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webrtc/rtc_tools/rtc_event_log_visualizer/analyzer.cc
Sebastian Jansson 74f96eccd6 Removes unused late feedback plot from analyzer. 
Due to changes in how the transport feedback is processed, the late
feedback results plot doesn't get any entries anymore.

Bug: webrtc:9883
Change-Id: I9df8e86a35bedddf78407128f0ab0b6b321a6f28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158668
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29643}
2019-10-29 18:48:55 +00:00

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/*
* 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 <limits>
#include <map>
#include <memory>
#include <string>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
#include "api/function_view.h"
#include "api/transport/field_trial_based_config.h"
#include "api/transport/goog_cc_factory.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "call/call.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
#include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
#include "modules/pacing/paced_sender.h"
#include "modules/pacing/packet_router.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet.h"
#include "modules/rtp_rtcp/source/rtcp_packet/common_header.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/sender_report.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "rtc_base/checks.h"
#include "rtc_base/format_macros.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/sequence_number_util.h"
#include "rtc_base/rate_statistics.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_tools/rtc_event_log_visualizer/log_simulation.h"
#ifndef BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
#define BWE_TEST_LOGGING_COMPILE_TIME_ENABLE 0
#endif // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
namespace webrtc {
namespace {
const int kNumMicrosecsPerSec = 1000000;
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 stardard RTP frequency.";
return absl::nullopt;
}
constexpr float kLeftMargin = 0.01f;
constexpr float kRightMargin = 0.02f;
constexpr float kBottomMargin = 0.02f;
constexpr float kTopMargin = 0.05f;
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;
}
// For each element in data_view, use |f()| to extract a y-coordinate and
// store the result in a TimeSeries.
template <typename DataType, typename IterableType>
void ProcessPoints(rtc::FunctionView<float(const DataType&)> fx,
rtc::FunctionView<absl::optional<float>(const DataType&)> fy,
const IterableType& data_view,
TimeSeries* result) {
for (size_t i = 0; i < data_view.size(); i++) {
const DataType& elem = data_view[i];
float x = fx(elem);
absl::optional<float> y = fy(elem);
if (y)
result->points.emplace_back(x, *y);
}
}
// For each pair of adjacent elements in |data|, use |f()| to extract a
// y-coordinate and store the result in a TimeSeries. Note that the x-coordinate
// will be the time of the second element in the pair.
template <typename DataType, typename ResultType, typename IterableType>
void ProcessPairs(
rtc::FunctionView<float(const DataType&)> fx,
rtc::FunctionView<absl::optional<ResultType>(const DataType&,
const DataType&)> fy,
const IterableType& data,
TimeSeries* result) {
for (size_t i = 1; i < data.size(); i++) {
float x = fx(data[i]);
absl::optional<ResultType> y = fy(data[i - 1], data[i]);
if (y)
result->points.emplace_back(x, static_cast<float>(*y));
}
}
// For each pair of adjacent elements in |data|, use |f()| to extract a
// y-coordinate and store the result in a TimeSeries. Note that the x-coordinate
// will be the time of the second element in the pair.
template <typename DataType, typename ResultType, typename IterableType>
void AccumulatePairs(
rtc::FunctionView<float(const DataType&)> fx,
rtc::FunctionView<absl::optional<ResultType>(const DataType&,
const DataType&)> fy,
const IterableType& data,
TimeSeries* result) {
ResultType sum = 0;
for (size_t i = 1; i < data.size(); i++) {
float x = fx(data[i]);
absl::optional<ResultType> y = fy(data[i - 1], data[i]);
if (y) {
sum += *y;
result->points.emplace_back(x, static_cast<float>(sum));
}
}
}
// Calculates a moving average of |data| and stores the result in a TimeSeries.
// A data point is generated every |step| microseconds from |begin_time|
// to |end_time|. The value of each data point is the average of the data
// during the preceding |window_duration_us| microseconds.
