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webrtc / src / 1545cdc76b2120398ff2b47dbad454e4da8c1675 / . / rtc_tools / rtc_event_log_visualizer / analyzer.cc
blob: 5ed4783082cf7d547146042edf6017072186522a [file] [log] [blame]
/*
* Copyright (c) 2016 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/analyzer.h"
#include <algorithm>
#include <cmath>
#include <cstddef>
#include <cstdint>
#include <deque>
#include <limits>
#include <map>
#include <memory>
#include <string>
#include <tuple>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/functional/bind_front.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/dtls_transport_interface.h"
#include "api/environment/environment_factory.h"
#include "api/function_view.h"
#include "api/media_types.h"
#include "api/network_state_predictor.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_headers.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/events/rtc_event_generic_packet_received.h"
#include "logging/rtc_event_log/events/rtc_event_generic_packet_sent.h"
#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "logging/rtc_event_log/rtc_event_processor.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/receiver_report.h"
#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.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 "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_base/strings/string_builder.h"
#include "rtc_tools/rtc_event_log_visualizer/analyze_audio.h"
#include "rtc_tools/rtc_event_log_visualizer/analyzer_common.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"
#include "test/explicit_key_value_config.h"
namespace webrtc {
namespace {
std::string SsrcToString(uint32_t ssrc) {
rtc::StringBuilder ss;
ss << "SSRC " << ssrc;
return ss.Release();
}
// Checks whether an SSRC is contained in the list of desired SSRCs.
// Note that an empty SSRC list matches every SSRC.
bool MatchingSsrc(uint32_t ssrc, const std::vector<uint32_t>& desired_ssrc) {
if (desired_ssrc.empty())
return true;
return std::find(desired_ssrc.begin(), desired_ssrc.end(), ssrc) !=
desired_ssrc.end();
}
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;
}
// Computes the difference `later` - `earlier` where `later` and `earlier`
// are counters that wrap at `modulus`. The difference is chosen to have the
// least absolute value. For example if `modulus` is 8, then the difference will
// be chosen in the range [-3, 4]. If `modulus` is 9, then the difference will
// be in [-4, 4].
int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus) {
RTC_DCHECK_LE(1, modulus);
RTC_DCHECK_LT(later, modulus);
RTC_DCHECK_LT(earlier, modulus);
int64_t difference =
static_cast<int64_t>(later) - static_cast<int64_t>(earlier);
int64_t max_difference = modulus / 2;
int64_t min_difference = max_difference - modulus + 1;
if (difference > max_difference) {
difference -= modulus;
}
if (difference < min_difference) {
difference += modulus;
}
if (difference > max_difference / 2 || difference < min_difference / 2) {
RTC_LOG(LS_WARNING) << "Difference between" << later << " and " << earlier
<< " expected to be in the range ("
<< min_difference / 2 << "," << max_difference / 2
<< ") but is " << difference
<< ". Correct unwrapping is uncertain.";
}
return difference;
}
// This is much more reliable for outgoing streams than for incoming streams.
template <typename RtpPacketContainer>
absl::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 absl::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 absl::nullopt;
}
absl::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 absl::nullopt;
}
}
absl::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;
}
template <typename T>
TimeSeries CreateRtcpTypeTimeSeries(const std::vector<T>& rtcp_list,
AnalyzerConfig config,
std::string rtcp_name,
int category_id) {
TimeSeries time_series(rtcp_name, LineStyle::kNone, PointStyle::kHighlight);
for (const auto& rtcp : rtcp_list) {
float x = config.GetCallTimeSec(rtcp.timestamp);
float y = category_id;
time_series.points.emplace_back(x, y);
}
return time_series;
}
const char kUnknownEnumValue[] = "unknown";
// TODO(tommi): This should be "host".
const char kIceCandidateTypeLocal[] = "local";
// TODO(tommi): This should be "srflx".
const char kIceCandidateTypeStun[] = "stun";
const char kIceCandidateTypePrflx[] = "prflx";
const char kIceCandidateTypeRelay[] = "relay";
const char kProtocolUdp[] = "udp";
const char kProtocolTcp[] = "tcp";
const char kProtocolSsltcp[] = "ssltcp";
const char kProtocolTls[] = "tls";
const char kAddressFamilyIpv4[] = "ipv4";
const char kAddressFamilyIpv6[] = "ipv6";
const char kNetworkTypeEthernet[] = "ethernet";
const char kNetworkTypeLoopback[] = "loopback";
const char kNetworkTypeWifi[] = "wifi";
const char kNetworkTypeVpn[] = "vpn";
const char kNetworkTypeCellular[] = "cellular";
absl::string_view GetIceCandidateTypeAsString(IceCandidateType type) {
switch (type) {
case IceCandidateType::kHost:
return kIceCandidateTypeLocal;
case IceCandidateType::kSrflx:
return kIceCandidateTypeStun;
case IceCandidateType::kPrflx:
return kIceCandidateTypePrflx;
case IceCandidateType::kRelay:
return kIceCandidateTypeRelay;
default:
RTC_DCHECK_NOTREACHED();
return kUnknownEnumValue;
}
}
std::string GetProtocolAsString(webrtc::IceCandidatePairProtocol protocol) {
switch (protocol) {
case webrtc::IceCandidatePairProtocol::kUdp:
return kProtocolUdp;
case webrtc::IceCandidatePairProtocol::kTcp:
return kProtocolTcp;
case webrtc::IceCandidatePairProtocol::kSsltcp:
return kProtocolSsltcp;
case webrtc::IceCandidatePairProtocol::kTls:
return kProtocolTls;
default:
return kUnknownEnumValue;
}
}
std::string GetAddressFamilyAsString(
webrtc::IceCandidatePairAddressFamily family) {
switch (family) {
case webrtc::IceCandidatePairAddressFamily::kIpv4:
return kAddressFamilyIpv4;
case webrtc::IceCandidatePairAddressFamily::kIpv6:
return kAddressFamilyIpv6;
default:
return kUnknownEnumValue;
}
}
std::string GetNetworkTypeAsString(webrtc::IceCandidateNetworkType type) {
switch (type) {
case webrtc::IceCandidateNetworkType::kEthernet:
return kNetworkTypeEthernet;
case webrtc::IceCandidateNetworkType::kLoopback:
return kNetworkTypeLoopback;
case webrtc::IceCandidateNetworkType::kWifi:
return kNetworkTypeWifi;
case webrtc::IceCandidateNetworkType::kVpn:
return kNetworkTypeVpn;
case webrtc::IceCandidateNetworkType::kCellular:
return kNetworkTypeCellular;
default:
return kUnknownEnumValue;
}
}
std::string GetCandidatePairLogDescriptionAsString(
const LoggedIceCandidatePairConfig& config) {
// Example: stun:wifi->relay(tcp):cellular@udp:ipv4
// represents a pair of a local server-reflexive candidate on a WiFi network
// and a remote relay candidate using TCP as the relay protocol on a cell
// network, when the candidate pair communicates over UDP using IPv4.
