| /* |
| * Copyright (c) 2026 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "rtc_tools/rtc_event_log_visualizer/analyze_bwe.h" |
| |
| #include <algorithm> |
| #include <cmath> |
| #include <cstddef> |
| #include <cstdint> |
| #include <limits> |
| #include <map> |
| #include <memory> |
| #include <optional> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/functional/bind_front.h" |
| #include "absl/strings/string_view.h" |
| #include "api/environment/environment.h" |
| #include "api/environment/environment_factory.h" |
| #include "api/field_trials.h" |
| #include "api/function_view.h" |
| #include "api/media_types.h" |
| #include "api/rtp_headers.h" |
| #include "api/transport/bandwidth_usage.h" |
| #include "api/transport/goog_cc_factory.h" |
| #include "api/transport/network_control.h" |
| #include "api/transport/network_types.h" |
| #include "api/units/data_rate.h" |
| #include "api/units/time_delta.h" |
| #include "api/units/timestamp.h" |
| #include "logging/rtc_event_log/events/logged_rtp_rtcp.h" |
| #include "logging/rtc_event_log/rtc_event_log_parser.h" |
| #include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h" |
| #include "modules/congestion_controller/include/receive_side_congestion_controller.h" |
| #include "modules/congestion_controller/rtp/transport_feedback_adapter.h" |
| #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
| #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/remb.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_packet_to_send.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/network/sent_packet.h" |
| #include "rtc_base/numerics/sequence_number_unwrapper.h" |
| #include "rtc_base/rate_statistics.h" |
| #include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h" |
| #include "rtc_tools/rtc_event_log_visualizer/log_scream_simulation.h" |
| #include "rtc_tools/rtc_event_log_visualizer/log_simulation.h" |
| #include "rtc_tools/rtc_event_log_visualizer/plot_base.h" |
| #include "system_wrappers/include/clock.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| double AbsSendTimeToMicroseconds(int64_t abs_send_time) { |
| // The timestamp is a fixed point representation with 6 bits for seconds |
| // and 18 bits for fractions of a second. Thus, we divide by 2^18 to get the |
| // time in seconds and then multiply by kNumMicrosecsPerSec to convert to |
| // microseconds. |
| static constexpr double kTimestampToMicroSec = |
| static_cast<double>(kNumMicrosecsPerSec) / static_cast<double>(1ul << 18); |
| return abs_send_time * kTimestampToMicroSec; |
| } |
| |
| // This is much more reliable for outgoing streams than for incoming streams. |
| template <typename RtpPacketContainer> |
| std::optional<uint32_t> EstimateRtpClockFrequency( |
| const RtpPacketContainer& packets, |
| int64_t end_time_us) { |
| RTC_CHECK(packets.size() >= 2); |
| SeqNumUnwrapper<uint32_t> unwrapper; |
| int64_t first_rtp_timestamp = |
| unwrapper.Unwrap(packets[0].rtp.header.timestamp); |
| int64_t first_log_timestamp = packets[0].log_time_us(); |
| int64_t last_rtp_timestamp = first_rtp_timestamp; |
| int64_t last_log_timestamp = first_log_timestamp; |
| for (size_t i = 1; i < packets.size(); i++) { |
| if (packets[i].log_time_us() > end_time_us) |
| break; |
| last_rtp_timestamp = unwrapper.Unwrap(packets[i].rtp.header.timestamp); |
| last_log_timestamp = packets[i].log_time_us(); |
| } |
| if (last_log_timestamp - first_log_timestamp < kNumMicrosecsPerSec) { |
| RTC_LOG(LS_WARNING) |
| << "Failed to estimate RTP clock frequency: Stream too short. (" |
| << packets.size() << " packets, " |
| << last_log_timestamp - first_log_timestamp << " us)"; |
| return std::nullopt; |
| } |
| double duration = |
| static_cast<double>(last_log_timestamp - first_log_timestamp) / |
| kNumMicrosecsPerSec; |
| double estimated_frequency = |
| (last_rtp_timestamp - first_rtp_timestamp) / duration; |
| for (uint32_t f : {8000, 16000, 32000, 48000, 90000}) { |
| if (std::fabs(estimated_frequency - f) < 0.15 * f) { |
| return f; |
| } |
| } |
| RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate " |
| << estimated_frequency |
| << " not close to any standard RTP frequency." |
| << " Last timestamp " << last_rtp_timestamp |
| << " first timestamp " << first_rtp_timestamp; |
| return std::nullopt; |
| } |
| |
| std::optional<double> NetworkDelayDiff_AbsSendTime( |
| const LoggedRtpPacketIncoming& old_packet, |
| const LoggedRtpPacketIncoming& new_packet) { |
| if (old_packet.rtp.header.extension.hasAbsoluteSendTime && |
| new_packet.rtp.header.extension.hasAbsoluteSendTime) { |
| int64_t send_time_diff = WrappingDifference( |
| new_packet.rtp.header.extension.absoluteSendTime, |
| old_packet.rtp.header.extension.absoluteSendTime, 1ul << 24); |
| int64_t recv_time_diff = |
| new_packet.log_time_us() - old_packet.log_time_us(); |
| double delay_change_us = |
| recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff); |
| return delay_change_us / 1000; |
| } else { |
| return std::nullopt; |
| } |
| } |
| |
| std::optional<double> NetworkDelayDiff_CaptureTime( |
| const LoggedRtpPacketIncoming& old_packet, |
| const LoggedRtpPacketIncoming& new_packet, |
| const double sample_rate) { |
| int64_t send_time_diff = |
| WrappingDifference(new_packet.rtp.header.timestamp, |
| old_packet.rtp.header.timestamp, 1ull << 32); |
| int64_t recv_time_diff = new_packet.log_time_us() - old_packet.log_time_us(); |
| |
| double delay_change = |
| static_cast<double>(recv_time_diff) / 1000 - |
| static_cast<double>(send_time_diff) / sample_rate * 1000; |
| if (delay_change < -10000 || 10000 < delay_change) { |
| RTC_LOG(LS_WARNING) << "Very large delay change. Timestamps correct?"; |
