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import json
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import logging
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import threading
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import numpy as np
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class ServeClientBase(object):
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RATE = 16000
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SERVER_READY = "SERVER_READY"
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DISCONNECT = "DISCONNECT"
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def __init__(self, client_uid, websocket):
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self.client_uid = client_uid
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self.websocket = websocket
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self.frames = b""
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self.timestamp_offset = 0.0
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self.frames_np = None
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self.frames_offset = 0.0
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self.text = []
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self.current_out = ''
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self.prev_out = ''
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self.t_start = None
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self.exit = False
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self.same_output_count = 0
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self.show_prev_out_thresh = 5 # if pause(no output from whisper) show previous output for 5 seconds
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self.add_pause_thresh = 3 # add a blank to segment list as a pause(no speech) for 3 seconds
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self.transcript = []
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self.send_last_n_segments = 10
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# text formatting
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self.pick_previous_segments = 2
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# threading
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self.lock = threading.Lock()
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def speech_to_text(self):
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raise NotImplementedError
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def transcribe_audio(self):
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raise NotImplementedError
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def handle_transcription_output(self):
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raise NotImplementedError
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def add_frames(self, frame_np):
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"""
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Add audio frames to the ongoing audio stream buffer.
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This method is responsible for maintaining the audio stream buffer, allowing the continuous addition
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of audio frames as they are received. It also ensures that the buffer does not exceed a specified size
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to prevent excessive memory usage.
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If the buffer size exceeds a threshold (45 seconds of audio data), it discards the oldest 30 seconds
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of audio data to maintain a reasonable buffer size. If the buffer is empty, it initializes it with the provided
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audio frame. The audio stream buffer is used for real-time processing of audio data for transcription.
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Args:
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frame_np (numpy.ndarray): The audio frame data as a NumPy array.
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"""
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self.lock.acquire()
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if self.frames_np is not None and self.frames_np.shape[0] > 45*self.RATE:
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self.frames_offset += 30.0
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self.frames_np = self.frames_np[int(30*self.RATE):]
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# check timestamp offset(should be >= self.frame_offset)
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# this basically means that there is no speech as timestamp offset hasnt updated
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# and is less than frame_offset
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if self.timestamp_offset < self.frames_offset:
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self.timestamp_offset = self.frames_offset
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if self.frames_np is None:
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self.frames_np = frame_np.copy()
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else:
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self.frames_np = np.concatenate((self.frames_np, frame_np), axis=0)
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self.lock.release()
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def clip_audio_if_no_valid_segment(self):
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"""
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Update the timestamp offset based on audio buffer status.
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Clip audio if the current chunk exceeds 30 seconds, this basically implies that
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no valid segment for the last 30 seconds from whisper
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"""
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with self.lock:
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if self.frames_np[int((self.timestamp_offset - self.frames_offset)*self.RATE):].shape[0] > 25 * self.RATE:
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duration = self.frames_np.shape[0] / self.RATE
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self.timestamp_offset = self.frames_offset + duration - 5
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def get_audio_chunk_for_processing(self):
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"""
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Retrieves the next chunk of audio data for processing based on the current offsets.
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Calculates which part of the audio data should be processed next, based on
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the difference between the current timestamp offset and the frame's offset, scaled by
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the audio sample rate (RATE). It then returns this chunk of audio data along with its
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duration in seconds.
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Returns:
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tuple: A tuple containing:
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- input_bytes (np.ndarray): The next chunk of audio data to be processed.
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- duration (float): The duration of the audio chunk in seconds.
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"""
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with self.lock:
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samples_take = max(0, (self.timestamp_offset - self.frames_offset) * self.RATE)
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input_bytes = self.frames_np[int(samples_take):].copy()
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duration = input_bytes.shape[0] / self.RATE
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return input_bytes, duration
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def prepare_segments(self, last_segment=None):
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"""
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Prepares the segments of transcribed text to be sent to the client.
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This method compiles the recent segments of transcribed text, ensuring that only the
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specified number of the most recent segments are included. It also appends the most
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recent segment of text if provided (which is considered incomplete because of the possibility
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of the last word being truncated in the audio chunk).
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Args:
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last_segment (str, optional): The most recent segment of transcribed text to be added
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to the list of segments. Defaults to None.
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Returns:
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list: A list of transcribed text segments to be sent to the client.
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"""
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segments = []
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if len(self.transcript) >= self.send_last_n_segments:
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segments = self.transcript[-self.send_last_n_segments:].copy()
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else:
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segments = self.transcript.copy()
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if last_segment is not None:
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segments = segments + [last_segment]
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return segments
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def get_audio_chunk_duration(self, input_bytes):
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"""
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Calculates the duration of the provided audio chunk.
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Args:
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input_bytes (numpy.ndarray): The audio chunk for which to calculate the duration.
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Returns:
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float: The duration of the audio chunk in seconds.
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"""
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return input_bytes.shape[0] / self.RATE
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def send_transcription_to_client(self, segments):
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"""
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Sends the specified transcription segments to the client over the websocket connection.
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This method formats the transcription segments into a JSON object and attempts to send
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this object to the client. If an error occurs during the send operation, it logs the error.
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Returns:
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segments (list): A list of transcription segments to be sent to the client.
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"""
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try:
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self.websocket.send(
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json.dumps({
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"uid": self.client_uid,
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"segments": segments,
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})
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)
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except Exception as e:
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logging.error(f"[ERROR]: Sending data to client: {e}")
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def disconnect(self):
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"""
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Notify the client of disconnection and send a disconnect message.
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This method sends a disconnect message to the client via the WebSocket connection to notify them
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that the transcription service is disconnecting gracefully.
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"""
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self.websocket.send(json.dumps({
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"uid": self.client_uid,
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"message": self.DISCONNECT
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}))
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def cleanup(self):
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"""
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Perform cleanup tasks before exiting the transcription service.
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This method performs necessary cleanup tasks, including stopping the transcription thread, marking
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the exit flag to indicate the transcription thread should exit gracefully, and destroying resources
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associated with the transcription process.
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"""
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logging.info("Cleaning up.")
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self.exit = True
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Reference in New Issue
Block a user