Driver/receive: Implement audio reorder/jitter buffer (#156)

This PR Introduces a new `VoiceTick` event which collects and reorders all RTP packets to smooth over network instability, as well as to synchronise user audio streams. Raw packet events have been moved to `RtpPacket`, while `SpeakingUpdate`s have been removed as they can be easily computed using the `silent`/`speaking` audio maps included in each event.

Closes #146.
This commit is contained in:
Kyle Simpson
2023-01-09 00:22:30 +00:00
parent ab18f9e092
commit c60c454cf5
19 changed files with 923 additions and 611 deletions

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@@ -6,13 +6,13 @@ mod disconnect;
#[cfg(feature = "receive")]
mod rtcp;
#[cfg(feature = "receive")]
mod speaking;
mod rtp;
#[cfg(feature = "receive")]
mod voice;
#[cfg(feature = "receive")]
use discortp::{rtcp::Rtcp, rtp::Rtp};
use bytes::Bytes;
pub use self::{connect::*, disconnect::*};
#[cfg(feature = "receive")]
pub use self::{rtcp::*, speaking::*, voice::*};
pub use self::{rtcp::*, rtp::*, voice::*};

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@@ -1,3 +1,5 @@
use discortp::rtcp::RtcpPacket;
use super::*;
#[derive(Clone, Debug, Eq, PartialEq)]
@@ -5,11 +7,22 @@ use super::*;
/// Telemetry/statistics packet, received from another stream (detailed in `packet`).
/// `payload_offset` contains the true payload location within the raw packet's `payload()`,
/// to allow manual decoding of `Rtcp` packet bodies.
pub struct RtcpData<'a> {
pub struct RtcpData {
/// Raw RTCP packet data.
pub packet: &'a Rtcp,
pub packet: Bytes,
/// Byte index into the packet body (after headers) for where the payload begins.
pub payload_offset: usize,
/// Number of bytes at the end of the packet to discard.
pub payload_end_pad: usize,
}
impl RtcpData {
/// Create a zero-copy view of the inner RTCP packet.
///
/// This allows easy access to packet header fields, taking them from the underlying
/// `Bytes` as needed while handling endianness etc.
pub fn rtcp(&'_ self) -> RtcpPacket<'_> {
RtcpPacket::new(&self.packet)
.expect("FATAL: leaked illegally small RTP packet from UDP Rx task.")
}
}

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@@ -0,0 +1,30 @@
use discortp::rtp::RtpPacket;
use super::*;
#[derive(Clone, Debug, Eq, PartialEq)]
#[non_exhaustive]
/// Opus audio packet, received from another stream (detailed in `packet`).
/// `payload_offset` contains the true payload location within the raw packet's `payload()`,
/// if extensions or raw packet data are required.
pub struct RtpData {
/// Raw RTP packet data.
///
/// Includes the SSRC (i.e., sender) of this packet.
pub packet: Bytes,
/// Byte index into the packet body (after headers) for where the payload begins.
pub payload_offset: usize,
/// Number of bytes at the end of the packet to discard.
pub payload_end_pad: usize,
}
impl RtpData {
/// Create a zero-copy view of the inner RTP packet.
///
/// This allows easy access to packet header fields, taking them from the underlying
/// `Bytes` as needed while handling endianness etc.
pub fn rtp(&'_ self) -> RtpPacket<'_> {
RtpPacket::new(&self.packet)
.expect("FATAL: leaked illegally small RTP packet from UDP Rx task.")
}
}

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@@ -1,14 +0,0 @@
#[derive(Clone, Debug, Eq, Hash, PartialEq)]
#[non_exhaustive]
/// Speaking state transition, describing whether a given source has started/stopped
/// transmitting. This fires in response to a silent burst, or the first packet
/// breaking such a burst.
pub struct SpeakingUpdateData {
/// Whether this user is currently speaking.
pub speaking: bool,
/// Synchronisation Source of the user who has begun speaking.
///
/// This must be combined with another event class to map this back to
/// its original UserId.
pub ssrc: u32,
}

