From a40d6f49942eef1c3ab5c1e8a896849973152ff2 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= <sebastian@centricular.com>
Date: Thu, 8 Oct 2020 18:49:57 +0300
Subject: [PATCH] Revert "rtpsender: Add API to set the priority"

This reverts commit a8b287c76472c8d7fd38800807c482d020ff4a63.

It breaks the CI until the C# bindings are fixed.
---
 gst-libs/gst/webrtc/rtpsender.c | 68 ++++++---------------------------
 gst-libs/gst/webrtc/rtpsender.h |  9 +----
 2 files changed, 12 insertions(+), 65 deletions(-)

diff --git a/gst-libs/gst/webrtc/rtpsender.c b/gst-libs/gst/webrtc/rtpsender.c
index e14227c0bd..3a8a9044f0 100644
--- a/gst-libs/gst/webrtc/rtpsender.c
+++ b/gst-libs/gst/webrtc/rtpsender.c
@@ -51,7 +51,10 @@ enum
 enum
 {
   PROP_0,
-  PROP_PRIORITY
+  PROP_MID,
+  PROP_SENDER,
+  PROP_STOPPED,
+  PROP_DIRECTION,
 };
 
 //static guint gst_webrtc_rtp_sender_signals[LAST_SIGNAL] = { 0 };
@@ -82,38 +85,11 @@ gst_webrtc_rtp_sender_set_rtcp_transport (GstWebRTCRTPSender * sender,
   GST_OBJECT_UNLOCK (sender);
 }
 
-/**
- * gst_webrtc_rtp_sender_set_priority:
- * @sender: a #GstWebRTCRTPSender
- * @priority: The priority of this sender
- *
- * Sets the content of the IPv4 Type of Service (ToS), also known as DSCP
- * (Differentiated Services Code Point).
- * This also sets the Traffic Class field of IPv6.
- *
- * Since: 1.20
- */
-
-void
-gst_webrtc_rtp_sender_set_priority (GstWebRTCRTPSender * sender,
-    GstWebRTCPriorityType priority)
-{
-  GST_OBJECT_LOCK (sender);
-  sender->priority = priority;
-  GST_OBJECT_UNLOCK (sender);
-  g_object_notify (G_OBJECT (sender), "priority");
-}
-
 static void
 gst_webrtc_rtp_sender_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec)
 {
-  GstWebRTCRTPSender *sender = GST_WEBRTC_RTP_SENDER (object);
-
   switch (prop_id) {
-    case PROP_PRIORITY:
-      gst_webrtc_rtp_sender_set_priority (sender, g_value_get_uint (value));
-      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -124,14 +100,7 @@ static void
 gst_webrtc_rtp_sender_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec)
 {
-  GstWebRTCRTPSender *sender = GST_WEBRTC_RTP_SENDER (object);
-
   switch (prop_id) {
-    case PROP_PRIORITY:
-      GST_OBJECT_LOCK (sender);
-      g_value_set_uint (value, sender->priority);
-      GST_OBJECT_UNLOCK (sender);
-      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -141,15 +110,15 @@ gst_webrtc_rtp_sender_get_property (GObject * object, guint prop_id,
 static void
 gst_webrtc_rtp_sender_finalize (GObject * object)
 {
-  GstWebRTCRTPSender *sender = GST_WEBRTC_RTP_SENDER (object);
+  GstWebRTCRTPSender *webrtc = GST_WEBRTC_RTP_SENDER (object);
 
-  if (sender->transport)
-    gst_object_unref (sender->transport);
-  sender->transport = NULL;
+  if (webrtc->transport)
+    gst_object_unref (webrtc->transport);
+  webrtc->transport = NULL;
 
-  if (sender->rtcp_transport)
-    gst_object_unref (sender->rtcp_transport);
-  sender->rtcp_transport = NULL;
+  if (webrtc->rtcp_transport)
+    gst_object_unref (webrtc->rtcp_transport);
+  webrtc->rtcp_transport = NULL;
 
   G_OBJECT_CLASS (parent_class)->finalize (object);
 }
@@ -162,21 +131,6 @@ gst_webrtc_rtp_sender_class_init (GstWebRTCRTPSenderClass * klass)
   gobject_class->get_property = gst_webrtc_rtp_sender_get_property;
   gobject_class->set_property = gst_webrtc_rtp_sender_set_property;
   gobject_class->finalize = gst_webrtc_rtp_sender_finalize;
-
-  /**
-   * GstWebRTCRTPSender:priority:
-   *
-   * The priority from which to set the DSCP field on packets
-   *
-   * Since: 1.20
-   */
-  g_object_class_install_property (gobject_class,
-      PROP_PRIORITY,
-      g_param_spec_enum ("priority",
-          "Priority",
-          "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));
 }
 
 static void
diff --git a/gst-libs/gst/webrtc/rtpsender.h b/gst-libs/gst/webrtc/rtpsender.h
index 0c5c07751a..bcaf93c604 100644
--- a/gst-libs/gst/webrtc/rtpsender.h
+++ b/gst-libs/gst/webrtc/rtpsender.h
@@ -37,10 +37,6 @@ GType gst_webrtc_rtp_sender_get_type(void);
 
 /**
  * GstWebRTCRTPSender:
- * @transport: The transport for RTP packets
- * @rtcp_transport: The transport for RTCP packets without rtcp-mux
- * @send_encodings: Unused
- * @priority: The priority of the stream (Since: 1.20)
  */
 struct _GstWebRTCRTPSender
 {
@@ -51,7 +47,6 @@ struct _GstWebRTCRTPSender
   GstWebRTCDTLSTransport            *rtcp_transport;
 
   GArray                            *send_encodings;
-  GstWebRTCPriorityType              priority;
 
   gpointer                          _padding[GST_PADDING];
 };
@@ -72,9 +67,7 @@ void                        gst_webrtc_rtp_sender_set_transport         (GstWebR
 GST_WEBRTC_API
 void                        gst_webrtc_rtp_sender_set_rtcp_transport    (GstWebRTCRTPSender * sender,
                                                                          GstWebRTCDTLSTransport * transport);
-GST_WEBRTC_API
-void                        gst_webrtc_rtp_sender_set_priority          (GstWebRTCRTPSender *sender,
-                                                                         GstWebRTCPriorityType priority);
+
 
 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPSender, gst_object_unref)