| /* |
| * 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/tools/event_log_visualizer/analyzer.h" |
| |
| #include <algorithm> |
| #include <limits> |
| #include <map> |
| #include <sstream> |
| #include <string> |
| #include <utility> |
| |
| #include "webrtc/audio_receive_stream.h" |
| #include "webrtc/audio_send_stream.h" |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/logging.h" |
| #include "webrtc/call.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/congestion_controller/include/congestion_controller.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/rtp_utility.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" |
| #include "webrtc/video_receive_stream.h" |
| #include "webrtc/video_send_stream.h" |
| |
| namespace webrtc { |
| namespace plotting { |
| |
| namespace { |
| |
| 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>(1 << 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; |
| } |
| return difference; |
| } |
| |
| void RegisterHeaderExtensions( |
| const std::vector<webrtc::RtpExtension>& extensions, |
| webrtc::RtpHeaderExtensionMap* extension_map) { |
| extension_map->Erase(); |
| for (const webrtc::RtpExtension& extension : extensions) { |
| extension_map->Register(webrtc::StringToRtpExtensionType(extension.uri), |
| extension.id); |
| } |
| } |
| |
| constexpr float kLeftMargin = 0.01f; |
| constexpr float kRightMargin = 0.02f; |
| constexpr float kBottomMargin = 0.02f; |
| constexpr float kTopMargin = 0.05f; |
| |
| } // namespace |
| |
| bool EventLogAnalyzer::StreamId::operator<(const StreamId& other) const { |
| if (ssrc_ < other.ssrc_) { |
| return true; |
| } |
| if (ssrc_ == other.ssrc_) { |
| if (direction_ < other.direction_) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| bool EventLogAnalyzer::StreamId::operator==(const StreamId& other) const { |
| return ssrc_ == other.ssrc_ && direction_ == other.direction_; |
| } |
| |
| |
| 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(); |
| |
| // Maps a stream identifier consisting of ssrc and direction |
| // to the header extensions used by that stream, |
| std::map<StreamId, RtpHeaderExtensionMap> extension_maps; |
| |
| PacketDirection direction; |
| uint8_t header[IP_PACKET_SIZE]; |
| size_t header_length; |
| size_t total_length; |
| |
| 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: { |
| VideoReceiveStream::Config config(nullptr); |
| parsed_log_.GetVideoReceiveConfig(i, &config); |
| StreamId stream(config.rtp.remote_ssrc, kIncomingPacket); |
| RegisterHeaderExtensions(config.rtp.extensions, |
| &extension_maps[stream]); |
| for (auto kv : config.rtp.rtx) { |
| StreamId rtx_stream(kv.second.ssrc, kIncomingPacket); |
| RegisterHeaderExtensions(config.rtp.extensions, |
| &extension_maps[rtx_stream]); |
| } |
| break; |
| } |
| case ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT: { |
| VideoSendStream::Config config(nullptr); |
| parsed_log_.GetVideoSendConfig(i, &config); |
| for (auto ssrc : config.rtp.ssrcs) { |
| StreamId stream(ssrc, kOutgoingPacket); |
| RegisterHeaderExtensions(config.rtp.extensions, |
| &extension_maps[stream]); |
| } |
| for (auto ssrc : config.rtp.rtx.ssrcs) { |
| StreamId stream(ssrc, kOutgoingPacket); |
| RegisterHeaderExtensions(config.rtp.extensions, |
| &extension_maps[stream]); |
| } |
| break; |
| } |
| case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: { |
| AudioReceiveStream::Config config; |
| // TODO(terelius): Parse the audio configs once we have them. |
| break; |
| } |
| case ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: { |
| AudioSendStream::Config config(nullptr); |
| // TODO(terelius): Parse the audio configs once we have them. |
| break; |
| } |
| case ParsedRtcEventLog::RTP_EVENT: { |
| MediaType media_type; |
| parsed_log_.GetRtpHeader(i, &direction, &media_type, header, |
| &header_length, &total_length); |
| // Parse header to get SSRC. |
| RtpUtility::RtpHeaderParser rtp_parser(header, header_length); |
| RTPHeader parsed_header; |
| rtp_parser.Parse(&parsed_header); |
| StreamId stream(parsed_header.ssrc, direction); |
| // Look up the extension_map and parse it again to get the extensions. |
| if (extension_maps.count(stream) == 1) { |
| RtpHeaderExtensionMap* extension_map = &extension_maps[stream]; |
| rtp_parser.Parse(&parsed_header, extension_map); |
| } |
| uint64_t timestamp = parsed_log_.GetTimestamp(i); |
| rtp_packets_[stream].push_back( |