| RTC_LOG(LS_WARNING) << "Old capture time " |
| << old_packet.rtp.header.timestamp << ", received time " |
| << old_packet.log_time_us(); |
| RTC_LOG(LS_WARNING) << "New capture time " |
| << new_packet.rtp.header.timestamp << ", received time " |
| << new_packet.log_time_us(); |
| RTC_LOG(LS_WARNING) << "Receive time difference " << recv_time_diff << " = " |
| << static_cast<double>(recv_time_diff) / |
| kNumMicrosecsPerSec |
| << "s"; |
| RTC_LOG(LS_WARNING) << "Send time difference " << send_time_diff << " = " |
| << static_cast<double>(send_time_diff) / sample_rate |
| << "s"; |
| } |
| return delay_change; |
| } |
| |
| 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 |
| |
| class BitrateObserver : public RemoteBitrateObserver { |
| public: |
| BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {} |
| |
| 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, |
| uint32_t bitrate) override {} |
| |
| uint32_t last_bitrate_bps() const { return last_bitrate_bps_; } |
| bool GetAndResetBitrateUpdated() { |
| bool bitrate_updated = bitrate_updated_; |
| bitrate_updated_ = false; |
| return bitrate_updated; |
| } |
| |
| private: |
| uint32_t last_bitrate_bps_; |
| bool bitrate_updated_; |
| }; |
| |
| void CreateIncomingDelayGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot) { |
| for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) { |
| // Filter on SSRC. |
| if (!MatchingSsrc(stream.ssrc, config.desired_ssrc_) || |
| IsRtxSsrc(parsed_log, kIncomingPacket, stream.ssrc)) { |
| continue; |
| } |
| |
| const std::vector<LoggedRtpPacketIncoming>& packets = |
| stream.incoming_packets; |
| if (packets.size() < 100) { |
| RTC_LOG(LS_WARNING) << "Can't estimate the RTP clock frequency with " |
| << packets.size() << " packets in the stream."; |
| continue; |
| } |
| int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us(); |
| std::optional<uint32_t> estimated_frequency = |
| EstimateRtpClockFrequency(packets, segment_end_us); |
| if (!estimated_frequency) |
| continue; |
| const double frequency_hz = *estimated_frequency; |
| if (IsVideoSsrc(parsed_log, kIncomingPacket, stream.ssrc) && |
| frequency_hz != 90000) { |
| RTC_LOG(LS_WARNING) |
| << "Video stream should use a 90 kHz clock but appears to use " |
| << frequency_hz / 1000 << ". Discarding."; |
| continue; |
| } |
| |
| auto ToCallTime = [&config](const LoggedRtpPacketIncoming& packet) { |
| return config.GetCallTimeSec(packet.log_time()); |
| }; |
| auto ToNetworkDelay = [frequency_hz]( |
| const LoggedRtpPacketIncoming& old_packet, |
| const LoggedRtpPacketIncoming& new_packet) { |
| return NetworkDelayDiff_CaptureTime(old_packet, new_packet, frequency_hz); |
| }; |
| |
| TimeSeries capture_time_data( |
| GetStreamName(parsed_log, kIncomingPacket, stream.ssrc) + |
| " capture-time", |
| LineStyle::kLine); |
| AccumulatePairs<LoggedRtpPacketIncoming, double>( |
| ToCallTime, ToNetworkDelay, packets, &capture_time_data); |
| plot->AppendTimeSeries(std::move(capture_time_data)); |
| |
| TimeSeries send_time_data( |
| GetStreamName(parsed_log, kIncomingPacket, stream.ssrc) + |
| " abs-send-time", |
| LineStyle::kLine); |
| AccumulatePairs<LoggedRtpPacketIncoming, double>( |
| ToCallTime, NetworkDelayDiff_AbsSendTime, packets, &send_time_data); |
| plot->AppendTimeSeriesIfNotEmpty(std::move(send_time_data)); |
| } |
| |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 1, "Delay (ms)", kBottomMargin, kTopMargin); |
| plot->SetTitle("Incoming network delay (relative to first packet)"); |
| } |
| |
| // Plot the fraction of packets lost (as perceived by the loss-based BWE). |
| void CreateFractionLossGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot) { |
| TimeSeries time_series("Fraction lost", LineStyle::kLine, |
| PointStyle::kHighlight); |
| for (auto& bwe_update : parsed_log.bwe_loss_updates()) { |
| float x = config.GetCallTimeSec(bwe_update.log_time()); |
| float y = static_cast<float>(bwe_update.fraction_lost) / 255 * 100; |
| time_series.points.emplace_back(x, y); |
| } |
| |
| plot->AppendTimeSeries(std::move(time_series)); |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 10, "Loss rate (in %)", kBottomMargin, kTopMargin); |
| plot->SetTitle("Outgoing packet loss (as reported by BWE)"); |
| } |
| |
| // Plot the total bandwidth used by all RTP streams. |
| void CreateTotalIncomingBitrateGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot) { |
| // TODO(terelius): This could be provided by the parser. |
| std::multimap<Timestamp, size_t> packets_in_order; |
| for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) { |
| for (const LoggedRtpPacketIncoming& packet : stream.incoming_packets) |
| packets_in_order.insert( |
| std::make_pair(packet.rtp.log_time(), packet.rtp.total_length)); |
| } |
| |
| auto window_begin = packets_in_order.begin(); |
| auto window_end = packets_in_order.begin(); |
| size_t bytes_in_window = 0; |
| |
| if (!packets_in_order.empty()) { |
| // Calculate a moving average of the bitrate and store in a TimeSeries. |
| TimeSeries bitrate_series("Bitrate", LineStyle::kLine); |
| for (Timestamp time = config.begin_time_; |
| time < config.end_time_ + config.step_; time += config.step_) { |
| while (window_end != packets_in_order.end() && window_end->first < time) { |
| bytes_in_window += window_end->second; |
| ++window_end; |
| } |
| while (window_begin != packets_in_order.end() && |
| window_begin->first < time - config.window_duration_) { |
| RTC_DCHECK_LE(window_begin->second, bytes_in_window); |
| bytes_in_window -= window_begin->second; |
| ++window_begin; |
| } |
| float window_duration_in_seconds = |
