Refactor FakeAudioDevice to have separate methods for starting recording and playout.

webrtc/modules/audio_device/include/fake_audio_device.h

 /*
  *  Copyright (c) 2013 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_MODULES_AUDIO_DEVICE_INCLUDE_FAKE_AUDIO_DEVICE_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_FAKE_AUDIO_DEVICE_H_
 
 #include "webrtc/modules/audio_device/include/audio_device.h"
 
 namespace webrtc {
 
 class FakeAudioDeviceModule : public AudioDeviceModule {
  public:
   FakeAudioDeviceModule() {}
   virtual ~FakeAudioDeviceModule() {}
   virtual int32_t AddRef() const { return 0; }
   virtual int32_t Release() const { return 0; }
+
+ private:
   virtual int32_t RegisterEventObserver(AudioDeviceObserver* eventCallback) {
     return 0;
   }
   virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) {
     return 0;
   }
   virtual int32_t Init() { return 0; }
   virtual int32_t InitSpeaker() { return 0; }
   virtual int32_t SetPlayoutDevice(uint16_t index) { return 0; }
   virtual int32_t SetPlayoutDevice(WindowsDeviceType device) { return 0; }
@@ skipping 80 matching lines @@
   virtual int32_t StereoRecording(bool* enabled) const { return 0; }
   virtual int32_t SetRecordingChannel(const ChannelType channel) { return 0; }
   virtual int32_t RecordingChannel(ChannelType* channel) const { return 0; }
   virtual int32_t SetPlayoutBuffer(const BufferType type,
                            uint16_t sizeMS = 0) {
     return 0;
   }
   virtual int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS) const {
     return 0;
   }
-  virtual int32_t PlayoutDelay(uint16_t* delayMS) const { return 0; }
+  virtual int32_t PlayoutDelay(uint16_t* delayMS) const {
+    *delayMS = 0;
+    return 0;
+  }
   virtual int32_t RecordingDelay(uint16_t* delayMS) const { return 0; }
   virtual int32_t CPULoad(uint16_t* load) const { return 0; }
   virtual int32_t StartRawOutputFileRecording(
       const char pcmFileNameUTF8[kAdmMaxFileNameSize]) {
     return 0;
   }
   virtual int32_t StopRawOutputFileRecording() { return 0; }
   virtual int32_t StartRawInputFileRecording(
       const char pcmFileNameUTF8[kAdmMaxFileNameSize]) {
     return 0;
@@ skipping 25 matching lines @@
   }
   virtual int GetRecordAudioParameters(AudioParameters* params) const {
     return -1;
   }
 #endif  // WEBRTC_IOS
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_FAKE_AUDIO_DEVICE_H_