mirror of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2024-11-22 03:21:00 +00:00
net/webrtc: Extract BaseWebRTCSrc
Define a Base for all the webrtcsrc type elements so they can all be derived from it. Similar to base element defined for webrtcsink type elements Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
This commit is contained in:
Taruntej Kanakamalla
GStreamer Marge Bot
parent
3fcab67570
commit
a0638ec983
3 changed files with 229 additions and 170 deletions
|
|
@ -6440,6 +6440,7 @@
|
|||
"description": "WebRTC src",
|
||||
"hierarchy": [
|
||||
"GstWebRTCSrc",
|
||||
"GstBaseWebRTCSrc",
|
||||
"GstBin",
|
||||
"GstElement",
|
||||
"GstObject",
|
||||
|
|
@ -6466,108 +6467,7 @@
|
|||
"type": "GstWebRTCSrcPad"
|
||||
}
|
||||
},
|
||||
"properties": {
|
||||
"audio-codecs": {
|
||||
"blurb": "Names of audio codecs to be be used during the SDP negotiation. Valid values: [OPUS]",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": false,
|
||||
"controllable": false,
|
||||
"mutable": "ready",
|
||||
"readable": true,
|
||||
"type": "GstValueArray",
|
||||
"writable": true
|
||||
},
|
||||
"enable-data-channel-navigation": {
|
||||
"blurb": "Enable navigation events through a dedicated WebRTCDataChannel",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": false,
|
||||
"controllable": false,
|
||||
"default": "false",
|
||||
"mutable": "ready",
|
||||
"readable": true,
|
||||
"type": "gboolean",
|
||||
"writable": true
|
||||
},
|
||||
"meta": {
|
||||
"blurb": "Free form metadata about the consumer",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": false,
|
||||
"controllable": false,
|
||||
"mutable": "ready",
|
||||
"readable": true,
|
||||
"type": "GstStructure",
|
||||
"writable": true
|
||||
},
|
||||
"signaller": {
|
||||
"blurb": "The Signallable object to use to handle WebRTC Signalling",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": true,
|
||||
"controllable": false,
|
||||
"mutable": "null",
|
||||
"readable": true,
|
||||
"type": "GstRSWebRTCSignallableIface",
|
||||
"writable": true
|
||||
},
|
||||
"stun-server": {
|
||||
"blurb": "The STUN server of the form stun://host:port",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": false,
|
||||
"controllable": false,
|
||||
"default": "stun://stun.l.google.com:19302",
|
||||
"mutable": "ready",
|
||||
"readable": true,
|
||||
"type": "gchararray",
|
||||
"writable": true
|
||||
},
|
||||
"turn-servers": {
|
||||
"blurb": "The TURN servers of the form <\"turn(s)://username:password@host:port\", \"turn(s)://username1:password1@host1:port1\">",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": false,
|
||||
"controllable": false,
|
||||
"mutable": "ready",
|
||||
"readable": true,
|
||||
"type": "GstValueArray",
|
||||
"writable": true
|
||||
},
|
||||
"video-codecs": {
|
||||
"blurb": "Names of video codecs to be be used during the SDP negotiation. Valid values: [VP8, H264, VP9, H265]",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": false,
|
||||
"controllable": false,
|
||||
"mutable": "ready",
|
||||
"readable": true,
|
||||
