Move some send stream configuration into webrtc::AudioSendStream.

webrtc/call/call_unittest.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
 #include <list>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/audio_state.h"
 #include "webrtc/call.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace {
 
 struct CallHelper {
   CallHelper() {
-    EXPECT_CALL(voice_engine_,
-        RegisterVoiceEngineObserver(testing::_)).WillOnce(testing::Return(0));
-    EXPECT_CALL(voice_engine_,
-        DeRegisterVoiceEngineObserver()).WillOnce(testing::Return(0));
-    EXPECT_CALL(voice_engine_,
-        GetEventLog()).WillOnce(testing::Return(nullptr));
     webrtc::AudioState::Config audio_state_config;
     audio_state_config.voice_engine = &voice_engine_;
     webrtc::Call::Config config;
     config.audio_state = webrtc::AudioState::Create(audio_state_config);
     call_.reset(webrtc::Call::Create(config));
   }
 
   webrtc::Call* operator->() { return call_.get(); }
 
  private:
-  webrtc::test::MockVoiceEngine voice_engine_;
+  testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_;
   rtc::scoped_ptr<webrtc::Call> call_;
 };
 }  // namespace
 
 namespace webrtc {
 
 TEST(CallTest, ConstructDestruct) {
   CallHelper call;
 }
 
@@ skipping 56 matching lines @@
         streams.push_front(stream);
       }
     }
     for (auto s : streams) {
       call->DestroyAudioReceiveStream(s);
     }
     streams.clear();
   }
 }
 }  // namespace webrtc