| static_cast<float>(config.window_duration_.us()) / |
| kNumMicrosecsPerSec; |
| float x = config.GetCallTimeSec(time); |
| float y = bytes_in_window * 8 / window_duration_in_seconds / 1000; |
| bitrate_series.points.emplace_back(x, y); |
| } |
| plot->AppendTimeSeries(std::move(bitrate_series)); |
| } |
| |
| // Overlay the outgoing REMB over incoming bitrate. |
| TimeSeries remb_series("Remb", LineStyle::kStep); |
| for (const auto& rtcp : parsed_log.rembs(kOutgoingPacket)) { |
| float x = config.GetCallTimeSec(rtcp.log_time()); |
| float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000; |
| remb_series.points.emplace_back(x, y); |
| } |
| plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series)); |
| |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin); |
| plot->SetTitle("Incoming RTP bitrate"); |
| } |
| |
| // Plot the total bandwidth used by all RTP streams. |
| void CreateTotalOutgoingBitrateGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot, |
| bool show_detector_state, |
| bool show_alr_state, |
| bool show_link_capacity) { |
| // TODO(terelius): This could be provided by the parser. |
| std::multimap<Timestamp, size_t> packets_in_order; |
| for (const auto& stream : parsed_log.outgoing_rtp_packets_by_ssrc()) { |
| for (const LoggedRtpPacketOutgoing& packet : stream.outgoing_packets) |
| packets_in_order.insert( |
| std::make_pair(packet.rtp.log_time(), packet.rtp.total_length)); |
| } |
| |
| auto window_begin = packets_in_order.begin(); |
| auto window_end = packets_in_order.begin(); |
| size_t bytes_in_window = 0; |
| |
| if (!packets_in_order.empty()) { |
| // Calculate a moving average of the bitrate and store in a TimeSeries. |
| TimeSeries bitrate_series("Bitrate", LineStyle::kLine); |
| for (Timestamp time = config.begin_time_; |
| time < config.end_time_ + config.step_; time += config.step_) { |
| while (window_end != packets_in_order.end() && window_end->first < time) { |
| bytes_in_window += window_end->second; |
| ++window_end; |
| } |
| while (window_begin != packets_in_order.end() && |
| window_begin->first < time - config.window_duration_) { |
| RTC_DCHECK_LE(window_begin->second, bytes_in_window); |
| bytes_in_window -= window_begin->second; |
| ++window_begin; |
| } |
| float window_duration_in_seconds = |
| static_cast<float>(config.window_duration_.us()) / |
| kNumMicrosecsPerSec; |
| float x = config.GetCallTimeSec(time); |
| float y = bytes_in_window * 8 / window_duration_in_seconds / 1000; |
| bitrate_series.points.emplace_back(x, y); |
| } |
| plot->AppendTimeSeries(std::move(bitrate_series)); |
| } |
| |
| // Overlay the send-side bandwidth estimate over the outgoing bitrate. |
| TimeSeries loss_series("Loss-based estimate", LineStyle::kStep); |
| for (auto& loss_update : parsed_log.bwe_loss_updates()) { |
| float x = config.GetCallTimeSec(loss_update.log_time()); |
| float y = static_cast<float>(loss_update.bitrate_bps) / 1000; |
| loss_series.points.emplace_back(x, y); |
| } |
| |
| TimeSeries link_capacity_lower_series("Link-capacity-lower", |
| LineStyle::kStep); |
| TimeSeries link_capacity_upper_series("Link-capacity-upper", |
| LineStyle::kStep); |
| for (auto& remote_estimate_event : parsed_log.remote_estimate_events()) { |
| float x = config.GetCallTimeSec(remote_estimate_event.log_time()); |
| if (remote_estimate_event.link_capacity_lower.has_value()) { |
| float link_capacity_lower = static_cast<float>( |
| remote_estimate_event.link_capacity_lower.value().kbps()); |
| link_capacity_lower_series.points.emplace_back(x, link_capacity_lower); |
| } |
| if (remote_estimate_event.link_capacity_upper.has_value()) { |
| float link_capacity_upper = static_cast<float>( |
| remote_estimate_event.link_capacity_upper.value().kbps()); |
| link_capacity_upper_series.points.emplace_back(x, link_capacity_upper); |
| } |
| } |
| |
| TimeSeries delay_series("Delay-based estimate", LineStyle::kStep); |
| IntervalSeries overusing_series("Overusing", "#ff8e82", |
| IntervalSeries::kHorizontal); |
| IntervalSeries underusing_series("Underusing", "#5092fc", |
| IntervalSeries::kHorizontal); |
| IntervalSeries normal_series("Normal", "#c4ffc4", |
| IntervalSeries::kHorizontal); |
| IntervalSeries* last_series = &normal_series; |
| float last_detector_switch = 0.0; |
| |
| BandwidthUsage last_detector_state = BandwidthUsage::kBwNormal; |
| |
| for (auto& delay_update : parsed_log.bwe_delay_updates()) { |
| float x = config.GetCallTimeSec(delay_update.log_time()); |
| float y = static_cast<float>(delay_update.bitrate_bps) / 1000; |
| |
| if (last_detector_state != delay_update.detector_state) { |
| last_series->intervals.emplace_back(last_detector_switch, x); |
| last_detector_state = delay_update.detector_state; |
| last_detector_switch = x; |
| |
| switch (delay_update.detector_state) { |
| case BandwidthUsage::kBwNormal: |
| last_series = &normal_series; |
| break; |
| case BandwidthUsage::kBwUnderusing: |
| last_series = &underusing_series; |
| break; |
| case BandwidthUsage::kBwOverusing: |
| last_series = &overusing_series; |
| break; |
| case BandwidthUsage::kLast: |
| RTC_DCHECK_NOTREACHED(); |
| } |
| } |
| |
| delay_series.points.emplace_back(x, y); |
| } |
| |
| RTC_CHECK(last_series); |
| last_series->intervals.emplace_back(last_detector_switch, |
| config.CallEndTimeSec()); |
| |
| TimeSeries scream_series("Scream target rate", LineStyle::kStep); |
| for (auto& scream_update : parsed_log.bwe_scream_updates()) { |
| float x = config.GetCallTimeSec(scream_update.log_time()); |
| float y = static_cast<float>(scream_update.target_rate.kbps()); |
| scream_series.points.emplace_back(x, y); |