template <typename DataType, typename ResultType, typename IterableType>
void MovingAverage(
rtc::FunctionView<absl::optional<ResultType>(const DataType&)> fy,
const IterableType& data_view,
AnalyzerConfig config,
TimeSeries* result) {
size_t window_index_begin = 0;
size_t window_index_end = 0;
ResultType sum_in_window = 0;
for (int64_t t = config.begin_time_; t < config.end_time_ + config.step_;
t += config.step_) {
while (window_index_end < data_view.size() &&
data_view[window_index_end].log_time_us() < t) {
absl::optional<ResultType> value = fy(data_view[window_index_end]);
if (value)
sum_in_window += *value;
++window_index_end;
}
while (window_index_begin < data_view.size() &&
data_view[window_index_begin].log_time_us() <
t - config.window_duration_) {
absl::optional<ResultType> value = fy(data_view[window_index_begin]);
if (value)
sum_in_window -= *value;
++window_index_begin;
}
float window_duration_s =
static_cast<float>(config.window_duration_) / kNumMicrosecsPerSec;
float x = config.GetCallTimeSec(t);
float y = sum_in_window / window_duration_s;
result->points.emplace_back(x, y);
}
}
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.log_time_us());
float y = category_id;
time_series.points.emplace_back(x, y);
}
return time_series;
}
const char kUnknownEnumValue[] = "unknown";
const char kIceCandidateTypeLocal[] = "local";
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";
std::string GetIceCandidateTypeAsString(webrtc::IceCandidateType type) {
switch (type) {
case webrtc::IceCandidateType::kLocal:
return kIceCandidateTypeLocal;
case webrtc::IceCandidateType::kStun:
return kIceCandidateTypeStun;
case webrtc::IceCandidateType::kPrflx:
return kIceCandidateTypePrflx;
case webrtc::IceCandidateType::kRelay:
return kIceCandidateTypeRelay;
default:
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;
std::string local_candidate_type =
GetIceCandidateTypeAsString(config.local_candidate_type);
std::string remote_candidate_type =
GetIceCandidateTypeAsString(config.remote_candidate_type);
if (config.local_candidate_type == webrtc::IceCandidateType::kRelay) {
local_candidate_type +=
"(" + GetProtocolAsString(config.local_relay_protocol) + ")";
}
ss << local_candidate_type << ":"
<< GetNetworkTypeAsString(config.local_network_type) << ":"
<< GetAddressFamilyAsString(config.local_address_family) << "->"
<< 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";
}
}
} // namespace
EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log,
bool normalize_time)
: parsed_log_(log) {
config_.window_duration_ = 250000;
config_.step_ = 10000;
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_ = 0;
}
const auto& log_start_events = parsed_log_.start_log_events();
const auto& log_end_events = parsed_log_.stop_log_events();
auto start_iter = log_start_events.begin();
auto end_iter = log_end_events.begin();
while (start_iter != log_start_events.end()) {
int64_t start = start_iter->log_time_us();
++start_iter;
absl::optional<int64_t> next_start;
if (start_iter != log_start_events.end())
next_start.emplace(start_iter->log_time_us());
if (end_iter != log_end_events.end() &&
end_iter->log_time_us() <=
next_start.value_or(std::numeric_limits<int64_t>::max())) {
int64_t end = end_iter->log_time_us();
RTC_DCHECK_LE(start, end);
log_segments_.push_back(std::make_pair(start, end));
++end_iter;
} else {
// we're missing an end event. Assume that it occurred just before the
// next start.
log_segments_.push_back(
std::make_pair(start, next_start.value_or(config_.end_time_)));
}
}
RTC_LOG(LS_INFO) << "Found " << log_segments_.size()
<< " (LOG_START, LOG_END) segments in log.";
}
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::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(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.log_time_us());
};
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->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"}});
}
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_us());
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(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");
}
// 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_us());
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");
}
// 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(direction, stream.ssrc))
continue;
TimeSeries time_series(GetStreamName(direction, stream.ssrc),
LineStyle::kLine);
for (auto& packet : stream.packet_view) {
if (packet.header.extension.hasAudioLevel) {
float x = config_.GetCallTimeSec(packet.log_time_us());
// 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.audioLevel);
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(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_us());
};
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(kIncomingPacket, stream.ssrc),
LineStyle::kLine, PointStyle::kHighlight);
// TODO(terelius): Should the window and step size be read from the class
// instead?
const int64_t kWindowUs = 1000000;
const int64_t kStep = 1000000;
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 (int64_t t = config_.begin_time_; t < config_.end_time_ + kStep;
t += kStep) {
while (window_index_end < packets.size() &&
packets[window_index_end].rtp.log_time_us() < 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_us() < t - kWindowUs) {
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(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 end_time_us = log_segments_.empty()
? std::numeric_limits<int64_t>::max()
: log_segments_.front().second;
absl::optional<uint32_t> estimated_frequency =
EstimateRtpClockFrequency(packets, end_time_us);
if (!estimated_frequency)
continue;
const double frequency_hz = *estimated_frequency;
if (IsVideoSsrc(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_us());
};
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(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(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_us());
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<int64_t, 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_us(), 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 (int64_t 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_) / 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_us());
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) {
// TODO(terelius): This could be provided by the parser.
