SqMod/vendor/DPP/include/dpp/discordvoiceclient.h

/************************************************************************************
 *
 * D++, A Lightweight C++ library for Discord
 *
 * Copyright 2021 Craig Edwards and D++ contributors 
 * (https://github.com/brainboxdotcc/DPP/graphs/contributors)
 *
 * Licensed under the Apache License, Version 2.0 (the "License");
 * you may not use this file except in compliance with the License.
 * You may obtain a copy of the License at
 *
 * http://www.apache.org/licenses/LICENSE-2.0
 *
 * Unless required by applicable law or agreed to in writing, software
 * distributed under the License is distributed on an "AS IS" BASIS,
 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 * See the License for the specific language governing permissions and
 * limitations under the License.
 *
 ************************************************************************************/
#pragma once

#include <dpp/export.h>

#include <cerrno>
#include <cstdio>
#include <cstdlib>
#include <sys/types.h>
#include <fcntl.h>
#include <csignal>
#include <cstring>
#include <string>
#include <map>
#include <vector>
#include <dpp/json_fwd.h>
#include <dpp/wsclient.h>
#include <dpp/dispatcher.h>
#include <dpp/cluster.h>
#include <dpp/discordevents.h>
#include <dpp/socket.h>
#include <queue>
#include <thread>
#include <deque>
#include <mutex>
#include <shared_mutex>
#include <memory>
#include <future>
#include <functional>
#include <chrono>



struct OpusDecoder;
struct OpusEncoder;
struct OpusRepacketizer;

namespace dpp {

using json = nlohmann::json;

// Forward declaration
class cluster;

/**
 * @brief An opus-encoded RTP packet to be sent out to a voice channel
 */
struct DPP_EXPORT voice_out_packet {
	/** 
	 * @brief Each string is a UDP packet.
	 * Generally these will be RTP.
	 */
	std::string packet;
	/**
	 * @brief Duration of packet
	 */
	uint64_t duration;
};

#define AUDIO_TRACK_MARKER (uint16_t)0xFFFF

#define AUDIO_OVERLAP_SLEEP_SAMPLES 30

/** @brief Implements a discord voice connection.
 * Each discord_voice_client connects to one voice channel and derives from a websocket client.
 */
class DPP_EXPORT discord_voice_client : public websocket_client
{
	/**
	 * @brief Clean up resources
	 */
	void cleanup();

	/**
	 * @brief Mutex for outbound packet stream
	 */
	std::mutex stream_mutex;

	/**
	 * @brief Mutex for message queue
	 */
	std::shared_mutex queue_mutex;

	/**
	 * @brief Queue of outbound messages
	 */
	std::deque<std::string> message_queue;

	/**
	 * @brief Thread this connection is executing on
	 */
	std::thread* runner;

	/**
	 * @brief Run shard loop under a thread
	 */
	void thread_run();

	/**
	 * @brief Last connect time of voice session
	 */
	time_t connect_time;

	/**
	 * @brief IP of UDP/RTP endpoint
	 */
	std::string ip;

	/**
	 * @brief Port number of UDP/RTP endpoint
	 */
	uint16_t port;

	/**
	 * @brief SSRC value 
	 */
	uint64_t ssrc;

	/**
	 * @brief List of supported audio encoding modes
	 */
	std::vector<std::string> modes;

	/**
	 * @brief Timescale in nanoseconds
	 */
	uint64_t timescale;

	/**
	 * @brief Output buffer
	 */
	std::vector<voice_out_packet> outbuf;

	/**
	 * @brief Data type of RTP packet sequence number field.
	 */
	using rtp_seq_t = uint16_t;
	using rtp_timestamp_t = uint32_t;

	/**
	 * @brief Keeps track of the voice payload to deliver to voice handlers.
	 */
	struct voice_payload {
		/**
		 * @brief The sequence number of the RTP packet that generated this
		 * voice payload.
		 */
		rtp_seq_t seq;
		/**
		 * @brief The timestamp of the RTP packet that generated this voice
		 * payload.
		 *
		 * The timestamp is used to detect the order around where sequence
		 * number wraps around.
		 */
		rtp_timestamp_t timestamp;
		/**
		 * @brief The event payload that voice handlers receive.
		 */
		std::unique_ptr<voice_receive_t> vr;
		
