Use RTCAudioSessionDelegate in AudioDeviceIOS.

webrtc/modules/audio_device/ios/audio_device_ios.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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
 
 #include <memory>
 
 #include <AudioUnit/AudioUnit.h>
 
+#include "webrtc/base/objc/RTCMacros.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 
+RTC_FWD_DECL_OBJC_CLASS(RTCAudioSessionDelegateAdapter);
+
 namespace webrtc {
 
 class FineAudioBuffer;
 
 // Implements full duplex 16-bit mono PCM audio support for iOS using a
 // Voice-Processing (VP) I/O audio unit in Core Audio. The VP I/O audio unit
 // supports audio echo cancellation. It also adds automatic gain control,
 // adjustment of voice-processing quality and muting.
 //
 // An instance must be created and destroyed on one and the same thread.
@@ skipping 113 matching lines @@
   int32_t CPULoad(uint16_t& load) const override;
   bool PlayoutWarning() const override;
   bool PlayoutError() const override;
   bool RecordingWarning() const override;
   bool RecordingError() const override;
   void ClearPlayoutWarning() override {}
   void ClearPlayoutError() override {}
   void ClearRecordingWarning() override {}
   void ClearRecordingError() override {}
 
+  // These methods should be called in response to audio events.

[henrika_webrtc, 2016/03/15 08:53:44]

Any restrictions on the calling thread?

[tkchin_webrtc, 2016/03/15 20:14:58]

Done.

+  void OnInterruptionBegin();
+  void OnInterruptionEnd();
+  void OnValidRouteChange();
+
  private:
   // Uses current |playout_parameters_| and |record_parameters_| to inform the
   // audio device buffer (ADB) about our internal audio parameters.
   void UpdateAudioDeviceBuffer();
 
-  // Registers observers for the AVAudioSessionRouteChangeNotification and
-  // AVAudioSessionInterruptionNotification notifications.
-  void RegisterNotificationObservers();
-  void UnregisterNotificationObservers();
-
   // Since the preferred audio parameters are only hints to the OS, the actual
   // values may be different once the AVAudioSession has been activated.
   // This method asks for the current hardware parameters and takes actions
   // if they should differ from what we have asked for initially. It also
   // defines |playout_parameters_| and |record_parameters_|.
   void SetupAudioBuffersForActiveAudioSession();
 
   // Creates a Voice-Processing I/O unit and configures it for full-duplex
   // audio. The selected stream format is selected to avoid internal resampling
   // and to match the 10ms callback rate for WebRTC as well as possible.
@@ skipping 105 matching lines @@
 
   // Set to true after successful call to Init(), false otherwise.
   bool initialized_;
 
   // Set to true after successful call to InitRecording(), false otherwise.
   bool rec_is_initialized_;
 
   // Set to true after successful call to InitPlayout(), false otherwise.
   bool play_is_initialized_;
 
+  // Set to true if audio session is interrupted, false otherwise.
+  bool is_interrupted_;
+
   // Audio interruption observer instance.
-  void* audio_interruption_observer_;
-  void* route_change_observer_;
+  RTCAudioSessionDelegateAdapter* audio_session_observer_;
 
   // Contains the audio data format specification for a stream of audio.
   AudioStreamBasicDescription application_format_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_