mirror of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2024-12-25 19:40:29 +00:00
Add a webrtcsrc element
Updating the docker image to include: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3236 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
This commit is contained in:
Thibault Saunier
parent
0ae637f531
commit
ce3bb2f1d4
12 changed files with 2611 additions and 5 deletions
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@ -99,9 +99,17 @@ foreach plugin_name: list_plugin_res.stdout().split(':')
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gst_index: 'plugins/index.md',
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include_paths: join_paths(meson.current_source_dir(), '..'),
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gst_smart_index: true,
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gst_c_source_filters: [
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'../target/*/*.rs',
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'../target/*/*/*.rs',
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'../target/*/*/*/*.rs',
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'../target/*/*/*/*/*.rs',
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'../target/*/*/*/*/*/*.rs',
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],
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gst_c_sources: [
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'../*/*/*/*.rs',
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'../*/*/*/*/*.rs',
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'../*/*/*/*/*/*.rs',
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],
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dependencies: [gst_dep],
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gst_order_generated_subpages: true,
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@ -6150,11 +6150,240 @@
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"when": "last"
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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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"GInterface"
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],
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"kind": "interface",
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"signals": {
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"error": {
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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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"return-type": "void",
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"when": "last"
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},
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"handle-ice": {
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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": "guint"
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},
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{
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"name": "arg2",
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"type": "gchararray"
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},
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{
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"name": "arg3",
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"type": "gchararray"
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}
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],
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"return-type": "void",
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"when": "last"
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},
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"producer-added": {
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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": "GstStructure"
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}
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],
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"return-type": "void",
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"when": "last"
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},
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"producer-removed": {
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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": "GstStructure"
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}
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],
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"return-type": "void",
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"when": "last"
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},
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"request-meta": {
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"args": [],
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"return-type": "GstStructure",
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"when": "last"
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},
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"session-description": {
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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": "GstWebRTCSessionDescription"
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}
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],
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"return-type": "void",
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"when": "last"
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},
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"session-ended": {
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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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"return-type": "void",
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"when": "last"
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},
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"session-requested": {
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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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"return-type": "void",
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"when": "last"
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},
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"session-started": {
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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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"return-type": "void",
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"when": "last"
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},
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"start": {
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"action": true,
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"args": [],
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"return-type": "void",
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"when": "last"
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},
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"stop": {
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"action": true,
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"args": [],
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"return-type": "void",
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"when": "last"
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}
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}
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},
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"GstWebRTCSinkCongestionControl": {
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"kind": "enum",
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"values": [
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@ -6174,6 +6403,18 @@
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"value": "2"
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}
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]
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},
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"GstWebRTCSrcPad": {
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"hierarchy": [
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"GstWebRTCSrcPad",
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"GstGhostPad",
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"GstProxyPad",
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"GstPad",
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"GstObject",
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"GInitiallyUnowned",
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"GObject"
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],
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"kind": "object"
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}
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},
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"package": "gst-plugin-webrtc",
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@ -16,6 +16,8 @@ gst-webrtc = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", pack
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gst-sdp = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-sdp", features = ["v1_20"] }
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gst-rtp = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-rtp", features = ["v1_20"] }
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gst-utils = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-utils" }
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gst-base = { git="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs", package = "gstreamer-base" }
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once_cell = "1.0"
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anyhow = "1"
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thiserror = "1"
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@ -29,6 +31,7 @@ serde_json = "1"
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fastrand = "1.0"
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gst_plugin_webrtc_protocol = { path="protocol", package = "gst-plugin-webrtc-signalling-protocol" }
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human_bytes = "0.4"
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url = "2"
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[dev-dependencies]
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tracing = { version = "0.1", features = ["log"] }
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@ -277,10 +277,9 @@ impl Handler {
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},
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)?;
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self.peers.get(consumer_id).map_or_else(
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|| Err(anyhow!("No consumer with ID: '{consumer_id}'")),
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Ok,
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)?;
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self.peers
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.get(consumer_id)
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.map_or_else(|| Err(anyhow!("No consumer with ID: '{consumer_id}'")), Ok)?;
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let session_id = uuid::Uuid::new_v4().to_string();
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self.sessions.insert(
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@ -7,14 +7,17 @@
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* Since: plugins-rs-0.9
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*/
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use gst::glib;
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use tokio::runtime;
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use once_cell::sync::Lazy;
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use tokio::runtime;
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mod signaller;
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pub mod utils;
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pub mod webrtcsink;
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pub mod webrtcsrc;
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fn plugin_init(plugin: &gst::Plugin) -> Result<(), glib::BoolError> {
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webrtcsink::register(plugin)?;
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webrtcsrc::register(Some(plugin))?;
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Ok(())
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}
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87
net/webrtc/src/utils.rs
Normal file
87
net/webrtc/src/utils.rs
Normal file
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@ -0,0 +1,87 @@
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use std::collections::HashMap;
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use gst::{glib, prelude::*};
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pub fn gvalue_to_json(val: &gst::glib::Value) -> Option<serde_json::Value> {
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match val.type_() {
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glib::Type::STRING => Some(val.get::<String>().unwrap().into()),
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glib::Type::BOOL => Some(val.get::<bool>().unwrap().into()),
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glib::Type::I32 => Some(val.get::<i32>().unwrap().into()),
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glib::Type::U32 => Some(val.get::<u32>().unwrap().into()),
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glib::Type::I_LONG | glib::Type::I64 => Some(val.get::<i64>().unwrap().into()),
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glib::Type::U_LONG | glib::Type::U64 => Some(val.get::<u64>().unwrap().into()),
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glib::Type::F32 => Some(val.get::<f32>().unwrap().into()),
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glib::Type::F64 => Some(val.get::<f64>().unwrap().into()),
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_ => {
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if let Ok(s) = val.get::<gst::Structure>() {
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serde_json::to_value(
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s.iter()
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.filter_map(|(name, value)| {
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gvalue_to_json(value).map(|value| (name.to_string(), value))
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})
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.collect::<HashMap<String, serde_json::Value>>(),
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)
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.ok()
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} else if let Ok(a) = val.get::<gst::Array>() {
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serde_json::to_value(
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a.iter()
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.filter_map(|value| gvalue_to_json(value))
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.collect::<Vec<serde_json::Value>>(),
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)
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.ok()
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} else if let Some((_klass, values)) = gst::glib::FlagsValue::from_value(val) {
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Some(
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values
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.iter()
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.map(|value| value.nick())
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.collect::<Vec<&str>>()
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.join("+")
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.into(),
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)
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} else if let Ok(value) = val.serialize() {
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Some(value.as_str().into())
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} else {
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None
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}
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}
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}
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}
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fn json_to_gststructure(val: &serde_json::Value) -> Option<glib::SendValue> {
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match val {
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serde_json::Value::Bool(v) => Some(v.to_send_value()),
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serde_json::Value::Number(n) => {
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if n.is_u64() {
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Some(n.as_u64().unwrap().to_send_value())
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} else if n.is_i64() {
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Some(n.as_i64().unwrap().to_send_value())
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} else if n.is_f64() {
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Some(n.as_f64().unwrap().to_send_value())
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} else {
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todo!("Unhandled case {n:?}");
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}
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}
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serde_json::Value::String(v) => Some(v.to_send_value()),
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serde_json::Value::Array(v) => {
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let array = v
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.iter()
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.filter_map(json_to_gststructure)
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.collect::<Vec<glib::SendValue>>();
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Some(gst::Array::from_values(array).to_send_value())
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}
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serde_json::Value::Object(v) => Some(serialize_json_object(v).to_send_value()),
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_ => None,
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}
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}
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pub fn serialize_json_object(val: &serde_json::Map<String, serde_json::Value>) -> gst::Structure {
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let mut res = gst::Structure::new_empty("v");
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val.iter().for_each(|(k, v)| {
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if let Some(gvalue) = json_to_gststructure(v) {
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res.set_value(k, gvalue);
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}
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});
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res
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}
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1099
net/webrtc/src/webrtcsrc/imp.rs
Normal file
1099
net/webrtc/src/webrtcsrc/imp.rs
Normal file
File diff suppressed because it is too large
Load diff
65
net/webrtc/src/webrtcsrc/mod.rs
Normal file
65
net/webrtc/src/webrtcsrc/mod.rs
Normal file
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@ -0,0 +1,65 @@
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// SPDX-License-Identifier: MPL-2.0
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use crate::webrtcsrc::signaller::WebRTCSignallerRole;
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use gst::prelude::*;
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use gst::{glib, prelude::StaticType};
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/**
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* element-webrtcsrc:
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*
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* `webrtcsrc` is the source counterpart of the #webrtcsink element and can be
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* used to receive streams from it, it can also be used to easily playback WebRTC
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* streams coming from a web browser.
