Removes analyzer dependency on legacy congestion controller.
Bug: webrtc:9586
Change-Id: Ic1f2445d6432202aeba9164acd49b75261e91aa0
Reviewed-on: https://webrtc-review.googlesource.com/c/105107
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25183}diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index 56e3a3a..c0716b4 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -255,10 +255,13 @@
# TODO(kwiberg): Remove this dependency.
"../api/audio_codecs:audio_codecs_api",
+ "../api/transport:goog_cc",
"../modules/congestion_controller",
"../modules/congestion_controller/goog_cc:delay_based_bwe",
"../modules/congestion_controller/goog_cc:estimators",
+ "../modules/congestion_controller/rtp:transport_feedback",
"../modules/pacing",
+ "../modules/remote_bitrate_estimator",
"../modules/rtp_rtcp",
"//third_party/abseil-cpp/absl/memory",
]
diff --git a/rtc_tools/DEPS b/rtc_tools/DEPS
index d9b162c..4015ae2 100644
--- a/rtc_tools/DEPS
+++ b/rtc_tools/DEPS
@@ -7,6 +7,7 @@
"+modules/audio_coding/neteq/tools",
"+modules/audio_processing",
"+modules/bitrate_controller",
+ "+modules/remote_bitrate_estimator",
"+modules/congestion_controller",
"+modules/pacing",
"+modules/rtp_rtcp",
diff --git a/rtc_tools/event_log_visualizer/analyzer.cc b/rtc_tools/event_log_visualizer/analyzer.cc
index 43cbd69..2cc86c2 100644
--- a/rtc_tools/event_log_visualizer/analyzer.cc
+++ b/rtc_tools/event_log_visualizer/analyzer.cc
@@ -18,6 +18,7 @@
#include <utility>
#include "absl/memory/memory.h"
+#include "api/transport/goog_cc_factory.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "call/call.h"
@@ -36,8 +37,9 @@
#include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
-#include "modules/congestion_controller/include/send_side_congestion_controller.h"
+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
#include "modules/pacing/packet_router.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet.h"
@@ -486,17 +488,15 @@
<< " (LOG_START, LOG_END) segments in log.";
}
-class BitrateObserver : public NetworkChangedObserver,
- public RemoteBitrateObserver {
+class BitrateObserver : public RemoteBitrateObserver {
public:
BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}
- void OnNetworkChanged(uint32_t bitrate_bps,
- uint8_t fraction_lost,
- int64_t rtt_ms,
- int64_t probing_interval_ms) override {
- last_bitrate_bps_ = bitrate_bps;
- bitrate_updated_ = true;
+ void Update(NetworkControlUpdate update) {
+ if (update.target_rate) {
+ last_bitrate_bps_ = update.target_rate->target_rate.bps();
+ bitrate_updated_ = true;
+ }
}
void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
@@ -1081,10 +1081,14 @@
RtcEventLogNullImpl null_event_log;
PacketRouter packet_router;
PacedSender pacer(&clock, &packet_router, &null_event_log);
- SendSideCongestionController cc(&clock, &observer, &null_event_log, &pacer);
+ webrtc_cc::TransportFeedbackAdapter transport_feedback(&clock);
+ auto factory = GoogCcNetworkControllerFactory(&null_event_log);
+ TimeDelta process_interval = factory.GetProcessInterval();
// TODO(holmer): Log the call config and use that here instead.