std::multimap<int64_t, 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_us(), 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 (int64_t 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_) / 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_us());
float y = static_cast<float>(loss_update.bitrate_bps) / 1000;
loss_series.points.emplace_back(x, y);
}
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;
double 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_us());
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_NOTREACHED();
}
}
delay_series.points.emplace_back(x, y);
}
RTC_CHECK(last_series);
last_series->intervals.emplace_back(last_detector_switch, config_.end_time_);
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_us());
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_us());
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_us());
probe_failures_series.points.emplace_back(x, 0);
}
IntervalSeries alr_state("ALR", "#555555", IntervalSeries::kHorizontal);
bool previously_in_alr = false;
int64_t alr_start = 0;
for (auto& alr : parsed_log_.alr_state_events()) {
float y = config_.GetCallTimeSec(alr.log_time_us());
if (!previously_in_alr && alr.in_alr) {
alr_start = alr.log_time_us();
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));
}
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_us());
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(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_us());
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.us()),
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_us()),
logged.bitrate_bps / 1000);
for (const auto& logged : parsed_log_.bwe_probe_success_events())
probe_results.points.emplace_back(
config_.GetCallTimeSec(logged.log_time_us()),
logged.bitrate_bps / 1000);
for (const auto& logged : parsed_log_.bwe_loss_updates())
loss_based.points.emplace_back(config_.GetCallTimeSec(logged.log_time_us()),
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::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;
PacketRouter packet_router;
PacedSender pacer(&clock, &packet_router, &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::us(clock.TimeInMicroseconds());
cc_config.constraints.starting_rate = DataRate::bps(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("Acked bitrate", LineStyle::kLine,
PointStyle::kHighlight);
TimeSeries acked_estimate_time_series(
"Acked 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 acked_bitrate(250, 8000);
#if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
FieldTrialBasedConfig field_trial_config_;
// The event_log_visualizer should normally not be compiled with
// BWE_TEST_LOGGING_COMPILE_TIME_ENABLE since the normal plots won't work.
// However, compiling with BWE_TEST_LOGGING, running with --plot_sendside_bwe
// and piping the output to plot_dynamics.py can be used as a hack to get the
// internal state of various BWE components. In this case, it is important
// we don't instantiate the AcknowledgedBitrateEstimator both here and in
// GoogCcNetworkController since that would lead to duplicate outputs.
AcknowledgedBitrateEstimator acknowledged_bitrate_estimator(
&field_trial_config_,
std::make_unique<BitrateEstimator>(&field_trial_config_));
#endif // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
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) {
RTC_DCHECK(rtp_packet.rtp.header.extension.hasTransportSequenceNumber);
RtpPacketSendInfo packet_info;
packet_info.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.has_rtp_sequence_number = true;
packet_info.length = rtp_packet.rtp.total_length;
transport_feedback.AddPacket(
packet_info,
0u, // Per packet overhead bytes.
Timestamp::us(rtp_packet.rtp.log_time_us()));
rtc::SentPacket sent_packet(
rtp_packet.rtp.header.extension.transportSequenceNumber,
rtp_packet.rtp.log_time_us() / 1000);
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::ms(clock.TimeInMilliseconds()));
absl::optional<uint32_t> bitrate_bps;
if (feedback_msg) {
observer.Update(goog_cc->OnTransportPacketsFeedback(*feedback_msg));
std::vector<PacketResult> feedback =
feedback_msg->SortedByReceiveTime();
if (!feedback.empty()) {
#if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
acknowledged_bitrate_estimator.IncomingPacketFeedbackVector(feedback);
#endif // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
for (const PacketResult& packet : feedback)
acked_bitrate.Update(packet.sent_packet.size.bytes(),
packet.receive_time.ms());
bitrate_bps = acked_bitrate.Rate(feedback.back().receive_time.ms());
}
}
float x = config_.GetCallTimeSec(clock.TimeInMicroseconds());
float y = bitrate_bps.value_or(0) / 1000;
acked_time_series.points.emplace_back(x, y);
#if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
y = acknowledged_bitrate_estimator.bitrate()
.value_or(DataRate::Zero())
.kbps();
acked_estimate_time_series.points.emplace_back(x, y);
#endif // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
++rtcp_iterator;
}
if (clock.TimeInMicroseconds() >= NextProcessTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
ProcessInterval msg;
msg.at_time = Timestamp::us(clock.TimeInMicroseconds());
observer.Update(goog_cc->OnProcessInterval(msg));
next_process_time_us_ += process_interval.us();
}
if (observer.GetAndResetBitrateUpdated() ||
time_us - last_update_us >= 1e6) {
uint32_t y = observer.last_bitrate_bps() / 1000;
float x = config_.GetCallTimeSec(clock.TimeInMicroseconds());
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(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 RembInterceptingPacketRouter : public PacketRouter {
public:
void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
uint32_t bitrate_bps) override {
last_bitrate_bps_ = bitrate_bps;
bitrate_updated_ = true;
PacketRouter::OnReceiveBitrateChanged(ssrcs, bitrate_bps);
}
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_;
};
std::multimap<int64_t, const RtpPacketType*> incoming_rtp;
for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
if (IsVideoSsrc(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);
RembInterceptingPacketRouter packet_router;
// TODO(terelius): The PacketRouter is used as the RemoteBitrateObserver.
// Is this intentional?
