Remove AudioDeviceObserver and make ADM not inherit from the Module interface.

modules/audio_device/dummy/file_audio_device.cc

 /*
  *  Copyright (c) 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 380 matching lines @@
   enabled = true;
   return 0;
 }
 
 int32_t FileAudioDevice::PlayoutDelay(uint16_t& delayMS) const {
   return 0;
 }
 
 int32_t FileAudioDevice::RecordingDelay(uint16_t& delayMS) const { return -1; }
 
-bool FileAudioDevice::PlayoutWarning() const { return false; }
-
-bool FileAudioDevice::PlayoutError() const { return false; }
-
-bool FileAudioDevice::RecordingWarning() const { return false; }
-
-bool FileAudioDevice::RecordingError() const { return false; }
-
-void FileAudioDevice::ClearPlayoutWarning() {}
-
-void FileAudioDevice::ClearPlayoutError() {}
-
-void FileAudioDevice::ClearRecordingWarning() {}
-
-void FileAudioDevice::ClearRecordingError() {}
-
 void FileAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
   rtc::CritScope lock(&_critSect);
 
   _ptrAudioBuffer = audioBuffer;
 
   // Inform the AudioBuffer about default settings for this implementation.
   // Set all values to zero here since the actual settings will be done by
   // InitPlayout and InitRecording later.
   _ptrAudioBuffer->SetRecordingSampleRate(0);
   _ptrAudioBuffer->SetPlayoutSampleRate(0);
@@ skipping 72 matching lines @@
 
     int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
     if (deltaTimeMillis < 10) {
       SleepMs(10 - deltaTimeMillis);
     }
 
     return true;
 }
 
 }  // namespace webrtc