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webrtc / src / 0040b66ad3c38c51153de721657e135b30406ff9 / . / rtc_tools / event_log_visualizer / analyzer.cc
blob: 3bed18bb5735524dda185a23b0a58545114e5e1c [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/event_log_visualizer/analyzer.h"
#include <algorithm>
#include <cmath>
#include <limits>
#include <map>
#include <string>
#include <utility>
#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 "common_types.h" // NOLINT(build/include)
#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/delay_based_bwe.h"
#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
#include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
#include "modules/congestion_controller/include/send_side_congestion_controller.h"
#include "modules/include/module_common_types.h"
#include "modules/pacing/packet_router.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/function_view.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/sequence_number_util.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/rate_statistics.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;
void SortPacketFeedbackVector(std::vector<PacketFeedback>* vec) {
auto pred = [](const PacketFeedback& packet_feedback) {
return packet_feedback.arrival_time_ms == PacketFeedback::kNotReceived;
};
vec->erase(std::remove_if(vec->begin(), vec->end(), pred), vec->end());
std::sort(vec->begin(), vec->end(), PacketFeedbackComparator());
}
std::string SsrcToString(uint32_t ssrc) {
std::stringstream ss;
ss << "SSRC " << ssrc;
return ss.str();
}
// 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.size() == 0)
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>
rtc::Optional<uint32_t> EstimateRtpClockFrequency(
const RtpPacketContainer& packets,
int64_t end_time_us) {
RTC_CHECK(packets.size() >= 2);
SeqNumUnwrapper<uint32_t> unwrapper;
uint64_t first_rtp_timestamp =
unwrapper.Unwrap(packets[0].rtp.header.timestamp);
int64_t first_log_timestamp = packets[0].log_time_us();
uint64_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 rtc::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) < 0.05 * f) {
return f;
}
}
RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate "
<< estimated_frequency
<< "not close to any stardard RTP frequency.";
return rtc::nullopt;
}
constexpr float kLeftMargin = 0.01f;
constexpr float kRightMargin = 0.02f;
constexpr float kBottomMargin = 0.02f;
constexpr float kTopMargin = 0.05f;
rtc::Optional<double> NetworkDelayDiff_AbsSendTime(
const LoggedRtpPacket& old_packet,
const LoggedRtpPacket& new_packet) {
if (old_packet.header.extension.hasAbsoluteSendTime &&
new_packet.header.extension.hasAbsoluteSendTime) {
int64_t send_time_diff = WrappingDifference(
new_packet.header.extension.absoluteSendTime,
old_packet.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 rtc::nullopt;
}
}
rtc::Optional<double> NetworkDelayDiff_CaptureTime(
const LoggedRtpPacket& old_packet,
const LoggedRtpPacket& new_packet) {
int64_t send_time_diff = WrappingDifference(
new_packet.header.timestamp, old_packet.header.timestamp, 1ull << 32);
int64_t recv_time_diff = new_packet.log_time_us() - old_packet.log_time_us();
const double kVideoSampleRate = 90000;
// TODO(terelius): We treat all streams as video for now, even though
// audio might be sampled at e.g. 16kHz, because it is really difficult to
// figure out the true sampling rate of a stream. The effect is that the
// delay will be scaled incorrectly for non-video streams.
double delay_change =
static_cast<double>(recv_time_diff) / 1000 -
static_cast<double>(send_time_diff) / kVideoSampleRate * 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.header.timestamp
<< ", received time " << old_packet.log_time_us();
RTC_LOG(LS_WARNING) << "New capture time " << new_packet.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) /
kVideoSampleRate
<< "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<rtc::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);
rtc::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<rtc::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]);
rtc::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<rtc::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]);
rtc::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 preceeding |window_duration_us| microseconds.
