mirror of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2024-05-20 09:18:15 +00:00
webrtc: extract a BaseWebRTCSink
For documentation purposes, AwsKVSWebRTCSink should not inherit from another element. + Mark base class as plugin API and update plugin cache Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
This commit is contained in:
Mathieu Duponchelle
GStreamer Marge Bot
parent
367b98bfcb
commit
f1fd8d84c3
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@ -6065,10 +6065,10 @@
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"elements": {
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"awskvswebrtcsink": {
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"author": "Mathieu Duponchelle <mathieu@centricular.com>",
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"description": "WebRTC sink",
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"description": "WebRTC sink with kinesis video streams signaller",
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"hierarchy": [
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"GstAwsKvsWebRTCSink",
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"GstWebRTCSink",
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"GstBaseWebRTCSink",
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"GstBin",
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"GstElement",
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"GstObject",
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@ -6096,9 +6096,10 @@
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},
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"webrtcsink": {
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"author": "Mathieu Duponchelle <mathieu@centricular.com>",
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"description": "WebRTC sink",
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"description": "WebRTC sink with custom protocol signaller",
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"hierarchy": [
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"GstWebRTCSink",
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"GstBaseWebRTCSink",
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"GstBin",
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"GstElement",
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"GstObject",
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@ -6110,7 +6111,6 @@
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"GstNavigation"
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],
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"klass": "Sink/Network/WebRTC",
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"long-name": "WebRTCSink",
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"pad-templates": {
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"audio_%%u": {
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"caps": "audio/x-raw:\n",
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@ -6123,6 +6123,117 @@
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"presence": "request"
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}
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},
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"rank": "none"
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},
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"webrtcsrc": {
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"author": "Thibault Saunier <tsaunier@igalia.com>",
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"description": "WebRTC src",
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"hierarchy": [
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"GstWebRTCSrc",
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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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"GstURIHandler"
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],
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"klass": "Source/Network/WebRTC",
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"long-name": "WebRTCSrc",
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"pad-templates": {
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"audio_%%u": {
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"caps": "audio/x-raw(ANY):\naudio/x-opus:\napplication/x-rtp:\n",
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"direction": "src",
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"presence": "sometimes",
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"type": "GstWebRTCSrcPad"
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},
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"video_%%u": {
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"caps": "video/x-raw(ANY):\napplication/x-rtp:\n",
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"direction": "src",
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"presence": "sometimes",
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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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"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": false,
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"controllable": false,
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"mutable": "ready",
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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": "NULL",
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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": "null",
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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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"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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}
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},
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"filename": "gstrswebrtc",
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"license": "MPL-2.0",
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"other-types": {
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"GstBaseWebRTCSink": {
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"hierarchy": [
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"GstBaseWebRTCSink",
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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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"GstNavigation"
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],
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"kind": "object",
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"properties": {
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"audio-caps": {
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"blurb": "Governs what audio codecs will be proposed",
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@ -6308,7 +6419,6 @@
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"writable": true
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}
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},
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"rank": "none",
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"signals": {
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"consumer-added": {
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"args": [
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@ -6364,101 +6474,6 @@
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}
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}
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},
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"webrtcsrc": {
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"author": "Thibault Saunier <tsaunier@igalia.com>",
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"description": "WebRTC src",
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"hierarchy": [
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"GstWebRTCSrc",
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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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"GstURIHandler"
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],
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"klass": "Source/Network/WebRTC",
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"long-name": "WebRTCSrc",
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"pad-templates": {
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"audio_%%u": {
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"caps": "audio/x-raw(ANY):\naudio/x-opus:\napplication/x-rtp:\n",
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"direction": "src",
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"presence": "sometimes",
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"type": "GstWebRTCSrcPad"
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},
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"video_%%u": {
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"caps": "video/x-raw(ANY):\napplication/x-rtp:\n",
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"direction": "src",
