Change how event_log_visualizer ignore duplicate incoming RTCP packets.
Instead of only looking at Audio or ANY media type, we ignore duplicate RTCP packets.
BUG=webrtc:7538
Review-Url: https://codereview.webrtc.org/2892913002
Cr-Commit-Position: refs/heads/master@{#18213}
diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc
index 88ca339..5ff0e58 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.cc
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc
@@ -308,6 +308,9 @@
size_t header_length;
size_t total_length;
+ uint8_t last_incoming_rtcp_packet[IP_PACKET_SIZE];
+ uint8_t last_incoming_rtcp_packet_length = 0;
+
// Make a default extension map for streams without configuration information.
// TODO(ivoc): Once configuration of audio streams is stored in the event log,
// this can be removed. Tracking bug: webrtc:6399
@@ -403,46 +406,54 @@
MediaType media_type;
parsed_log_.GetRtcpPacket(i, &direction, &media_type, packet,
&total_length);
-
- // Currently feedback is logged twice, both for audio and video.
- // Only act on one of them.
- if (media_type == MediaType::AUDIO || media_type == MediaType::ANY) {
- rtcp::CommonHeader header;
- const uint8_t* packet_end = packet + total_length;
- for (const uint8_t* block = packet; block < packet_end;
- block = header.NextPacket()) {
- RTC_CHECK(header.Parse(block, packet_end - block));
- if (header.type() == rtcp::TransportFeedback::kPacketType &&
- header.fmt() == rtcp::TransportFeedback::kFeedbackMessageType) {
- std::unique_ptr<rtcp::TransportFeedback> rtcp_packet(
- new rtcp::TransportFeedback());
- if (rtcp_packet->Parse(header)) {
- uint32_t ssrc = rtcp_packet->sender_ssrc();
- StreamId stream(ssrc, direction);
- uint64_t timestamp = parsed_log_.GetTimestamp(i);
- rtcp_packets_[stream].push_back(LoggedRtcpPacket(
- timestamp, kRtcpTransportFeedback, std::move(rtcp_packet)));
- }
- } else if (header.type() == rtcp::SenderReport::kPacketType) {
- std::unique_ptr<rtcp::SenderReport> rtcp_packet(
- new rtcp::SenderReport());
- if (rtcp_packet->Parse(header)) {
- uint32_t ssrc = rtcp_packet->sender_ssrc();
- StreamId stream(ssrc, direction);
- uint64_t timestamp = parsed_log_.GetTimestamp(i);
- rtcp_packets_[stream].push_back(LoggedRtcpPacket(
- timestamp, kRtcpSr, std::move(rtcp_packet)));
- }
- } else if (header.type() == rtcp::ReceiverReport::kPacketType) {
- std::unique_ptr<rtcp::ReceiverReport> rtcp_packet(
- new rtcp::ReceiverReport());
- if (rtcp_packet->Parse(header)) {
- uint32_t ssrc = rtcp_packet->sender_ssrc();
- StreamId stream(ssrc, direction);
- uint64_t timestamp = parsed_log_.GetTimestamp(i);
- rtcp_packets_[stream].push_back(LoggedRtcpPacket(
- timestamp, kRtcpRr, std::move(rtcp_packet)));
- }
+ // Currently incoming RTCP packets are logged twice, both for audio and
+ // video. Only act on one of them. Compare against the previous parsed
+ // incoming RTCP packet.
+ if (direction == webrtc::kIncomingPacket) {
+ RTC_CHECK_LE(total_length, IP_PACKET_SIZE);
+ if (total_length == last_incoming_rtcp_packet_length &&
+ memcmp(last_incoming_rtcp_packet, packet, total_length) == 0) {
+ continue;
+ } else {
+ memcpy(last_incoming_rtcp_packet, packet, total_length);
+ last_incoming_rtcp_packet_length = total_length;
+ }
+ }
+ rtcp::CommonHeader header;
+ const uint8_t* packet_end = packet + total_length;
+ for (const uint8_t* block = packet; block < packet_end;
+ block = header.NextPacket()) {
+ RTC_CHECK(header.Parse(block, packet_end - block));
+ if (header.type() == rtcp::TransportFeedback::kPacketType &&
+ header.fmt() == rtcp::TransportFeedback::kFeedbackMessageType) {
+ std::unique_ptr<rtcp::TransportFeedback> rtcp_packet(
+ new rtcp::TransportFeedback());
+ if (rtcp_packet->Parse(header)) {
+ uint32_t ssrc = rtcp_packet->sender_ssrc();
+ StreamId stream(ssrc, direction);
+ uint64_t timestamp = parsed_log_.GetTimestamp(i);
+ rtcp_packets_[stream].push_back(LoggedRtcpPacket(
+ timestamp, kRtcpTransportFeedback, std::move(rtcp_packet)));
+ }
+ } else if (header.type() == rtcp::SenderReport::kPacketType) {
+ std::unique_ptr<rtcp::SenderReport> rtcp_packet(
+ new rtcp::SenderReport());
+ if (rtcp_packet->Parse(header)) {
+ uint32_t ssrc = rtcp_packet->sender_ssrc();
+ StreamId stream(ssrc, direction);
+ uint64_t timestamp = parsed_log_.GetTimestamp(i);
+ rtcp_packets_[stream].push_back(
+ LoggedRtcpPacket(timestamp, kRtcpSr, std::move(rtcp_packet)));
+ }
+ } else if (header.type() == rtcp::ReceiverReport::kPacketType) {
+ std::unique_ptr<rtcp::ReceiverReport> rtcp_packet(
+ new rtcp::ReceiverReport());
+ if (rtcp_packet->Parse(header)) {
+ uint32_t ssrc = rtcp_packet->sender_ssrc();
+ StreamId stream(ssrc, direction);
+ uint64_t timestamp = parsed_log_.GetTimestamp(i);
+ rtcp_packets_[stream].push_back(
+ LoggedRtcpPacket(timestamp, kRtcpRr, std::move(rtcp_packet)));
}
}
}