""" The AudioToTextRecorder class in the provided code facilitates fast speech-to-text transcription. The class employs the faster_whisper library to transcribe the recorded audio into text using machine learning models, which can be run either on a GPU or CPU. Voice activity detection (VAD) is built in, meaning the software can automatically start or stop recording based on the presence or absence of speech. It integrates wake word detection through the pvporcupine library, allowing the software to initiate recording when a specific word or phrase is spoken. The system provides real-time feedback and can be further customized. Features: - Voice Activity Detection: Automatically starts/stops recording when speech is detected or when speech ends. - Wake Word Detection: Starts recording when a specified wake word (or words) is detected. - Event Callbacks: Customizable callbacks for when recording starts or finishes. - Fast Transcription: Returns the transcribed text from the audio as fast as possible. Author: Kolja Beigel """ from typing import Iterable, List, Optional, Union import torch.multiprocessing as mp import torch from typing import List, Union from ctypes import c_bool from scipy.signal import resample from scipy import signal import faster_whisper import collections import numpy as np import pvporcupine import traceback import threading import webrtcvad import itertools import platform import pyaudio import logging import struct import halo import time import copy import os import re import gc # Set OpenMP runtime duplicate library handling to OK (Use only for development!) os.environ['KMP_DUPLICATE_LIB_OK'] = 'TRUE' INIT_MODEL_TRANSCRIPTION = "tiny" INIT_MODEL_TRANSCRIPTION_REALTIME = "tiny" INIT_REALTIME_PROCESSING_PAUSE = 0.2 INIT_SILERO_SENSITIVITY = 0.4 INIT_WEBRTC_SENSITIVITY = 3 INIT_POST_SPEECH_SILENCE_DURATION = 0.6 INIT_MIN_LENGTH_OF_RECORDING = 0.5 INIT_MIN_GAP_BETWEEN_RECORDINGS = 0 INIT_WAKE_WORDS_SENSITIVITY = 0.6 INIT_PRE_RECORDING_BUFFER_DURATION = 1.0 INIT_WAKE_WORD_ACTIVATION_DELAY = 0.0 INIT_WAKE_WORD_TIMEOUT = 5.0 ALLOWED_LATENCY_LIMIT = 10 TIME_SLEEP = 0.02 SAMPLE_RATE = 16000 BUFFER_SIZE = 512 INT16_MAX_ABS_VALUE = 32768.0 INIT_HANDLE_BUFFER_OVERFLOW = False if platform.system() != 'Darwin': INIT_HANDLE_BUFFER_OVERFLOW = True class AudioToTextRecorder: """ A class responsible for capturing audio from the microphone, detecting voice activity, and then transcribing the captured audio using the `faster_whisper` model. """ def __init__(self, model: str = INIT_MODEL_TRANSCRIPTION, language: str = "", compute_type: str = "default", input_device_index: int = 0, gpu_device_index: Union[int, List[int]] = 0, on_recording_start=None, on_recording_stop=None, on_transcription_start=None, ensure_sentence_starting_uppercase=True, ensure_sentence_ends_with_period=True, use_microphone=True, spinner=True, level=logging.WARNING, # Realtime transcription parameters enable_realtime_transcription=False, realtime_model_type=INIT_MODEL_TRANSCRIPTION_REALTIME, realtime_processing_pause=INIT_REALTIME_PROCESSING_PAUSE, on_realtime_transcription_update=None, on_realtime_transcription_stabilized=None, # Voice activation parameters silero_sensitivity: float = INIT_SILERO_SENSITIVITY, silero_use_onnx: bool = False, webrtc_sensitivity: int = INIT_WEBRTC_SENSITIVITY, post_speech_silence_duration: float = ( INIT_POST_SPEECH_SILENCE_DURATION ), min_length_of_recording: float = ( INIT_MIN_LENGTH_OF_RECORDING ), min_gap_between_recordings: float = ( INIT_MIN_GAP_BETWEEN_RECORDINGS ), pre_recording_buffer_duration: float = ( INIT_PRE_RECORDING_BUFFER_DURATION ), on_vad_detect_start=None, on_vad_detect_stop=None, # Wake word parameters wake_words: str = "", wake_words_sensitivity: float = INIT_WAKE_WORDS_SENSITIVITY, wake_word_activation_delay: float = ( INIT_WAKE_WORD_ACTIVATION_DELAY ), wake_word_timeout: float = INIT_WAKE_WORD_TIMEOUT, on_wakeword_detected=None, on_wakeword_timeout=None, on_wakeword_detection_start=None, on_wakeword_detection_end=None, on_recorded_chunk=None, debug_mode=False, handle_buffer_overflow: bool = INIT_HANDLE_BUFFER_OVERFLOW, beam_size: int = 5, beam_size_realtime: int = 3, buffer_size: int = BUFFER_SIZE, sample_rate: int = SAMPLE_RATE, initial_prompt: Optional[Union[str, Iterable[int]]] = None, suppress_tokens: Optional[List[int]] = [-1], ): """ Initializes an audio recorder and transcription and wake word detection. Args: - model (str, default="tiny"): Specifies the size of the transcription model to use or the path to a converted model directory. Valid options are 'tiny', 'tiny.en', 'base', 'base.en', 'small', 'small.en', 'medium', 'medium.en', 'large-v1', 'large-v2'. If a specific size is provided, the model is downloaded from the Hugging Face Hub. - language (str, default=""): Language code for speech-to-text engine. If not specified, the model will attempt to detect the language automatically. - compute_type (str, default="default"): Specifies the type of computation to be used for transcription. See https://opennmt.net/CTranslate2/quantization.html. - input_device_index (int, default=0): The index of the audio input device to use. - gpu_device_index (int, default=0): Device ID to use. The model can also be loaded on multiple GPUs by passing a list of IDs (e.g. [0, 1, 2, 3]). In that case, multiple transcriptions can run in parallel when transcribe() is called from multiple Python threads - on_recording_start (callable, default=None): Callback function to be called when recording of audio to be transcripted starts. - on_recording_stop (callable, default=None): Callback function to be called when recording of audio to be transcripted stops. - on_transcription_start (callable, default=None): Callback function to be called when transcription of audio to text starts. - ensure_sentence_starting_uppercase (bool, default=True): Ensures that every sentence detected by the algorithm starts with an uppercase letter. - ensure_sentence_ends_with_period (bool, default=True): Ensures that every sentence that doesn't end with punctuation such as "?", "!" ends with a period - use_microphone (bool, default=True): Specifies whether to use the microphone as the audio input source. If set to False, the audio input source will be the audio data sent through the feed_audio() method. - spinner (bool, default=True): Show