static const uint32_t kDefaultStartBitrateBps = 300000;
- cc.SetBweBitrates(0, kDefaultStartBitrateBps, -1);
+ NetworkControllerConfig cc_config;
+ cc_config.constraints.starting_rate = DataRate::bps(kDefaultStartBitrateBps);
+ auto goog_cc = factory.Create(cc_config);
TimeSeries time_series("Delay-based estimate", LineStyle::kStep,
PointStyle::kHighlight);
@@ -1107,12 +1111,12 @@
return static_cast<int64_t>(rtcp_iterator->log_time_us());
return std::numeric_limits<int64_t>::max();
};
+ int64_t next_process_time_us_ = clock.TimeInMicroseconds();
auto NextProcessTime = [&]() {
if (rtcp_iterator != incoming_rtcp.end() ||
rtp_iterator != outgoing_rtp.end()) {
- return clock.TimeInMicroseconds() +
- std::max<int64_t>(cc.TimeUntilNextProcess() * 1000, 0);
+ return next_process_time_us_;
}
return std::numeric_limits<int64_t>::max();
};
@@ -1129,24 +1133,32 @@
AcknowledgedBitrateEstimator acknowledged_bitrate_estimator(
absl::make_unique<BitrateEstimator>());
#endif // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
- int64_t time_us = std::min(NextRtpTime(), NextRtcpTime());
+ int64_t time_us =
+ std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
int64_t last_update_us = 0;
while (time_us != std::numeric_limits<int64_t>::max()) {
clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
- cc.OnTransportFeedback(rtcp_iterator->transport_feedback);
- std::vector<PacketFeedback> feedback = cc.GetTransportFeedbackVector();
- SortPacketFeedbackVector(&feedback);
+
+ auto feedback_msg = transport_feedback.ProcessTransportFeedback(
+ rtcp_iterator->transport_feedback);
absl::optional<uint32_t> bitrate_bps;
- if (!feedback.empty()) {
+ if (feedback_msg) {
+ observer.Update(goog_cc->OnTransportPacketsFeedback(*feedback_msg));
+ std::vector<PacketFeedback> feedback =
+ transport_feedback.GetTransportFeedbackVector();
+ SortPacketFeedbackVector(&feedback);
+ if (!feedback.empty()) {
#if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
- acknowledged_bitrate_estimator.IncomingPacketFeedbackVector(feedback);
+ acknowledged_bitrate_estimator.IncomingPacketFeedbackVector(feedback);
#endif // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
- for (const PacketFeedback& packet : feedback)
- acked_bitrate.Update(packet.payload_size, packet.arrival_time_ms);
- bitrate_bps = acked_bitrate.Rate(feedback.back().arrival_time_ms);
+ for (const PacketFeedback& packet : feedback)
+ acked_bitrate.Update(packet.payload_size, packet.arrival_time_ms);
+ bitrate_bps = acked_bitrate.Rate(feedback.back().arrival_time_ms);
+ }
}
+
float x = ToCallTimeSec(clock.TimeInMicroseconds());
float y = bitrate_bps.value_or(0) / 1000;
acked_time_series.points.emplace_back(x, y);
@@ -1161,19 +1173,25 @@
const RtpPacketType& rtp_packet = *rtp_iterator->second;
if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
RTC_DCHECK(rtp_packet.rtp.header.extension.hasTransportSequenceNumber);
- cc.AddPacket(rtp_packet.rtp.header.ssrc,
- rtp_packet.rtp.header.extension.transportSequenceNumber,
- rtp_packet.rtp.total_length, PacedPacketInfo());
+ transport_feedback.AddPacket(
+ rtp_packet.rtp.header.ssrc,
+ rtp_packet.rtp.header.extension.transportSequenceNumber,
+ rtp_packet.rtp.total_length, PacedPacketInfo());
rtc::SentPacket sent_packet(
rtp_packet.rtp.header.extension.transportSequenceNumber,
rtp_packet.rtp.log_time_us() / 1000);
- cc.OnSentPacket(sent_packet);
+ auto sent_msg = transport_feedback.ProcessSentPacket(sent_packet);
+ if (sent_msg)
+ observer.Update(goog_cc->OnSentPacket(*sent_msg));
}
++rtp_iterator;
}
if (clock.TimeInMicroseconds() >= NextProcessTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
- cc.Process();
+ ProcessInterval msg;
+ msg.at_time = Timestamp::us(clock.TimeInMicroseconds());
+ observer.Update(goog_cc->OnProcessInterval(msg));
+ next_process_time_us_ += process_interval.us();
}
if (observer.GetAndResetBitrateUpdated() ||
time_us - last_update_us >= 1e6) {