webrtcsink: Add LiveKit WebRTC sink and signaller

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
This commit is contained in:
Olivier Crête 2023-06-21 15:54:00 -04:00 committed by GStreamer Marge Bot
parent de6abf1439
commit 793ee66afa
8 changed files with 844 additions and 1 deletions

View file

@ -6118,6 +6118,37 @@
},
"rank": "none"
},
"livekitwebrtcsink": {
"author": "Olivier Crête <olivier.crete@collabora.com>",
"description": "WebRTC sink with LiveKit signaller",
"hierarchy": [
"GstLiveKitWebRTCSink",
"GstBaseWebRTCSink",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy",
"GstNavigation"
],
"klass": "Sink/Network/WebRTC",
"pad-templates": {
"audio_%%u": {
"caps": "audio/x-raw:\naudio/x-opus:\n",
"direction": "sink",
"presence": "request"
},
"video_%%u": {
"caps": "video/x-raw:\n\nvideo/x-raw(memory:CUDAMemory):\n\nvideo/x-raw(memory:GLMemory):\n\nvideo/x-raw(memory:NVMM):\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\n",
"direction": "sink",
"presence": "request"
}
},
"rank": "none"
},
"webrtcsink": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "WebRTC sink with custom protocol signaller",

View file

@ -27,7 +27,7 @@ tokio = { version = "1", features = ["fs", "macros", "rt-multi-thread", "time"]
tokio-native-tls = "0.3.0"
tokio-stream = "0.1.11"
async-tungstenite = { version = "0.22", features = ["tokio-runtime", "tokio-native-tls"] }
serde = "1"
serde = { version = "1", features = ["derive"] }
serde_json = "1"
fastrand = "2.0"
gst_plugin_webrtc_protocol = { path="protocol", package = "gst-plugin-webrtc-signalling-protocol" }
@ -46,11 +46,15 @@ http = "0.2.7"
chrono = "0.4"
data-encoding = "2.3.3"
url-escape = "0.1.1"
regex = "1"
reqwest = { version = "0.11", features = ["default-tls"] }
parse_link_header = {version = "0.3", features = ["url"]}
async-recursion = "1.0.0"
livekit-protocol = { version = "0.1.3" }
livekit-api = { version = "0.1.3", default-features = false, features = ["signal-client", "access-token", "native-tls"] }
[dev-dependencies]
tracing = { version = "0.1", features = ["log"] }
tracing-subscriber = { version = "0.3", features = ["registry", "env-filter"] }

View file

@ -258,5 +258,28 @@ gst-launch-1.0 -e uridecodebin uri=file:///home/meh/path/to/video/file ! \
You should see a second video displayed in the videoroomtest web page.
## Using the LiveKit Signaller
Testing the LiveKit signaller can be done by setting up [LiveKit] and creating a room.
You can connect either by given the API key and secret:
``` shell
gst-launch-1.0 -e uridecodebin uri=file:///home/meh/path/to/video/file ! \
videoconvert ! video/x-raw ! queue ! \
livekitwebrtcsink signaller::ws-url=ws://127.0.0.1:7880 signaller::api-key=devkey signaller::secret-key=secret signaller::room-name=testroom
```
Or by using a separately created authentication token
``` shell
gst-launch-1.0 -e uridecodebin uri=file:///home/meh/path/to/video/file ! \
videoconvert ! video/x-raw ! queue ! \
livekitwebrtcsink signaller::ws-url=ws://127.0.0.1:7880 signaller::auth-token=mygeneratedtoken signaller::room-name=testroom
```
You should see a second video displayed in the videoroomtest web page.
[LiveKit]: https://livekit.io/
[janus]: https://github.com/meetecho/janus-gateway
[simple whip server]: https://github.com/meetecho/simple-whip-server/

View file

@ -15,6 +15,7 @@ use once_cell::sync::Lazy;
use tokio::runtime;
mod aws_kvs_signaller;
mod livekit_signaller;
pub mod signaller;
pub mod utils;
pub mod webrtcsink;