rtc::StringBuilder ss;
ss << GetIceCandidateTypeAsString(config.local_candidate_type);
if (config.local_candidate_type == IceCandidateType::kRelay) {
ss << "(" << GetProtocolAsString(config.local_relay_protocol) << ")";
}
ss << ":" << GetNetworkTypeAsString(config.local_network_type) << ":"
<< GetAddressFamilyAsString(config.local_address_family) << "->"
<< GetIceCandidateTypeAsString(config.remote_candidate_type) << ":"
<< GetAddressFamilyAsString(config.remote_address_family) << "@"
<< GetProtocolAsString(config.candidate_pair_protocol);
return ss.Release();
}
std::string GetDirectionAsString(PacketDirection direction) {
if (direction == kIncomingPacket) {
return "Incoming";
} else {
return "Outgoing";
}
}
std::string GetDirectionAsShortString(PacketDirection direction) {
if (direction == kIncomingPacket) {
return "In";
} else {
return "Out";
}
}
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;
}
struct PacketLossSummary {
size_t num_packets = 0;
size_t num_lost_packets = 0;
Timestamp base_time = Timestamp::MinusInfinity();
};
float GetHighestSeqNumber(const webrtc::rtcp::ReportBlock& block) {
return block.extended_high_seq_num();
}
float GetFractionLost(const webrtc::rtcp::ReportBlock& block) {
return static_cast<double>(block.fraction_lost()) / 256 * 100;
}
float GetCumulativeLost(const webrtc::rtcp::ReportBlock& block) {
return block.cumulative_lost();
}
float DelaySinceLastSr(const webrtc::rtcp::ReportBlock& block) {
return static_cast<double>(block.delay_since_last_sr()) / 65536;
}
} // namespace
EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
bool normalize_time)
: parsed_log_(log) {
config_.window_duration_ = TimeDelta::Millis(250);
config_.step_ = TimeDelta::Millis(10);
if (!log.start_log_events().empty()) {
config_.rtc_to_utc_offset_ = log.start_log_events()[0].utc_time() -
log.start_log_events()[0].log_time();
}
config_.normalize_time_ = normalize_time;
config_.begin_time_ = parsed_log_.first_timestamp();
config_.end_time_ = parsed_log_.last_timestamp();
if (config_.end_time_ < config_.begin_time_) {
RTC_LOG(LS_WARNING) << "No useful events in the log.";
config_.begin_time_ = config_.end_time_ = Timestamp::Zero();
}
RTC_LOG(LS_INFO) << "Log is "
<< (parsed_log_.last_timestamp().ms() -
parsed_log_.first_timestamp().ms()) /
1000
<< " seconds long.";
}
EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
const AnalyzerConfig& config)
: parsed_log_(log), config_(config) {
RTC_LOG(LS_INFO) << "Log is "
<< (parsed_log_.last_timestamp().ms() -
parsed_log_.first_timestamp().ms()) /
1000
<< " seconds long.";
}
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 EventLogAnalyzer::InitializeMapOfNamedGraphs(bool show_detector_state,
bool show_alr_state,
bool show_link_capacity) {
plots_.RegisterPlot("incoming_packet_sizes", [this](Plot* plot) {
this->CreatePacketGraph(webrtc::kIncomingPacket, plot);
});
plots_.RegisterPlot("outgoing_packet_sizes", [this](Plot* plot) {
this->CreatePacketGraph(webrtc::kOutgoingPacket, plot);
});
plots_.RegisterPlot("incoming_rtcp_types", [this](Plot* plot) {
this->CreateRtcpTypeGraph(webrtc::kIncomingPacket, plot);
});
plots_.RegisterPlot("outgoing_rtcp_types", [this](Plot* plot) {
this->CreateRtcpTypeGraph(webrtc::kOutgoingPacket, plot);
});
plots_.RegisterPlot("incoming_packet_count", [this](Plot* plot) {
this->CreateAccumulatedPacketsGraph(webrtc::kIncomingPacket, plot);
});
plots_.RegisterPlot("outgoing_packet_count", [this](Plot* plot) {
this->CreateAccumulatedPacketsGraph(webrtc::kOutgoingPacket, plot);
});
plots_.RegisterPlot("incoming_packet_rate", [this](Plot* plot) {
this->CreatePacketRateGraph(webrtc::kIncomingPacket, plot);
});
plots_.RegisterPlot("outgoing_packet_rate", [this](Plot* plot) {
this->CreatePacketRateGraph(webrtc::kOutgoingPacket, plot);
});
plots_.RegisterPlot("total_incoming_packet_rate", [this](Plot* plot) {
this->CreateTotalPacketRateGraph(webrtc::kIncomingPacket, plot);
});
plots_.RegisterPlot("total_outgoing_packet_rate", [this](Plot* plot) {
this->CreateTotalPacketRateGraph(webrtc::kOutgoingPacket, plot);
});
plots_.RegisterPlot("audio_playout",
[this](Plot* plot) { this->CreatePlayoutGraph(plot); });
plots_.RegisterPlot("neteq_set_minimum_delay", [this](Plot* plot) {
this->CreateNetEqSetMinimumDelay(plot);
});
plots_.RegisterPlot("incoming_audio_level", [this](Plot* plot) {
this->CreateAudioLevelGraph(webrtc::kIncomingPacket, plot);
});
plots_.RegisterPlot("outgoing_audio_level", [this](Plot* plot) {
this->CreateAudioLevelGraph(webrtc::kOutgoingPacket, plot);
});
plots_.RegisterPlot("incoming_sequence_number_delta", [this](Plot* plot) {
this->CreateSequenceNumberGraph(plot);
});
plots_.RegisterPlot("incoming_delay", [this](Plot* plot) {
this->CreateIncomingDelayGraph(plot);
});
plots_.RegisterPlot("incoming_loss_rate", [this](Plot* plot) {
this->CreateIncomingPacketLossGraph(plot);
});
plots_.RegisterPlot("incoming_bitrate", [this](Plot* plot) {
this->CreateTotalIncomingBitrateGraph(plot);
});
plots_.RegisterPlot(
"outgoing_bitrate", [this, show_detector_state, show_alr_state,
show_link_capacity](Plot* plot) {
this->CreateTotalOutgoingBitrateGraph(
plot, show_detector_state, show_alr_state, show_link_capacity);
});
