Index: webrtc/tools/event_log_visualizer/analyzer.cc |
diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc |
index f068759ec2a276f5b0bfff1302e2d1f0cab79273..b5b05dd218db29b3e4eb202be4a2adfb94a32c53 100644 |
--- a/webrtc/tools/event_log_visualizer/analyzer.cc |
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc |
@@ -19,6 +19,7 @@ |
#include "webrtc/base/checks.h" |
#include "webrtc/base/logging.h" |
+#include "webrtc/base/ptr_util.h" |
#include "webrtc/base/rate_statistics.h" |
#include "webrtc/call/audio_receive_stream.h" |
#include "webrtc/call/audio_send_stream.h" |
@@ -30,6 +31,7 @@ |
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h" |
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" |
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h" |
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" |
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" |
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" |
@@ -420,7 +422,7 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log) |
if (header.type() == rtcp::TransportFeedback::kPacketType && |
header.fmt() == rtcp::TransportFeedback::kFeedbackMessageType) { |
std::unique_ptr<rtcp::TransportFeedback> rtcp_packet( |
- new rtcp::TransportFeedback()); |
+ rtc::MakeUnique<rtcp::TransportFeedback>()); |
if (rtcp_packet->Parse(header)) { |
uint32_t ssrc = rtcp_packet->sender_ssrc(); |
StreamId stream(ssrc, direction); |
@@ -430,7 +432,7 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log) |
} |
} else if (header.type() == rtcp::SenderReport::kPacketType) { |
std::unique_ptr<rtcp::SenderReport> rtcp_packet( |
- new rtcp::SenderReport()); |
+ rtc::MakeUnique<rtcp::SenderReport>()); |
if (rtcp_packet->Parse(header)) { |
uint32_t ssrc = rtcp_packet->sender_ssrc(); |
StreamId stream(ssrc, direction); |
@@ -440,7 +442,7 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log) |
} |
} else if (header.type() == rtcp::ReceiverReport::kPacketType) { |
std::unique_ptr<rtcp::ReceiverReport> rtcp_packet( |
- new rtcp::ReceiverReport()); |
+ rtc::MakeUnique<rtcp::ReceiverReport>()); |
if (rtcp_packet->Parse(header)) { |
uint32_t ssrc = rtcp_packet->sender_ssrc(); |
StreamId stream(ssrc, direction); |
@@ -448,6 +450,17 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log) |
rtcp_packets_[stream].push_back( |
LoggedRtcpPacket(timestamp, kRtcpRr, std::move(rtcp_packet))); |
} |
+ } else if (header.type() == rtcp::Remb::kPacketType && |
+ header.fmt() == rtcp::Remb::kFeedbackMessageType) { |
+ std::unique_ptr<rtcp::Remb> rtcp_packet( |
+ rtc::MakeUnique<rtcp::Remb>()); |
+ 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, kRtcpRemb, std::move(rtcp_packet))); |
+ } |
} |
} |
break; |
@@ -975,6 +988,32 @@ void EventLogAnalyzer::CreateTotalBitrateGraph( |
plot->AppendTimeSeries(std::move(result_series)); |
} |
+ // Overlay the incoming REMB over the outgoing bitrate |
+ // and outgoing REMB over incoming bitrate. |
+ PacketDirection remb_direction = |
+ desired_direction == kOutgoingPacket ? kIncomingPacket : kOutgoingPacket; |
+ TimeSeries remb_series("Remb", LINE_STEP_GRAPH); |
+ std::multimap<uint64_t, const LoggedRtcpPacket*> remb_packets; |
+ for (const auto& kv : rtcp_packets_) { |
+ if (kv.first.GetDirection() == remb_direction) { |
+ for (const LoggedRtcpPacket& rtcp_packet : kv.second) { |
+ if (rtcp_packet.type == kRtcpRemb) { |
+ remb_packets.insert( |
+ std::make_pair(rtcp_packet.timestamp, &rtcp_packet)); |
+ } |
+ } |
+ } |
+ } |
+ |
+ for (const auto& kv : remb_packets) { |
+ const LoggedRtcpPacket* const rtcp = kv.second; |
+ const rtcp::Remb* const remb = static_cast<rtcp::Remb*>(rtcp->packet.get()); |
+ float x = static_cast<float>(rtcp->timestamp - begin_time_) / 1000000; |
+ float y = static_cast<float>(remb->bitrate_bps()) / 1000; |
+ remb_series.points.emplace_back(x, y); |
+ } |
+ plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series)); |
+ |
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin); |
plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin); |
if (desired_direction == webrtc::PacketDirection::kIncomingPacket) { |