Trivial initialization fix in AudioDeviceIOS

webrtc/modules/audio_device/ios/audio_device_ios.mm

 /*
  *  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 180 matching lines @@
 #endif  // !defined(NDEBUG)
 
 AudioDeviceIOS::AudioDeviceIOS()
     : audio_device_buffer_(nullptr),
       vpio_unit_(nullptr),
       recording_(0),
       playing_(0),
       initialized_(false),
       rec_is_initialized_(false),
       play_is_initialized_(false),
-      audio_interruption_observer_(nullptr) {
+      audio_interruption_observer_(nullptr),
+      route_change_observer_(nullptr) {
   LOGI() << "ctor" << ios::GetCurrentThreadDescription();
 }
 
 AudioDeviceIOS::~AudioDeviceIOS() {
   LOGI() << "~dtor";
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   Terminate();
 }
 
 void AudioDeviceIOS::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
@@ skipping 705 matching lines @@
   // Read decoded 16-bit PCM samples from WebRTC (using a size that matches
   // the native I/O audio unit) to a preallocated intermediate buffer and
   // copy the result to the audio buffer in the |io_data| destination.
   SInt8* source = playout_audio_buffer_.get();
   fine_audio_buffer_->GetPlayoutData(source);
   memcpy(destination, source, dataSizeInBytes);
   return noErr;
 }
 
 }  // namespace webrtc