| Index: webrtc/call/call.cc
|
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
|
| index 326c1bad3e25c9538ab497eb3358b4c807cdde06..dfec9d58a2eb42e331f63b723f4730abdeaa4212 100644
|
| --- a/webrtc/call/call.cc
|
| +++ b/webrtc/call/call.cc
|
| @@ -668,6 +668,7 @@ 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
|
| @@ -701,6 +702,7 @@ 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;
|
|
|