Google Git
Sign in
webrtc / src / c20978e581a3f71d96ef2f6078fbad350922fff8 / . / webrtc / rtc_tools / event_log_visualizer / analyzer.cc
blob: 78993f4f7cada132678d8aa1456ea339285ee1d0 [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 "webrtc/rtc_tools/event_log_visualizer/analyzer.h"
#include <algorithm>
#include <limits>
#include <map>
#include <sstream>
#include <string>
#include <utility>
#include "webrtc/call/audio_receive_stream.h"
#include "webrtc/call/audio_send_stream.h"
#include "webrtc/call/call.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
#include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_test.h"
#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/format_macros.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/rate_statistics.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
namespace webrtc {
namespace plotting {
namespace {
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 1000000 to convert to microseconds.
static constexpr double kTimestampToMicroSec =
1000000.0 / 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) {
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;
}
// Return default values for header extensions, to use on streams without stored
// mapping data. Currently this only applies to audio streams, since the mapping
// is not stored in the event log.
// TODO(ivoc): Remove this once this mapping is stored in the event log for
// audio streams. Tracking bug: webrtc:6399
webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() {
webrtc::RtpHeaderExtensionMap default_map;
default_map.Register<AudioLevel>(webrtc::RtpExtension::kAudioLevelDefaultId);
default_map.Register<AbsoluteSendTime>(
webrtc::RtpExtension::kAbsSendTimeDefaultId);
return default_map;
}
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.timestamp - old_packet.timestamp;
double delay_change_us =
recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff);
return rtc::Optional<double>(delay_change_us / 1000);
} else {
return rtc::Optional<double>();
}
}
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.timestamp - old_packet.timestamp;
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) {
LOG(LS_WARNING) << "Very large delay change. Timestamps correct?";
LOG(LS_WARNING) << "Old capture time " << old_packet.header.timestamp
<< ", received time " << old_packet.timestamp;
LOG(LS_WARNING) << "New capture time " << new_packet.header.timestamp
<< ", received time " << new_packet.timestamp;
LOG(LS_WARNING) << "Receive time difference " << recv_time_diff << " = "
<< static_cast<double>(recv_time_diff) / 1000000 << "s";
LOG(LS_WARNING) << "Send time difference " << send_time_diff << " = "
<< static_cast<double>(send_time_diff) / kVideoSampleRate
<< "s";
}
return rtc::Optional<double>(delay_change);
}
// For each element in data, use |get_y()| to extract a y-coordinate and
// store the result in a TimeSeries.
template <typename DataType>
void ProcessPoints(
rtc::FunctionView<rtc::Optional<float>(const DataType&)> get_y,
const std::vector<DataType>& data,
uint64_t begin_time,
TimeSeries* result) {
for (size_t i = 0; i < data.size(); i++) {
float x = static_cast<float>(data[i].timestamp - begin_time) / 1000000;
rtc::Optional<float> y = get_y(data[i]);
if (y)
result->points.emplace_back(x, *y);
}
}
// For each pair of adjacent elements in |data|, use |get_y| 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>
void ProcessPairs(
rtc::FunctionView<rtc::Optional<ResultType>(const DataType&,
const DataType&)> get_y,
const std::vector<DataType>& data,
uint64_t begin_time,
TimeSeries* result) {
for (size_t i = 1; i < data.size(); i++) {
float x = static_cast<float>(data[i].timestamp - begin_time) / 1000000;
rtc::Optional<ResultType> y = get_y(data[i - 1], data[i]);
if (y)
result->points.emplace_back(x, static_cast<float>(*y));
}
}
// For each element in data, use |extract()| to extract a y-coordinate and
// store the result in a TimeSeries.
template <typename DataType, typename ResultType>
void AccumulatePoints(
rtc::FunctionView<rtc::Optional<ResultType>(const DataType&)> extract,
const std::vector<DataType>& data,
uint64_t begin_time,
TimeSeries* result) {
ResultType sum = 0;
for (size_t i = 0; i < data.size(); i++) {
float x = static_cast<float>(data[i].timestamp - begin_time) / 1000000;
rtc::Optional<ResultType> y = extract(data[i]);
if (y) {
sum += *y;
result->points.emplace_back(x, static_cast<float>(sum));
}
}
}
// For each pair of adjacent elements in |data|, use |extract()| 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>
void AccumulatePairs(
rtc::FunctionView<rtc::Optional<ResultType>(const DataType&,
const DataType&)> extract,
const std::vector<DataType>& data,
uint64_t begin_time,
TimeSeries* result) {
ResultType sum = 0;
for (size_t i = 1; i < data.size(); i++) {
float x = static_cast<float>(data[i].timestamp - begin_time) / 1000000;
rtc::Optional<ResultType> y = extract(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>
void MovingAverage(
rtc::FunctionView<rtc::Optional<ResultType>(const DataType&)> extract,
const std::vector<DataType>& data,
uint64_t begin_time,
uint64_t end_time,
uint64_t window_duration_us,
uint64_t step,
webrtc::plotting::TimeSeries* result) {
size_t window_index_begin = 0;
size_t window_index_end = 0;
ResultType sum_in_window = 0;
for (uint64_t t = begin_time; t < end_time + step; t += step) {
while (window_index_end < data.size() &&
data[window_index_end].timestamp < t) {
rtc::Optional<ResultType> value = extract(data[window_index_end]);
if (value)
sum_in_window += *value;
++window_index_end;
}
while (window_index_begin < data.size() &&
data[window_index_begin].timestamp < t - window_duration_us) {
rtc::Optional<ResultType> value = extract(data[window_index_begin]);
if (value)
sum_in_window -= *value;
++window_index_begin;
}
float window_duration_s = static_cast<float>(window_duration_us) / 1000000;
float x = static_cast<float>(t - begin_time) / 1000000;
float y = sum_in_window / window_duration_s;
result->points.emplace_back(x, y);
}
}
} // namespace
EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
: parsed_log_(log), window_duration_(250000), step_(10000) {
uint64_t first_timestamp = std::numeric_limits<uint64_t>::max();
uint64_t last_timestamp = std::numeric_limits<uint64_t>::min();
PacketDirection direction;
uint8_t header[IP_PACKET_SIZE];
size_t header_length;
size_t total_length;
uint8_t last_incoming_rtcp_packet[IP_PACKET_SIZE];
uint8_t last_incoming_rtcp_packet_length = 0;
// Make a default extension map for streams without configuration information.
