Split out dscp option from VideoOptions to new struct MediaChannelOptions.

talk/app/webrtc/webrtcsession_unittest.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 39 matching lines @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/physicalsocketserver.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/virtualsocketserver.h"
 #include "webrtc/media/base/fakemediaengine.h"
+#include "webrtc/media/base/fakenetworkinterface.h"
 #include "webrtc/media/base/fakevideorenderer.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/webrtc/fakewebrtccall.h"
 #include "webrtc/p2p/base/stunserver.h"
 #include "webrtc/p2p/base/teststunserver.h"
 #include "webrtc/p2p/base/testturnserver.h"
 #include "webrtc/p2p/base/transportchannel.h"
 #include "webrtc/p2p/client/basicportallocator.h"
 
 #define MAYBE_SKIP_TEST(feature)                    \
@@ skipping 4012 matching lines @@
   SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   SetLocalDescriptionWithoutError(offer);
 
   video_channel_ = media_engine_->GetVideoChannel(0);
   voice_channel_ = media_engine_->GetVoiceChannel(0);
 
   ASSERT_TRUE(video_channel_ != NULL);
   ASSERT_TRUE(voice_channel_ != NULL);
-  const cricket::AudioOptions& audio_options = voice_channel_->options();
-  const cricket::VideoOptions& video_options = video_channel_->options();
-  EXPECT_EQ(rtc::Optional<bool>(true), audio_options.dscp);
-  EXPECT_EQ(rtc::Optional<bool>(true), video_options.dscp);
+
+  cricket::FakeNetworkInterface video_network_interface;
+  cricket::FakeNetworkInterface voice_network_interface;
+  video_channel_->SetInterface(&video_network_interface);
+  voice_channel_->SetInterface(&voice_network_interface);
+  EXPECT_EQ(rtc::DSCP_AF41, video_network_interface.dscp());
+  EXPECT_EQ(rtc::DSCP_EF, voice_network_interface.dscp());
+  video_channel_->SetInterface(NULL);
+  voice_channel_->SetInterface(NULL);
 }
 
 TEST_F(WebRtcSessionTest, TestSuspendBelowMinBitrateConstraint) {
   constraints_.reset(new FakeConstraints());
   constraints_->AddOptional(
       webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
       true);
   Init();
   SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
@@ skipping 204 matching lines @@
 }
 
 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled.  That test
 // currently fails because upon disconnection and reconnection OnIceComplete is
 // called more than once without returning to IceGatheringGathering.
 
 INSTANTIATE_TEST_CASE_P(WebRtcSessionTests,
                         WebRtcSessionTest,
                         testing::Values(ALREADY_GENERATED,
                                         DTLS_IDENTITY_STORE));