| LoggedRtpPacket(timestamp, parsed_header, total_length)); |
| break; |
| } |
| case ParsedRtcEventLog::RTCP_EVENT: { |
| uint8_t packet[IP_PACKET_SIZE]; |
| MediaType media_type; |
| parsed_log_.GetRtcpPacket(i, &direction, &media_type, packet, |
| &total_length); |
| |
| RtpUtility::RtpHeaderParser rtp_parser(packet, total_length); |
| RTPHeader parsed_header; |
| RTC_CHECK(rtp_parser.ParseRtcp(&parsed_header)); |
| uint32_t ssrc = parsed_header.ssrc; |
| |
| RTCPUtility::RTCPParserV2 rtcp_parser(packet, total_length, true); |
| RTC_CHECK(rtcp_parser.IsValid()); |
| |
| RTCPUtility::RTCPPacketTypes packet_type = rtcp_parser.Begin(); |
| while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) { |
| switch (packet_type) { |
| case RTCPUtility::RTCPPacketTypes::kTransportFeedback: { |
| // Currently feedback is logged twice, both for audio and video. |
| // Only act on one of them. |
| if (media_type == MediaType::VIDEO) { |
| std::unique_ptr<rtcp::RtcpPacket> rtcp_packet( |
| rtcp_parser.ReleaseRtcpPacket()); |
| StreamId stream(ssrc, direction); |
| uint64_t timestamp = parsed_log_.GetTimestamp(i); |
| rtcp_packets_[stream].push_back(LoggedRtcpPacket( |
| timestamp, kRtcpTransportFeedback, std::move(rtcp_packet))); |
| } |
| break; |
| } |
| default: |
| break; |
| } |
| rtcp_parser.Iterate(); |
| packet_type = rtcp_parser.PacketType(); |
| } |
| break; |
| } |
| case ParsedRtcEventLog::LOG_START: { |
| break; |
| } |
| case ParsedRtcEventLog::LOG_END: { |
| break; |
| } |
| case ParsedRtcEventLog::BWE_PACKET_LOSS_EVENT: { |
| BwePacketLossEvent bwe_update; |
| bwe_update.timestamp = parsed_log_.GetTimestamp(i); |
| parsed_log_.GetBwePacketLossEvent(i, &bwe_update.new_bitrate, |
| &bwe_update.fraction_loss, |
| &bwe_update.expected_packets); |
| bwe_loss_updates_.push_back(bwe_update); |
| break; |
| } |
| case ParsedRtcEventLog::BWE_PACKET_DELAY_EVENT: { |
| break; |
| } |
| case ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT: { |
| 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; |
| } |
| |
| class BitrateObserver : public CongestionController::Observer, |
| public RemoteBitrateObserver { |
| public: |
| BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {} |
| |
| void OnNetworkChanged(uint32_t bitrate_bps, |
| uint8_t fraction_loss, |
| int64_t rtt_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_; |
| }; |
| |
| void EventLogAnalyzer::CreatePacketGraph(PacketDirection desired_direction, |
| Plot* plot) { |
| std::map<uint32_t, TimeSeries> time_series; |
| |
| PacketDirection direction; |
| MediaType media_type; |
| uint8_t header[IP_PACKET_SIZE]; |
| size_t header_length, total_length; |
| |
| 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, &media_type, header, |
| &header_length, &total_length); |
| if (direction == desired_direction) { |
| // Parse header to get SSRC. |
| RtpUtility::RtpHeaderParser rtp_parser(header, header_length); |
| RTPHeader parsed_header; |
| rtp_parser.Parse(&parsed_header); |
| // Filter on SSRC. |
| if (MatchingSsrc(parsed_header.ssrc, desired_ssrc_)) { |
| uint64_t timestamp = parsed_log_.GetTimestamp(i); |
| float x = static_cast<float>(timestamp - begin_time_) / 1000000; |
| float y = total_length; |
| time_series[parsed_header.ssrc].points.push_back( |
| TimeSeriesPoint(x, y)); |
| } |
| } |
| } |
| } |
| |
| // Set labels and put in graph. |
| for (auto& kv : time_series) { |
| kv.second.label = SsrcToString(kv.first); |
| kv.second.style = BAR_GRAPH; |
| plot->series_list_.push_back(std::move(kv.second)); |
| } |
| |
| 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"); |
| } |
| } |
| |
| // 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->series_list_.push_back(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 each SSRC, plot the time between the consecutive playouts. |
| void EventLogAnalyzer::CreateSequenceNumberGraph(Plot* plot) { |
| std::map<uint32_t, TimeSeries> time_series; |
| std::map<uint32_t, uint16_t> last_seqno; |
| |
| PacketDirection direction; |
| MediaType media_type; |
| uint8_t header[IP_PACKET_SIZE]; |
| size_t header_length, total_length; |
| |
| 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, &media_type, header, |
| &header_length, &total_length); |
| uint64_t timestamp = parsed_log_.GetTimestamp(i); |
| if (direction == PacketDirection::kIncomingPacket) { |
| // Parse header to get SSRC. |
| RtpUtility::RtpHeaderParser rtp_parser(header, header_length); |