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"presence": "sometimes",
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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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"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": false,
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"controllable": false,
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"mutable": "ready",
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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": "NULL",
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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": "null",
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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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"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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}
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},
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"filename": "gstrswebrtc",
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"license": "MPL-2.0",
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"other-types": {
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"GstRSWebRTCSignallableIface": {
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"hierarchy": [
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"GstRSWebRTCSignallableIface",
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@ -110,7 +110,7 @@ impl CongestionController {
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fn update_delay(
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&mut self,
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element: &super::WebRTCSink,
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element: &super::BaseWebRTCSink,
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twcc_stats: &gst::StructureRef,
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rtt: f64,
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) -> CongestionControlOp {
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@ -291,7 +291,7 @@ impl CongestionController {
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pub fn loss_control(
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&mut self,
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element: &super::WebRTCSink,
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element: &super::BaseWebRTCSink,
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stats: &gst::StructureRef,
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encoders: &mut Vec<VideoEncoder>,
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) {
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@ -316,7 +316,7 @@ impl CongestionController {
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pub fn delay_control(
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&mut self,
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element: &super::WebRTCSink,
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element: &super::BaseWebRTCSink,
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stats: &gst::StructureRef,
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encoders: &mut Vec<VideoEncoder>,
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) {
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@ -328,7 +328,7 @@ impl CongestionController {
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fn apply_control_op(
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&mut self,
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element: &super::WebRTCSink,
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element: &super::BaseWebRTCSink,
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encoders: &mut Vec<VideoEncoder>,
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control_op: CongestionControlOp,
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controller_type: ControllerType,
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@ -220,7 +220,7 @@ struct State {
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signaller_signals: Option<SignallerSignals>,
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}
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fn create_navigation_event(sink: &super::WebRTCSink, msg: &str) {
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fn create_navigation_event(sink: &super::BaseWebRTCSink, msg: &str) {
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let event: Result<NavigationEvent, _> = serde_json::from_str(msg);
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if let Ok(event) = event {
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@ -281,7 +281,7 @@ struct NavigationEventHandler((glib::SignalHandlerId, WebRTCDataChannel));
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/// Our instance structure
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#[derive(Default)]
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pub struct WebRTCSink {
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pub struct BaseWebRTCSink {
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state: Mutex<State>,
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settings: Mutex<Settings>,
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}
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@ -486,7 +486,7 @@ fn configure_encoder(enc: &gst::Element, start_bitrate: u32) {
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/// the codec discovery code, and this gets the job done.
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#[allow(clippy::too_many_arguments)] // This needs some more refactoring and it will happen soon
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fn setup_encoding(
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element: &super::WebRTCSink,
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element: &super::BaseWebRTCSink,
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pipeline: &gst::Pipeline,
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src: &gst::Element,
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input_caps: &gst::Caps,
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@ -681,7 +681,7 @@ impl VideoEncoder {
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(width + 1) & !1
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}
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pub(crate) fn set_bitrate(&mut self, element: &super::WebRTCSink, bitrate: i32) {
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pub(crate) fn set_bitrate(&mut self, element: &super::BaseWebRTCSink, bitrate: i32) {
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match self.factory_name.as_str() {
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"vp8enc" | "vp9enc" => self.element.set_property("target-bitrate", bitrate),
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"x264enc" | "nvh264enc" | "vaapih264enc" | "vaapivp8enc" => self
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@ -795,7 +795,7 @@ impl State {
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}
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}
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fn should_start_signaller(&mut self, element: &super::WebRTCSink) -> bool {
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fn should_start_signaller(&mut self, element: &super::BaseWebRTCSink) -> bool {
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self.signaller_state == SignallerState::Stopped
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&& element.current_state() >= gst::State::Paused
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&& self.codec_discovery_done
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@ -854,7 +854,7 @@ impl Session {
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/// to a given WebRTCPad
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fn connect_input_stream(
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&mut self,
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element: &super::WebRTCSink,
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element: &super::BaseWebRTCSink,
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producer: &StreamProducer,
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webrtc_pad: &WebRTCPad,
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codecs: &BTreeMap<i32, Codec>,
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@ -1045,7 +1045,7 @@ impl Drop for PipelineWrapper {
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impl InputStream {
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/// Called when transitioning state up to Paused
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fn prepare(&mut self, element: &super::WebRTCSink) -> Result<(), Error> {
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fn prepare(&mut self, element: &super::BaseWebRTCSink) -> Result<(), Error> {
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let clocksync = make_element("clocksync", None)?;
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let appsink = make_element("appsink", None)?
