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Fix decoder issues with RTP extensions

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Signed-off-by: Aron Heinecke <aron.heinecke@t-online.de>
This commit is contained in:
Aron Heinecke 2021-08-05 16:49:54 +02:00
parent 74377b4816
commit 7c77c2cd52
3 changed files with 45 additions and 17 deletions
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47
src/discord.rs
View file

@ -1,7 +1,11 @@
//! Discord handler
use std::sync::Arc;
use audiopus::{Channels, SampleRate};
use audiopus::coder::Decoder;
use serde::Deserialize;
use serenity::prelude::Mentionable;
use serenity::prelude::{Mentionable, Mutex};
use slog::error;
// This trait adds the `register_songbird` and `register_songbird_with` methods
@ -24,6 +28,8 @@ use serenity::{
model::{channel::Message, gateway::Ready},
Result as SerenityResult,
};
use songbird::packet::PacketSize;
use songbird::packet::rtp::RtpExtensionPacket;
use songbird::{
model::payload::{ClientConnect, ClientDisconnect, Speaking},
CoreEvent,
@ -330,6 +336,7 @@ fn check_msg(result: SerenityResult<Message>) {
struct Receiver{
sink: crate::AudioBufferDiscord,
decoder: Arc<Mutex<Decoder>>,
}
impl Receiver {
@ -338,6 +345,7 @@ impl Receiver {
// you can later store them in intervals.
Self {
sink: voice_receiver,
decoder: Arc::new(Mutex::new(Decoder::new(SampleRate::Hz48000, Channels::Stereo).unwrap()))
}
}
}
@ -383,19 +391,38 @@ impl VoiceEventHandler for Receiver {
// get raw opus package, we don't decode here and leave that to the AudioHandler
let last_bytes = packet.payload.len() - payload_end_pad;
let opus_slice = &packet.payload[*payload_offset..last_bytes];
let data = &packet.payload[*payload_offset..last_bytes];
let start = if packet.extension != 0 {
match RtpExtensionPacket::new(data) {
Some(v) => v.packet_size(),
None => {
eprintln!("Extension packet indicated, but insufficient space.");
return None;
}
}
} else {
0
};
let opus_slice = &data[start..];
let dur;
{
let time = std::time::Instant::now();
let mut lock = self.sink.lock().await;
dur = time.elapsed();
if let Err(e) = lock.handle_packet(packet.ssrc, packet.sequence.0.0, opus_slice.to_vec()) {
eprintln!("Failed to handle Discord voice packet: {}",e);
let mut lock_decoder = self.decoder.lock().await;
let mut decoded: [i16; 48000 *2 ] = [0; 48000 * 2];
if let Err(e) = lock_decoder.decode(Some(opus_slice), &mut decoded[..], false) {
eprintln!("Failed to handle Discord voice packet: {:?}",e);
} else {
let time = std::time::Instant::now();
let mut lock = self.sink.lock().await;
dur = time.elapsed();
if let Err(e) = lock.handle_packet(packet.ssrc, packet.sequence.0.0, opus_slice.to_vec()) {
eprintln!("Failed to handle Discord voice packet: {}",e);
}
if dur.as_millis() > 1 {
eprintln!("Acquiring lock took {}ms",dur.as_millis());
}
}
}
if dur.as_millis() > 1 {
eprintln!("Acquiring lock took {}ms",dur.as_millis());
}
},
Ctx::RtcpPacket {packet, payload_offset, payload_end_pad} => {
// An event which fires for every received rtcp packet,

6
src/discord_audiohandler.rs
View file

@ -17,7 +17,7 @@ use std::hash::Hash;
use audiopus::coder::Decoder;
use audiopus::{packet, Channels, SampleRate};
use slog::{debug, o, trace, warn, Logger};
use slog::{Logger, debug, info, o, trace, warn};
use tsclientlib::audio::Error;
use tsproto_packets::packets::{AudioData, CodecType, InAudioBuf};
@ -244,7 +244,7 @@ impl AudioQueue {
}
fn decode_packet(&mut self, packet: Option<&QueuePacket>, fec: bool) -> Result<()> {
trace!(self.logger, "Decoding packet"; "has_packet" => packet.is_some(), "fec" => fec);
debug!(self.logger, "Decoding packet"; "has_packet" => packet.is_some(), "fec" => fec);
let packet_data;
let len;
if let Some(p) = packet {
@ -347,7 +347,7 @@ impl AudioQueue {
cur_id
);
// Packet loss
debug!(self.logger, "Audio packet loss"; "need" => cur_id, "have" => packet.id);
info!(self.logger, "Audio packet loss"; "need" => cur_id, "have" => packet.id);
if packet.id == self.next_id {
// Can use forward-error-correction
self.decode_packet(Some(&packet), true)?;

9
src/main.rs
View file

@ -92,7 +92,7 @@ impl TypeMapKey for ListenerHolder {
/// teamspeak audio fragment timer
/// We want to run every 20ms, but we only get ~1ms correctness
const TICK_TIME: u64 = 18;
const TICK_TIME: u64 = 20;
const FRAME_SIZE_MS: usize = 20;
const SAMPLE_RATE: usize = 48000;
const STEREO_20MS: usize = SAMPLE_RATE * 2 * FRAME_SIZE_MS / 1000;
@ -104,6 +104,7 @@ const I16_CONVERSION_DIVIDER: f32 = 0x8000 as f32;
const MAX_OPUS_FRAME_SIZE: usize = 1275;
#[tokio::main]
async fn main() -> Result<()> {
dbg!(STEREO_20MS);
tracing_subscriber::fmt::init();
// init logging stuff used by tsclientlib
let config: Config = toml::from_str(&std::fs::read_to_string(".credentials.toml").unwrap()).unwrap();
@ -127,7 +128,7 @@ async fn main() -> Result<()> {
let songbird = Songbird::serenity();
songbird.set_config(
DriverConfig::default()
.decode_mode(DecodeMode::Decode)
.decode_mode(DecodeMode::Decrypt)
);
// init discord client
@ -255,12 +256,12 @@ async fn process_discord_audio(voice_buffer: &AudioBufferDiscord, encoder: &Arc<
// buffer_map = std::mem::replace(&mut *lock, HashMap::new());
// }
let mut data = [0.0; STEREO_20MS];
let mut data = [0.0; 1920];
{
let mut lock = voice_buffer.lock().await;
lock.fill_buffer(&mut data);
}
let mut encoded = [0; 1024];
let mut encoded = [0; 1920];
let encoder_c = encoder.clone();
// don't block the async runtime
let res = task::spawn_blocking(move || {