mirror of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2024-11-22 11:30:59 +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
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@ -6440,6 +6440,7 @@
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"description": "WebRTC src",
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"hierarchy": [
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"GstWebRTCSrc",
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"GstBaseWebRTCSrc",
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"GstBin",
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"GstElement",
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"GstObject",
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@ -6466,108 +6467,7 @@
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"type": "GstWebRTCSrcPad"
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}
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},
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"properties": {
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"audio-codecs": {
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"blurb": "Names of audio codecs to be be used during the SDP negotiation. Valid values: [OPUS]",
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"conditionally-available": false,
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"construct": false,
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"construct-only": false,
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"controllable": false,
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"mutable": "ready",
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"readable": true,
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"type": "GstValueArray",
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"writable": true
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},
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"enable-data-channel-navigation": {
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"blurb": "Enable navigation events through a dedicated WebRTCDataChannel",
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"conditionally-available": false,
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"construct": false,
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"construct-only": false,
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"controllable": false,
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"default": "false",
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"mutable": "ready",
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"readable": true,
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"type": "gboolean",
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"writable": true
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},
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"meta": {
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"blurb": "Free form metadata about the consumer",
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"conditionally-available": false,
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"construct": false,
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"construct-only": false,
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"controllable": false,
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"mutable": "ready",
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"readable": true,
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"type": "GstStructure",
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"writable": true
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},
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"signaller": {
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"blurb": "The Signallable object to use to handle WebRTC Signalling",
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"conditionally-available": false,
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"construct": false,
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"construct-only": true,
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"controllable": false,
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"mutable": "null",
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"readable": true,
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"type": "GstRSWebRTCSignallableIface",
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"writable": true
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},
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"stun-server": {
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"blurb": "The STUN server of the form stun://host:port",
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"conditionally-available": false,
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"construct": false,
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"construct-only": false,
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"controllable": false,
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"default": "stun://stun.l.google.com:19302",
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"mutable": "ready",
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"readable": true,
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"type": "gchararray",
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"writable": true
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},
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"turn-servers": {
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"blurb": "The TURN servers of the form <\"turn(s)://username:password@host:port\", \"turn(s)://username1:password1@host1:port1\">",
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"conditionally-available": false,
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"construct": false,
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"construct-only": false,
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"controllable": false,
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"mutable": "ready",
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"readable": true,
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"type": "GstValueArray",
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"writable": true
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},
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"video-codecs": {
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"blurb": "Names of video codecs to be be used during the SDP negotiation. Valid values: [VP8, H264, VP9, H265]",
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"conditionally-available": false,
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"construct": false,
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"construct-only": false,
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"controllable": false,
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"mutable": "ready",
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"readable": true,
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"type": "GstValueArray",
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"writable": true
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}
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},
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"rank": "primary",
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"signals": {
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"request-encoded-filter": {
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"args": [
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{
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"name": "arg0",
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"type": "gchararray"
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},
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{
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"name": "arg1",
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"type": "gchararray"
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},
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{
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"name": "arg2",
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"type": "GstCaps"
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}
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],
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"return-type": "GstElement",
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"when": "last"
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}
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}
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"rank": "primary"
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},
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"whipclientsink": {
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"author": "Taruntej Kanakamalla <taruntej@asymptotic.io>",
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@ -6890,6 +6790,121 @@
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}
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}
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},
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"GstBaseWebRTCSrc": {
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"hierarchy": [
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"GstBaseWebRTCSrc",
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"GstBin",
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"GstElement",
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"GstObject",
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"GInitiallyUnowned",
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"GObject"
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],
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"interfaces": [
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"GstChildProxy"
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],
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"kind": "object",
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"properties": {
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"audio-codecs": {
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"blurb": "Names of audio codecs to be be used during the SDP negotiation. Valid values: [OPUS]",
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"conditionally-available": false,
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"construct": false,
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"construct-only": false,
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"controllable": false,
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"mutable": "ready",
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"readable": true,
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"type": "GstValueArray",
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"writable": true
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},
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"enable-data-channel-navigation": {
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"blurb": "Enable navigation events through a dedicated WebRTCDataChannel",
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"conditionally-available": false,
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"construct": false,