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.downcast::<gst_app::AppSink>()
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@ -1072,7 +1072,7 @@ impl InputStream {
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}
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/// Called when transitioning state back down to Ready
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fn unprepare(&mut self, element: &super::WebRTCSink) {
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fn unprepare(&mut self, element: &super::BaseWebRTCSink) {
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self.sink_pad.set_target(None::<&gst::Pad>).unwrap();
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if let Some(clocksync) = self.clocksync.take() {
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@ -1089,7 +1089,7 @@ impl InputStream {
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}
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impl NavigationEventHandler {
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fn new(element: &super::WebRTCSink, webrtcbin: &gst::Element) -> Self {
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fn new(element: &super::BaseWebRTCSink, webrtcbin: &gst::Element) -> Self {
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gst::info!(CAT, "Creating navigation data channel");
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let channel = webrtcbin.emit_by_name::<WebRTCDataChannel>(
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"create-data-channel",
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@ -1117,8 +1117,11 @@ impl NavigationEventHandler {
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}
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}
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impl WebRTCSink {
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fn generate_ssrc(element: &super::WebRTCSink, webrtc_pads: &HashMap<u32, WebRTCPad>) -> u32 {
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impl BaseWebRTCSink {
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fn generate_ssrc(
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element: &super::BaseWebRTCSink,
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webrtc_pads: &HashMap<u32, WebRTCPad>,
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) -> u32 {
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loop {
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let ret = fastrand::u32(..);
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@ -1130,12 +1133,12 @@ impl WebRTCSink {
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}
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fn request_inactive_webrtcbin_pad(
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element: &super::WebRTCSink,
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element: &super::BaseWebRTCSink,
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webrtcbin: &gst::Element,
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webrtc_pads: &mut HashMap<u32, WebRTCPad>,
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is_video: bool,
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) {
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let ssrc = WebRTCSink::generate_ssrc(element, webrtc_pads);
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let ssrc = BaseWebRTCSink::generate_ssrc(element, webrtc_pads);
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let media_idx = webrtc_pads.len() as i32;
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let pad = webrtcbin
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@ -1169,7 +1172,7 @@ impl WebRTCSink {
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}
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async fn request_webrtcbin_pad(
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element: &super::WebRTCSink,
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element: &super::BaseWebRTCSink,
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webrtcbin: &gst::Element,
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stream: &InputStream,
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media: Option<&gst_sdp::SDPMediaRef>,
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@ -1177,7 +1180,7 @@ impl WebRTCSink {
