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In rtc event log visualizer use modern api of the ReceiveSideCongestionController

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Bug: webrtc:14859
Change-Id: I121905a892aaafb54c1fff62f818bfd767b92da0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296823
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39755}
This commit is contained in:
Danil Chapovalov 2023-03-30 14:17:19 +02:00 committed by WebRTC LUCI CQ
parent 40a0e3191a
commit 4b61f3a0c6
1 changed files with 77 additions and 6 deletions
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83
rtc_tools/rtc_event_log_visualizer/analyzer.cc
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@ -46,6 +46,7 @@
#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
@ -339,6 +340,72 @@ std::string GetDirectionAsShortString(PacketDirection direction) {
}
}
struct FakeExtensionSmall {
static constexpr RTPExtensionType kId = kRtpExtensionMid;
static constexpr absl::string_view Uri() { return "fake-extension-small"; }
};
struct FakeExtensionLarge {
static constexpr RTPExtensionType kId = kRtpExtensionRtpStreamId;
static constexpr absl::string_view Uri() { return "fake-extension-large"; }
};
RtpPacketReceived RtpPacketForBWEFromHeader(const RTPHeader& header) {
RtpHeaderExtensionMap rtp_header_extensions(/*extmap_allow_mixed=*/true);
// ReceiveSideCongestionController doesn't need to know extensions ids as
// long as it able to get extensions by type. So any ids would work here.
rtp_header_extensions.Register<TransmissionOffset>(1);
rtp_header_extensions.Register<AbsoluteSendTime>(2);
rtp_header_extensions.Register<TransportSequenceNumber>(3);
rtp_header_extensions.Register<FakeExtensionSmall>(4);
// Use id > 14 to force two byte header per rtp header when this one is used.
rtp_header_extensions.Register<FakeExtensionLarge>(16);
RtpPacketReceived rtp_packet(&rtp_header_extensions);
// Set only fields that might be relevant for the bandwidth estimatior.
rtp_packet.SetSsrc(header.ssrc);
rtp_packet.SetTimestamp(header.timestamp);
size_t num_bwe_extensions = 0;
if (header.extension.hasTransmissionTimeOffset) {
rtp_packet.SetExtension<TransmissionOffset>(
header.extension.transmissionTimeOffset);
++num_bwe_extensions;
}
if (header.extension.hasAbsoluteSendTime) {
rtp_packet.SetExtension<AbsoluteSendTime>(
header.extension.absoluteSendTime);
++num_bwe_extensions;
}
if (header.extension.hasTransportSequenceNumber) {
rtp_packet.SetExtension<TransportSequenceNumber>(
header.extension.transportSequenceNumber);
++num_bwe_extensions;
}
// All parts of the RTP header are 32bit aligned.
RTC_CHECK_EQ(header.headerLength % 4, 0);
// Original packet could have more extensions, there could be csrcs that are
// not propagated by the rtc event log, i.e. logged header size might be
// larger that rtp_packet.header_size(). Increase it by setting an extra fake
// extension.
RTC_CHECK_GE(header.headerLength, rtp_packet.headers_size());
size_t bytes_to_add = header.headerLength - rtp_packet.headers_size();
if (bytes_to_add > 0) {
if (bytes_to_add <= 16) {
// one-byte header rtp header extension allows to add up to 16 bytes.
rtp_packet.AllocateExtension(FakeExtensionSmall::kId, bytes_to_add - 1);
} else {
// two-byte header rtp header extension would also add one byte per
// already set extension.
rtp_packet.AllocateExtension(FakeExtensionLarge::kId,
bytes_to_add - 2 - num_bwe_extensions);
}
}
RTC_CHECK_EQ(rtp_packet.headers_size(), header.headerLength);
return rtp_packet;
}
} // namespace
EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
@ -1468,12 +1535,16 @@ void EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(Plot* plot) {
int64_t last_update_us = 0;
for (const auto& kv : incoming_rtp) {
const RtpPacketType& packet = *kv.second;
int64_t arrival_time_ms = packet.rtp.log_time_us() / 1000;
size_t payload = packet.rtp.total_length; /*Should subtract header?*/
clock.AdvanceTimeMicroseconds(packet.rtp.log_time_us() -
clock.TimeInMicroseconds());
rscc.OnReceivedPacket(arrival_time_ms, payload, packet.rtp.header);
acked_bitrate.Update(payload, arrival_time_ms);
RtpPacketReceived rtp_packet = RtpPacketForBWEFromHeader(packet.rtp.header);
rtp_packet.set_arrival_time(packet.rtp.log_time());
rtp_packet.SetPayloadSize(packet.rtp.total_length -
rtp_packet.headers_size());
clock.AdvanceTime(rtp_packet.arrival_time() - clock.CurrentTime());
rscc.OnReceivedPacket(rtp_packet, MediaType::VIDEO);
int64_t arrival_time_ms = packet.rtp.log_time().ms();
acked_bitrate.Update(packet.rtp.total_length, arrival_time_ms);
absl::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
if (bitrate_bps) {
uint32_t y = *bitrate_bps / 1000;