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*
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* To try the element, you should run #webrtcsink as described in its documentation,
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* finding its `peer-id` (in the signalling server logs for example) and then
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* run:
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*
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* ``` bash
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* gst-launch-1.0 webrtcsrc signaller::producer-peer-id=<webrtcsink-peer-id> ! videoconvert ! autovideosink
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* ```
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*
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* or directly using `playbin`:
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*
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* ``` bash
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* gst-launch-1.0 playbin3 uri="gstwebrtc://localhost:8443?peer-id=<webrtcsink-peer-id>"
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* ```
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*
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* ## Decoding
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*
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* To be able to precisely negotiate the WebRTC SDP, `webrtcsrc` is able to decode streams.
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* During SDP negotiation we expose our pads based on the peer offer and right after query caps
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* to see what downstream supports.
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* In practice in `uridecodebinX` or `playbinX`, decoding will happen
|
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* in `decodebinX` but for the case where a `videoconvert` is placed after a `video_XX` pad,
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* decoding will happen inside `webrtcsrc`.
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*
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* Since: 0.10
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*/
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mod imp;
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mod pad;
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pub mod signaller;
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pub use signaller::{SignallableImpl, SignallableImplExt};
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||||
use self::signaller::Signallable;
|
||||
|
||||
glib::wrapper! {
|
||||
pub struct WebRTCSrc(ObjectSubclass<imp::WebRTCSrc>) @extends gst::Bin, gst::Element, gst::Object, @implements gst::URIHandler, gst::ChildProxy;
|
||||
}
|
||||
|
||||
glib::wrapper! {
|
||||
pub struct WebRTCSrcPad(ObjectSubclass<pad::WebRTCSrcPad>) @extends gst::GhostPad, gst::ProxyPad, gst::Pad, gst::Object;
|
||||
}
|
||||
|
||||
pub fn register(plugin: Option<&gst::Plugin>) -> Result<(), glib::BoolError> {
|
||||
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());
|
||||
gst::Element::register(
|
||||
plugin,
|
||||
"webrtcsrc",
|
||||
gst::Rank::Primary,
|
||||
WebRTCSrc::static_type(),
|
||||
)
|
||||
}
|
||||
45
net/webrtc/src/webrtcsrc/pad.rs
Normal file
45
net/webrtc/src/webrtcsrc/pad.rs
Normal file
|
|
@ -0,0 +1,45 @@
|
|||
// SPDX-License-Identifier: MPL-2.0
|
||||
|
||||
use gst::glib;
|
||||
use gst::subclass::prelude::*;
|
||||
use std::sync::atomic::AtomicBool;
|
||||
use std::sync::atomic::Ordering;
|
||||
use std::sync::Mutex;
|
||||
|
||||
#[derive(Default)]
|
||||
pub struct WebRTCSrcPad {
|
||||
needs_raw: AtomicBool,
|
||||
stream_id: Mutex<Option<String>>,
|
||||
}
|
||||
|
||||
impl WebRTCSrcPad {
|
||||
pub fn set_needs_decoding(&self, raw_wanted: bool) {
|
||||
self.needs_raw.store(raw_wanted, Ordering::SeqCst);
|
||||
}
|
||||
|
||||
pub fn needs_decoding(&self) -> bool {
|
||||
self.needs_raw.load(Ordering::SeqCst)
|
||||
}
|
||||
|
||||
pub fn set_stream_id(&self, stream_id: &str) {
|
||||
*self.stream_id.lock().unwrap() = Some(stream_id.to_string());
|
||||
}
|
||||
|
||||
pub fn stream_id(&self) -> String {
|
||||
let stream_id = self.stream_id.lock().unwrap();
|
||||
stream_id.as_ref().unwrap().clone()
|
||||
}
|
||||
}
|
||||
|
||||
#[glib::object_subclass]
|
||||
impl ObjectSubclass for WebRTCSrcPad {
|
||||
const NAME: &'static str = "GstWebRTCSrcPad";
|
||||
type Type = super::WebRTCSrcPad;
|
||||
type ParentType = gst::GhostPad;
|
||||
}
|
||||
|
||||
impl ObjectImpl for WebRTCSrcPad {}
|
||||
impl GstObjectImpl for WebRTCSrcPad {}
|
||||
impl PadImpl for WebRTCSrcPad {}
|
||||
impl ProxyPadImpl for WebRTCSrcPad {}
|
||||
impl GhostPadImpl for WebRTCSrcPad {}
|
||||
426
net/webrtc/src/webrtcsrc/signaller/iface.rs
Normal file
426
net/webrtc/src/webrtcsrc/signaller/iface.rs
Normal file
|
|
@ -0,0 +1,426 @@
|
|||
use gst::glib;
|
||||
use gst::glib::subclass::*;
|
||||
use gst::prelude::*;
|
||||
use gst::subclass::prelude::*;
|
||||
use once_cell::sync::Lazy;
|
||||
|
||||
#[derive(Copy, Clone)]
|
||||
pub struct Signallable {
|
||||
_parent: glib::gobject_ffi::GTypeInterface,
|
||||
pub start: fn(&super::Signallable),
|
||||
pub stop: fn(&super::Signallable),
|
||||
pub send_sdp: fn(&super::Signallable, &str, &gst_webrtc::WebRTCSessionDescription),
|
||||
pub add_ice: fn(&super::Signallable, &str, &str, Option<u32>, Option<String>),
|
||||
pub end_session: fn(&super::Signallable, &str),
|
||||
}
|
||||
|
||||
impl Signallable {
|
||||
fn request_meta(_iface: &super::Signallable) -> Option<gst::Structure> {
|
||||
None
|
||||
}
|
||||
fn start(_iface: &super::Signallable) {}
|
||||
fn stop(_iface: &super::Signallable) {}
|
||||
fn send_sdp(
|
||||
_iface: &super::Signallable,
|
||||
_session_id: &str,
|
||||
_sdp: &gst_webrtc::WebRTCSessionDescription,
|
||||
) {
|
||||
}
|
||||
fn add_ice(
|
||||
_iface: &super::Signallable,
|
||||
_session_id: &str,
|
||||
_candidate: &str,
|
||||
_sdp_m_line_index: Option<u32>,
|
||||
_sdp_mid: Option<String>,
|
||||
) {
|
||||
}
|
||||
fn end_session(_iface: &super::Signallable, _session_id: &str) {}
|
||||
}
|
||||
|
||||
#[glib::object_interface]
|
||||
unsafe impl prelude::ObjectInterface for Signallable {
|
||||
const NAME: &'static ::std::primitive::str = "GstRSWebRTCSignallableIface";
|
||||
type Prerequisites = (glib::Object,);
|
||||
|
||||
fn interface_init(&mut self) {
|
||||
self.start = Signallable::start;
|
||||
self.stop = Signallable::stop;
|
||||
self.send_sdp = Signallable::send_sdp;
|
||||
self.add_ice = Signallable::add_ice;
|
||||
self.end_session = Signallable::end_session;
|
||||
}
|
||||
|
||||
fn signals() -> &'static [Signal] {
|
||||
static SIGNALS: Lazy<Vec<Signal>> = Lazy::new(|| {
|
||||
vec![
|
||||
/**
|
||||
* GstRSWebRTCSignallableIface::session-ended:
|
||||
* @self: The object implementing #GstRSWebRTCSignallableIface
|
||||
* @session-id: The ID of the session that ended
|
||||
*
|
||||
* Some WebRTC Session was closed.