spinner animation with current state. - level (int, default=logging.WARNING): Logging level. - enable_realtime_transcription (bool, default=False): Enables or disables real-time transcription of audio. When set to True, the audio will be transcribed continuously as it is being recorded. - realtime_model_type (str, default="tiny"): Specifies the machine learning model to be used for real-time transcription. Valid options include 'tiny', 'tiny.en', 'base', 'base.en', 'small', 'small.en', 'medium', 'medium.en', 'large-v1', 'large-v2'. - realtime_processing_pause (float, default=0.1): Specifies the time interval in seconds after a chunk of audio gets transcribed. Lower values will result in more "real-time" (frequent) transcription updates but may increase computational load. - on_realtime_transcription_update = A callback function that is triggered whenever there's an update in the real-time transcription. The function is called with the newly transcribed text as its argument. - on_realtime_transcription_stabilized = A callback function that is triggered when the transcribed text stabilizes in quality. The stabilized text is generally more accurate but may arrive with a slight delay compared to the regular real-time updates. - silero_sensitivity (float, default=SILERO_SENSITIVITY): Sensitivity for the Silero Voice Activity Detection model ranging from 0 (least sensitive) to 1 (most sensitive). Default is 0.5. - silero_use_onnx (bool, default=False): Enables usage of the pre-trained model from Silero in the ONNX (Open Neural Network Exchange) format instead of the PyTorch format. This is recommended for faster performance. - webrtc_sensitivity (int, default=WEBRTC_SENSITIVITY): Sensitivity for the WebRTC Voice Activity Detection engine ranging from 0 (least aggressive / most sensitive) to 3 (most aggressive, least sensitive). Default is 3. - post_speech_silence_duration (float, default=0.2): Duration in seconds of silence that must follow speech before the recording is considered to be completed. This ensures that any brief pauses during speech don't prematurely end the recording. - min_gap_between_recordings (float, default=1.0): Specifies the minimum time interval in seconds that should exist between the end of one recording session and the beginning of another to prevent rapid consecutive recordings. - min_length_of_recording (float, default=1.0): Specifies the minimum duration in seconds that a recording session should last to ensure meaningful audio capture, preventing excessively short or fragmented recordings. - pre_recording_buffer_duration (float, default=0.2): Duration in seconds for the audio buffer to maintain pre-roll audio (compensates speech activity detection latency) - on_vad_detect_start (callable, default=None): Callback function to be called when the system listens for voice activity. - on_vad_detect_stop (callable, default=None): Callback function to be called when the system stops listening for voice activity. - wake_words (str, default=""): Comma-separated string of wake words to initiate recording. Supported wake words include: 'alexa', 'americano', 'blueberry', 'bumblebee', 'computer', 'grapefruits', 'grasshopper', 'hey google', 'hey siri', 'jarvis', 'ok google', 'picovoice', 'porcupine', 'terminator'. - wake_words_sensitivity (float, default=0.5): Sensitivity for wake word detection, ranging from 0 (least sensitive) to 1 (most sensitive). Default is 0.5. - wake_word_activation_delay (float, default=0): Duration in seconds after the start of monitoring before the system switches to wake word activation if no voice is initially detected. If set to zero, the system uses wake word activation immediately. - wake_word_timeout (float, default=5): Duration in seconds after a wake word is recognized. If no subsequent voice activity is detected within this window, the system transitions back to an inactive state, awaiting the next wake word or voice activation. - on_wakeword_detected (callable, default=None): Callback function to be called when a wake word is detected. - on_wakeword_timeout (callable, default=None): Callback function to be called when the system goes back to an inactive state after when no speech was detected after wake word activation - on_wakeword_detection_start (callable, default=None): Callback function to be called when the system starts to listen for wake words - on_wakeword_detection_end (callable, default=None): Callback function to be called when the system stops to listen for wake words (e.g. because of timeout or wake word detected) - on_recorded_chunk (callable, default=None): Callback function to be called when a chunk of audio is recorded. The function is called with the recorded audio chunk as its argument. - debug_mode (bool, default=False): If set to True, the system will print additional debug information to the console. - handle_buffer_overflow (bool, default=True): If set to True, the system will log a warning when an input overflow occurs during recording and remove the data from the buffer. - beam_size (int, default=5): The beam size to use for beam search decoding. - beam_size_realtime (int, default=3): The beam size to use for beam search decoding in the real-time transcription model. - buffer_size (int, default=512): The buffer size to use for audio recording. Changing this may break functionality. - sample_rate (int, default=16000): The sample rate to use for audio recording. Changing this will very probably functionality (as the WebRTC VAD model is very sensitive towards the sample rate). - initial_prompt (str or iterable of int, default=None): Initial prompt to be fed to the transcription models. - suppress_tokens (list of int, default=[-1]): Tokens to be suppressed from the transcription output. Raises: Exception: Errors related to initializing transcription model, wake word detection, or audio recording. """ self.language = language self.compute_type = compute_type self.input_device_index = input_device_index self.gpu_device_index = gpu_device_index self.wake_words = wake_words self.wake_word_activation_delay = wake_word_activation_delay self.wake_word_timeout = wake_word_timeout self.ensure_sentence_starting_uppercase = ( ensure_sentence_starting_uppercase ) self.ensure_sentence_ends_with_period = ( ensure_sentence_ends_with_period ) self.use_microphone = mp.Value(c_bool, use_microphone) self.min_gap_between_recordings = min_gap_between_recordings self.min_length_of_recording = min_length_of_recording self.pre_recording_buffer_duration = pre_recording_buffer_duration self.post_speech_silence_duration = post_speech_silence_duration self.on_recording_start = on_recording_start self.on_recording_stop = on_recording_stop self.on_wakeword_detected = on_wakeword_detected self.on_wakeword_timeout = on_wakeword_timeout self.on_vad_detect_start = on_vad_detect_start self.on_vad_detect_stop = on_vad_detect_stop self.on_wakeword_detection_start = on_wakeword_detection_start self.on_wakeword_detection_end = on_wakeword_detection_end self.on_recorded_chunk = on_recorded_chunk self.on_transcription_start = on_transcription_start self.enable_realtime_transcription = enable_realtime_transcription self.realtime_model_type = realtime_model_type self.realtime_processing_pause = realtime_processing_pause self.on_realtime_transcription_update = ( on_realtime_transcription_update ) self.on_realtime_transcription_stabilized = ( on_realtime_transcription_stabilized ) self.debug_mode = debug_mode self.handle_buffer_overflow = handle_buffer_overflow self.beam_size = beam_size self.beam_size_realtime = beam_size_realtime self.allowed_latency_limit = ALLOWED_LATENCY_LIMIT self.level = level self.audio_queue = mp.Queue() self.buffer_size = buffer_size self.sample_rate = sample_rate self.recording_start_time = 0 self.recording_stop_time = 0 self.wake_word_detect_time = 0 self.silero_check_time = 0 self.silero_working = False self.speech_end_silence_start = 0 self.silero_sensitivity = silero_sensitivity self.listen_start = 0 self.spinner = spinner self.halo = None self.state = "inactive" self.wakeword_detected = False self.text_storage = [] self.realtime_stabilized_text = "" self.realtime_stabilized_safetext = "" self.is_webrtc_speech_active = False self.is_silero_speech_active = False self.recording_thread = None self.realtime_thread = None self.audio_interface = None self.audio = None self.stream = None self.start_recording_event = threading.Event() self.stop_recording_event = threading.Event() self.last_transcription_bytes = None self.initial_prompt = initial_prompt self.suppress_tokens = suppress_tokens # Initialize the logging configuration with the specified level log_format = 'RealTimeSTT: %(name)s - %(levelname)s - %(message)s' # Create a logger logger = logging.getLogger() logger.setLevel(level) # Set the root logger's level # Create a file handler and set its level file_handler = logging.FileHandler('realtimesst.log') file_handler.setLevel(logging.DEBUG) file_handler.setFormatter(logging.Formatter(log_format)) # Create a console handler and set its level console_handler = logging.StreamHandler() console_handler.setLevel(level) console_handler.setFormatter(logging.Formatter(log_format)) # Add the handlers to the logger logger.addHandler(file_handler) logger.addHandler(console_handler) self.is_shut_down = False self.shutdown_event = mp.Event() try: logging.debug("Explicitly setting the multiprocessing start method to 'spawn'") mp.set_start_method('spawn') except RuntimeError as e: logging.debug("Start method has already been set. Details:", e) logging.info("Starting RealTimeSTT") self.interrupt_stop_event = mp.Event() self.was_interrupted = mp.Event() self.main_transcription_ready_event = mp.Event() self.parent_transcription_pipe, child_transcription_pipe = mp.Pipe() self.transcript_process = mp.Process( target=AudioToTextRecorder._transcription_worker, args=( child_transcription_pipe, model, self.compute_type, self.gpu_device_index, self.main_transcription_ready_event, self.shutdown_event, self.interrupt_stop_event, self.beam_size, self.initial_prompt, self.suppress_tokens ) ) self.transcript_process.start() # Start audio data reading process if self.use_microphone.value: logging.info("Initializing audio recording" " (creating pyAudio input stream," f" sample rate: {self.sample_rate}" f" buffer size: {self.buffer_size}" ) self.reader_process = mp.Process( target=AudioToTextRecorder._audio_data_worker, args=( self.audio_queue, self.sample_rate, self.buffer_size, self.input_device_index, self.shutdown_event, self.interrupt_stop_event, self.use_microphone ) ) self.reader_process.start() # Initialize the realtime transcription model if self.enable_realtime_transcription: try: logging.info("Initializing faster_whisper realtime " f"transcription model {self.realtime_model_type}" ) self.realtime_model_type = faster_whisper.WhisperModel( model_size_or_path=self.realtime_model_type, device='cuda' if torch.cuda.is_available() else 'cpu', compute_type=self.compute_type, device_index=self.gpu_device_index ) except Exception as e: logging.exception("Error initializing faster_whisper " f"realtime transcription model: {e}" ) raise logging.debug("Faster_whisper realtime speech to text " "transcription model initialized successfully") # Setup wake word detection if wake_words: self.wake_words_list = [ word.strip() for word in wake_words.lower().split(',') ] sensitivity_list = [ float(wake_words_sensitivity) for _ in range(len(self.wake_words_list)) ] try: self.porcupine = pvporcupine.create( keywords=self.wake_words_list, sensitivities=sensitivity_list ) self.buffer_size = self.porcupine.frame_length self.sample_rate = self.porcupine.sample_rate except Exception as e: logging.exception("Error initializing porcupine " f"wake word detection engine: {e}" ) raise logging.debug("Porcupine wake word detection " "engine initialized successfully" ) # Setup voice activity detection model WebRTC try: logging.info("Initializing WebRTC voice with " f"Sensitivity {webrtc_sensitivity}" ) self.webrtc_vad_model = webrtcvad.Vad() self.webrtc_vad_model.set_mode(webrtc_sensitivity) except Exception as e: logging.exception("Error initializing WebRTC voice " f"activity detection engine: {e}" ) raise logging.debug("WebRTC VAD voice activity detection " "engine initialized successfully" ) # Setup voice activity detection model Silero VAD try: self.silero_vad_model, _ = torch.hub.load( repo_or_dir="snakers4/silero-vad", model="silero_vad", verbose=False, onnx=silero_use_onnx ) except Exception as e: logging.exception(f"Error initializing Silero VAD " f"voice activity detection engine: {e}" ) raise logging.debug("Silero VAD