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"construct-only": false,
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"controllable": false,
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"default": "false",
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"mutable": "ready",
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"readable": true,
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"type": "gboolean",
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"writable": true
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},
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"meta": {
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"blurb": "Free form metadata about the consumer",
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"conditionally-available": false,
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"construct": false,
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"construct-only": false,
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"controllable": false,
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"mutable": "ready",
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"readable": true,
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"type": "GstStructure",
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"writable": true
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},
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"signaller": {
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"blurb": "The Signallable object to use to handle WebRTC Signalling",
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"conditionally-available": false,
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"construct": false,
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"construct-only": true,
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"controllable": false,
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"mutable": "null",
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"readable": true,
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"type": "GstRSWebRTCSignallableIface",
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"writable": true
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},
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"stun-server": {
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"blurb": "The STUN server of the form stun://host:port",
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"conditionally-available": false,
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"construct": false,
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"construct-only": false,
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"controllable": false,
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"default": "stun://stun.l.google.com:19302",
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"mutable": "ready",
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"readable": true,
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"type": "gchararray",
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"writable": true
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},
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"turn-servers": {
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"blurb": "The TURN servers of the form <\"turn(s)://username:password@host:port\", \"turn(s)://username1:password1@host1:port1\">",
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"conditionally-available": false,
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"construct": false,
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"construct-only": false,
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"controllable": false,
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"mutable": "ready",
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"readable": true,
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"type": "GstValueArray",
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"writable": true
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},
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"video-codecs": {
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"blurb": "Names of video codecs to be be used during the SDP negotiation. Valid values: [VP8, H264, VP9, H265]",
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"conditionally-available": false,
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"construct": false,
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"construct-only": false,
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"controllable": false,
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"mutable": "ready",
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"readable": true,
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"type": "GstValueArray",
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"writable": true
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}
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},
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"signals": {
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"request-encoded-filter": {
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"args": [
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{
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"name": "arg0",
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"type": "gchararray"
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},
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{
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"name": "arg1",
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"type": "gchararray"
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},
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{
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"name": "arg2",
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"type": "GstCaps"
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}
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],
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"return-type": "GstElement",
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"when": "last"
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}
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}
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},
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"GstRSWebRTCSignallableIface": {
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"hierarchy": [
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"GstRSWebRTCSignallableIface",
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@ -43,7 +43,7 @@ struct Settings {
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}
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#[derive(Default)]
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pub struct WebRTCSrc {
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pub struct BaseWebRTCSrc {
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settings: Mutex<Settings>,
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n_video_pads: AtomicU16,
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n_audio_pads: AtomicU16,
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@ -51,14 +51,21 @@ pub struct WebRTCSrc {
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}
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#[glib::object_subclass]
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impl ObjectSubclass for WebRTCSrc {
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const NAME: &'static str = "GstWebRTCSrc";
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type Type = super::WebRTCSrc;
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impl ObjectSubclass for BaseWebRTCSrc {
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const NAME: &'static str = "GstBaseWebRTCSrc";
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type Type = super::BaseWebRTCSrc;
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type ParentType = gst::Bin;
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type Interfaces = (gst::URIHandler, gst::ChildProxy);
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type Interfaces = (gst::ChildProxy,);
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}
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impl ObjectImpl for WebRTCSrc {
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unsafe impl<T: BaseWebRTCSrcImpl> IsSubclassable<T> for super::BaseWebRTCSrc {
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fn class_init(class: &mut glib::Class<Self>) {
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Self::parent_class_init::<T>(class);
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}
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}
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pub(crate) trait BaseWebRTCSrcImpl: BinImpl {}
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impl ObjectImpl for BaseWebRTCSrc {
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fn properties() -> &'static [glib::ParamSpec] {
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static PROPS: Lazy<Vec<glib::ParamSpec>> = Lazy::new(|| {
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vec![
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@ -205,7 +212,7 @@ impl ObjectImpl for WebRTCSrc {
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static SIGNALS: Lazy<Vec<glib::subclass::Signal>> = Lazy::new(|| {
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vec![
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/**
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* WebRTCSrc::request-encoded-filter:
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* BaseWebRTCSrc::request-encoded-filter:
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* @producer_id: Identifier of the producer
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* @pad_name: The name of the output pad
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* @allowed_caps: the allowed caps for the output pad
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@ -278,7 +285,7 @@ struct SignallerSignals {
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handle_ice: glib::SignalHandlerId,
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}
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impl WebRTCSrc {
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impl BaseWebRTCSrc {
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fn webrtcbin(&self) -> gst::Bin {
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let state = self.state.lock().unwrap();
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let webrtcbin = state
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@ -562,15 +569,22 @@ impl WebRTCSrc {
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let mline = transceiver.map_or(mline, |t| Some(t.mlineindex()));
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// Same logic as gst_pad_create_stream_id and friends, making a hash of
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// the URI and adding `:<some-id>`, here the ID is the mline of the
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// the URI (session id, if URI doesn't exist) and adding `:<some-id>`, here the ID is the mline of the
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// stream in the SDP.
