From 86c009e7aa9e9900dfac13be654301a39fd38a7d Mon Sep 17 00:00:00 2001
From: Mathieu Duponchelle <mathieu@centricular.com>
Date: Tue, 12 Jan 2021 19:12:42 +0100
Subject: [PATCH] webrtc: expose transport property on sender and receiver

As advised by !1366#note_629558 , the nice transport should be
accessed through:

> transceiver->sender/receiver->transport/rtcp_transport->icetransport

All the objects on the path can be accessed through properties
except sender/receiver->transport. This patch addresses that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1952>
---
 ext/webrtc/webrtctransceiver.c    |  9 +++++++--
 gst-libs/gst/webrtc/rtpreceiver.c | 21 +++++++++++++++++++++
 gst-libs/gst/webrtc/rtpsender.c   | 22 +++++++++++++++++++++-
 3 files changed, 49 insertions(+), 3 deletions(-)

diff --git a/ext/webrtc/webrtctransceiver.c b/ext/webrtc/webrtctransceiver.c
index db012f3922..aa9f1e2a05 100644
--- a/ext/webrtc/webrtctransceiver.c
+++ b/ext/webrtc/webrtctransceiver.c
@@ -59,12 +59,17 @@ webrtc_transceiver_set_transport (WebRTCTransceiver * trans,
 
   gst_object_replace ((GstObject **) & trans->stream, (GstObject *) stream);
 
-  if (rtp_trans->sender)
+  if (rtp_trans->sender) {
     gst_object_replace ((GstObject **) & rtp_trans->sender->transport,
         (GstObject *) stream->transport);
-  if (rtp_trans->receiver)
+    g_object_notify (G_OBJECT (rtp_trans->sender), "transport");
+  }
+
+  if (rtp_trans->receiver) {
     gst_object_replace ((GstObject **) & rtp_trans->receiver->transport,
         (GstObject *) stream->transport);
+    g_object_notify (G_OBJECT (rtp_trans->receiver), "transport");
+  }
 }
 
 GstWebRTCDTLSTransport *
diff --git a/gst-libs/gst/webrtc/rtpreceiver.c b/gst-libs/gst/webrtc/rtpreceiver.c
index fb8f6e658c..5f4c4d4e50 100644
--- a/gst-libs/gst/webrtc/rtpreceiver.c
+++ b/gst-libs/gst/webrtc/rtpreceiver.c
@@ -49,6 +49,7 @@ enum
 enum
 {
   PROP_0,
+  PROP_TRANSPORT,
 };
 
 //static guint gst_webrtc_rtp_receiver_signals[LAST_SIGNAL] = { 0 };
@@ -68,7 +69,13 @@ static void
 gst_webrtc_rtp_receiver_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec)
 {
+  GstWebRTCRTPReceiver *receiver = GST_WEBRTC_RTP_RECEIVER (object);
   switch (prop_id) {
+    case PROP_TRANSPORT:
+      GST_OBJECT_LOCK (receiver);
+      g_value_set_object (value, receiver->transport);
+      GST_OBJECT_UNLOCK (receiver);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -95,6 +102,20 @@ gst_webrtc_rtp_receiver_class_init (GstWebRTCRTPReceiverClass * klass)
   gobject_class->get_property = gst_webrtc_rtp_receiver_get_property;
   gobject_class->set_property = gst_webrtc_rtp_receiver_set_property;
   gobject_class->finalize = gst_webrtc_rtp_receiver_finalize;
+
+  /**
+   * GstWebRTCRTPReceiver:transport:
+   *
+   * The DTLS transport for this receiver
+   *
+   * Since: 1.20
+   */
+  g_object_class_install_property (gobject_class,
+      PROP_TRANSPORT,
+      g_param_spec_object ("transport", "Transport",
+          "The DTLS transport for this receiver",
+          GST_TYPE_WEBRTC_DTLS_TRANSPORT,
+          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
 }
 
 static void
diff --git a/gst-libs/gst/webrtc/rtpsender.c b/gst-libs/gst/webrtc/rtpsender.c
index 388c48d171..90221dd56c 100644
--- a/gst-libs/gst/webrtc/rtpsender.c
+++ b/gst-libs/gst/webrtc/rtpsender.c
@@ -51,7 +51,8 @@ enum
 enum
 {
   PROP_0,
-  PROP_PRIORITY
+  PROP_PRIORITY,
+  PROP_TRANSPORT,
 };
 
 //static guint gst_webrtc_rtp_sender_signals[LAST_SIGNAL] = { 0 };
@@ -106,6 +107,11 @@ gst_webrtc_rtp_sender_get_property (GObject * object, guint prop_id,
       g_value_set_uint (value, sender->priority);
       GST_OBJECT_UNLOCK (sender);
       break;
+    case PROP_TRANSPORT:
+      GST_OBJECT_LOCK (sender);
+      g_value_set_object (value, sender->transport);
+      GST_OBJECT_UNLOCK (sender);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -147,6 +153,20 @@ gst_webrtc_rtp_sender_class_init (GstWebRTCRTPSenderClass * klass)
           "The priority from which to set the DSCP field on packets",
           GST_TYPE_WEBRTC_PRIORITY_TYPE, GST_WEBRTC_PRIORITY_TYPE_LOW,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstWebRTCRTPSender:transport:
+   *
+   * The DTLS transport for this sender
+   *
+   * Since: 1.20
+   */
+  g_object_class_install_property (gobject_class,
+      PROP_TRANSPORT,
+      g_param_spec_object ("transport", "Transport",
+          "The DTLS transport for this sender",
+          GST_TYPE_WEBRTC_DTLS_TRANSPORT,
+          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
 }
 
 static void