		/**
		 * @brief For priority_queue sorting.
		 * @return true if "this" has lower priority that "other",
		 *         i.e. appears later in the queue; false otherwise.
		 */
		bool operator<(const voice_payload& other) const;
	};

	struct voice_payload_parking_lot {
		/**
		 * @brief The range of RTP packet sequence number and timestamp in the lot.
		 *
		 * The minimum is used to drop packets that arrive too late. Packets
		 * less than the minimum have been delivered to voice handlers and
		 * there is no going back. Unfortunately we just have to drop them.
		 *
		 * The maximum is used, at flush time, to calculate the minimum for
		 * the next batch. The maximum is also updated every time we receive an
		 * RTP packet with a larger value.
		 */
		struct seq_range_t {
			rtp_seq_t min_seq, max_seq;
			rtp_timestamp_t min_timestamp, max_timestamp;
		} range;
		/**
		 * @brief The queue of parked voice payloads.
		 * 
		 * We group payloads and deliver them to handlers periodically as the
		 * handling of out-of-order RTP packets. Payloads in between flushes
		 * are parked and sorted in this queue.
		 */
		std::priority_queue<voice_payload> parked_payloads;
		/**
		 * @brief The decoder ctls to be set on the decoder.
		 */
		std::vector<std::function<void(OpusDecoder&)>> pending_decoder_ctls;
		/**
		 * @brief libopus decoder
		 *
		 * Shared with the voice courier thread that does the decoding.
		 * This is not protected by a mutex because only the courier thread
		 * uses the decoder.
		 */
		std::shared_ptr<OpusDecoder> decoder;
	};
	/**
	 * @brief Thread used to deliver incoming voice data to handlers.
	 */
	std::thread voice_courier;
	/**
	 * @brief Shared state between this voice client and the courier thread.
	 */
	struct courier_shared_state_t {
		/**
		 * @brief Protects all following members.
		 */
		std::mutex mtx;
		/**
		 * @brief Signaled when there is a new payload to deliver or terminating state has changed.
		 */
		std::condition_variable signal_iteration;
		/**
		 * @brief Voice buffers to be reported to handler, grouped by speaker.
		 *
		 * Buffers are parked here and flushed every 500ms.
		 */
		std::map<snowflake, voice_payload_parking_lot> parked_voice_payloads;
		/**
		 * @brief Used to signal termination.
		 *
		 * @note Pending payloads are delivered first before termination.
		 */
		bool terminating = false;
	} voice_courier_shared_state;
	/**
	 * @brief The run loop of the voice courier thread.
	 */
	static void voice_courier_loop(discord_voice_client&, courier_shared_state_t&);

	/**
	 * @brief If true, audio packet sending is paused
	 */
	bool paused;

#ifdef HAVE_VOICE
	/**
	 * @brief libopus encoder
	 */
	OpusEncoder* encoder;

	/**
	 * @brief libopus repacketizer
	 * (merges frames into one packet)
	 */
	OpusRepacketizer* repacketizer;
#else
	/**
	 * @brief libopus encoder
	 */
	void* encoder;

	/**
	 * @brief libopus repacketizer
	 * (merges frames into one packet)
	 */
	void* repacketizer;
#endif

	/**
	 * @brief File descriptor for UDP connection
	 */
	dpp::socket fd;

	/**
	 * @brief Secret key for encrypting voice.
	 * If it has been sent, this is non-null and points to a 
	 * sequence of exactly 32 bytes.
	 */
	uint8_t* secret_key;

	/**
	 * @brief Sequence number of outbound audio. This is incremented
	 * once per frame sent.
	 */
	uint16_t sequence;

	/**
	 * @brief Timestamp value used in outbound audio. Each packet
	 * has the timestamp value which is incremented to match
	 * how many frames are sent.
	 */
	uint32_t timestamp;

	/**
	 * @brief Last sent packet high-resolution timestamp
	 */
	std::chrono::high_resolution_clock::time_point last_timestamp;

	/**
	 * @brief Fraction of the sleep that was not executed after the last audio packet was sent
	 */
	std::chrono::nanoseconds last_sleep_remainder;

	/**
	 * @brief Maps receiving ssrc to user id
	 */
	std::unordered_map<uint32_t, snowflake> ssrc_map;

	/**
	 * @brief This is set to true if we have started sending audio.
	 * When this moves from false to true, this causes the
	 * client to send the 'talking' notification to the websocket.
	 */
	bool sending;