template <typename DataType, typename ResultType, typename IterableType>
void MovingAverage(
rtc::FunctionView<float(int64_t)> fx,
rtc::FunctionView<rtc::Optional<ResultType>(const DataType&)> fy,
const IterableType& data_view,
int64_t begin_time,
int64_t end_time,
int64_t window_duration_us,
int64_t step,
TimeSeries* result) {
size_t window_index_begin = 0;
size_t window_index_end = 0;
ResultType sum_in_window = 0;
for (int64_t t = begin_time; t < end_time + step; t += step) {
while (window_index_end < data_view.size() &&
data_view[window_index_end].log_time_us() < t) {
rtc::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 - window_duration_us) {
rtc::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>(window_duration_us) / kNumMicrosecsPerSec;
float x = fx(t);
float y = sum_in_window / window_duration_s;
result->points.emplace_back(x, y);
}
}
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.
std::stringstream 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.str();
}
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 ParsedRtcEventLogNew& log,
bool normalize_time)
: parsed_log_(log),
window_duration_(250000),
step_(10000),
normalize_time_(normalize_time) {
begin_time_ = parsed_log_.first_timestamp();
end_time_ = parsed_log_.last_timestamp();
if (end_time_ < begin_time_) {
RTC_LOG(LS_WARNING) << "No useful events in the log.";
begin_time_ = end_time_ = 0;
}
call_duration_s_ = ToCallTimeSec(end_time_);
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;
rtc::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(end_time_)));
}
}
RTC_LOG(LS_INFO) << "Found " << log_segments_.size()
<< " (LOG_START, LOG_END) segments in log.";
}
class BitrateObserver : public NetworkChangedObserver,
public RemoteBitrateObserver {
public:
BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}
void OnNetworkChanged(uint32_t bitrate_bps,
uint8_t fraction_lost,
int64_t rtt_ms,
int64_t probing_interval_ms) override {
last_bitrate_bps_ = bitrate_bps;
bitrate_updated_ = true;
}
void 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_;
};
int64_t EventLogAnalyzer::ToCallTimeUs(int64_t timestamp) const {
int64_t begin_time = 0;
if (normalize_time_) {
begin_time = begin_time_;
}
return timestamp - begin_time;
}
float EventLogAnalyzer::ToCallTimeSec(int64_t timestamp) const {
return static_cast<float>(ToCallTimeUs(timestamp)) / kNumMicrosecsPerSec;
}
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 rtc::Optional<float>(packet.total_length);
};
auto ToCallTime = [this](const LoggedRtpPacket& packet) {
return this->ToCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedRtpPacket>(ToCallTime, GetPacketSize,
stream.packet_view, &time_series);
plot->AppendTimeSeries(std::move(time_series));
}
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Packet size (bytes)", kBottomMargin,
kTopMargin);
plot->SetTitle(GetDirectionAsString(direction) + " RTP packets");
}
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 = ToCallTimeSec(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(ToCallTimeSec(begin_time_), call_duration_s_, "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;
rtc::Optional<int64_t> last_playout_ms;
TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
for (const auto& playout_event : playout_stream.second) {
float x = ToCallTimeSec(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(ToCallTimeSec(begin_time_), call_duration_s_, "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 = ToCallTimeSec(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(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetYAxis(-127, 0, "Audio level (dBov)", kBottomMargin,
kTopMargin);
plot->SetTitle(GetDirectionAsString(direction) + " audio level");
}
// For each SSRC, plot the time between the consecutive playouts.
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->ToCallTimeSec(packet.log_time_us());
};
ProcessPairs<LoggedRtpPacketIncoming, float>(
ToCallTime, GetSequenceNumberDiff, stream.incoming_packets,
&time_series);
plot->AppendTimeSeries(std::move(time_series));
}
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Difference since last packet", kBottomMargin,
kTopMargin);
plot->SetTitle("Sequence number");
}
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.size() == 0) {
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 = begin_time_; t < 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 = ToCallTimeSec(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(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Estimated loss rate (%)", kBottomMargin,
kTopMargin);
plot->SetTitle("Estimated incoming loss rate");
}
void EventLogAnalyzer::CreateIncomingDelayDeltaGraph(Plot* plot) {
for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(kIncomingPacket)) {
// Filter on SSRC.