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webrtc_pads: &mut HashMap<u32, WebRTCPad>,
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codecs: &mut BTreeMap<i32, Codec>,
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) {
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let ssrc = WebRTCSink::generate_ssrc(element, webrtc_pads);
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let ssrc = BaseWebRTCSink::generate_ssrc(element, webrtc_pads);
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let media_idx = webrtc_pads.len() as i32;
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let mut payloader_caps = match media {
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|
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@ -1192,7 +1195,7 @@ impl WebRTCSink {
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gst::Rank::Marginal,
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);
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let codec = WebRTCSink::select_codec(
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let codec = BaseWebRTCSink::select_codec(
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element,
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&encoders,
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&payloaders,
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|
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@ -1224,7 +1227,7 @@ impl WebRTCSink {
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};
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if payloader_caps.is_empty() {
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WebRTCSink::request_inactive_webrtcbin_pad(
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BaseWebRTCSink::request_inactive_webrtcbin_pad(
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element,
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webrtcbin,
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webrtc_pads,
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@ -1336,7 +1339,7 @@ impl WebRTCSink {
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/// Prepare for accepting consumers, by setting
|
||||
/// up StreamProducers for each of our sink pads
|
||||
fn prepare(&self, element: &super::WebRTCSink) -> Result<(), Error> {
|
||||
fn prepare(&self, element: &super::BaseWebRTCSink) -> Result<(), Error> {
|
||||
gst::debug!(CAT, obj: element, "preparing");
|
||||
|
||||
self.state
|
||||
|
|
@ -1351,7 +1354,7 @@ impl WebRTCSink {
|
|||
|
||||
/// Unprepare by stopping consumers, then the signaller object.
|
||||
/// Might abort codec discovery
|
||||
fn unprepare(&self, element: &super::WebRTCSink) -> Result<(), Error> {
|
||||
fn unprepare(&self, element: &super::BaseWebRTCSink) -> Result<(), Error> {
|
||||
gst::info!(CAT, obj: element, "unpreparing");
|
||||
|
||||
let settings = self.settings.lock().unwrap();
|
||||
|
|
@ -1503,14 +1506,14 @@ impl WebRTCSink {
|
|||
}
|
||||
|
||||
/// Called by the signaller when it wants to shut down gracefully
|
||||
fn shutdown(&self, element: &super::WebRTCSink) {
|
||||
fn shutdown(&self, element: &super::BaseWebRTCSink) {
|
||||
gst::info!(CAT, "Shutting down");
|
||||
let _ = element.post_message(gst::message::Eos::builder().src(element).build());
|
||||
}
|
||||
|
||||
fn on_offer_created(
|
||||
&self,
|
||||
_element: &super::WebRTCSink,
|
||||
_element: &super::BaseWebRTCSink,
|
||||
offer: gst_webrtc::WebRTCSessionDescription,
|
||||
session_id: &str,
|
||||
) {
|
||||
|
|
@ -1531,7 +1534,7 @@ impl WebRTCSink {
|
|||
|
||||
fn on_answer_created(
|
||||
&self,
|
||||
element: &super::WebRTCSink,
|
||||
element: &super::BaseWebRTCSink,
|
||||
answer: gst_webrtc::WebRTCSessionDescription,
|
||||
session_id: &str,
|
||||
) {
|
||||
|
|
@ -1567,7 +1570,7 @@ impl WebRTCSink {
|
|||
}
|
||||
}
|
||||
|
||||
fn on_remote_description_offer_set(&self, element: &super::WebRTCSink, session_id: String) {
|
||||
fn on_remote_description_offer_set(&self, element: &super::BaseWebRTCSink, session_id: String) {
|
||||
let state = self.state.lock().unwrap();
|
||||
|
||||
if let Some(session) = state.sessions.get(&session_id) {
|
||||
|
|
@ -1663,7 +1666,7 @@ impl WebRTCSink {
|
|||
}
|
||||
|
||||
async fn select_codec(
|
||||
element: &super::WebRTCSink,
|
||||
element: &super::BaseWebRTCSink,