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@@ -1,28 +1,38 @@
use std::collections::{HashMap, HashSet};
use super::*;
#[derive(Clone, Debug, Eq, PartialEq)]
#[non_exhaustive]
/// Opus audio packet, received from another stream (detailed in `packet`).
/// `payload_offset` contains the true payload location within the raw packet's `payload()`,
/// if extensions or raw packet data are required.
/// Audio data from all users in a voice channel, fired every 20ms.
///
/// Valid audio data (`Some(audio)` where `audio.len >= 0`) contains up to 20ms of 16-bit stereo PCM audio
/// at 48kHz, using native endianness. Songbird will not send audio for silent regions, these should
/// be inferred using [`SpeakingUpdate`]s (and filled in by the user if required using arrays of zeroes).
/// Songbird implements a jitter buffer to sycnhronise user packets, smooth out network latency, and
/// handle packet reordering by the network. Packet playout via this event is delayed by approximately
/// [`Config::playout_buffer_length`]` * 20ms` from its original arrival.
///
/// If `audio.len() == 0`, then this packet arrived out-of-order. If `None`, songbird was not configured
/// to decode received packets.
///
/// [`SpeakingUpdate`]: crate::events::CoreEvent::SpeakingUpdate
pub struct VoiceData<'a> {
/// Decoded audio from this packet.
pub audio: &'a Option<Vec<i16>>,
/// Raw RTP packet data.
///
/// Includes the SSRC (i.e., sender) of this packet.
pub packet: &'a Rtp,
/// Byte index into the packet body (after headers) for where the payload begins.
pub payload_offset: usize,
/// Number of bytes at the end of the packet to discard.
pub payload_end_pad: usize,
/// [`Config::playout_buffer_length`]: crate::Config::playout_buffer_length
pub struct VoiceTick {
/// Decoded voice data and source packets sent by each user.
pub speaking: HashMap<u32, VoiceData>,
/// Set of all SSRCs currently known in the call who aren't included in [`Self::speaking`].
pub silent: HashSet<u32>,
}
#[derive(Clone, Debug, Eq, PartialEq)]
#[non_exhaustive]
/// Voice packet and audio data for a single user, from a single tick.
pub struct VoiceData {
/// RTP packet clocked out for this tick.
///
/// If `None`, then the packet was lost, and [`Self::decoded_voice`] may include
/// around one codec delay's worth of audio.
pub packet: Option<RtpData>,
/// PCM audio obtained from a user.
///
/// Valid audio data (`Some(audio)` where `audio.len >= 0`) typically contains 20ms of 16-bit stereo PCM audio
/// at 48kHz, using native endianness. Channels are interleaved (i.e., `L, R, L, R, ...`).
///
/// This value will be `None` if Songbird is not configured to decode audio.
pub decoded_voice: Option<Vec<i16>>,
}

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@@ -41,53 +41,36 @@ impl<'a> From<&'a InternalDisconnect> for DisconnectData<'a> {
#[cfg(feature = "receive")]
mod receive {
use super::*;
use discortp::{rtcp::Rtcp, rtp::Rtp};
#[derive(Clone, Debug, Eq, Hash, PartialEq)]
pub struct InternalSpeakingUpdate {
pub ssrc: u32,
pub speaking: bool,
}
use bytes::Bytes;
#[derive(Clone, Debug, Eq, PartialEq)]
pub struct InternalVoicePacket {
pub audio: Option<Vec<i16>>,
pub packet: Rtp,
pub struct InternalRtpPacket {
pub packet: Bytes,
pub payload_offset: usize,
pub payload_end_pad: usize,
}
#[derive(Clone, Debug, Eq, PartialEq)]
pub struct InternalRtcpPacket {
pub packet: Rtcp,
pub packet: Bytes,
pub payload_offset: usize,
pub payload_end_pad: usize,
}
impl<'a> From<&'a InternalSpeakingUpdate> for SpeakingUpdateData {
fn from(val: &'a InternalSpeakingUpdate) -> Self {
impl<'a> From<&'a InternalRtpPacket> for RtpData {
fn from(val: &'a InternalRtpPacket) -> Self {
Self {
speaking: val.speaking,
ssrc: val.ssrc,
}
}
}
impl<'a> From<&'a InternalVoicePacket> for VoiceData<'a> {
fn from(val: &'a InternalVoicePacket) -> Self {
Self {
audio: &val.audio,
packet: &val.packet,
packet: val.packet.clone(),
payload_offset: val.payload_offset,
payload_end_pad: val.payload_end_pad,
}
}
}
impl<'a> From<&'a InternalRtcpPacket> for RtcpData<'a> {
impl<'a> From<&'a InternalRtcpPacket> for RtcpData {
fn from(val: &'a InternalRtcpPacket) -> Self {
Self {
packet: &val.packet,
packet: val.packet.clone(),
payload_offset: val.payload_offset,
payload_end_pad: val.payload_end_pad,
}

