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 79 matching lines @@
   }
 }
 #endif  // !defined(NDEBUG)
 
 AudioDeviceIOS::AudioDeviceIOS()
     : audio_device_buffer_(nullptr),
       audio_unit_(nullptr),
       recording_(0),
       playing_(0),
       initialized_(false),
-      rec_is_initialized_(false),
-      play_is_initialized_(false),
+      audio_is_initialized_(false),
       is_interrupted_(false),
       has_configured_session_(false) {
   LOGI() << "ctor" << ios::GetCurrentThreadDescription();
   thread_ = rtc::Thread::Current();
   audio_session_observer_ =
       [[RTCAudioSessionDelegateAdapter alloc] initWithObserver:this];
 }
 
 AudioDeviceIOS::~AudioDeviceIOS() {
   LOGI() << "~dtor" << ios::GetCurrentThreadDescription();
@@ skipping 46 matching lines @@
   StopPlayout();
   StopRecording();
   initialized_ = false;
   return 0;
 }
 
 int32_t AudioDeviceIOS::InitPlayout() {
   LOGI() << "InitPlayout";
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   RTC_DCHECK(initialized_);
-  RTC_DCHECK(!play_is_initialized_);
+  RTC_DCHECK(!audio_is_initialized_);
   RTC_DCHECK(!playing_);
-  if (!rec_is_initialized_) {
+  if (!audio_is_initialized_) {
     if (!InitPlayOrRecord()) {
       LOG_F(LS_ERROR) << "InitPlayOrRecord failed for InitPlayout!";
       return -1;
     }
   }
-  play_is_initialized_ = true;
+  audio_is_initialized_ = true;
   return 0;
 }
 
 int32_t AudioDeviceIOS::InitRecording() {
   LOGI() << "InitRecording";
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   RTC_DCHECK(initialized_);
-  RTC_DCHECK(!rec_is_initialized_);
+  RTC_DCHECK(!audio_is_initialized_);
   RTC_DCHECK(!recording_);
-  if (!play_is_initialized_) {
+  if (!audio_is_initialized_) {
     if (!InitPlayOrRecord()) {
       LOG_F(LS_ERROR) << "InitPlayOrRecord failed for InitRecording!";
       return -1;
     }
   }
-  rec_is_initialized_ = true;
+  audio_is_initialized_ = true;
   return 0;
 }
 
 int32_t AudioDeviceIOS::StartPlayout() {
   LOGI() << "StartPlayout";
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  RTC_DCHECK(play_is_initialized_);
+  RTC_DCHECK(audio_is_initialized_);
   RTC_DCHECK(!playing_);
   RTC_DCHECK(audio_unit_);
   if (fine_audio_buffer_) {
     fine_audio_buffer_->ResetPlayout();
   }
   if (!recording_ &&
       audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
     if (!audio_unit_->Start()) {
       RTCLogError(@"StartPlayout failed to start audio unit.");
       return -1;
     }
     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 (!audio_is_initialized_ || !playing_) {
     return 0;
   }
   if (!recording_) {
     ShutdownPlayOrRecord();
+    audio_is_initialized_ = false;
   }
-  play_is_initialized_ = false;
   rtc::AtomicOps::ReleaseStore(&playing_, 0);
   return 0;
 }
 
 int32_t AudioDeviceIOS::StartRecording() {
   LOGI() << "StartRecording";
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  RTC_DCHECK(rec_is_initialized_);
+  RTC_DCHECK(audio_is_initialized_);
   RTC_DCHECK(!recording_);
   RTC_DCHECK(audio_unit_);
   if (fine_audio_buffer_) {
     fine_audio_buffer_->ResetRecord();
   }
   if (!playing_ &&
       audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
     if (!audio_unit_->Start()) {
       RTCLogError(@"StartRecording failed to start audio unit.");
       return -1;
     }
     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 (!audio_is_initialized_ || !recording_) {
     return 0;
   }
   if (!playing_) {
     ShutdownPlayOrRecord();
+    audio_is_initialized_ = false;
   }
-  rec_is_initialized_ = false;
   rtc::AtomicOps::ReleaseStore(&recording_, 0);
   return 0;
 }
 
 // Change the default receiver playout route to speaker.
 int32_t AudioDeviceIOS::SetLoudspeakerStatus(bool enable) {
   LOGI() << "SetLoudspeakerStatus(" << enable << ")";
 
   RTCAudioSession* session = [RTCAudioSession sharedInstance];
   [session lockForConfiguration];
@@ skipping 402 matching lines @@
   RTCLog(@"Updating audio unit state. CanPlayOrRecord=%d IsInterrupted=%d",
          can_play_or_record, is_interrupted_);
 
   if (is_interrupted_) {
     RTCLog(@"Ignoring audio unit update due to interruption.");
     return;
   }
 
   // If we're not initialized we don't need to do anything. Audio unit will
   // be initialized on initialization.
-  if (!rec_is_initialized_ && !play_is_initialized_)
+  if (!audio_is_initialized_)
     return;
 
   // If we're initialized, we must have an audio unit.
   RTC_DCHECK(audio_unit_);
 
   bool should_initialize_audio_unit = false;
   bool should_uninitialize_audio_unit = false;
   bool should_start_audio_unit = false;
   bool should_stop_audio_unit = false;
 
@@ skipping 138 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