py
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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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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.process_speaker_data if speaker else self.process_speaker_data
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}
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def transcribe_audio_queue(self, audio_queue, language):
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# while True:
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audio, time_spoken = audio_queue.get()
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self.update_last_sample_and_phrase_status(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])
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except Exception as e:
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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.update_transcript(text)
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def update_last_sample_and_phrase_status(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 process_mic_data(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 process_speaker_data(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 update_transcript(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 get_transcript(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 clear_transcript_data(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
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52
models/transcription/transcription_utils.py
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52
models/transcription/transcription_utils.py
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from pyaudiowpatch import PyAudio, paWASAPI
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def get_input_device_list():
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devices = {}
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with PyAudio() as p:
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for host_index in range(0, p.get_host_api_count()):
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host = p.get_host_api_info_by_index(host_index)
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for device_index in range(0, p.get_host_api_info_by_index(host_index)['deviceCount']):
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device = p.get_device_info_by_host_api_device_index(host_index, device_index)
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if device["maxInputChannels"] > 0 and device["isLoopbackDevice"] is False:
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if host["name"] in devices.keys():
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devices[host["name"]].append(device)
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else:
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devices[host["name"]] = [device]
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return devices
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def get_output_device_list():
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devices =[]
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with PyAudio() as p:
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wasapi_info = p.get_host_api_info_by_type(paWASAPI)
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for device in p.get_loopback_device_info_generator():
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if device["hostApi"] == wasapi_info["index"] and device["isLoopbackDevice"] is True:
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devices.append(device)
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return devices
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def get_default_input_device():
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with PyAudio() as p:
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api_info = p.get_default_host_api_info()
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defaultInputDevice = api_info["defaultInputDevice"]
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for host_index in range(0, p.get_host_api_count()):
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host = p.get_host_api_info_by_index(host_index)
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for device_index in range(0, p. get_host_api_info_by_index(host_index)['deviceCount']):
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device = p.get_device_info_by_host_api_device_index(host_index, device_index)
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if device["index"] == defaultInputDevice:
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return {"host":host, "device": device}
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def get_default_output_device():
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with PyAudio() as p:
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wasapi_info = p.get_host_api_info_by_type(paWASAPI)
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defaultOutputDevice = wasapi_info["defaultOutputDevice"]
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for host_index in range(0, p.get_host_api_count()):
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for device_index in range(0, p. get_host_api_info_by_index(host_index)['deviceCount']):
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device = p.get_device_info_by_host_api_device_index(host_index, device_index)
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if device["index"] == defaultOutputDevice:
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default_speakers = device
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if not default_speakers["isLoopbackDevice"]:
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for loopback in p.get_loopback_device_info_generator():
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if default_speakers["name"] in loopback["name"]:
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default_device = loopback
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return default_device
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