ReceiveSideCongestionController rscc(&clock, &packet_router);
// 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;
int64_t arrival_time_ms = packet.rtp.log_time_us() / 1000;
size_t payload = packet.rtp.total_length; /*Should subtract header?*/
clock.AdvanceTimeMicroseconds(packet.rtp.log_time_us() -
clock.TimeInMicroseconds());
rscc.OnReceivedPacket(arrival_time_ms, payload, packet.rtp.header);
acked_bitrate.Update(payload, arrival_time_ms);
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.TimeInMicroseconds());
acked_time_series.points.emplace_back(x, y);
}
if (packet_router.GetAndResetBitrateUpdated() ||
clock.TimeInMicroseconds() - last_update_us >= 1e6) {
uint32_t y = packet_router.last_bitrate_bps() / 1000;
float x = config_.GetCallTimeSec(clock.TimeInMicroseconds());
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();
int64_t prev_y = 0;
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;
});
for (const auto& packet : matched_rtp_rtcp) {
if (packet.arrival_time_ms == MatchedSendArrivalTimes::kNotReceived)
continue;
float x = config_.GetCallTimeSec(1000 * packet.feedback_arrival_time_ms);
int64_t y = packet.arrival_time_ms - packet.send_time_ms;
prev_y = y;
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(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 end_time_us = log_segments_.empty()
? std::numeric_limits<int64_t>::max()
: log_segments_.front().second;
absl::optional<uint32_t> estimated_frequency =
EstimateRtpClockFrequency(packets, end_time_us);
if (!estimated_frequency)
continue;
if (IsVideoSsrc(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(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_us());
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(direction, stream.ssrc) + " capture-time",
LineStyle::kLine, PointStyle::kHighlight);
for (const auto& packet : stream.packet_view) {
float x = config_.GetCallTimeSec(packet.log_time_us());
float y = packet.header.timestamp;
rtp_timestamps.points.emplace_back(x, y);
}
plot->AppendTimeSeries(std::move(rtp_timestamps));
TimeSeries rtcp_timestamps(
GetStreamName(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_us());
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_us());
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(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_us());
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(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::CreateAudioEncoderTargetBitrateGraph(Plot* plot) {
TimeSeries time_series("Audio encoder target bitrate", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaBitrateBps = [](const LoggedAudioNetworkAdaptationEvent& ana_event)
-> absl::optional<float> {
if (ana_event.config.bitrate_bps)
return absl::optional<float>(
static_cast<float>(*ana_event.config.bitrate_bps));
return absl::nullopt;
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->config_.GetCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaBitrateBps,
parsed_log_.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Bitrate (bps)", kBottomMargin, kTopMargin);
plot->SetTitle("Reported audio encoder target bitrate");
}
void EventLogAnalyzer::CreateAudioEncoderFrameLengthGraph(Plot* plot) {
TimeSeries time_series("Audio encoder frame length", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaFrameLengthMs =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.frame_length_ms)
return absl::optional<float>(
static_cast<float>(*ana_event.config.frame_length_ms));
return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->config_.GetCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaFrameLengthMs,
parsed_log_.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Frame length (ms)", kBottomMargin, kTopMargin);
plot->SetTitle("Reported audio encoder frame length");
}
void EventLogAnalyzer::CreateAudioEncoderPacketLossGraph(Plot* plot) {
TimeSeries time_series("Audio encoder uplink packet loss fraction",
LineStyle::kLine, PointStyle::kHighlight);
auto GetAnaPacketLoss =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.uplink_packet_loss_fraction)
return absl::optional<float>(static_cast<float>(
*ana_event.config.uplink_packet_loss_fraction));
return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->config_.GetCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaPacketLoss,
parsed_log_.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 10, "Percent lost packets", kBottomMargin,
kTopMargin);
plot->SetTitle("Reported audio encoder lost packets");
}
void EventLogAnalyzer::CreateAudioEncoderEnableFecGraph(Plot* plot) {
TimeSeries time_series("Audio encoder FEC", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaFecEnabled =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.enable_fec)
return absl::optional<float>(
static_cast<float>(*ana_event.config.enable_fec));
return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->config_.GetCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaFecEnabled,
parsed_log_.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "FEC (false/true)", kBottomMargin, kTopMargin);
plot->SetTitle("Reported audio encoder FEC");
}
void EventLogAnalyzer::CreateAudioEncoderEnableDtxGraph(Plot* plot) {
TimeSeries time_series("Audio encoder DTX", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaDtxEnabled =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.enable_dtx)
return absl::optional<float>(
static_cast<float>(*ana_event.config.enable_dtx));
return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->config_.GetCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaDtxEnabled,
parsed_log_.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "DTX (false/true)", kBottomMargin, kTopMargin);
plot->SetTitle("Reported audio encoder DTX");
}
void EventLogAnalyzer::CreateAudioEncoderNumChannelsGraph(Plot* plot) {
TimeSeries time_series("Audio encoder number of channels", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaNumChannels =
[](const LoggedAudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.num_channels)
return absl::optional<float>(
static_cast<float>(*ana_event.config.num_channels));
return absl::optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->config_.GetCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaNumChannels,
parsed_log_.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))",
kBottomMargin, kTopMargin);
plot->SetTitle("Reported audio encoder number of channels");
}
class NetEqStreamInput : public test::NetEqInput {
public:
// Does not take any ownership, and all pointers must refer to valid objects
// that outlive the one constructed.
