Late initialize MediaController, for less resource i.e. ProcessThread, usage by PeerConnection.

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 48 matching lines @@
 // Error return values
 const char kNotFound[] = "NOT FOUND";
 
 // Constant names for track identification.
 const char kLocalTrackId[] = "local_track_id";
 const char kRemoteTrackId[] = "remote_track_id";
 const uint32_t kSsrcOfTrack = 1234;
 
 class MockWebRtcSession : public webrtc::WebRtcSession {
  public:
-  explicit MockWebRtcSession(webrtc::MediaControllerInterface* media_controller)
-      : WebRtcSession(media_controller,
-                      rtc::Thread::Current(),
+  explicit MockWebRtcSession()
+      : WebRtcSession(rtc::Thread::Current(),
                       rtc::Thread::Current(),
                       nullptr) {}
   MOCK_METHOD0(voice_channel, cricket::VoiceChannel*());
   MOCK_METHOD0(video_channel, cricket::VideoChannel*());
   // Libjingle uses "local" for a outgoing track, and "remote" for a incoming
   // track.
   MOCK_METHOD2(GetLocalTrackIdBySsrc, bool(uint32_t, std::string*));
   MOCK_METHOD2(GetRemoteTrackIdBySsrc, bool(uint32_t, std::string*));
   MOCK_METHOD1(GetTransportStats, bool(SessionStats*));
   MOCK_METHOD2(GetLocalCertificate,
@@ skipping 397 matching lines @@
   }
 
  private:
   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(cricket::MediaConfig(),
-                                                     rtc::Thread::Current(),
-                                                     channel_manager_.get())),
-        session_(media_controller_.get()) {
+        session_() {
     // 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 234 matching lines @@
                   TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA),
               dtls_cipher_suite);
     std::string srtp_crypto_suite =
         ExtractStatsValue(StatsReport::kStatsReportTypeComponent, reports,
                           StatsReport::kStatsValueNameSrtpCipher);
     EXPECT_EQ(rtc::SrtpCryptoSuiteToName(rtc::SRTP_AES128_CM_SHA1_80),
               srtp_crypto_suite);
   }
 
   cricket::FakeMediaEngine* media_engine_;
-  rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
-  rtc::scoped_ptr<webrtc::MediaControllerInterface> media_controller_;
   MockWebRtcSession session_;
   MockPeerConnection pc_;
   FakeDataChannelProvider data_channel_provider_;
   SessionStats session_stats_;
   rtc::scoped_refptr<webrtc::MediaStream> stream_;
   rtc::scoped_refptr<webrtc::VideoTrack> track_;
   rtc::scoped_refptr<FakeAudioTrack> audio_track_;
   std::vector<rtc::scoped_refptr<DataChannel>> data_channels_;
 };
 
@@ skipping 969 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