// TODO(ivoc): Once configuration of audio streams is stored in the event log,
// this can be removed. Tracking bug: webrtc:6399
RtpHeaderExtensionMap default_extension_map = GetDefaultHeaderExtensionMap();
rtc::Optional<uint64_t> last_log_start;
for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
if (event_type != ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT &&
event_type != ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT &&
event_type != ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT &&
event_type != ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT &&
event_type != ParsedRtcEventLog::LOG_START &&
event_type != ParsedRtcEventLog::LOG_END) {
uint64_t timestamp = parsed_log_.GetTimestamp(i);
first_timestamp = std::min(first_timestamp, timestamp);
last_timestamp = std::max(last_timestamp, timestamp);
}
switch (parsed_log_.GetEventType(i)) {
case ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: {
rtclog::StreamConfig config = parsed_log_.GetVideoReceiveConfig(i);
StreamId stream(config.remote_ssrc, kIncomingPacket);
video_ssrcs_.insert(stream);
StreamId rtx_stream(config.rtx_ssrc, kIncomingPacket);
video_ssrcs_.insert(rtx_stream);
rtx_ssrcs_.insert(rtx_stream);
break;
}
case ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT: {
std::vector<rtclog::StreamConfig> configs =
parsed_log_.GetVideoSendConfig(i);
for (const auto& config : configs) {
StreamId stream(config.local_ssrc, kOutgoingPacket);
video_ssrcs_.insert(stream);
StreamId rtx_stream(config.rtx_ssrc, kOutgoingPacket);
video_ssrcs_.insert(rtx_stream);
rtx_ssrcs_.insert(rtx_stream);
}
break;
}
case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: {
rtclog::StreamConfig config = parsed_log_.GetAudioReceiveConfig(i);
StreamId stream(config.remote_ssrc, kIncomingPacket);
audio_ssrcs_.insert(stream);
break;
}
case ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
rtclog::StreamConfig config = parsed_log_.GetAudioSendConfig(i);
StreamId stream(config.local_ssrc, kOutgoingPacket);
audio_ssrcs_.insert(stream);
break;
}
case ParsedRtcEventLog::RTP_EVENT: {
RtpHeaderExtensionMap* extension_map = parsed_log_.GetRtpHeader(
i, &direction, header, &header_length, &total_length);
RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
RTPHeader parsed_header;
if (extension_map != nullptr) {
rtp_parser.Parse(&parsed_header, extension_map);
} else {
// Use the default extension map.
// TODO(ivoc): Once configuration of audio streams is stored in the
// event log, this can be removed.
// Tracking bug: webrtc:6399
rtp_parser.Parse(&parsed_header, &default_extension_map);
}
uint64_t timestamp = parsed_log_.GetTimestamp(i);
StreamId stream(parsed_header.ssrc, direction);
rtp_packets_[stream].push_back(
LoggedRtpPacket(timestamp, parsed_header, total_length));
break;
}
case ParsedRtcEventLog::RTCP_EVENT: {
uint8_t packet[IP_PACKET_SIZE];
parsed_log_.GetRtcpPacket(i, &direction, packet, &total_length);
// Currently incoming RTCP packets are logged twice, both for audio and
// video. Only act on one of them. Compare against the previous parsed
// incoming RTCP packet.
if (direction == webrtc::kIncomingPacket) {
RTC_CHECK_LE(total_length, IP_PACKET_SIZE);
if (total_length == last_incoming_rtcp_packet_length &&
memcmp(last_incoming_rtcp_packet, packet, total_length) == 0) {
continue;
} else {
memcpy(last_incoming_rtcp_packet, packet, total_length);
last_incoming_rtcp_packet_length = total_length;
}
}
rtcp::CommonHeader header;
const uint8_t* packet_end = packet + total_length;
for (const uint8_t* block = packet; block < packet_end;
block = header.NextPacket()) {
RTC_CHECK(header.Parse(block, packet_end - block));
if (header.type() == rtcp::TransportFeedback::kPacketType &&
header.fmt() == rtcp::TransportFeedback::kFeedbackMessageType) {
std::unique_ptr<rtcp::TransportFeedback> rtcp_packet(
rtc::MakeUnique<rtcp::TransportFeedback>());
if (rtcp_packet->Parse(header)) {
uint32_t ssrc = rtcp_packet->sender_ssrc();
StreamId stream(ssrc, direction);
uint64_t timestamp = parsed_log_.GetTimestamp(i);
rtcp_packets_[stream].push_back(LoggedRtcpPacket(
timestamp, kRtcpTransportFeedback, std::move(rtcp_packet)));
}
} else if (header.type() == rtcp::SenderReport::kPacketType) {
std::unique_ptr<rtcp::SenderReport> rtcp_packet(
rtc::MakeUnique<rtcp::SenderReport>());
if (rtcp_packet->Parse(header)) {
uint32_t ssrc = rtcp_packet->sender_ssrc();
StreamId stream(ssrc, direction);
uint64_t timestamp = parsed_log_.GetTimestamp(i);
rtcp_packets_[stream].push_back(
LoggedRtcpPacket(timestamp, kRtcpSr, std::move(rtcp_packet)));
}
} else if (header.type() == rtcp::ReceiverReport::kPacketType) {
std::unique_ptr<rtcp::ReceiverReport> rtcp_packet(
rtc::MakeUnique<rtcp::ReceiverReport>());
if (rtcp_packet->Parse(header)) {
uint32_t ssrc = rtcp_packet->sender_ssrc();
StreamId stream(ssrc, direction);
uint64_t timestamp = parsed_log_.GetTimestamp(i);
rtcp_packets_[stream].push_back(
LoggedRtcpPacket(timestamp, kRtcpRr, std::move(rtcp_packet)));
}
} else if (header.type() == rtcp::Remb::kPacketType &&
header.fmt() == rtcp::Remb::kFeedbackMessageType) {
std::unique_ptr<rtcp::Remb> rtcp_packet(
rtc::MakeUnique<rtcp::Remb>());
if (rtcp_packet->Parse(header)) {
uint32_t ssrc = rtcp_packet->sender_ssrc();
StreamId stream(ssrc, direction);
uint64_t timestamp = parsed_log_.GetTimestamp(i);
rtcp_packets_[stream].push_back(LoggedRtcpPacket(
timestamp, kRtcpRemb, std::move(rtcp_packet)));
}
}
}
break;
}
case ParsedRtcEventLog::LOG_START: {
if (last_log_start) {
// A LOG_END event was missing. Use last_timestamp.
