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webrtc: Stop using deprecated API and require GStreamer 1.22 or newer

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2243>
This commit is contained in:
Sebastian Dröge 2025-05-14 08:40:39 +03:00
parent d8d988427f
commit 236608f48e
3 changed files with 37 additions and 25 deletions
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19
net/webrtc/Cargo.toml
View file

@ -9,14 +9,14 @@ repository.workspace = true
rust-version.workspace = true
[dependencies]
gst = { workspace = true, features = ["v1_20", "serde"] }
gst-app = { workspace = true, features = ["v1_20"] }
gst-audio = { workspace = true, features = ["v1_20", "serde"] }
gst-video = { workspace = true, features = ["v1_20", "serde"] }
gst-net = { workspace = true, features = ["v1_20"] }
gst-webrtc = { workspace = true, features = ["v1_20"] }
gst-sdp = { workspace = true, features = ["v1_20"] }
gst-rtp = { workspace = true, features = ["v1_20"] }
gst = { workspace = true, features = ["v1_22", "serde"] }
gst-app = { workspace = true, features = ["v1_22"] }
gst-audio = { workspace = true, features = ["v1_22", "serde"] }
gst-video = { workspace = true, features = ["v1_22", "serde"] }
gst-net = { workspace = true, features = ["v1_22"] }
gst-webrtc = { workspace = true, features = ["v1_22"] }
gst-sdp = { workspace = true, features = ["v1_22"] }
gst-rtp = { workspace = true, features = ["v1_22"] }
gst-utils.workspace = true
gst-base.workspace = true
uuid = { version = "1", features = ["v4"] }
@ -82,10 +82,9 @@ path = "src/lib.rs"
gst-plugin-version-helper.workspace = true
[features]
default = ["v1_22", "janus", "whip", "web_server"]
default = ["janus", "whip", "web_server"]
static = []
capi = []
v1_22 = ["gst/v1_22", "gst-app/v1_22", "gst-video/v1_22", "gst-webrtc/v1_22", "gst-sdp/v1_22", "gst-rtp/v1_22"]
doc = []
aws = ["dep:aws-config", "dep:aws-smithy-runtime", "dep:aws-smithy-runtime-api", "dep:aws-types", "dep:aws-credential-types",

25
net/webrtc/src/webrtcsink/imp.rs
View file

@ -63,11 +63,8 @@ const DEFAULT_MIN_BITRATE: u32 = 1000;
* my local network, possibly related to chrome's pretty low UDP
* buffer sizes */
const DEFAULT_MAX_BITRATE: u32 = 8192000;
const DEFAULT_CONGESTION_CONTROL: WebRTCSinkCongestionControl = if cfg!(feature = "v1_22") {
WebRTCSinkCongestionControl::GoogleCongestionControl
} else {
WebRTCSinkCongestionControl::Disabled
};
const DEFAULT_CONGESTION_CONTROL: WebRTCSinkCongestionControl =
WebRTCSinkCongestionControl::GoogleCongestionControl;
const DEFAULT_DO_FEC: bool = true;
const DEFAULT_DO_RETRANSMISSION: bool = true;
const DEFAULT_DO_CLOCK_SIGNALLING: bool = false;
@ -90,7 +87,6 @@ const DEFAULT_WEB_SERVER_HOST_ADDR: &str = "http://127.0.0.1:8080";
const DEFAULT_FORWARD_METAS: &str = "";
/* Start adding some FEC when the bitrate > 2Mbps as we found experimentally
* that it is not worth it below that threshold */
#[cfg(feature = "v1_22")]
const DO_FEC_THRESHOLD: u32 = 2000000;
#[derive(Debug, Clone, Copy)]
@ -302,7 +298,6 @@ struct SessionInner {
pipeline: gst::Pipeline,
webrtcbin: gst::Element,
#[cfg(feature = "v1_22")]
rtprtxsend: Option<gst::Element>,
webrtc_pads: HashMap<u32, WebRTCPad>,
peer_id: String,
@ -1292,7 +1287,6 @@ impl SessionInner {
webrtcbin,
peer_id,
cc_info,
#[cfg(feature = "v1_22")]
rtprtxsend: None,
congestion_controller,
rtpgccbwe,
@ -1596,7 +1590,13 @@ impl ControlRequestHandler {
}
Ok(msg) => match serde_json::to_string(&msg).ok() {
Some(s) => {
channel.send_string(Some(s.as_str()));
if let Err(err) = channel.send_string_full(Some(s.as_str())) {
gst::error!(
CAT,
obj = element,
"Failed sending control request to peer: {err}",
);
}
}
None => {
gst::error!(
@ -2740,7 +2740,9 @@ impl BaseWebRTCSink {
info: utils::Info::Meta(meta),
}) {
Ok(msg) => {
handler.0 .1.send_string(Some(msg.as_str()));
if let Err(err) = handler.0 .1.send_string_full(Some(msg.as_str())) {
gst::error!(CAT, imp = self, "Failed sending meta to peer: {err}",);
}
}
Err(err) => {
gst::warning!(CAT, imp = self, "Failed to serialize info message: {err:?}",);
@ -2801,7 +2803,6 @@ impl BaseWebRTCSink {
}
let rtpgccbwe = match settings.cc_info.heuristic {
#[cfg(feature = "v1_22")]
WebRTCSinkCongestionControl::GoogleCongestionControl => {
let rtpgccbwe = match gst::ElementFactory::make("rtpgccbwe").build() {
Err(err) => {
@ -3459,7 +3460,6 @@ impl BaseWebRTCSink {
webrtcbin.emit_by_name::<()>("get-stats", &[&None::<gst::Pad>, &promise]);
}
#[cfg(feature = "v1_22")]
fn set_rtptrxsend(&self, session_id: &str, rtprtxsend: gst::Element) {
let mut state = self.state.lock().unwrap();
@ -3468,7 +3468,6 @@ impl BaseWebRTCSink {
}
}
#[cfg(feature = "v1_22")]
fn set_bitrate(&self, session_id: &str, bitrate: u32) {
let settings = self.settings.lock().unwrap();
let mut state = self.state.lock().unwrap();

18
net/webrtc/src/webrtcsrc/imp.rs
View file

@ -426,7 +426,14 @@ impl Session {
"Sending navigation event to peer for session {}",
self.id
);
data_channel.send_string(Some(str.as_str()));
if let Err(err) = data_channel.send_string_full(Some(str.as_str())) {
gst::error!(
CAT,
obj = element,
"Failed sending navigation event to peer for session {}: {err}",
self.id,
);
}
}
None => {
gst::error!(
@ -460,7 +467,14 @@ impl Session {
"Sending control request to peer for session {}",
self.id
);
data_channel.send_string(Some(str.as_str()));
if let Err(err) = data_channel.send_string_full(Some(str.as_str())) {
gst::error!(
CAT,
obj = element,
"Failed sending control request to peer for session {}: {err}",
self.id,
);
}
}
None => {
gst::error!(