"type": "GstValueArray",
|
||||
"writable": true
|
||||
}
|
||||
},
|
||||
"rank": "primary",
|
||||
"signals": {
|
||||
"request-encoded-filter": {
|
||||
"args": [
|
||||
{
|
||||
"name": "arg0",
|
||||
"type": "gchararray"
|
||||
},
|
||||
{
|
||||
"name": "arg1",
|
||||
"type": "gchararray"
|
||||
},
|
||||
{
|
||||
"name": "arg2",
|
||||
"type": "GstCaps"
|
||||
}
|
||||
],
|
||||
"return-type": "GstElement",
|
||||
"when": "last"
|
||||
}
|
||||
}
|
||||
"rank": "primary"
|
||||
},
|
||||
"whipclientsink": {
|
||||
"author": "Taruntej Kanakamalla <taruntej@asymptotic.io>",
|
||||
|
|
@ -6890,6 +6790,121 @@
|
|||
}
|
||||
}
|
||||
},
|
||||
"GstBaseWebRTCSrc": {
|
||||
"hierarchy": [
|
||||
"GstBaseWebRTCSrc",
|
||||
"GstBin",
|
||||
"GstElement",
|
||||
"GstObject",
|
||||
"GInitiallyUnowned",
|
||||
"GObject"
|
||||
],
|
||||
"interfaces": [
|
||||
"GstChildProxy"
|
||||
],
|
||||
"kind": "object",
|
||||
"properties": {
|
||||
"audio-codecs": {
|
||||
"blurb": "Names of audio codecs to be be used during the SDP negotiation. Valid values: [OPUS]",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": false,
|
||||
"controllable": false,
|
||||
"mutable": "ready",
|
||||
"readable": true,
|
||||
"type": "GstValueArray",
|
||||
"writable": true
|
||||
},
|
||||
"enable-data-channel-navigation": {
|
||||
"blurb": "Enable navigation events through a dedicated WebRTCDataChannel",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": false,
|
||||
"controllable": false,
|
||||
"default": "false",
|
||||
"mutable": "ready",
|
||||
"readable": true,
|
||||
"type": "gboolean",
|
||||
"writable": true
|
||||
},
|
||||
"meta": {
|
||||
"blurb": "Free form metadata about the consumer",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": false,
|
||||
"controllable": false,
|
||||
"mutable": "ready",
|
||||
"readable": true,
|
||||
"type": "GstStructure",
|
||||
"writable": true
|
||||
},
|
||||
"signaller": {
|
||||
"blurb": "The Signallable object to use to handle WebRTC Signalling",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": true,
|
||||
"controllable": false,
|
||||
"mutable": "null",
|
||||
"readable": true,
|
||||
"type": "GstRSWebRTCSignallableIface",
|
||||
"writable": true
|
||||
},
|
||||
"stun-server": {
|
||||
"blurb": "The STUN server of the form stun://host:port",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": false,
|
||||
"controllable": false,
|
||||
"default": "stun://stun.l.google.com:19302",
|
||||
"mutable": "ready",
|
||||
"readable": true,
|
||||
"type": "gchararray",
|
||||
"writable": true
|
||||
},
|
||||
"turn-servers": {
|
||||
"blurb": "The TURN servers of the form <\"turn(s)://username:password@host:port\", \"turn(s)://username1:password1@host1:port1\">",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": false,
|
||||
"controllable": false,
|
||||
"mutable": "ready",
|
||||
"readable": true,
|
||||
"type": "GstValueArray",
|
||||
"writable": true
|
||||
},
|
||||
"video-codecs": {
|
||||
"blurb": "Names of video codecs to be be used during the SDP negotiation. Valid values: [VP8, H264, VP9, H265]",
|