plots_.RegisterPlot("incoming_stream_bitrate", [this](Plot* plot) {
this->CreateStreamBitrateGraph(webrtc::kIncomingPacket, plot);
});
plots_.RegisterPlot("outgoing_stream_bitrate", [this](Plot* plot) {
this->CreateStreamBitrateGraph(webrtc::kOutgoingPacket, plot);
});
plots_.RegisterPlot("incoming_layer_bitrate_allocation", [this](Plot* plot) {
this->CreateBitrateAllocationGraph(webrtc::kIncomingPacket, plot);
});
plots_.RegisterPlot("outgoing_layer_bitrate_allocation", [this](Plot* plot) {
this->CreateBitrateAllocationGraph(webrtc::kOutgoingPacket, plot);
});
plots_.RegisterPlot("simulated_receiveside_bwe", [this](Plot* plot) {
this->CreateReceiveSideBweSimulationGraph(plot);
});
plots_.RegisterPlot("simulated_sendside_bwe", [this](Plot* plot) {
this->CreateSendSideBweSimulationGraph(plot);
});
plots_.RegisterPlot("simulated_goog_cc", [this](Plot* plot) {
this->CreateGoogCcSimulationGraph(plot);
});
plots_.RegisterPlot("outgoing_twcc_loss", [this](Plot* plot) {
this->CreateOutgoingTWCCLossRateGraph(plot);
});
plots_.RegisterPlot("network_delay_feedback", [this](Plot* plot) {
this->CreateNetworkDelayFeedbackGraph(plot);
});
plots_.RegisterPlot("fraction_loss_feedback", [this](Plot* plot) {
this->CreateFractionLossGraph(plot);
});
plots_.RegisterPlot("incoming_timestamps", [this](Plot* plot) {
this->CreateTimestampGraph(webrtc::kIncomingPacket, plot);
});
plots_.RegisterPlot("outgoing_timestamps", [this](Plot* plot) {
this->CreateTimestampGraph(webrtc::kOutgoingPacket, plot);
});
plots_.RegisterPlot("incoming_rtcp_fraction_lost", [this](Plot* plot) {
this->CreateSenderAndReceiverReportPlot(
webrtc::kIncomingPacket, GetFractionLost,
"Fraction lost (incoming RTCP)", "Loss rate (percent)", plot);
});
plots_.RegisterPlot("outgoing_rtcp_fraction_lost", [this](Plot* plot) {
this->CreateSenderAndReceiverReportPlot(
webrtc::kOutgoingPacket, GetFractionLost,
"Fraction lost (outgoing RTCP)", "Loss rate (percent)", plot);
});
plots_.RegisterPlot("incoming_rtcp_cumulative_lost", [this](Plot* plot) {
this->CreateSenderAndReceiverReportPlot(
webrtc::kIncomingPacket, GetCumulativeLost,
"Cumulative lost packets (incoming RTCP)", "Packets", plot);
});
plots_.RegisterPlot("outgoing_rtcp_cumulative_lost", [this](Plot* plot) {
this->CreateSenderAndReceiverReportPlot(
webrtc::kOutgoingPacket, GetCumulativeLost,
"Cumulative lost packets (outgoing RTCP)", "Packets", plot);
});
plots_.RegisterPlot("incoming_rtcp_highest_seq_number", [this](Plot* plot) {
this->CreateSenderAndReceiverReportPlot(
webrtc::kIncomingPacket, GetHighestSeqNumber,
"Highest sequence number (incoming RTCP)", "Sequence number", plot);
});
plots_.RegisterPlot("outgoing_rtcp_highest_seq_number", [this](Plot* plot) {
this->CreateSenderAndReceiverReportPlot(
webrtc::kOutgoingPacket, GetHighestSeqNumber,
"Highest sequence number (outgoing RTCP)", "Sequence number", plot);
});
plots_.RegisterPlot("incoming_rtcp_delay_since_last_sr", [this](Plot* plot) {
this->CreateSenderAndReceiverReportPlot(
webrtc::kIncomingPacket, DelaySinceLastSr,
"Delay since last received sender report (incoming RTCP)", "Time (s)",
plot);
});
plots_.RegisterPlot("outgoing_rtcp_delay_since_last_sr", [this](Plot* plot) {
this->CreateSenderAndReceiverReportPlot(
webrtc::kOutgoingPacket, DelaySinceLastSr,
"Delay since last received sender report (outgoing RTCP)", "Time (s)",
plot);
});
plots_.RegisterPlot(
"pacer_delay", [this](Plot* plot) { this->CreatePacerDelayGraph(plot); });
plots_.RegisterPlot("audio_encoder_bitrate", [this](Plot* plot) {
CreateAudioEncoderTargetBitrateGraph(this->parsed_log_, this->config_,
plot);
});
plots_.RegisterPlot("audio_encoder_frame_length", [this](Plot* plot) {
CreateAudioEncoderFrameLengthGraph(this->parsed_log_, this->config_, plot);
});
plots_.RegisterPlot("audio_encoder_packet_loss", [this](Plot* plot) {
CreateAudioEncoderPacketLossGraph(this->parsed_log_, this->config_, plot);
});
plots_.RegisterPlot("audio_encoder_fec", [this](Plot* plot) {
CreateAudioEncoderEnableFecGraph(this->parsed_log_, this->config_, plot);
});
plots_.RegisterPlot("audio_encoder_dtx", [this](Plot* plot) {
CreateAudioEncoderEnableDtxGraph(this->parsed_log_, this->config_, plot);
});
plots_.RegisterPlot("audio_encoder_num_channels", [this](Plot* plot) {
CreateAudioEncoderNumChannelsGraph(this->parsed_log_, this->config_, plot);
});
plots_.RegisterPlot("ice_candidate_pair_config", [this](Plot* plot) {
this->CreateIceCandidatePairConfigGraph(plot);
});
plots_.RegisterPlot("ice_connectivity_check", [this](Plot* plot) {
this->CreateIceConnectivityCheckGraph(plot);
});
plots_.RegisterPlot("dtls_transport_state", [this](Plot* plot) {
this->CreateDtlsTransportStateGraph(plot);
});
plots_.RegisterPlot("dtls_writable_state", [this](Plot* plot) {
this->CreateDtlsWritableStateGraph(plot);
});
}
void EventLogAnalyzer::CreateGraphsByName(const std::vector<std::string>& names,
PlotCollection* collection) {
for (const auto& plot : plots_) {
if (absl::c_find(names, plot.label) != names.end()) {
Plot* output = collection->AppendNewPlot(plot.label);
plot.plot_func(output);
}
}
}
void EventLogAnalyzer::CreatePacketGraph(PacketDirection direction,
Plot* plot) {
for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
// Filter on SSRC.
