Adding time profiling support to AudioFrame

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 746db8114c600ab0d7afa7c0e2b4007d78b4b8da..d2790199fdd39c6e08fb36ca926e2ee914aac13a 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -2725,6 +2725,10 @@ void Channel::ProcessAndEncodeAudio(const AudioFrame& audio_input) {
   // either into pool of frames or into the task itself.
   audio_frame->CopyFrom(audio_input);
   audio_frame->id_ = ChannelId();
+  // Profile time between when the audio frame is added to the task queue (t0)
+  // and when the task is actually executed (t1). Goal is to keep track of
+  // unwanted extra latency added by the queue.
+  audio_frame->UpdateProfileTime();
   encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
       new ProcessAndEncodeAudioTask(std::move(audio_frame), this)));
 }
@@ -2764,6 +2768,10 @@ void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) {
   RTC_DCHECK_LE(audio_input->num_channels_, 2);
   RTC_DCHECK_EQ(audio_input->id_, ChannelId());
 
+  // TODO(henrika): add comment and histogram here...
+  // dT = t1 - t0.
+  int64_t dT = audio_input->TimeSinceLastProfile();
+
   if (channel_state_.Get().input_file_playing) {
     MixOrReplaceAudioWithFile(audio_input);
   }