-
-

Welcome to whisper_live’s documentation!

+
+

Welcome to Whisper Live documentation!

@@ -219,9 +219,201 @@ port (int): The port number to bind the server.

+
+
+class whisper_live.client.Client(host=None, port=None, is_multilingual=False, lang=None, translate=False)
+

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.

+
+
+
+
+static 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.

+
+
+
+ +
+
+close_websocket()
+

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.

+
+ +
+
+get_client_socket()
+

Get the WebSocket client socket instance.

+
+
Returns:

WebSocketApp: The WebSocket client socket instance currently in use by the client.

+
+
+
+ +
+
+static 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.

+
+
+
+ +
+
+static 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.

+
+
+
+ +
+
+play_file(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.

+
+
+
+ +
+
+record(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”.

+
+
+
+ +
+
+send_packet_to_server(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.

+
+
+
+ +
+
+write_audio_frames_to_file(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.

+
+
+
+ +
+
+write_output_recording(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.

+
+
+
+ +
+ +
+
+class whisper_live.client.TranscriptionClient(host, port, is_multilingual=False, lang=None, translate=False)
+

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() +`

+
+
+
+ +
+
+whisper_live.client.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

+
+
Args:

file (str): The audio file to open +sr (int): The sample rate to resample the audio if necessary

+
+
Returns:

resampled_file (str): The resampled audio file

+
+
+
+
-

Indices and tables

+

Indices and tables

  • Index

  • Module Index

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