RTC_DCHECK_GE(last_timestamp, *last_log_start);
log_segments_.push_back(
std::make_pair(*last_log_start, last_timestamp));
}
last_log_start = rtc::Optional<uint64_t>(parsed_log_.GetTimestamp(i));
break;
}
case ParsedRtcEventLog::LOG_END: {
RTC_DCHECK(last_log_start);
log_segments_.push_back(
std::make_pair(*last_log_start, parsed_log_.GetTimestamp(i)));
last_log_start.reset();
break;
}
case ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT: {
uint32_t this_ssrc;
parsed_log_.GetAudioPlayout(i, &this_ssrc);
audio_playout_events_[this_ssrc].push_back(parsed_log_.GetTimestamp(i));
break;
}
case ParsedRtcEventLog::LOSS_BASED_BWE_UPDATE: {
LossBasedBweUpdate bwe_update;
bwe_update.timestamp = parsed_log_.GetTimestamp(i);
parsed_log_.GetLossBasedBweUpdate(i, &bwe_update.new_bitrate,
&bwe_update.fraction_loss,
&bwe_update.expected_packets);
bwe_loss_updates_.push_back(bwe_update);
break;
}
case ParsedRtcEventLog::DELAY_BASED_BWE_UPDATE: {
bwe_delay_updates_.push_back(parsed_log_.GetDelayBasedBweUpdate(i));
break;
}
case ParsedRtcEventLog::AUDIO_NETWORK_ADAPTATION_EVENT: {
AudioNetworkAdaptationEvent ana_event;
ana_event.timestamp = parsed_log_.GetTimestamp(i);
parsed_log_.GetAudioNetworkAdaptation(i, &ana_event.config);
audio_network_adaptation_events_.push_back(ana_event);
break;
}
case ParsedRtcEventLog::BWE_PROBE_CLUSTER_CREATED_EVENT: {
bwe_probe_cluster_created_events_.push_back(
parsed_log_.GetBweProbeClusterCreated(i));
break;
}
case ParsedRtcEventLog::BWE_PROBE_RESULT_EVENT: {
bwe_probe_result_events_.push_back(parsed_log_.GetBweProbeResult(i));
break;
}
case ParsedRtcEventLog::UNKNOWN_EVENT: {
break;
}
}
}
if (last_timestamp < first_timestamp) {
// No useful events in the log.
first_timestamp = last_timestamp = 0;
}
begin_time_ = first_timestamp;
end_time_ = last_timestamp;
call_duration_s_ = static_cast<float>(end_time_ - begin_time_) / 1000000;
if (last_log_start) {
// The log was missing the last LOG_END event. Fake it.
log_segments_.push_back(std::make_pair(*last_log_start, end_time_));
}
}
class BitrateObserver : public CongestionController::Observer,
public RemoteBitrateObserver {
public:
BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}
// TODO(minyue): remove this when old OnNetworkChanged is deprecated. See
// https://bugs.chromium.org/p/webrtc/issues/detail?id=6796
using CongestionController::Observer::OnNetworkChanged;
void OnNetworkChanged(uint32_t bitrate_bps,
uint8_t fraction_loss,
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_;
};
bool EventLogAnalyzer::IsRtxSsrc(StreamId stream_id) const {
return rtx_ssrcs_.count(stream_id) == 1;
}
bool EventLogAnalyzer::IsVideoSsrc(StreamId stream_id) const {
return video_ssrcs_.count(stream_id) == 1;
}
bool EventLogAnalyzer::IsAudioSsrc(StreamId stream_id) const {
return audio_ssrcs_.count(stream_id) == 1;
}
std::string EventLogAnalyzer::GetStreamName(StreamId stream_id) const {
std::stringstream name;
if (IsAudioSsrc(stream_id)) {
name << "Audio ";
} else if (IsVideoSsrc(stream_id)) {
name << "Video ";
} else {
name << "Unknown ";
}
if (IsRtxSsrc(stream_id))
name << "RTX ";
if (stream_id.GetDirection() == kIncomingPacket) {
name << "(In) ";
} else {
name << "(Out) ";
}
name << SsrcToString(stream_id.GetSsrc());
return name.str();
}
void EventLogAnalyzer::CreatePacketGraph(PacketDirection desired_direction,
Plot* plot) {
for (auto& kv : rtp_packets_) {
StreamId stream_id = kv.first;
const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
// Filter on direction and SSRC.
if (stream_id.GetDirection() != desired_direction ||
!MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
continue;
}
TimeSeries time_series(GetStreamName(stream_id), BAR_GRAPH);
ProcessPoints<LoggedRtpPacket>(
[](const LoggedRtpPacket& packet) -> rtc::Optional<float> {
return rtc::Optional<float>(packet.total_length);
},
packet_stream, begin_time_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
}
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Packet size (bytes)", kBottomMargin,
kTopMargin);
if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
plot->SetTitle("Incoming RTP packets");
} else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
plot->SetTitle("Outgoing RTP packets");
}
}
template <typename T>
void EventLogAnalyzer::CreateAccumulatedPacketsTimeSeries(
PacketDirection desired_direction,
Plot* plot,
const std::map<StreamId, std::vector<T>>& packets,
const std::string& label_prefix) {
for (auto& kv : packets) {
StreamId stream_id = kv.first;
const std::vector<T>& packet_stream = kv.second;
// Filter on direction and SSRC.
