commit 3a1b92772f661b6b52710fee27c5707bcfdda4a6
author Bjorn A Mellem <mellem@webrtc.org> Fri May 24 23:13:08 2019
committer Commit Bot <commit-bot@chromium.org> Fri May 24 23:58:46 2019
tree aef94b23b6ab3f37c3f9160a5fe15576c9817e7b
parent 8b096a03b48562b5f4e84784a3d6b06917dee334

Remove rtp_ and rtcp_packet_transport() from the RtpTransport interface.

RtpTransportInternal does not need to expose these.  They are only used
by tests and for setting options.  Instead, it can expose a SetRtpOption
and SetRtcpOption to set options relevant to each of its transports.

Also updates tests to work around no longer having access to internals.

This will simplify the composite needed during negotiation of different
RTP transport types, as we no longer need to have composites of both
RtpTransport and PacketTransport.

Bug: webrtc:9719
Change-Id: I91bfa6e95b7aa384d10497f47e7d2483c2e0bef2
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138282
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28066}

pc/peer_connection_bundle_unittest.cc

diff --git a/pc/peer_connection_bundle_unittest.cc b/pc/peer_connection_bundle_unittest.cc
index b369db2..57d059c 100644
--- a/pc/peer_connection_bundle_unittest.cc
+++ b/pc/peer_connection_bundle_unittest.cc
@@ -83,14 +83,8 @@
     return false;
   }
 
-  rtc::PacketTransportInternal* voice_rtp_transport() {
-    return (voice_channel() ? voice_channel()->rtp_packet_transport()
-                            : nullptr);
-  }
-
-  rtc::PacketTransportInternal* voice_rtcp_transport() {
-    return (voice_channel() ? voice_channel()->rtcp_packet_transport()
-                            : nullptr);
+  RtpTransportInternal* voice_rtp_transport() {
+    return (voice_channel() ? voice_channel()->rtp_transport() : nullptr);
   }
 
   cricket::VoiceChannel* voice_channel() {
@@ -104,14 +98,8 @@
     return nullptr;
   }
 
-  rtc::PacketTransportInternal* video_rtp_transport() {
-    return (video_channel() ? video_channel()->rtp_packet_transport()
-                            : nullptr);
-  }
-
-  rtc::PacketTransportInternal* video_rtcp_transport() {
-    return (video_channel() ? video_channel()->rtcp_packet_transport()
-                            : nullptr);
+  RtpTransportInternal* video_rtp_transport() {
+    return (video_channel() ? video_channel()->rtp_transport() : nullptr);
   }
 
   cricket::VideoChannel* video_channel() {
@@ -552,14 +540,14 @@
 
   ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
 
-  EXPECT_FALSE(caller->voice_rtcp_transport());
-  EXPECT_FALSE(caller->video_rtcp_transport());
+  EXPECT_FALSE(caller->voice_rtp_transport()->rtcp_mux_enabled());
+  EXPECT_FALSE(caller->video_rtp_transport()->rtcp_mux_enabled());
 
   ASSERT_TRUE(
       caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
 
-  EXPECT_FALSE(caller->voice_rtcp_transport());
-  EXPECT_FALSE(caller->video_rtcp_transport());
+  EXPECT_TRUE(caller->voice_rtp_transport()->rtcp_mux_enabled());
+  EXPECT_TRUE(caller->video_rtp_transport()->rtcp_mux_enabled());
 }
 
 // When negotiating RTCP multiplexing, the PeerConnection makes RTCP transports
@@ -573,14 +561,14 @@
 
   ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
 
-  EXPECT_TRUE(caller->voice_rtcp_transport());
-  EXPECT_TRUE(caller->video_rtcp_transport());
+  EXPECT_FALSE(caller->voice_rtp_transport()->rtcp_mux_enabled());
+  EXPECT_FALSE(caller->video_rtp_transport()->rtcp_mux_enabled());
 
   ASSERT_TRUE(
       caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
 
-  EXPECT_FALSE(caller->voice_rtcp_transport());
-  EXPECT_FALSE(caller->video_rtcp_transport());
+  EXPECT_TRUE(caller->voice_rtp_transport()->rtcp_mux_enabled());
+  EXPECT_TRUE(caller->video_rtp_transport()->rtcp_mux_enabled());
 }
 
 TEST_P(PeerConnectionBundleTest, FailToSetDescriptionWithBundleAndNoRtcpMux) {