Voice Related

Objects

Attributes

• average_latency
• channel
• endpoint
• guild
• latency
• mode
• privacy_code
• secret_key
• session_id
• source
• ssrc
• timeout
• token
• user

Methods

• asyncdisconnect
• defelapsed
• defis_connected
• defis_dave_connection
• defis_paused
• defis_playing
• defis_recording
• defis_speaking
• asyncmove_to
• defpause
• defplay
• defresume
• defsend_audio_packet
• defstart_listening
• defstart_recording
• defstop
• defstop_listening
• defstop_recording

class discord.voice.VoiceClient(client, channel)

Represents a Discord voice connection.

You do not create these, you typically get them from e.g. VoiceChannel.connect().

session_id

The voice connection session ID.

Type:

str

token

The voice connection token.

Type:

str

endpoint

The endpoint we are connecting to.

Type:

str

channel

The channel we are connected to.

Type:

Union[VoiceChannel, StageChannel]

警告

In order to use PCM based AudioSources, you must have the opus library installed on your system and loaded through opus.load_opus(). Otherwise, your AudioSources must be opus encoded (e.g. using FFmpegOpusAudio) or the library will not be able ot transmit audio.

パラメータ:

• client (Client)

• channel (Connectable)

property guild: Guild

Returns the guild the channel we're connecting to is bound to.

property user: ClientUser

The user connected to voice (i.e. ourselves)

property session_id: str | None

The session ID of the current voice call.

property token: str | None

The token of the voice connection. You should not share this.

property endpoint: str | None

The endpoint where the client is connected.

property ssrc: int

The SSRC of the current user in the voice call.

property mode: Literal['xsalsa20_poly1305_lite', 'xsalsa20_poly1305_suffix', 'xsalsa20_poly1305', 'aead_xchacha20_poly1305_rtpsize']

The encryption / decryption mode the voice client is currently using.

property secret_key: list[int]

Returns the secret key of the current connected voice call.

property timeout: float

The amount of ms the client waits before killing connect attempts.

is_dave_connection()

Whether the voice client is connected to a DAVE call.

戻り値の型:

bool

property latency: float

Latency between a HEARTBEAT and a HEARBEAT_ACK in seconds.

This chould be referred to as the Discord Voice WebSocket latency and is and analogue of user's voice latencies as seen in the Discord client.

Added in version 1.4.

property average_latency: float

Average of most recent 20 HEARBEAT latencies in seconds.

Added in version 1.4.

property privacy_code: str | None

Returns the current voice session's privacy code, only available if the call has upgraded to use the DAVE protocol

await disconnect(*, force=False)

This function is a coroutine.

Disconnects this voice client from voice.

パラメータ:

force (bool)

戻り値の型:

None

await move_to(channel, *, timeout=30.0)

This function is a coroutine.

moves you to a different voice channel.

パラメータ:

• channel (Snowflake | None) -- The channel to move to. If this is None, it is an equivalent of calling disconnect().

• timeout (float | None) -- The maximum time in seconds to wait for the channel move to be completed, defaults to 30. If it is None, then there is no timeout.

例外:

asyncio.TimeoutError -- Waiting for channel move timed out.

戻り値の型:

None

is_connected()

Whether the voice client is connected to voice.

戻り値の型:

bool

is_playing()

INdicates if we're playing audio.

戻り値の型:

bool

is_paused()

Indicates if we're playing audio, but if we're paused.

戻り値の型:

bool

play(source, *, after=None, application='audio', bitrate=128, fec=True, expected_packet_loss=0.15, bandwidth='full', signal_type='auto', wait_finish=False)

Plays an AudioSource.

The finalizer, after is called after the source has been exhausted or an error occurred.

IF an error happens while the audio player is running, the exception is caught and the audio player is then stopped. If no after callback is passed, any caught exception will be displayed as if it were raised.

パラメータ:

• source (AudioSource) -- The audio source we're reading from.

