Move some send stream configuration into webrtc::AudioSendStream.

webrtc/test/mock_voice_engine.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_AUDIO_MOCK_VOICE_ENGINE_H_
 #define WEBRTC_AUDIO_MOCK_VOICE_ENGINE_H_
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 namespace test {
 
 // NOTE: This class inherits from VoiceEngineImpl so that its clients will be
 // able to get the various interfaces as usual, via T::GetInterface().
-class MockVoiceEngine final : public VoiceEngineImpl {
+class MockVoiceEngine : public VoiceEngineImpl {
  public:
   MockVoiceEngine() : VoiceEngineImpl(new Config(), true) {
     // Increase ref count so this object isn't automatically deleted whenever
     // interfaces are Release():d.
     ++_ref_count;
   }
   ~MockVoiceEngine() override {
     // Decrease ref count before base class d-tor is called; otherwise it will
     // trigger an assertion.
     --_ref_count;
@@ skipping 280 matching lines @@
                int(int channel, unsigned& level));
   MOCK_METHOD2(SetChannelOutputVolumeScaling, int(int channel, float scaling));
   MOCK_METHOD2(GetChannelOutputVolumeScaling, int(int channel, float& scaling));
   MOCK_METHOD3(SetOutputVolumePan, int(int channel, float left, float right));
   MOCK_METHOD3(GetOutputVolumePan, int(int channel, float& left, float& right));
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_MOCK_VOICE_ENGINE_H_