Introduce struct MediaConfig, with construction-time settings.

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 346 matching lines @@
       : 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::MediaConfig(),
                                                      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 3690 matching lines @@
   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) {
+// TODO(nisse): Test needs refactoring, see cl
+// https://codereview.webrtc.org/1646253004
+TEST_F(WebRtcSessionTest, DISABLED_TestDscpConstraint) {
+#if 0

[pthatcher1, 2016/02/05 17:11:30]

Are you going to fix this in this CL?

[nisse-webrtc, 2016/02/08 09:02:32]

Yes. I think the previous cl (https://codereview.webrtc.org/1646253004/)
includes the needed changes, patchset 4, I just have to move it over.

The plan is to use FakeNetworkInterface and check the dscp property there, and
not use the options() method. 

(I'd actually prefer to delete the options() method, SetOptions should be able
to apply options in any way it needs without keeping a literal copy, but that's
unrelated to this cl).

[nisse-webrtc, 2016/02/08 12:22:30]

I'm getting into trouble with the webrtcsession unittests.

Problem is, the MediaController is created by PeerConnection::Initialize. So
populating MediaConfig, and looking for the corresponding constraints, is moved
to this method. After creating the MediaController, it creates a WebRtcSession
and passes the MediaController to it.

But in WebRtcSessionTest.TestDscpConstraint, there's no PeerConnection involved,
and hence the code to handle the constraint, in PeerConnection::Initialize,
simply has no place in the test. I guess the sanest solution is to somehow move
this test to the peerconnection unittests?

[pthatcher1, 2016/02/08 19:03:01]

I agree that the right thing to do is move it to peerconnection unittests.

   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);
+#endif
 }
 
 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));