• after (Callable[[Exception | None], Any] | None) -- The finalizer that is called after the stream is exhausted. This function must have a single parameter, error, that denotes an optional exception that was raised during playing.

• application (Literal['audio', 'voip', 'lowdelay']) -- The intended application encoder application type. Must be one of audio, voip, or lowdelay. Defaults to audio.

• bitrate (int) -- The encoder's bitrate. Must be between 16 and 512. Defaults to 128.

• fec (bool) -- Configures the encoder's use of inband forward error correction (fec). Defaults to True.

• expected_packet_loss (float) -- How much packet loss percentage is expected from the encoder. This requires fec to be set to True. Defaults to 0.15.

• bandwidth (Literal['narrow', 'medium', 'wide', 'superwide', 'full']) -- The encoder's bandpass. Must be one of narrow, medium, wide, superwide, or full. Defaults to full.

• signal_type (Literal['auto', 'voice', 'music']) -- The type of signal being encoded. Must be one of auto, voice, music. Defaults to auto.

• wait_finish (bool) --

  If True, then an awaitable is returned that waits for the audio source to be exhausted, and will return an optional exception that could have been raised.

  If False, None is returned and the function does not block.

  Added in version 2.5.

例外:

• ClientException -- Already playing audio, or not connected to voice.

• TypeError -- Source is not a AudioSource, or after is not a callable.

• OpusNotLoaded -- Source is not opus encoded and opus is not loaded.

戻り値の型:

None | Future[None]

stop()

Stops playing audio, if applicable.

戻り値の型:

None

pause()

Pauses the audio playing.

戻り値の型:

None

resume()

Resumes the audio playing.

戻り値の型:

None

property source: AudioSource | None

The audio source being player, if playing.

This property can also be used to change the audio source currently being played.

send_audio_packet(data, *, encode=True)

Sends an audio packet composed of the data.

You must be connected to play audio.

パラメータ:

• data (bytes) -- The bytes-like object denoting PCM or Opus voice data.

• encode (bool) -- Indicates if data should be encoded into Opus.

例外:

• ClientException -- You are not connected.

• opus.OpusError -- Encoding the data failed.

戻り値の型:

None

elapsed()

Returns the elapsed time of the playing audio.

戻り値の型:

timedelta

start_recording(sink, callback=None, *args, sync_start=...)

Start recording the audio from the current connected channel to the provided sink.

Added in version 2.0.

警告

Recording may not work as expected due to the new DAVE (End-to-End Encryption) for voice calls.

パラメータ:

• sink (Sink) -- A Sink in which all audio packets will be processed in.

• callback (Callable[[Exception | None], Any] | None) --

  A function which is called after the bot has stopped recording. This must take exactly one positonal(-only) parameter, exception, which is the exception that was raised during the recording of the Sink.

  バージョン 2.7 で変更: This parameter is now optional, and must take exactly one parameter, exception.

• *args (Any) --

  The arguments to pass to the callback coroutine.

  バージョン 2.7 で非推奨: Passing custom arguments to the callback is now deprecated and ignored.

• sync_start (bool) --

  If True, the recordings of subsequent users will start with silence. This is useful for recording audio just as it was heard.

  バージョン 2.7 で非推奨: This parameter is now ignored and deprecated.

例外:

• RecordingException -- Not connected to a voice channel

• TypeError -- You did not provide a Sink object.

戻り値の型:

None

start_listening(sink, callback=None, *args, sync_start=...)

Start recording the audio from the current connected channel to the provided sink.

Added in version 2.0.

警告

Recording may not work as expected due to the new DAVE (End-to-End Encryption) for voice calls.

パラメータ:

• sink (Sink) -- A Sink in which all audio packets will be processed in.

• callback (Callable[[Exception | None], Any] | None) --

  A function which is called after the bot has stopped recording. This must take exactly one positonal(-only) parameter, exception, which is the exception that was raised during the recording of the Sink.

  バージョン 2.7 で変更: This parameter is now optional, and must take exactly one parameter, exception.

• *args (Any) --

  The arguments to pass to the callback coroutine.