View file

@ -0,0 +1,714 @@
// SPDX-License-Identifier: MPL-2.0
use crate::signaller::{Signallable, SignallableImpl};
use crate::utils::{wait_async, WaitError};
use crate::RUNTIME;
use anyhow::anyhow;
use futures::executor::block_on;
use gst::glib;
use gst::prelude::*;
use gst::subclass::prelude::*;
use once_cell::sync::Lazy;
use serde::{Deserialize, Serialize};
use std::collections::HashMap;
use std::sync::{Arc, Mutex};
use tokio::sync::oneshot;
use tokio::task::JoinHandle;
use livekit_api::access_token::{AccessToken, VideoGrants};
use livekit_api::signal_client;
use livekit_protocol as proto;
static CAT: Lazy<gst::DebugCategory> = Lazy::new(|| {
gst::DebugCategory::new(
"webrtc-livekit-signaller",
gst::DebugColorFlags::empty(),
Some("WebRTC LiveKit signaller"),
)
});
const DEFAULT_TRACK_PUBLISH_TIMEOUT: u32 = 10;
#[derive(Clone)]
struct Settings {
wsurl: Option<String>,
api_key: Option<String>,
secret_key: Option<String>,
participant_name: Option<String>,
identity: Option<String>,
room_name: Option<String>,
auth_token: Option<String>,
timeout: u32,
}
impl Default for Settings {
fn default() -> Self {
Self {
wsurl: Some("ws://127.0.0.1:7880".to_string()),
api_key: None,
secret_key: None,
participant_name: Some("GStreamer".to_string()),
identity: Some("gstreamer".to_string()),
room_name: None,
auth_token: None,
timeout: DEFAULT_TRACK_PUBLISH_TIMEOUT,
}
}
}
#[derive(Default)]
pub struct Signaller {
settings: Mutex<Settings>,
connection: Mutex<Option<Connection>>,
join_canceller: Mutex<Option<futures::future::AbortHandle>>,
signal_task_canceller: Mutex<Option<futures::future::AbortHandle>>,
}
struct Channels {
reliable_channel: gst_webrtc::WebRTCDataChannel,
lossy_channel: gst_webrtc::WebRTCDataChannel,
}
struct Connection {
signal_client: Arc<signal_client::SignalClient>,
pending_tracks: HashMap<String, oneshot::Sender<proto::TrackInfo>>,
signal_task: JoinHandle<()>,
early_candidates: Option<Vec<String>>,
channels: Option<Channels>,
}
#[derive(Serialize, Deserialize)]
#[serde(rename_all = "camelCase")]
struct IceCandidateJson {
pub sdp_mid: String,
pub sdp_m_line_index: i32,
pub candidate: String,
}
impl Signaller {
fn raise_error(&self, msg: String) {
self.obj()
.emit_by_name::<()>("error", &[&format!("Error: {msg}")]);
}
async fn signal_task(&self, mut signal_events: signal_client::SignalEvents) {
loop {
match wait_async(&self.signal_task_canceller, signal_events.recv(), 0).await {
Ok(Some(signal)) => match signal {
signal_client::SignalEvent::Open => {}
signal_client::SignalEvent::Signal(signal) => {
self.on_signal_event(signal).await;
}
signal_client::SignalEvent::Close => {
gst::debug!(CAT, imp: self, "Close");
self.raise_error("Server disconnected".to_string());
break;
}
},
Ok(None) => {}
Err(err) => match err {
WaitError::FutureAborted => {
gst::debug!(CAT, imp: self, "Closing signal_task");
break;
}
WaitError::FutureError(err) => self.raise_error(err.to_string()),
},
}
}
}
async fn on_signal_event(&self, event: proto::signal_response::Message) {
match event {
proto::signal_response::Message::Answer(answer) => {
gst::debug!(CAT, imp: self, "Received publisher answer: {:?}", answer);
let sdp = match gst_sdp::SDPMessage::parse_buffer(answer.sdp.as_bytes()) {
Ok(sdp) => sdp,
Err(_) => {
self.raise_error("Couldn't parse Answer SDP".to_string());
return;
}
};
let answer = gst_webrtc::WebRTCSessionDescription::new(
gst_webrtc::WebRTCSDPType::Answer,
sdp,
);
self.obj()
