Adding time profiling support to AudioFrame

webrtc/voice_engine/channel.cc

 /*
  *  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 2707 matching lines @@
   // Avoid posting any new tasks if sending was already stopped in StopSend().
   rtc::CritScope cs(&encoder_queue_lock_);
   if (!encoder_queue_is_active_) {
     return;
   }
   std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
   // TODO(henrika): try to avoid copying by moving ownership of audio frame
   // 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)));
 }
 
 void Channel::ProcessAndEncodeAudio(const int16_t* audio_data,
                                     int sample_rate,
                                     size_t number_of_frames,
                                     size_t number_of_channels) {
   // Avoid posting as new task if sending was already stopped in StopSend().
   rtc::CritScope cs(&encoder_queue_lock_);
@@ skipping 19 matching lines @@
   encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
       new ProcessAndEncodeAudioTask(std::move(audio_frame), this)));
 }
 
 void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) {
   RTC_DCHECK_RUN_ON(encoder_queue_);
   RTC_DCHECK_GT(audio_input->samples_per_channel_, 0);
   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);
   }
 
   bool is_muted = InputMute();
   AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted);
 
   if (_includeAudioLevelIndication) {
     size_t length =
         audio_input->samples_per_channel_ * audio_input->num_channels_;
@@ skipping 355 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc