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Make our "loss based control" algorithm closer to what is defined in [GCC]

Browse source 
As specified in Google Congestion Control we should run the packet loss
estimation algorithm "every time feedback from the receiver is
received".

And, also as defined by GCC, we now have 2 different estimated bitrates,
one for the delay-based controller value and one for the loss-based one,
and we use the minimum value between those 2 as our current estimation.

[GCC]: https://datatracker.ietf.org/doc/html/draft-ietf-rmcat-gcc-02
This commit is contained in:
Thibault Saunier 2022-03-30 15:39:19 +00:00 committed by Mathieu Duponchelle
parent 3c81afa7b2
commit 075a625305
3 changed files with 300 additions and 150 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

7
Cargo.lock generated
View file

@ -903,6 +903,12 @@ version = "1.6.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "9100414882e15fb7feccb4897e5f0ff0ff1ca7d1a86a23208ada4d7a18e6c6c4"
[[package]]
name = "human_bytes"
version = "0.3.1"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "88a0d4dc39ec942e44c1c306aa196da67f2bd6a30dc7b4a475465c13ccf28817"
[[package]]
name = "idna"
version = "0.2.3"
@ -1847,6 +1853,7 @@ dependencies = [
"gstreamer-sdp",
"gstreamer-video",
"gstreamer-webrtc",
"human_bytes",
"once_cell",
"serde",
"serde_json",

1
plugins/Cargo.toml
View file

@ -27,6 +27,7 @@ serde = "1"
serde_json = "1"
fastrand = "1.0"
webrtcsink-protocol = { version = "0.1", path="../protocol" }
human_bytes = "0.3.1"
[dev-dependencies]
tracing = { version = "0.1", features = ["log"] }