.emit_by_name::<()>("session-description", &[&"unique", &answer]);
}
proto::signal_response::Message::Trickle(trickle) => {
let target = if let Some(target) = proto::SignalTarget::from_i32(trickle.target) {
target
} else {
gst::warning!(
CAT,
imp: self,
"Received ice_candidate {:?} from invalid target, ignoring",
trickle
);
return;
};
gst::debug!(CAT, imp: self, "Received ice_candidate {:?}", trickle);
if target == proto::SignalTarget::Publisher {
if let Ok(json) =
serde_json::from_str::<IceCandidateJson>(&trickle.candidate_init)
{
let mline = json.sdp_m_line_index as u32;
self.obj().emit_by_name::<()>(
"handle-ice",
&[&"unique", &mline, &Some(json.sdp_mid), &json.candidate],
);
}
}
}
proto::signal_response::Message::ConnectionQuality(quality) => {
gst::debug!(CAT, imp: self, "Connection quality: {:?}", quality);
}
proto::signal_response::Message::TrackPublished(publish_res) => {
gst::debug!(CAT, imp: self, "Track published: {:?}", publish_res);
if let Some(connection) = &mut *self.connection.lock().unwrap() {
if let Some(tx) = connection.pending_tracks.remove(&publish_res.cid) {
let _ = tx.send(publish_res.track.unwrap());
}
}
}
proto::signal_response::Message::Leave(leave) => {
gst::debug!(CAT, imp: self, "Leave: {:?}", leave);
}
_ => {}
}
}
}
impl SignallableImpl for Signaller {
fn start(&self) {
gst::debug!(CAT, imp: self, "Connecting");
let wsurl = if let Some(wsurl) = &self.settings.lock().unwrap().wsurl {
wsurl.clone()
} else {
self.raise_error("WebSocket URL must be set".to_string());
return;
};
let auth_token = {
let settings = self.settings.lock().unwrap();
if let Some(auth_token) = &settings.auth_token {
auth_token.clone()
} else if let (
Some(api_key),
Some(secret_key),
Some(identity),
Some(participant_name),
Some(room_name),
) = (
&settings.api_key,
&settings.secret_key,
&settings.identity,
&settings.participant_name,
&settings.room_name,
) {
let grants = VideoGrants {
room_join: true,
can_subscribe: false,
room: room_name.clone(),
..Default::default()
};
let access_token = AccessToken::with_api_key(api_key, secret_key)
.with_name(participant_name)
.with_identity(identity)
.with_grants(grants);
match access_token.to_jwt() {
Ok(token) => token,
Err(err) => {
self.raise_error(format!(
"{:?}",
anyhow!("Could not create auth token {err}")
));
return;
}
}
} else {
self.raise_error("Either auth-token or (api-key and secret-key and identity and room-name) must be set".to_string());
return;
}
};
gst::debug!(CAT, imp: self, "We have an authentication token");
let weak_imp = self.downgrade();
RUNTIME.spawn(async move {
let imp = if let Some(imp) = weak_imp.upgrade() {
imp
} else {
return;
};
let options = signal_client::SignalOptions::default();
gst::debug!(CAT, imp: imp, "Connecting to {}", wsurl);
let res = signal_client::SignalClient::connect(&wsurl, &auth_token, options).await;
let (signal_client, join_response, signal_events) = match res {
Err(err) => {
imp.obj()
.emit_by_name::<()>("error", &[&format!("{:?}", anyhow!("Error: {err}"))]);
return;
}
Ok(ok) => ok,
};
let signal_client = Arc::new(signal_client);
gst::debug!(
CAT,
imp: imp,
"Connected with JoinResponse: {:?}",
join_response
);
let weak_imp = imp.downgrade();
let signal_task = RUNTIME.spawn(async move {
if let Some(imp) = weak_imp.upgrade() {
imp.signal_task(signal_events).await;
}
});
let weak_imp = imp.downgrade();
imp.obj().connect_closure(
"consumer-added",
false,
glib::closure!(|_signaler: &super::LiveKitSignaller,
