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Fix fec ratio, max-bitrate and consider encoders

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nazar 2022-08-22 19:58:01 +01:00 committed by Mathieu Duponchelle
parent 5ebf9913a1
commit ef9c818522
1 changed files with 69 additions and 55 deletions
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124
plugins/src/webrtcsink/imp.rs
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@ -48,6 +48,9 @@ const DEFAULT_DO_FEC: bool = true;
const DEFAULT_DO_RETRANSMISSION: bool = true;
const DEFAULT_ENABLE_DATA_CHANNEL_NAVIGATION: bool = false;
const DEFAULT_START_BITRATE: u32 = 2048000;
/* Start adding some FEC when the bitrate > 2Mbps as we found experimentally
* that it is not worth it below that threshold */
const DO_FEC_THRESHOLD: u32 = 2000000;
#[derive(Debug, Clone, Copy)]
struct CCInfo {
@ -139,8 +142,12 @@ struct Session {
webrtc_pads: HashMap<u32, WebRTCPad>,
peer_id: String,
encoders: Vec<VideoEncoder>,
// Our Homegrown controller
// Our Homegrown controller (if cc_info.heuristic == Homegrown)
congestion_controller: Option<CongestionController>,
// Our BandwidthEstimator (if cc_info.heuristic == GoogleCongestionControl)
rtpgccbwe: Option<gst::Element>,
sdp: Option<gst_sdp::SDPMessage>,
stats: gst::Structure,
@ -732,6 +739,7 @@ impl Session {
webrtcbin: gst::Element,
peer_id: String,
congestion_controller: Option<CongestionController>,
rtpgccbwe: Option<gst::Element>,
cc_info: CCInfo,
) -> Self {
Self {
@ -742,6 +750,7 @@ impl Session {
cc_info,
rtprtxsend: None,
congestion_controller,
rtpgccbwe,
stats: gst::Structure::new_empty("application/x-webrtc-stats"),
sdp: None,
webrtc_pads: HashMap::new(),
@ -961,6 +970,11 @@ impl Session {
}
self.encoders.push(enc);
if let Some(ref rtpgccbwe) = self.rtpgccbwe.as_ref() {
let max_bitrate = self.cc_info.max_bitrate * (self.encoders.len() as u32);
rtpgccbwe.set_property("max-bitrate", max_bitrate);
}
}
let appsrc = appsrc.downcast::<gst_app::AppSrc>().unwrap();
@ -1362,48 +1376,49 @@ impl WebRTCSink {
webrtcbin.set_property("turn-server", turn_server);
}
match settings.cc_info.heuristic {
let rtpgccbwe = match settings.cc_info.heuristic {
WebRTCSinkCongestionControl::GoogleCongestionControl => {
webrtcbin.connect_closure(
"request-aux-sender",
false,
glib::closure!(@watch element, @strong session_id
=> move |_webrtcbin: gst::Element, _transport: gst::Object| {
let rtpgccbwe = match gst::ElementFactory::make("rtpgccbwe", None) {
Err(err) => {
glib::g_warning!(
"webrtcsink",
"The `rtpgccbwe` element is not available \
not doing any congestion control: {err:?}"
);
None
}
Ok(cc) => {
webrtcbin.connect_closure(
"request-aux-sender",
false,
glib::closure!(@watch element, @strong session_id, @weak-allow-none cc
=> move |_webrtcbin: gst::Element, _transport: gst::Object| {
let settings = element.imp().settings.lock().unwrap();
let cc = match gst::ElementFactory::make("rtpgccbwe", None) {
Err(err) => {
glib::g_warning!("webrtcsink",
"The `rtpgccbwe` element is not available \
not doing any congestion control: {err:?}"
);
return None;
},
Ok(e) => {
let max_bitrate = (settings.cc_info.max_bitrate as f64 * if settings.do_fec { 1.5 } else { 1.}) as u32;
e.set_properties(&[