388
plugins/src/webrtcsink/imp.rs
View file

@ -126,14 +126,21 @@ struct VideoEncoder {
}
struct CongestionController {
/// Overall bitrate target for all video streams.
/// Note: The target bitrate applied is the min of
/// target_bitrate_on_delay and target_bitrate_on_loss
///
/// Bitrate target based on delay factor for all video streams.
/// Hasn't been tested with multiple video streams, but
/// current design is simply to divide bitrate equally.
bitrate_ema: Option<f64>,
target_bitrate_on_delay: i32,
/// Bitrate target based on loss for all video streams.
target_bitrate_on_loss: i32,
/// Exponential moving average, updated when bitrate is
/// decreased, discarded when increased again past last
/// congestion window. Smoothing factor hardcoded.
target_bitrate: i32,
bitrate_ema: Option<f64>,
/// Exponentially weighted moving variance, recursively
/// updated along with bitrate_ema. sqrt'd to obtain standard
/// deviation, used to determine whether to increase bitrate
@ -162,11 +169,23 @@ enum CongestionControlOp {
/// Don't update target bitrate
Hold,
/// Decrease target bitrate
Decrease(f64),
Decrease {
factor: f64,
#[allow(dead_code)]
reason: String, // for Debug
},
/// Increase target bitrate, either additively or multiplicatively
Increase(IncreaseType),
}
#[derive(Debug, Clone, Copy)]
enum ControllerType {
// Running the "delay-based controller"
Delay,
// Running the "loss based controller"
Loss,
}
struct Consumer {
pipeline: gst::Pipeline,
webrtcbin: gst::Element,
@ -179,6 +198,8 @@ struct Consumer {
stats: gst::Structure,
max_bitrate: u32,
stats_sigid: Option<glib::SignalHandlerId>,
}
#[derive(PartialEq)]
@ -215,6 +236,7 @@ fn create_navigation_event<N: IsA<gst_video::Navigation>>(sink: &N, msg: &str) {
gst_error!(CAT, "Invalid navigation event: {:?}", msg);
}
}
/// Simple utility for tearing down a pipeline cleanly
struct PipelineWrapper(gst::Pipeline);
@ -465,20 +487,6 @@ fn setup_encoding(
Ok((enc, conv_filter, pay))
}
fn lookup_remote_inbound_rtp_stats(stats: &gst::StructureRef) -> Option<gst::Structure> {
for (_, field_value) in stats {
if let Ok(s) = field_value.get::<gst::Structure>() {
if let Ok(type_) = s.get::<gst_webrtc::WebRTCStatsType>("type") {
if type_ == gst_webrtc::WebRTCStatsType::RemoteInboundRtp {
return Some(s);
}
}
}
}
None
}
fn lookup_transport_stats(stats: &gst::StructureRef) -> Option<gst::Structure> {
for (_, field_value) in stats {
if let Ok(s) = field_value.get::<gst::Structure>() {
@ -627,7 +635,8 @@ impl VideoEncoder {
impl CongestionController {
fn new(peer_id: &str, min_bitrate: u32, max_bitrate: u32) -> Self {
Self {
target_bitrate: 0,
target_bitrate_on_delay: 0,
target_bitrate_on_loss: 0,
bitrate_ema: None,
bitrate_emvar: 0.,
last_update_time: None,
@ -637,19 +646,21 @@ impl CongestionController {
}
}
fn update(
fn update_delay(
&mut self,
element: &super::WebRTCSink,
twcc_stats: &gst::StructureRef,
rtt: f64,
) -> CongestionControlOp {
let target_bitrate = self.target_bitrate as f64;
let target_bitrate = f64::min(
self.target_bitrate_on_delay as f64,
self.target_bitrate_on_loss as f64,
);
// Unwrap, all those fields must be there or there's been an API
// break, which qualifies as programming error
let bitrate_sent = twcc_stats.get::<u32>("bitrate-sent").unwrap();
let bitrate_recv = twcc_stats.get::<u32>("bitrate-recv").unwrap();
let delta_of_delta = twcc_stats.get::<i64>("avg-delta-of-delta").unwrap();
let loss_percentage = twcc_stats.get::<f64>("packet-loss-pct").unwrap();
let sent_minus_received = bitrate_sent.saturating_sub(bitrate_recv);
@ -665,61 +676,22 @@ impl CongestionController {
);
if delay_factor > 0.1 {
CongestionControlOp::Decrease(if delay_factor < 0.64 {
gst_trace!(
CAT,
obj: element,
"consumer {}: low delay factor {}",
self.peer_id,
delay_factor,
);
0.96
let (factor, reason) = if delay_factor < 0.64 {
(0.96, format!("low delay factor {}", delay_factor))
} else {