  バージョン 2.7 で非推奨: Passing custom arguments to the callback is now deprecated and ignored.

• sync_start (bool) --

  If True, the recordings of subsequent users will start with silence. This is useful for recording audio just as it was heard.

  バージョン 2.7 で非推奨: This parameter is now ignored and deprecated.

例外:

• RecordingException -- Not connected to a voice channel

• TypeError -- You did not provide a Sink object.

戻り値の型:

None

stop_recording()

Stops the recording of the provided sink, or all recording sinks.

Added in version 2.0.

例外:

RecordingException -- You are not recording.

戻り値の型:

None

stop_listening()

Stops the recording of the provided sink, or all recording sinks.

Added in version 2.0.

例外:

RecordingException -- You are not recording.

戻り値の型:

None

is_recording()

Whether the current client is recording in any sink.

戻り値の型:

bool

is_speaking(member)

Whether a user is speaking.

This is an approximate calculation and may have outdated or wrong data.

If the member speaking status has not been yet saved, it returns None.

Added in version 2.7.

パラメータ:

member (Member | User)

戻り値の型:

bool | None

Methods

• defcleanup
• asyncconnect
• asyncdisconnect
• asyncon_voice_server_update
• asyncon_voice_state_update

class discord.voice.VoiceProtocol(client, channel)

A class that represents the Discord voice protocol.

警告

If you are an end user, you should not construct this manually but instead take it from the return type in abc.Connectable.connect. The parameters and methods being documented here is so third party libraries can refer to it when implementing their own VoiceProtocol types.

This is an abstract class. The library provides a concrete implementation under VoiceClient.

This class allows you to implement a protocol to allow for an external method of sending voice, such as Lavalink or a native library implementation.

These classes are passed to abc.Connectable.connect.

パラメータ:

• client (TypeVar(ClientT, bound= Client, covariant=True)) -- The client (or its subclasses) that started the connection request.

• channel (Connectable) -- The voice channel that is being connected to.

await on_voice_state_update(data)

This function is a coroutine.

A method called when the client's voice state has changed. This corresponds to the VOICE_STATE_UPDATE event.

パラメータ:

data (RawVoiceStateUpdateEvent) --

The voice state payload.

バージョン 2.7 で変更: This now gets passed a RawVoiceStateUpdateEvent object instead of a dict, but accessing keys via data[key] or data.get(key) is still supported, but deprecated.

戻り値の型:

None

await on_voice_server_update(data)

This function is a coroutine.

A method called when the client's intially connecting to voice. This corresponds to the VOICE_SERVER_UPDATE event.

パラメータ:

data (RawVoiceServerUpdateEvent) --

The voice server payload.

バージョン 2.7 で変更: This now gets passed a RawVoiceServerUpdateEvent object instead of a dict, but accessing keys via data[key] or data.get(key) is still supported, but deprecated.

戻り値の型:

None

await connect(*, timeout, reconnect)

This function is a coroutine.

A method called to initialise the connection.

The library initialises this class and calls __init__, and then connect() when attempting to start a connection to the voice. If an error ocurrs, it calls disconnect(), so if you need to implement any cleanup, you should manually call it in disconnect() as the library will not do so for you.

Within this method, to start the voice connection flow, it is recommened to use Guild.change_voice_state() to start the flow. After which on_voice_server_update() and on_voice_state_update() will be called, although this could vary and cause unexpected behaviour, but that falls under Discord's way of handling the voice connection.

パラメータ:

• timeout (float) -- The timeout for the connection.

• reconnect (bool) -- Whether reconnection is expected.

戻り値の型:

None

await disconnect(*, force)

This function is a coroutine.

A method called to terminate the voice connection.

This can be either called manually when forcing a disconnection, or when an exception in connect() ocurrs.

It is recommended to call cleanup() here.

パラメータ:

force (bool) -- Whether the disconnection was forced.

戻り値の型:

None

cleanup()

This method must be called to ensure proper clean-up during a disconnect.

