Improved error handling in iOS ADM to avoid race during init

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.
  */
 
@@ skipping 164 matching lines @@
 
   // Restarts active audio streams using a new sample rate. Required when e.g.
   // a BT headset is enabled or disabled.
   bool RestartAudioUnitWithNewFormat(float sample_rate);
 
   // Activates our audio session, creates and initializes the voice-processing
   // audio unit and verifies that we got the preferred native audio parameters.
   bool InitPlayOrRecord();
 
   // Closes and deletes the voice-processing I/O unit.
-  bool ShutdownPlayOrRecord();
+  void ShutdownPlayOrRecord();
+
+  // Helper method for destroying the existing audio unit.
+  void DisposeAudioUnit();
 
   // Callback function called on a real-time priority I/O thread from the audio
   // unit. This method is used to signal that recorded audio is available.
   static OSStatus RecordedDataIsAvailable(
       void* in_ref_con,
       AudioUnitRenderActionFlags* io_action_flags,
       const AudioTimeStamp* time_stamp,
       UInt32 in_bus_number,
       UInt32 in_number_frames,
       AudioBufferList* io_data);
@@ skipping 90 matching lines @@
   void* audio_interruption_observer_;
   void* route_change_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_