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Unified Diff: webrtc/call/call.cc

Issue 1457383002: Implement standalone event tracing in AppRTCDemo. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: readd jitter_buffer trace, rebase Created 5 years ago
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Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 9209c7c97a06ee96b014087a0e42facb3b5e1495..f6efa6d46a2d331f6b56b6bc2691025cc20e86ec 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;
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