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 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"

[perkj_webrtc, 2016/02/11 08:04:06]

used?

[nisse-webrtc, 2016/02/11 09:16:00]

Done.

 #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 286 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));