||||
"conditionally-available": false,
|
||||
"construct": false,
|
||||
"construct-only": false,
|
||||
"controllable": false,
|
||||
"mutable": "ready",
|
||||
"readable": true,
|
||||
"type": "GstValueArray",
|
||||
"writable": true
|
||||
}
|
||||
},
|
||||
"signals": {
|
||||
"request-encoded-filter": {
|
||||
"args": [
|
||||
{
|
||||
"name": "arg0",
|
||||
"type": "gchararray"
|
||||
},
|
||||
{
|
||||
"name": "arg1",
|
||||
"type": "gchararray"
|
||||
},
|
||||
{
|
||||
"name": "arg2",
|
||||
"type": "GstCaps"
|
||||
}
|
||||
],
|
||||
"return-type": "GstElement",
|
||||
"when": "last"
|
||||
}
|
||||
}
|
||||
},
|
||||
"GstRSWebRTCSignallableIface": {
|
||||
"hierarchy": [
|
||||
"GstRSWebRTCSignallableIface",
|
||||
|
|
|
|||
|
|
@ -43,7 +43,7 @@ struct Settings {
|
|||
}
|
||||
|
||||
#[derive(Default)]
|
||||
pub struct WebRTCSrc {
|
||||
pub struct BaseWebRTCSrc {
|
||||
settings: Mutex<Settings>,
|
||||
n_video_pads: AtomicU16,
|
||||
n_audio_pads: AtomicU16,
|
||||
|
|
@ -51,14 +51,21 @@ pub struct WebRTCSrc {
|
|||
}
|
||||
|
||||
#[glib::object_subclass]
|
||||
impl ObjectSubclass for WebRTCSrc {
|
||||
const NAME: &'static str = "GstWebRTCSrc";
|
||||
type Type = super::WebRTCSrc;
|
||||
impl ObjectSubclass for BaseWebRTCSrc {
|
||||
const NAME: &'static str = "GstBaseWebRTCSrc";
|
||||
type Type = super::BaseWebRTCSrc;
|
||||
type ParentType = gst::Bin;
|
||||
type Interfaces = (gst::URIHandler, gst::ChildProxy);
|
||||
type Interfaces = (gst::ChildProxy,);
|
||||
}
|
||||
|
||||
impl ObjectImpl for WebRTCSrc {
|
||||
unsafe impl<T: BaseWebRTCSrcImpl> IsSubclassable<T> for super::BaseWebRTCSrc {
|
||||
fn class_init(class: &mut glib::Class<Self>) {
|
||||
Self::parent_class_init::<T>(class);
|
||||
}
|
||||
}
|
||||
pub(crate) trait BaseWebRTCSrcImpl: BinImpl {}
|
||||
|
||||
impl ObjectImpl for BaseWebRTCSrc {
|
||||
fn properties() -> &'static [glib::ParamSpec] {
|
||||
static PROPS: Lazy<Vec<glib::ParamSpec>> = Lazy::new(|| {
|
||||
vec![
|
||||
|
|
@ -205,7 +212,7 @@ impl ObjectImpl for WebRTCSrc {
|
|||
static SIGNALS: Lazy<Vec<glib::subclass::Signal>> = Lazy::new(|| {
|
||||
vec![
|
||||
/**
|
||||
* WebRTCSrc::request-encoded-filter:
|
||||
* BaseWebRTCSrc::request-encoded-filter:
|
||||
* @producer_id: Identifier of the producer
|
||||
* @pad_name: The name of the output pad
|
||||
* @allowed_caps: the allowed caps for the output pad
|
||||
|
|
@ -278,7 +285,7 @@ struct SignallerSignals {
|
|||
handle_ice: glib::SignalHandlerId,
|
||||
}
|
||||
|
||||
impl WebRTCSrc {
|
||||
impl BaseWebRTCSrc {
|
||||
fn webrtcbin(&self) -> gst::Bin {
|
||||
let state = self.state.lock().unwrap();
|
||||
let webrtcbin = state
|
||||
|
|
@ -562,15 +569,22 @@ impl WebRTCSrc {
|
|||
let mline = transceiver.map_or(mline, |t| Some(t.mlineindex()));
|
||||
|
||||
// Same logic as gst_pad_create_stream_id and friends, making a hash of
|
||||
// the URI and adding `:<some-id>`, here the ID is the mline of the
|
||||
// the URI (session id, if URI doesn't exist) and adding `:<some-id>`, here the ID is the mline of the
|
||||
// stream in the SDP.