|
||||
*/
|
||||
Signal::builder("session-ended")
|
||||
.param_types([str::static_type()])
|
||||
.build(),
|
||||
/**
|
||||
* GstRSWebRTCSignallableIface::producer-added:
|
||||
* @self: The object implementing #GstRSWebRTCSignallableIface
|
||||
* @producer_id: The ID of the producer that was added
|
||||
* @meta: The metadata structure of the producer
|
||||
*
|
||||
* Some new producing peer is ready to produce a WebRTC stream.
|
||||
*/
|
||||
Signal::builder("producer-added")
|
||||
.param_types([str::static_type(), <Option<gst::Structure>>::static_type()])
|
||||
.build(),
|
||||
/**
|
||||
* GstRSWebRTCSignallableIface::producer-removed:
|
||||
* @self: The object implementing #GstRSWebRTCSignallableIface
|
||||
* @producer_id: The ID of the producer that was added
|
||||
* @meta: The metadata structure of the producer
|
||||
*
|
||||
* Some new producing peer is stopped producing streams.
|
||||
*/
|
||||
Signal::builder("producer-removed")
|
||||
.param_types([str::static_type(), <Option<gst::Structure>>::static_type()])
|
||||
.build(),
|
||||
/**
|
||||
* GstRSWebRTCSignallableIface::session-started:
|
||||
* @self: The object implementing #GstRSWebRTCSignallableIface
|
||||
*
|
||||
* A new session started,
|
||||
*/
|
||||
Signal::builder("session-started")
|
||||
.param_types([str::static_type(), str::static_type()])
|
||||
.build(),
|
||||
/**
|
||||
* GstRSWebRTCSignallableIface::session-requested:
|
||||
* @self: The object implementing #GstRSWebRTCSignallableIface
|
||||
* @session_id: The ID of the producer that was added
|
||||
* @peer_id: The ID of the consumer peer who wants to initiate a
|
||||
* session
|
||||
*/
|
||||
Signal::builder("session-requested")
|
||||
.param_types([str::static_type(), str::static_type()])
|
||||
.build(),
|
||||
/**
|
||||
* GstRSWebRTCSignallableIface::error:
|
||||
* @self: The object implementing #GstRSWebRTCSignallableIface
|
||||
* @error: The error message as a string
|
||||
*/
|
||||
Signal::builder("error")
|
||||
.param_types([str::static_type()])
|
||||
.build(),
|
||||
/**
|
||||
* GstRSWebRTCSignallableIface::request-meta:
|
||||
* @self: The object implementing #GstRSWebRTCSignallableIface
|
||||
*
|
||||
* The signaller requests a meta about the peer using it
|
||||
*
|
||||
* Return: The metadata about the peer represented by the signaller
|
||||
*/
|
||||
Signal::builder("request-meta")
|
||||
.return_type::<Option<gst::Structure>>()
|
||||
.class_handler(|_token, args| {
|
||||
let arg0 = args[0usize]
|
||||
.get::<&super::Signallable>()
|
||||
.unwrap_or_else(|e| {
|
||||
panic!("Wrong type for argument {}: {:?}", 0usize, e)
|
||||
});
|
||||
Some(Signallable::request_meta(arg0).to_value())
|
||||
})
|
||||
.build(),
|
||||
/**
|
||||
* GstRSWebRTCSignallableIface::handle-ice:
|
||||
* @self: The object implementing #GstRSWebRTCSignallableIface
|
||||
* @session_id: Id of the session the ice information is about
|
||||
* @sdp_m_line_index: The mlineindex of the ice candidate
|
||||
* @sdp_mid: Media ID of the ice candidate
|
||||
* @candiate: Information about the candidate
|
||||
*/
|
||||
Signal::builder("handle-ice")
|
||||
.param_types([
|
||||
str::static_type(),
|
||||
u32::static_type(),
|
||||
<Option<String>>::static_type(),
|
||||
str::static_type(),
|
||||
])
|
||||
.build(),
|
||||
/**
|
||||
* GstRSWebRTCSignallableIface::session-description:
|
||||
* @self: The object implementing #GstRSWebRTCSignallableIface
|
||||
* @session_id: Id of the session being described
|
||||
* @description: The WebRTC session description
|
||||
*/
|
||||
Signal::builder("session-description")
|
||||
.param_types([
|
||||
str::static_type(),
|
||||
gst_webrtc::WebRTCSessionDescription::static_type(),
|
||||
])
|
||||
.build(),
|
||||
/**
|
||||
* GstRSWebRTCSignallableIface::start:
|
||||
* @self: The object implementing #GstRSWebRTCSignallableIface
|
||||
*
|
||||
* Starts the signaller, connecting it to the signalling server.
|
||||
*/
|
||||
Signal::builder("start")
|
||||
.flags(glib::SignalFlags::ACTION)
|
||||
.class_handler(|_token, args| {
|
||||
let arg0 = args[0usize]
|
||||
.get::<&super::Signallable>()
|
||||
.unwrap_or_else(|e| {
|
||||
panic!("Wrong type for argument {}: {:?}", 0usize, e)
|
||||
});
|
||||
Signallable::start(arg0);
|
||||
|
||||
None
|
||||
})
|
||||
.build(),
|
||||
/**
|
||||
* GstRSWebRTCSignallableIface::stop:
|
||||
* @self: The object implementing #GstRSWebRTCSignallableIface
|
||||
*
|
||||
* Stops the signaller, disconnecting it to the signalling server.