if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
continue;
}
TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
LineStyle::kBar);
auto GetPacketSize = [](const LoggedRtpPacket& packet) {
return absl::optional<float>(packet.total_length);
};
auto ToCallTime = [this](const LoggedRtpPacket& packet) {
return this->config_.GetCallTimeSec(packet.timestamp);
};
ProcessPoints<LoggedRtpPacket>(ToCallTime, GetPacketSize,
stream.packet_view, &time_series);
plot->AppendTimeSeries(std::move(time_series));
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Packet size (bytes)", kBottomMargin,
kTopMargin);
plot->SetTitle(GetDirectionAsString(direction) + " RTP packets");
}
void EventLogAnalyzer::CreateRtcpTypeGraph(PacketDirection direction,
Plot* plot) {
plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
parsed_log_.transport_feedbacks(direction), config_, "TWCC", 1));
plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
parsed_log_.receiver_reports(direction), config_, "RR", 2));
plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
parsed_log_.sender_reports(direction), config_, "SR", 3));
plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(
parsed_log_.extended_reports(direction), config_, "XR", 4));
plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(parsed_log_.nacks(direction),
config_, "NACK", 5));
plot->AppendTimeSeries(CreateRtcpTypeTimeSeries(parsed_log_.rembs(direction),
config_, "REMB", 6));
plot->AppendTimeSeries(
CreateRtcpTypeTimeSeries(parsed_log_.firs(direction), config_, "FIR", 7));
plot->AppendTimeSeries(
CreateRtcpTypeTimeSeries(parsed_log_.plis(direction), config_, "PLI", 8));
plot->AppendTimeSeries(
CreateRtcpTypeTimeSeries(parsed_log_.byes(direction), config_, "BYE", 9));
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "RTCP type", kBottomMargin, kTopMargin);
plot->SetTitle(GetDirectionAsString(direction) + " RTCP packets");
plot->SetYAxisTickLabels({{1, "TWCC"},
{2, "RR"},
{3, "SR"},
{4, "XR"},
{5, "NACK"},
{6, "REMB"},
{7, "FIR"},
{8, "PLI"},
{9, "BYE"}});
}
template <typename IterableType>
void EventLogAnalyzer::CreateAccumulatedPacketsTimeSeries(
Plot* plot,
const IterableType& packets,
const std::string& label) {
TimeSeries time_series(label, LineStyle::kStep);
for (size_t i = 0; i < packets.size(); i++) {
float x = config_.GetCallTimeSec(packets[i].log_time());
time_series.points.emplace_back(x, i + 1);
}
plot->AppendTimeSeries(std::move(time_series));
}
void EventLogAnalyzer::CreateAccumulatedPacketsGraph(PacketDirection direction,
Plot* plot) {
for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
if (!MatchingSsrc(stream.ssrc, desired_ssrc_))
continue;
std::string label = std::string("RTP ") +
GetStreamName(parsed_log_, direction, stream.ssrc);
CreateAccumulatedPacketsTimeSeries(plot, stream.packet_view, label);
}
std::string label =
std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")";
if (direction == kIncomingPacket) {
CreateAccumulatedPacketsTimeSeries(
plot, parsed_log_.incoming_rtcp_packets(), label);
} else {
CreateAccumulatedPacketsTimeSeries(
plot, parsed_log_.outgoing_rtcp_packets(), label);
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Received Packets", kBottomMargin, kTopMargin);
plot->SetTitle(std::string("Accumulated ") + GetDirectionAsString(direction) +
" RTP/RTCP packets");
}
void EventLogAnalyzer::CreatePacketRateGraph(PacketDirection direction,
Plot* plot) {
auto CountPackets = [](auto packet) { return 1.0; };
for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
// Filter on SSRC.
if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
continue;
}
TimeSeries time_series(
std::string("RTP ") +
GetStreamName(parsed_log_, direction, stream.ssrc),
LineStyle::kLine);
MovingAverage<LoggedRtpPacket, double>(CountPackets, stream.packet_view,
config_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
}
TimeSeries time_series(
std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")",
LineStyle::kLine);
if (direction == kIncomingPacket) {
MovingAverage<LoggedRtcpPacketIncoming, double>(
CountPackets, parsed_log_.incoming_rtcp_packets(), config_,
&time_series);
} else {
MovingAverage<LoggedRtcpPacketOutgoing, double>(
CountPackets, parsed_log_.outgoing_rtcp_packets(), config_,
&time_series);
}
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
kTopMargin);
plot->SetTitle("Rate of " + GetDirectionAsString(direction) +
" RTP/RTCP packets");
}
void EventLogAnalyzer::CreateTotalPacketRateGraph(PacketDirection direction,
Plot* plot) {
// Contains a log timestamp to enable counting logged events of different
// types using MovingAverage().
class LogTime {
public:
explicit LogTime(Timestamp log_time) : log_time_(log_time) {}
Timestamp log_time() const { return log_time_; }
private:
Timestamp log_time_;
};
std::vector<LogTime> packet_times;
auto handle_rtp = [&packet_times](const LoggedRtpPacket& packet) {
packet_times.emplace_back(packet.log_time());
};
RtcEventProcessor process;
for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
process.AddEvents(stream.packet_view, handle_rtp, direction);
}
if (direction == kIncomingPacket) {
auto handle_incoming_rtcp =
[&packet_times](const LoggedRtcpPacketIncoming& packet) {
packet_times.emplace_back(packet.log_time());
};
process.AddEvents(parsed_log_.incoming_rtcp_packets(),
handle_incoming_rtcp);
} else {
auto handle_outgoing_rtcp =
[&packet_times](const LoggedRtcpPacketOutgoing& packet) {
packet_times.emplace_back(packet.log_time());
};
process.AddEvents(parsed_log_.outgoing_rtcp_packets(),
handle_outgoing_rtcp);
}
process.ProcessEventsInOrder();
TimeSeries time_series(std::string("Total ") + "(" +
GetDirectionAsShortString(direction) + ") packets",
LineStyle::kLine);
MovingAverage<LogTime, uint64_t>([](auto packet) { return 1; }, packet_times,
config_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Packet Rate (packets/s)", kBottomMargin,
kTopMargin);
plot->SetTitle("Rate of all " + GetDirectionAsString(direction) +
" RTP/RTCP packets");
}
// For each SSRC, plot the time between the consecutive playouts.
