import os import argparse import wave import numpy as np import scipy import ffmpeg import pyaudio import threading import textwrap import json import websocket import uuid import time def resample(file: str, sr: int = 16000): """ # https://github.com/openai/whisper/blob/7858aa9c08d98f75575035ecd6481f462d66ca27/whisper/audio.py#L22 Open an audio file and read as mono waveform, resampling as necessary, save the resampled audio Parameters ---------- file: str The audio file to open sr: int The sample rate to resample the audio if necessary """ try: # This launches a subprocess to decode audio while down-mixing and resampling as necessary. # Requires the ffmpeg CLI and `ffmpeg-python` package to be installed. out, _ = ( ffmpeg.input(file, threads=0) .output("-", format="s16le", acodec="pcm_s16le", ac=1, ar=sr) .run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True) ) except ffmpeg.Error as e: raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e np_buffer = np.frombuffer(out, dtype=np.int16) resampled_file = f"{file.split('.')[0]}_resampled.wav" scipy.io.wavfile.write(resampled_file, sr, np_buffer.astype(np.int16)) return resampled_file class Client: """ Represents a client for audio recording and streaming to a server using WebSocket communication. This class allows audio recording from the microphone or playing audio from a file while streaming it to a server for transcription or translation. It uses PyAudio for audio recording and playback and WebSocket for communication with the server. Attributes: CHUNK (int): The size of audio chunks for recording and playback. FORMAT: The audio format used by PyAudio (paInt16 for 16-bit PCM). CHANNELS (int): The number of audio channels (1 for mono). RATE (int): The audio sampling rate in Hz (samples per second). RECORD_SECONDS (int): The maximum duration for audio recording in seconds. RECORDING (bool): Indicates whether recording is currently active. multilingual (bool): Indicates if multilingual transcription is enabled. language (str): The selected language for transcription. task (str): The transcription or translation task to be performed. uid (str): A unique identifier for the client. WAITING (bool): Indicates if the client is waiting for server availability. LAST_RESPONSE_RECIEVED (float): Timestamp of the last response received from the server. DISCONNECT_IF_NO_RESPONSE_FOR (int): Maximum time without server response before disconnection. """ CHUNK = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 16000 RECORD_SECONDS = 60000 RECORDING = False multilingual = False language = None task = "transcribe" uid = str(uuid.uuid4()) WAITING = False LAST_RESPONSE_RECIEVED = None DISCONNECT_IF_NO_RESPONSE_FOR = 15 def __init__(self, host=None, port=None, is_multilingual=False, lang=None, translate=False): """ Initializes a Client instance for audio recording and streaming to a server. If host and port are not provided, the WebSocket connection will not be established. When translate is True, the task will be set to "translate" instead of "transcribe". he audio recording starts immediately upon initialization. Args: host (str): The hostname or IP address of the server. port (int): The port number for the WebSocket server. is_multilingual (bool, optional): Specifies if multilingual transcription is enabled. Default is False. lang (str, optional): The selected language for transcription when multilingual is disabled. Default is None. translate (bool, optional): Specifies if the task is translation. Default is False. Attributes: timestamp_offset (float): A timestamp offset for tracking audio timing. audio_bytes (bytes): A buffer for storing audio data. p (pyaudio.PyAudio): An instance of PyAudio for audio streaming. stream (pyaudio.Stream): The audio stream for recording. client_socket (websocket.WebSocketApp): The WebSocket client for server communication. ws_thread (threading.Thread): A thread for running the WebSocket client. frames (bytes): A buffer for accumulating audio frames. """ Client.multilingual = is_multilingual Client.language = lang if is_multilingual else "en" if translate: Client.task = "translate" self.timestamp_offset = 0.0 self.audio_bytes = None self.p = pyaudio.PyAudio() self.stream = self.p.open(format=self.FORMAT, channels=self.CHANNELS, rate=self.RATE, input=True, frames_per_buffer=self.CHUNK) if host is not None and port is not None: socket_url = f"ws://{host}:{port}" self.client_socket = websocket.WebSocketApp(socket_url, on_open=Client.on_open, on_message=Client.on_message, on_error=Client.on_error, on_close=Client.on_close) else: print("[ERROR]: No host or port specified.") return self.ws_thread = threading.Thread(target=self.client_socket.run_forever) self.ws_thread.setDaemon(True) self.ws_thread.start() self.frames = b"" print("[INFO]: * recording") @staticmethod def on_message(ws, message): """ Callback function called when a message is received from the server. It updates various attributes of the client based on the received message, including recording status, language detection, and server messages. If a disconnect message is received, it sets the recording status to False. Args: ws (websocket.WebSocketApp): The WebSocket client instance. message (str): The received message from the server. """ Client.LAST_RESPONSE_RECIEVED = time.time() message = json.loads(message) if message.get('uid')!