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mline.map(|mline| {
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let mut cs = glib::Checksum::new(glib::ChecksumType::Sha256).unwrap();
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cs.update(
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self.uri()
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.expect("get_stream_id should never be called if no URI has been set")
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.as_bytes(),
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);
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let data: String = if self
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.signaller()
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.has_property("uri", Some(String::static_type()))
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{
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self.signaller().property::<Option<String>>("uri").unwrap()
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} else {
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// use the session id
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self.state.lock().unwrap().session_id.clone().unwrap()
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};
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cs.update(data.as_bytes());
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format!("{}:{mline}", cs.string().unwrap())
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})
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@ -983,20 +997,7 @@ impl WebRTCSrc {
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}
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}
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impl ElementImpl for WebRTCSrc {
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fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
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static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
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gst::subclass::ElementMetadata::new(
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"WebRTCSrc",
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"Source/Network/WebRTC",
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"WebRTC src",
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"Thibault Saunier <tsaunier@igalia.com>",
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)
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});
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Some(&*ELEMENT_METADATA)
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}
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impl ElementImpl for BaseWebRTCSrc {
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fn pad_templates() -> &'static [gst::PadTemplate] {
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static PAD_TEMPLATES: Lazy<Vec<gst::PadTemplate>> = Lazy::new(|| {
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let mut video_caps_builder = gst::Caps::builder_full()
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@ -1095,11 +1096,11 @@ impl ElementImpl for WebRTCSrc {
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}
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}
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impl GstObjectImpl for WebRTCSrc {}
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impl GstObjectImpl for BaseWebRTCSrc {}
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impl BinImpl for WebRTCSrc {}
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impl BinImpl for BaseWebRTCSrc {}
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impl ChildProxyImpl for WebRTCSrc {
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impl ChildProxyImpl for BaseWebRTCSrc {
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fn child_by_index(&self, index: u32) -> Option<glib::Object> {
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if index == 0 {
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Some(self.signaller().upcast())
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@ -1123,42 +1124,6 @@ impl ChildProxyImpl for WebRTCSrc {
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}
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}
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impl URIHandlerImpl for WebRTCSrc {
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const URI_TYPE: gst::URIType = gst::URIType::Src;
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fn protocols() -> &'static [&'static str] {
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&["gstwebrtc", "gstwebrtcs"]
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}
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fn uri(&self) -> Option<String> {
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self.signaller().property::<Option<String>>("uri")
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}
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fn set_uri(&self, uri: &str) -> Result<(), glib::Error> {
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let uri = Url::from_str(uri)
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.map_err(|err| glib::Error::new(gst::URIError::BadUri, &format!("{:?}", err)))?;
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let socket_scheme = match uri.scheme() {
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"gstwebrtc" => Ok("ws"),
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"gstwebrtcs" => Ok("wss"),
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_ => Err(glib::Error::new(
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gst::URIError::BadUri,
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&format!("Invalid protocol: {}", uri.scheme()),
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)),
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}?;
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let mut url_str = uri.to_string();
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// Not using `set_scheme()` because it doesn't work with `http`
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// See https://github.com/servo/rust-url/pull/768 for a PR implementing that
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url_str.replace_range(0..uri.scheme().len(), socket_scheme);
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self.signaller().set_property("uri", &url_str);
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Ok(())
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}
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}
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#[derive(PartialEq)]
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enum SignallerState {
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Started,
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@ -1186,3 +1151,77 @@ impl Default for State {
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}
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}
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}
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#[derive(Default)]
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pub struct WebRTCSrc {}
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impl ObjectImpl for WebRTCSrc {}
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impl GstObjectImpl for WebRTCSrc {}
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impl BinImpl for WebRTCSrc {}
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impl ElementImpl for WebRTCSrc {
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fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
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static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
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gst::subclass::ElementMetadata::new(
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"WebRTCSrc",
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"Source/Network/WebRTC",
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"WebRTC src",
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"Thibault Saunier <tsaunier@igalia.com>",
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)
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});
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Some(&*ELEMENT_METADATA)
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}
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}
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impl BaseWebRTCSrcImpl for WebRTCSrc {}
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impl URIHandlerImpl for WebRTCSrc {
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const URI_TYPE: gst::URIType = gst::URIType::Src;
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fn protocols() -> &'static [&'static str] {
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&["gstwebrtc", "gstwebrtcs"]
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}
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fn uri(&self) -> Option<String> {
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let obj = self.obj();
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let base = obj.upcast_ref::<super::BaseWebRTCSrc>().imp();
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base.signaller().property::<Option<String>>("uri")
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}
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fn set_uri(&self, uri: &str) -> Result<(), glib::Error> {
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let uri = Url::from_str(uri)
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.map_err(|err| glib::Error::new(gst::URIError::BadUri, &format!("{:?}", err)))?;
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let socket_scheme = match uri.scheme() {
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"gstwebrtc" => Ok("ws"),
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"gstwebrtcs" => Ok("wss"),
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_ => Err(glib::Error::new(
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gst::URIError::BadUri,
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&format!("Invalid protocol: {}", uri.scheme()),
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)),
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}?;
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let mut url_str = uri.to_string();
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// Not using `set_scheme()` because it doesn't work with `http`
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// See https://github.com/servo/rust-url/pull/768 for a PR implementing that
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url_str.replace_range(0..uri.scheme().len(), socket_scheme);
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let obj = self.obj();
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let base = obj.upcast_ref::<super::BaseWebRTCSrc>().imp();
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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