| 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 e80ca38d720a2b1d3e7d5627751a43cecb25288c..15b3e821b843f9647f92e0be053d68025964f4b2 100644
|
| --- a/webrtc/tools/event_log_visualizer/analyzer.cc
|
| +++ b/webrtc/tools/event_log_visualizer/analyzer.cc
|
| @@ -24,6 +24,7 @@
|
| #include "webrtc/call/audio_send_stream.h"
|
| #include "webrtc/call/call.h"
|
| #include "webrtc/common_types.h"
|
| +#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
|
| #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
|
| #include "webrtc/modules/include/module_common_types.h"
|
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
|
| @@ -1071,9 +1072,11 @@
|
| RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
|
| const LoggedRtcpPacket& rtcp = *rtcp_iterator->second;
|
| if (rtcp.type == kRtcpTransportFeedback) {
|
| - cc.OnTransportFeedback(
|
| - *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get()));
|
| - std::vector<PacketFeedback> feedback = cc.GetTransportFeedbackVector();
|
| + TransportFeedbackObserver* observer = cc.GetTransportFeedbackObserver();
|
| + observer->OnTransportFeedback(*static_cast<rtcp::TransportFeedback*>(
|
| + rtcp.packet.get()));
|
| + std::vector<PacketFeedback> feedback =
|
| + observer->GetTransportFeedbackVector();
|
| SortPacketFeedbackVector(&feedback);
|
| rtc::Optional<uint32_t> bitrate_bps;
|
| if (!feedback.empty()) {
|
| @@ -1095,8 +1098,9 @@
|
| const LoggedRtpPacket& rtp = *rtp_iterator->second;
|
| if (rtp.header.extension.hasTransportSequenceNumber) {
|
| RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
|
| - cc.AddPacket(rtp.header.extension.transportSequenceNumber,
|
| - rtp.total_length, PacedPacketInfo());
|
| + cc.GetTransportFeedbackObserver()->AddPacket(
|
| + rtp.header.extension.transportSequenceNumber, rtp.total_length,
|
| + PacedPacketInfo());
|
| rtc::SentPacket sent_packet(
|
| rtp.header.extension.transportSequenceNumber, rtp.timestamp / 1000);
|
| cc.OnSentPacket(sent_packet);
|
| @@ -1126,6 +1130,34 @@
|
| plot->SetTitle("Simulated BWE behavior");
|
| }
|
|
|
| +// TODO(holmer): Remove once TransportFeedbackAdapter no longer needs a
|
| +// BitrateController.
|
| +class NullBitrateController : public BitrateController {
|
| + public:
|
| + ~NullBitrateController() override {}
|
| + RtcpBandwidthObserver* CreateRtcpBandwidthObserver() override {
|
| + return nullptr;
|
| + }
|
| + void SetStartBitrate(int start_bitrate_bps) override {}
|
| + void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) override {}
|
| + void SetBitrates(int start_bitrate_bps,
|
| + int min_bitrate_bps,
|
| + int max_bitrate_bps) override {}
|
| + void ResetBitrates(int bitrate_bps,
|
| + int min_bitrate_bps,
|
| + int max_bitrate_bps) override {}
|
| + void OnDelayBasedBweResult(const DelayBasedBwe::Result& result) override {}
|
| + bool AvailableBandwidth(uint32_t* bandwidth) const override { return false; }
|
| + void SetReservedBitrate(uint32_t reserved_bitrate_bps) override {}
|
| + bool GetNetworkParameters(uint32_t* bitrate,
|
| + uint8_t* fraction_loss,
|
| + int64_t* rtt) override {
|
| + return false;
|
| + }
|
| + int64_t TimeUntilNextProcess() override { return 0; }
|
| + void Process() override {}
|
| +};
|
| +
|
| void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) {
|
| std::map<uint64_t, const LoggedRtpPacket*> outgoing_rtp;
|
| std::map<uint64_t, const LoggedRtcpPacket*> incoming_rtcp;
|
| @@ -1146,7 +1178,9 @@
|
| }
|
|
|
| SimulatedClock clock(0);
|
| - TransportFeedbackAdapter feedback_adapter(&clock);
|
| + NullBitrateController null_controller;
|
| + TransportFeedbackAdapter feedback_adapter(nullptr, &clock, &null_controller);
|
| + feedback_adapter.InitBwe();
|
|
|
| TimeSeries time_series;
|
| time_series.label = "Network Delay Change";
|
|
|