voice activity detection " "engine initialized successfully" ) self.audio_buffer = collections.deque( maxlen=int((self.sample_rate // self.buffer_size) * self.pre_recording_buffer_duration) ) self.frames = [] # Recording control flags self.is_recording = False self.is_running = True self.start_recording_on_voice_activity = False self.stop_recording_on_voice_deactivity = False # Start the recording worker thread self.recording_thread = threading.Thread(target=self._recording_worker) self.recording_thread.daemon = True self.recording_thread.start() # Start the realtime transcription worker thread self.realtime_thread = threading.Thread(target=self._realtime_worker) self.realtime_thread.daemon = True self.realtime_thread.start() # Wait for transcription models to start logging.debug('Waiting for main transcription model to start') self.main_transcription_ready_event.wait() logging.debug('Main transcription model ready') logging.debug('RealtimeSTT initialization completed successfully') @staticmethod def _transcription_worker(conn, model_path, compute_type, gpu_device_index, ready_event, shutdown_event, interrupt_stop_event, beam_size, initial_prompt, suppress_tokens ): """ Worker method that handles the continuous process of transcribing audio data. This method runs in a separate process and is responsible for: - Initializing the `faster_whisper` model used for transcription. - Receiving audio data sent through a pipe and using the model to transcribe it. - Sending transcription results back through the pipe. - Continuously checking for a shutdown event to gracefully terminate the transcription process. Args: conn (multiprocessing.Connection): The connection endpoint used for receiving audio data and sending transcription results. model_path (str): The path to the pre-trained faster_whisper model for transcription. compute_type (str): Specifies the type of computation to be used for transcription. gpu_device_index (int): Device ID to use. ready_event (threading.Event): An event that is set when the transcription model is successfully initialized and ready. shutdown_event (threading.Event): An event that, when set, signals this worker method to terminate. interrupt_stop_event (threading.Event): An event that, when set, signals this worker method to stop processing audio data. beam_size (int): The beam size to use for beam search decoding. initial_prompt (str or iterable of int): Initial prompt to be fed to the transcription model. suppress_tokens (list of int): Tokens to be suppressed from the transcription output. Raises: Exception: If there is an error while initializing the transcription model. """ logging.info("Initializing faster_whisper " f"main transcription model {model_path}" ) try: model = faster_whisper.WhisperModel( model_size_or_path=model_path, device='cuda' if torch.cuda.is_available() else 'cpu', compute_type=compute_type, device_index=gpu_device_index, ) except Exception as e: logging.exception("Error initializing main " f"faster_whisper transcription model: {e}" ) raise ready_event.set() logging.debug("Faster_whisper main speech to text " "transcription model initialized successfully" ) while not shutdown_event.is_set(): try: if conn.poll(0.5): audio, language = conn.recv() try: segments = model.transcribe( audio, language=language if language else None, beam_size=beam_size, initial_prompt=initial_prompt, suppress_tokens=suppress_tokens ) segments = segments[0] transcription = " ".join(seg.text for seg in segments) transcription = transcription.strip() conn.send(('success', transcription)) except Exception as e: logging.error(f"General transcription error: {e}") conn.send(('error', str(e))) else: # If there's no data, sleep / prevent busy waiting time.sleep(0.02) except KeyboardInterrupt: interrupt_stop_event.set() logging.debug("Transcription worker process " "finished due to KeyboardInterrupt" ) break @staticmethod def _audio_data_worker(audio_queue, sample_rate, buffer_size, input_device_index, shutdown_event, interrupt_stop_event, use_microphone): """ Worker method that handles the audio recording process. This method runs in a separate process and is responsible for: - Setting up the audio input stream for recording. - Continuously reading audio data from the input stream and placing it in a queue. - Handling errors during the recording process, including input overflow. - Gracefully terminating the recording process when a shutdown event is set. Args: audio_queue (queue.Queue): A queue where recorded audio data is placed. sample_rate (int): The sample rate of the audio input stream. buffer_size (int): The size of the buffer used in the audio input stream. input_device_index (int): The index of the audio input device shutdown_event (threading.Event): An event that, when set, signals this worker method to terminate. Raises: Exception: If there is an error while initializing the audio recording. """ try: audio_interface = pyaudio.PyAudio() stream = audio_interface.open( rate=sample_rate, format=pyaudio.paInt16, channels=1, input=True, frames_per_buffer=buffer_size, input_device_index=input_device_index, ) except Exception as e: logging.exception("Error initializing pyaudio " f"audio recording: {e}" ) raise logging.debug("Audio recording (pyAudio input " "stream) initialized successfully" ) try: while not shutdown_event.is_set(): try: data = stream.read(buffer_size) except OSError as e: if e.errno == pyaudio.paInputOverflowed: logging.warning("Input overflowed. Frame dropped.") else: logging.error(f"Error during recording: {e}") tb_str = traceback.format_exc() print(f"Traceback: {tb_str}") print(f"Error: {e}") continue except Exception as e: logging.error(f"Error during recording: {e}") tb_str = traceback.format_exc() print(f"Traceback: {tb_str}") print(f"Error: {e}") continue if use_microphone.value: audio_queue.put(data) except KeyboardInterrupt: interrupt_stop_event.set() logging.debug("Audio data worker process " "finished due to KeyboardInterrupt" ) finally: stream.stop_stream() stream.close() audio_interface.terminate() def wakeup(self): """ If in wake work modus, wake up as if a wake word was spoken. """ self.listen_start = time.time() def abort(self): self.start_recording_on_voice_activity = False self.stop_recording_on_voice_deactivity = False