Minor refactoring of the AudioDeviceBuffer class

webrtc/modules/audio_device/audio_device_buffer.h

 /*
  *  Copyright (c) 2012 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 10 matching lines @@
 const uint32_t kPulsePeriodMs = 1000;
 const size_t kMaxBufferSizeBytes = 3840;  // 10ms in stereo @ 96kHz
 
 class AudioDeviceObserver;
 
 class AudioDeviceBuffer {
  public:
   AudioDeviceBuffer();
   virtual ~AudioDeviceBuffer();
 
-  void SetId(uint32_t id);
+  void SetId(uint32_t id) {};
   int32_t RegisterAudioCallback(AudioTransport* audioCallback);
 
   int32_t InitPlayout();
   int32_t InitRecording();
 
   virtual int32_t SetRecordingSampleRate(uint32_t fsHz);
   virtual int32_t SetPlayoutSampleRate(uint32_t fsHz);
   int32_t RecordingSampleRate() const;
   int32_t PlayoutSampleRate() const;
 
@@ skipping 14 matching lines @@
   virtual int32_t GetPlayoutData(void* audioBuffer);
 
   int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]);
   int32_t StopInputFileRecording();
   int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]);
   int32_t StopOutputFileRecording();
 
   int32_t SetTypingStatus(bool typingStatus);
 
  private:
-  int32_t _id;
   CriticalSectionWrapper& _critSect;
   CriticalSectionWrapper& _critSectCb;
 
   AudioTransport* _ptrCbAudioTransport;
 
   uint32_t _recSampleRate;
   uint32_t _playSampleRate;
 
   size_t _recChannels;
   size_t _playChannels;
@@ skipping 29 matching lines @@
 
   int _playDelayMS;
   int _recDelayMS;
   int _clockDrift;
   int high_delay_counter_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H