	/**
	 * @brief Number of track markers in the buffer. For example if there
	 * are two track markers in the buffer there are 3 tracks.
	 * 
	 * **Special case:**
	 * 
	 * If the buffer is empty, there are zero tracks in the
	 * buffer.
	 */
	uint32_t tracks;

	/**
	 * @brief Meta data associated with each track.
	 * Arbitrary string that the user can set via
	 * dpp::discord_voice_client::add_marker
	 */
	std::vector<std::string> track_meta;

	/** 
	 * @brief Encoding buffer for opus repacketizer and encode
	 */
	uint8_t encode_buffer[65536];

	/**
	 * @brief Send data to UDP socket immediately.
	 * 
	 * @param data data to send
	 * @param length length of data to send
	 * @return int bytes sent. Will return -1 if we cannot send
	 */
	int udp_send(const char* data, size_t length);

	/**
	 * @brief Receive data from UDP socket immediately.
	 * 
	 * @param data data to receive
	 * @param max_length size of data receiving buffer
	 * @return int bytes received. -1 if there is an error
	 * (e.g. EAGAIN)
	 */
	int udp_recv(char* data, size_t max_length);

	/**
	 * @brief This hooks the ssl_client, returning the file
	 * descriptor if we want to send buffered data, or
	 * -1 if there is nothing to send
	 * 
	 * @return int file descriptor or -1
	 */
	dpp::socket want_write();

	/**
	 * @brief This hooks the ssl_client, returning the file
	 * descriptor if we want to receive buffered data, or
	 * -1 if we are not wanting to receive
	 * 
	 * @return int file descriptor or -1
	 */
	dpp::socket want_read();

	/**
	 * @brief Called by ssl_client when the socket is ready
	 * for writing, at this point we pick the head item off
	 * the buffer and send it. So long as it doesn't error
	 * completely, we pop it off the head of the queue.
	 */
	void write_ready();

	/**
	 * @brief Called by ssl_client when there is data to be
	 * read. At this point we insert that data into the
	 * input queue.
	 * @throw dpp::voice_exception if voice support is not compiled into D++
	 */
	void read_ready();

	/**
	 * @brief Send data to the UDP socket, using the buffer.
	 * 
	 * @param packet packet data
	 * @param len length of packet
	 * @param duration duration of opus packet
	 */
	void send(const char* packet, size_t len, uint64_t duration);

	/**
	 * @brief Queue a message to be sent via the websocket
	 * 
	 * @param j The JSON data of the message to be sent
	 * @param to_front If set to true, will place the message at the front of the queue not the back
	 * (this is for urgent messages such as heartbeat, presence, so they can take precedence over
	 * chunk requests etc)
	 */
	void queue_message(const std::string &j, bool to_front = false);

	/**
	 * @brief Clear the outbound message queue
	 * 
	 */
	void clear_queue();

	/**
	 * @brief Get the size of the outbound message queue
	 * 
	 * @return The size of the queue
	 */
	size_t get_queue_size();

	/**
	 * @brief Encode a byte buffer using opus codec.
	 * Multiple opus frames (2880 bytes each) will be encoded into one packet for sending.
	 * 
	 * @param input Input data as raw bytes of PCM data
	 * @param inDataSize Input data length
	 * @param output Output data as an opus encoded packet
	 * @param outDataSize Output data length, should be at least equal to the input size.
	 * Will be adjusted on return to the actual compressed data size.
	 * @return size_t The compressed data size that was encoded.
	 * @throw dpp::voice_exception If data length to encode is invalid or voice support not compiled into D++
	 */
	size_t encode(uint8_t *input, size_t inDataSize, uint8_t *output, size_t &outDataSize);

public:

	/**
	 * @brief Owning cluster
	 */
	class dpp::cluster* creator;

	/**
	 * @brief This needs to be static, we only initialise libsodium once per program start,
	 * so initialising it on first use in a voice connection is best.
	 */
	static bool sodium_initialised;

	/**
	 * @brief True when the thread is shutting down
	 */
	bool terminating;

	/**
	 * @brief Heartbeat interval for sending heartbeat keepalive
	 */
	uint32_t heartbeat_interval;

	/**
	 * @brief Last voice channel websocket heartbeat
	 */
	time_t last_heartbeat;