void EventLogAnalyzer::CreatePlayoutGraph(Plot* plot) {
for (const auto& playout_stream : parsed_log_.audio_playout_events()) {
uint32_t ssrc = playout_stream.first;
if (!MatchingSsrc(ssrc, desired_ssrc_))
continue;
absl::optional<int64_t> last_playout_ms;
TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
for (const auto& playout_event : playout_stream.second) {
float x = config_.GetCallTimeSec(playout_event.log_time());
int64_t playout_time_ms = playout_event.log_time_ms();
// If there were no previous playouts, place the point on the x-axis.
float y = playout_time_ms - last_playout_ms.value_or(playout_time_ms);
time_series.points.push_back(TimeSeriesPoint(x, y));
last_playout_ms.emplace(playout_time_ms);
}
plot->AppendTimeSeries(std::move(time_series));
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Time since last playout (ms)", kBottomMargin,
kTopMargin);
plot->SetTitle("Audio playout");
}
void EventLogAnalyzer::CreateNetEqSetMinimumDelay(Plot* plot) {
for (const auto& playout_stream :
parsed_log_.neteq_set_minimum_delay_events()) {
uint32_t ssrc = playout_stream.first;
if (!MatchingSsrc(ssrc, desired_ssrc_))
continue;
TimeSeries time_series(SsrcToString(ssrc), LineStyle::kStep,
PointStyle::kHighlight);
for (const auto& event : playout_stream.second) {
float x = config_.GetCallTimeSec(event.log_time());
float y = event.minimum_delay_ms;
time_series.points.push_back(TimeSeriesPoint(x, y));
}
plot->AppendTimeSeries(std::move(time_series));
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1000, "Minimum Delay (ms)", kBottomMargin,
kTopMargin);
plot->SetTitle("Set Minimum Delay");
}
// For audio SSRCs, plot the audio level.
void EventLogAnalyzer::CreateAudioLevelGraph(PacketDirection direction,
Plot* plot) {
for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
if (!IsAudioSsrc(parsed_log_, direction, stream.ssrc))
continue;
TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
LineStyle::kLine);
for (auto& packet : stream.packet_view) {
if (packet.header.extension.audio_level()) {
float x = config_.GetCallTimeSec(packet.log_time());
// The audio level is stored in -dBov (so e.g. -10 dBov is stored as 10)
// Here we convert it to dBov.
float y =
static_cast<float>(-packet.header.extension.audio_level()->level());
time_series.points.emplace_back(TimeSeriesPoint(x, y));
}
}
plot->AppendTimeSeries(std::move(time_series));
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetYAxis(-127, 0, "Audio level (dBov)", kBottomMargin, kTopMargin);
plot->SetTitle(GetDirectionAsString(direction) + " audio level");
}
// For each SSRC, plot the sequence number difference between consecutive
// incoming packets.
void EventLogAnalyzer::CreateSequenceNumberGraph(Plot* plot) {
for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
// Filter on SSRC.
if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
continue;
}
TimeSeries time_series(
GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc),
LineStyle::kBar);
auto GetSequenceNumberDiff = [](const LoggedRtpPacketIncoming& old_packet,
const LoggedRtpPacketIncoming& new_packet) {
int64_t diff =
WrappingDifference(new_packet.rtp.header.sequenceNumber,
old_packet.rtp.header.sequenceNumber, 1ul << 16);
return diff;
};
auto ToCallTime = [this](const LoggedRtpPacketIncoming& packet) {
return this->config_.GetCallTimeSec(packet.log_time());
};
ProcessPairs<LoggedRtpPacketIncoming, float>(
ToCallTime, GetSequenceNumberDiff, stream.incoming_packets,
&time_series);
plot->AppendTimeSeries(std::move(time_series));
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Difference since last packet", kBottomMargin,
kTopMargin);
plot->SetTitle("Incoming sequence number delta");
}
void EventLogAnalyzer::CreateIncomingPacketLossGraph(Plot* plot) {
for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
const std::vector<LoggedRtpPacketIncoming>& packets =
stream.incoming_packets;
// Filter on SSRC.
if (!MatchingSsrc(stream.ssrc, desired_ssrc_) || packets.empty()) {
continue;
}
TimeSeries time_series(
GetStreamName(parsed_log_, kIncomingPacket, stream.ssrc),
LineStyle::kLine, PointStyle::kHighlight);
// TODO(terelius): Should the window and step size be read from the class
// instead?
const TimeDelta kWindow = TimeDelta::Millis(1000);
const TimeDelta kStep = TimeDelta::Millis(1000);
SeqNumUnwrapper<uint16_t> unwrapper_;
SeqNumUnwrapper<uint16_t> prior_unwrapper_;
size_t window_index_begin = 0;
size_t window_index_end = 0;
uint64_t highest_seq_number =
unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
uint64_t highest_prior_seq_number =
prior_unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
for (Timestamp t = config_.begin_time_; t < config_.end_time_ + kStep;
t += kStep) {
while (window_index_end < packets.size() &&
packets[window_index_end].rtp.log_time() < t) {
uint64_t sequence_number = unwrapper_.Unwrap(
packets[window_index_end].rtp.header.sequenceNumber);
highest_seq_number = std::max(highest_seq_number, sequence_number);
++window_index_end;
}
while (window_index_begin < packets.size() &&
packets[window_index_begin].rtp.log_time() < t - kWindow) {
uint64_t sequence_number = prior_unwrapper_.Unwrap(
packets[window_index_begin].rtp.header.sequenceNumber);
highest_prior_seq_number =
std::max(highest_prior_seq_number, sequence_number);
++window_index_begin;
}
float x = config_.GetCallTimeSec(t);
uint64_t expected_packets = highest_seq_number - highest_prior_seq_number;
if (expected_packets > 0) {
int64_t received_packets = window_index_end - window_index_begin;
int64_t lost_packets = expected_packets - received_packets;
float y = static_cast<float>(lost_packets) / expected_packets * 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, 1, "Loss rate (in %)", kBottomMargin, kTopMargin);
plot->SetTitle("Incoming packet loss (derived from incoming packets)");
}
void EventLogAnalyzer::CreateIncomingDelayGraph(Plot* plot) {
for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
// Filter on SSRC.
