Discord -> teamspeak pipe without audio hearable
Teamspeak bot displays "talking" but nothing can be heard. Just piping opus 1:1 from discord to ts. Won't work for >1 person in a channel. Signed-off-by: Aron Heinecke <aron.heinecke@t-online.de>
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2 changed files with 231 additions and 16 deletions
183
src/discord.rs
183
src/discord.rs
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@ -1,11 +1,13 @@
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use std::env;
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use std::{env, sync::Arc, time::Duration};
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use serde::Deserialize;
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use tokio::sync::mpsc;
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use tsclientlib::{ClientId, Connection, DisconnectOptions, Identity, StreamItem};
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use tsproto_packets::packets::{AudioData, CodecType, OutAudio, OutPacket};
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use tsproto_packets::packets::{AudioData, CodecType, Direction, OutAudio, OutPacket};
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use audiopus::coder::Encoder;
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use futures::prelude::*;
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use sdl2::audio::{AudioCallback, AudioDevice, AudioSpec, AudioSpecDesired, AudioStatus};
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use sdl2::AudioSubsystem;
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use serenity::prelude::Mentionable;
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// This trait adds the `register_songbird` and `register_songbird_with` methods
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// to the client builder below, making it easy to install this voice client.
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@ -28,6 +30,17 @@ use serenity::{
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model::{channel::Message, gateway::Ready},
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Result as SerenityResult,
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};
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use songbird::{
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driver::{Config as DriverConfig, DecodeMode},
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model::payload::{ClientConnect, ClientDisconnect, Speaking},
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CoreEvent,
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Event,
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EventContext,
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EventHandler as VoiceEventHandler,
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Songbird,
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};
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use crate::ListenerHolder;
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pub(crate) struct Handler;
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@ -105,7 +118,51 @@ async fn join(ctx: &Context, msg: &Message) -> CommandResult {
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let manager = songbird::get(ctx).await
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.expect("Songbird Voice client placed in at initialisation.").clone();
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let _handler = manager.join(guild_id, connect_to).await;
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let (handler_lock, conn_result) = manager.join(guild_id, connect_to).await;
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if let Ok(_) = conn_result {
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// NOTE: this skips listening for the actual connection result.
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let channel: Arc<mpsc::Sender<OutPacket>>;
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{
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let data_read = ctx.data.read().await;
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channel = data_read.get::<ListenerHolder>().expect("Expected CommandCounter in TypeMap.").clone();
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}
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let mut handler = handler_lock.lock().await;
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handler.add_global_event(
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CoreEvent::SpeakingStateUpdate.into(),
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Receiver::new(channel.clone()),
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);
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handler.add_global_event(
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CoreEvent::SpeakingUpdate.into(),
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Receiver::new(channel.clone()),
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);
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handler.add_global_event(
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CoreEvent::VoicePacket.into(),
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Receiver::new(channel.clone()),
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);
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handler.add_global_event(
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CoreEvent::RtcpPacket.into(),
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Receiver::new(channel.clone()),
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);
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handler.add_global_event(
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CoreEvent::ClientConnect.into(),
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Receiver::new(channel.clone()),
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);
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handler.add_global_event(
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CoreEvent::ClientDisconnect.into(),
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Receiver::new(channel),
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);
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check_msg(msg.channel_id.say(&ctx.http, &format!("Joined {}", connect_to.mention())).await);
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} else {
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check_msg(msg.channel_id.say(&ctx.http, "Error joining the channel").await);
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}
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Ok(())
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}
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@ -273,3 +330,123 @@ fn check_msg(result: SerenityResult<Message>) {
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println!("Error sending message: {:?}", why);
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}
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}
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struct Receiver{
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sink: Arc<mpsc::Sender<OutPacket>>,
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}
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impl Receiver {
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pub fn new(voice_receiver: Arc<mpsc::Sender<OutPacket>>) -> Self {
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// You can manage state here, such as a buffer of audio packet bytes so
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// you can later store them in intervals.