It is advisable to call this from within disconnect() when you are completely done with the voice protocol instance.

This method removes it from the internal state cache that keeps track of the currently alive voice clients. Failure to clean-up will cause subsequent connections to report that it's still connected.

The library will NOT automatically call this for you, unlike connect() and disconnect().

戻り値の型:

None

Methods

• defcleanup
• defis_opus
• defread

class discord.AudioSource

Represents an audio stream.

The audio stream can be Opus encoded or not, however if the audio stream is not Opus encoded then the audio format must be 16-bit 48KHz stereo PCM.

警告

The audio source reads are done in a separate thread.

read()

Reads 20ms worth of audio.

Subclasses must implement this.

If the audio is complete, then returning an empty bytes-like object to signal this is the way to do so.

If is_opus() method returns True, then it must return 20ms worth of Opus encoded audio. Otherwise, it must be 20ms worth of 16-bit 48KHz stereo PCM, which is about 3,840 bytes per frame (20ms worth of audio).

戻り値:

A bytes like object that represents the PCM or Opus data.

戻り値の型:

bytes

is_opus()

Checks if the audio source is already encoded in Opus.

戻り値の型:

bool

cleanup()

Called when clean-up is needed to be done.

Useful for clearing buffer data or processes after it is done playing audio.

戻り値の型:

None

Attributes

• stream

Methods

• defread

class discord.PCMAudio(stream)

Represents raw 16-bit 48KHz stereo PCM audio source.

stream

A file-like object that reads byte data representing raw PCM.

Type:

file object

パラメータ:

stream (BufferedIOBase)

read()

Reads 20ms worth of audio.

Subclasses must implement this.

If the audio is complete, then returning an empty bytes-like object to signal this is the way to do so.

If is_opus() method returns True, then it must return 20ms worth of Opus encoded audio. Otherwise, it must be 20ms worth of 16-bit 48KHz stereo PCM, which is about 3,840 bytes per frame (20ms worth of audio).

戻り値:

A bytes like object that represents the PCM or Opus data.

戻り値の型:

bytes

Methods

• defcleanup

class discord.FFmpegAudio(source, *, executable='ffmpeg', args, **subprocess_kwargs)

Represents an FFmpeg (or AVConv) based AudioSource.

User created AudioSources using FFmpeg differently from how FFmpegPCMAudio and FFmpegOpusAudio work should subclass this.

Added in version 1.3.

パラメータ:

• source (str | BufferedIOBase)

• executable (str)

• args (Any)

• subprocess_kwargs (Any)

cleanup()

Called when clean-up is needed to be done.

Useful for clearing buffer data or processes after it is done playing audio.

戻り値の型:

None

Methods

• defis_opus
• defread

class discord.FFmpegPCMAudio(source, *, executable='ffmpeg', pipe=False, stderr=None, before_options=None, options=None)

An audio source from FFmpeg (or AVConv).

This launches a sub-process to a specific input file given.

警告

You must have the ffmpeg or avconv executable in your path environment variable in order for this to work.

パラメータ:

• source (str | BufferedIOBase) -- The input that ffmpeg will take and convert to PCM bytes. If pipe is True then this is a file-like object that is passed to the stdin of ffmpeg.

• executable (str) --

  The executable name (and path) to use. Defaults to ffmpeg.

  警告

  Since this class spawns a subprocess, care should be taken to not pass in an arbitrary executable name when using this parameter.

• pipe (bool) -- If True, denotes that source parameter will be passed to the stdin of ffmpeg. Defaults to False.

• stderr (IO[bytes] | None) -- A file-like object to pass to the Popen constructor.

• before_options (str | None) -- Extra command line arguments to pass to ffmpeg before the -i flag.

• options (str | None) -- Extra command line arguments to pass to ffmpeg after the -i flag.

例外:

ClientException -- The subprocess failed to be created.

read()

Reads 20ms worth of audio.

Subclasses must implement this.

If the audio is complete, then returning an empty bytes-like object to signal this is the way to do so.