if (!MatchingSsrc(stream.ssrc, 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();
absl::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 = [this](const LoggedRtpPacketIncoming& packet) {
return this->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 EventLogAnalyzer::CreateFractionLossGraph(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 EventLogAnalyzer::CreateTotalIncomingBitrateGraph(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));
if (!parsed_log_.generic_packets_received().empty()) {
TimeSeries time_series("Incoming generic bitrate", LineStyle::kLine);
auto GetPacketSizeKilobits = [](const LoggedGenericPacketReceived& packet) {
return packet.packet_length * 8.0 / 1000.0;
};
MovingAverage<LoggedGenericPacketReceived, double>(
GetPacketSizeKilobits, parsed_log_.generic_packets_received(), config_,
&time_series);
plot->AppendTimeSeries(std::move(time_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 EventLogAnalyzer::CreateTotalOutgoingBitrateGraph(
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 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->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));
if (!parsed_log_.generic_packets_sent().empty()) {
{
TimeSeries time_series("Outgoing generic total bitrate",
LineStyle::kLine);
auto GetPacketSizeKilobits = [](const LoggedGenericPacketSent& packet) {
return packet.packet_length() * 8.0 / 1000.0;
};
MovingAverage<LoggedGenericPacketSent, double>(
GetPacketSizeKilobits, parsed_log_.generic_packets_sent(), config_,
&time_series);
plot->AppendTimeSeries(std::move(time_series));
}
{
TimeSeries time_series("Outgoing generic payload bitrate",
LineStyle::kLine);
auto GetPacketSizeKilobits = [](const LoggedGenericPacketSent& packet) {
return packet.payload_length * 8.0 / 1000.0;
};
MovingAverage<LoggedGenericPacketSent, double>(
GetPacketSizeKilobits, parsed_log_.generic_packets_sent(), config_,
&time_series);
plot->AppendTimeSeries(std::move(time_series));
}
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
plot->SetTitle("Outgoing RTP bitrate");
}
// For each SSRC, plot the bandwidth used by that stream.
void EventLogAnalyzer::CreateStreamBitrateGraph(PacketDirection direction,
Plot* plot) {
for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
// Filter on SSRC.
if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
continue;
}
TimeSeries time_series(GetStreamName(parsed_log_, direction, stream.ssrc),
LineStyle::kLine);
auto GetPacketSizeKilobits = [](const LoggedRtpPacket& packet) {
return packet.total_length * 8.0 / 1000.0;
};
MovingAverage<LoggedRtpPacket, double>(
GetPacketSizeKilobits, stream.packet_view, config_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
plot->SetTitle(GetDirectionAsString(direction) + " bitrate per stream");
}
// Plot the bitrate allocation for each temporal and spatial layer.
// Computed from RTCP XR target bitrate block, so the graph is only populated if
// those are sent.
void EventLogAnalyzer::CreateBitrateAllocationGraph(PacketDirection direction,
Plot* plot) {
std::map<LayerDescription, TimeSeries> time_series;
const auto& xr_list = parsed_log_.extended_reports(direction);
for (const auto& rtcp : xr_list) {
const absl::optional<rtcp::TargetBitrate>& target_bitrate =
rtcp.xr.target_bitrate();
if (!target_bitrate.has_value())
continue;
for (const auto& bitrate_item : target_bitrate->GetTargetBitrates()) {
LayerDescription layer(rtcp.xr.sender_ssrc(), bitrate_item.spatial_layer,
bitrate_item.temporal_layer);
auto time_series_it = time_series.find(layer);
if (time_series_it == time_series.end()) {
std::string layer_name = GetLayerName(layer);
bool inserted;
std::tie(time_series_it, inserted) = time_series.insert(
std::make_pair(layer, TimeSeries(layer_name, LineStyle::kStep)));
RTC_DCHECK(inserted);
}
float x = config_.GetCallTimeSec(rtcp.log_time());
float y = bitrate_item.target_bitrate_kbps;
time_series_it->second.points.emplace_back(x, y);
}
}
for (auto& layer : time_series) {
plot->AppendTimeSeries(std::move(layer.second));
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
if (direction == kIncomingPacket)
plot->SetTitle("Target bitrate per incoming layer");
else
plot->SetTitle("Target bitrate per outgoing layer");
}
void EventLogAnalyzer::CreateGoogCcSimulationGraph(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(
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 EventLogAnalyzer::CreateOutgoingTWCCLossRateGraph(Plot* plot) {
TimeSeries loss_rate_series("Loss rate (from packet feedback)",
LineStyle::kLine, PointStyle::kHighlight);
TimeSeries average_loss_rate_series("Average loss rate last 5s",
LineStyle::kLine, PointStyle::kHighlight);
TimeSeries missing_feedback_series("Missing feedback", LineStyle::kNone,
PointStyle::kHighlight);
PacketLossSummary window_summary;
Timestamp last_observation_receive_time = Timestamp::Zero();
// Use loss based bwe 2 observation duration and observation window size.
constexpr TimeDelta kObservationDuration = TimeDelta::Millis(250);
constexpr uint32_t kObservationWindowSize = 20;
std::deque<PacketLossSummary> observations;
SeqNumUnwrapper<uint16_t> unwrapper;
int64_t last_acked = 1;
if (!parsed_log_.transport_feedbacks(kIncomingPacket).empty()) {
last_acked =
unwrapper.Unwrap(parsed_log_.transport_feedbacks(kIncomingPacket)[0]
.transport_feedback.GetBaseSequence());
}
for (auto& feedback : parsed_log_.transport_feedbacks(kIncomingPacket)) {
const rtcp::TransportFeedback& transport_feedback =
feedback.transport_feedback;
size_t base_seq_num =
unwrapper.Unwrap(transport_feedback.GetBaseSequence());
// Collect packets that do not have feedback, which are from the last acked
// packet, to the current base packet.
for (size_t seq_num = last_acked; seq_num < base_seq_num; ++seq_num) {
missing_feedback_series.points.emplace_back(
config_.GetCallTimeSec(feedback.timestamp),
100 + seq_num - last_acked);
}
last_acked = base_seq_num + transport_feedback.GetPacketStatusCount();
// Compute loss rate from the transport feedback.
auto loss_rate =
static_cast<float>((transport_feedback.GetPacketStatusCount() -
transport_feedback.GetReceivedPackets().size()) *
100.0 / transport_feedback.GetPacketStatusCount());
loss_rate_series.points.emplace_back(
config_.GetCallTimeSec(feedback.timestamp), loss_rate);
// Compute loss rate in a window of kObservationWindowSize.
if (window_summary.num_packets == 0) {
window_summary.base_time = feedback.log_time();
}
window_summary.num_packets += transport_feedback.GetPacketStatusCount();
window_summary.num_lost_packets +=
transport_feedback.GetPacketStatusCount() -
transport_feedback.GetReceivedPackets().size();
const Timestamp last_received_time = feedback.log_time();
const TimeDelta observation_duration =
window_summary.base_time == Timestamp::Zero()
? TimeDelta::Zero()
: last_received_time - window_summary.base_time;
if (observation_duration > kObservationDuration) {
last_observation_receive_time = last_received_time;
observations.push_back(window_summary);
if (observations.size() > kObservationWindowSize) {
observations.pop_front();
}
// Compute average loss rate in a number of windows.