	/**
	 * @brief Thread ID
	 */
	std::thread::native_handle_type thread_id;

	/**
	 * @brief Discord voice session token
	 */
	std::string token;

	/**
	 * @brief Discord voice session id
	 */
	std::string sessionid;

	/**
	 * @brief Server ID
	 */
	snowflake server_id;

	/**
	 * @brief Channel ID
	 */
	snowflake channel_id;

	/**
	 * @brief The audio type to be sent. The default type is recorded audio.
	 *
	 * If the audio is recorded, the sending of audio packets is throttled.
	 * Otherwise, if the audio is live, the sending is not throttled.
	 *
	 * Discord voice engine is expecting audio data as if they were from
	 * some audio device, e.g. microphone, where the data become available
	 * as they get captured from the audio device.
	 *
	 * In case of recorded audio, unlike from a device, the audio data are
	 * usually instantly available in large chunks. Throttling is needed to
	 * simulate audio data coming from an audio device. In case of live audio,
	 * the throttling is by nature, so no extra throttling is needed.
	 *
	 * Using live audio mode for recorded audio can cause Discord to skip
	 * audio data because Discord does not expect to receive, say, 3 minutes'
	 * worth of audio data in 1 second.
	 *
	 * There are some inaccuracies in the throttling method used by the recorded
	 * audio mode on some systems (mainly Windows) which causes gaps and stutters
	 * in the resulting audio stream. The overlap audio mode provides a different 
	 * implementation that fixes the issue. This method is slightly more CPU 
	 * intensive, and should only be used if you encounter issues with recorded audio 
	 * on your system.
	 * 
	 * Use discord_voice_client::set_send_audio_type to change this value as
	 * it ensures thread safety.
	 */
	enum send_audio_type_t
	{
	    satype_recorded_audio,
	    satype_live_audio,
		satype_overlap_audio
	} send_audio_type = satype_recorded_audio;

	/**
	 * @brief Sets the gain for the specified user.
	 *
	 * Similar to the User Volume slider, controls the listening volume per user.
	 * Uses native Opus gain control, so clients don't have to perform extra
	 * audio processing.
	 *
	 * The gain setting will affect the both individual and combined voice audio.
	 *
	 * The gain value can also be set even before the user connects to the voice
	 * channel.
	 *
	 * @param user_id The ID of the user where the gain is to be controlled.
	 * @param factor Nonnegative factor to scale the amplitude by, where 1.f reverts
	 *               to the default volume.
	 */
	void set_user_gain(snowflake user_id, float factor);

	/**
	 * @brief Log a message to whatever log the user is using.
	 * The logged message is passed up the chain to the on_log event in user code which can then do whatever
	 * it wants to do with it.
	 * @param severity The log level from dpp::loglevel
	 * @param msg The log message to output
	 */
	virtual void log(dpp::loglevel severity, const std::string &msg) const;

	/**
	 * @brief Fires every second from the underlying socket I/O loop, used for sending heartbeats
	 * @throw dpp::exception if the socket needs to disconnect
	 */
	virtual void one_second_timer();

	/**
	 * @brief voice client is ready to stream audio.
	 * The voice client is considered ready if it has a secret key.
	 * 
	 * @return true if ready to stream audio
	 */
	bool is_ready();

	/**
	 * @brief Returns true if the voice client is connected to the websocket
	 * 
	 * @return True if connected
	 */
	bool is_connected();

	/**
	 * @brief Returns the connection time of the voice client
	 * 
	 * @return dpp::utility::uptime Detail of how long the voice client has been connected for
	 */
	dpp::utility::uptime get_uptime();

	/** Constructor takes shard id, max shards and token.
	 * @param _cluster The cluster which owns this voice connection, for related logging, REST requests etc
	 * @param _channel_id The channel id to identify the voice connection as
	 * @param _server_id The server id (guild id) to identify the voice connection as
	 * @param _token The voice session token to use for identifying to the websocket
	 * @param _session_id The voice session id to identify with
	 * @param _host The voice server hostname to connect to (hostname:port format)
	 * @throw dpp::voice_exception Sodium or Opus failed to initialise, or D++ is not compiled with voice support
	 */
	discord_voice_client(dpp::cluster* _cluster, snowflake _channel_id, snowflake _server_id, const std::string &_token, const std::string &_session_id, const std::string &_host);