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@@ -33,18 +33,16 @@ pub enum EventContext<'a> {
SpeakingStateUpdate(Speaking),
#[cfg(feature = "receive")]
/// Speaking state transition, describing whether a given source has started/stopped
/// transmitting. This fires in response to a silent burst, or the first packet
/// breaking such a burst.
SpeakingUpdate(SpeakingUpdateData),
/// Reordered and decoded audio packets, received every 20ms.
VoiceTick(VoiceTick),
#[cfg(feature = "receive")]
/// Opus audio packet, received from another stream.
VoicePacket(VoiceData<'a>),
RtpPacket(RtpData),
#[cfg(feature = "receive")]
/// Telemetry/statistics packet, received from another stream.
RtcpPacket(RtcpData<'a>),
RtcpPacket(RtcpData),
/// Fired whenever a client disconnects.
ClientDisconnect(ClientDisconnect),
@@ -63,9 +61,9 @@ pub enum EventContext<'a> {
pub enum CoreContext {
SpeakingStateUpdate(Speaking),
#[cfg(feature = "receive")]
SpeakingUpdate(InternalSpeakingUpdate),
VoiceTick(VoiceTick),
#[cfg(feature = "receive")]
VoicePacket(InternalVoicePacket),
RtpPacket(InternalRtpPacket),
#[cfg(feature = "receive")]
RtcpPacket(InternalRtcpPacket),
ClientDisconnect(ClientDisconnect),
@@ -79,10 +77,9 @@ impl<'a> CoreContext {
match self {
Self::SpeakingStateUpdate(evt) => EventContext::SpeakingStateUpdate(*evt),
#[cfg(feature = "receive")]
Self::SpeakingUpdate(evt) =>
EventContext::SpeakingUpdate(SpeakingUpdateData::from(evt)),
Self::VoiceTick(evt) => EventContext::VoiceTick(evt.clone()),
#[cfg(feature = "receive")]
Self::VoicePacket(evt) => EventContext::VoicePacket(VoiceData::from(evt)),
Self::RtpPacket(evt) => EventContext::RtpPacket(RtpData::from(evt)),
#[cfg(feature = "receive")]
Self::RtcpPacket(evt) => EventContext::RtcpPacket(RtcpData::from(evt)),
Self::ClientDisconnect(evt) => EventContext::ClientDisconnect(*evt),
@@ -102,9 +99,9 @@ impl EventContext<'_> {
match self {
Self::SpeakingStateUpdate(_) => Some(CoreEvent::SpeakingStateUpdate),
#[cfg(feature = "receive")]
Self::SpeakingUpdate(_) => Some(CoreEvent::SpeakingUpdate),
Self::VoiceTick(_) => Some(CoreEvent::VoiceTick),
#[cfg(feature = "receive")]
Self::VoicePacket(_) => Some(CoreEvent::VoicePacket),
Self::RtpPacket(_) => Some(CoreEvent::RtpPacket),
#[cfg(feature = "receive")]
Self::RtcpPacket(_) => Some(CoreEvent::RtcpPacket),
Self::ClientDisconnect(_) => Some(CoreEvent::ClientDisconnect),

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@@ -9,14 +9,11 @@
/// when a client leaves the session ([`ClientDisconnect`]).
///
/// When the `"receive"` feature is enabled, songbird can also handle voice packets
#[cfg_attr(feature = "receive", doc = "([`VoicePacket`](Self::VoicePacket)),")]
#[cfg_attr(not(feature = "receive"), doc = "(`VoicePacket`),")]
/// detect speech starting/stopping
#[cfg_attr(
feature = "receive",
doc = "([`SpeakingUpdate`](Self::SpeakingUpdate)),"
)]
#[cfg_attr(not(feature = "receive"), doc = "(`SpeakingUpdate`),")]
#[cfg_attr(feature = "receive", doc = "([`RtpPacket`](Self::RtpPacket)),")]
#[cfg_attr(not(feature = "receive"), doc = "(`RtpPacket`),")]
/// decode and track speaking users
#[cfg_attr(feature = "receive", doc = "([`VoiceTick`](Self::VoiceTick)),")]
#[cfg_attr(not(feature = "receive"), doc = "(`VoiceTick`),")]
/// and handle telemetry data
#[cfg_attr(feature = "receive", doc = "([`RtcpPacket`](Self::RtcpPacket)).")]
#[cfg_attr(not(feature = "receive"), doc = "(`RtcpPacket`).")]
@@ -49,9 +46,9 @@ pub enum CoreEvent {
SpeakingStateUpdate,
#[cfg(feature = "receive")]
/// Fires when a source starts speaking, or stops speaking
/// (*i.e.*, 5 consecutive silent frames).
SpeakingUpdate,
/// Fires every 20ms, containing the scheduled voice packet and decoded audio
/// data for each live user.
VoiceTick,
#[cfg(feature = "receive")]
/// Fires on receipt of a voice packet from another stream in the voice call.
@@ -59,7 +56,7 @@ pub enum CoreEvent {
/// As RTP packets do not map to Discord's notion of users, SSRCs must be mapped
/// back using the user IDs seen through client connection, disconnection,
/// or speaking state update.
VoicePacket,
RtpPacket,
#[cfg(feature = "receive")]
/// Fires on receipt of an RTCP packet, containing various call stats