Experimental patch for delaying mic permission dialog in AppRTCMobile

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 751 matching lines @@
   // the native audio unit buffer size. Use a reasonable capacity to avoid
   // reallocations while audio is played to reduce risk of glitches.
   RTC_DCHECK(audio_device_buffer_);
   const size_t capacity_in_bytes = 2 * playout_parameters_.GetBytesPerBuffer();
   fine_audio_buffer_.reset(new FineAudioBuffer(
       audio_device_buffer_, playout_parameters_.sample_rate(), capacity_in_bytes));
 }
 
 bool AudioDeviceIOS::CreateAudioUnit() {
   RTC_DCHECK(!audio_unit_);
+  RTCLog(@"CreateAudioUnit");
 
   audio_unit_.reset(new VoiceProcessingAudioUnit(this));
   if (!audio_unit_->Init()) {
     audio_unit_.reset();
     return false;
   }
 
   return true;
 }
 
@@ skipping 138 matching lines @@
     RTCLogError(@"Failed to begin WebRTC session: %@",
                 error.localizedDescription);
     return false;
   }
 
   // If we are ready to play or record, initialize the audio unit.
   if (session.canPlayOrRecord) {
     ConfigureAudioSession();
     SetupAudioBuffersForActiveAudioSession();
     audio_unit_->Initialize(playout_parameters_.sample_rate());
+  } else {
+    RTCLogWarning(@"Not ready to start play or record yet.");
   }
 
   // Release the lock.
   [session unlockForConfiguration];
 
   return true;
 }
 
 void AudioDeviceIOS::ShutdownPlayOrRecord() {
   LOGI() << "ShutdownPlayOrRecord";
@@ skipping 15 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