gst_trace!(
CAT,
obj: element,
"consumer {}: High delay factor",
self.peer_id
);
delay_factor.sqrt().sqrt().clamp(0.8, 0.96)
})
} else if delta_of_delta > 1000000 {
CongestionControlOp::Decrease(if loss_percentage < 10. {
gst_trace!(
CAT,
obj: element,
"consumer {}: moderate loss high delta",
self.peer_id
);
0.97
} else {
gst_log!(
CAT,
obj: element,
"consumer: {}: high loss high delta",
self.peer_id
);
((100. - loss_percentage) / 100.).clamp(0.7, 0.98)
})
} else if loss_percentage > 10. {
CongestionControlOp::Decrease(
((100. - (0.5 * loss_percentage)) / 100.).clamp(0.7, 0.98),
(
delay_factor.sqrt().sqrt().clamp(0.8, 0.96),
format!("High delay factor {}", delay_factor),
)
} else if loss_percentage > 2. {
gst_trace!(
CAT,
obj: element,
"consumer {}: moderate loss",
self.peer_id
);
CongestionControlOp::Hold
};
CongestionControlOp::Decrease { factor, reason }
} else if delta_of_delta > 1_000_000 {
CongestionControlOp::Decrease {
factor: 0.97,
reason: format!("High delta: {}", delta_of_delta),
}
} else {
gst_trace!(
CAT,
obj: element,
"consumer {}: no detected congestion",
self.peer_id
);
CongestionControlOp::Increase(if let Some(ema) = self.bitrate_ema {
let bitrate_stdev = self.bitrate_emvar.sqrt();
@ -800,30 +772,107 @@ impl CongestionController {
}
}
fn clamp_bitrate(&mut self, bitrate: i32, n_encoders: i32) {
self.target_bitrate = bitrate.clamp(
fn clamp_bitrate(&mut self, bitrate: i32, n_encoders: i32, controller_type: ControllerType) {
match controller_type {
ControllerType::Loss => {
self.target_bitrate_on_loss = bitrate.clamp(
self.min_bitrate as i32 * n_encoders,
self.max_bitrate as i32 * n_encoders,
);
)
}
fn control(
ControllerType::Delay => {
self.target_bitrate_on_delay = bitrate.clamp(
self.min_bitrate as i32 * n_encoders,
self.max_bitrate as i32 * n_encoders,
)
}
}
}
fn get_remote_inbound_stats(&self, stats: &gst::StructureRef) -> Vec<gst::Structure> {
let mut inbound_rtp_stats: Vec<gst::Structure> = Default::default();
for (_, field_value) in stats {
if let Ok(s) = field_value.get::<gst::Structure>() {
if let Ok(type_) = s.get::<gst_webrtc::WebRTCStatsType>("type") {
if type_ == gst_webrtc::WebRTCStatsType::RemoteInboundRtp {
inbound_rtp_stats.push(s);
}
}
}
}
inbound_rtp_stats
}
fn lookup_rtt(&self, stats: &gst::StructureRef) -> f64 {
let inbound_rtp_stats = self.get_remote_inbound_stats(stats);
let mut rtt = 0.;
let mut n_rtts = 0u64;
for inbound_stat in &inbound_rtp_stats {
if let Err(err) = (|| -> Result<(), gst::structure::GetError> {
rtt += inbound_stat.get::<f64>("round-trip-time")?;
n_rtts += 1;
Ok(())
})() {
gst_debug!(CAT, "{:?}", err);
}
}
rtt /= f64::max(1., n_rtts as f64);
gst_log!(CAT, "Round trip time: {}", rtt);
rtt
}
fn loss_control(
&mut self,
element: &super::WebRTCSink,
stats: &gst::StructureRef,
encoders: &mut Vec<VideoEncoder>,
) {
let n_encoders = encoders.len() as i32;
let loss_percentage = stats.get::<f64>("packet-loss-pct").unwrap();
let rtt = lookup_remote_inbound_rtp_stats(stats)
.and_then(|s| s.get::<f64>("round-trip-time").ok())
.unwrap_or(0.);
self.apply_control_op(
element,
encoders,
if loss_percentage > 10. {
CongestionControlOp::Decrease {
factor: ((100. - (0.5 * loss_percentage)) / 100.).clamp(0.7, 0.98),
reason: format!("High loss: {}", loss_percentage),
}
} else if loss_percentage > 2. {
CongestionControlOp::Hold
} else {
CongestionControlOp::Increase(IncreaseType::Multiplicative(1.05))
},
ControllerType::Loss,
);
}
fn delay_control(
&mut self,
element: &super::WebRTCSink,
stats: &gst::StructureRef,
encoders: &mut Vec<VideoEncoder>,
) {
if let Some(twcc_stats) = lookup_transport_stats(stats).and_then(|transport_stats| {
transport_stats.get::<gst::Structure>("gst-twcc-stats").ok()
}) {
let control_op = self.update(element, &twcc_stats, rtt);
let op = self.update_delay(element, &twcc_stats, self.lookup_rtt(stats));
self.apply_control_op(element, encoders, op, ControllerType::Delay);