|
||||
*/
|
||||
Signal::builder("stop")
|
||||
.flags(glib::SignalFlags::ACTION)
|
||||
.class_handler(|_tokens, args| {
|
||||
let arg0 = args[0usize]
|
||||
.get::<&super::Signallable>()
|
||||
.unwrap_or_else(|e| {
|
||||
panic!("Wrong type for argument {}: {:?}", 0usize, e)
|
||||
});
|
||||
Signallable::stop(arg0);
|
||||
|
||||
None
|
||||
})
|
||||
.build(),
|
||||
]
|
||||
});
|
||||
SIGNALS.as_ref()
|
||||
}
|
||||
}
|
||||
|
||||
unsafe impl<Obj: SignallableImpl> types::IsImplementable<Obj> for super::Signallable
|
||||
where
|
||||
<Obj as types::ObjectSubclass>::Type: glib::IsA<glib::Object>,
|
||||
{
|
||||
fn interface_init(iface: &mut glib::Interface<Self>) {
|
||||
let iface = ::std::convert::AsMut::as_mut(iface);
|
||||
|
||||
fn vstart_trampoline<Obj: types::ObjectSubclass + SignallableImpl>(
|
||||
obj: &super::Signallable,
|
||||
) {
|
||||
let this = obj
|
||||
.dynamic_cast_ref::<<Obj as types::ObjectSubclass>::Type>()
|
||||
.unwrap()
|
||||
.imp();
|
||||
SignallableImpl::start(this)
|
||||
}
|
||||
iface.start = vstart_trampoline::<Obj>;
|
||||
|
||||
fn vstop_trampoline<Obj: types::ObjectSubclass + SignallableImpl>(
|
||||
this: &super::Signallable,
|
||||
) {
|
||||
let this = this
|
||||
.dynamic_cast_ref::<<Obj as types::ObjectSubclass>::Type>()
|
||||
.unwrap();
|
||||
SignallableImpl::stop(this.imp())
|
||||
}
|
||||
iface.stop = vstop_trampoline::<Obj>;
|
||||
|
||||
fn send_sdp_trampoline<Obj: types::ObjectSubclass + SignallableImpl>(
|
||||
this: &super::Signallable,
|
||||
session_id: &str,
|
||||
sdp: &gst_webrtc::WebRTCSessionDescription,
|
||||
) {
|
||||
let this = this
|
||||
.dynamic_cast_ref::<<Obj as types::ObjectSubclass>::Type>()
|
||||
.unwrap();
|
||||
SignallableImpl::send_sdp(this.imp(), session_id, sdp)
|
||||
}
|
||||
iface.send_sdp = send_sdp_trampoline::<Obj>;
|
||||
|
||||
fn add_ice_trampoline<Obj: types::ObjectSubclass + SignallableImpl>(
|
||||
this: &super::Signallable,
|
||||
session_id: &str,
|
||||
candidate: &str,
|
||||
sdp_m_line_index: Option<u32>,
|
||||
sdp_mid: Option<String>,
|
||||
) {
|
||||
let this = this
|
||||
.dynamic_cast_ref::<<Obj as types::ObjectSubclass>::Type>()
|
||||
.unwrap();
|
||||
SignallableImpl::add_ice(this.imp(), session_id, candidate, sdp_m_line_index, sdp_mid)
|
||||
}
|
||||
iface.add_ice = add_ice_trampoline::<Obj>;
|
||||
|
||||
fn end_session_trampoline<Obj: types::ObjectSubclass + SignallableImpl>(
|
||||
this: &super::Signallable,
|
||||
session_id: &str,
|
||||
) {
|
||||
let this = this
|
||||
.dynamic_cast_ref::<<Obj as types::ObjectSubclass>::Type>()
|
||||
.unwrap();
|
||||
SignallableImpl::end_session(this.imp(), session_id)
|
||||
}
|
||||
iface.end_session = end_session_trampoline::<Obj>;
|
||||
}
|
||||
}
|
||||
|
||||
pub trait SignallableImpl: object::ObjectImpl + 'static {
|
||||
fn start(&self) {
|
||||
SignallableImplExt::parent_vstart(self)
|
||||
}
|
||||
fn stop(&self) {
|
||||
SignallableImplExt::parent_vstop(self)
|
||||
}
|
||||
fn send_sdp(&self, session_id: &str, sdp: &gst_webrtc::WebRTCSessionDescription) {
|
||||
SignallableImplExt::parent_send_sdp(self, session_id, sdp)
|
||||
}
|
||||
fn add_ice(
|
||||
&self,
|
||||
session_id: &str,
|
||||
candidate: &str,
|
||||
sdp_m_line_index: Option<u32>,
|
||||
sdp_mid: Option<String>,
|
||||
) {
|
||||
SignallableImplExt::parent_add_ice(self, session_id, candidate, sdp_m_line_index, sdp_mid)
|
||||
}
|
||||
fn end_session(&self, session_id: &str) {
|
||||
SignallableImplExt::parent_end_session(self, session_id)
|
||||
}
|
||||
}
|
||||
|
||||
pub trait SignallableImplExt: types::ObjectSubclass {
|
||||
fn parent_vstart(&self);
|
||||
fn parent_vstop(&self);
|
||||
fn parent_send_sdp(&self, session_id: &str, sdp: &gst_webrtc::WebRTCSessionDescription);
|
||||
fn parent_add_ice(
|
||||
&self,
|
||||
session_id: &str,
|
||||
candidate: &str,
|
||||
sdp_m_line_index: Option<u32>,
|
||||
sdp_mid: Option<String>,
|
||||
);
|
||||
fn parent_end_session(&self, session_id: &str);
|
||||
}
|
||||
|
||||
type ClassType = *mut <super::Signallable as glib::object::ObjectType>::GlibClassType;
|
||||
impl<Obj: SignallableImpl> SignallableImplExt for Obj {
|
||||
fn parent_vstart(&self) {
|
||||
let obj = self.obj();
|
||||
let obj = unsafe { obj.unsafe_cast_ref::<super::Signallable>() };
|
||||
let vtable = unsafe {
|
||||
&*(Self::type_data()
|
||||
.as_ref()
|
||||
.parent_interface::<super::Signallable>() as ClassType)
|
||||
};
|
||||
(vtable.start)(obj)
|
||||
}
|
||||
fn parent_vstop(&self) {
|
||||
let obj = self.obj();
|
||||
let obj = unsafe { obj.unsafe_cast_ref::<super::Signallable>() };
|
||||
let vtable = unsafe {
|
||||
&*(Self::type_data()
|
||||
.as_ref()
|
||||
.parent_interface::<super::Signallable>() as ClassType)
|
||||
};
|
||||
(vtable.stop)(obj)
|
||||
}
|
||||
fn parent_send_sdp(&self, session_id: &str, sdp: &gst_webrtc::WebRTCSessionDescription) {
|
||||
let obj = self.obj();
|
||||
let obj = unsafe { obj.unsafe_cast_ref::<super::Signallable>() };
|
||||
let vtable = unsafe {
|
||||
&*(Self::type_data()
|
||||
.as_ref()
|
||||
.parent_interface::<super::Signallable>() as ClassType)
|
||||
};
|
||||
(vtable.send_sdp)(obj, session_id, sdp)
|
||||
}
|
||||
fn parent_add_ice(
|
||||
&self,
|
||||
session_id: &str,
|
||||
candidate: &str,
|
||||
sdp_m_line_index: Option<u32>,
|
||||
sdp_mid: Option<String>,
|
||||
) {
|
||||
let obj = self.obj();
|
||||
let obj = unsafe { obj.unsafe_cast_ref::<super::Signallable>() };
|
||||
let vtable = unsafe {
|
||||
&*(Self::type_data()
|
||||
.as_ref()
|
||||
.parent_interface::<super::Signallable>() as ClassType)
