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|
// SPDX-License-Identifier: MPL-2.0
use crate::signaller::Signallable;
/**
* SECTION:element-webrtcsink
* @symbols:
* - GstBaseWebRTCSink
* - GstRSWebRTCSignallableIface
*
* `webrtcsink` is an element that can be used to serve media streams
* to multiple consumers through WebRTC.
*
* It uses a signaller that implements the protocol supported by the default
* signalling server we additionally provide, take a look at the subclasses of
* #GstBaseWebRTCSink for other supported protocols, or implement your own.
*
* See the [documentation of the plugin](plugin-rswebrtc) for more information
* on features and usage.
*/
/**
* GstBaseWebRTCSink:
* @title: Base class for WebRTC producers
*
* Base class for WebRTC sinks to implement and provide their own protocol for.
*/
/**
* GstRSWebRTCSignallableIface:
* @title: Interface for WebRTC signalling protocols
*
* Interface that WebRTC elements can implement their own protocol with.
*/
use gst::glib;
use gst::prelude::*;
use gst::subclass::prelude::*;
mod homegrown_cc;
mod imp;
glib::wrapper! {
pub struct BaseWebRTCSink(ObjectSubclass<imp::BaseWebRTCSink>) @extends gst::Bin, gst::Element, gst::Object, @implements gst::ChildProxy, gst_video::Navigation;
}
glib::wrapper! {
pub struct WebRTCSink(ObjectSubclass<imp::WebRTCSink>) @extends BaseWebRTCSink, gst::Bin, gst::Element, gst::Object, @implements gst::ChildProxy, gst_video::Navigation;
}
glib::wrapper! {
pub struct AwsKvsWebRTCSink(ObjectSubclass<imp::AwsKvsWebRTCSink>) @extends BaseWebRTCSink, gst::Bin, gst::Element, gst::Object, @implements gst::ChildProxy, gst_video::Navigation;
}
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;
}
glib::wrapper! {
pub struct JanusVRWebRTCSink(ObjectSubclass<imp::JanusVRWebRTCSink>) @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")]
NoSessionWithId(String),
#[error("consumer refused media")]
ConsumerRefusedMedia { session_id: String, media_idx: u32 },
#[error("consumer did not provide valid payload for media")]
ConsumerNoValidPayload { session_id: String, media_idx: u32 },
#[error("SDP mline index is currently mandatory")]
MandatorySdpMlineIndex,
#[error("duplicate session id")]
DuplicateSessionId(String),
#[error("error setting up consumer pipeline")]
SessionPipelineError {
session_id: String,
peer_id: String,
details: String,
},
}
impl Default for BaseWebRTCSink {
fn default() -> Self {
glib::Object::new()
}
}
impl BaseWebRTCSink {
pub fn with_signaller(signaller: Signallable) -> Self {
let ret: BaseWebRTCSink = glib::Object::new();
let ws = ret.imp();
ws.set_signaller(signaller).unwrap();
ret
}
}
#[derive(Debug, Eq, PartialEq, Ord, PartialOrd, Hash, Clone, Copy, glib::Enum)]
#[repr(u32)]
#[enum_type(name = "GstWebRTCSinkCongestionControl")]
pub enum WebRTCSinkCongestionControl {
#[enum_value(name = "Disabled: no congestion control is applied", nick = "disabled")]
Disabled,
#[enum_value(name = "Homegrown: simple sender-side heuristic", nick = "homegrown")]
Homegrown,
#[enum_value(name = "Google Congestion Control algorithm", nick = "gcc")]
GoogleCongestionControl,
}
#[glib::flags(name = "GstWebRTCSinkMitigationMode")]
enum WebRTCSinkMitigationMode {
#[flags_value(name = "No mitigation applied", nick = "none")]
NONE = 0b00000000,
#[flags_value(name = "Lowered resolution", nick = "downscaled")]
DOWNSCALED = 0b00000001,
#[flags_value(name = "Lowered framerate", nick = "downsampled")]
DOWNSAMPLED = 0b00000010,
}
pub fn register(plugin: &gst::Plugin) -> Result<(), glib::BoolError> {
BaseWebRTCSink::static_type().mark_as_plugin_api(gst::PluginAPIFlags::empty());
WebRTCSinkCongestionControl::static_type().mark_as_plugin_api(gst::PluginAPIFlags::empty());
gst::Element::register(
Some(plugin),
"webrtcsink",
gst::Rank::NONE,
WebRTCSink::static_type(),
)?;
gst::Element::register(
Some(plugin),
"awskvswebrtcsink",
gst::Rank::NONE,
AwsKvsWebRTCSink::static_type(),
)?;
gst::Element::register(
Some(plugin),
"whipclientsink",
gst::Rank::NONE,
WhipWebRTCSink::static_type(),
)?;
gst::Element::register(
Some(plugin),
"livekitwebrtcsink",
gst::Rank::NONE,
LiveKitWebRTCSink::static_type(),
)?;
/**
* element-janusvrwebrtcsink:
*
* The `JanusVRWebRTCSink` is a plugin that integrates with the [Video Room plugin](https://janus.conf.meetecho.com/docs/videoroom) of the [Janus Gateway](https://github.com/meetecho/janus-gateway). It basically streams whatever data you pipe to it (video, audio) into WebRTC using Janus as the signaller.
*
* ## How to use it
*
* You'll need to have:
*
* - A Janus server endpoint;
* - Any WebRTC browser application that uses Janus as the signaller, eg: the `html` folder of [janus-gateway repository](https://github.com/meetecho/janus-gateway).
*
* You can pipe the video like this (if you don't happen to run Janus locally, you can set the endpoint
* like this: `signaller::janus-endpoint=ws://127.0.0.1:8188`):
*
* ```bash
* $ gst-launch-1.0 videotestsrc ! janusvrwebrtcsink signaller::room-id=1234
* ```
*
* And for audio (yes you can do both at the same time, you just need to pipe it properly).
*
* ```bash
* $ gst-launch-1.0 audiotestsrc ! janusvrwebrtcsink signaller::room-id=1234
* ```
*
* And you can set the display name via `signaller::display-name`, eg:
*
* ```bash
* $ gst-launch-1.0 videotestsrc ! janusvrwebrtcsink signaller::room-id=1234 signaller::display-name=ana
* ```
*
* You should see the GStreamer `videotestsrc`/`audiotestsrc` output in your browser now!
*
* If for some reason you can't run Janus locally, you can use their open [demo webpage](https://janus.conf.meetecho.com/demos/videoroom.html), and point to its WebSocket server:
*
* ```bash
* $ gst-launch-1.0 videotestsrc ! janusvrwebrtcsink signaller::room-id=1234 signaller::janus-endpoint=wss://janus.conf.meetecho.com/ws
* ```
*
* ## Reference links
*
* - [Janus REST/WebSockets docs](https://janus.conf.meetecho.com/docs/rest.html)
* - [Example implementation in GStreamer](https://gitlab.freedesktop.org/gstreamer/gstreamer/-/blob/269ab858813e670d521cc4b6a71cc0ec4a6e70ed/subprojects/gst-examples/webrtc/janus/rust/src/janus.rs)
*
* ## Notes
*
* - This plugin supports both the legacy Video Room plugin as well as the `multistream` one;
* - If you see a warning in the logs related to `rtpgccbwe`, you're probably missing the `gst-plugin-rtp` in your system.
*/
gst::Element::register(
Some(plugin),
"janusvrwebrtcsink",
gst::Rank::NONE,
JanusVRWebRTCSink::static_type(),
)?;
Ok(())
}
|