if (!MatchingSsrc(stream.ssrc, desired_ssrc_) ||
IsAudioSsrc(kIncomingPacket, stream.ssrc) ||
!IsVideoSsrc(kIncomingPacket, stream.ssrc) ||
IsRtxSsrc(kIncomingPacket, stream.ssrc)) {
continue;
}
TimeSeries capture_time_data(
GetStreamName(kIncomingPacket, stream.ssrc) + " capture-time",
LineStyle::kBar);
auto ToCallTime = [this](const LoggedRtpPacket& packet) {
return this->ToCallTimeSec(packet.log_time_us());
};
ProcessPairs<LoggedRtpPacket, double>(
ToCallTime, NetworkDelayDiff_CaptureTime, stream.packet_view,
&capture_time_data);
plot->AppendTimeSeries(std::move(capture_time_data));
TimeSeries send_time_data(
GetStreamName(kIncomingPacket, stream.ssrc) + " abs-send-time",
LineStyle::kBar);
ProcessPairs<LoggedRtpPacket, double>(ToCallTime,
NetworkDelayDiff_AbsSendTime,
stream.packet_view, &send_time_data);
plot->AppendTimeSeries(std::move(send_time_data));
}
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Latency change (ms)", kBottomMargin,
kTopMargin);
plot->SetTitle("Network latency difference between consecutive packets");
}
void EventLogAnalyzer::CreateIncomingDelayGraph(Plot* plot) {
for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(kIncomingPacket)) {
// Filter on SSRC.
if (!MatchingSsrc(stream.ssrc, desired_ssrc_) ||
IsAudioSsrc(kIncomingPacket, stream.ssrc) ||
!IsVideoSsrc(kIncomingPacket, stream.ssrc) ||
IsRtxSsrc(kIncomingPacket, stream.ssrc)) {
continue;
}
TimeSeries capture_time_data(
GetStreamName(kIncomingPacket, stream.ssrc) + " capture-time",
LineStyle::kLine);
auto ToCallTime = [this](const LoggedRtpPacket& packet) {
return this->ToCallTimeSec(packet.log_time_us());
};
AccumulatePairs<LoggedRtpPacket, double>(
ToCallTime, NetworkDelayDiff_CaptureTime, stream.packet_view,
&capture_time_data);
plot->AppendTimeSeries(std::move(capture_time_data));
TimeSeries send_time_data(
GetStreamName(kIncomingPacket, stream.ssrc) + " abs-send-time",
LineStyle::kLine);
AccumulatePairs<LoggedRtpPacket, double>(
ToCallTime, NetworkDelayDiff_AbsSendTime, stream.packet_view,
&send_time_data);
plot->AppendTimeSeries(std::move(send_time_data));
}
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Latency change (ms)", kBottomMargin,
kTopMargin);
plot->SetTitle("Network latency (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 = ToCallTimeSec(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(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 10, "Percent lost packets", kBottomMargin,
kTopMargin);
plot->SetTitle("Reported packet loss");
}
// 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;
// Calculate a moving average of the bitrate and store in a TimeSeries.
TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
for (int64_t time = begin_time_; time < end_time_ + step_; time += 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 - 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>(window_duration_) / kNumMicrosecsPerSec;
float x = ToCallTimeSec(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 = ToCallTimeSec(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));
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "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;
// Calculate a moving average of the bitrate and store in a TimeSeries.
TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
for (int64_t time = begin_time_; time < end_time_ + step_; time += 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 - 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>(window_duration_) / kNumMicrosecsPerSec;
float x = ToCallTimeSec(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 = ToCallTimeSec(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 = ToCallTimeSec(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, end_time_);
TimeSeries created_series("Probe cluster created.", LineStyle::kNone,
PointStyle::kHighlight);
for (auto& cluster : parsed_log_.bwe_probe_cluster_created_events()) {
float x = ToCallTimeSec(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 = ToCallTimeSec(result.log_time_us());
float y = static_cast<float>(result.bitrate_bps) / 1000;
result_series.points.emplace_back(x, y);
}
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 = ToCallTimeSec(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 = ToCallTimeSec(alr_start);
alr_state.intervals.emplace_back(x, y);
previously_in_alr = false;
}
}
if (previously_in_alr) {
float x = ToCallTimeSec(alr_start);
float y = ToCallTimeSec(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->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 = ToCallTimeSec(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));
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "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;
};
auto ToCallTime = [this](int64_t time) {
return this->ToCallTimeSec(time);
};
MovingAverage<LoggedRtpPacket, double>(
ToCallTime, GetPacketSizeKilobits, stream.packet_view, begin_time_,
end_time_, window_duration_, step_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
}
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
plot->SetTitle(GetDirectionAsString(direction) + " bitrate per stream");
}
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;
RtcEventLogNullImpl null_event_log;
PacketRouter packet_router;
PacedSender pacer(&clock, &packet_router, &null_event_log);
SendSideCongestionController cc(&clock, &observer, &null_event_log, &pacer);
// TODO(holmer): Log the call config and use that here instead.
static const uint32_t kDefaultStartBitrateBps = 300000;
cc.SetBweBitrates(0, kDefaultStartBitrateBps, -1);
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();
};
auto NextProcessTime = [&]() {
if (rtcp_iterator != incoming_rtcp.end() ||
rtp_iterator != outgoing_rtp.end()) {
return clock.TimeInMicroseconds() +
std::max<int64_t>(cc.TimeUntilNextProcess() * 1000, 0);
}
return std::numeric_limits<int64_t>::max();
};
RateStatistics acked_bitrate(250, 8000);
#if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
// 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, runnning 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
// SendSideCongestionController since that would lead to duplicate outputs.
AcknowledgedBitrateEstimator acknowledged_bitrate_estimator(
rtc::MakeUnique<BitrateEstimator>());
#endif // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
int64_t time_us = std::min(NextRtpTime(), NextRtcpTime());
int64_t last_update_us = 0;
while (time_us != std::numeric_limits<int64_t>::max()) {
clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
cc.OnTransportFeedback(rtcp_iterator->transport_feedback);
std::vector<PacketFeedback> feedback = cc.GetTransportFeedbackVector();
SortPacketFeedbackVector(&feedback);
rtc::Optional<uint32_t> bitrate_bps;
if (!feedback.empty()) {
#if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
acknowledged_bitrate_estimator.IncomingPacketFeedbackVector(feedback);
#endif // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
for (const PacketFeedback& packet : feedback)
acked_bitrate.Update(packet.payload_size, packet.arrival_time_ms);
bitrate_bps = acked_bitrate.Rate(feedback.back().arrival_time_ms);
}
float x = ToCallTimeSec(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_bps().value_or(0) / 1000;
acked_estimate_time_series.points.emplace_back(x, y);
#endif // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
++rtcp_iterator;
}
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);
cc.AddPacket(rtp_packet.rtp.header.ssrc,
rtp_packet.rtp.header.extension.transportSequenceNumber,
rtp_packet.rtp.total_length, PacedPacketInfo());
rtc::SentPacket sent_packet(
rtp_packet.rtp.header.extension.transportSequenceNumber,
rtp_packet.rtp.log_time_us() / 1000);
cc.OnSentPacket(sent_packet);
}
++rtp_iterator;
}
if (clock.TimeInMicroseconds() >= NextProcessTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
cc.Process();
}
if (observer.GetAndResetBitrateUpdated() ||
time_us - last_update_us >= 1e6) {
uint32_t y = observer.last_bitrate_bps() / 1000;
float x = ToCallTimeSec(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(ToCallTimeSec(begin_time_), call_duration_s_, "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);
rtc::Optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
if (bitrate_bps) {
uint32_t y = *bitrate_bps / 1000;
float x = ToCallTimeSec(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 = ToCallTimeSec(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(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
plot->SetTitle("Simulated receive-side BWE behavior");
}
void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(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);
TransportFeedbackAdapter feedback_adapter(&clock);
TimeSeries late_feedback_series("Late feedback results.", LineStyle::kNone,
PointStyle::kHighlight);
TimeSeries time_series("Network Delay Change", LineStyle::kLine,
PointStyle::kHighlight);
int64_t estimated_base_delay_ms = std::numeric_limits<int64_t>::max();
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 time_us = std::min(NextRtpTime(), NextRtcpTime());