|
||||
};
|
||||
(vtable.add_ice)(obj, session_id, candidate, sdp_m_line_index, sdp_mid)
|
||||
}
|
||||
fn parent_end_session(&self, session_id: &str) {
|
||||
let obj = self.obj();
|
||||
let obj = unsafe { obj.unsafe_cast_ref::<super::Signallable>() };
|
||||
let vtable = unsafe {
|
||||
&*(Self::type_data()
|
||||
.as_ref()
|
||||
.parent_interface::<super::Signallable>() as ClassType)
|
||||
};
|
||||
(vtable.end_session)(obj, session_id)
|
||||
}
|
||||
}
|
||||
|
||||
pub trait SignallableExt: 'static {
|
||||
fn start(&self);
|
||||
fn stop(&self);
|
||||
fn send_sdp(&self, session_id: &str, sdp: &gst_webrtc::WebRTCSessionDescription);
|
||||
fn add_ice(
|
||||
&self,
|
||||
session_id: &str,
|
||||
candidate: &str,
|
||||
sdp_m_line_index: Option<u32>,
|
||||
sdp_mid: Option<String>,
|
||||
);
|
||||
fn end_session(&self, session_id: &str);
|
||||
}
|
||||
|
||||
impl<Obj: glib::IsA<super::Signallable>> SignallableExt for Obj {
|
||||
fn start(&self) {
|
||||
let obj = self.upcast_ref::<super::Signallable>();
|
||||
let vtable = obj.interface::<super::Signallable>().unwrap();
|
||||
let vtable = vtable.as_ref();
|
||||
(vtable.start)(obj)
|
||||
}
|
||||
|
||||
fn stop(&self) {
|
||||
let obj = self.upcast_ref::<super::Signallable>();
|
||||
let vtable = obj.interface::<super::Signallable>().unwrap();
|
||||
let vtable = vtable.as_ref();
|
||||
(vtable.stop)(obj)
|
||||
}
|
||||
|
||||
fn send_sdp(&self, session_id: &str, sdp: &gst_webrtc::WebRTCSessionDescription) {
|
||||
let obj = self.upcast_ref::<super::Signallable>();
|
||||
let vtable = obj.interface::<super::Signallable>().unwrap();
|
||||
let vtable = vtable.as_ref();
|
||||
(vtable.send_sdp)(obj, session_id, sdp)
|
||||
}
|
||||
|
||||
fn add_ice(
|
||||
&self,
|
||||
session_id: &str,
|
||||
candidate: &str,
|
||||
sdp_m_line_index: Option<u32>,
|
||||
sdp_mid: Option<String>,
|
||||
) {
|
||||
let obj = self.upcast_ref::<super::Signallable>();
|
||||
let vtable = obj.interface::<super::Signallable>().unwrap();
|
||||
let vtable = vtable.as_ref();
|
||||
(vtable.add_ice)(obj, session_id, candidate, sdp_m_line_index, sdp_mid)
|
||||
}
|
||||
|
||||
fn end_session(&self, session_id: &str) {
|
||||
let obj = self.upcast_ref::<super::Signallable>();
|
||||
let vtable = obj.interface::<super::Signallable>().unwrap();
|
||||
let vtable = vtable.as_ref();
|
||||
(vtable.end_session)(obj, session_id)
|
||||
}
|
||||
}
|
||||
584
net/webrtc/src/webrtcsrc/signaller/imp.rs
Normal file
584
net/webrtc/src/webrtcsrc/signaller/imp.rs
Normal file
|
|
@ -0,0 +1,584 @@
|
|||
use crate::utils::{gvalue_to_json, serialize_json_object};
|
||||
use crate::webrtcsrc::signaller::{prelude::*, Signallable};
|
||||
use crate::RUNTIME;
|
||||
use anyhow::{anyhow, Error};
|
||||
use async_tungstenite::tungstenite::Message as WsMessage;
|
||||
use futures::channel::mpsc;
|
||||
use futures::prelude::*;
|
||||
use gst::glib;
|
||||
use gst::glib::prelude::*;
|
||||
use gst::subclass::prelude::*;
|
||||
use gst_plugin_webrtc_protocol as p;
|
||||
use once_cell::sync::Lazy;
|
||||
use std::collections::HashSet;
|
||||
use std::ops::ControlFlow;
|
||||
use std::str::FromStr;
|
||||
use std::sync::Mutex;
|
||||
use std::time::Duration;
|
||||
use tokio::{task, time::timeout};
|
||||
use url::Url;
|
||||
|
||||
use super::CAT;
|
||||
|
||||
#[derive(Debug, Eq, PartialEq, Clone, Copy, glib::Enum, Default)]
|
||||
#[repr(u32)]
|
||||
#[enum_type(name = "GstRSWebRTCSignallerRole")]
|
||||
pub enum WebRTCSignallerRole {
|
||||
#[default]
|
||||
Consumer,
|
||||
Producer,
|
||||
Listener,
|
||||
}
|
||||
|
||||
pub struct Settings {
|
||||
uri: Url,
|
||||
producer_peer_id: Option<String>,
|
||||
cafile: Option<String>,
|
||||
role: WebRTCSignallerRole,
|
||||
}
|
||||
|
||||
impl Default for Settings {
|
||||
fn default() -> Self {
|
||||
Self {
|
||||
uri: Url::from_str("ws://127.0.0.1:8443").unwrap(),
|
||||
producer_peer_id: None,
|
||||
cafile: Default::default(),
|
||||
role: Default::default(),
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
#[derive(Default)]
|
||||
pub struct Signaller {
|
||||
state: Mutex<State>,
|
||||
settings: Mutex<Settings>,
|
||||
}
|
||||
|
||||
#[derive(Default)]
|
||||
struct State {
|
||||
/// Sender for the websocket messages
|
||||
websocket_sender: Option<mpsc::Sender<p::IncomingMessage>>,
|
||||
send_task_handle: Option<task::JoinHandle<Result<(), Error>>>,
|
||||
receive_task_handle: Option<task::JoinHandle<()>>,
|
||||
producers: HashSet<String>,
|
||||
}
|
||||
|
||||
impl Signaller {
|
||||
fn uri(&self) -> Url {
|
||||
self.settings.lock().unwrap().uri.clone()
|
||||
}
|
||||
|
||||
fn set_uri(&self, uri: &str) -> Result<(), Error> {
|
||||
let mut settings = self.settings.lock().unwrap();
|
||||
let mut uri = Url::from_str(uri).map_err(|err| anyhow!("{err:?}"))?;
|
||||
|
||||
if let Some(peer_id) = uri
|
||||
.query_pairs()
|
||||
.find(|(k, _)| k == "peer-id")
|
||||
.map(|v| v.1.to_string())
|
||||
{
|
||||
if !matches!(settings.role, WebRTCSignallerRole::Consumer) {
|
||||
gst::warning!(
|
||||
CAT,
|
||||
"Setting peer-id doesn't make sense for {:?}",
|
||||
settings.role
|
||||
);
|
||||
} else {
|
||||
settings.producer_peer_id = Some(peer_id);
|
||||
}
|
||||
}
|
||||
|
||||
if let Some(peer_id) = &settings.producer_peer_id {
|
||||
uri.query_pairs_mut()
|
||||
.clear()
|
||||
.append_pair("peer-id", peer_id);
|
||||
}
|
||||
|
||||
settings.uri = uri;
|
||||
|
||||
Ok(())
|
||||
}
|
||||
|
||||
async fn connect(&self) -> Result<(), Error> {
|
||||
let obj = self.obj();
|
||||
|
||||
let role = self.settings.lock().unwrap().role;