|
||||
encoders: &gst::glib::List<gst::ElementFactory>,
|
||||
payloaders: &gst::glib::List<gst::ElementFactory>,
|
||||
media: &gst_sdp::SDPMediaRef,
|
||||
|
|
@ -1713,7 +1716,7 @@ impl WebRTCSink {
|
|||
let encoding_name = s.get::<String>("encoding-name").unwrap();
|
||||
|
||||
if let Some(codec) =
|
||||
WebRTCSink::build_codec(&encoding_name, payload, encoders, payloaders)
|
||||
BaseWebRTCSink::build_codec(&encoding_name, payload, encoders, payloaders)
|
||||
{
|
||||
for (user_caps, codecs_and_caps) in ordered_codecs_and_caps.iter_mut() {
|
||||
if codec.caps.is_subset(user_caps) {
|
||||
|
|
@ -1751,7 +1754,7 @@ impl WebRTCSink {
|
|||
.iter()
|
||||
.flat_map(|(_, codecs_and_caps)| codecs_and_caps)
|
||||
.map(|(codec, caps)| async move {
|
||||
WebRTCSink::run_discovery_pipeline(element, codec, in_caps, caps, twcc_idx)
|
||||
BaseWebRTCSink::run_discovery_pipeline(element, codec, in_caps, caps, twcc_idx)
|
||||
.await
|
||||
.map(|s| {
|
||||
let mut codec = codec.clone();
|
||||
|
|
@ -1772,7 +1775,7 @@ impl WebRTCSink {
|
|||
|
||||
fn negotiate(
|
||||
&self,
|
||||
element: &super::WebRTCSink,
|
||||
element: &super::BaseWebRTCSink,
|
||||
session_id: &str,
|
||||
offer: Option<&gst_webrtc::WebRTCSessionDescription>,
|
||||
) {
|
||||
|
|
@ -1866,7 +1869,7 @@ impl WebRTCSink {
|
|||
|
||||
fn on_ice_candidate(
|
||||
&self,
|
||||
_element: &super::WebRTCSink,
|
||||
_element: &super::BaseWebRTCSink,
|
||||
session_id: String,
|
||||
sdp_m_line_index: u32,
|
||||
candidate: String,
|
||||
|
|
@ -2267,7 +2270,7 @@ impl WebRTCSink {
|
|||
media_is_video == stream_is_video
|
||||
}) {
|
||||
let stream = streams.remove(idx);
|
||||
WebRTCSink::request_webrtcbin_pad(
|
||||
BaseWebRTCSink::request_webrtcbin_pad(
|
||||
&element,
|
||||
&webrtcbin,
|
||||
&stream,
|
||||
|
|
@ -2278,7 +2281,7 @@ impl WebRTCSink {
|
|||
)
|
||||
.await;
|
||||
} else {
|
||||
WebRTCSink::request_inactive_webrtcbin_pad(
|
||||
BaseWebRTCSink::request_inactive_webrtcbin_pad(
|
||||
&element,
|
||||
&webrtcbin,
|
||||
&mut webrtc_pads,
|
||||
|
|
@ -2288,7 +2291,7 @@ impl WebRTCSink {
|
|||
}
|
||||
} else {
|
||||
for stream in streams {
|
||||
WebRTCSink::request_webrtcbin_pad(
|
||||
BaseWebRTCSink::request_webrtcbin_pad(
|
||||
&element,
|
||||
&webrtcbin,
|
||||
&stream,
|
||||
|
|
@ -2359,7 +2362,7 @@ impl WebRTCSink {
|
|||
/// Called by the signaller to remove a consumer
|
||||
fn remove_session(
|
||||
&self,
|
||||
element: &super::WebRTCSink,
|
||||
element: &super::BaseWebRTCSink,
|
||||
session_id: &str,
|
||||
signal: bool,
|
||||
) -> Result<(), WebRTCSinkError> {
|
||||
|
|
@ -2387,7 +2390,7 @@ impl WebRTCSink {
|
|||
|
||||
fn process_loss_stats(
|
||||
&self,
|
||||
element: &super::WebRTCSink,
|
||||
element: &super::BaseWebRTCSink,
|
||||
session_id: &str,
|
||||
stats: &gst::Structure,
|
||||
) {
|
||||
|
|
@ -2402,7 +2405,7 @@ impl WebRTCSink {
|
|||
|
||||
fn process_stats(
|
||||
&self,
|
||||
element: &super::WebRTCSink,
|
||||
element: &super::BaseWebRTCSink,
|
||||
webrtcbin: gst::Element,
|
||||
session_id: &str,
|
||||
) {
|
||||
|
|
@ -2425,7 +2428,12 @@ impl WebRTCSink {
|
|||
webrtcbin.emit_by_name::<()>("get-stats", &[&None::<gst::Pad>, &promise]);
|
||||
}
|
||||
|
||||
fn set_rtptrxsend(&self, element: &super::WebRTCSink, peer_id: &str, rtprtxsend: gst::Element) {
|
||||
fn set_rtptrxsend(
|
||||
&self,
|
||||
element: &super::BaseWebRTCSink,
|
||||
peer_id: &str,
|
||||
rtprtxsend: gst::Element,
|
||||
) {
|
||||
let mut state = element.imp().state.lock().unwrap();
|
||||
|
||||
if let Some(session) = state.sessions.get_mut(peer_id) {
|
||||
|
|
@ -2433,7 +2441,7 @@ impl WebRTCSink {
|
|||
}
|
||||
}
|
||||
|
||||
fn set_bitrate(&self, element: &super::WebRTCSink, peer_id: &str, bitrate: u32) {
|
||||
fn set_bitrate(&self, element: &super::BaseWebRTCSink, peer_id: &str, bitrate: u32) {
|
||||
let settings = element.imp().settings.lock().unwrap();
|
||||
let mut state = element.imp().state.lock().unwrap();
|
||||
|
||||
|
|
@ -2467,7 +2475,7 @@ impl WebRTCSink {
|
|||
}
|
||||
}
|
||||
|
||||
fn on_remote_description_set(&self, element: &super::WebRTCSink, session_id: String) {
|
||||
fn on_remote_description_set(&self, element: &super::BaseWebRTCSink, session_id: String) {
|
||||
let mut state = self.state.lock().unwrap();
|
||||
