Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index 4d758d99a62662e25fccafa1e496a1fdd621fe6b..936b4590f477a7b55a0ace2daf67dc1173b79348 100644 |
--- a/webrtc/call/call.cc |
+++ b/webrtc/call/call.cc |
@@ -655,6 +655,7 @@ void Call::ConfigureSync(const std::string& sync_group) { |
PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type, |
const uint8_t* packet, |
size_t length) { |
+ TRACE_EVENT0("webrtc", "Call::DeliverRtcp"); |
// TODO(pbos): Figure out what channel needs it actually. |
// Do NOT broadcast! Also make sure it's a valid packet. |
// Return DELIVERY_UNKNOWN_SSRC if it can be determined that |
@@ -688,6 +689,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, |
const uint8_t* packet, |
size_t length, |
const PacketTime& packet_time) { |
+ TRACE_EVENT0("webrtc", "Call::DeliverRtp"); |
// Minimum RTP header size. |
if (length < 12) |
return DELIVERY_PACKET_ERROR; |