If is_opus() method returns True, then it must return 20ms worth of Opus encoded audio. Otherwise, it must be 20ms worth of 16-bit 48KHz stereo PCM, which is about 3,840 bytes per frame (20ms worth of audio).

戻り値:

A bytes like object that represents the PCM or Opus data.

戻り値の型:

bytes

is_opus()

Checks if the audio source is already encoded in Opus.

戻り値の型:

bool

Methods

• asyncfrom_probe
• defis_opus
• asyncprobe
• defread

class discord.FFmpegOpusAudio(source, *, bitrate=None, codec=None, executable='ffmpeg', pipe=False, stderr=None, before_options=None, options=None)

An audio source from FFmpeg (or AVConv).

This launches a sub-process to a specific input file given. However, rather than producing PCM packets like FFmpegPCMAudio does that need to be encoded to Opus, this class produces Opus packets, skipping the encoding step done by the library.

Alternatively, instead of instantiating this class directly, you can use FFmpegOpusAudio.from_probe() to probe for bitrate and codec information. This can be used to opportunistically skip pointless re-encoding of existing Opus audio data for a boost in performance at the cost of a short initial delay to gather the information. The same can be achieved by passing copy to the codec parameter, but only if you know that the input source is Opus encoded beforehand.

Added in version 1.3.

警告

You must have the ffmpeg or avconv executable in your path environment variable in order for this to work.

パラメータ:

• source (str | BufferedIOBase) -- The input that ffmpeg will take and convert to Opus bytes. If pipe is True then this is a file-like object that is passed to the stdin of ffmpeg.

• bitrate (int | None) -- The bitrate in kbps to encode the output to. Defaults to 128.

• codec (str | None) --

  The codec to use to encode the audio data. Normally this would be just libopus, but is used by FFmpegOpusAudio.from_probe() to opportunistically skip pointlessly re-encoding Opus audio data by passing copy as the codec value. Any values other than copy, opus, or libopus will be considered libopus. Defaults to libopus.

  警告

  Do not provide this parameter unless you are certain that the audio input is already Opus encoded. For typical use FFmpegOpusAudio.from_probe() should be used to determine the proper value for this parameter.

• executable (str) -- The executable name (and path) to use. Defaults to ffmpeg.

• pipe (bool) -- If True, denotes that source parameter will be passed to the stdin of ffmpeg. Defaults to False.

• stderr (IO[bytes] | None) -- A file-like object to pass to the Popen constructor.

• before_options (str | None) -- Extra command line arguments to pass to ffmpeg before the -i flag.

• options (str | None) -- Extra command line arguments to pass to ffmpeg after the -i flag.

例外:

ClientException -- The subprocess failed to be created.

classmethod await from_probe(source, *, method=None, **kwargs)

This function is a coroutine.

A factory method that creates a FFmpegOpusAudio after probing the input source for audio codec and bitrate information.

サンプル

Use this function to create an FFmpegOpusAudio instance instead of the constructor:

source = await discord.FFmpegOpusAudio.from_probe("song.webm")
voice_client.play(source)

If you are on Windows and don't have ffprobe installed, use the fallback method to probe using ffmpeg instead:

source = await discord.FFmpegOpusAudio.from_probe("song.webm", method='fallback')
voice_client.play(source)

Using a custom method of determining codec and bitrate:

def custom_probe(source, executable):
    # some analysis code here
    return codec, bitrate

source = await discord.FFmpegOpusAudio.from_probe("song.webm", method=custom_probe)
voice_client.play(source)

パラメータ:

• source (str) -- Identical to the source parameter for the constructor.

• method (str | Callable[[str, str], tuple[str | None, int | None]] | None) -- The probing method used to determine bitrate and codec information. As a string, valid values are native to use ffprobe (or avprobe) and fallback to use ffmpeg (or avconv). As a callable, it must take two string arguments, source and executable. Both parameters are the same values passed to this factory function. executable will default to ffmpeg if not provided as a keyword argument.