self._set_state("inactive") self.interrupt_stop_event.set() self.was_interrupted.wait() self.was_interrupted.clear() def wait_audio(self): """ Waits for the start and completion of the audio recording process. This method is responsible for: - Waiting for voice activity to begin recording if not yet started. - Waiting for voice inactivity to complete the recording. - Setting the audio buffer from the recorded frames. - Resetting recording-related attributes. Side effects: - Updates the state of the instance. - Modifies the audio attribute to contain the processed audio data. """ self.listen_start = time.time() # If not yet started recording, wait for voice activity to initiate. if not self.is_recording and not self.frames: self._set_state("listening") self.start_recording_on_voice_activity = True # Wait until recording starts while not self.interrupt_stop_event.is_set(): if self.start_recording_event.wait(timeout=0.02): break # If recording is ongoing, wait for voice inactivity # to finish recording. if self.is_recording: self.stop_recording_on_voice_deactivity = True # Wait until recording stops while not self.interrupt_stop_event.is_set(): if (self.stop_recording_event.wait(timeout=0.02)): break # Convert recorded frames to the appropriate audio format. audio_array = np.frombuffer(b''.join(self.frames), dtype=np.int16) self.audio = audio_array.astype(np.float32) / INT16_MAX_ABS_VALUE self.frames.clear() # Reset recording-related timestamps self.recording_stop_time = 0 self.listen_start = 0 self._set_state("inactive") def transcribe(self): """ Transcribes audio captured by this class instance using the `faster_whisper` model. Automatically starts recording upon voice activity if not manually started using `recorder.start()`. Automatically stops recording upon voice deactivity if not manually stopped with `recorder.stop()`. Processes the recorded audio to generate transcription. Args: on_transcription_finished (callable, optional): Callback function to be executed when transcription is ready. If provided, transcription will be performed asynchronously, and the callback will receive the transcription as its argument. If omitted, the transcription will be performed synchronously, and the result will be returned. Returns (if no callback is set): str: The transcription of the recorded audio. Raises: Exception: If there is an error during the transcription process. """ self._set_state("transcribing") audio_copy = copy.deepcopy(self.audio) self.parent_transcription_pipe.send((self.audio, self.language)) status, result = self.parent_transcription_pipe.recv() self._set_state("inactive") if status == 'success': self.last_transcription_bytes = audio_copy return self._preprocess_output(result) else: logging.error(result) raise Exception(result) def text(self, on_transcription_finished=None, ): """ Transcribes audio captured by this class instance using the `faster_whisper` model. - Automatically starts recording upon voice activity if not manually started using `recorder.start()`. - Automatically stops recording upon voice deactivity if not manually stopped with `recorder.stop()`. - Processes the recorded audio to generate transcription. Args: on_transcription_finished (callable, optional): Callback function to be executed when transcription is ready. If provided, transcription will be performed asynchronously, and the callback will receive the transcription as its argument. If omitted, the transcription will be performed synchronously, and the result will be returned. Returns (if not callback is set): str: The transcription of the recorded audio """ self.interrupt_stop_event.clear() self.was_interrupted.clear() self.wait_audio() if self.is_shut_down or self.interrupt_stop_event.is_set(): if self.interrupt_stop_event.is_set(): self.was_interrupted.set() return "" if on_transcription_finished: threading.Thread(target=on_transcription_finished, args=(self.transcribe(),)).start() else: return self.transcribe() def start(self): """ Starts recording audio directly without waiting for voice activity. """ # Ensure there's a minimum interval # between stopping and starting recording if (time.time() - self.recording_stop_time < self.min_gap_between_recordings): logging.info("Attempted to start recording " "too soon after stopping." ) return self logging.info("recording started") self._set_state("recording") self.text_storage = [] self.realtime_stabilized_text = "" self.realtime_stabilized_safetext = "" self.wakeword_detected = False self.wake_word_detect_time = 0 self.frames = [] self.is_recording = True self.recording_start_time = time.time() self.is_silero_speech_active = False self.is_webrtc_speech_active = False self.stop_recording_event.clear() self.start_recording_event.set() if self.on_recording_start: self.on_recording_start() return self def stop(self): """ Stops recording audio. """ # Ensure there's a minimum interval # between starting and stopping recording if (time.time() - self.recording_start_time < self.min_length_of_recording): logging.info("Attempted to stop recording " "too soon after starting." ) return self logging.info("recording stopped") self.is_recording = False self.recording_stop_time = time.time() self.is_silero_speech_active = False self.is_webrtc_speech_active = False self.silero_check_time = 0 self.start_recording_event.clear() self.stop_recording_event.set() if self.on_recording_stop: self.on_recording_stop() return self def feed_audio(self, chunk, original_sample_rate=16000): """ Feed an audio chunk into the processing pipeline. Chunks are accumulated until the buffer size is reached, and then the accumulated data is fed into the audio_queue. """ # Check if the buffer attribute exists, if not, initialize it if not hasattr(self, 'buffer'): self.buffer = bytearray() # Check if input is a NumPy array if isinstance(chunk, np.ndarray): # Handle stereo to mono conversion if necessary if chunk.ndim == 2: chunk = np.mean(chunk, axis=1) # Resample to 16000 Hz if necessary if original_sample_rate != 16000: num_samples = int(len(chunk) * 16000 / original_sample_rate) chunk = resample(chunk, num_samples) # Ensure data type is int16 chunk = chunk.astype(np.int16) # Convert the NumPy array to bytes chunk = chunk.tobytes() # Append the chunk to the buffer self.buffer += chunk buf_size = 2 * self.buffer_size # silero complains if too short # Check if the buffer has reached