Adding more detail to MessageQueue::Dispatch logging.

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 318 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_);
-  thread_->Post(this, kMessageTypeInterruptionBegin);
+  thread_->Post(RTC_FROM_HERE, this, kMessageTypeInterruptionBegin);
 }
 
 void AudioDeviceIOS::OnInterruptionEnd() {
   RTC_DCHECK(thread_);
-  thread_->Post(this, kMessageTypeInterruptionEnd);
+  thread_->Post(RTC_FROM_HERE, this, kMessageTypeInterruptionEnd);
 }
 
 void AudioDeviceIOS::OnValidRouteChange() {
   RTC_DCHECK(thread_);
-  thread_->Post(this, kMessageTypeValidRouteChange);
+  thread_->Post(RTC_FROM_HERE, this, kMessageTypeValidRouteChange);
 }
 
 void AudioDeviceIOS::OnCanPlayOrRecordChange(bool can_play_or_record) {
   RTC_DCHECK(thread_);
-  thread_->Post(this, kMessageTypeCanPlayOrRecordChange,
+  thread_->Post(RTC_FROM_HERE, this, kMessageTypeCanPlayOrRecordChange,
                 new rtc::TypedMessageData<bool>(can_play_or_record));
 }
 
 OSStatus AudioDeviceIOS::OnDeliverRecordedData(
     AudioUnitRenderActionFlags* flags,
     const AudioTimeStamp* time_stamp,
     UInt32 bus_number,
     UInt32 num_frames,
     AudioBufferList* /* io_data */) {
   OSStatus result = noErr;
@@ skipping 452 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