int64_t prev_y = 0;
while (time_us != std::numeric_limits<int64_t>::max()) {
clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
feedback_adapter.OnTransportFeedback(rtcp_iterator->transport_feedback);
std::vector<PacketFeedback> feedback =
feedback_adapter.GetTransportFeedbackVector();
SortPacketFeedbackVector(&feedback);
for (const PacketFeedback& packet : feedback) {
float x = ToCallTimeSec(clock.TimeInMicroseconds());
if (packet.send_time_ms == PacketFeedback::kNoSendTime) {
late_feedback_series.points.emplace_back(x, prev_y);
continue;
}
int64_t y = packet.arrival_time_ms - packet.send_time_ms;
prev_y = y;
estimated_base_delay_ms = std::min(y, estimated_base_delay_ms);
time_series.points.emplace_back(x, y);
}
++rtcp_iterator;
}
if (clock.TimeInMicroseconds() >= NextRtpTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
const RtpPacketType& rtp_packet = *rtp_iterator->second;
if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
feedback_adapter.AddPacket(
rtp_packet.rtp.header.ssrc,
rtp_packet.rtp.header.extension.transportSequenceNumber,
rtp_packet.rtp.total_length, PacedPacketInfo());
feedback_adapter.OnSentPacket(
rtp_packet.rtp.header.extension.transportSequenceNumber,
rtp_packet.rtp.log_time_us() / 1000);
}
++rtp_iterator;
}
time_us = std::min(NextRtpTime(), NextRtcpTime());
}
// We assume that the base network delay (w/o queues) is the min delay
// observed during the call.
for (TimeSeriesPoint& point : time_series.points)
point.y -= estimated_base_delay_ms;
for (TimeSeriesPoint& point : late_feedback_series.points)
point.y -= estimated_base_delay_ms;
// Add the data set to the plot.
plot->AppendTimeSeriesIfNotEmpty(std::move(time_series));
plot->AppendTimeSeriesIfNotEmpty(std::move(late_feedback_series));
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
plot->SetTitle("Network Delay Change.");
}
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 (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;
rtc::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 = ToCallTimeSec(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(ToCallTimeSec(begin_time_), call_duration_s_, "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 = ToCallTimeSec(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 = ToCallTimeSec(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(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "RTP timestamp", kBottomMargin, kTopMargin);
plot->SetTitle(GetDirectionAsString(direction) + " timestamps");
}
void EventLogAnalyzer::CreateAudioEncoderTargetBitrateGraph(Plot* plot) {
TimeSeries time_series("Audio encoder target bitrate", LineStyle::kLine,
PointStyle::kHighlight);
auto GetAnaBitrateBps = [](const LoggedAudioNetworkAdaptationEvent& ana_event)
-> rtc::Optional<float> {
if (ana_event.config.bitrate_bps)
return rtc::Optional<float>(
static_cast<float>(*ana_event.config.bitrate_bps));
return rtc::nullopt;
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaBitrateBps,
parsed_log_.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "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 rtc::Optional<float>(
static_cast<float>(*ana_event.config.frame_length_ms));
return rtc::Optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaFrameLengthMs,
parsed_log_.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "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 rtc::Optional<float>(static_cast<float>(
*ana_event.config.uplink_packet_loss_fraction));
return rtc::Optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaPacketLoss,
parsed_log_.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "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 rtc::Optional<float>(
static_cast<float>(*ana_event.config.enable_fec));
return rtc::Optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaFecEnabled,
parsed_log_.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "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 rtc::Optional<float>(
static_cast<float>(*ana_event.config.enable_dtx));
return rtc::Optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaDtxEnabled,
parsed_log_.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "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 rtc::Optional<float>(
static_cast<float>(*ana_event.config.num_channels));
return rtc::Optional<float>();
};
auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
return this->ToCallTimeSec(packet.log_time_us());
};
ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
ToCallTime, GetAnaNumChannels,
parsed_log_.audio_network_adaptation_events(), &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "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,
rtc::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);
}
rtc::Optional<int64_t> NextPacketTime() const override {
if (packet_stream_it_ == packet_stream_.end()) {
return rtc::nullopt;
}
if (end_time_ms_ && packet_stream_it_->rtp.log_time_ms() > *end_time_ms_) {
return rtc::nullopt;
}
return packet_stream_it_->rtp.log_time_ms();
}
rtc::Optional<int64_t> NextOutputEventTime() const override {
if (output_events_it_ == output_events_end_) {
return rtc::nullopt;
}
if (end_time_ms_ && output_events_it_->log_time_ms() > *end_time_ms_) {
return rtc::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(); }
rtc::Optional<RTPHeader> NextHeader() const override {
if (packet_stream_it_ == packet_stream_.end()) {
return rtc::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 rtc::Optional<int64_t> end_time_ms_;
};
namespace {
// 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,
rtc::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());
test::NetEqTest::DecoderMap codecs;
// Create a "replacement decoder" that produces the decoded audio by reading
// from a file rather than from the encoded payloads.