	/**
	 * @brief Destroy the discord voice client object
	 */
	virtual ~discord_voice_client();

	/**
	 * @brief Handle JSON from the websocket.
	 * @param buffer The entire buffer content from the websocket client
	 * @return bool True if a frame has been handled
	 * @throw dpp::exception If there was an error processing the frame, or connection to UDP socket failed
	 */
	virtual bool handle_frame(const std::string &buffer);

	/**
	 * @brief Handle a websocket error.
	 * @param errorcode The error returned from the websocket
	 */
	virtual void error(uint32_t errorcode);

	/**
	 * @brief Start and monitor I/O loop
	 */
	void run();

	/**
	 * @brief Send raw audio to the voice channel.
	 * 
	 * You should send an audio packet of 11520 bytes.
	 * Note that this function can be costly as it has to opus encode
	 * the PCM audio on the fly, and also encrypt it with libsodium.
	 * 
	 * @note Because this function encrypts and encodes packets before
	 * pushing them onto the output queue, if you have a complete stream
	 * ready to send and know its length it is advisable to call this
	 * method multiple times to enqueue the entire stream audio so that
	 * it is all encoded at once (unless you have set use_opus to false).
	 * Constantly calling this from the dpp::on_voice_buffer_send callback
	 * can and will eat a TON of cpu!
	 * 
	 * @param audio_data Raw PCM audio data. Channels are interleaved,
	 * with each channel's amplitude being a 16 bit value.
	 * 
	 * The audio data should be 48000Hz signed 16 bit audio.
	 * 
	 * @param length The length of the audio data. The length should
	 * be a multiple of 4 (2x 16 bit stereo channels) with a maximum
	 * length of 11520, which is a complete opus frame at highest
	 * quality.
	 * 
	 * @return discord_voice_client& Reference to self
	 * 
	 * @throw dpp::voice_exception If data length is invalid or voice support not compiled into D++
	 */
	discord_voice_client& send_audio_raw(uint16_t* audio_data, const size_t length);

	/**
	 * @brief Send opus packets to the voice channel
	 * 
	 * Some containers such as .ogg may contain OPUS
	 * encoded data already. In this case, we don't need to encode the
	 * frames using opus here. We can bypass the codec, only applying 
	 * libsodium to the stream.
	 * 
	 * @param opus_packet Opus packets. Discord expects opus frames 
	 * to be encoded at 48000Hz
	 * 
	 * @param length The length of the audio data. 
	 * 
	 * @param duration Generally duration is 2.5, 5, 10, 20, 40 or 60
	 * if the timescale is 1000000 (1ms) 
	 * 
	 * @return discord_voice_client& Reference to self
	 * 
	 * @note It is your responsibility to ensure that packets of data 
	 * sent to send_audio are correctly repacketized for streaming, 
	 * e.g. that audio frames are not too large or contain
	 * an incorrect format. Discord will still expect the same frequency
	 * and bit width of audio and the same signedness.
	 * 
	 * @throw dpp::voice_exception If data length is invalid or voice support not compiled into D++
	 */
	discord_voice_client& send_audio_opus(uint8_t* opus_packet, const size_t length, uint64_t duration);

	/**
	 * @brief Send opus packets to the voice channel
	 * 
	 * Some containers such as .ogg may contain OPUS
	 * encoded data already. In this case, we don't need to encode the
	 * frames using opus here. We can bypass the codec, only applying 
	 * libsodium to the stream.
	 * 
	 * Duration is calculated internally
	 * 
	 * @param opus_packet Opus packets. Discord expects opus frames 
	 * to be encoded at 48000Hz
	 * 
	 * @param length The length of the audio data. 
	 * 
	 * @return discord_voice_client& Reference to self
	 * 
	 * @note It is your responsibility to ensure that packets of data 
	 * sent to send_audio are correctly repacketized for streaming, 
	 * e.g. that audio frames are not too large or contain
	 * an incorrect format. Discord will still expect the same frequency
	 * and bit width of audio and the same signedness.
	 * 
	 * @throw dpp::voice_exception If data length is invalid or voice support not compiled into D++
	 */
	discord_voice_client& send_audio_opus(uint8_t* opus_packet, const size_t length);