|
||||
if let super::WebRTCSignallerRole::Consumer = role {
|
||||
self.producer_peer_id()
|
||||
.ok_or_else(|| anyhow!("No target producer peer id set"))?;
|
||||
}
|
||||
|
||||
let connector = if let Some(path) = obj.property::<Option<String>>("cafile") {
|
||||
let cert = tokio::fs::read_to_string(&path).await?;
|
||||
let cert = tokio_native_tls::native_tls::Certificate::from_pem(cert.as_bytes())?;
|
||||
let mut connector_builder = tokio_native_tls::native_tls::TlsConnector::builder();
|
||||
let connector = connector_builder.add_root_certificate(cert).build()?;
|
||||
Some(tokio_native_tls::TlsConnector::from(connector))
|
||||
} else {
|
||||
None
|
||||
};
|
||||
|
||||
let mut uri = self.uri();
|
||||
uri.set_query(None);
|
||||
let (ws, _) = timeout(
|
||||
// FIXME: Make the timeout configurable
|
||||
Duration::from_secs(20),
|
||||
async_tungstenite::tokio::connect_async_with_tls_connector(uri.to_string(), connector),
|
||||
)
|
||||
.await??;
|
||||
|
||||
gst::info!(CAT, imp: self, "connected");
|
||||
|
||||
// Channel for asynchronously sending out websocket message
|
||||
let (mut ws_sink, mut ws_stream) = ws.split();
|
||||
|
||||
// 1000 is completely arbitrary, we simply don't want infinite piling
|
||||
// up of messages as with unbounded
|
||||
let (websocket_sender, mut websocket_receiver) = mpsc::channel::<p::IncomingMessage>(1000);
|
||||
let send_task_handle =
|
||||
RUNTIME.spawn(glib::clone!(@weak-allow-none self as this => async move {
|
||||
while let Some(msg) = websocket_receiver.next().await {
|
||||
gst::log!(CAT, "Sending websocket message {:?}", msg);
|
||||
ws_sink
|
||||
.send(WsMessage::Text(serde_json::to_string(&msg).unwrap()))
|
||||
.await?;
|
||||
}
|
||||
|
||||
let msg = "Done sending";
|
||||
this.map_or_else(|| gst::info!(CAT, "{msg}"),
|
||||
|this| gst::info!(CAT, imp: this, "{msg}")
|
||||
);
|
||||
|
||||
ws_sink.send(WsMessage::Close(None)).await?;
|
||||
ws_sink.close().await?;
|
||||
|
||||
Ok::<(), Error>(())
|
||||
}));
|
||||
|
||||
let obj = self.obj();
|
||||
let meta =
|
||||
if let Some(meta) = obj.emit_by_name::<Option<gst::Structure>>("request-meta", &[]) {
|
||||
gvalue_to_json(&meta.to_value())
|
||||
} else {
|
||||
None
|
||||
};
|
||||
|
||||
let receive_task_handle =
|
||||
RUNTIME.spawn(glib::clone!(@weak-allow-none self as this => async move {
|
||||
while let Some(msg) = tokio_stream::StreamExt::next(&mut ws_stream).await {
|
||||
if let Some(ref this) = this {
|
||||
if let ControlFlow::Break(_) = this.handle_message(msg, &meta) {
|
||||
break;
|
||||
}
|
||||
} else {
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
let msg = "Stopped websocket receiving";
|
||||
this.map_or_else(|| gst::info!(CAT, "{msg}"),
|
||||
|this| gst::info!(CAT, imp: this, "{msg}")
|
||||
);
|
||||
}));
|
||||
|
||||
let mut state = self.state.lock().unwrap();
|
||||
state.websocket_sender = Some(websocket_sender);
|
||||
state.send_task_handle = Some(send_task_handle);
|
||||
state.receive_task_handle = Some(receive_task_handle);
|
||||
|
||||
Ok(())
|
||||
}
|
||||
|
||||
fn set_status(&self, meta: &Option<serde_json::Value>, peer_id: &str) {
|
||||
let role = self.settings.lock().unwrap().role;
|
||||
self.send(p::IncomingMessage::SetPeerStatus(match role {
|
||||
super::WebRTCSignallerRole::Consumer => p::PeerStatus {
|
||||
meta: meta.clone(),
|
||||
peer_id: Some(peer_id.to_string()),
|
||||
roles: vec![],
|
||||
},
|
||||
super::WebRTCSignallerRole::Producer => p::PeerStatus {
|
||||
meta: meta.clone(),
|
||||
peer_id: Some(peer_id.to_string()),
|
||||
roles: vec![p::PeerRole::Producer],
|
||||
},
|
||||
super::WebRTCSignallerRole::Listener => p::PeerStatus {
|
||||
meta: meta.clone(),
|
||||
peer_id: Some(peer_id.to_string()),
|
||||
roles: vec![p::PeerRole::Listener],
|
||||
},
|
||||
}));
|
||||
}
|
||||
|
||||
fn producer_peer_id(&self) -> Option<String> {
|
||||
let settings = self.settings.lock().unwrap();
|
||||
|
||||
settings.producer_peer_id.clone()
|
||||
}
|
||||
|
||||
fn send(&self, msg: p::IncomingMessage) {
|
||||
let state = self.state.lock().unwrap();
|
||||
if let Some(mut sender) = state.websocket_sender.clone() {
|
||||
RUNTIME.spawn(glib::clone!(@weak self as this => async move {
|
||||
if let Err(err) = sender.send(msg).await {
|
||||
this.obj().emit_by_name::<()>("error", &[&format!("Error: {}", err)]);
|
||||
}
|
||||
}));
|
||||
}
|
||||
}
|
||||
|
||||
pub fn start_session(&self) {
|
||||
let role = self.settings.lock().unwrap().role;
|
||||
if matches!(role, super::WebRTCSignallerRole::Consumer) {
|
||||
let target_producer = self.producer_peer_id().unwrap();
|
||||
|
||||
self.send(p::IncomingMessage::StartSession(p::StartSessionMessage {
|
||||
peer_id: target_producer.clone(),
|
||||
}));
|
||||
|
||||
gst::info!(
|
||||
CAT,
|
||||
imp: self,
|
||||
"Started session with producer peer id {target_producer}",
|
||||
);
|
||||
}
|
||||
}
|
||||
|
||||
fn handle_message(
|
||||
&self,
|
||||
msg: Result<WsMessage, async_tungstenite::tungstenite::Error>,
|
||||
meta: &Option<serde_json::Value>,
|
||||
) -> ControlFlow<()> {
|
||||
match msg {
|
||||
Ok(WsMessage::Text(msg)) => {
|
||||
gst::trace!(CAT, imp: self, "Received message {}", msg);
|
||||
|
||||
if let Ok(msg) = serde_json::from_str::<p::OutgoingMessage>(&msg) {
|
||||
match msg {
|
||||
p::OutgoingMessage::Welcome { peer_id } => {
|
||||
self.set_status(meta, &peer_id);
|
||||
self.start_session();
|
||||
}
|
||||
p::OutgoingMessage::PeerStatusChanged(p::PeerStatus {
|
||||
meta,
|
||||
roles,
|
||||
peer_id,
|
||||
}) => {
|
||||
let meta = meta.and_then(|m| match m {
