Ensures that ADM for Android and iOS uses identical states when stopping audio

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 199 matching lines @@
     }
     LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
   }
   rtc::AtomicOps::ReleaseStore(&playing_, 1);
   return 0;
 }
 
 int32_t AudioDeviceIOS::StopPlayout() {
   LOGI() << "StopPlayout";
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  if (!play_is_initialized_) {
-    return 0;
-  }
-  if (!playing_) {
-    play_is_initialized_ = false;
+  if (!play_is_initialized_ || !playing_) {

[sophiechang-webrtc, 2016/09/21 10:43:54]

as mentioned offline, i think it would be good to merge the rec_is_initialized_
and play_is_initialized_ to some sort of audio_is_initialized_ variable since
they both just control the audio unit. and also only set audio_is_initialized_
to false when you call ShutdownPlayOrRecord so stick it into that if block in
line 223 and the same thing around StopRecording. 

     return 0;
   }
   if (!recording_) {
     ShutdownPlayOrRecord();
   }
   play_is_initialized_ = false;
   rtc::AtomicOps::ReleaseStore(&playing_, 0);
   return 0;
 }
 
@@ skipping 14 matching lines @@
     }
     LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
   }
   rtc::AtomicOps::ReleaseStore(&recording_, 1);
   return 0;
 }
 
 int32_t AudioDeviceIOS::StopRecording() {
   LOGI() << "StopRecording";
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  if (!rec_is_initialized_) {
-    return 0;
-  }
-  if (!recording_) {
-    rec_is_initialized_ = false;
+  if (!rec_is_initialized_ || !recording_) {
     return 0;
   }
   if (!playing_) {
     ShutdownPlayOrRecord();
   }
   rec_is_initialized_ = false;
   rtc::AtomicOps::ReleaseStore(&recording_, 0);
   return 0;
 }
 
@@ skipping 567 matching lines @@
 
   // All I/O should be stopped or paused prior to deactivating the audio
   // session, hence we deactivate as last action.
   [session lockForConfiguration];
   UnconfigureAudioSession();
   [session endWebRTCSession:nil];
   [session unlockForConfiguration];
 }
 
 }  // namespace webrtc