if (stream_id.GetDirection() != desired_direction ||
!MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
continue;
}
std::string label = label_prefix + " " + GetStreamName(stream_id);
TimeSeries time_series(label, LINE_STEP_GRAPH);
for (size_t i = 0; i < packet_stream.size(); i++) {
float x = static_cast<float>(packet_stream[i].timestamp - begin_time_) /
1000000;
time_series.points.emplace_back(x, i + 1);
}
plot->AppendTimeSeries(std::move(time_series));
}
}
void EventLogAnalyzer::CreateAccumulatedPacketsGraph(
PacketDirection desired_direction,
Plot* plot) {
CreateAccumulatedPacketsTimeSeries(desired_direction, plot, rtp_packets_,
"RTP");
CreateAccumulatedPacketsTimeSeries(desired_direction, plot, rtcp_packets_,
"RTCP");
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Received Packets", kBottomMargin, kTopMargin);
if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
plot->SetTitle("Accumulated Incoming RTP/RTCP packets");
} else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
plot->SetTitle("Accumulated Outgoing RTP/RTCP packets");
}
}
// For each SSRC, plot the time between the consecutive playouts.
void EventLogAnalyzer::CreatePlayoutGraph(Plot* plot) {
std::map<uint32_t, TimeSeries> time_series;
std::map<uint32_t, uint64_t> last_playout;
uint32_t ssrc;
for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
if (event_type == ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT) {
parsed_log_.GetAudioPlayout(i, &ssrc);
uint64_t timestamp = parsed_log_.GetTimestamp(i);
if (MatchingSsrc(ssrc, desired_ssrc_)) {
float x = static_cast<float>(timestamp - begin_time_) / 1000000;
float y = static_cast<float>(timestamp - last_playout[ssrc]) / 1000;
if (time_series[ssrc].points.size() == 0) {
// There were no previusly logged playout for this SSRC.
// Generate a point, but place it on the x-axis.
y = 0;
}
time_series[ssrc].points.push_back(TimeSeriesPoint(x, y));
last_playout[ssrc] = timestamp;
}
}
}
// Set labels and put in graph.
for (auto& kv : time_series) {
kv.second.label = SsrcToString(kv.first);
kv.second.style = BAR_GRAPH;
plot->AppendTimeSeries(std::move(kv.second));
}
plot->SetXAxis(0, 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(Plot* plot) {
std::map<StreamId, TimeSeries> time_series;
for (auto& kv : rtp_packets_) {
StreamId stream_id = kv.first;
const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
// TODO(ivoc): When audio send/receive configs are stored in the event
// log, a check should be added here to only process audio
// streams. Tracking bug: webrtc:6399
for (auto& packet : packet_stream) {
if (packet.header.extension.hasAudioLevel) {
float x = static_cast<float>(packet.timestamp - begin_time_) / 1000000;
// 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[stream_id].points.emplace_back(TimeSeriesPoint(x, y));
}
}
}
for (auto& series : time_series) {
series.second.label = GetStreamName(series.first);
series.second.style = LINE_GRAPH;
plot->AppendTimeSeries(std::move(series.second));
}
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetYAxis(-127, 0, "Audio level (dBov)", kBottomMargin,
kTopMargin);
plot->SetTitle("Audio level");
}
// For each SSRC, plot the time between the consecutive playouts.
void EventLogAnalyzer::CreateSequenceNumberGraph(Plot* plot) {
for (auto& kv : rtp_packets_) {
StreamId stream_id = kv.first;
const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
// Filter on direction and SSRC.
if (stream_id.GetDirection() != kIncomingPacket ||
!MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
continue;
}
TimeSeries time_series(GetStreamName(stream_id), BAR_GRAPH);
ProcessPairs<LoggedRtpPacket, float>(
[](const LoggedRtpPacket& old_packet,
const LoggedRtpPacket& new_packet) {
int64_t diff =
WrappingDifference(new_packet.header.sequenceNumber,
old_packet.header.sequenceNumber, 1ul << 16);
return rtc::Optional<float>(diff);
},
packet_stream, begin_time_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
}
plot->SetXAxis(0, 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 (auto& kv : rtp_packets_) {
StreamId stream_id = kv.first;
const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
// Filter on direction and SSRC.
if (stream_id.GetDirection() != kIncomingPacket ||
!MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_) ||
packet_stream.size() == 0) {
continue;
}
TimeSeries time_series(GetStreamName(stream_id), LINE_DOT_GRAPH);
const uint64_t kWindowUs = 1000000;
const uint64_t kStep = 1000000;
SequenceNumberUnwrapper unwrapper_;
SequenceNumberUnwrapper prior_unwrapper_;
size_t window_index_begin = 0;
size_t window_index_end = 0;
int64_t highest_seq_number =
unwrapper_.Unwrap(packet_stream[0].header.sequenceNumber) - 1;
int64_t highest_prior_seq_number =
prior_unwrapper_.Unwrap(packet_stream[0].header.sequenceNumber) - 1;
for (uint64_t t = begin_time_; t < end_time_ + kStep; t += kStep) {
while (window_index_end < packet_stream.size() &&
packet_stream[window_index_end].timestamp < t) {
int64_t sequence_number = unwrapper_.Unwrap(
packet_stream[window_index_end].header.sequenceNumber);
highest_seq_number = std::max(highest_seq_number, sequence_number);
++window_index_end;
}
while (window_index_begin < packet_stream.size() &&
packet_stream[window_index_begin].timestamp < t - kWindowUs) {
int64_t sequence_number = prior_unwrapper_.Unwrap(
packet_stream[window_index_begin].header.sequenceNumber);
highest_prior_seq_number =
std::max(highest_prior_seq_number, sequence_number);
++window_index_begin;
}
float x = static_cast<float>(t - begin_time_) / 1000000;
int64_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(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Estimated loss rate (%)", kBottomMargin,
kTopMargin);
plot->SetTitle("Estimated incoming loss rate");
}
void EventLogAnalyzer::CreateDelayChangeGraph(Plot* plot) {
for (auto& kv : rtp_packets_) {
StreamId stream_id = kv.first;
const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
// Filter on direction and SSRC.
if (stream_id.GetDirection() != kIncomingPacket ||
!MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_) ||
IsAudioSsrc(stream_id) || !IsVideoSsrc(stream_id) ||
IsRtxSsrc(stream_id)) {
continue;
}
TimeSeries capture_time_data(GetStreamName(stream_id) + " capture-time",
BAR_GRAPH);
ProcessPairs<LoggedRtpPacket, double>(NetworkDelayDiff_CaptureTime,
packet_stream, begin_time_,
&capture_time_data);
plot->AppendTimeSeries(std::move(capture_time_data));
TimeSeries send_time_data(GetStreamName(stream_id) + " abs-send-time",
BAR_GRAPH);
ProcessPairs<LoggedRtpPacket, double>(NetworkDelayDiff_AbsSendTime,
packet_stream, begin_time_,
&send_time_data);
plot->AppendTimeSeries(std::move(send_time_data));
}
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Latency change (ms)", kBottomMargin,
kTopMargin);
plot->SetTitle("Network latency change between consecutive packets");
}
void EventLogAnalyzer::CreateAccumulatedDelayChangeGraph(Plot* plot) {
for (auto& kv : rtp_packets_) {
StreamId stream_id = kv.first;
const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
// Filter on direction and SSRC.
if (stream_id.GetDirection() != kIncomingPacket ||
!MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_) ||
IsAudioSsrc(stream_id) || !IsVideoSsrc(stream_id) ||
IsRtxSsrc(stream_id)) {
continue;
}
TimeSeries capture_time_data(GetStreamName(stream_id) + " capture-time",
LINE_GRAPH);
AccumulatePairs<LoggedRtpPacket, double>(NetworkDelayDiff_CaptureTime,
packet_stream, begin_time_,
&capture_time_data);
plot->AppendTimeSeries(std::move(capture_time_data));
TimeSeries send_time_data(GetStreamName(stream_id) + " abs-send-time",
LINE_GRAPH);
AccumulatePairs<LoggedRtpPacket, double>(NetworkDelayDiff_AbsSendTime,
packet_stream, begin_time_,
&send_time_data);
plot->AppendTimeSeries(std::move(send_time_data));
}
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Latency change (ms)", kBottomMargin,
kTopMargin);
plot->SetTitle("Accumulated network latency change");
}
// Plot the fraction of packets lost (as perceived by the loss-based BWE).
void EventLogAnalyzer::CreateFractionLossGraph(Plot* plot) {
TimeSeries time_series("Fraction lost", LINE_DOT_GRAPH);
for (auto& bwe_update : bwe_loss_updates_) {
float x = static_cast<float>(bwe_update.timestamp - begin_time_) / 1000000;
float y = static_cast<float>(bwe_update.fraction_loss) / 255 * 100;
time_series.points.emplace_back(x, y);
}
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 10, "Percent lost packets", kBottomMargin,
kTopMargin);
plot->SetTitle("Reported packet loss");
plot->AppendTimeSeries(std::move(time_series));
}
// Plot the total bandwidth used by all RTP streams.
void EventLogAnalyzer::CreateTotalBitrateGraph(
PacketDirection desired_direction,
Plot* plot) {
struct TimestampSize {
TimestampSize(uint64_t t, size_t s) : timestamp(t), size(s) {}
uint64_t timestamp;
size_t size;
};
std::vector<TimestampSize> packets;
PacketDirection direction;
size_t total_length;
// Extract timestamps and sizes for the relevant packets.
for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
if (event_type == ParsedRtcEventLog::RTP_EVENT) {
parsed_log_.GetRtpHeader(i, &direction, nullptr, nullptr, &total_length);
if (direction == desired_direction) {
uint64_t timestamp = parsed_log_.GetTimestamp(i);
packets.push_back(TimestampSize(timestamp, total_length));
}
}
}
size_t window_index_begin = 0;
size_t window_index_end = 0;
size_t bytes_in_window = 0;
// Calculate a moving average of the bitrate and store in a TimeSeries.
TimeSeries bitrate_series("Bitrate", LINE_GRAPH);
for (uint64_t time = begin_time_; time < end_time_ + step_; time += step_) {
while (window_index_end < packets.size() &&
packets[window_index_end].timestamp < time) {
bytes_in_window += packets[window_index_end].size;
++window_index_end;
}
while (window_index_begin < packets.size() &&
packets[window_index_begin].timestamp < time - window_duration_) {
RTC_DCHECK_LE(packets[window_index_begin].size, bytes_in_window);
bytes_in_window -= packets[window_index_begin].size;
++window_index_begin;
}
float window_duration_in_seconds =
static_cast<float>(window_duration_) / 1000000;
float x = static_cast<float>(time - begin_time_) / 1000000;
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.
if (desired_direction == kOutgoingPacket) {
TimeSeries loss_series("Loss-based estimate", LINE_STEP_GRAPH);
for (auto& loss_update : bwe_loss_updates_) {
float x =
static_cast<float>(loss_update.timestamp - begin_time_) / 1000000;
float y = static_cast<float>(loss_update.new_bitrate) / 1000;
loss_series.points.emplace_back(x, y);
}
TimeSeries delay_series("Delay-based estimate", LINE_STEP_GRAPH);
for (auto& delay_update : bwe_delay_updates_) {
float x =
static_cast<float>(delay_update.timestamp - begin_time_) / 1000000;
float y = static_cast<float>(delay_update.bitrate_bps) / 1000;
delay_series.points.emplace_back(x, y);
}
TimeSeries created_series("Probe cluster created.", DOT_GRAPH);
for (auto& cluster : bwe_probe_cluster_created_events_) {
float x = static_cast<float>(cluster.timestamp - begin_time_) / 1000000;
float y = static_cast<float>(cluster.bitrate_bps) / 1000;
created_series.points.emplace_back(x, y);
}
TimeSeries result_series("Probing results.", DOT_GRAPH);
for (auto& result : bwe_probe_result_events_) {
if (result.bitrate_bps) {
float x = static_cast<float>(result.timestamp - begin_time_) / 1000000;
float y = static_cast<float>(*result.bitrate_bps) / 1000;
result_series.points.emplace_back(x, y);
}
}
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
// and outgoing REMB over incoming bitrate.
