Add logging around audio session interruptions

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 308 matching lines @@
 int AudioDeviceIOS::GetRecordAudioParameters(AudioParameters* params) const {
   LOGI() << "GetRecordAudioParameters";
   RTC_DCHECK(record_parameters_.is_valid());
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   *params = record_parameters_;
   return 0;
 }
 
 void AudioDeviceIOS::OnInterruptionBegin() {
   RTC_DCHECK(thread_);
+  LOGI() << "OnInterruptionBegin";
   thread_->Post(RTC_FROM_HERE, this, kMessageTypeInterruptionBegin);
 }
 
 void AudioDeviceIOS::OnInterruptionEnd() {
   RTC_DCHECK(thread_);
+  LOGI() << "OnInterruptionEnd";
   thread_->Post(RTC_FROM_HERE, this, kMessageTypeInterruptionEnd);
 }
 
 void AudioDeviceIOS::OnValidRouteChange() {
   RTC_DCHECK(thread_);
   thread_->Post(RTC_FROM_HERE, this, kMessageTypeValidRouteChange);
 }
 
 void AudioDeviceIOS::OnCanPlayOrRecordChange(bool can_play_or_record) {
   RTC_DCHECK(thread_);
@@ skipping 117 matching lines @@
       HandleCanPlayOrRecordChange(data->data());
       delete data;
       break;
     }
   }
 }
 
 void AudioDeviceIOS::HandleInterruptionBegin() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 
+  RTCLog(@"Interruption begin. IsInterrupted changed from %d to 1.",
+         is_interrupted_);
   if (audio_unit_ &&
       audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
     RTCLog(@"Stopping the audio unit due to interruption begin.");
     if (!audio_unit_->Stop()) {
       RTCLogError(@"Failed to stop the audio unit for interruption begin.");
     }
   }
   is_interrupted_ = true;
 }
 
 void AudioDeviceIOS::HandleInterruptionEnd() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 
+  RTCLog(@"Interruption ended. IsInterrupted changed from %d to 0. "
+         "Updating audio unit state.", is_interrupted_);
   is_interrupted_ = false;
-  RTCLog(@"Interruption ended. Updating audio unit state.");
   UpdateAudioUnit([RTCAudioSession sharedInstance].canPlayOrRecord);
 }
 
 void AudioDeviceIOS::HandleValidRouteChange() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 
   RTCAudioSession* session = [RTCAudioSession sharedInstance];
   RTCLog(@"%@", session);
   HandleSampleRateChange(session.sampleRate);
 }
@@ skipping 330 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