let mut remove = false;
|
||||
let codecs = state.codecs.clone();
|
||||
|
|
@ -2587,7 +2595,7 @@ impl WebRTCSink {
|
|||
|
||||
fn handle_sdp_answer(
|
||||
&self,
|
||||
element: &super::WebRTCSink,
|
||||
element: &super::BaseWebRTCSink,
|
||||
session_id: &str,
|
||||
desc: &gst_webrtc::WebRTCSessionDescription,
|
||||
) {
|
||||
|
|
@ -2681,7 +2689,7 @@ impl WebRTCSink {
|
|||
}
|
||||
|
||||
async fn run_discovery_pipeline(
|
||||
element: &super::WebRTCSink,
|
||||
element: &super::BaseWebRTCSink,
|
||||
codec: &Codec,
|
||||
caps: &gst::Caps,
|
||||
output_caps: &gst::Caps,
|
||||
|
|
@ -2802,7 +2810,7 @@ impl WebRTCSink {
|
|||
}
|
||||
|
||||
async fn lookup_caps(
|
||||
element: &super::WebRTCSink,
|
||||
element: &super::BaseWebRTCSink,
|
||||
name: String,
|
||||
in_caps: gst::Caps,
|
||||
output_caps: gst::Caps,
|
||||
|
|
@ -2823,7 +2831,7 @@ impl WebRTCSink {
|
|||
.iter()
|
||||
.filter(|(_, codec)| codec.is_video() == is_video)
|
||||
.map(|(_, codec)| {
|
||||
WebRTCSink::run_discovery_pipeline(
|
||||
BaseWebRTCSink::run_discovery_pipeline(
|
||||
element,
|
||||
codec,
|
||||
&sink_caps,
|
||||
|
|
@ -2854,7 +2862,7 @@ impl WebRTCSink {
|
|||
(name, payloader_caps)
|
||||
}
|
||||
|
||||
async fn lookup_streams_caps(&self, element: &super::WebRTCSink) -> Result<(), Error> {
|
||||
async fn lookup_streams_caps(&self, element: &super::BaseWebRTCSink) -> Result<(), Error> {
|
||||
let codecs = self.lookup_codecs();
|
||||
|
||||
gst::debug!(CAT, obj: element, "Looked up codecs {codecs:?}");
|
||||
|
|
@ -2866,7 +2874,7 @@ impl WebRTCSink {
|
|||
.streams
|
||||
.iter()
|
||||
.map(|(name, stream)| {
|
||||
WebRTCSink::lookup_caps(
|
||||
BaseWebRTCSink::lookup_caps(
|
||||
element,
|
||||
name.to_owned(),
|
||||
stream.in_caps.as_ref().unwrap().to_owned(),
|
||||
|
|
@ -2907,7 +2915,12 @@ impl WebRTCSink {
|
|||
)
|
||||
}
|
||||
|
||||
fn sink_event(&self, pad: &gst::Pad, element: &super::WebRTCSink, event: gst::Event) -> bool {
|
||||
fn sink_event(
|
||||
&self,
|
||||
pad: &gst::Pad,
|
||||
element: &super::BaseWebRTCSink,
|
||||
event: gst::Event,
|
||||
) -> bool {
|
||||
use gst::EventView;
|
||||
|
||||
match event.view() {
|
||||
|
|
@ -2995,22 +3008,22 @@ impl WebRTCSink {
|
|||
}
|
||||
|
||||
#[glib::object_subclass]
|
||||
impl ObjectSubclass for WebRTCSink {
|
||||
const NAME: &'static str = "GstWebRTCSink";
|
||||
type Type = super::WebRTCSink;
|
||||
impl ObjectSubclass for BaseWebRTCSink {
|
||||
const NAME: &'static str = "GstBaseWebRTCSink";
|
||||
type Type = super::BaseWebRTCSink;
|
||||
type ParentType = gst::Bin;
|
||||
type Interfaces = (gst::ChildProxy, gst_video::Navigation);
|
||||
}
|
||||
|
||||
unsafe impl<T: WebRTCSinkImpl> IsSubclassable<T> for super::WebRTCSink {
|
||||
unsafe impl<T: BaseWebRTCSinkImpl> IsSubclassable<T> for super::BaseWebRTCSink {
|
||||
fn class_init(class: &mut glib::Class<Self>) {
|
||||
Self::parent_class_init::<T>(class);
|
||||
}
|
||||
}
|
||||
|
||||
pub(crate) trait WebRTCSinkImpl: BinImpl {}
|
||||
pub(crate) trait BaseWebRTCSinkImpl: BinImpl {}
|
||||
|
||||
impl ObjectImpl for WebRTCSink {
|
||||
impl ObjectImpl for BaseWebRTCSink {
|
||||
fn properties() -> &'static [glib::ParamSpec] {
|
||||
static PROPERTIES: Lazy<Vec<glib::ParamSpec>> = Lazy::new(|| {
|
||||
vec![
|
||||
|
|
@ -3247,7 +3260,7 @@ impl ObjectImpl for WebRTCSink {
|
|||
static SIGNALS: Lazy<Vec<glib::subclass::Signal>> = Lazy::new(|| {
|
||||
vec![
|
||||
/**
|
||||
* RsWebRTCSink::consumer-added:
|
||||
* RsBaseWebRTCSink::consumer-added:
|
||||
* @consumer_id: Identifier of the consumer added
|
||||
* @webrtcbin: The new webrtcbin
|
||||
*
|
||||
|
|
@ -3258,7 +3271,7 @@ impl ObjectImpl for WebRTCSink {
|
|||
.param_types([String::static_type(), gst::Element::static_type()])
|
||||
.build(),
|
||||
/**
|
||||
* RsWebRTCSink::consumer_removed:
|
||||
* RsBaseWebRTCSink::consumer_removed:
|
||||
* @consumer_id: Identifier of the consumer that was removed
|
||||
* @webrtcbin: The webrtcbin connected to the newly removed consumer
|
||||
*
|
||||
|
|
@ -3269,14 +3282,14 @@ impl ObjectImpl for WebRTCSink {
|
|||
.param_types([String::static_type(), gst::Element::static_type()])
|
||||
.build(),
|
||||
/**
|
||||
* RsWebRTCSink::get_sessions:
|
||||
* RsBaseWebRTCSink::get_sessions:
|
||||
*
|
||||
* List all sessions (by ID).