| } |
| |
| TimeSeries created_series("Probe cluster created.", LineStyle::kNone, |
| PointStyle::kHighlight); |
| for (auto& cluster : parsed_log.bwe_probe_cluster_created_events()) { |
| float x = config.GetCallTimeSec(cluster.log_time()); |
| float y = static_cast<float>(cluster.bitrate_bps) / 1000; |
| created_series.points.emplace_back(x, y); |
| } |
| |
| TimeSeries result_series("Probing results.", LineStyle::kNone, |
| PointStyle::kHighlight); |
| for (auto& result : parsed_log.bwe_probe_success_events()) { |
| float x = config.GetCallTimeSec(result.log_time()); |
| float y = static_cast<float>(result.bitrate_bps) / 1000; |
| result_series.points.emplace_back(x, y); |
| } |
| |
| TimeSeries probe_failures_series("Probe failed", LineStyle::kNone, |
| PointStyle::kHighlight); |
| for (auto& failure : parsed_log.bwe_probe_failure_events()) { |
| float x = config.GetCallTimeSec(failure.log_time()); |
| probe_failures_series.points.emplace_back(x, 0); |
| } |
| |
| IntervalSeries alr_state("ALR", "#555555", IntervalSeries::kHorizontal); |
| bool previously_in_alr = false; |
| Timestamp alr_start = Timestamp::Zero(); |
| for (auto& alr : parsed_log.alr_state_events()) { |
| float y = config.GetCallTimeSec(alr.log_time()); |
| if (!previously_in_alr && alr.in_alr) { |
| alr_start = alr.log_time(); |
| previously_in_alr = true; |
| } else if (previously_in_alr && !alr.in_alr) { |
| float x = config.GetCallTimeSec(alr_start); |
| alr_state.intervals.emplace_back(x, y); |
| previously_in_alr = false; |
| } |
| } |
| |
| if (previously_in_alr) { |
| float x = config.GetCallTimeSec(alr_start); |
| float y = config.GetCallTimeSec(config.end_time_); |
| alr_state.intervals.emplace_back(x, y); |
| } |
| |
| if (show_detector_state) { |
| plot->AppendIntervalSeries(std::move(overusing_series)); |
| plot->AppendIntervalSeries(std::move(underusing_series)); |
| plot->AppendIntervalSeries(std::move(normal_series)); |
| } |
| |
| if (show_alr_state) { |
| plot->AppendIntervalSeries(std::move(alr_state)); |
| } |
| |
| if (show_link_capacity) { |
| plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_lower_series)); |
| plot->AppendTimeSeriesIfNotEmpty(std::move(link_capacity_upper_series)); |
| } |
| |
| plot->AppendTimeSeries(std::move(loss_series)); |
| plot->AppendTimeSeriesIfNotEmpty(std::move(probe_failures_series)); |
| plot->AppendTimeSeries(std::move(delay_series)); |
| plot->AppendTimeSeriesIfNotEmpty(std::move(scream_series)); |
| plot->AppendTimeSeries(std::move(created_series)); |
| plot->AppendTimeSeries(std::move(result_series)); |
| |
| // Overlay the incoming REMB over the outgoing bitrate. |
| TimeSeries remb_series("Remb", LineStyle::kStep); |
| for (const auto& rtcp : parsed_log.rembs(kIncomingPacket)) { |
| float x = config.GetCallTimeSec(rtcp.log_time()); |
| float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000; |
| remb_series.points.emplace_back(x, y); |
| } |
| plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series)); |
| |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin); |
| plot->SetTitle("Outgoing RTP bitrate"); |
| } |
| |
| void CreateGoogCcSimulationGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot) { |
| TimeSeries target_rates("Simulated target rate", LineStyle::kStep, |
| PointStyle::kHighlight); |
| TimeSeries delay_based("Logged delay-based estimate", LineStyle::kStep, |
| PointStyle::kHighlight); |
| TimeSeries loss_based("Logged loss-based estimate", LineStyle::kStep, |
| PointStyle::kHighlight); |
| TimeSeries probe_results("Logged probe success", LineStyle::kNone, |
| PointStyle::kHighlight); |
| |
| LogBasedNetworkControllerSimulation simulation( |
| config.env_, std::make_unique<GoogCcNetworkControllerFactory>(), |
| [&](const NetworkControlUpdate& update, Timestamp at_time) { |
| if (update.target_rate) { |
| target_rates.points.emplace_back( |
| config.GetCallTimeSec(at_time), |
| update.target_rate->target_rate.kbps<float>()); |
| } |
| }); |
| |
| simulation.ProcessEventsInLog(parsed_log); |
| for (const auto& logged : parsed_log.bwe_delay_updates()) |
| delay_based.points.emplace_back(config.GetCallTimeSec(logged.log_time()), |
| logged.bitrate_bps / 1000); |
| for (const auto& logged : parsed_log.bwe_probe_success_events()) |
| probe_results.points.emplace_back(config.GetCallTimeSec(logged.log_time()), |
| logged.bitrate_bps / 1000); |
| for (const auto& logged : parsed_log.bwe_loss_updates()) |
| loss_based.points.emplace_back(config.GetCallTimeSec(logged.log_time()), |
| logged.bitrate_bps / 1000); |
| |
| plot->AppendTimeSeries(std::move(delay_based)); |
| plot->AppendTimeSeries(std::move(loss_based)); |
| plot->AppendTimeSeries(std::move(probe_results)); |
| plot->AppendTimeSeries(std::move(target_rates)); |
| |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin); |
| plot->SetTitle("Simulated BWE behavior"); |
| } |
| |
| void CreateScreamSimulationBitrateGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot) { |
| TimeSeries target_rate_series("Target rate", LineStyle::kStep); |
| TimeSeries pacing_rate_series("Pacing rate", LineStyle::kStep); |
| TimeSeries send_rate_series("Send rate", LineStyle::kStep); |
| |
| LogScreamSimulation simulation({.rate_window = config.window_duration_}, |
| config.env_); |
| simulation.ProcessEventsInLog(parsed_log); |
| |
| for (const LogScreamSimulation::State& state : simulation.updates()) { |
| target_rate_series.points.emplace_back(config.GetCallTimeSec(state.time), |
| state.target_rate.bps() / 1000); |
| pacing_rate_series.points.emplace_back(config.GetCallTimeSec(state.time), |
| state.pacing_rate.bps() / 1000); |