_consumer_identifier: &str,
webrtcbin: &gst::Element| {
gst::info!(CAT, "Adding data channels");
let reliable_channel = webrtcbin.emit_by_name::<gst_webrtc::WebRTCDataChannel>(
"create-data-channel",
&[
&"_reliable",
&gst::Structure::builder("config")
.field("ordered", true)
.build(),
],
);
let lossy_channel = webrtcbin.emit_by_name::<gst_webrtc::WebRTCDataChannel>(
"create-data-channel",
&[
&"_lossy",
&gst::Structure::builder("config")
.field("ordered", true)
.field("max-retransmits", 0)
.build(),
],
);
if let Some(imp) = weak_imp.upgrade() {
let mut connection = imp.connection.lock().unwrap();
if let Some(connection) = connection.as_mut() {
connection.channels = Some(Channels {
reliable_channel,
lossy_channel,
});
}
}
}),
);
let connection = Connection {
signal_client,
signal_task,
pending_tracks: Default::default(),
early_candidates: Some(Vec::new()),
channels: None,
};
if let Ok(mut sc) = imp.connection.lock() {
*sc = Some(connection);
}
imp.obj().emit_by_name::<()>(
"session-requested",
&[
&"unique",
&"unique",
&None::<gst_webrtc::WebRTCSessionDescription>,
],
);
});
}
fn send_sdp(&self, _session_id: &str, sessdesc: &gst_webrtc::WebRTCSessionDescription) {
gst::debug!(CAT, imp: self, "Created offer SDP {:#?}", sessdesc.sdp());
assert!(sessdesc.type_() == gst_webrtc::WebRTCSDPType::Offer);
let weak_imp = self.downgrade();
let sessdesc = sessdesc.clone();
RUNTIME.spawn(async move {
if let Some(imp) = weak_imp.upgrade() {
let sdp = sessdesc.sdp();
let signal_client = imp
.connection
.lock()
.unwrap()
.as_ref()
.unwrap()
.signal_client
.clone();
let timeout = imp.settings.lock().unwrap().timeout;
for media in sdp.medias() {
if let Some(mediatype) = media.media() {
let (mtype, msource) = if mediatype == "audio" {
(
proto::TrackType::Audio,
proto::TrackSource::Microphone as i32,
)
} else if mediatype == "video" {
(proto::TrackType::Video, proto::TrackSource::Camera as i32)
} else {
continue;
};
let mut disable_red = true;
if mtype == proto::TrackType::Audio {
for format in media.formats() {
if let Ok(pt) = format.parse::<i32>() {
if let Some(caps) = media.caps_from_media(pt) {
let s = caps.structure(0).unwrap();
let encoding_name = s.get::<&str>("encoding-name").unwrap();
if encoding_name == "RED" {
disable_red = false;
}
}
}
}
}
// Our SDP should always have a mid
let mid = media.attribute_val("mid").unwrap().to_string();
let mut trackid = "";
for attr in media.attributes() {
if attr.key() == "ssrc" {
if let Some(val) = attr.value() {
let split: Vec<&str> = val.split_whitespace().collect();
if split.len() == 3 && split[1].starts_with("msid:") {
trackid = split[2];
break;
}
}
}
}
// let layers = if mtype == proto::TrackType::Video {
// let ssrc = if let Some(attr) = media.attribute_val("ssrc") {
// let mut split = attr.split_whitespace();
// if let Some(ssrc_str) = split.next() {
// ssrc_str.parse().unwrap_or(0)
// } else {
// 0
// }
// } else {
// 0 as u32
// };
// gst::debug!(CAT, imp: imp, "Adding video track {mid} with ssrc {ssrc}");
// vec![ proto::VideoLayer {
// quality: proto::VideoQuality::High as i32,
// width: 1280,
// height: 720,
// bitrate: 5000,
// ssrc
// }]
// } else {
// gst::debug!(CAT, imp: imp, "Adding audio track {mid}");
// Vec::new()
// };
let req = proto::AddTrackRequest {
cid: trackid.to_string(),
name: mid.clone(),
r#type: mtype as i32,
muted: false,
source: msource,
disable_dtx: true,
disable_red,
// layers: layers,
..Default::default()