|
||||
*/
|
||||
glib::subclass::Signal::builder("get-sessions")
|
||||
.action()
|
||||
.class_handler(|_, args| {
|
||||
let element = args[0].get::<super::WebRTCSink>().expect("signal arg");
|
||||
let element = args[0].get::<super::BaseWebRTCSink>().expect("signal arg");
|
||||
let this = element.imp();
|
||||
|
||||
let res = Some(
|
||||
|
|
@ -3294,7 +3307,7 @@ impl ObjectImpl for WebRTCSink {
|
|||
.return_type::<Vec<String>>()
|
||||
.build(),
|
||||
/**
|
||||
* RsWebRTCSink::encoder-setup:
|
||||
* RsBaseWebRTCSink::encoder-setup:
|
||||
* @consumer_id: Identifier of the consumer
|
||||
* @pad_name: The name of the corresponding input pad
|
||||
* @encoder: The constructed encoder
|
||||
|
|
@ -3313,7 +3326,7 @@ impl ObjectImpl for WebRTCSink {
|
|||
.return_type::<bool>()
|
||||
.accumulator(|_hint, _ret, value| !value.get::<bool>().unwrap())
|
||||
.class_handler(|_, args| {
|
||||
let element = args[0].get::<super::WebRTCSink>().expect("signal arg");
|
||||
let element = args[0].get::<super::BaseWebRTCSink>().expect("signal arg");
|
||||
let enc = args[3].get::<gst::Element>().unwrap();
|
||||
|
||||
gst::debug!(
|
||||
|
|
@ -3349,22 +3362,9 @@ impl ObjectImpl for WebRTCSink {
|
|||
}
|
||||
}
|
||||
|
||||
impl GstObjectImpl for WebRTCSink {}
|
||||
|
||||
impl ElementImpl for WebRTCSink {
|
||||
fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
|
||||
static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
|
||||
gst::subclass::ElementMetadata::new(
|
||||
"WebRTCSink",
|
||||
"Sink/Network/WebRTC",
|
||||
"WebRTC sink",
|
||||
"Mathieu Duponchelle <mathieu@centricular.com>",
|
||||
)
|
||||
});
|
||||
|
||||
Some(&*ELEMENT_METADATA)
|
||||
}
|
||||
impl GstObjectImpl for BaseWebRTCSink {}
|
||||
|
||||
impl ElementImpl for BaseWebRTCSink {
|
||||
fn pad_templates() -> &'static [gst::PadTemplate] {
|
||||
static PAD_TEMPLATES: Lazy<Vec<gst::PadTemplate>> = Lazy::new(|| {
|
||||
let caps = gst::Caps::builder_full()
|
||||
|
|
@ -3436,7 +3436,7 @@ impl ElementImpl for WebRTCSink {
|
|||
|
||||
let sink_pad = gst::GhostPad::builder_with_template(templ, Some(name.as_str()))
|
||||
.event_function(|pad, parent, event| {
|
||||
WebRTCSink::catch_panic_pad_function(
|
||||
BaseWebRTCSink::catch_panic_pad_function(
|
||||
parent,
|
||||
|| false,
|
||||
|this| this.sink_event(pad.upcast_ref(), &this.obj(), event),
|
||||
|
|
@ -3514,9 +3514,9 @@ impl ElementImpl for WebRTCSink {
|
|||
}
|
||||
}
|
||||
|
||||
impl BinImpl for WebRTCSink {}
|
||||
impl BinImpl for BaseWebRTCSink {}
|
||||
|
||||
impl ChildProxyImpl for WebRTCSink {
|
||||
impl ChildProxyImpl for BaseWebRTCSink {
|
||||
fn child_by_index(&self, _index: u32) -> Option<glib::Object> {
|
||||
None
|
||||
}
|
||||
|
|
@ -3533,7 +3533,7 @@ impl ChildProxyImpl for WebRTCSink {
|
|||
}
|
||||
}
|
||||
|
||||
impl NavigationImpl for WebRTCSink {
|
||||
impl NavigationImpl for BaseWebRTCSink {
|
||||
fn send_event(&self, event_def: gst::Structure) {
|
||||
let mut state = self.state.lock().unwrap();
|
||||
let event = gst::event::Navigation::new(event_def);
|
||||
|
|
@ -3550,13 +3550,46 @@ impl NavigationImpl for WebRTCSink {
|
|||
}
|
||||
}
|
||||
|
||||
#[derive(Default)]
|
||||
pub struct WebRTCSink {}
|
||||
|
||||
impl ObjectImpl for WebRTCSink {}
|
||||
|
||||
impl GstObjectImpl for WebRTCSink {}
|
||||
|
||||
impl ElementImpl for WebRTCSink {
|
||||
fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
|
||||
static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
|
||||
gst::subclass::ElementMetadata::new(
|
||||
"WebRTCSink",
|
