Add performance tracing for PlatformThread and parts of the video code.

webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc

 /*
  *  Copyright (c) 2015 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/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
 
 namespace webrtc {
 namespace rtcp {
 namespace {
 // Header size:
 // * 4 bytes Common RTCP Packet Header
 // * 8 bytes Common Packet Format for RTCP Feedback Messages
@@ skipping 380 matching lines @@
   std::vector<int64_t> us_deltas;
   for (const auto& packet : packets_)
     us_deltas.push_back(packet.delta_us());
   return us_deltas;
 }
 
 // De-serialize packet.
 bool TransportFeedback::Parse(const CommonHeader& packet) {
   RTC_DCHECK_EQ(packet.type(), kPacketType);
   RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);
+  TRACE_EVENT0("webrtc", "TransportFeedback::Parse");
 
   if (packet.payload_size_bytes() < kMinPayloadSizeBytes) {
     LOG(LS_WARNING) << "Buffer too small (" << packet.payload_size_bytes()
                     << " bytes) to fit a "
                        "FeedbackPacket. Minimum size = "
                     << kMinPayloadSizeBytes;
     return false;
   }
 
   const uint8_t* const payload = packet.payload();
@@ skipping 236 matching lines @@
 
   encoded_chunks_.push_back(last_chunk_->Emit());
   size_bytes_ += kChunkSizeBytes;
   last_chunk_->Add(delta_size);
   ++num_seq_no_;
   return true;
 }
 
 }  // namespace rtcp
 }  // namespace webrtc