| send_rate_series.points.emplace_back(config.GetCallTimeSec(state.time), |
| state.send_rate.bps() / 1000); |
| } |
| plot->AppendTimeSeries(std::move(target_rate_series)); |
| plot->AppendTimeSeries(std::move(pacing_rate_series)); |
| plot->AppendTimeSeries(std::move(send_rate_series)); |
| |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 100, "Kbps", kBottomMargin, kTopMargin); |
| plot->SetTitle("Simulated Scream rates"); |
| } |
| |
| void CreateScreamSimulationRefWindowGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot) { |
| using SendWindowUsage = LogScreamSimulation::State::SendWindowUsage; |
| TimeSeries ref_window_series("RefWindow", LineStyle::kStep); |
| TimeSeries ref_window_i_series("RefWindowI", LineStyle::kStep); |
| TimeSeries max_data_in_flight("Max allowed data in flight", LineStyle::kStep); |
| TimeSeries data_in_flight("Data in flight", LineStyle::kStep); |
| IntervalSeries send_window_above_max_series( |
| "Data in flight > Max allowed", "#ff8e82", IntervalSeries::kHorizontal); |
| IntervalSeries send_window_below_ref_window_series( |
| "Data in flight < RefWindow", "#c5dff2", IntervalSeries::kHorizontal); |
| IntervalSeries send_window_above_ref_window_series( |
| "Data in flight >= RefWindow", "#b9fad8", IntervalSeries::kHorizontal); |
| IntervalSeries* last_series = &send_window_below_ref_window_series; |
| |
| LogScreamSimulation simulation({.rate_window = config.window_duration_}, |
| config.env_); |
| simulation.ProcessEventsInLog(parsed_log); |
| if (simulation.updates().empty()) { |
| RTC_LOG(LS_ERROR) << "Empty simulation."; |
| return; |
| } |
| |
| float send_window_state_switch = |
| config.GetCallTimeSec(simulation.updates().front().time); |
| SendWindowUsage send_window_usage = SendWindowUsage::kBelowRefWindow; |
| |
| for (const LogScreamSimulation::State& state : simulation.updates()) { |
| ref_window_series.points.emplace_back(config.GetCallTimeSec(state.time), |
| state.ref_window.bytes()); |
| ref_window_i_series.points.emplace_back(config.GetCallTimeSec(state.time), |
| state.ref_window_i.bytes()); |
| max_data_in_flight.points.emplace_back(config.GetCallTimeSec(state.time), |
| state.max_data_in_flight.bytes()); |
| data_in_flight.points.emplace_back(config.GetCallTimeSec(state.time), |
| state.data_in_flight.bytes()); |
| if (state.send_window_usage != send_window_usage) { |
| last_series->intervals.emplace_back(send_window_state_switch, |
| config.GetCallTimeSec(state.time)); |
| send_window_usage = state.send_window_usage; |
| send_window_state_switch = config.GetCallTimeSec(state.time); |
| switch (send_window_usage) { |
| case SendWindowUsage::kAboveRefWindow: |
| last_series = &send_window_above_ref_window_series; |
| break; |
| case SendWindowUsage::kBelowRefWindow: |
| last_series = &send_window_below_ref_window_series; |
| break; |
| case SendWindowUsage::kAboveScreamMax: |
| last_series = &send_window_above_max_series; |
| break; |
| } |
| } |
| } |
| last_series->intervals.emplace_back(send_window_state_switch, |
| config.CallEndTimeSec()); |
| plot->AppendTimeSeries(std::move(ref_window_series)); |
| plot->AppendTimeSeries(std::move(ref_window_i_series)); |
| plot->AppendTimeSeries(std::move(max_data_in_flight)); |
| plot->AppendTimeSeries(std::move(data_in_flight)); |
| plot->AppendIntervalSeries(std::move(send_window_above_max_series)); |
| plot->AppendIntervalSeries(std::move(send_window_below_ref_window_series)); |
| plot->AppendIntervalSeries(std::move(send_window_above_ref_window_series)); |
| |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 10, "Bytes", kBottomMargin, kTopMargin); |
| plot->SetTitle("Simulated Scream RefWindow"); |
| } |
| |
| void CreateScreamSimulationRatiosGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot) { |
| TimeSeries queue_delay_dev_norm_series("QueueDelayDevNorm", LineStyle::kStep); |
| TimeSeries l4s_alpha_series("L4sAlpha", LineStyle::kStep); |
| TimeSeries l4s_alpha_v_series("L4sAlphaV", LineStyle::kStep); |
| TimeSeries ref_window_scale_factor_due_to_increased_delay( |
| "RefWindowScaleFactorDueToIncreasedDelay", LineStyle::kStep); |
| TimeSeries ref_window_scale_factor_due_to_delay_variation( |
| "RefWindowScaleFactorDueToDelayVariation", LineStyle::kStep); |
| |
| LogScreamSimulation simulation({.rate_window = config.window_duration_}, |
| config.env_); |
| simulation.ProcessEventsInLog(parsed_log); |
| |
| for (const LogScreamSimulation::State& state : simulation.updates()) { |
| queue_delay_dev_norm_series.points.emplace_back( |
| config.GetCallTimeSec(state.time), state.queue_delay_dev_norm); |
| l4s_alpha_series.points.emplace_back(config.GetCallTimeSec(state.time), |
| state.l4s_alpha); |
| l4s_alpha_v_series.points.emplace_back(config.GetCallTimeSec(state.time), |
| state.l4s_alpha_v); |
| ref_window_scale_factor_due_to_increased_delay.points.emplace_back( |
| config.GetCallTimeSec(state.time), |
| state.ref_window_scale_factor_due_to_increased_delay); |
| ref_window_scale_factor_due_to_delay_variation.points.emplace_back( |
| config.GetCallTimeSec(state.time), |
| state.ref_window_scale_factor_due_to_delay_variation); |
| } |
| plot->AppendTimeSeries(std::move(queue_delay_dev_norm_series)); |
| plot->AppendTimeSeries(std::move(l4s_alpha_series)); |
| plot->AppendTimeSeries(std::move(l4s_alpha_v_series)); |
| plot->AppendTimeSeries( |
| std::move(ref_window_scale_factor_due_to_increased_delay)); |
| plot->AppendTimeSeries( |
| std::move(ref_window_scale_factor_due_to_delay_variation)); |
| |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 1, "Ratios", kBottomMargin, kTopMargin); |
| plot->SetTitle("Simulated Scream Ratios"); |
| } |
| |