||||
"Sink/Network/WebRTC",
|
||||
"WebRTC sink with custom protocol signaller",
|
||||
"Mathieu Duponchelle <mathieu@centricular.com>",
|
||||
)
|
||||
});
|
||||
|
||||
Some(&*ELEMENT_METADATA)
|
||||
}
|
||||
}
|
||||
|
||||
impl BinImpl for WebRTCSink {}
|
||||
|
||||
impl BaseWebRTCSinkImpl for WebRTCSink {}
|
||||
|
||||
#[glib::object_subclass]
|
||||
impl ObjectSubclass for WebRTCSink {
|
||||
const NAME: &'static str = "GstWebRTCSink";
|
||||
type Type = super::WebRTCSink;
|
||||
type ParentType = super::BaseWebRTCSink;
|
||||
}
|
||||
|
||||
#[derive(Default)]
|
||||
pub struct AwsKvsWebRTCSink {}
|
||||
|
||||
impl ObjectImpl for AwsKvsWebRTCSink {
|
||||
fn constructed(&self) {
|
||||
let element = self.obj();
|
||||
let ws = element.upcast_ref::<super::WebRTCSink>().imp();
|
||||
let ws = element.upcast_ref::<super::BaseWebRTCSink>().imp();
|
||||
|
||||
let _ = ws.set_signaller(AwsKvsSignaller::default().upcast());
|
||||
}
|
||||
|
|
@ -3564,15 +3597,28 @@ impl ObjectImpl for AwsKvsWebRTCSink {
|
|||
|
||||
impl GstObjectImpl for AwsKvsWebRTCSink {}
|
||||
|
||||
impl ElementImpl for AwsKvsWebRTCSink {}
|
||||
impl ElementImpl for AwsKvsWebRTCSink {
|
||||
fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
|
||||
static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
|
||||
gst::subclass::ElementMetadata::new(
|
||||
"AwsKvsWebRTCSink",
|
||||
"Sink/Network/WebRTC",
|
||||
"WebRTC sink with kinesis video streams signaller",
|
||||
"Mathieu Duponchelle <mathieu@centricular.com>",
|
||||
)
|
||||
});
|
||||
|
||||
Some(&*ELEMENT_METADATA)
|
||||
}
|
||||
}
|
||||
|
||||
impl BinImpl for AwsKvsWebRTCSink {}
|
||||
|
||||
impl WebRTCSinkImpl for AwsKvsWebRTCSink {}
|
||||
impl BaseWebRTCSinkImpl for AwsKvsWebRTCSink {}
|
||||
|
||||
#[glib::object_subclass]
|
||||
impl ObjectSubclass for AwsKvsWebRTCSink {
|
||||
const NAME: &'static str = "GstAwsKvsWebRTCSink";
|
||||
type Type = super::AwsKvsWebRTCSink;
|
||||
type ParentType = super::WebRTCSink;
|
||||
type ParentType = super::BaseWebRTCSink;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -16,11 +16,15 @@ mod homegrown_cc;
|
|||
mod imp;
|
||||
|
||||
glib::wrapper! {
|
||||
pub struct WebRTCSink(ObjectSubclass<imp::WebRTCSink>) @extends gst::Bin, gst::Element, gst::Object, @implements gst::ChildProxy, gst_video::Navigation;
|
||||
pub struct BaseWebRTCSink(ObjectSubclass<imp::BaseWebRTCSink>) @extends gst::Bin, gst::Element, gst::Object, @implements gst::ChildProxy, gst_video::Navigation;
|
||||
}
|
||||
|
||||
glib::wrapper! {
|
||||
pub struct AwsKvsWebRTCSink(ObjectSubclass<imp::AwsKvsWebRTCSink>) @extends WebRTCSink, gst::Bin, gst::Element, gst::Object, @implements gst::ChildProxy, gst_video::Navigation;
|
||||
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;
|
||||
}
|
||||
|
||||
#[derive(thiserror::Error, Debug)]
|
||||
|
|
@ -43,15 +47,15 @@ pub enum WebRTCSinkError {
|
|||
},
|
||||
}
|
||||
|
||||
impl Default for WebRTCSink {
|
||||
impl Default for BaseWebRTCSink {
|
||||
fn default() -> Self {
|
||||
glib::Object::new()
|
||||
}
|
||||
}
|
||||
|
||||
impl WebRTCSink {
|
||||
impl BaseWebRTCSink {
|
||||
pub fn with_signaller(signaller: Signallable) -> Self {
|
||||
let ret: WebRTCSink = glib::Object::new();
|
||||
let ret: BaseWebRTCSink = glib::Object::new();
|
||||
|
||||
let ws = ret.imp();
|
||||
ws.set_signaller(signaller).unwrap();
|
||||
|
|
@ -83,6 +87,7 @@ enum WebRTCSinkMitigationMode {
|
|||
}
|
||||
|
||||
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),
|
||||
|
|
|
|||
Loading…
Reference in a new issue