|
||||
serde_json::Value::Object(v) => Some(serialize_json_object(&v)),
|
||||
_ => {
|
||||
gst::error!(CAT, imp: self, "Invalid json value: {m:?}");
|
||||
None
|
||||
}
|
||||
});
|
||||
|
||||
let peer_id =
|
||||
peer_id.expect("Status changed should always contain a peer ID");
|
||||
let mut state = self.state.lock().unwrap();
|
||||
if roles.iter().any(|r| matches!(r, p::PeerRole::Producer)) {
|
||||
if !state.producers.contains(&peer_id) {
|
||||
state.producers.insert(peer_id.clone());
|
||||
drop(state);
|
||||
|
||||
self.obj()
|
||||
.emit_by_name::<()>("producer-added", &[&peer_id, &meta]);
|
||||
}
|
||||
} else if state.producers.remove(&peer_id) {
|
||||
drop(state);
|
||||
|
||||
self.obj()
|
||||
.emit_by_name::<()>("producer-removed", &[&peer_id, &meta]);
|
||||
}
|
||||
}
|
||||
p::OutgoingMessage::SessionStarted {
|
||||
peer_id,
|
||||
session_id,
|
||||
} => {
|
||||
self.obj()
|
||||
.emit_by_name::<()>("session-started", &[&session_id, &peer_id]);
|
||||
}
|
||||
p::OutgoingMessage::StartSession {
|
||||
session_id,
|
||||
peer_id,
|
||||
} => {
|
||||
assert!(matches!(
|
||||
self.obj().property::<WebRTCSignallerRole>("role"),
|
||||
super::WebRTCSignallerRole::Producer
|
||||
));
|
||||
|
||||
self.obj()
|
||||
.emit_by_name::<()>("session-requested", &[&session_id, &peer_id]);
|
||||
}
|
||||
p::OutgoingMessage::EndSession(p::EndSessionMessage { session_id }) => {
|
||||
gst::info!(CAT, imp: self, "Session {session_id} ended");
|
||||
|
||||
self.obj()
|
||||
.emit_by_name::<()>("session-ended", &[&session_id]);
|
||||
}
|
||||
p::OutgoingMessage::Peer(p::PeerMessage {
|
||||
session_id,
|
||||
peer_message,
|
||||
}) => match peer_message {
|
||||
p::PeerMessageInner::Sdp(reply) => {
|
||||
let (sdp, desc_type) = match reply {
|
||||
p::SdpMessage::Answer { sdp } => {
|
||||
(sdp, gst_webrtc::WebRTCSDPType::Answer)
|
||||
}
|
||||
p::SdpMessage::Offer { sdp } => {
|
||||
(sdp, gst_webrtc::WebRTCSDPType::Offer)
|
||||
}
|
||||
};
|
||||
let sdp = match gst_sdp::SDPMessage::parse_buffer(sdp.as_bytes()) {
|
||||
Ok(sdp) => sdp,
|
||||
Err(err) => {
|
||||
self.obj().emit_by_name::<()>(
|
||||
"error",
|
||||
&[&format!("Error parsing SDP: {sdp} {err:?}")],
|
||||
);
|
||||
|
||||
return ControlFlow::Break(());
|
||||
}
|
||||
};
|
||||
|
||||
let desc =
|
||||
gst_webrtc::WebRTCSessionDescription::new(desc_type, sdp);
|
||||
self.obj().emit_by_name::<()>(
|
||||
"session-description",
|
||||
&[&session_id, &desc],
|
||||
);
|
||||
}
|
||||
p::PeerMessageInner::Ice {
|
||||
candidate,
|
||||
sdp_m_line_index,
|
||||
} => {
|
||||
let sdp_mid: Option<String> = None;
|
||||
self.obj().emit_by_name::<()>(
|
||||
"handle-ice",
|
||||
&[&session_id, &sdp_m_line_index, &sdp_mid, &candidate],
|
||||
);
|
||||
}
|
||||
},
|
||||
p::OutgoingMessage::Error { details } => {
|
||||
self.obj().emit_by_name::<()>(
|
||||
"error",
|
||||
&[&format!("Error message from server: {details}")],
|
||||
);
|
||||
}
|
||||
_ => {
|
||||
gst::warning!(CAT, imp: self, "Ignoring unsupported message {:?}", msg);
|
||||
}
|
||||
}
|
||||
} else {
|
||||
gst::error!(CAT, imp: self, "Unknown message from server: {}", msg);
|
||||
|
||||
self.obj().emit_by_name::<()>(
|
||||
"error",
|
||||
&[&format!("Unknown message from server: {}", msg)],
|
||||
);
|
||||
}
|
||||
}
|
||||
Ok(WsMessage::Close(reason)) => {
|
||||
gst::info!(CAT, imp: self, "websocket connection closed: {:?}", reason);
|
||||
return ControlFlow::Break(());
|
||||
}
|
||||
Ok(_) => (),
|
||||
Err(err) => {
|
||||
self.obj()
|
||||
.emit_by_name::<()>("error", &[&format!("Error receiving: {}", err)]);
|
||||
return ControlFlow::Break(());
|
||||
}
|
||||
}
|
||||
ControlFlow::Continue(())
|
||||
}
|
||||
}
|
||||
|
||||
#[glib::object_subclass]
|
||||
impl ObjectSubclass for Signaller {
|
||||
const NAME: &'static str = "GstWebRTCSignaller";
|
||||
type Type = super::Signaller;
|
||||
type ParentType = glib::Object;
|
||||
type Interfaces = (Signallable,);
|
||||
}
|
||||
|
||||
impl ObjectImpl for Signaller {
|
||||
fn properties() -> &'static [glib::ParamSpec] {
|
||||
static PROPS: Lazy<Vec<glib::ParamSpec>> = Lazy::new(|| {
|
||||
vec![
|
||||
glib::ParamSpecString::builder("uri")
|
||||
.flags(glib::ParamFlags::READWRITE)
|
||||
.build(),
|
||||
glib::ParamSpecString::builder("producer-peer-id")
|
||||
.flags(glib::ParamFlags::READWRITE)
|
||||
.build(),
|
||||
glib::ParamSpecString::builder("cafile")
|
||||
.flags(glib::ParamFlags::READWRITE)
|
||||
.build(),
|
||||
glib::ParamSpecEnum::builder_with_default("role", WebRTCSignallerRole::Consumer)
|
||||
.flags(glib::ParamFlags::READWRITE)
|
||||
.build(),
|
||||
]
|
||||
});
|
||||
|
||||
PROPS.as_ref()
|
||||
}
|
||||
|
||||
fn set_property(&self, _id: usize, value: &glib::Value, pspec: &glib::ParamSpec) {
|
||||
match pspec.name() {
|
||||
"uri" => {
|
||||
if let Err(e) = self.set_uri(value.get::<&str>().expect("type checked upstream")) {
|
||||
gst::error!(CAT, "Couldn't set URI: {e:?}");
|
||||
}
|
||||
}
|
||||
"producer-peer-id" => {
|
||||
let mut settings = self.settings.lock().unwrap();
|
||||
|
||||
if !matches!(settings.role, WebRTCSignallerRole::Consumer) {
|
||||
gst::warning!(
|
||||
CAT,
|
||||
"Setting `producer-peer-id` doesn't make sense for {:?}",
|
||||
settings.role
|
||||
);
|
||||
} else {
|
||||
settings.producer_peer_id = value
|
||||
.get::<Option<String>>()
|
||||
.expect("type checked upstream");
|
||||
}
|
||||
}
|
||||
"cafile" => {
|
||||
self.settings.lock().unwrap().cafile = value
|
||||
.get::<Option<String>>()
|
||||
.expect("type checked upstream")
|
