Introduce struct MediaConfig, with construction-time settings.

webrtc/api/webrtcsession_unittest.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 311 matching lines @@
   WebRtcSessionTest()
       : media_engine_(new cricket::FakeMediaEngine()),
         data_engine_(new cricket::FakeDataEngine()),
         channel_manager_(
             new cricket::ChannelManager(media_engine_,
                                         data_engine_,
                                         new cricket::CaptureManager(),
                                         rtc::Thread::Current())),
         fake_call_(webrtc::Call::Config()),
         media_controller_(
-            webrtc::MediaControllerInterface::Create(rtc::Thread::Current(),
+            webrtc::MediaControllerInterface::Create(cricket::MediaConfig(),
+                                                     rtc::Thread::Current(),
                                                      channel_manager_.get())),
         tdesc_factory_(new cricket::TransportDescriptionFactory()),
         desc_factory_(
             new cricket::MediaSessionDescriptionFactory(channel_manager_.get(),
                                                         tdesc_factory_.get())),
         pss_(new rtc::PhysicalSocketServer),
         vss_(new rtc::VirtualSocketServer(pss_.get())),
         fss_(new rtc::FirewallSocketServer(vss_.get())),
         ss_scope_(fss_.get()),
         stun_socket_addr_(
@@ skipping 3689 matching lines @@
   TransportInfo* audio = sdp->GetTransportInfoByName("audio");
   ASSERT_TRUE(audio != NULL);
   ASSERT_TRUE(audio->description.identity_fingerprint.get() == NULL);
   audio->description.identity_fingerprint.reset(
       rtc::SSLFingerprint::CreateFromRfc4572(
           rtc::DIGEST_SHA_256, kFakeDtlsFingerprint));
   SetRemoteDescriptionOfferExpectError(kSdpWithoutSdesCrypto,
                                        offer);
 }
 
-// This test verifies DSCP is properly applied on the media channels.
-TEST_F(WebRtcSessionTest, TestDscpConstraint) {
-  constraints_.reset(new FakeConstraints());
-  constraints_->AddOptional(
-      webrtc::MediaConstraintsInterface::kEnableDscp, true);
-  Init();
-  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);
-}
-
 TEST_F(WebRtcSessionTest, TestSuspendBelowMinBitrateConstraint) {
   constraints_.reset(new FakeConstraints());
   constraints_->AddOptional(
       webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
       true);
   Init();
   SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
 
   SetLocalDescriptionWithoutError(offer);
@@ skipping 202 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));