or exceeded the buffer_size while len(self.buffer) >= buf_size: # Extract self.buffer_size amount of data from the buffer to_process = self.buffer[:buf_size] self.buffer = self.buffer[buf_size:] # Feed the extracted data to the audio_queue self.audio_queue.put(to_process) def set_microphone(self, microphone_on=True): """ Set the microphone on or off. """ logging.info("Setting microphone to: " + str(microphone_on)) self.use_microphone.value = microphone_on def shutdown(self): """ Safely shuts down the audio recording by stopping the recording worker and closing the audio stream. """ # Force wait_audio() and text() to exit self.is_shut_down = True self.start_recording_event.set() self.stop_recording_event.set() self.shutdown_event.set() self.is_recording = False self.is_running = False logging.debug('Finishing recording thread') if self.recording_thread: self.recording_thread.join() logging.debug('Terminating reader process') # Give it some time to finish the loop and cleanup. if self.use_microphone: self.reader_process.join(timeout=10) if self.reader_process.is_alive(): logging.warning("Reader process did not terminate " "in time. Terminating forcefully." ) self.reader_process.terminate() logging.debug('Terminating transcription process') self.transcript_process.join(timeout=10) if self.transcript_process.is_alive(): logging.warning("Transcript process did not terminate " "in time. Terminating forcefully." ) self.transcript_process.terminate() self.parent_transcription_pipe.close() logging.debug('Finishing realtime thread') if self.realtime_thread: self.realtime_thread.join() if self.enable_realtime_transcription: if self.realtime_model_type: del self.realtime_model_type self.realtime_model_type = None gc.collect() def _recording_worker(self): """ The main worker method which constantly monitors the audio input for voice activity and accordingly starts/stops the recording. """ logging.debug('Starting recording worker') try: was_recording = False delay_was_passed = False # Continuously monitor audio for voice activity while self.is_running: try: data = self.audio_queue.get() if self.on_recorded_chunk: self.on_recorded_chunk(data) if self.handle_buffer_overflow: # Handle queue overflow if (self.audio_queue.qsize() > self.allowed_latency_limit): logging.warning("Audio queue size exceeds " "latency limit. Current size: " f"{self.audio_queue.qsize()}. " "Discarding old audio chunks." ) while (self.audio_queue.qsize() > self.allowed_latency_limit): data = self.audio_queue.get() except BrokenPipeError: print("BrokenPipeError _recording_worker") self.is_running = False break if not self.is_recording: # Handle not recording state time_since_listen_start = (time.time() - self.listen_start if self.listen_start else 0) wake_word_activation_delay_passed = ( time_since_listen_start > self.wake_word_activation_delay ) # Handle wake-word timeout callback if wake_word_activation_delay_passed \ and not delay_was_passed: if self.wake_words and self.wake_word_activation_delay: if self.on_wakeword_timeout: self.on_wakeword_timeout() delay_was_passed = wake_word_activation_delay_passed # Set state and spinner text if not self.recording_stop_time: if self.wake_words \ and wake_word_activation_delay_passed \ and not self.wakeword_detected: self._set_state("wakeword") else: if self.listen_start: self._set_state("listening") else: self._set_state("inactive") # Detect wake words if applicable if self.wake_words and wake_word_activation_delay_passed: try: pcm = struct.unpack_from( "h" * self.buffer_size, data ) wakeword_index = self.porcupine.process(pcm) except struct.error: logging.error("Error unpacking audio data " "for wake word processing.") continue except Exception as e: logging.error(f"Wake word processing error: {e}") continue # If a wake word is detected if wakeword_index >= 0: # Removing the wake word from the recording samples_for_0_1_sec = int(self.sample_rate * 0.1) start_index = max( 0, len(self.audio_buffer) - samples_for_0_1_sec ) temp_samples = collections.deque( itertools.islice( self.audio_buffer, start_index, None) ) self.audio_buffer.clear() self.audio_buffer.extend(temp_samples) self.wake_word_detect_time = time.time() self.wakeword_detected = True if self.on_wakeword_detected: self.on_wakeword_detected() # Check for voice activity to # trigger the start of recording if ((not self.wake_words or not wake_word_activation_delay_passed) and self.start_recording_on_voice_activity) \ or self.wakeword_detected: if self._is_voice_active(): logging.info("voice activity detected") self.start() if self.is_recording: self.start_recording_on_voice_activity = False # Add the buffered audio # to the recording frames self.frames.extend(list(self.audio_buffer)) self.audio_buffer.clear() self.silero_vad_model.reset_states() else: data_copy = data[:] self._check_voice_activity(data_copy) self.speech_end_silence_start = 0 else: # If we are currently recording # Stop the recording if silence is detected after speech if self.stop_recording_on_voice_deactivity: if not self._is_webrtc_speech(data, True): # Voice deactivity was detected, so we start # measuring silence time before stopping recording if self.speech_end_silence_start == 0: self.speech_end_silence_start = time.time() else: self.speech_end_silence_start = 0 # Wait for silence to stop recording after speech if self.speech_end_silence_start and time.time() - \ self.speech_end_silence_start > \ self.post_speech_silence_duration: logging.info("voice deactivity detected") self.stop() if not self.is_recording and was_recording: # Reset after stopping recording to ensure clean state self.stop_recording_on_voice_deactivity = False if time.time() - self.silero_check_time > 0.1: self.silero_check_time = 0 # Handle wake word timeout (waited to long initiating # speech after wake word detection) if self.wake_word_detect_time and time.time() - \ self.wake_word_detect_time > self.wake_word_timeout: self.wake_word_detect_time = 0 if self.wakeword_detected and self.on_wakeword_timeout: self.on_wakeword_timeout() self.wakeword_detected = False was_recording = self.is_recording if self.is_recording: self.frames.append(data) if not self.is_recording or self.speech_end_silence_start: self.audio_buffer.append(data) except Exception as e: if not self.interrupt_stop_event.is_set(): logging.error(f"Unhandled