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

modules/audio_device/android/audio_device_template.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.
  */
 
@@ skipping 373 matching lines @@
   }
 
   int32_t RecordingDelay(uint16_t& delay_ms) const override {
     // Best guess we can do is to use half of the estimated total delay.
     LOG(INFO) << __FUNCTION__;
     delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2;
     RTC_DCHECK_GT(delay_ms, 0);
     return 0;
   }
 
-  bool PlayoutWarning() const override {
-    return false;
-  }
-
-  bool PlayoutError() const override {
-    return false;
-  }
-
-  bool RecordingWarning() const override {
-    return false;
-  }
-
-  bool RecordingError() const override {
-    return false;
-  }
-
-  void ClearPlayoutWarning() override { LOG(INFO) << __FUNCTION__; }
-
-  void ClearPlayoutError() override { LOG(INFO) << __FUNCTION__; }
-
-  void ClearRecordingWarning() override { LOG(INFO) << __FUNCTION__; }
-
-  void ClearRecordingError() override { LOG(INFO) << __FUNCTION__; }
-
   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override {
     LOG(INFO) << __FUNCTION__;
     output_.AttachAudioBuffer(audioBuffer);
     input_.AttachAudioBuffer(audioBuffer);
   }
 
   // TODO(henrika): remove
   int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) override {
     FATAL() << "Should never be called";
     return -1;
@@ skipping 81 matching lines @@
   OutputType output_;
 
   InputType input_;
 
   bool initialized_;
 };
 
 }  // namespace webrtc
 
 #endif  // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_