mirror of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2025-04-04 14:59:36 +00:00
Fix fec ratio, max-bitrate and consider encoders
This commit is contained in:
nazar
Mathieu Duponchelle
parent
5ebf9913a1
commit
ef9c818522
1 changed files with 69 additions and 55 deletions
|
|
@ -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);
|
||||
|
|
|
|||
Loading…
Reference in a new issue