std::unique_ptr<test::ResampleInputAudioFile> replacement_file(
new test::ResampleInputAudioFile(replacement_file_name,
file_sample_rate_hz));
replacement_file->set_output_rate_hz(48000);
std::unique_ptr<AudioDecoder> replacement_decoder(
new test::FakeDecodeFromFile(std::move(replacement_file), 48000, false));
test::NetEqTest::ExtDecoderMap ext_codecs;
ext_codecs[kReplacementPt] = {replacement_decoder.get(),
NetEqDecoder::kDecoderArbitrary,
"replacement codec"};
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, codecs, ext_codecs, 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();
}
rtc::Optional<int64_t> end_time_ms =
log_segments_.empty()
? rtc::nullopt
: rtc::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;
}
// Plots the jitter buffer delay profile. This will plot only for the first
// incoming audio SSRC. If the stream contains more than one incoming audio
// SSRC, all but the first will be ignored.
void EventLogAnalyzer::CreateAudioJitterBufferGraph(
const NetEqStatsGetterMap& neteq_stats,
Plot* plot) const {
if (neteq_stats.size() < 1)
return;
const uint32_t ssrc = neteq_stats.begin()->first;
std::vector<float> send_times_s;
std::vector<float> arrival_delay_ms;
std::vector<float> corrected_arrival_delay_ms;
std::vector<rtc::Optional<float>> playout_delay_ms;
std::vector<rtc::Optional<float>> target_delay_ms;
neteq_stats.at(ssrc)->delay_analyzer()->CreateGraphs(
&send_times_s, &arrival_delay_ms, &corrected_arrival_delay_ms,
&playout_delay_ms, &target_delay_ms);
RTC_DCHECK_EQ(send_times_s.size(), arrival_delay_ms.size());
RTC_DCHECK_EQ(send_times_s.size(), corrected_arrival_delay_ms.size());
RTC_DCHECK_EQ(send_times_s.size(), playout_delay_ms.size());
RTC_DCHECK_EQ(send_times_s.size(), target_delay_ms.size());
std::map<uint32_t, TimeSeries> time_series_packet_arrival;
std::map<uint32_t, TimeSeries> time_series_relative_packet_arrival;
std::map<uint32_t, TimeSeries> time_series_play_time;
std::map<uint32_t, TimeSeries> time_series_target_time;
float min_y_axis = 0.f;
float max_y_axis = 0.f;
for (size_t i = 0; i < send_times_s.size(); ++i) {
time_series_packet_arrival[ssrc].points.emplace_back(
TimeSeriesPoint(send_times_s[i], arrival_delay_ms[i]));
time_series_relative_packet_arrival[ssrc].points.emplace_back(
TimeSeriesPoint(send_times_s[i], corrected_arrival_delay_ms[i]));
min_y_axis = std::min(min_y_axis, corrected_arrival_delay_ms[i]);
max_y_axis = std::max(max_y_axis, corrected_arrival_delay_ms[i]);
if (playout_delay_ms[i]) {
time_series_play_time[ssrc].points.emplace_back(
TimeSeriesPoint(send_times_s[i], *playout_delay_ms[i]));
min_y_axis = std::min(min_y_axis, *playout_delay_ms[i]);
max_y_axis = std::max(max_y_axis, *playout_delay_ms[i]);
}
if (target_delay_ms[i]) {
time_series_target_time[ssrc].points.emplace_back(
TimeSeriesPoint(send_times_s[i], *target_delay_ms[i]));
min_y_axis = std::min(min_y_axis, *target_delay_ms[i]);
max_y_axis = std::max(max_y_axis, *target_delay_ms[i]);
}
}
// This code is adapted for a single stream. The creation of the streams above
// guarantee that no more than one steam is included. If multiple streams are
// to be plotted, they should likely be given distinct labels below.
