Implement standalone event tracing in AppRTCDemo.

webrtc/modules/video_coding/jitter_buffer.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.
  */
 #include "webrtc/modules/video_coding/jitter_buffer.h"
@@ skipping 1166 matching lines @@
   bool key_frame_found = false;
   int dropped_frames = 0;
   dropped_frames += incomplete_frames_.RecycleFramesUntilKeyFrame(
       &key_frame_it, &free_frames_);
   key_frame_found = key_frame_it != incomplete_frames_.end();
   if (dropped_frames == 0) {
     dropped_frames += decodable_frames_.RecycleFramesUntilKeyFrame(
         &key_frame_it, &free_frames_);
     key_frame_found = key_frame_it != decodable_frames_.end();
   }
-  TRACE_EVENT_INSTANT0("webrtc", "JB::RecycleFramesUntilKeyFrame");
   if (key_frame_found) {
     LOG(LS_INFO) << "Found key frame while dropping frames.";
     // Reset last decoded state to make sure the next frame decoded is a key
     // frame, and start NACKing from here.
     last_decoded_state_.Reset();
     DropPacketsFromNackList(EstimatedLowSequenceNumber(*key_frame_it->second));
   } else if (decodable_frames_.empty()) {
     // All frames dropped. Reset the decoding state and clear missing sequence
     // numbers as we're starting fresh.
     last_decoded_state_.Reset();
@@ skipping 115 matching lines @@
   }
   // Evaluate if the RTT is higher than |high_rtt_nack_threshold_ms_|, and in
   // that case we don't wait for retransmissions.
   if (high_rtt_nack_threshold_ms_ >= 0 &&
       rtt_ms_ >= high_rtt_nack_threshold_ms_) {
     return false;
   }
   return true;
 }
 }  // namespace webrtc