From 43ee6bfc1c991b8fd8117d7f57df2c39f0f9377c Mon Sep 17 00:00:00 2001
From: Taruntej Kanakamalla <taruntej@asymptotic.io>
Date: Wed, 19 Jul 2023 10:35:33 +0530
Subject: 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>
---
 docs/plugins/gst_plugins_cache.json | 32 ++++++++++++++++++++++++++++++++
 1 file changed, 32 insertions(+)

(limited to 'docs')

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",
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