PacketDirection remb_direction =
desired_direction == kOutgoingPacket ? kIncomingPacket : kOutgoingPacket;
TimeSeries remb_series("Remb", LINE_STEP_GRAPH);
std::multimap<uint64_t, const LoggedRtcpPacket*> remb_packets;
for (const auto& kv : rtcp_packets_) {
if (kv.first.GetDirection() == remb_direction) {
for (const LoggedRtcpPacket& rtcp_packet : kv.second) {
if (rtcp_packet.type == kRtcpRemb) {
remb_packets.insert(
std::make_pair(rtcp_packet.timestamp, &rtcp_packet));
}
}
}
}
for (const auto& kv : remb_packets) {
const LoggedRtcpPacket* const rtcp = kv.second;
const rtcp::Remb* const remb = static_cast<rtcp::Remb*>(rtcp->packet.get());
float x = static_cast<float>(rtcp->timestamp - begin_time_) / 1000000;
float y = static_cast<float>(remb->bitrate_bps()) / 1000;
remb_series.points.emplace_back(x, y);
}
plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
plot->SetTitle("Incoming RTP bitrate");
} else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
plot->SetTitle("Outgoing RTP bitrate");
}
}
// For each SSRC, plot the bandwidth used by that stream.
void EventLogAnalyzer::CreateStreamBitrateGraph(
PacketDirection desired_direction,
Plot* plot) {
for (auto& kv : rtp_packets_) {
StreamId stream_id = kv.first;
const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
// Filter on direction and SSRC.
if (stream_id.GetDirection() != desired_direction ||
!MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
continue;
}
TimeSeries time_series(GetStreamName(stream_id), LINE_GRAPH);
MovingAverage<LoggedRtpPacket, double>(
[](const LoggedRtpPacket& packet) {
return rtc::Optional<double>(packet.total_length * 8.0 / 1000.0);
},
packet_stream, begin_time_, end_time_, window_duration_, step_,
&time_series);
plot->AppendTimeSeries(std::move(time_series));
}
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
plot->SetTitle("Incoming bitrate per stream");
} else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
plot->SetTitle("Outgoing bitrate per stream");
}
}
void EventLogAnalyzer::CreateBweSimulationGraph(Plot* plot) {
std::multimap<uint64_t, const LoggedRtpPacket*> outgoing_rtp;
std::multimap<uint64_t, const LoggedRtcpPacket*> incoming_rtcp;
for (const auto& kv : rtp_packets_) {
if (kv.first.GetDirection() == PacketDirection::kOutgoingPacket) {
for (const LoggedRtpPacket& rtp_packet : kv.second)
outgoing_rtp.insert(std::make_pair(rtp_packet.timestamp, &rtp_packet));
}
}
for (const auto& kv : rtcp_packets_) {
if (kv.first.GetDirection() == PacketDirection::kIncomingPacket) {
for (const LoggedRtcpPacket& rtcp_packet : kv.second)
incoming_rtcp.insert(
std::make_pair(rtcp_packet.timestamp, &rtcp_packet));
}
}
SimulatedClock clock(0);
BitrateObserver observer;
RtcEventLogNullImpl null_event_log;
PacketRouter packet_router;
CongestionController cc(&clock, &observer, &observer, &null_event_log,
&packet_router);
// 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", LINE_DOT_GRAPH);
TimeSeries acked_time_series("Acked bitrate", LINE_DOT_GRAPH);
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->first);
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);
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());
const LoggedRtcpPacket& rtcp = *rtcp_iterator->second;
if (rtcp.type == kRtcpTransportFeedback) {
cc.OnTransportFeedback(
*static_cast<rtcp::TransportFeedback*>(rtcp.packet.get()));
std::vector<PacketFeedback> feedback = cc.GetTransportFeedbackVector();
SortPacketFeedbackVector(&feedback);
rtc::Optional<uint32_t> bitrate_bps;
if (!feedback.empty()) {
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);
}
uint32_t y = 0;
if (bitrate_bps)
y = *bitrate_bps / 1000;
float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
1000000;
acked_time_series.points.emplace_back(x, y);
}
++rtcp_iterator;
}
if (clock.TimeInMicroseconds() >= NextRtpTime()) {
RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
const LoggedRtpPacket& rtp = *rtp_iterator->second;
if (rtp.header.extension.hasTransportSequenceNumber) {
RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
cc.AddPacket(rtp.header.ssrc,
rtp.header.extension.transportSequenceNumber,
rtp.total_length, PacedPacketInfo());
rtc::SentPacket sent_packet(
rtp.header.extension.transportSequenceNumber, rtp.timestamp / 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 = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
1000000;
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->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
plot->SetTitle("Simulated BWE behavior");
}
void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) {
std::multimap<uint64_t, const LoggedRtpPacket*> outgoing_rtp;
std::multimap<uint64_t, const LoggedRtcpPacket*> incoming_rtcp;
for (const auto& kv : rtp_packets_) {
if (kv.first.GetDirection() == PacketDirection::kOutgoingPacket) {
for (const LoggedRtpPacket& rtp_packet : kv.second)
outgoing_rtp.insert(std::make_pair(rtp_packet.timestamp, &rtp_packet));
}
}
for (const auto& kv : rtcp_packets_) {
if (kv.first.GetDirection() == PacketDirection::kIncomingPacket) {
for (const LoggedRtcpPacket& rtcp_packet : kv.second)
incoming_rtcp.insert(
std::make_pair(rtcp_packet.timestamp, &rtcp_packet));
}
}
SimulatedClock clock(0);
TransportFeedbackAdapter feedback_adapter(&clock);
TimeSeries time_series("Network Delay Change", LINE_DOT_GRAPH);
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->first);
return std::numeric_limits<int64_t>::max();
};
int64_t time_us = std::min(NextRtpTime(), NextRtcpTime());
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());
const LoggedRtcpPacket& rtcp = *rtcp_iterator->second;
if (rtcp.type == kRtcpTransportFeedback) {
feedback_adapter.OnTransportFeedback(
*static_cast<rtcp::TransportFeedback*>(rtcp.packet.get()));
std::vector<PacketFeedback> feedback =
feedback_adapter.GetTransportFeedbackVector();
SortPacketFeedbackVector(&feedback);
for (const PacketFeedback& packet : feedback) {
int64_t y = packet.arrival_time_ms - packet.send_time_ms;
float x =
static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
1000000;
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 LoggedRtpPacket& rtp = *rtp_iterator->second;
if (rtp.header.extension.hasTransportSequenceNumber) {
RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
feedback_adapter.AddPacket(rtp.header.ssrc,
rtp.header.extension.transportSequenceNumber,
rtp.total_length, PacedPacketInfo());
feedback_adapter.OnSentPacket(
rtp.header.extension.transportSequenceNumber, rtp.timestamp / 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;
// Add the data set to the plot.