||||
}
|
||||
"role" => {
|
||||
self.settings.lock().unwrap().role = value
|
||||
.get::<WebRTCSignallerRole>()
|
||||
.expect("type checked upstream")
|
||||
}
|
||||
_ => unimplemented!(),
|
||||
}
|
||||
}
|
||||
|
||||
fn property(&self, _id: usize, pspec: &glib::ParamSpec) -> glib::Value {
|
||||
let settings = self.settings.lock().unwrap();
|
||||
match pspec.name() {
|
||||
"uri" => settings.uri.to_string().to_value(),
|
||||
"producer-peer-id" => {
|
||||
if !matches!(settings.role, WebRTCSignallerRole::Consumer) {
|
||||
gst::warning!(
|
||||
CAT,
|
||||
"`producer-peer-id` doesn't make sense for {:?}",
|
||||
settings.role
|
||||
);
|
||||
}
|
||||
|
||||
settings.producer_peer_id.to_value()
|
||||
}
|
||||
"cafile" => settings.cafile.to_value(),
|
||||
"role" => settings.role.to_value(),
|
||||
_ => unimplemented!(),
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
impl SignallableImpl for Signaller {
|
||||
fn start(&self) {
|
||||
gst::info!(CAT, imp: self, "Starting");
|
||||
RUNTIME.spawn(glib::clone!(@weak self as this => async move {
|
||||
if let Err(err) = this.connect().await {
|
||||
this.obj().emit_by_name::<()>("error", &[&format!("Error receiving: {}", err)]);
|
||||
}
|
||||
}));
|
||||
}
|
||||
|
||||
fn stop(&self) {
|
||||
gst::info!(CAT, imp: self, "Stopping now");
|
||||
|
||||
let mut state = self.state.lock().unwrap();
|
||||
let send_task_handle = state.send_task_handle.take();
|
||||
let receive_task_handle = state.receive_task_handle.take();
|
||||
if let Some(mut sender) = state.websocket_sender.take() {
|
||||
RUNTIME.block_on(async move {
|
||||
sender.close_channel();
|
||||
|
||||
if let Some(handle) = send_task_handle {
|
||||
if let Err(err) = handle.await {
|
||||
gst::warning!(CAT, imp: self, "Error while joining send task: {}", err);
|
||||
}
|
||||
}
|
||||
|
||||
if let Some(handle) = receive_task_handle {
|
||||
if let Err(err) = handle.await {
|
||||
gst::warning!(CAT, imp: self, "Error while joining receive task: {}", err);
|
||||
}
|
||||
}
|
||||
});
|
||||
}
|
||||
}
|
||||
|
||||
fn send_sdp(&self, session_id: &str, sdp: &gst_webrtc::WebRTCSessionDescription) {
|
||||
gst::debug!(CAT, imp: self, "Sending SDP {sdp:#?}");
|
||||
|
||||
let role = self.settings.lock().unwrap().role;
|
||||
let is_consumer = matches!(role, super::WebRTCSignallerRole::Consumer);
|
||||
|
||||
let msg = p::IncomingMessage::Peer(p::PeerMessage {
|
||||
session_id: session_id.to_owned(),
|
||||
peer_message: p::PeerMessageInner::Sdp(if is_consumer {
|
||||
p::SdpMessage::Answer {
|
||||
sdp: sdp.sdp().as_text().unwrap(),
|
||||
}
|
||||
} else {
|
||||
p::SdpMessage::Offer {
|
||||
sdp: sdp.sdp().as_text().unwrap(),
|
||||
}
|
||||
}),
|
||||
});
|
||||
|
||||
self.send(msg);
|
||||
}
|
||||
|
||||
fn add_ice(
|
||||
&self,
|
||||
session_id: &str,
|
||||
candidate: &str,
|
||||
sdp_m_line_index: Option<u32>,
|
||||
_sdp_mid: Option<String>,
|
||||
) {
|
||||
gst::debug!(
|
||||
CAT,
|
||||
imp: self,
|
||||
"Adding ice candidate {candidate:?} for {sdp_m_line_index:?} on session {session_id}"
|
||||
);
|
||||
|
||||
let msg = p::IncomingMessage::Peer(p::PeerMessage {
|
||||
session_id: session_id.to_string(),
|
||||
peer_message: p::PeerMessageInner::Ice {
|
||||
candidate: candidate.to_string(),
|
||||
sdp_m_line_index: sdp_m_line_index.unwrap(),
|
||||
},
|
||||
});
|
||||
|
||||
self.send(msg);
|
||||
}
|
||||
|
||||
fn end_session(&self, session_id: &str) {
|
||||
gst::debug!(CAT, imp: self, "Signalling session done {}", session_id);
|
||||
|
||||
let state = self.state.lock().unwrap();
|
||||
let session_id = session_id.to_string();
|
||||
if let Some(mut sender) = state.websocket_sender.clone() {
|
||||
RUNTIME.spawn(glib::clone!(@weak self as this => async move {
|
||||
if let Err(err) = sender
|
||||
.send(p::IncomingMessage::EndSession(p::EndSessionMessage {
|
||||
session_id,
|
||||
}))
|
||||
.await
|
||||
{
|
||||
this.obj().emit_by_name::<()>("error", &[&format!("Error: {}", err)]);
|
||||
}
|
||||
}));
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
impl GstObjectImpl for Signaller {}
|
||||
46
net/webrtc/src/webrtcsrc/signaller/mod.rs
Normal file
46
net/webrtc/src/webrtcsrc/signaller/mod.rs
Normal file
|
|
@ -0,0 +1,46 @@
|
|||
mod iface;
|
||||
mod imp;
|
||||
use gst::glib;
|
||||
|
||||
use once_cell::sync::Lazy;
|
||||
// Expose traits and objects from the module itself so it exactly looks like
|
||||
// generated bindings
|
||||
pub use imp::WebRTCSignallerRole;
|
||||
pub mod prelude {
|
||||
pub use {super::SignallableExt, super::SignallableImpl};
|
||||
}
|
||||
|
||||
pub static CAT: Lazy<gst::DebugCategory> = Lazy::new(|| {
|
||||
gst::DebugCategory::new(
|
||||
"webrtcsrc-signaller",
|
||||
gst::DebugColorFlags::empty(),
|
||||
Some("WebRTC src signaller"),
|
||||
)
|
||||
});
|
||||
|
||||
glib::wrapper! {
|
||||
pub struct Signallable(ObjectInterface<iface::Signallable>);
|
||||
}
|
||||
|
||||
glib::wrapper! {
|
||||
pub struct Signaller(ObjectSubclass <imp::Signaller>) @implements Signallable;
|
||||
}
|
||||
|
||||
impl Default for Signaller {
|
||||
fn default() -> Self {
|
||||
glib::Object::builder().build()
|
||||
}
|
||||
}
|
||||
|
||||
impl Signaller {
|
||||
pub fn new(mode: WebRTCSignallerRole) -> Self {
|
||||
glib::Object::builder().property("role", &mode).build()
|
||||
}
|
||||
}
|
||||
|
||||
pub use iface::SignallableExt;
|
||||
pub use iface::SignallableImpl;
|
||||
pub use iface::SignallableImplExt;
|
||||
|
||||
unsafe impl Send for Signallable {}
|
||||
unsafe impl Sync for Signallable {}
|
||||
Loading…
Reference in a new issue