exeption in _recording_worker: {e}") raise def _realtime_worker(self): """ Performs real-time transcription if the feature is enabled. The method is responsible transcribing recorded audio frames in real-time based on the specified resolution interval. The transcribed text is stored in `self.realtime_transcription_text` and a callback function is invoked with this text if specified. """ try: logging.debug('Starting realtime worker') # Return immediately if real-time transcription is not enabled if not self.enable_realtime_transcription: return # Continue running as long as the main process is active while self.is_running: # Check if the recording is active if self.is_recording: # Sleep for the duration of the transcription resolution time.sleep(self.realtime_processing_pause) # Convert the buffer frames to a NumPy array audio_array = np.frombuffer( b''.join(self.frames), dtype=np.int16 ) # Normalize the array to a [-1, 1] range audio_array = audio_array.astype(np.float32) / \ INT16_MAX_ABS_VALUE # Perform transcription and assemble the text segments = self.realtime_model_type.transcribe( audio_array, language=self.language if self.language else None, beam_size=self.beam_size_realtime, initial_prompt=self.initial_prompt, suppress_tokens=self.suppress_tokens, ) # double check recording state # because it could have changed mid-transcription if self.is_recording and time.time() - \ self.recording_start_time > 0.5: logging.debug('Starting realtime transcription') self.realtime_transcription_text = " ".join( seg.text for seg in segments[0] ) self.realtime_transcription_text = \ self.realtime_transcription_text.strip() self.text_storage.append( self.realtime_transcription_text ) # Take the last two texts in storage, if they exist if len(self.text_storage) >= 2: last_two_texts = self.text_storage[-2:] # Find the longest common prefix # between the two texts prefix = os.path.commonprefix( [last_two_texts[0], last_two_texts[1]] ) # This prefix is the text that was transcripted # two times in the same way # Store as "safely detected text" if len(prefix) >= \ len(self.realtime_stabilized_safetext): # Only store when longer than the previous # as additional security self.realtime_stabilized_safetext = prefix # Find parts of the stabilized text # in the freshly transcripted text matching_pos = self._find_tail_match_in_text( self.realtime_stabilized_safetext, self.realtime_transcription_text ) if matching_pos < 0: if self.realtime_stabilized_safetext: self._on_realtime_transcription_stabilized( self._preprocess_output( self.realtime_stabilized_safetext, True ) ) else: self._on_realtime_transcription_stabilized( self._preprocess_output( self.realtime_transcription_text, True ) ) else: # We found parts of the stabilized text # in the transcripted text # We now take the stabilized text # and add only the freshly transcripted part to it output_text = self.realtime_stabilized_safetext + \ self.realtime_transcription_text[matching_pos:] # This yields us the "left" text part as stabilized # AND at the same time delivers fresh detected # parts on the first run without the need for # two transcriptions self._on_realtime_transcription_stabilized( self._preprocess_output(output_text, True) ) # Invoke the callback with the transcribed text self._on_realtime_transcription_update( self._preprocess_output( self.realtime_transcription_text, True ) ) # If not recording, sleep briefly before checking again else: time.sleep(TIME_SLEEP) except Exception as e: logging.error(f"Unhandled exeption in _realtime_worker: {e}") raise def _is_silero_speech(self, chunk): """ Returns true if speech is detected in the provided audio data Args: data (bytes): raw bytes of audio data (1024 raw bytes with 16000 sample rate and 16 bits per sample) """ if self.sample_rate != 16000: pcm_data = np.frombuffer(chunk, dtype=np.int16) data_16000 = signal.resample_poly( pcm_data, 16000, self.sample_rate) chunk = data_16000.astype(np.int16).tobytes() self.silero_working = True audio_chunk = np.frombuffer(chunk, dtype=np.int16) audio_chunk = audio_chunk.astype(np.float32) / INT16_MAX_ABS_VALUE vad_prob = self.silero_vad_model( torch.from_numpy(audio_chunk), SAMPLE_RATE).item() is_silero_speech_active = vad_prob > (1 - self.silero_sensitivity) if is_silero_speech_active: self.is_silero_speech_active = True self.silero_working = False return is_silero_speech_active def _is_webrtc_speech(self, chunk, all_frames_must_be_true=False): """ Returns true if speech is detected in the provided audio data Args: data (bytes): raw bytes of audio data (1024 raw bytes with 16000 sample rate and 16 bits per sample) """ if self.sample_rate != 16000: pcm_data = np.frombuffer(chunk, dtype=np.int16) data_16000 = signal.resample_poly( pcm_data, 16000, self.sample_rate) chunk = data_16000.astype(np.int16).tobytes() # Number of audio frames per millisecond frame_length = int(16000 * 0.01) # for 10ms frame num_frames = int(len(chunk) / (2 * frame_length)) speech_frames = 0 for i in range(num_frames): start_byte = i * frame_length * 2 end_byte = start_byte + frame_length * 2 frame = chunk[start_byte:end_byte] if self.webrtc_vad_model.is_speech(frame, 16000): speech_frames += 1 if not all_frames_must_be_true: if self.debug_mode: print(f"Speech detected in frame {i + 1}" f" of {num_frames}") return True if all_frames_must_be_true: if self.debug_mode and speech_frames == num_frames: print(f"Speech detected in {speech_frames} of " f"{num_frames} frames") elif self.debug_mode: print(f"Speech not detected in all {num_frames} frames") return speech_frames == num_frames else: if self.debug_mode: print(f"Speech not detected in any of {num_frames} frames") return False def _check_voice_activity(self, data): """ Initiate check if voice is active based on the provided data. Args: data: The audio data to be checked for voice activity. """ self.is_webrtc_speech_active = self._is_webrtc_speech(data) # First quick performing check for voice activity using WebRTC if self.is_webrtc_speech_active: if not self.silero_working: self.silero_working = True # Run the intensive check in a separate thread threading.Thread( target=self._is_silero_speech, args=(data,)).start() def _is_voice_active(self): """ Determine if voice is active. Returns: bool: True if voice is active, False otherwise. """ return self.is_webrtc_speech_active