| RTPHeader parsed_header; |
| rtp_parser.Parse(&parsed_header); |
| // Filter on SSRC. |
| if (MatchingSsrc(parsed_header.ssrc, desired_ssrc_)) { |
| float x = static_cast<float>(timestamp - begin_time_) / 1000000; |
| int y = WrappingDifference(parsed_header.sequenceNumber, |
| last_seqno[parsed_header.ssrc], 1ul << 16); |
| if (time_series[parsed_header.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[parsed_header.ssrc].points.push_back( |
| TimeSeriesPoint(x, y)); |
| last_seqno[parsed_header.ssrc] = parsed_header.sequenceNumber; |
| } |
| } |
| } |
| } |
| |
| // Set labels and put in graph. |
| for (auto& kv : time_series) { |
| kv.second.label = SsrcToString(kv.first); |
| kv.second.style = BAR_GRAPH; |
| plot->series_list_.push_back(std::move(kv.second)); |
| } |
| |
| 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::CreateDelayChangeGraph(Plot* plot) { |
| for (auto& kv : rtp_packets_) { |
| StreamId stream_id = kv.first; |
| // Filter on direction and SSRC. |
| if (stream_id.GetDirection() != kIncomingPacket || |
| !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) { |
| continue; |
| } |
| |
| TimeSeries time_series; |
| time_series.label = SsrcToString(stream_id.GetSsrc()); |
| time_series.style = BAR_GRAPH; |
| const std::vector<LoggedRtpPacket>& packet_stream = kv.second; |
| int64_t last_abs_send_time = 0; |
| int64_t last_timestamp = 0; |
| for (const LoggedRtpPacket& packet : packet_stream) { |
| if (packet.header.extension.hasAbsoluteSendTime) { |
| int64_t send_time_diff = |
| WrappingDifference(packet.header.extension.absoluteSendTime, |
| last_abs_send_time, 1ul << 24); |
| int64_t recv_time_diff = packet.timestamp - last_timestamp; |
| |
| last_abs_send_time = packet.header.extension.absoluteSendTime; |
| last_timestamp = packet.timestamp; |
| |
| float x = static_cast<float>(packet.timestamp - begin_time_) / 1000000; |
| double y = |
| static_cast<double>(recv_time_diff - |
| AbsSendTimeToMicroseconds(send_time_diff)) / |
| 1000; |
| if (time_series.points.size() == 0) { |
| // There were no previously logged packets for this SSRC. |
| // Generate a point, but place it on the x-axis. |
| y = 0; |
| } |
| time_series.points.emplace_back(x, y); |
| } |
| } |
| // Add the data set to the plot. |
| plot->series_list_.push_back(std::move(time_series)); |
| } |
| |
| 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; |
| // Filter on direction and SSRC. |
| if (stream_id.GetDirection() != kIncomingPacket || |
| !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) { |
| continue; |
| } |
| TimeSeries time_series; |
| time_series.label = SsrcToString(stream_id.GetSsrc()); |
| time_series.style = LINE_GRAPH; |
| const std::vector<LoggedRtpPacket>& packet_stream = kv.second; |
| int64_t last_abs_send_time = 0; |
| int64_t last_timestamp = 0; |
| double accumulated_delay_ms = 0; |
| for (const LoggedRtpPacket& packet : packet_stream) { |
| if (packet.header.extension.hasAbsoluteSendTime) { |
| int64_t send_time_diff = |
| WrappingDifference(packet.header.extension.absoluteSendTime, |
| last_abs_send_time, 1ul << 24); |
| int64_t recv_time_diff = packet.timestamp - last_timestamp; |
| |
| last_abs_send_time = packet.header.extension.absoluteSendTime; |
| last_timestamp = packet.timestamp; |
| |
| float x = static_cast<float>(packet.timestamp - begin_time_) / 1000000; |
| accumulated_delay_ms += |
| static_cast<double>(recv_time_diff - |
| AbsSendTimeToMicroseconds(send_time_diff)) / |
| 1000; |
| if (time_series.points.size() == 0) { |
| // There were no previously logged packets for this SSRC. |
| // Generate a point, but place it on the x-axis. |
| accumulated_delay_ms = 0; |
| } |
| time_series.points.emplace_back(x, accumulated_delay_ms); |
| } |
| } |
| // Add the data set to the plot. |
| plot->series_list_.push_back(std::move(time_series)); |
| } |
| |
| 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) { |
| plot->series_list_.push_back(TimeSeries()); |
| 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; |
| plot->series_list_.back().points.emplace_back(x, y); |
| } |
| plot->series_list_.back().label = "Fraction lost"; |
| plot->series_list_.back().style = LINE_DOT_GRAPH; |
| |
| 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 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, 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. |
| plot->series_list_.push_back(TimeSeries()); |