RTC_DCHECK_EQ(time_series_relative_packet_arrival.size(), 1);
for (auto& series : time_series_relative_packet_arrival) {
series.second.label = "Relative packet arrival delay";
series.second.line_style = LineStyle::kLine;
plot->AppendTimeSeries(std::move(series.second));
}
RTC_DCHECK_EQ(time_series_play_time.size(), 1);
for (auto& series : time_series_play_time) {
series.second.label = "Playout delay";
series.second.line_style = LineStyle::kLine;
plot->AppendTimeSeries(std::move(series.second));
}
RTC_DCHECK_EQ(time_series_target_time.size(), 1);
for (auto& series : time_series_target_time) {
series.second.label = "Target delay";
series.second.line_style = LineStyle::kLine;
series.second.point_style = PointStyle::kHighlight;
plot->AppendTimeSeries(std::move(series.second));
}
plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetYAxis(min_y_axis, max_y_axis, "Relative delay (ms)", kBottomMargin,
kTopMargin);
plot->SetTitle("NetEq timing for " + GetStreamName(kIncomingPacket, ssrc));
}
void EventLogAnalyzer::CreateNetEqStatsGraph(
const NetEqStatsGetterMap& neteq_stats,
rtc::FunctionView<float(const NetEqNetworkStatistics&)> stats_extractor,
const std::string& plot_name,
Plot* plot) const {
if (neteq_stats.size() < 1)
return;
std::map<uint32_t, TimeSeries> time_series;
float min_y_axis = std::numeric_limits<float>::max();
float max_y_axis = std::numeric_limits<float>::min();
for (const auto& st : neteq_stats) {
const uint32_t ssrc = st.first;
const auto& stats = st.second->stats();
for (size_t i = 0; i < stats.size(); ++i) {
const float time = ToCallTimeSec(stats[i].first * 1000); // ms to us.
const float value = stats_extractor(stats[i].second);
time_series[ssrc].points.emplace_back(TimeSeriesPoint(time, value));
min_y_axis = std::min(min_y_axis, value);
max_y_axis = std::max(max_y_axis, 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(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetYAxis(min_y_axis, max_y_axis, plot_name, kBottomMargin, kTopMargin);
plot->SetTitle(plot_name);
}
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 = ToCallTimeSec(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(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 3, "Numeric Config Type", kBottomMargin,
kTopMargin);
plot->SetTitle("[IceEventLog] ICE candidate pair configs");
}
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) {
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 = ToCallTimeSec(event.log_time_us());
float y = static_cast<float>(event.type);
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(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 4, "Numeric Connectivity State", kBottomMargin,
kTopMargin);
plot->SetTitle("[IceEventLog] ICE connectivity checks");
}
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;
rtc::Optional<int64_t> last_seq_num;
SeqNumUnwrapper<uint32_t> capture_time_unwrapper;
rtc::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, ToCallTimeSec(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, ToCallTimeSec(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);
}
rtc::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, ToCallTimeSec(timestamp), duration / 1000);
}
last_rtp_time.emplace(timestamp);
}
rtc::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, ToCallTimeSec(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, ToCallTimeSec(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