NetEqStreamInput(const std::vector<LoggedRtpPacketIncoming>* packet_stream,
const std::vector<LoggedAudioPlayoutEvent>* output_events,
absl::optional<int64_t> end_time_ms)
: packet_stream_(*packet_stream),
packet_stream_it_(packet_stream_.begin()),
output_events_it_(output_events->begin()),
output_events_end_(output_events->end()),
end_time_ms_(end_time_ms) {
RTC_DCHECK(packet_stream);
RTC_DCHECK(output_events);
}
absl::optional<int64_t> NextPacketTime() const override {
if (packet_stream_it_ == packet_stream_.end()) {
return absl::nullopt;
}
if (end_time_ms_ && packet_stream_it_->rtp.log_time_ms() > *end_time_ms_) {
return absl::nullopt;
}
return packet_stream_it_->rtp.log_time_ms();
}
absl::optional<int64_t> NextOutputEventTime() const override {
if (output_events_it_ == output_events_end_) {
return absl::nullopt;
}
if (end_time_ms_ && output_events_it_->log_time_ms() > *end_time_ms_) {
return absl::nullopt;
}
return output_events_it_->log_time_ms();
}
std::unique_ptr<PacketData> PopPacket() override {
if (packet_stream_it_ == packet_stream_.end()) {
return std::unique_ptr<PacketData>();
}
std::unique_ptr<PacketData> packet_data(new PacketData());
packet_data->header = packet_stream_it_->rtp.header;
packet_data->time_ms = packet_stream_it_->rtp.log_time_ms();
// This is a header-only "dummy" packet. Set the payload to all zeros, with
// length according to the virtual length.
packet_data->payload.SetSize(packet_stream_it_->rtp.total_length -
packet_stream_it_->rtp.header_length);
std::fill_n(packet_data->payload.data(), packet_data->payload.size(), 0);
++packet_stream_it_;
return packet_data;
}
void AdvanceOutputEvent() override {
if (output_events_it_ != output_events_end_) {
++output_events_it_;
}
}
bool ended() const override { return !NextEventTime(); }
absl::optional<RTPHeader> NextHeader() const override {
if (packet_stream_it_ == packet_stream_.end()) {
return absl::nullopt;
}
return packet_stream_it_->rtp.header;
}
private:
const std::vector<LoggedRtpPacketIncoming>& packet_stream_;
std::vector<LoggedRtpPacketIncoming>::const_iterator packet_stream_it_;
std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_it_;
const std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_end_;
const absl::optional<int64_t> end_time_ms_;
};
namespace {
// Factory to create a "replacement decoder" that produces the decoded audio
// by reading from a file rather than from the encoded payloads.
class ReplacementAudioDecoderFactory : public AudioDecoderFactory {
public:
ReplacementAudioDecoderFactory(const absl::string_view replacement_file_name,
int file_sample_rate_hz)
: replacement_file_name_(replacement_file_name),
file_sample_rate_hz_(file_sample_rate_hz) {}
std::vector<AudioCodecSpec> GetSupportedDecoders() override {
RTC_NOTREACHED();
return {};
}
bool IsSupportedDecoder(const SdpAudioFormat& format) override {
return true;
}
std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format,
absl::optional<AudioCodecPairId> codec_pair_id) override {
auto replacement_file = std::make_unique<test::ResampleInputAudioFile>(
replacement_file_name_, file_sample_rate_hz_);
replacement_file->set_output_rate_hz(48000);
return std::make_unique<test::FakeDecodeFromFile>(
std::move(replacement_file), 48000, false);
}
private:
const std::string replacement_file_name_;
const int file_sample_rate_hz_;
};
// Creates a NetEq test object and all necessary input and output helpers. Runs
// the test and returns the NetEqDelayAnalyzer object that was used to
// instrument the test.