|
||||
mline.map(|mline| {
|
||||
let mut cs = glib::Checksum::new(glib::ChecksumType::Sha256).unwrap();
|
||||
cs.update(
|
||||
self.uri()
|
||||
.expect("get_stream_id should never be called if no URI has been set")
|
||||
.as_bytes(),
|
||||
);
|
||||
|
||||
let data: String = if self
|
||||
.signaller()
|
||||
.has_property("uri", Some(String::static_type()))
|
||||
{
|
||||
self.signaller().property::<Option<String>>("uri").unwrap()
|
||||
} else {
|
||||
// use the session id
|
||||
self.state.lock().unwrap().session_id.clone().unwrap()
|
||||
};
|
||||
|
||||
cs.update(data.as_bytes());
|
||||
|
||||
format!("{}:{mline}", cs.string().unwrap())
|
||||
})
|
||||
|
|
@ -983,20 +997,7 @@ impl WebRTCSrc {
|
|||
}
|
||||
}
|
||||
|
||||
impl ElementImpl for WebRTCSrc {
|
||||
fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
|
||||
static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
|
||||
gst::subclass::ElementMetadata::new(
|
||||
"WebRTCSrc",
|
||||
"Source/Network/WebRTC",
|
||||
"WebRTC src",
|
||||
"Thibault Saunier <tsaunier@igalia.com>",
|
||||
)
|
||||
});
|
||||
|
||||
Some(&*ELEMENT_METADATA)
|
||||
}
|
||||
|
||||
impl ElementImpl for BaseWebRTCSrc {
|
||||
fn pad_templates() -> &'static [gst::PadTemplate] {
|
||||
static PAD_TEMPLATES: Lazy<Vec<gst::PadTemplate>> = Lazy::new(|| {
|
||||
let mut video_caps_builder = gst::Caps::builder_full()
|
||||
|
|
@ -1095,11 +1096,11 @@ impl ElementImpl for WebRTCSrc {
|
|||
}
|
||||
}
|
||||
|
||||
impl GstObjectImpl for WebRTCSrc {}
|
||||
impl GstObjectImpl for BaseWebRTCSrc {}
|
||||
|
||||
impl BinImpl for WebRTCSrc {}
|
||||
impl BinImpl for BaseWebRTCSrc {}
|
||||
|
||||
impl ChildProxyImpl for WebRTCSrc {
|
||||
impl ChildProxyImpl for BaseWebRTCSrc {
|
||||
fn child_by_index(&self, index: u32) -> Option<glib::Object> {
|
||||
if index == 0 {
|
||||
Some(self.signaller().upcast())
|
||||
|
|
@ -1123,42 +1124,6 @@ impl ChildProxyImpl for WebRTCSrc {
|
|||
}
|
||||
}
|
||||
|
||||
impl URIHandlerImpl for WebRTCSrc {
|
||||
const URI_TYPE: gst::URIType = gst::URIType::Src;
|
||||
|
||||
fn protocols() -> &'static [&'static str] {
|
||||
&["gstwebrtc", "gstwebrtcs"]
|
||||
}
|
||||
|
||||
fn uri(&self) -> Option<String> {
|
||||
self.signaller().property::<Option<String>>("uri")
|
||||
}
|
||||
|
||||
fn set_uri(&self, uri: &str) -> Result<(), glib::Error> {
|
||||
let uri = Url::from_str(uri)
|
||||
.map_err(|err| glib::Error::new(gst::URIError::BadUri, &format!("{:?}", err)))?;
|
||||
|
||||
let socket_scheme = match uri.scheme() {
|
||||
"gstwebrtc" => Ok("ws"),
|
||||
"gstwebrtcs" => Ok("wss"),
|
||||
_ => Err(glib::Error::new(
|
||||
gst::URIError::BadUri,
|
||||
&format!("Invalid protocol: {}", uri.scheme()),
|
||||
)),
|
||||
}?;
|
||||
|
||||
let mut url_str = uri.to_string();
|
||||
|
||||
// Not using `set_scheme()` because it doesn't work with `http`
|
||||
// See https://github.com/servo/rust-url/pull/768 for a PR implementing that
|
||||
url_str.replace_range(0..uri.scheme().len(), socket_scheme);
|
||||
|
||||
self.signaller().set_property("uri", &url_str);
|
||||
|
||||
Ok(())
|
||||
}
|
||||
}
|
||||
|
||||
#[derive(PartialEq)]
|
||||
enum SignallerState {
|
||||
Started,
|
||||
|
|
@ -1186,3 +1151,77 @@ impl Default for State {
|
|||
}
|
||||
}
|
||||
}
|
||||
|
||||
#[derive(Default)]
|
||||
pub struct WebRTCSrc {}
|
||||
|
||||
impl ObjectImpl for WebRTCSrc {}
