In rtc event log visualizer use modern api of the ReceiveSideCongestionController
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}
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
index a5b8b49..07a7dd4 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
@@ -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 @@
}
}
+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 @@
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;