std::unique_ptr<test::NetEqStatsGetter> CreateNetEqTestAndRun(
const std::vector<LoggedRtpPacketIncoming>* packet_stream,
const std::vector<LoggedAudioPlayoutEvent>* output_events,
absl::optional<int64_t> end_time_ms,
const std::string& replacement_file_name,
int file_sample_rate_hz) {
std::unique_ptr<test::NetEqInput> input(
new NetEqStreamInput(packet_stream, output_events, end_time_ms));
constexpr int kReplacementPt = 127;
std::set<uint8_t> cn_types;
std::set<uint8_t> forbidden_types;
input.reset(new test::NetEqReplacementInput(std::move(input), kReplacementPt,
cn_types, forbidden_types));
NetEq::Config config;
config.max_packets_in_buffer = 200;
config.enable_fast_accelerate = true;
std::unique_ptr<test::VoidAudioSink> output(new test::VoidAudioSink());
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory =
new rtc::RefCountedObject<ReplacementAudioDecoderFactory>(
replacement_file_name, file_sample_rate_hz);
test::NetEqTest::DecoderMap codecs = {
{kReplacementPt, SdpAudioFormat("l16", 48000, 1)}};
std::unique_ptr<test::NetEqDelayAnalyzer> delay_cb(
new test::NetEqDelayAnalyzer);
std::unique_ptr<test::NetEqStatsGetter> neteq_stats_getter(
new test::NetEqStatsGetter(std::move(delay_cb)));
test::DefaultNetEqTestErrorCallback error_cb;
test::NetEqTest::Callbacks callbacks;
callbacks.error_callback = &error_cb;
callbacks.post_insert_packet = neteq_stats_getter->delay_analyzer();
callbacks.get_audio_callback = neteq_stats_getter.get();
test::NetEqTest test(config, decoder_factory, codecs, nullptr,
std::move(input), std::move(output), callbacks);
test.Run();
return neteq_stats_getter;
}
} // namespace
EventLogAnalyzer::NetEqStatsGetterMap EventLogAnalyzer::SimulateNetEq(
const std::string& replacement_file_name,
int file_sample_rate_hz) const {
NetEqStatsGetterMap neteq_stats;
for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
const uint32_t ssrc = stream.ssrc;
if (!IsAudioSsrc(kIncomingPacket, ssrc))
continue;
const std::vector<LoggedRtpPacketIncoming>* audio_packets =
&stream.incoming_packets;
if (audio_packets == nullptr) {
// No incoming audio stream found.
continue;
}
RTC_DCHECK(neteq_stats.find(ssrc) == neteq_stats.end());
std::map<uint32_t, std::vector<LoggedAudioPlayoutEvent>>::const_iterator
output_events_it = parsed_log_.audio_playout_events().find(ssrc);
if (output_events_it == parsed_log_.audio_playout_events().end()) {
// Could not find output events with SSRC matching the input audio stream.
// Using the first available stream of output events.
output_events_it = parsed_log_.audio_playout_events().cbegin();
}
absl::optional<int64_t> end_time_ms =
log_segments_.empty()
? absl::nullopt
: absl::optional<int64_t>(log_segments_.front().second / 1000);
neteq_stats[ssrc] = CreateNetEqTestAndRun(
audio_packets, &output_events_it->second, end_time_ms,
replacement_file_name, file_sample_rate_hz);
}
return neteq_stats;
}
// Given a NetEqStatsGetter and the SSRC that the NetEqStatsGetter was created
// for, this method generates a plot for the jitter buffer delay profile.
void EventLogAnalyzer::CreateAudioJitterBufferGraph(
uint32_t ssrc,
const test::NetEqStatsGetter* stats_getter,
Plot* plot) const {
test::NetEqDelayAnalyzer::Delays arrival_delay_ms;
test::NetEqDelayAnalyzer::Delays corrected_arrival_delay_ms;
test::NetEqDelayAnalyzer::Delays playout_delay_ms;
test::NetEqDelayAnalyzer::Delays target_delay_ms;
stats_getter->delay_analyzer()->CreateGraphs(
&arrival_delay_ms, &corrected_arrival_delay_ms, &playout_delay_ms,
&target_delay_ms);
TimeSeries time_series_packet_arrival("packet arrival delay",
LineStyle::kLine);
TimeSeries time_series_relative_packet_arrival(
"Relative packet arrival delay", LineStyle::kLine);
TimeSeries time_series_play_time("Playout delay", LineStyle::kLine);
TimeSeries time_series_target_time("Target delay", LineStyle::kLine,
PointStyle::kHighlight);
for (const auto& data : arrival_delay_ms) {
const float x = config_.GetCallTimeSec(data.first * 1000); // ms to us.
const float y = data.second;
time_series_packet_arrival.points.emplace_back(TimeSeriesPoint(x, y));
}
for (const auto& data : corrected_arrival_delay_ms) {
const float x = config_.GetCallTimeSec(data.first * 1000); // ms to us.
const float y = data.second;
time_series_relative_packet_arrival.points.emplace_back(
TimeSeriesPoint(x, y));
}
for (const auto& data : playout_delay_ms) {
const float x = config_.GetCallTimeSec(data.first * 1000); // ms to us.
const float y = data.second;
time_series_play_time.points.emplace_back(TimeSeriesPoint(x, y));
}
for (const auto& data : target_delay_ms) {
const float x = config_.GetCallTimeSec(data.first * 1000); // ms to us.