};
let (tx, rx) = oneshot::channel();
if let Some(connection) = &mut *imp.connection.lock().unwrap() {
let pendings_tracks = &mut connection.pending_tracks;
if pendings_tracks.contains_key(&req.cid) {
panic!("track already published");
}
pendings_tracks.insert(req.cid.clone(), tx);
}
let cid = req.cid.clone();
signal_client
.send(proto::signal_request::Message::AddTrack(req))
.await;
if let Err(err) = wait_async(&imp.join_canceller, rx, timeout).await {
if let Some(connection) = &mut *imp.connection.lock().unwrap() {
connection.pending_tracks.remove(&cid);
}
match err {
WaitError::FutureAborted => {
gst::warning!(CAT, imp: imp, "Future aborted")
}
WaitError::FutureError(err) => imp.raise_error(err.to_string()),
};
}
}
}
gst::debug!(CAT, imp: imp, "Sending SDP now");
signal_client
.send(proto::signal_request::Message::Offer(
proto::SessionDescription {
r#type: "offer".to_string(),
sdp: sessdesc.sdp().to_string(),
},
))
.await;
if let Some(imp) = weak_imp.upgrade() {
let early_candidates =
if let Some(connection) = &mut *imp.connection.lock().unwrap() {
connection.early_candidates.take()
} else {
None
};
if let Some(mut early_candidates) = early_candidates {
while let Some(candidate_str) = early_candidates.pop() {
gst::debug!(
CAT,
imp: imp,
"Sending delayed ice candidate {candidate_str:?}"
);
signal_client
.send(proto::signal_request::Message::Trickle(
proto::TrickleRequest {
candidate_init: candidate_str,
target: proto::SignalTarget::Publisher as i32,
},
))
.await;
}
}
}
}
});
}
fn add_ice(
&self,
_session_id: &str,
candidate: &str,
sdp_m_line_index: u32,
sdp_mid: Option<String>,
) {
let candidate_str = serde_json::to_string(&IceCandidateJson {
sdp_mid: sdp_mid.unwrap_or("".to_string()),
sdp_m_line_index: sdp_m_line_index as i32,
candidate: candidate.to_string(),
})
.unwrap();
if let Some(connection) = &mut *self.connection.lock().unwrap() {
if let Some(early_candidates) = connection.early_candidates.as_mut() {
gst::debug!(CAT, imp: self, "Delaying ice candidate {candidate_str:?}");
early_candidates.push(candidate_str);
return;
}
};
gst::debug!(CAT, imp: self, "Sending ice candidate {candidate_str:?}");
let imp = self.downgrade();
RUNTIME.spawn(async move {
if let Some(imp) = imp.upgrade() {
let signal_client = if let Some(connection) = &mut *imp.connection.lock().unwrap() {
connection.signal_client.clone()
} else {
return;
};
signal_client
.send(proto::signal_request::Message::Trickle(
proto::TrickleRequest {
candidate_init: candidate_str,
target: proto::SignalTarget::Publisher as i32,
},
))
.await;
};
});
}
fn stop(&self) {
if let Some(canceller) = &*self.join_canceller.lock().unwrap() {
canceller.abort();
}
if let Some(canceller) = &*self.signal_task_canceller.lock().unwrap() {
canceller.abort();
}
if let Some(connection) = self.connection.lock().unwrap().take() {
block_on(connection.signal_task).unwrap();
block_on(connection.signal_client.close());
}
}
fn end_session(&self, session_id: &str) {
assert_eq!(session_id, "unique");
}
}
#[glib::object_subclass]
impl ObjectSubclass for Signaller {
const NAME: &'static str = "GstLiveKitWebRTCSinkSignaller";
type Type = super::LiveKitSignaller;
type ParentType = glib::Object;
type Interfaces = (Signallable,);
}
impl ObjectImpl for Signaller {
fn properties() -> &'static [glib::ParamSpec] {
static PROPERTIES: Lazy<Vec<glib::ParamSpec>> = Lazy::new(|| {
vec![
glib::ParamSpecString::builder("ws-url")
.nick("WebSocket URL")