| void CreateScreamRefWindowGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot) { |
| TimeSeries ref_window_series("RefWindow", LineStyle::kStep); |
| for (auto& scream_update : parsed_log.bwe_scream_updates()) { |
| float x = config.GetCallTimeSec(scream_update.log_time()); |
| float y = static_cast<float>(scream_update.ref_window.bytes()); |
| ref_window_series.points.emplace_back(x, y); |
| } |
| plot->AppendTimeSeries(std::move(ref_window_series)); |
| |
| TimeSeries data_in_flight_series("Data in flight", LineStyle::kLine); |
| for (auto& scream_update : parsed_log.bwe_scream_updates()) { |
| float x = config.GetCallTimeSec(scream_update.log_time()); |
| float y = static_cast<float>(scream_update.data_in_flight.bytes()); |
| data_in_flight_series.points.emplace_back(x, y); |
| } |
| plot->AppendTimeSeries(std::move(data_in_flight_series)); |
| |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 3000, "Bytes", kBottomMargin, kTopMargin); |
| plot->SetTitle("Scream Ref Window"); |
| } |
| |
| void CreateScreamDelayEstimateGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot) { |
| TimeSeries smoothed_rtt_series("Smoothed RTT", LineStyle::kStep); |
| TimeSeries avg_queue_delay_series("Avg queue delay", LineStyle::kStep); |
| |
| for (auto& scream_update : parsed_log.bwe_scream_updates()) { |
| float x = config.GetCallTimeSec(scream_update.log_time()); |
| float smoothed_rtt_ms = static_cast<float>(scream_update.smoothed_rtt.ms()); |
| smoothed_rtt_series.points.emplace_back(x, smoothed_rtt_ms); |
| float avg_queue_delay_ms = |
| static_cast<float>(scream_update.avg_queue_delay.ms()); |
| avg_queue_delay_series.points.emplace_back(x, avg_queue_delay_ms); |
| } |
| |
| plot->AppendTimeSeries(std::move(smoothed_rtt_series)); |
| plot->AppendTimeSeries(std::move(avg_queue_delay_series)); |
| |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 50, "Delay (ms)", kBottomMargin, kTopMargin); |
| plot->SetTitle("Scream delay estimates"); |
| } |
| |
| void CreateSendSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot) { |
| using RtpPacketType = LoggedRtpPacketOutgoing; |
| using TransportFeedbackType = LoggedRtcpPacketTransportFeedback; |
| |
| // TODO(terelius): The parser could provide a clearer view of the |
| // streams, so that we don't have to recalculate it. |
| std::multimap<int64_t, const RtpPacketType*> outgoing_rtp; |
| for (const auto& stream : parsed_log.outgoing_rtp_packets_by_ssrc()) { |
| for (const RtpPacketType& rtp_packet : stream.outgoing_packets) |
| outgoing_rtp.insert( |
| std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet)); |
| } |
| |
| const std::vector<TransportFeedbackType>& incoming_rtcp = |
| parsed_log.transport_feedbacks(kIncomingPacket); |
| |
| SimulatedClock clock(0); |
| BitrateObserver observer; |
| TransportFeedbackAdapter transport_feedback; |
| auto factory = GoogCcNetworkControllerFactory(); |
| TimeDelta process_interval = factory.GetProcessInterval(); |
| // TODO(holmer): Log the call config and use that here instead. |
| static const uint32_t kDefaultStartBitrateBps = 300000; |
| NetworkControllerConfig cc_config(config.env_); |
| cc_config.constraints.at_time = clock.CurrentTime(); |
| cc_config.constraints.starting_rate = |
| DataRate::BitsPerSec(kDefaultStartBitrateBps); |
| auto goog_cc = factory.Create(cc_config); |
| |
| TimeSeries time_series("Delay-based estimate", LineStyle::kStep, |
| PointStyle::kHighlight); |
| TimeSeries acked_time_series("Raw acked bitrate", LineStyle::kLine, |
| PointStyle::kHighlight); |
| TimeSeries robust_time_series("Robust throughput estimate", LineStyle::kLine, |
| PointStyle::kHighlight); |
| TimeSeries acked_estimate_time_series("Ackednowledged bitrate estimate", |
| LineStyle::kLine, |
| PointStyle::kHighlight); |
| |
| auto rtp_iterator = outgoing_rtp.begin(); |
| auto rtcp_iterator = incoming_rtcp.begin(); |
| |
| auto NextRtpTime = [&]() { |
| if (rtp_iterator != outgoing_rtp.end()) |
| return static_cast<int64_t>(rtp_iterator->first); |
| return std::numeric_limits<int64_t>::max(); |
| }; |
| |
| auto NextRtcpTime = [&]() { |
| if (rtcp_iterator != incoming_rtcp.end()) |
| return static_cast<int64_t>(rtcp_iterator->log_time_us()); |
| return std::numeric_limits<int64_t>::max(); |
| }; |
| int64_t next_process_time_us_ = std::min({NextRtpTime(), NextRtcpTime()}); |
| |
| auto NextProcessTime = [&]() { |
| if (rtcp_iterator != incoming_rtcp.end() || |
| rtp_iterator != outgoing_rtp.end()) { |
| return next_process_time_us_; |
| } |
| return std::numeric_limits<int64_t>::max(); |
| }; |
| |
| RateStatistics raw_acked_bitrate(750, 8000); |
| FieldTrials throughput_config( |
| "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:true/"); |
| std::unique_ptr<AcknowledgedBitrateEstimatorInterface> |
| robust_throughput_estimator( |
| AcknowledgedBitrateEstimatorInterface::Create(&throughput_config)); |
| FieldTrials acked_bitrate_config( |
| "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:false/"); |
| std::unique_ptr<AcknowledgedBitrateEstimatorInterface> |
| acknowledged_bitrate_estimator( |
| AcknowledgedBitrateEstimatorInterface::Create(&acked_bitrate_config)); |
| 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() >= NextRtpTime()) { |
| RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime()); |
| const RtpPacketType& rtp_packet = *rtp_iterator->second; |
| if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) { |
| RtpPacketToSend send_packet(/*extensions=*/nullptr); |
| send_packet.set_transport_sequence_number( |
| rtp_packet.rtp.header.extension.transportSequenceNumber); |
| send_packet.SetSsrc(rtp_packet.rtp.header.ssrc); |