	/**
	 * @brief Send silence to the voice channel
	 * 
	 * @param duration How long to send silence for. With the standard
	 * timescale this is in milliseconds. Allowed values are 2.5,
	 * 5, 10, 20, 40 or 60 milliseconds.
	 * @return discord_voice_client& Reference to self
	 * @throw dpp::voice_exception if voice support is not compiled into D++
	 */
	discord_voice_client& send_silence(const uint64_t duration);

	/**
	 * @brief Sets the audio type that will be sent with send_audio_* methods.
	 *
	 * @see send_audio_type_t
	 */
	discord_voice_client& set_send_audio_type(send_audio_type_t type);

	/**
	 * @brief Set the timescale in nanoseconds.
	 * 
	 * @param new_timescale Timescale to set. This defaults to 1000000,
	 * which means 1 millisecond.
	 * @return discord_voice_client& Reference to self
	 * @throw dpp::voice_exception If data length is invalid or voice support not compiled into D++
	 */
	discord_voice_client& set_timescale(uint64_t new_timescale);

	/**
	 * @brief Get the current timescale, this will default to 1000000
	 * which means 1 millisecond.
	 * 
	 * @return uint64_t timescale in nanoseconds
	 */
	uint64_t get_timescale();

	/**
	 * @brief Mark the voice connection as 'speaking'.
	 * This sends a JSON message to the voice websocket which tells discord
	 * that the user is speaking. The library automatically calls this for you
	 * whenever you send audio.
	 * 
	 * @return discord_voice_client& Reference to self
	 */
	discord_voice_client& speak();

	/**
	 * @brief Pause sending of audio
	 * 
	 * @param pause True to pause, false to resume
	 * @return reference to self
	 */
	discord_voice_client& pause_audio(bool pause);

	/**
	 * @brief Immediately stop all audio.
	 * Clears the packet queue.
	 * @return reference to self
	 */
	discord_voice_client& stop_audio();

	/**
	 * @brief Returns true if we are playing audio
	 * 
	 * @return true if audio is playing
	 */
	bool is_playing();

	/**
	 * @brief Get the number of seconds remaining
	 * of the audio output buffer
	 * 
	 * @return float number of seconds remaining 
	 */
	float get_secs_remaining();

	/**
	 * @brief Get the number of tracks remaining
	 * in the output buffer.
	 * This is calculated by the number of track
	 * markers plus one.
	 * @return uint32_t Number of tracks in the
	 * buffer
	 */
	uint32_t get_tracks_remaining();

	/**
	 * @brief Get the time remaining to send the
	 * audio output buffer in hours:minutes:seconds
	 * 
	 * @return dpp::utility::uptime length of buffer
	 */
	dpp::utility::uptime get_remaining();

	/**
	 * @brief Insert a track marker into the audio
	 * output buffer.
	 * A track marker is an arbitrary flag in the
	 * buffer contents that indicates the end of some
	 * block of audio of significance to the sender.
	 * This may be a song from a streaming site, or
	 * some voice audio/speech, a sound effect, or
	 * whatever you choose. You can later skip
	 * to the next marker using the
	 * dpp::discord_voice_client::skip_to_next_marker
	 * function.
	 * @param metadata Arbitrary information related to this
	 * track
	 * @return reference to self
	 */
	discord_voice_client& insert_marker(const std::string& metadata = "");

	/**
	 * @brief Skip tp the next track marker,
	 * previously inserted by using the
	 * dpp::discord_voice_client::insert_marker
	 * function. If there are no markers in the
	 * output buffer, then this skips to the end
	 * of the buffer and is equivalent to the
	 * dpp::discord_voice_client::stop_audio
	 * function.
	 * @note It is possible to use this function
	 * while the output stream is paused.
	 * @return reference to self
	 */
	discord_voice_client& skip_to_next_marker();

	/**
	 * @brief Get the metadata string associated with each inserted marker.
	 * 
	 * @return const std::vector<std::string>& list of metadata strings
	 */
	const std::vector<std::string> get_marker_metadata();

	/**
	 * @brief Returns true if the audio is paused.
	 * You can unpause with
	 * dpp::discord_voice_client::pause_audio.
	 * 
	 * @return true if paused
	 */
	bool is_paused();

	/**
	 * @brief Discord external IP detection.
	 * @return std::string Your external IP address
	 * @note This is a blocking operation that waits
	 * for a single packet from Discord's voice servers.
	 */
	std::string discover_ip();
};

};