int total_packets = 0;
int total_loss = 0;
for (const auto& observation : observations) {
total_loss += observation.num_lost_packets;
total_packets += observation.num_packets;
}
if (total_packets > 0) {
float average_loss_rate = total_loss * 100.0 / total_packets;
average_loss_rate_series.points.emplace_back(
config_.GetCallTimeSec(feedback.timestamp), average_loss_rate);
} else {
average_loss_rate_series.points.emplace_back(
config_.GetCallTimeSec(feedback.timestamp), 0);
}
window_summary = PacketLossSummary();
}
}
// Add the data set to the plot.
plot->AppendTimeSeriesIfNotEmpty(std::move(loss_rate_series));
plot->AppendTimeSeriesIfNotEmpty(std::move(average_loss_rate_series));
plot->AppendTimeSeriesIfNotEmpty(std::move(missing_feedback_series));
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 100, "Loss rate (percent)", kBottomMargin,
kTopMargin);
plot->SetTitle("Outgoing loss rate (from TWCC feedback)");
}
void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
using RtpPacketType = LoggedRtpPacketOutgoing;
using TransportFeedbackType = LoggedRtcpPacketTransportFeedback;
// TODO(terelius): This could be provided by the parser.
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;
RtcEventLogNull null_event_log;
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;
cc_config.constraints.at_time = Timestamp::Micros(clock.TimeInMicroseconds());
cc_config.constraints.starting_rate =
DataRate::BitsPerSec(kDefaultStartBitrateBps);
cc_config.event_log = &null_event_log;
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);
test::ExplicitKeyValueConfig throughput_config(
"WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:true/");
std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
robust_throughput_estimator(
AcknowledgedBitrateEstimatorInterface::Create(&throughput_config));
test::ExplicitKeyValueConfig 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) {
RtpPacketSendInfo packet_info;
packet_info.media_ssrc = rtp_packet.rtp.header.ssrc;
packet_info.transport_sequence_number =
rtp_packet.rtp.header.extension.transportSequenceNumber;
packet_info.rtp_sequence_number = rtp_packet.rtp.header.sequenceNumber;
packet_info.length = 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)) {
packet_info.packet_type = RtpPacketMediaType::kVideo;
} else if (IsAudioSsrc(parsed_log_, PacketDirection::kOutgoingPacket,
rtp_packet.rtp.header.ssrc)) {
packet_info.packet_type = RtpPacketMediaType::kAudio;
}
transport_feedback.AddPacket(
packet_info,
0u, // Per packet overhead bytes.
Timestamp::Micros(rtp_packet.rtp.log_time_us()));
}
rtc::SentPacket 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,
Timestamp::Millis(clock.TimeInMilliseconds()));
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());
}
absl::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);
}
absl::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);
}
absl::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 = Timestamp::Micros(clock.TimeInMicroseconds());
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 EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(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);
RembInterceptor remb_interceptor;
ReceiveSideCongestionController rscc(
CreateEnvironment(&clock), [](auto...) {},
absl::bind_front(&RembInterceptor::SendRemb, &remb_interceptor), nullptr);
// 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);
absl::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 EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(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 EventLogAnalyzer::CreatePacerDelayGraph(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();
absl::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.)");
}
void EventLogAnalyzer::CreateTimestampGraph(PacketDirection direction,
Plot* plot) {
for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
TimeSeries rtp_timestamps(
GetStreamName(parsed_log_, direction, stream.ssrc) + " capture-time",
LineStyle::kLine, PointStyle::kHighlight);
for (const auto& packet : stream.packet_view) {
float x = config_.GetCallTimeSec(packet.log_time());
float y = packet.header.timestamp;
rtp_timestamps.points.emplace_back(x, y);
}
plot->AppendTimeSeries(std::move(rtp_timestamps));
TimeSeries rtcp_timestamps(
GetStreamName(parsed_log_, direction, stream.ssrc) +
" rtcp capture-time",
LineStyle::kLine, PointStyle::kHighlight);
// TODO(terelius): Why only sender reports?
const auto& sender_reports = parsed_log_.sender_reports(direction);
for (const auto& rtcp : sender_reports) {
if (rtcp.sr.sender_ssrc() != stream.ssrc)
continue;
float x = config_.GetCallTimeSec(rtcp.log_time());
float y = rtcp.sr.rtp_timestamp();
rtcp_timestamps.points.emplace_back(x, y);
}
plot->AppendTimeSeriesIfNotEmpty(std::move(rtcp_timestamps));
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "RTP timestamp", kBottomMargin, kTopMargin);
plot->SetTitle(GetDirectionAsString(direction) + " timestamps");
}
void EventLogAnalyzer::CreateSenderAndReceiverReportPlot(
PacketDirection direction,
rtc::FunctionView<float(const rtcp::ReportBlock&)> fy,
std::string title,
std::string yaxis_label,
Plot* plot) {
std::map<uint32_t, TimeSeries> sr_reports_by_ssrc;
const auto& sender_reports = parsed_log_.sender_reports(direction);
for (const auto& rtcp : sender_reports) {
float x = config_.GetCallTimeSec(rtcp.log_time());
uint32_t ssrc = rtcp.sr.sender_ssrc();
for (const auto& block : rtcp.sr.report_blocks()) {
float y = fy(block);
auto sr_report_it = sr_reports_by_ssrc.find(ssrc);
bool inserted;
if (sr_report_it == sr_reports_by_ssrc.end()) {
std::tie(sr_report_it, inserted) = sr_reports_by_ssrc.emplace(
ssrc, TimeSeries(GetStreamName(parsed_log_, direction, ssrc) +
" Sender Reports",
LineStyle::kLine, PointStyle::kHighlight));
}
sr_report_it->second.points.emplace_back(x, y);
}
}
for (auto& kv : sr_reports_by_ssrc) {
plot->AppendTimeSeries(std::move(kv.second));
}
std::map<uint32_t, TimeSeries> rr_reports_by_ssrc;
const auto& receiver_reports = parsed_log_.receiver_reports(direction);
for (const auto& rtcp : receiver_reports) {
float x = config_.GetCallTimeSec(rtcp.log_time());
uint32_t ssrc = rtcp.rr.sender_ssrc();
for (const auto& block : rtcp.rr.report_blocks()) {
float y = fy(block);
auto rr_report_it = rr_reports_by_ssrc.find(ssrc);
bool inserted;
if (rr_report_it == rr_reports_by_ssrc.end()) {
std::tie(rr_report_it, inserted) = rr_reports_by_ssrc.emplace(
ssrc, TimeSeries(GetStreamName(parsed_log_, direction, ssrc) +
" Receiver Reports",
LineStyle::kLine, PointStyle::kHighlight));
}
rr_report_it->second.points.emplace_back(x, y);
}
}
for (auto& kv : rr_reports_by_ssrc) {
plot->AppendTimeSeries(std::move(kv.second));
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, yaxis_label, kBottomMargin, kTopMargin);
plot->SetTitle(title);
}
void EventLogAnalyzer::CreateIceCandidatePairConfigGraph(Plot* plot) {
std::map<uint32_t, TimeSeries> configs_by_cp_id;
for (const auto& config : parsed_log_.ice_candidate_pair_configs()) {
if (configs_by_cp_id.find(config.candidate_pair_id) ==
configs_by_cp_id.end()) {
const std::string candidate_pair_desc =
GetCandidatePairLogDescriptionAsString(config);
configs_by_cp_id[config.candidate_pair_id] =
TimeSeries("[" + std::to_string(config.candidate_pair_id) + "]" +
candidate_pair_desc,
LineStyle::kNone, PointStyle::kHighlight);
candidate_pair_desc_by_id_[config.candidate_pair_id] =
candidate_pair_desc;
}
float x = config_.GetCallTimeSec(config.log_time());
float y = static_cast<float>(config.type);
configs_by_cp_id[config.candidate_pair_id].points.emplace_back(x, y);
}
// TODO(qingsi): There can be a large number of candidate pairs generated by
// certain calls and the frontend cannot render the chart in this case due to
// the failure of generating a palette with the same number of colors.
