Index: webrtc/call/rtc_event_log_unittest.cc |
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc |
index 0998071827d93a00b5591394a6fb1f23e5a85f21..70032039303129242ea52cfb4509adc361993c80 100644 |
--- a/webrtc/call/rtc_event_log_unittest.cc |
+++ b/webrtc/call/rtc_event_log_unittest.cc |
@@ -314,7 +314,8 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector, |
nullptr, // SendTimeObserver* |
nullptr, // BitrateStatisticsObserver* |
nullptr, // FrameCountObserver* |
- nullptr); // SendSideDelayObserver* |
+ nullptr, // SendSideDelayObserver* |
+ nullptr); // RtcEventLog* |
std::vector<uint32_t> csrcs; |
for (unsigned i = 0; i < csrcs_count; i++) { |
@@ -480,12 +481,12 @@ void LogSessionAndReadBack(size_t rtp_count, |
size_t bwe_loss_index = 1; |
for (size_t i = 1; i <= rtp_count; i++) { |
log_dumper->LogRtpHeader( |
- (i % 2 == 0), // Every second packet is incoming. |
+ (i % 2 == 0) ? kIncomingPacket : kOutgoingPacket, |
(i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, |
rtp_packets[i - 1].data(), rtp_packets[i - 1].size()); |
if (i * rtcp_count >= rtcp_index * rtp_count) { |
log_dumper->LogRtcpPacket( |
- rtcp_index % 2 == 0, // Every second packet is incoming |
+ (rtcp_index % 2 == 0) ? kIncomingPacket : kOutgoingPacket, |
rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO, |
rtcp_packets[rtcp_index - 1]->Buffer(), |
rtcp_packets[rtcp_index - 1]->Length()); |
@@ -643,16 +644,18 @@ void DropOldEvents(uint32_t extensions_bitvector, |
log_dumper->SetBufferDuration(50000); |
log_dumper->LogVideoReceiveStreamConfig(receiver_config); |
log_dumper->LogVideoSendStreamConfig(sender_config); |
- log_dumper->LogRtpHeader(false, MediaType::AUDIO, old_rtp_packet.data(), |
- old_rtp_packet.size()); |
- log_dumper->LogRtcpPacket(true, MediaType::AUDIO, old_rtcp_packet->Buffer(), |
+ log_dumper->LogRtpHeader(kOutgoingPacket, MediaType::AUDIO, |
+ old_rtp_packet.data(), old_rtp_packet.size()); |
+ log_dumper->LogRtcpPacket(kIncomingPacket, MediaType::AUDIO, |
+ old_rtcp_packet->Buffer(), |
old_rtcp_packet->Length()); |
// Sleep 55 ms to let old events be removed from the queue. |
rtc::Thread::SleepMs(55); |
log_dumper->StartLogging(temp_filename, 10000000); |
- log_dumper->LogRtpHeader(true, MediaType::VIDEO, recent_rtp_packet.data(), |
+ log_dumper->LogRtpHeader(kIncomingPacket, MediaType::VIDEO, |
+ recent_rtp_packet.data(), |
recent_rtp_packet.size()); |
- log_dumper->LogRtcpPacket(false, MediaType::VIDEO, |
+ log_dumper->LogRtcpPacket(kOutgoingPacket, MediaType::VIDEO, |
recent_rtcp_packet->Buffer(), |
recent_rtcp_packet->Length()); |
} |