• **kwargs (Any) -- The remaining parameters to be passed to the FFmpegOpusAudio constructor, excluding bitrate and codec.

例外:

• AttributeError -- Invalid probe method, must be 'native' or 'fallback'.

• TypeError -- Invalid value for probe parameter, must be str or a callable.

戻り値:

An instance of this class.

戻り値の型:

Self

classmethod await probe(source, *, method=None, executable=None)

This function is a coroutine.

Probes the input source for bitrate and codec information.

パラメータ:

• source (str) -- Identical to the source parameter for FFmpegOpusAudio.

• method (str | Callable[[str, str], tuple[str | None, int | None]] | None) -- Identical to the method parameter for FFmpegOpusAudio.from_probe().

• executable (str | None) -- Identical to the executable parameter for FFmpegOpusAudio.

戻り値:

A 2-tuple with the codec and bitrate of the input source.

戻り値の型:

tuple[str | None, int | None]

例外:

• AttributeError -- Invalid probe method, must be 'native' or 'fallback'.

• TypeError -- Invalid value for probe parameter, must be str or a callable.

read()

Reads 20ms worth of audio.

Subclasses must implement this.

If the audio is complete, then returning an empty bytes-like object to signal this is the way to do so.

If is_opus() method returns True, then it must return 20ms worth of Opus encoded audio. Otherwise, it must be 20ms worth of 16-bit 48KHz stereo PCM, which is about 3,840 bytes per frame (20ms worth of audio).

戻り値:

A bytes like object that represents the PCM or Opus data.

戻り値の型:

bytes

is_opus()

Checks if the audio source is already encoded in Opus.

戻り値の型:

bool

Attributes

• volume

Methods

• defcleanup
• defread

class discord.PCMVolumeTransformer(original, volume=1.0)

Transforms a previous AudioSource to have volume controls.

This does not work on audio sources that have AudioSource.is_opus() set to True.

パラメータ:

• original (TypeVar(AT, bound= AudioSource)) -- The original AudioSource to transform.

• volume (float) -- The initial volume to set it to. See volume for more info.

例外:

• TypeError -- Not an audio source.

• ClientException -- The audio source is opus encoded.

property volume: float

Retrieves or sets the volume as a floating point percentage (e.g. 1.0 for 100%).

cleanup()

Called when clean-up is needed to be done.

Useful for clearing buffer data or processes after it is done playing audio.

戻り値の型:

None

read()

Reads 20ms worth of audio.

Subclasses must implement this.

If the audio is complete, then returning an empty bytes-like object to signal this is the way to do so.

If is_opus() method returns True, then it must return 20ms worth of Opus encoded audio. Otherwise, it must be 20ms worth of 16-bit 48KHz stereo PCM, which is about 3,840 bytes per frame (20ms worth of audio).

戻り値:

A bytes like object that represents the PCM or Opus data.

戻り値の型:

bytes

Opus Library

discord.opus.load_opus(name)

Loads the libopus shared library for use with voice.

If this function is not called then the library uses the function ctypes.util.find_library() and then loads that one if available.

Not loading a library and attempting to use PCM based AudioSources will lead to voice not working.

This function propagates the exceptions thrown.

警告

The bitness of the library must match the bitness of your python interpreter. If the library is 64-bit then your python interpreter must be 64-bit as well. Usually if there's a mismatch in bitness then the load will throw an exception.

注釈

On Windows, this function should not need to be called as the binaries are automatically loaded.

注釈

On Windows, the .dll extension is not necessary. However, on Linux the full extension is required to load the library, e.g. libopus.so.1. On Linux however, ctypes.util.find_library() will usually find the library automatically without you having to call this.

パラメータ:

name (str) -- The filename of the shared library.

戻り値の型:

None

discord.opus.is_loaded()

Function to check if opus lib is successfully loaded either via the ctypes.util.find_library() call of load_opus().

This must return True for voice to work.

戻り値:

Indicates if the opus library has been loaded.

戻り値の型:

bool