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Add time line for acked bitrate.

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R=philipel@webrtc.org

Review URL: https://codereview.webrtc.org/2310943002 .

Cr-Commit-Position: refs/heads/master@{#14098}
This commit is contained in:
Stefan Holmer
2016-09-07 09:58:20 +02:00
parent b9d8d10d42
commit 60e4346955
7 changed files with 52 additions and 13 deletions
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31
webrtc/tools/event_log_visualizer/analyzer.cc
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@ -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"
@ -835,6 +836,9 @@ 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();
@ -860,6 +864,8 @@ 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());
@ -867,8 +873,23 @@ 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()));
TransportFeedbackObserver* observer = cc.GetTransportFeedbackObserver();
observer->OnTransportFeedback(*static_cast<rtcp::TransportFeedback*>(
rtcp.packet.get()));
std::vector<PacketInfo> feedback =
observer->GetTransportFeedbackVector();
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;
}
@ -900,6 +921,7 @@ 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);
@ -955,9 +977,10 @@ void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
const LoggedRtcpPacket& rtcp = *rtcp_iterator->second;
if (rtcp.type == kRtcpTransportFeedback) {
feedback_adapter.OnTransportFeedback(
*static_cast<rtcp::TransportFeedback*>(rtcp.packet.get()));
std::vector<PacketInfo> feedback =
feedback_adapter.GetPacketFeedbackVector(
*static_cast<rtcp::TransportFeedback*>(rtcp.packet.get()));
feedback_adapter.GetTransportFeedbackVector();
for (const PacketInfo& packet : feedback) {
int64_t y = packet.arrival_time_ms - packet.send_time_ms;
float x =