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
plot->SetTitle("Network Delay Change.");
}
std::vector<std::pair<int64_t, int64_t>> EventLogAnalyzer::GetFrameTimestamps()
const {
std::vector<std::pair<int64_t, int64_t>> timestamps;
size_t largest_stream_size = 0;
const std::vector<LoggedRtpPacket>* largest_video_stream = nullptr;
// Find the incoming video stream with the most number of packets that is
// not rtx.
for (const auto& kv : rtp_packets_) {
if (kv.first.GetDirection() == kIncomingPacket &&
video_ssrcs_.find(kv.first) != video_ssrcs_.end() &&
rtx_ssrcs_.find(kv.first) == rtx_ssrcs_.end() &&
kv.second.size() > largest_stream_size) {
largest_stream_size = kv.second.size();
largest_video_stream = &kv.second;
}
}
if (largest_video_stream == nullptr) {
for (auto& packet : *largest_video_stream) {
if (packet.header.markerBit) {
int64_t capture_ms = packet.header.timestamp / 90.0;
int64_t arrival_ms = packet.timestamp / 1000.0;
timestamps.push_back(std::make_pair(capture_ms, arrival_ms));
}
}
}
return timestamps;
}
void EventLogAnalyzer::CreateTimestampGraph(Plot* plot) {
for (const auto& kv : rtp_packets_) {
const std::vector<LoggedRtpPacket>& rtp_packets = kv.second;
StreamId stream_id = kv.first;
{
TimeSeries timestamp_data(GetStreamName(stream_id) + " capture-time",
LINE_DOT_GRAPH);
for (LoggedRtpPacket packet : rtp_packets) {
float x = static_cast<float>(packet.timestamp - begin_time_) / 1000000;
float y = packet.header.timestamp;
timestamp_data.points.emplace_back(x, y);
}
plot->AppendTimeSeries(std::move(timestamp_data));
}
{
auto kv = rtcp_packets_.find(stream_id);
if (kv != rtcp_packets_.end()) {
const auto& packets = kv->second;
TimeSeries timestamp_data(
GetStreamName(stream_id) + " rtcp capture-time", LINE_DOT_GRAPH);
for (const LoggedRtcpPacket& rtcp : packets) {
if (rtcp.type != kRtcpSr)
continue;
rtcp::SenderReport* sr;
sr = static_cast<rtcp::SenderReport*>(rtcp.packet.get());
float x = static_cast<float>(rtcp.timestamp - begin_time_) / 1000000;
float y = sr->rtp_timestamp();
timestamp_data.points.emplace_back(x, y);
}
plot->AppendTimeSeries(std::move(timestamp_data));
}
}
}
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetSuggestedYAxis(0, 1, "Timestamp (90khz)", kBottomMargin, kTopMargin);
plot->SetTitle("Timestamps");
}
void EventLogAnalyzer::CreateAudioEncoderTargetBitrateGraph(Plot* plot) {
TimeSeries time_series("Audio encoder target bitrate", LINE_DOT_GRAPH);
ProcessPoints<AudioNetworkAdaptationEvent>(
[](const AudioNetworkAdaptationEvent& 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::Optional<float>();
},
audio_network_adaptation_events_, begin_time_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(0, 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", LINE_DOT_GRAPH);
ProcessPoints<AudioNetworkAdaptationEvent>(
[](const AudioNetworkAdaptationEvent& 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>();
},
audio_network_adaptation_events_, begin_time_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(0, 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::CreateAudioEncoderUplinkPacketLossFractionGraph(
Plot* plot) {
TimeSeries time_series("Audio encoder uplink packet loss fraction",
LINE_DOT_GRAPH);
ProcessPoints<AudioNetworkAdaptationEvent>(
[](const AudioNetworkAdaptationEvent& 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>();
},
audio_network_adaptation_events_, begin_time_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(0, 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", LINE_DOT_GRAPH);
ProcessPoints<AudioNetworkAdaptationEvent>(
[](const AudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.enable_fec)
return rtc::Optional<float>(
static_cast<float>(*ana_event.config.enable_fec));
return rtc::Optional<float>();
},
audio_network_adaptation_events_, begin_time_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(0, 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", LINE_DOT_GRAPH);
ProcessPoints<AudioNetworkAdaptationEvent>(
[](const AudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.enable_dtx)
return rtc::Optional<float>(
static_cast<float>(*ana_event.config.enable_dtx));
return rtc::Optional<float>();
},
audio_network_adaptation_events_, begin_time_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(0, 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", LINE_DOT_GRAPH);
ProcessPoints<AudioNetworkAdaptationEvent>(
[](const AudioNetworkAdaptationEvent& ana_event) {
if (ana_event.config.num_channels)
return rtc::Optional<float>(
static_cast<float>(*ana_event.config.num_channels));
return rtc::Optional<float>();
},
audio_network_adaptation_events_, begin_time_, &time_series);
plot->AppendTimeSeries(std::move(time_series));
plot->SetXAxis(0, 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<LoggedRtpPacket>* packet_stream,
const std::vector<uint64_t>* output_events_us,
rtc::Optional<uint64_t> end_time_us)
: packet_stream_(*packet_stream),
packet_stream_it_(packet_stream_.begin()),
output_events_us_it_(output_events_us->begin()),
output_events_us_end_(output_events_us->end()),
end_time_us_(end_time_us) {
RTC_DCHECK(packet_stream);
RTC_DCHECK(output_events_us);
}
rtc::Optional<int64_t> NextPacketTime() const override {
if (packet_stream_it_ == packet_stream_.end()) {
return rtc::Optional<int64_t>();
}
if (end_time_us_ && packet_stream_it_->timestamp > *end_time_us_) {
return rtc::Optional<int64_t>();
}
// Convert from us to ms.
