Introduce struct MediaConfig, with construction-time settings.

webrtc/api/statscollector_unittest.cc

 /*
  *  Copyright 2014 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 471 matching lines @@
   double time_now_;
 };
 
 class StatsCollectorTest : public testing::Test {
  protected:
   StatsCollectorTest()
       : media_engine_(new cricket::FakeMediaEngine()),
         channel_manager_(
             new cricket::ChannelManager(media_engine_, rtc::Thread::Current())),
         media_controller_(
-            webrtc::MediaControllerInterface::Create(rtc::Thread::Current(),
+            webrtc::MediaControllerInterface::Create(cricket::MediaConfig(),
+                                                     rtc::Thread::Current(),
                                                      channel_manager_.get())),
         session_(media_controller_.get()) {
     // By default, we ignore session GetStats calls.
     EXPECT_CALL(session_, GetTransportStats(_)).WillRepeatedly(Return(false));
     // Add default returns for mock classes.
     EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull());
     EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
     EXPECT_CALL(pc_, session()).WillRepeatedly(Return(&session_));
     EXPECT_CALL(pc_, sctp_data_channels())
         .WillRepeatedly(ReturnRef(data_channels_));
@@ skipping 1227 matching lines @@
   cricket::VoiceSenderInfo new_voice_sender_info;
   InitVoiceSenderInfo(&new_voice_sender_info);
   cricket::VoiceMediaInfo new_stats_read;
   reports.clear();
   SetupAndVerifyAudioTrackStats(
       new_audio_track.get(), stream_.get(), &stats, &voice_channel, kVcName,
       media_channel, &new_voice_sender_info, NULL, &new_stats_read, &reports);
 }
 
 }  // namespace webrtc