and self.is_silero_speech_active def _set_state(self, new_state): """ Update the current state of the recorder and execute corresponding state-change callbacks. Args: new_state (str): The new state to set. """ # Check if the state has actually changed if new_state == self.state: return # Store the current state for later comparison old_state = self.state # Update to the new state self.state = new_state # Execute callbacks based on transitioning FROM a particular state if old_state == "listening": if self.on_vad_detect_stop: self.on_vad_detect_stop() elif old_state == "wakeword": if self.on_wakeword_detection_end: self.on_wakeword_detection_end() # Execute callbacks based on transitioning TO a particular state if new_state == "listening": if self.on_vad_detect_start: self.on_vad_detect_start() self._set_spinner("speak now") if self.spinner and self.halo: self.halo._interval = 250 elif new_state == "wakeword": if self.on_wakeword_detection_start: self.on_wakeword_detection_start() self._set_spinner(f"say {self.wake_words}") if self.spinner and self.halo: self.halo._interval = 500 elif new_state == "transcribing": if self.on_transcription_start: self.on_transcription_start() self._set_spinner("transcribing") if self.spinner and self.halo: self.halo._interval = 50 elif new_state == "recording": self._set_spinner("recording") if self.spinner and self.halo: self.halo._interval = 100 elif new_state == "inactive": if self.spinner and self.halo: self.halo.stop() self.halo = None def _set_spinner(self, text): """ Update the spinner's text or create a new spinner with the provided text. Args: text (str): The text to be displayed alongside the spinner. """ if self.spinner: # If the Halo spinner doesn't exist, create and start it if self.halo is None: self.halo = halo.Halo(text=text) self.halo.start() # If the Halo spinner already exists, just update the text else: self.halo.text = text def _preprocess_output(self, text, preview=False): """ Preprocesses the output text by removing any leading or trailing whitespace, converting all whitespace sequences to a single space character, and capitalizing the first character of the text. Args: text (str): The text to be preprocessed. Returns: str: The preprocessed text. """ text = re.sub(r'\s+', ' ', text.strip()) if self.ensure_sentence_starting_uppercase: if text: text = text[0].upper() + text[1:] # Ensure the text ends with a proper punctuation # if it ends with an alphanumeric character if not preview: if self.ensure_sentence_ends_with_period: if text and text[-1].isalnum(): text += '.' return text def _find_tail_match_in_text(self, text1, text2, length_of_match=10): """ Find the position where the last 'n' characters of text1 match with a substring in text2. This method takes two texts, extracts the last 'n' characters from text1 (where 'n' is determined by the variable 'length_of_match'), and searches for an occurrence of this substring in text2, starting from the end of text2 and moving towards the beginning. Parameters: - text1 (str): The text containing the substring that we want to find in text2. - text2 (str): The text in which we want to find the matching substring. - length_of_match(int): The length of the matching string that we are looking for Returns: int: The position (0-based index) in text2 where the matching substring starts. If no match is found or either of the texts is too short, returns -1. """ # Check if either of the texts is too short if len(text1) < length_of_match or len(text2) < length_of_match: return -1 # The end portion of the first text that we want to compare target_substring = text1[-length_of_match:] # Loop through text2 from right to left for i in range(len(text2) - length_of_match + 1): # Extract the substring from text2 # to compare with the target_substring current_substring = text2[len(text2) - i - length_of_match: len(text2) - i] # Compare the current_substring with the target_substring if current_substring == target_substring: # Position in text2 where the match starts return len(text2) - i return -1 def _on_realtime_transcription_stabilized(self, text): """ Callback method invoked when the real-time transcription stabilizes. This method is called internally when the transcription text is considered "stable" meaning it's less likely to change significantly with additional audio input. It notifies any registered external listener about the stabilized text if recording is still ongoing. This is particularly useful for applications that need to display live transcription results to users and want to highlight parts of the transcription that are less likely to change. Args: text (str): The stabilized transcription text. """ if self.on_realtime_transcription_stabilized: if self.is_recording: self.on_realtime_transcription_stabilized(text) def _on_realtime_transcription_update(self, text): """ Callback method invoked when there's an update in the real-time transcription. This method is called internally whenever there's a change in the transcription text, notifying any registered external listener about the update if recording is still ongoing. This provides a mechanism for applications to receive and possibly display live transcription updates, which could be partial and still subject to change. Args: text (str): The updated transcription text. """ if self.on_realtime_transcription_update: if self.is_recording: self.on_realtime_transcription_update(text) def __enter__(self): """ Method to setup the context manager protocol. This enables the instance to be used in a `with` statement, ensuring proper resource management. When the `with` block is entered, this method is automatically called. Returns: self: The current instance of the class. """ return self def __exit__(self, exc_type, exc_value, traceback): """ Method to define behavior when the context manager protocol exits. This is called when exiting the `with` block and ensures that any necessary cleanup or resource release processes are executed, such as shutting down the system properly. Args: exc_type (Exception or None): The type of the exception that caused the context to be exited, if any. exc_value (Exception or None): The exception instance that caused the context to be exited, if any. traceback (Traceback or None): The traceback corresponding to the exception, if any. """ self.shutdown()