}
}
fn apply_control_op(
&mut self,
element: &super::WebRTCSink,
encoders: &mut Vec<VideoEncoder>,
control_op: CongestionControlOp,
controller_type: ControllerType,
) {
gst_trace!(
CAT,
obj: element,
@ -832,32 +881,62 @@ impl CongestionController {
control_op
);
match control_op {
CongestionControlOp::Hold => (),
let n_encoders = encoders.len() as i32;
let prev_bitrate = i32::min(self.target_bitrate_on_delay, self.target_bitrate_on_loss);
match &control_op {
CongestionControlOp::Hold => {}
CongestionControlOp::Increase(IncreaseType::Additive(value)) => {
self.clamp_bitrate(self.target_bitrate + value as i32, n_encoders);
self.clamp_bitrate(
self.target_bitrate_on_delay + *value as i32,
n_encoders,
controller_type,
);
}
CongestionControlOp::Increase(IncreaseType::Multiplicative(factor)) => {
self.clamp_bitrate((self.target_bitrate as f64 * factor) as i32, n_encoders);
self.clamp_bitrate(
(self.target_bitrate_on_delay as f64 * factor) as i32,
n_encoders,
controller_type,
);
}
CongestionControlOp::Decrease(factor) => {
self.clamp_bitrate((self.target_bitrate as f64 * factor) as i32, n_encoders);
CongestionControlOp::Decrease { factor, .. } => {
self.clamp_bitrate(
(self.target_bitrate_on_delay as f64 * factor) as i32,
n_encoders,
controller_type,
);
if let ControllerType::Delay = controller_type {
// Smoothing factor
let alpha = 0.75;
if let Some(ema) = self.bitrate_ema {
let sigma: f64 = (self.target_bitrate as f64) - ema;
let sigma: f64 = (self.target_bitrate_on_delay as f64) - ema;
self.bitrate_ema = Some(ema + (alpha * sigma));
self.bitrate_emvar =
(1. - alpha) * (self.bitrate_emvar + alpha * sigma.powi(2));
} else {
self.bitrate_ema = Some(self.target_bitrate as f64);
self.bitrate_ema = Some(self.target_bitrate_on_delay as f64);
self.bitrate_emvar = 0.;
}
}
}
}
let target_bitrate = self.target_bitrate / n_encoders;
let target_bitrate =
i32::min(self.target_bitrate_on_delay, self.target_bitrate_on_loss).clamp(
self.min_bitrate as i32 * n_encoders,
self.max_bitrate as i32 * n_encoders,
) / n_encoders;
if target_bitrate != prev_bitrate {
gst_info!(
CAT,
"{:?} {} => {}",
control_op,
human_bytes::human_bytes(prev_bitrate),
human_bytes::human_bytes(target_bitrate)
);
}
let fec_ratio = {
if target_bitrate <= 2000000 || self.max_bitrate <= 2000000 {
@ -876,7 +955,6 @@ impl CongestionController {
.set_property("fec-percentage", fec_percentage);
}
}
}
}
impl State {
@ -953,6 +1031,27 @@ impl State {
}
impl Consumer {
fn new(
pipeline: gst::Pipeline,
webrtcbin: gst::Element,
peer_id: String,
congestion_controller: Option<CongestionController>,
max_bitrate: u32,
) -> Self {
Self {
pipeline,
webrtcbin,
peer_id,
congestion_controller,
max_bitrate,
sdp: None,
stats: gst::Structure::new_empty("application/x-webrtc-stats"),
webrtc_pads: HashMap::new(),
encoders: Vec::new(),
stats_sigid: None,
}
}
fn gather_stats(&self) -> gst::Structure {
let mut ret = self.stats.to_owned();
@ -1140,7 +1239,9 @@ impl Consumer {
);
if let Some(congestion_controller) = self.congestion_controller.as_mut() {
congestion_controller.target_bitrate += enc.bitrate();
congestion_controller.target_bitrate_on_delay += enc.bitrate();
congestion_controller.target_bitrate_on_loss =
congestion_controller.target_bitrate_on_delay;
enc.transceiver.set_property("fec-percentage", 0u32);
} else {
/* If congestion control is disabled, we simply use the highest
@ -1658,12 +1759,11 @@ impl WebRTCSink {
}
});
let mut consumer = Consumer {
pipeline: pipeline.clone(),
webrtcbin: webrtcbin.clone(),
webrtc_pads: HashMap::new(),
peer_id: peer_id.to_string(),
congestion_controller: match settings.cc_heuristic {
let mut consumer = Consumer::new(
pipeline.clone(),
webrtcbin.clone(),
peer_id.to_string(),
match settings.cc_heuristic {
WebRTCSinkCongestionControl::Disabled => None,
WebRTCSinkCongestionControl::Homegrown => Some(CongestionController::new(
peer_id,