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Self {
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sink: voice_receiver,
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}
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}
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}
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#[async_trait]
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impl VoiceEventHandler for Receiver {
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#[allow(unused_variables)]
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async fn act(&self, ctx: &EventContext<'_>) -> Option<Event> {
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use EventContext as Ctx;
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match ctx {
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Ctx::SpeakingStateUpdate(
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Speaking {speaking, ssrc, user_id, ..}
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) => {
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// Discord voice calls use RTP, where every sender uses a randomly allocated
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// *Synchronisation Source* (SSRC) to allow receivers to tell which audio
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// stream a received packet belongs to. As this number is not derived from
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// the sender's user_id, only Discord Voice Gateway messages like this one
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// inform us about which random SSRC a user has been allocated. Future voice
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// packets will contain *only* the SSRC.
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//
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// You can implement logic here so that you can differentiate users'
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// SSRCs and map the SSRC to the User ID and maintain this state.
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// Using this map, you can map the `ssrc` in `voice_packet`
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// to the user ID and handle their audio packets separately.
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println!(
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"Speaking state update: user {:?} has SSRC {:?}, using {:?}",
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user_id,
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ssrc,
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speaking,
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);
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},
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Ctx::SpeakingUpdate {ssrc, speaking} => {
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// You can implement logic here which reacts to a user starting
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// or stopping speaking.
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println!(
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"Source {} has {} speaking.",
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ssrc,
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if *speaking {"started"} else {"stopped"},
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);
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},
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Ctx::VoicePacket {audio, packet, payload_offset, payload_end_pad} => {
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// An event which fires for every received audio packet,
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// containing the decoded data.
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let data: &[u8] = &packet.payload.as_slice()[*payload_offset..(packet.payload.len()-payload_end_pad)];
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let packet = OutAudio::new(&AudioData::C2S { id: 0, codec: CodecType::OpusMusic, data });
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if let Err(e) = self.sink.send_timeout(packet, Duration::from_millis(10)).await {
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eprint!("Can't send voice to sender: {}",e);
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}
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if let Some(audio) = audio {
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// println!("Audio packet's first 5 samples: {:?}", audio.get(..5.min(audio.len())));
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// // println!(
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// // "Audio packet sequence {:05} has {:04} bytes (decompressed from {}), SSRC {}",
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// // packet.sequence.0,
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// // audio.len() * std::mem::size_of::<i16>(),
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// // packet.payload.len(),
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// // packet.ssrc,
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// // );
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// let mut values_converted = Vec::with_capacity(2*audio.len());
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// for value in audio {
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// // maybe "be" ?
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// // TODO: we could optimize this, data isn't directly used
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// values_converted.extend(&value.to_le_bytes());
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// }
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// let packet =
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// OutAudio::new(&AudioData::C2S { id: 0, codec: CodecType::OpusMusic, data: &values_converted });
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// if let Err(e) = self.sink.send_timeout(packet, Duration::from_millis(10)).await {
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// eprint!("Can't send voice to sender: {}",e);
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// }
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} else {
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println!("RTP packet, but no audio. Driver may not be configured to decode.");
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}
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},
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Ctx::RtcpPacket {packet, payload_offset, payload_end_pad} => {
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// An event which fires for every received rtcp packet,
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// containing the call statistics and reporting information.
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println!("RTCP packet received: {:?}", packet);
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},
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Ctx::ClientConnect(
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ClientConnect {audio_ssrc, video_ssrc, user_id, ..}
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) => {
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// You can implement your own logic here to handle a user who has joined the
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// voice channel e.g., allocate structures, map their SSRC to User ID.
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println!(
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"Client connected: user {:?} has audio SSRC {:?}, video SSRC {:?}",
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user_id,
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audio_ssrc,
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video_ssrc,
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);
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},
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Ctx::ClientDisconnect(
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ClientDisconnect {user_id, ..}
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) => {
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// You can implement your own logic here to handle a user who has left the
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// voice channel e.g., finalise processing of statistics etc.
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// You will typically need to map the User ID to their SSRC; observed when
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// speaking or connecting.