return rtc::Optional<int64_t>(packet_stream_it_->timestamp / 1000);
}
rtc::Optional<int64_t> NextOutputEventTime() const override {
if (output_events_us_it_ == output_events_us_end_) {
return rtc::Optional<int64_t>();
}
if (end_time_us_ && *output_events_us_it_ > *end_time_us_) {
return rtc::Optional<int64_t>();
}
// Convert from us to ms.
return rtc::Optional<int64_t>(
rtc::checked_cast<int64_t>(*output_events_us_it_ / 1000));
}
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_->header;
// Convert from us to ms.
packet_data->time_ms = packet_stream_it_->timestamp / 1000.0;
// 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_->total_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_us_it_ != output_events_us_end_) {
++output_events_us_it_;
}
}
bool ended() const override { return !NextEventTime(); }
rtc::Optional<RTPHeader> NextHeader() const override {
if (packet_stream_it_ == packet_stream_.end()) {
return rtc::Optional<RTPHeader>();
}
return rtc::Optional<RTPHeader>(packet_stream_it_->header);
}
private:
const std::vector<LoggedRtpPacket>& packet_stream_;
std::vector<LoggedRtpPacket>::const_iterator packet_stream_it_;
std::vector<uint64_t>::const_iterator output_events_us_it_;
const std::vector<uint64_t>::const_iterator output_events_us_end_;
const rtc::Optional<uint64_t> end_time_us_;
};
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::NetEqDelayAnalyzer> CreateNetEqTestAndRun(
const std::vector<LoggedRtpPacket>* packet_stream,
const std::vector<uint64_t>* output_events_us,
rtc::Optional<uint64_t> end_time_us,
const std::string& replacement_file_name,
int file_sample_rate_hz) {
std::unique_ptr<test::NetEqInput> input(
new NetEqStreamInput(packet_stream, output_events_us, end_time_us));
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);
test::DefaultNetEqTestErrorCallback error_cb;
test::NetEqTest::Callbacks callbacks;
callbacks.error_callback = &error_cb;
callbacks.post_insert_packet = delay_cb.get();
callbacks.get_audio_callback = delay_cb.get();
test::NetEqTest test(config, codecs, ext_codecs, std::move(input),
std::move(output), callbacks);
test.Run();
return delay_cb;
}
} // namespace
// 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 std::string& replacement_file_name,
int file_sample_rate_hz,
Plot* plot) {
const auto& incoming_audio_kv = std::find_if(
rtp_packets_.begin(), rtp_packets_.end(),
[this](std::pair<StreamId, std::vector<LoggedRtpPacket>> kv) {
return kv.first.GetDirection() == kIncomingPacket &&
this->IsAudioSsrc(kv.first);
});
if (incoming_audio_kv == rtp_packets_.end()) {
// No incoming audio stream found.
return;
}
const uint32_t ssrc = incoming_audio_kv->first.GetSsrc();
std::map<uint32_t, std::vector<uint64_t>>::const_iterator output_events_it =
audio_playout_events_.find(ssrc);
if (output_events_it == 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 = audio_playout_events_.cbegin();
}
rtc::Optional<uint64_t> end_time_us =
log_segments_.empty()
? rtc::Optional<uint64_t>()
: rtc::Optional<uint64_t>(log_segments_.front().second);
auto delay_cb = CreateNetEqTestAndRun(
&incoming_audio_kv->second, &output_events_it->second, end_time_us,
replacement_file_name, file_sample_rate_hz);
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;
delay_cb->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<StreamId, TimeSeries> time_series_packet_arrival;
std::map<StreamId, TimeSeries> time_series_relative_packet_arrival;
std::map<StreamId, TimeSeries> time_series_play_time;
std::map<StreamId, TimeSeries> time_series_target_time;
float min_y_axis = 0.f;
float max_y_axis = 0.f;
const StreamId stream_id = incoming_audio_kv->first;
for (size_t i = 0; i < send_times_s.size(); ++i) {
time_series_packet_arrival[stream_id].points.emplace_back(
TimeSeriesPoint(send_times_s[i], arrival_delay_ms[i]));
time_series_relative_packet_arrival[stream_id].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[stream_id].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[stream_id].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.style = LINE_GRAPH;
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.style = LINE_GRAPH;
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.style = LINE_DOT_GRAPH;
plot->AppendTimeSeries(std::move(series.second));
}
plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
plot->SetYAxis(min_y_axis, max_y_axis, "Relative delay (ms)", kBottomMargin,
kTopMargin);
plot->SetTitle("NetEq timing");
}
} // namespace plotting
} // namespace webrtc