@ -1671,11 +1771,39 @@ impl WebRTCSink {
settings.max_bitrate,
)),
},
encoders: Vec::new(),
sdp: None,
stats: gst::Structure::new_empty("application/x-webrtc-stats"),
max_bitrate: settings.max_bitrate,
};
settings.max_bitrate,
);
let rtpbin = webrtcbin
.dynamic_cast_ref::<gst::ChildProxy>()
.unwrap()
.child_by_name("rtpbin")
.unwrap();
if consumer.congestion_controller.is_some() {
let peer_id_str = peer_id.to_string();
if consumer.stats_sigid.is_none() {
consumer.stats_sigid = Some(rtpbin.connect_closure("on-new-ssrc", true,
glib::closure!(@weak-allow-none element, @weak-allow-none webrtcbin
=> move |rtpbin: gst::Object, session_id: u32, _src: u32| {
let session = rtpbin.emit_by_name::<gst::Element>("get-session", &[&session_id]);
let element = element.expect("on-new-ssrc emited when webrtcsink has been disposed?");
let webrtcbin = webrtcbin.unwrap();
let mut state = element.imp().state.lock().unwrap();
if let Some(mut consumer) = state.consumers.get_mut(&peer_id_str) {
consumer.stats_sigid = Some(session.connect_notify(Some("twcc-stats"),
glib::clone!(@strong peer_id_str, @weak webrtcbin, @weak element => @default-panic, move |sess, pspec| {
// Run the Loss-based control algortithm on new peer TWCC feedbacks
element.imp().process_loss_stats(&element, &peer_id_str, &sess.property::<gst::Structure>(pspec.name()));
})
));
}
})
));
}
}
state
.streams
@ -1805,22 +1933,41 @@ impl WebRTCSink {
Ok(())
}
fn process_webrtcbin_stats(
fn process_loss_stats(
&self,
element: &super::WebRTCSink,
peer_id: &str,
stats: &gst::StructureRef,
stats: &gst::Structure,
) {
let mut state = self.state.lock().unwrap();
if let Some(consumer) = state.consumers.get_mut(peer_id) {
let mut state = element.imp().state.lock().unwrap();
if let Some(mut consumer) = state.consumers.get_mut(peer_id) {
if let Some(congestion_controller) = consumer.congestion_controller.as_mut() {
congestion_controller.control(element, stats, &mut consumer.encoders);
congestion_controller.loss_control(&element, stats, &mut consumer.encoders);
}
consumer.stats = stats.to_owned();
}
}
fn process_stats(&self, element: &super::WebRTCSink, webrtcbin: gst::Element, peer_id: &str) {
let peer_id = peer_id.to_string();
let promise = gst::Promise::with_change_func(
glib::clone!(@strong peer_id, @weak element => move |reply| {
if let Ok(Some(stats)) = reply {
let mut state = element.imp().state.lock().unwrap();
if let Some(mut consumer) = state.consumers.get_mut(&peer_id) {
if let Some(congestion_controller) = consumer.congestion_controller.as_mut() {
congestion_controller.delay_control(&element, stats, &mut consumer.encoders,);
}
consumer.stats = stats.to_owned();
}
}
}),
);
webrtcbin.emit_by_name::<()>("get-stats", &[&None::<gst::Pad>, &promise]);
}
fn on_remote_description_set(&self, element: &super::WebRTCSink, peer_id: String) {
let mut state = self.state.lock().unwrap();
let mut remove = false;
@ -1874,18 +2021,12 @@ impl WebRTCSink {
while interval.next().await.is_some() {
let element_clone = element_clone.clone();
let peer_id_clone = peer_id_clone.clone();
if let Some(webrtcbin) = webrtcbin.upgrade() {
let promise = gst::Promise::with_change_func(move |reply| {
if let Some(element) = element_clone.upgrade() {
let this = Self::from_instance(&element);
if let Ok(Some(stats)) = reply {
this.process_webrtcbin_stats(&element, &peer_id_clone, stats);
}
}
});
webrtcbin.emit_by_name::<()>("get-stats", &[&None::<gst::Pad>, &promise]);
if let (Some(webrtcbin), Some(element)) =
(webrtcbin.upgrade(), element_clone.upgrade())
{
element
.imp()
.process_stats(&element, webrtcbin, &peer_id_clone);
} else {
break;
}
@ -2426,6 +2567,7 @@ impl ObjectImpl for WebRTCSink {
.set_bitrate(&self.instance(), consumer.max_bitrate as i32);
encoder.transceiver.set_property("fec-percentage", 50u32);
}
consumer.stats_sigid.take();
}
WebRTCSinkCongestionControl::Homegrown => {
let _ = consumer.congestion_controller.insert(