("min-bitrate", &settings.cc_info.min_bitrate),
("estimated-bitrate", &settings.cc_info.start_bitrate),
("max-bitrate", &max_bitrate),
]);
let cc = cc.unwrap();
let settings = element.imp().settings.lock().unwrap();
// TODO: Bind properties with @element's
cc.set_properties(&[
("min-bitrate", &settings.cc_info.min_bitrate),
("estimated-bitrate", &settings.cc_info.start_bitrate),
("max-bitrate", &settings.cc_info.max_bitrate),
]);
e
}
};
cc.connect_notify(Some("estimated-bitrate"),
glib::clone!(@weak element, @strong session_id
=> move |bwe, pspec| {
element.imp().set_bitrate(&element, &session_id,
bwe.property::<u32>(pspec.name()));
}
));
cc.connect_notify(Some("estimated-bitrate"),
glib::clone!(@weak element, @strong session_id
=> move |bwe, pspec| {
element.imp().set_bitrate(&element, &session_id,
bwe.property::<u32>(pspec.name()));
}
));
Some(cc)
}),
);
Some(cc)
}),
);
}
};
webrtcbin.connect_closure(
"deep-element-added",
@ -1421,9 +1436,11 @@ impl WebRTCSink {
}
}),
);
rtpgccbwe
}
_ => (),
}
_ => None,
};
pipeline.add(&webrtcbin).unwrap();
@ -1553,18 +1570,14 @@ impl WebRTCSink {
webrtcbin.clone(),
peer_id.clone(),
match settings.cc_info.heuristic {
WebRTCSinkCongestionControl::Homegrown => {
let max_bitrate = (settings.cc_info.max_bitrate as f64
* if settings.do_fec { 1.5 } else { 1. })
as u32;
Some(CongestionController::new(
&peer_id,
settings.cc_info.min_bitrate,
max_bitrate,
))
}
WebRTCSinkCongestionControl::Homegrown => Some(CongestionController::new(
&peer_id,
settings.cc_info.min_bitrate,
settings.cc_info.max_bitrate,
)),
_ => None,
},
rtpgccbwe,
settings.cc_info,
);
@ -1589,7 +1602,7 @@ impl WebRTCSink {
session.stats_sigid = Some(rtp_session.connect_notify(Some("twcc-stats"),
glib::clone!(@strong session_id_str, @weak webrtcbin, @weak element => @default-panic, move |sess, pspec| {
// Run the Loss-based control algortithm on new peer TWCC feedbacks
// Run the Loss-based control algorithm on new peer TWCC feedbacks
element.imp().process_loss_stats(&element, &session_id_str, &sess.property::<gst::Structure>(pspec.name()));
})
));
@ -1778,22 +1791,23 @@ impl WebRTCSink {
}
fn set_bitrate(&self, element: &super::WebRTCSink, peer_id: &str, bitrate: u32) {
let settings = element.imp().settings.lock().unwrap();
let mut state = element.imp().state.lock().unwrap();
if let Some(session) = state.sessions.get_mut(peer_id) {
let fec_ratio = {
// Start adding some FEC when the bitrate > 2Mbps as we found experimentally
// that it is not worth it below that threshold
if bitrate <= 2_000_000 || session.cc_info.max_bitrate <= 2_000_000 {
0f64
if settings.do_fec && bitrate > DO_FEC_THRESHOLD {
(bitrate as f64 - DO_FEC_THRESHOLD as f64)
/ (session.cc_info.max_bitrate as f64 - DO_FEC_THRESHOLD as f64)
} else {
(bitrate as f64 - 2_000_000.)
/ (session.cc_info.max_bitrate as f64 - 2_000_000.)
0f64
}
};
let fec_percentage = fec_ratio * 50f64;
let encoders_bitrate = ((bitrate as f64) / (1. + (fec_percentage / 100.))) as i32;
let encoders_bitrate = ((bitrate as f64)
/ (1. + (fec_percentage / 100.))
/ (session.encoders.len() as f64)) as i32;
if let Some(ref rtpxsend) = session.rtprtxsend.as_ref() {
rtpxsend.set_property("stuffing-kbps", (bitrate as f64 / 1000.) as i32);