diff options
| author | Taruntej Kanakamalla <taruntej@asymptotic.io> | 2023-07-19 08:05:33 +0300 |
|---|---|---|
| committer | GStreamer Marge Bot <gitlab-merge-bot@gstreamer-foundation.org> | 2023-11-17 21:08:44 +0300 |
| commit | 43ee6bfc1c991b8fd8117d7f57df2c39f0f9377c (patch) | |
| tree | c36ec158c082b8da54ed19bed60af891a506e5ca /docs | |
| parent | ed3aa740bedf2cf308d91d89b017071af34260a2 (diff) | |
net/webrtc: add whipserversrc
Implement new signaller WhipServerSignaller
- an http server using 'warp'
- handlers for the POST, OPTIONS, PATCH and DELETE
- fixed path `/whip/endpoint` as the URI
- fixed value 'whip-client' as the producer peer id
- fixed resource url `/whip/resource/whip-client`
Derive whipserversrc element from BaseWebRTCSrc
- implement constructed method for ObjectImpl to set
non-default signaller, i.e., WhipServerSignaller
- bind the properties stun-server and turn-servers to those on
the Signaller
Connect to 'webrtcbin-ready' signal in the constructor of WhipServerSignaller
- it will be emitted by the webrtcsrc when the webrtcbin element is ready
- the closure for this signal will in turn connect to webrtcbin's ice-gathering-state
and perform send with the answer sdp via the channel
- the WhipServer will hold its HTTP response in POST handler until this signal
is received or timeout which happens early
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
Diffstat (limited to 'docs')
| -rw-r--r-- | docs/plugins/gst_plugins_cache.json | 32 |
1 files changed, 32 insertions, 0 deletions
diff --git a/docs/plugins/gst_plugins_cache.json b/docs/plugins/gst_plugins_cache.json index 1539cec57..fc7ebb5df 100644 --- a/docs/plugins/gst_plugins_cache.json +++ b/docs/plugins/gst_plugins_cache.json @@ -6499,6 +6499,38 @@ } }, "rank": "none" + }, + "whipserversrc": { + "author": "Taruntej Kanakamalla <taruntej@asymptotic.io>", + "description": "WebRTC source element using WHIP Server as the signaller", + "hierarchy": [ + "GstWhipServerSrc", + "GstBaseWebRTCSrc", + "GstBin", + "GstElement", + "GstObject", + "GInitiallyUnowned", + "GObject" + ], + "interfaces": [ + "GstChildProxy" + ], + "klass": "Source/Network/WebRTC", + "pad-templates": { + "audio_%%u": { + "caps": "audio/x-raw(ANY):\napplication/x-rtp:\naudio/x-opus:\n", + "direction": "src", + "presence": "sometimes", + "type": "GstWebRTCSrcPad" + }, + "video_%%u": { + "caps": "video/x-raw(ANY):\napplication/x-rtp:\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\n", + "direction": "src", + "presence": "sometimes", + "type": "GstWebRTCSrcPad" + } + }, + "rank": "primary" } }, "filename": "gstrswebrtc", |