for (auto& kv : configs_by_cp_id) {
plot->AppendTimeSeries(std::move(kv.second));
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 3, "Config Type", kBottomMargin, kTopMargin);
plot->SetTitle("[IceEventLog] ICE candidate pair configs");
plot->SetYAxisTickLabels(
{{static_cast<float>(IceCandidatePairConfigType::kAdded), "ADDED"},
{static_cast<float>(IceCandidatePairConfigType::kUpdated), "UPDATED"},
{static_cast<float>(IceCandidatePairConfigType::kDestroyed),
"DESTROYED"},
{static_cast<float>(IceCandidatePairConfigType::kSelected),
"SELECTED"}});
}
std::string EventLogAnalyzer::GetCandidatePairLogDescriptionFromId(
uint32_t candidate_pair_id) {
if (candidate_pair_desc_by_id_.find(candidate_pair_id) !=
candidate_pair_desc_by_id_.end()) {
return candidate_pair_desc_by_id_[candidate_pair_id];
}
for (const auto& config : parsed_log_.ice_candidate_pair_configs()) {
// TODO(qingsi): Add the handling of the "Updated" config event after the
// visualization of property change for candidate pairs is introduced.
if (candidate_pair_desc_by_id_.find(config.candidate_pair_id) ==
candidate_pair_desc_by_id_.end()) {
const std::string candidate_pair_desc =
GetCandidatePairLogDescriptionAsString(config);
candidate_pair_desc_by_id_[config.candidate_pair_id] =
candidate_pair_desc;
}
}
return candidate_pair_desc_by_id_[candidate_pair_id];
}
void EventLogAnalyzer::CreateIceConnectivityCheckGraph(Plot* plot) {
constexpr int kEventTypeOffset =
static_cast<int>(IceCandidatePairConfigType::kNumValues);
std::map<uint32_t, TimeSeries> checks_by_cp_id;
for (const auto& event : parsed_log_.ice_candidate_pair_events()) {
if (checks_by_cp_id.find(event.candidate_pair_id) ==
checks_by_cp_id.end()) {
checks_by_cp_id[event.candidate_pair_id] = TimeSeries(
"[" + std::to_string(event.candidate_pair_id) + "]" +
GetCandidatePairLogDescriptionFromId(event.candidate_pair_id),
LineStyle::kNone, PointStyle::kHighlight);
}
float x = config_.GetCallTimeSec(event.log_time());
float y = static_cast<float>(event.type) + kEventTypeOffset;
checks_by_cp_id[event.candidate_pair_id].points.emplace_back(x, y);
}
// TODO(qingsi): The same issue as in CreateIceCandidatePairConfigGraph.
for (auto& kv : checks_by_cp_id) {
plot->AppendTimeSeries(std::move(kv.second));
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 4, "Connectivity State", kBottomMargin,
kTopMargin);
plot->SetTitle("[IceEventLog] ICE connectivity checks");
plot->SetYAxisTickLabels(
{{static_cast<float>(IceCandidatePairEventType::kCheckSent) +
kEventTypeOffset,
"CHECK SENT"},
{static_cast<float>(IceCandidatePairEventType::kCheckReceived) +
kEventTypeOffset,
"CHECK RECEIVED"},
{static_cast<float>(IceCandidatePairEventType::kCheckResponseSent) +
kEventTypeOffset,
"RESPONSE SENT"},
{static_cast<float>(IceCandidatePairEventType::kCheckResponseReceived) +
kEventTypeOffset,
"RESPONSE RECEIVED"}});
}
void EventLogAnalyzer::CreateDtlsTransportStateGraph(Plot* plot) {
TimeSeries states("DTLS Transport State", LineStyle::kNone,
PointStyle::kHighlight);
for (const auto& event : parsed_log_.dtls_transport_states()) {
float x = config_.GetCallTimeSec(event.log_time());
float y = static_cast<float>(event.dtls_transport_state);
states.points.emplace_back(x, y);
}
plot->AppendTimeSeries(std::move(states));
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, static_cast<float>(DtlsTransportState::kNumValues),
"Transport State", kBottomMargin, kTopMargin);
plot->SetTitle("DTLS Transport State");
plot->SetYAxisTickLabels(
{{static_cast<float>(DtlsTransportState::kNew), "NEW"},
{static_cast<float>(DtlsTransportState::kConnecting), "CONNECTING"},
{static_cast<float>(DtlsTransportState::kConnected), "CONNECTED"},
{static_cast<float>(DtlsTransportState::kClosed), "CLOSED"},
{static_cast<float>(DtlsTransportState::kFailed), "FAILED"}});
}
void EventLogAnalyzer::CreateDtlsWritableStateGraph(Plot* plot) {
TimeSeries writable("DTLS Writable", LineStyle::kNone,
PointStyle::kHighlight);
for (const auto& event : parsed_log_.dtls_writable_states()) {
float x = config_.GetCallTimeSec(event.log_time());
float y = static_cast<float>(event.writable);
writable.points.emplace_back(x, y);
}
plot->AppendTimeSeries(std::move(writable));
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Writable", kBottomMargin, kTopMargin);
plot->SetTitle("DTLS Writable State");
}
} // namespace webrtc