=Client.uid: print("[ERROR]: invalid client uid") return if "status" in message.keys() and message["status"] == "WAIT": Client.WAITING = True print(f"[INFO]:Server is full. Estimated wait time {round(message['message'])} minutes.") if "message" in message.keys() and message["message"] == "DISCONNECT": print("[INFO]: Server overtime disconnected.") Client.RECORDING = False if "message" in message.keys() and message["message"] == "SERVER_READY": Client.RECORDING = True return if "language" in message.keys(): Client.language = message.get("language") lang_prob = message.get("language_prob") print(f"[INFO]: Server detected language {Client.language} with probability {lang_prob}") return if "segments" not in message.keys(): return message = message["segments"] text = [] if len(message): for seg in message: if len(text): if text[-1] != seg["text"]: text.append(seg["text"]) else: text.append(seg["text"]) if len(text) > 3: text = text[-3:] wrapper = textwrap.TextWrapper(width=60) word_list = wrapper.wrap(text="".join(text)) # Print each line. if os.name=='nt': os.system('cls') else: os.system('clear') for element in word_list: print(element) @staticmethod def on_error(ws, error): print(error) @staticmethod def on_close(ws, close_status_code, close_msg): print(f"[INFO]: Websocket connection closed.") @staticmethod def on_open(ws): """ Callback function called when the WebSocket connection is successfully opened. Sends an initial configuration message to the server, including client UID, multilingual mode, language selection, and task type. Args: ws (websocket.WebSocketApp): The WebSocket client instance. """ print(Client.multilingual, Client.language, Client.task) print("[INFO]: Opened connection") ws.send(json.dumps({ 'uid': Client.uid, 'multilingual': Client.multilingual, 'language': Client.language, 'task': Client.task })) @staticmethod def bytes_to_float_array(audio_bytes): """ Convert audio data from bytes to a NumPy float array. It assumes that the audio data is in 16-bit PCM format. The audio data is normalized to have values between -1 and 1. Args: audio_bytes (bytes): Audio data in bytes. Returns: np.ndarray: A NumPy array containing the audio data as float values normalized between -1 and 1. """ raw_data = np.frombuffer( buffer=audio_bytes, dtype=np.int16 ) return raw_data.astype(np.float32) / 32768.0 def send_packet_to_server(self, message): """ Send an audio packet to the server using WebSocket. Args: message (bytes): The audio data packet in bytes to be sent to the server. """ try: self.client_socket.send(message, websocket.ABNF.OPCODE_BINARY) except Exception as e: print(e) def play_file(self, filename): """ Play an audio file and send it to the server for processing. Reads an audio file, plays it through the audio output, and simultaneously sends the audio data to the server for processing. It uses PyAudio to create an audio stream for playback. The audio data is read from the file in chunks, converted to floating-point format, and sent to the server using WebSocket communication. This method is typically used when you want to process pre-recorded audio and send it to the server in real-time. Args: filename (str): The path to the audio file to be played and sent to the server. """ self.wf = wave.open(filename, 'rb') self.stream = self.p.open(format=self.p.get_format_from_width(self.wf.getsampwidth()), channels=self.wf.getnchannels(), rate=self.wf.getframerate(), input=True, output=True, frames_per_buffer=self.CHUNK) try: while Client.RECORDING: data = self.wf.readframes(self.CHUNK) if data==b'': break audio_array = Client.bytes_to_float_array(data) self.send_packet_to_server(audio_array.tobytes()) self.stream.write(data) self.wf.close() elapsed_time = time.time() - self.LAST_RESPONSE_RECIEVED while elapsed_time < self.DISCONNECT_IF_NO_RESPONSE_FOR: continue self.stream.close() self.close_websocket() except KeyboardInterrupt: self.wf.close() self.stream.stop_stream() self.stream.close() self.p.terminate() self.close_websocket() print("[INFO]: Keyboard interrupt.") def close_websocket(self): """ Close the WebSocket connection and join the WebSocket thread. First attempts to close the WebSocket connection using `self.client_socket.close()`. After closing the connection, it joins the WebSocket thread to ensure proper termination. """ try: self.client_socket.close() except Exception as e: print("[ERROR]: Error closing WebSocket:", e) try: self.ws_thread.join() except Exception as e: print("[ERROR:] Error joining WebSocket thread:", e) def get_client_socket(self): """ Get the WebSocket client socket instance. Returns: WebSocketApp: The WebSocket client socket instance currently in use by the client. """ return self.client_socket def write_audio_frames_to_file(self, frames, file_name): """ Write audio frames to a WAV file. The WAV file is created or overwritten with the specified name. The audio frames should be in the correct format and match the specified