|
||||
|
||||
impl GstObjectImpl for WebRTCSrc {}
|
||||
|
||||
impl BinImpl for WebRTCSrc {}
|
||||
|
||||
impl ElementImpl for WebRTCSrc {
|
||||
fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
|
||||
static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
|
||||
gst::subclass::ElementMetadata::new(
|
||||
"WebRTCSrc",
|
||||
"Source/Network/WebRTC",
|
||||
"WebRTC src",
|
||||
"Thibault Saunier <tsaunier@igalia.com>",
|
||||
)
|
||||
});
|
||||
|
||||
Some(&*ELEMENT_METADATA)
|
||||
}
|
||||
}
|
||||
|
||||
impl BaseWebRTCSrcImpl for WebRTCSrc {}
|
||||
|
||||
impl URIHandlerImpl for WebRTCSrc {
|
||||
const URI_TYPE: gst::URIType = gst::URIType::Src;
|
||||
|
||||
fn protocols() -> &'static [&'static str] {
|
||||
&["gstwebrtc", "gstwebrtcs"]
|
||||
}
|
||||
|
||||
fn uri(&self) -> Option<String> {
|
||||
let obj = self.obj();
|
||||
let base = obj.upcast_ref::<super::BaseWebRTCSrc>().imp();
|
||||
base.signaller().property::<Option<String>>("uri")
|
||||
}
|
||||
|
||||
fn set_uri(&self, uri: &str) -> Result<(), glib::Error> {
|
||||
let uri = Url::from_str(uri)
|
||||
.map_err(|err| glib::Error::new(gst::URIError::BadUri, &format!("{:?}", err)))?;
|
||||
|
||||
let socket_scheme = match uri.scheme() {
|
||||
"gstwebrtc" => Ok("ws"),
|
||||
"gstwebrtcs" => Ok("wss"),
|
||||
_ => Err(glib::Error::new(
|
||||
gst::URIError::BadUri,
|
||||
&format!("Invalid protocol: {}", uri.scheme()),
|
||||
)),
|
||||
}?;
|
||||
|
||||
let mut url_str = uri.to_string();
|
||||
|
||||
// Not using `set_scheme()` because it doesn't work with `http`
|
||||
// See https://github.com/servo/rust-url/pull/768 for a PR implementing that
|
||||
url_str.replace_range(0..uri.scheme().len(), socket_scheme);
|
||||
|
||||
let obj = self.obj();
|
||||
let base = obj.upcast_ref::<super::BaseWebRTCSrc>().imp();
|
||||
base.signaller().set_property("uri", &url_str);
|
||||
|
||||
Ok(())
|
||||
}
|
||||
}
|
||||
|
||||
#[glib::object_subclass]
|
||||
impl ObjectSubclass for WebRTCSrc {
|
||||
const NAME: &'static str = "GstWebRTCSrc";
|
||||
type Type = super::WebRTCSrc;
|
||||
type ParentType = super::BaseWebRTCSrc;
|
||||
type Interfaces = (gst::URIHandler,);
|
||||
}
|
||||
|
|
|
|||
|
|
@ -42,7 +42,11 @@ use gst::prelude::*;
|
|||
use gst::{glib, prelude::StaticType};
|
||||
|
||||
glib::wrapper! {
|
||||
pub struct WebRTCSrc(ObjectSubclass<imp::WebRTCSrc>) @extends gst::Bin, gst::Element, gst::Object, @implements gst::URIHandler, gst::ChildProxy;
|
||||
pub struct BaseWebRTCSrc(ObjectSubclass<imp::BaseWebRTCSrc>) @extends gst::Bin, gst::Element, gst::Object, @implements gst::ChildProxy;
|
||||
}
|
||||
|
||||
glib::wrapper! {
|
||||
pub struct WebRTCSrc(ObjectSubclass<imp::WebRTCSrc>) @extends BaseWebRTCSrc, gst::Bin, gst::Element, gst::Object, @implements gst::URIHandler, gst::ChildProxy;
|
||||
}
|
||||
|
||||
glib::wrapper! {
|
||||
|
|
@ -50,6 +54,7 @@ glib::wrapper! {
|
|||
}
|
||||
|
||||
pub fn register(plugin: Option<&gst::Plugin>) -> Result<(), glib::BoolError> {
|
||||
BaseWebRTCSrc::static_type().mark_as_plugin_api(gst::PluginAPIFlags::empty());
|
||||
WebRTCSignallerRole::static_type().mark_as_plugin_api(gst::PluginAPIFlags::empty());
|
||||
WebRTCSrcPad::static_type().mark_as_plugin_api(gst::PluginAPIFlags::empty());
|
||||
Signallable::static_type().mark_as_plugin_api(gst::PluginAPIFlags::empty());
|
||||
|
|
|
|||
Loading…
Reference in a new issue