const float y = data.second;
time_series_target_time.points.emplace_back(TimeSeriesPoint(x, y));
}
plot->AppendTimeSeries(std::move(time_series_packet_arrival));
plot->AppendTimeSeries(std::move(time_series_relative_packet_arrival));
plot->AppendTimeSeries(std::move(time_series_play_time));
plot->AppendTimeSeries(std::move(time_series_target_time));
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Relative delay (ms)", kBottomMargin,
kTopMargin);
plot->SetTitle("NetEq timing for " + GetStreamName(kIncomingPacket, ssrc));
}
template <typename NetEqStatsType>
void EventLogAnalyzer::CreateNetEqStatsGraphInternal(
const NetEqStatsGetterMap& neteq_stats,
rtc::FunctionView<const std::vector<std::pair<int64_t, NetEqStatsType>>*(
const test::NetEqStatsGetter*)> data_extractor,
rtc::FunctionView<float(const NetEqStatsType&)> stats_extractor,
const std::string& plot_name,
Plot* plot) const {
std::map<uint32_t, TimeSeries> time_series;
for (const auto& st : neteq_stats) {
const uint32_t ssrc = st.first;
const std::vector<std::pair<int64_t, NetEqStatsType>>* data_vector =
data_extractor(st.second.get());
for (const auto& data : *data_vector) {
const float time =
config_.GetCallTimeSec(data.first * 1000); // ms to us.
const float value = stats_extractor(data.second);
time_series[ssrc].points.emplace_back(TimeSeriesPoint(time, value));
}
}
for (auto& series : time_series) {
series.second.label = GetStreamName(kIncomingPacket, series.first);
series.second.line_style = LineStyle::kLine;
plot->AppendTimeSeries(std::move(series.second));
}
plot->SetXAxis(config_.CallBeginTimeSec(), config_.CallEndTimeSec(),
"Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, plot_name, kBottomMargin, kTopMargin);
plot->SetTitle(plot_name);
}
void EventLogAnalyzer::CreateNetEqNetworkStatsGraph(
const NetEqStatsGetterMap& neteq_stats,
rtc::FunctionView<float(const NetEqNetworkStatistics&)> stats_extractor,
const std::string& plot_name,
Plot* plot) const {
CreateNetEqStatsGraphInternal<NetEqNetworkStatistics>(
neteq_stats,
[](const test::NetEqStatsGetter* stats_getter) {
return stats_getter->stats();
},
stats_extractor, plot_name, plot);
}
void EventLogAnalyzer::CreateNetEqLifetimeStatsGraph(
const NetEqStatsGetterMap& neteq_stats,
rtc::FunctionView<float(const NetEqLifetimeStatistics&)> stats_extractor,
const std::string& plot_name,
Plot* plot) const {
CreateNetEqStatsGraphInternal<NetEqLifetimeStatistics>(
neteq_stats,
[](const test::NetEqStatsGetter* stats_getter) {
return stats_getter->lifetime_stats();
},
stats_extractor, plot_name, plot);
}
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_us());
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_us());
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_us());
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_us());
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");
}
void EventLogAnalyzer::PrintNotifications(FILE* file) {
fprintf(file, "========== TRIAGE NOTIFICATIONS ==========\n");
for (const auto& alert : incoming_rtp_recv_time_gaps_) {
fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
}
for (const auto& alert : incoming_rtcp_recv_time_gaps_) {
fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
}
for (const auto& alert : outgoing_rtp_send_time_gaps_) {
fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
}
for (const auto& alert : outgoing_rtcp_send_time_gaps_) {
fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
}
for (const auto& alert : incoming_seq_num_jumps_) {
fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
}
for (const auto& alert : incoming_capture_time_jumps_) {
fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
}
for (const auto& alert : outgoing_seq_num_jumps_) {
fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
}
for (const auto& alert : outgoing_capture_time_jumps_) {
fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
}
for (const auto& alert : outgoing_high_loss_alerts_) {
fprintf(file, " : %s\n", alert.ToString().c_str());
}
fprintf(file, "========== END TRIAGE NOTIFICATIONS ==========\n");
}
void EventLogAnalyzer::CreateStreamGapAlerts(PacketDirection direction) {
// With 100 packets/s (~800kbps), false positives would require 10 s without
// data.
constexpr int64_t kMaxSeqNumJump = 1000;
// With a 90 kHz clock, false positives would require 10 s without data.
constexpr int64_t kMaxCaptureTimeJump = 900000;
int64_t end_time_us = log_segments_.empty()
? std::numeric_limits<int64_t>::max()
: log_segments_.front().second;
SeqNumUnwrapper<uint16_t> seq_num_unwrapper;
absl::optional<int64_t> last_seq_num;
SeqNumUnwrapper<uint32_t> capture_time_unwrapper;
absl::optional<int64_t> last_capture_time;
// Check for gaps in sequence numbers and capture timestamps.
for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
for (const auto& packet : stream.packet_view) {
if (packet.log_time_us() > end_time_us) {
// Only process the first (LOG_START, LOG_END) segment.