.blurb("The URL of the websocket of the LiveKit server")
.mutable_ready()
.build(),
glib::ParamSpecString::builder("api-key")
.nick("API key")
.blurb("API key (combined into auth-token)")
.mutable_ready()
.build(),
glib::ParamSpecString::builder("secret-key")
.nick("Secret Key")
.blurb("Secret key (combined into auth-token)")
.mutable_ready()
.build(),
glib::ParamSpecString::builder("participant-name")
.nick("Participant name")
.blurb("Human readable name of the participant (combined into auth-token)")
.mutable_ready()
.build(),
glib::ParamSpecString::builder("identity")
.nick("Participant Identity")
.blurb("Identity of the participant (combined into auth-token)")
.mutable_ready()
.build(),
glib::ParamSpecString::builder("auth-token")
.nick("Authorization Token")
.blurb("Authentication token to use (contains api_key/secret/name/identity)")
.mutable_ready()
.build(),
glib::ParamSpecString::builder("room-name")
.nick("Room Name")
.blurb("Name of the room to join (mandatory)")
.mutable_ready()
.build(),
glib::ParamSpecUInt::builder("timeout")
.nick("Timeout")
.blurb("Value in seconds to timeout join requests.")
.maximum(3600)
.minimum(1)
.default_value(DEFAULT_TRACK_PUBLISH_TIMEOUT)
.build(),
glib::ParamSpecObject::builder::<gst_webrtc::WebRTCDataChannel>("reliable-channel")
.nick("Reliable Channel")
.blurb("Reliable Data Channel object.")
.flags(glib::ParamFlags::READABLE)
.build(),
glib::ParamSpecObject::builder::<gst_webrtc::WebRTCDataChannel>("lossy-channel")
.nick("Lossy Channel")
.blurb("Lossy Data Channel object.")
.flags(glib::ParamFlags::READABLE)
.build(),
]
});
PROPERTIES.as_ref()
}
fn set_property(&self, _id: usize, value: &glib::Value, pspec: &glib::ParamSpec) {
let mut settings = self.settings.lock().unwrap();
match pspec.name() {
"ws-url" => {
settings.wsurl = value.get().expect("WebSocket URL should be a string");
}
"api-key" => {
settings.api_key = value.get().expect("API Key should be a string");
}
"secret-key" => {
settings.secret_key = value.get().expect("Secret Key should be a string");
}
"participant-name" => {
settings.participant_name = value.get().expect("Participant Name should be a string");
}
"identity" => {
settings.identity = value.get().expect("Participant Identity should be a string");
}
"room-name" => {
settings.room_name = value.get().expect("Room Name should be a string");
}
"auth-token" => {
settings.auth_token = value.get().expect("Auth token should be a string");
}
"timeout" => {
settings.timeout = value.get().expect("type checked upstream");
}
_ => unimplemented!(),
}
}
fn property(&self, _id: usize, pspec: &glib::ParamSpec) -> glib::Value {
let settings = self.settings.lock().unwrap();
match pspec.name() {
"ws-url" => settings.wsurl.to_value(),
"api-key" => settings.api_key.to_value(),
"secret-key" => settings.secret_key.to_value(),
"participant-name" => settings.participant_name.to_value(),
"identity" => settings.identity.to_value(),
"room-name" => settings.room_name.to_value(),
"auth-token" => settings.auth_token.to_value(),
"timeout" => settings.timeout.to_value(),
channel @ ("reliable-channel" | "lossy-channel") => {
let channel = if let Some(connection) = &*self.connection.lock().unwrap() {
if let Some(channels) = &connection.channels {
if channel == "reliable-channel" {
Some(channels.reliable_channel.clone())
} else {
Some(channels.lossy_channel.clone())
}
} else {
None
}
} else {
None
};
channel.to_value()
}
_ => unimplemented!(),
}
}
}