| send_packet.SetSequenceNumber(rtp_packet.rtp.header.sequenceNumber); |
| send_packet.SetPayloadSize(rtp_packet.rtp.total_length - |
| send_packet.headers_size()); |
| RTC_DCHECK_EQ(send_packet.size(), rtp_packet.rtp.total_length); |
| if (IsRtxSsrc(parsed_log, PacketDirection::kOutgoingPacket, |
| rtp_packet.rtp.header.ssrc)) { |
| // Don't set the optional media type as we don't know if it is |
| // a retransmission, FEC or padding. |
| } else if (IsVideoSsrc(parsed_log, PacketDirection::kOutgoingPacket, |
| rtp_packet.rtp.header.ssrc)) { |
| send_packet.set_packet_type(RtpPacketMediaType::kVideo); |
| } else if (IsAudioSsrc(parsed_log, PacketDirection::kOutgoingPacket, |
| rtp_packet.rtp.header.ssrc)) { |
| send_packet.set_packet_type(RtpPacketMediaType::kAudio); |
| } |
| transport_feedback.AddPacket( |
| send_packet, PacedPacketInfo(), |
| 0u, // Per packet overhead bytes., |
| Timestamp::Micros(rtp_packet.rtp.log_time_us())); |
| } |
| SentPacketInfo sent_packet; |
| sent_packet.send_time_ms = rtp_packet.rtp.log_time_ms(); |
| sent_packet.info.included_in_allocation = true; |
| sent_packet.info.packet_size_bytes = rtp_packet.rtp.total_length; |
| if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) { |
| sent_packet.packet_id = |
| rtp_packet.rtp.header.extension.transportSequenceNumber; |
| sent_packet.info.included_in_feedback = true; |
| } |
| auto sent_msg = transport_feedback.ProcessSentPacket(sent_packet); |
| if (sent_msg) |
| observer.Update(goog_cc->OnSentPacket(*sent_msg)); |
| ++rtp_iterator; |
| } |
| if (clock.TimeInMicroseconds() >= NextRtcpTime()) { |
| RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime()); |
| |
| auto feedback_msg = transport_feedback.ProcessTransportFeedback( |
| rtcp_iterator->transport_feedback, clock.CurrentTime()); |
| if (feedback_msg) { |
| observer.Update(goog_cc->OnTransportPacketsFeedback(*feedback_msg)); |
| std::vector<PacketResult> feedback = |
| feedback_msg->SortedByReceiveTime(); |
| if (!feedback.empty()) { |
| acknowledged_bitrate_estimator->IncomingPacketFeedbackVector( |
| feedback); |
| robust_throughput_estimator->IncomingPacketFeedbackVector(feedback); |
| for (const PacketResult& packet : feedback) { |
| raw_acked_bitrate.Update(packet.sent_packet.size.bytes(), |
| packet.receive_time.ms()); |
| } |
| std::optional<uint32_t> raw_bitrate_bps = |
| raw_acked_bitrate.Rate(feedback.back().receive_time.ms()); |
| float x = config.GetCallTimeSec(clock.CurrentTime()); |
| if (raw_bitrate_bps) { |
| float y = raw_bitrate_bps.value() / 1000; |
| acked_time_series.points.emplace_back(x, y); |
| } |
| std::optional<DataRate> robust_estimate = |
| robust_throughput_estimator->bitrate(); |
| if (robust_estimate) { |
| float y = robust_estimate.value().kbps(); |
| robust_time_series.points.emplace_back(x, y); |
| } |
| std::optional<DataRate> acked_estimate = |
| acknowledged_bitrate_estimator->bitrate(); |
| if (acked_estimate) { |
| float y = acked_estimate.value().kbps(); |
| acked_estimate_time_series.points.emplace_back(x, y); |
| } |
| } |
| } |
| ++rtcp_iterator; |
| } |
| if (clock.TimeInMicroseconds() >= NextProcessTime()) { |
| RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime()); |
| ProcessInterval msg; |
| msg.at_time = clock.CurrentTime(); |
| observer.Update(goog_cc->OnProcessInterval(msg)); |
| next_process_time_us_ += process_interval.us(); |
| } |
| if (observer.GetAndResetBitrateUpdated() || |
| time_us - last_update_us >= 1e6) { |
| uint32_t y = observer.last_bitrate_bps() / 1000; |
| float x = config.GetCallTimeSec(clock.CurrentTime()); |
| time_series.points.emplace_back(x, y); |
| last_update_us = time_us; |
| } |
| time_us = std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()}); |
| } |
| // Add the data set to the plot. |
| plot->AppendTimeSeries(std::move(time_series)); |
| plot->AppendTimeSeries(std::move(robust_time_series)); |
| plot->AppendTimeSeries(std::move(acked_time_series)); |
| plot->AppendTimeSeriesIfNotEmpty(std::move(acked_estimate_time_series)); |
| |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin); |
| plot->SetTitle("Simulated send-side BWE behavior"); |
| } |
| |
| void CreateReceiveSideBweSimulationGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot) { |
| using RtpPacketType = LoggedRtpPacketIncoming; |
| class RembInterceptor { |
| public: |
| void SendRemb(uint32_t bitrate_bps, std::vector<uint32_t> ssrcs) { |
| last_bitrate_bps_ = bitrate_bps; |
| bitrate_updated_ = true; |
| } |
| uint32_t last_bitrate_bps() const { return last_bitrate_bps_; } |
| bool GetAndResetBitrateUpdated() { |
| bool bitrate_updated = bitrate_updated_; |
| bitrate_updated_ = false; |
| return bitrate_updated; |
| } |
| |
| private: |
| // We don't know the start bitrate, but assume that it is the default 300 |
| // kbps. |
| uint32_t last_bitrate_bps_ = 300000; |
| bool bitrate_updated_ = false; |
| }; |
| |
| std::multimap<int64_t, const RtpPacketType*> incoming_rtp; |
| |
| for (const auto& stream : parsed_log.incoming_rtp_packets_by_ssrc()) { |
| if (IsVideoSsrc(parsed_log, kIncomingPacket, stream.ssrc)) { |
| for (const auto& rtp_packet : stream.incoming_packets) |
| incoming_rtp.insert( |
| std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet)); |
| } |
| } |
| |
| SimulatedClock clock(0); |
| EnvironmentFactory env_factory(config.env_); |
| env_factory.Set(&clock); |
| RembInterceptor remb_interceptor; |
| ReceiveSideCongestionController rscc( |
| env_factory.Create(), [](auto...) {}, |
| absl::bind_front(&RembInterceptor::SendRemb, &remb_interceptor)); |
| // TODO(holmer): Log the call config and use that here instead. |
