Chromium Code Reviews| 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 2a23c94a7dbe325e5d66d96bc334e756db804a83..9bc0dddae6f2775346531a8cd11eb2200fcf415a 100644 |
| --- a/webrtc/tools/event_log_visualizer/analyzer.cc |
| +++ b/webrtc/tools/event_log_visualizer/analyzer.cc |
| @@ -21,6 +21,7 @@ |
| #include "webrtc/api/call/audio_send_stream.h" |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/logging.h" |
| +#include "webrtc/base/rate_statistics.h" |
| #include "webrtc/call.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/congestion_controller/include/congestion_controller.h" |
| @@ -788,6 +789,9 @@ void EventLogAnalyzer::CreateBweSimulationGraph(Plot* plot) { |
| TimeSeries time_series; |
| time_series.label = "Delay-based estimate"; |
| time_series.style = LINE_DOT_GRAPH; |
| + TimeSeries acked_time_series; |
| + acked_time_series.label = "Acked bitrate"; |
| + acked_time_series.style = LINE_DOT_GRAPH; |
| auto rtp_iterator = outgoing_rtp.begin(); |
| auto rtcp_iterator = incoming_rtcp.begin(); |
| @@ -813,6 +817,8 @@ void EventLogAnalyzer::CreateBweSimulationGraph(Plot* plot) { |
| return std::numeric_limits<int64_t>::max(); |
| }; |
| + RateStatistics acked_bitrate(1000, 8000); |
| + |
| int64_t time_us = std::min(NextRtpTime(), NextRtcpTime()); |
| while (time_us != std::numeric_limits<int64_t>::max()) { |
| clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds()); |
| @@ -820,8 +826,21 @@ void EventLogAnalyzer::CreateBweSimulationGraph(Plot* plot) { |
| RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime()); |
| const LoggedRtcpPacket& rtcp = *rtcp_iterator->second; |
| if (rtcp.type == kRtcpTransportFeedback) { |
| - cc.GetTransportFeedbackObserver()->OnTransportFeedback( |
| - *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get())); |
| + std::vector<PacketInfo> feedback = |
| + cc.GetTransportFeedbackObserver()->OnTransportFeedback( |
|
philipel
2016/09/05 13:30:05
I think it's cleaner if you call both TransportFee
stefan-webrtc
2016/09/05 14:03:39
Done.
philipel
2016/09/05 14:56:25
Maybe something even cleaner would be to add a Get
stefan-webrtc
2016/09/06 11:53:11
I want to get rid of the GetBitrateEstimator() api
|
| + *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get())); |
| + rtc::Optional<uint32_t> bitrate_bps; |
| + if (!feedback.empty()) { |
| + for (const PacketInfo& packet : feedback) |
| + acked_bitrate.Update(packet.payload_size, packet.arrival_time_ms); |
| + bitrate_bps = acked_bitrate.Rate(feedback.back().arrival_time_ms); |
| + } |
| + uint32_t y = 0; |
| + if (bitrate_bps) |
| + y = *bitrate_bps / 1000; |
| + float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) / |
| + 1000000; |
| + acked_time_series.points.emplace_back(x, y); |
| } |
| ++rtcp_iterator; |
| } |
| @@ -853,6 +872,7 @@ void EventLogAnalyzer::CreateBweSimulationGraph(Plot* plot) { |
| } |
| // Add the data set to the plot. |
| plot->series_list_.push_back(std::move(time_series)); |
| + plot->series_list_.push_back(std::move(acked_time_series)); |
| plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin); |
| @@ -909,7 +929,7 @@ void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) { |
| const LoggedRtcpPacket& rtcp = *rtcp_iterator->second; |
| if (rtcp.type == kRtcpTransportFeedback) { |
| std::vector<PacketInfo> feedback = |
| - feedback_adapter.GetPacketFeedbackVector( |
| + feedback_adapter.OnTransportFeedback( |
| *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get())); |
| for (const PacketInfo& packet : feedback) { |
| int64_t y = packet.arrival_time_ms - packet.send_time_ms; |