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println!("Client disconnected: user {:?}", user_id);
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},
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_ => {
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// We won't be registering this struct for any more event classes.
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unimplemented!()
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}
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}
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None
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}
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}
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64
src/main.rs
64
src/main.rs
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use std::env;
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use std::{collections::HashMap, env, sync::Arc};
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use serde::Deserialize;
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use tsclientlib::{ClientId, Connection, DisconnectOptions, Identity, StreamItem};
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use tsproto_packets::packets::{AudioData, CodecType, OutAudio, OutPacket};
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@ -20,10 +20,11 @@ struct ConnectionId(u64);
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// This trait adds the `register_songbird` and `register_songbird_with` methods
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// to the client builder below, making it easy to install this voice client.
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// The voice client can be retrieved in any command using `songbird::get(ctx).await`.
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use songbird::SerenityInit;
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use songbird::{SerenityInit, Songbird};
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use songbird::driver::{Config as DriverConfig, DecodeMode};
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// Import the `Context` to handle commands.
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use serenity::client::Context;
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use serenity::{client::Context, prelude::{RwLock, TypeMapKey}};
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use serenity::{
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async_trait,
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volume: f32,
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}
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struct ListenerHolder;
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impl TypeMapKey for ListenerHolder {
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type Value = Arc<mpsc::Sender<OutPacket>>;
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}
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#[tokio::main]
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async fn main() -> Result<()> {
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tracing_subscriber::fmt::init();
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.prefix("~"))
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.group(&discord::GENERAL_GROUP);
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// Here, we need to configure Songbird to decode all incoming voice packets.
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// If you want, you can do this on a per-call basis---here, we need it to
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// read the audio data that other people are sending us!
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let songbird = Songbird::serenity();
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songbird.set_config(
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DriverConfig::default()
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.decode_mode(DecodeMode::Decode)
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);
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let mut client = Client::builder(&config.discord_token)
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.event_handler(discord::Handler)
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.framework(framework)
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.register_songbird()
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.register_songbird_with(songbird.into())
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.await
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.expect("Err creating client");
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let (tx,mut rx) = mpsc::channel(50);
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let voice_pipes: Arc<mpsc::Sender<OutPacket>> = Arc::new(tx);
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{
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// Open the data lock in write mode, so keys can be inserted to it.
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let mut data = client.data.write().await;
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// The CommandCounter Value has the following type:
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// Arc<RwLock<HashMap<String, u64>>>
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// So, we have to insert the same type to it.
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data.insert::<ListenerHolder>(voice_pipes);
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}
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tokio::spawn(async move {
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let _ = client.start().await.map_err(|why| println!("Client ended: {:?}", why));
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});
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r?;
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}
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let (send, mut recv) = mpsc::channel(5);
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{
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let mut a2t = audiodata.a2ts.lock().unwrap();
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a2t.set_listener(send);
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a2t.set_volume(config.volume);
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a2t.set_playing(true);
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}
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// let (send, mut recv) = mpsc::channel(5);
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// {
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// let mut a2t = audiodata.a2ts.lock().unwrap();
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// a2t.set_listener(send);
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// a2t.set_volume(config.volume);
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// a2t.set_playing(true);
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// }
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loop {
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let t2a = audiodata.ts2a.clone();
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@ -134,7 +163,15 @@ async fn main() -> Result<()> {
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// Wait for ctrl + c
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tokio::select! {
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send_audio = recv.recv() => {
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// send_audio = recv.recv() => {
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// if let Some(packet) = send_audio {
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// con.send_audio(packet)?;
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// } else {
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// info!(logger, "Audio sending stream was canceled");
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// break;
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// }
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// }
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send_audio = rx.recv() => {
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if let Some(packet) = send_audio {
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con.send_audio(packet)?;
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} else {
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@ -149,9 +186,10 @@ async fn main() -> Result<()> {
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}
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};
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}
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println!("Disconnecting");
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// Disconnect
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con.disconnect(DisconnectOptions::new())?;
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con.events().for_each(|_| future::ready(())).await;
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println!("Disconnected");
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Ok(())
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}
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