break;
}
int64_t seq_num = seq_num_unwrapper.Unwrap(packet.header.sequenceNumber);
if (last_seq_num.has_value() &&
std::abs(seq_num - last_seq_num.value()) > kMaxSeqNumJump) {
Alert_SeqNumJump(direction,
config_.GetCallTimeSec(packet.log_time_us()),
packet.header.ssrc);
}
last_seq_num.emplace(seq_num);
int64_t capture_time =
capture_time_unwrapper.Unwrap(packet.header.timestamp);
if (last_capture_time.has_value() &&
std::abs(capture_time - last_capture_time.value()) >
kMaxCaptureTimeJump) {
Alert_CaptureTimeJump(direction,
config_.GetCallTimeSec(packet.log_time_us()),
packet.header.ssrc);
}
last_capture_time.emplace(capture_time);
}
}
}
void EventLogAnalyzer::CreateTransmissionGapAlerts(PacketDirection direction) {
constexpr int64_t kMaxRtpTransmissionGap = 500000;
constexpr int64_t kMaxRtcpTransmissionGap = 2000000;
int64_t end_time_us = log_segments_.empty()
? std::numeric_limits<int64_t>::max()
: log_segments_.front().second;
// TODO(terelius): The parser could provide a list of all packets, ordered
// by time, for each direction.
std::multimap<int64_t, const LoggedRtpPacket*> rtp_in_direction;
for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
for (const LoggedRtpPacket& rtp_packet : stream.packet_view)
rtp_in_direction.emplace(rtp_packet.log_time_us(), &rtp_packet);
}
absl::optional<int64_t> last_rtp_time;
for (const auto& kv : rtp_in_direction) {
int64_t timestamp = kv.first;
if (timestamp > end_time_us) {
// Only process the first (LOG_START, LOG_END) segment.
break;
}
int64_t duration = timestamp - last_rtp_time.value_or(0);
if (last_rtp_time.has_value() && duration > kMaxRtpTransmissionGap) {
// No packet sent/received for more than 500 ms.
Alert_RtpLogTimeGap(direction, config_.GetCallTimeSec(timestamp),
duration / 1000);
}
last_rtp_time.emplace(timestamp);
}
absl::optional<int64_t> last_rtcp_time;
if (direction == kIncomingPacket) {
for (const auto& rtcp : parsed_log_.incoming_rtcp_packets()) {
if (rtcp.log_time_us() > end_time_us) {
// Only process the first (LOG_START, LOG_END) segment.
break;
}
int64_t duration = rtcp.log_time_us() - last_rtcp_time.value_or(0);
if (last_rtcp_time.has_value() && duration > kMaxRtcpTransmissionGap) {
// No feedback sent/received for more than 2000 ms.
Alert_RtcpLogTimeGap(direction,
config_.GetCallTimeSec(rtcp.log_time_us()),
duration / 1000);
}
last_rtcp_time.emplace(rtcp.log_time_us());
}
} else {
for (const auto& rtcp : parsed_log_.outgoing_rtcp_packets()) {
if (rtcp.log_time_us() > end_time_us) {
// Only process the first (LOG_START, LOG_END) segment.
break;
}
int64_t duration = rtcp.log_time_us() - last_rtcp_time.value_or(0);
if (last_rtcp_time.has_value() && duration > kMaxRtcpTransmissionGap) {
// No feedback sent/received for more than 2000 ms.
Alert_RtcpLogTimeGap(direction,
config_.GetCallTimeSec(rtcp.log_time_us()),
duration / 1000);
}
last_rtcp_time.emplace(rtcp.log_time_us());
}
}
}
// TODO(terelius): Notifications could possibly be generated by the same code
// that produces the graphs. There is some code duplication that could be
// avoided, but that might be solved anyway when we move functionality from the
// analyzer to the parser.
void EventLogAnalyzer::CreateTriageNotifications() {
CreateStreamGapAlerts(kIncomingPacket);
CreateStreamGapAlerts(kOutgoingPacket);
CreateTransmissionGapAlerts(kIncomingPacket);
CreateTransmissionGapAlerts(kOutgoingPacket);
int64_t end_time_us = log_segments_.empty()
? std::numeric_limits<int64_t>::max()
: log_segments_.front().second;
constexpr double kMaxLossFraction = 0.05;
// Loss feedback
int64_t total_lost_packets = 0;
int64_t total_expected_packets = 0;
for (auto& bwe_update : parsed_log_.bwe_loss_updates()) {
if (bwe_update.log_time_us() > end_time_us) {
// Only process the first (LOG_START, LOG_END) segment.
break;
}
int64_t lost_packets = static_cast<double>(bwe_update.fraction_lost) / 255 *
bwe_update.expected_packets;
total_lost_packets += lost_packets;
total_expected_packets += bwe_update.expected_packets;
}
double avg_outgoing_loss =
static_cast<double>(total_lost_packets) / total_expected_packets;
if (avg_outgoing_loss > kMaxLossFraction) {
Alert_OutgoingHighLoss(avg_outgoing_loss);
}
}
} // namespace webrtc
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