| // static const uint32_t kDefaultStartBitrateBps = 300000; |
| // rscc.SetBweBitrates(0, kDefaultStartBitrateBps, -1); |
| |
| TimeSeries time_series("Receive side estimate", LineStyle::kLine, |
| PointStyle::kHighlight); |
| TimeSeries acked_time_series("Received bitrate", LineStyle::kLine); |
| |
| RateStatistics acked_bitrate(250, 8000); |
| int64_t last_update_us = 0; |
| for (const auto& kv : incoming_rtp) { |
| const RtpPacketType& packet = *kv.second; |
| |
| 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); |
| std::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms); |
| if (bitrate_bps) { |
| uint32_t y = *bitrate_bps / 1000; |
| float x = config.GetCallTimeSec(clock.CurrentTime()); |
| acked_time_series.points.emplace_back(x, y); |
| } |
| if (remb_interceptor.GetAndResetBitrateUpdated() || |
| clock.TimeInMicroseconds() - last_update_us >= 1e6) { |
| uint32_t y = remb_interceptor.last_bitrate_bps() / 1000; |
| float x = config.GetCallTimeSec(clock.CurrentTime()); |
| time_series.points.emplace_back(x, y); |
| last_update_us = clock.TimeInMicroseconds(); |
| } |
| } |
| // Add the data set to the plot. |
| plot->AppendTimeSeries(std::move(time_series)); |
| plot->AppendTimeSeries(std::move(acked_time_series)); |
| |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin); |
| plot->SetTitle("Simulated receive-side BWE behavior"); |
| } |
| |
| void CreateNetworkDelayFeedbackGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot) { |
| TimeSeries time_series("Network delay", LineStyle::kLine, |
| PointStyle::kHighlight); |
| int64_t min_send_receive_diff_ms = std::numeric_limits<int64_t>::max(); |
| int64_t min_rtt_ms = std::numeric_limits<int64_t>::max(); |
| |
| std::vector<MatchedSendArrivalTimes> matched_rtp_rtcp = |
| GetNetworkTrace(parsed_log); |
| absl::c_stable_sort(matched_rtp_rtcp, [](const MatchedSendArrivalTimes& a, |
| const MatchedSendArrivalTimes& b) { |
| return a.feedback_arrival_time_ms < b.feedback_arrival_time_ms || |
| (a.feedback_arrival_time_ms == b.feedback_arrival_time_ms && |
| a.arrival_time_ms < b.arrival_time_ms); |
| }); |
| for (const auto& packet : matched_rtp_rtcp) { |
| if (packet.arrival_time_ms == MatchedSendArrivalTimes::kNotReceived) |
| continue; |
| float x = config.GetCallTimeSecFromMs(packet.feedback_arrival_time_ms); |
| int64_t y = packet.arrival_time_ms - packet.send_time_ms; |
| int64_t rtt_ms = packet.feedback_arrival_time_ms - packet.send_time_ms; |
| min_rtt_ms = std::min(rtt_ms, min_rtt_ms); |
| min_send_receive_diff_ms = std::min(y, min_send_receive_diff_ms); |
| time_series.points.emplace_back(x, y); |
| } |
| |
| // We assume that the base network delay (w/o queues) is equal to half |
| // the minimum RTT. Therefore rescale the delays by subtracting the minimum |
| // observed 1-ways delay and add half the minimum RTT. |
| const int64_t estimated_clock_offset_ms = |
| min_send_receive_diff_ms - min_rtt_ms / 2; |
| for (TimeSeriesPoint& point : time_series.points) |
| point.y -= estimated_clock_offset_ms; |
| |
| // Add the data set to the plot. |
| plot->AppendTimeSeriesIfNotEmpty(std::move(time_series)); |
| |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin); |
| plot->SetTitle("Outgoing network delay (based on per-packet feedback)"); |
| } |
| |
| void CreatePacerDelayGraph(const ParsedRtcEventLog& parsed_log, |
| const AnalyzerConfig& config, |
| Plot* plot) { |
| for (const auto& stream : parsed_log.outgoing_rtp_packets_by_ssrc()) { |
| const std::vector<LoggedRtpPacketOutgoing>& packets = |
| stream.outgoing_packets; |
| |
| if (IsRtxSsrc(parsed_log, kOutgoingPacket, stream.ssrc)) { |
| continue; |
| } |
| |
| if (packets.size() < 2) { |
| RTC_LOG(LS_WARNING) |
| << "Can't estimate a the RTP clock frequency or the " |
| "pacer delay with less than 2 packets in the stream"; |
| continue; |
| } |
| int64_t segment_end_us = parsed_log.first_log_segment().stop_time_us(); |
| std::optional<uint32_t> estimated_frequency = |
| EstimateRtpClockFrequency(packets, segment_end_us); |
| if (!estimated_frequency) |
| continue; |
| if (IsVideoSsrc(parsed_log, kOutgoingPacket, stream.ssrc) && |
| *estimated_frequency != 90000) { |
| RTC_LOG(LS_WARNING) |
| << "Video stream should use a 90 kHz clock but appears to use " |
| << *estimated_frequency / 1000 << ". Discarding."; |
| continue; |
| } |
| |
| TimeSeries pacer_delay_series( |
| GetStreamName(parsed_log, kOutgoingPacket, stream.ssrc) + "(" + |
| std::to_string(*estimated_frequency / 1000) + " kHz)", |
| LineStyle::kLine, PointStyle::kHighlight); |
| SeqNumUnwrapper<uint32_t> timestamp_unwrapper; |
| uint64_t first_capture_timestamp = |
| timestamp_unwrapper.Unwrap(packets.front().rtp.header.timestamp); |
| uint64_t first_send_timestamp = packets.front().rtp.log_time_us(); |
| for (const auto& packet : packets) { |
| double capture_time_ms = (static_cast<double>(timestamp_unwrapper.Unwrap( |
| packet.rtp.header.timestamp)) - |
| first_capture_timestamp) / |
| *estimated_frequency * 1000; |
| double send_time_ms = |
| static_cast<double>(packet.rtp.log_time_us() - first_send_timestamp) / |
| 1000; |
| float x = config.GetCallTimeSec(packet.rtp.log_time()); |
| float y = send_time_ms - capture_time_ms; |
| pacer_delay_series.points.emplace_back(x, y); |
| } |
| plot->AppendTimeSeries(std::move(pacer_delay_series)); |
| } |
| |
| plot->SetXAxis(config.CallBeginTimeSec(), config.CallEndTimeSec(), "Time (s)", |
| kLeftMargin, kRightMargin); |
| plot->SetSuggestedYAxis(0, 10, "Pacer delay (ms)", kBottomMargin, kTopMargin); |
| plot->SetTitle( |
| "Delay from capture to send time. (First packet normalized to 0.)"); |
| } |
| |
| } // namespace webrtc |