channel, sample width, and sample rate. Args: frames (bytes): The audio frames to be written to the file. file_name (str): The name of the WAV file to which the frames will be written. """ wf = wave.open(file_name, 'wb') wf.setnchannels(self.CHANNELS) wf.setsampwidth(2) wf.setframerate(self.RATE) wf.writeframes(frames) wf.close() def record(self, out_file="output_recording.wav"): """ Record audio data from the input stream and save it to a WAV file. Continuously records audio data from the input stream, sends it to the server via a WebSocket connection, and simultaneously saves it to multiple WAV files in chunks. It stops recording when the `RECORD_SECONDS` duration is reached or when the `RECORDING` flag is set to `False`. Audio data is saved in chunks to the "chunks" directory. Each chunk is saved as a separate WAV file. The recording will continue until the specified duration is reached or until the `RECORDING` flag is set to `False`. The recording process can be interrupted by sending a KeyboardInterrupt (e.g., pressing Ctrl+C). After recording, the method combines all the saved audio chunks into the specified `out_file`. Args: out_file (str, optional): The name of the output WAV file to save the entire recording. Default is "output_recording.wav". """ n_audio_file = 0 if not os.path.exists("chunks"): os.makedirs("chunks", exist_ok=True) try: for _ in range(0, int(self.RATE / self.CHUNK * self.RECORD_SECONDS)): if not Client.RECORDING: break data = self.stream.read(self.CHUNK) self.frames += data audio_array = Client.bytes_to_float_array(data) self.send_packet_to_server(audio_array.tobytes()) # save frames if more than a minute if len(self.frames) > 60*self.RATE: t = threading.Thread( target=self.write_audio_frames_to_file, args=(self.frames[:], f"chunks/{n_audio_file}.wav", ) ) t.start() n_audio_file += 1 self.frames = b"" except KeyboardInterrupt: if len(self.frames): self.write_audio_frames_to_file( self.frames[:], f"chunks/{n_audio_file}.wav") n_audio_file += 1 self.stream.stop_stream() self.stream.close() self.p.terminate() self.close_websocket() self.write_output_recording(n_audio_file, out_file) def write_output_recording(self, n_audio_file, out_file): """ Combine and save recorded audio chunks into a single WAV file. The individual audio chunk files are expected to be located in the "chunks" directory. Reads each chunk file, appends its audio data to the final recording, and then deletes the chunk file. After combining and saving, the final recording is stored in the specified `out_file`. Args: n_audio_file (int): The number of audio chunk files to combine. out_file (str): The name of the output WAV file to save the final recording. """ input_files = [f"chunks/{i}.wav" for i in range(n_audio_file) if os.path.exists(f"chunks/{i}.wav")] wf = wave.open(out_file, 'wb') wf.setnchannels(self.CHANNELS) wf.setsampwidth(2) wf.setframerate(self.RATE) for in_file in input_files: w = wave.open(in_file, 'rb') while True: data = w.readframes(self.CHUNK) if data==b'': break wf.writeframes(data) w.close() # remove this file os.remove(in_file) wf.close() class TranscriptionClient: """ Client for handling audio transcription tasks via a WebSocket connection. Acts as a high-level client for audio transcription tasks using a WebSocket connection. It can be used to send audio data for transcription to a server and receive transcribed text segments. Args: host (str): The hostname or IP address of the server. port (int): The port number to connect to on the server. is_multilingual (bool, optional): Indicates whether the transcription should support multiple languages (default is False). lang (str, optional): The primary language for transcription (used if `is_multilingual` is False). Default is None, which defaults to English ('en'). translate (bool, optional): Indicates whether translation tasks are required (default is False). Attributes: client (Client): An instance of the underlying Client class responsible for handling the WebSocket connection. Example: To create a TranscriptionClient and start transcription on microphone audio: ```python transcription_client = TranscriptionClient(host="localhost", port=9090, is_multilingual=True) transcription_client() ``` """ def __init__(self, host, port, is_multilingual=False, lang=None, translate=False): self.client = Client(host, port, is_multilingual, lang, translate) def __call__(self, audio=None): """ Start the transcription process. Initiates the transcription process by connecting to the server via a WebSocket. It waits for the server to be ready to receive audio data and then sends audio for transcription. If an audio file is provided, it will be played and streamed to the server; otherwise, it will perform live recording. Args: audio (str, optional): Path to an audio file for transcription. Default is None, which triggers live recording. """ print("[INFO]: Waiting for server ready ...") while not Client.RECORDING: if Client.WAITING: self.client.close_websocket() return pass print("[INFO]: Server Ready!") if audio is not None: resampled_file = resample(audio) self.client.play_file(resampled_file) else: self.client.record()