Implement standalone event tracing in AppRTCDemo.

talk/session/media/channel.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 11 matching lines @@
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #include "talk/session/media/channel.h"
 
 #include "talk/media/base/constants.h"
 #include "talk/media/base/rtputils.h"
-#include "webrtc/p2p/base/transportchannel.h"
 #include "talk/session/media/channelmanager.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/p2p/base/transportchannel.h"
 
 namespace cricket {
 
 using rtc::Bind;
 
 enum {
   MSG_EARLYMEDIATIMEOUT = 1,
   MSG_SCREENCASTWINDOWEVENT,
   MSG_RTPPACKET,
   MSG_RTCPPACKET,
@@ skipping 414 matching lines @@
 
 void BaseChannel::OnWritableState(TransportChannel* channel) {
   ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
   UpdateWritableState_w();
 }
 
 void BaseChannel::OnChannelRead(TransportChannel* channel,
                                 const char* data, size_t len,
                                 const rtc::PacketTime& packet_time,
                                 int flags) {
+  TRACE_EVENT0("webrtc", "BaseChannel::OnChannelRead");
   // OnChannelRead gets called from P2PSocket; now pass data to MediaEngine
   ASSERT(worker_thread_ == rtc::Thread::Current());
 
   // When using RTCP multiplexing we might get RTCP packets on the RTP
   // transport. We feed RTP traffic into the demuxer to determine if it is RTCP.
   bool rtcp = PacketIsRtcp(channel, data, len);
   rtc::Buffer packet(data, len);
   HandlePacket(rtcp, &packet, packet_time);
 }
 
@@ skipping 781 matching lines @@
 
 void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension(
     const std::vector<RtpHeaderExtension>& extensions) {
   const RtpHeaderExtension* send_time_extension =
       FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
   rtp_abs_sendtime_extn_id_ =
       send_time_extension ? send_time_extension->id : -1;
 }
 
 void BaseChannel::OnMessage(rtc::Message *pmsg) {
+  TRACE_EVENT0("webrtc", "BaseChannel::OnMessage");
   switch (pmsg->message_id) {
     case MSG_RTPPACKET:
     case MSG_RTCPPACKET: {
       PacketMessageData* data = static_cast<PacketMessageData*>(pmsg->pdata);
       SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet,
                  data->options);
       delete data;  // because it is Posted
       break;
     }
     case MSG_FIRSTPACKETRECEIVED: {
@@ skipping 1017 matching lines @@
   return (data_channel_type_ == DCT_RTP) && BaseChannel::ShouldSetupDtlsSrtp();
 }
 
 void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
   rtc::TypedMessageData<uint32_t>* message =
       new rtc::TypedMessageData<uint32_t>(sid);
   signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message);
 }
 
 }  // namespace cricket