View file

@ -0,0 +1,19 @@
// SPDX-License-Identifier: MPL-2.0
use crate::signaller::Signallable;
use gst::glib;
mod imp;
glib::wrapper! {
pub struct LiveKitSignaller(ObjectSubclass<imp::Signaller>) @implements Signallable;
}
unsafe impl Send for LiveKitSignaller {}
unsafe impl Sync for LiveKitSignaller {}
impl Default for LiveKitSignaller {
fn default() -> Self {
glib::Object::new()
}
}

View file

@ -22,6 +22,7 @@ use std::sync::{mpsc, Arc, Condvar, Mutex};
use super::homegrown_cc::CongestionController;
use super::{WebRTCSinkCongestionControl, WebRTCSinkError, WebRTCSinkMitigationMode};
use crate::aws_kvs_signaller::AwsKvsSignaller;
use crate::livekit_signaller::LiveKitSignaller;
use crate::signaller::{prelude::*, Signallable, Signaller, WebRTCSignallerRole};
use crate::whip_signaller::WhipSignaller;
use crate::RUNTIME;
@ -3851,3 +3852,43 @@ impl ObjectSubclass for WhipWebRTCSink {
type Type = super::WhipWebRTCSink;
type ParentType = super::BaseWebRTCSink;
}
#[derive(Default)]
pub struct LiveKitWebRTCSink {}
impl ObjectImpl for LiveKitWebRTCSink {
fn constructed(&self) {
let element = self.obj();
let ws = element.upcast_ref::<super::BaseWebRTCSink>().imp();
let _ = ws.set_signaller(LiveKitSignaller::default().upcast());
}
}
impl GstObjectImpl for LiveKitWebRTCSink {}
impl ElementImpl for LiveKitWebRTCSink {
fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
gst::subclass::ElementMetadata::new(
"LiveKitWebRTCSink",
"Sink/Network/WebRTC",
"WebRTC sink with LiveKit signaller",
"Olivier Crête <olivier.crete@collabora.com>",
)
});
Some(&*ELEMENT_METADATA)
}
}
impl BinImpl for LiveKitWebRTCSink {}
impl BaseWebRTCSinkImpl for LiveKitWebRTCSink {}
#[glib::object_subclass]
impl ObjectSubclass for LiveKitWebRTCSink {
const NAME: &'static str = "GstLiveKitWebRTCSink";
type Type = super::LiveKitWebRTCSink;
type ParentType = super::BaseWebRTCSink;
}

View file

@ -56,6 +56,10 @@ glib::wrapper! {
pub struct WhipWebRTCSink(ObjectSubclass<imp::WhipWebRTCSink>) @extends BaseWebRTCSink, gst::Bin, gst::Element, gst::Object, @implements gst::ChildProxy, gst_video::Navigation;
}
glib::wrapper! {
pub struct LiveKitWebRTCSink(ObjectSubclass<imp::LiveKitWebRTCSink>) @extends BaseWebRTCSink, gst::Bin, gst::Element, gst::Object, @implements gst::ChildProxy, gst_video::Navigation;
}
#[derive(thiserror::Error, Debug)]
pub enum WebRTCSinkError {
#[error("no session with id")]
@ -136,6 +140,12 @@ pub fn register(plugin: &gst::Plugin) -> Result<(), glib::BoolError> {
gst::Rank::None,
WhipWebRTCSink::static_type(),
)?;
gst::Element::register(
Some(plugin),
"livekitwebrtcsink",
gst::Rank::None,
LiveKitWebRTCSink::static_type(),
)?;
Ok(())
}