| 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; |
| plot->series_list_.back().points.push_back(TimeSeriesPoint(x, y)); |
| } |
| |
| // Set labels. |
| if (desired_direction == webrtc::PacketDirection::kIncomingPacket) { |
| plot->series_list_.back().label = "Incoming bitrate"; |
| } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) { |
| plot->series_list_.back().label = "Outgoing bitrate"; |
| } |
| plot->series_list_.back().style = LINE_GRAPH; |
| |
| // Overlay the send-side bandwidth estimate over the outgoing bitrate. |
| if (desired_direction == kOutgoingPacket) { |
| plot->series_list_.push_back(TimeSeries()); |
| 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.new_bitrate) / 1000; |
| plot->series_list_.back().points.emplace_back(x, y); |
| } |
| plot->series_list_.back().label = "Loss-based estimate"; |
| plot->series_list_.back().style = LINE_GRAPH; |
| } |
| plot->series_list_.back().style = LINE_GRAPH; |
| 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) { |
| struct TimestampSize { |
| TimestampSize(uint64_t t, size_t s) : timestamp(t), size(s) {} |
| uint64_t timestamp; |
| size_t size; |
| }; |
| std::map<uint32_t, std::vector<TimestampSize>> packets; |
| |
| PacketDirection direction; |
| MediaType media_type; |
| uint8_t header[IP_PACKET_SIZE]; |
| size_t header_length, 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, &media_type, header, |
| &header_length, &total_length); |
| if (direction == desired_direction) { |
| // Parse header to get SSRC. |
| RtpUtility::RtpHeaderParser rtp_parser(header, header_length); |
| RTPHeader parsed_header; |
| rtp_parser.Parse(&parsed_header); |
| // Filter on SSRC. |
| if (MatchingSsrc(parsed_header.ssrc, desired_ssrc_)) { |
| uint64_t timestamp = parsed_log_.GetTimestamp(i); |
| packets[parsed_header.ssrc].push_back( |
| TimestampSize(timestamp, total_length)); |
| } |
| } |
| } |
| } |
| |
| for (auto& kv : packets) { |
| 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. |
| plot->series_list_.push_back(TimeSeries()); |
| for (uint64_t time = begin_time_; time < end_time_ + step_; time += step_) { |
| while (window_index_end < kv.second.size() && |
| kv.second[window_index_end].timestamp < time) { |
| bytes_in_window += kv.second[window_index_end].size; |
| window_index_end++; |
| } |
| while (window_index_begin < kv.second.size() && |
| kv.second[window_index_begin].timestamp < |
| time - window_duration_) { |
| RTC_DCHECK_LE(kv.second[window_index_begin].size, bytes_in_window); |
| bytes_in_window -= kv.second[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; |
| plot->series_list_.back().points.push_back(TimeSeriesPoint(x, y)); |
| } |
| |
| // Set labels. |
| plot->series_list_.back().label = SsrcToString(kv.first); |
| plot->series_list_.back().style = LINE_GRAPH; |
| } |
| |
| 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::CreateBweGraph(Plot* plot) { |
| std::map<uint64_t, const LoggedRtpPacket*> outgoing_rtp; |
| std::map<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; |
| CongestionController cc(&clock, &observer, &observer, &null_event_log); |
| // 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; |
| time_series.label = "BWE"; |
| time_series.style = 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(); |
| }; |
| |
| 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) { |
| cc.GetTransportFeedbackObserver()->OnTransportFeedback( |
| *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get())); |
| } |
| ++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.GetTransportFeedbackObserver()->AddPacket( |
| rtp.header.extension.transportSequenceNumber, rtp.total_length, 0); |
| 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()) { |
| 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); |
| } |
| time_us = std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()}); |
| } |
| // Add the data set to the plot. |
| plot->series_list_.push_back(std::move(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::CreateNetworkDelayFeebackGraph(Plot* plot) { |
| std::map<uint64_t, const LoggedRtpPacket*> outgoing_rtp; |
| std::map<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(nullptr, &clock); |
| |
| TimeSeries time_series; |
| time_series.label = "Network Delay Change"; |
| time_series.style = 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) { |
| std::vector<PacketInfo> feedback = |
| feedback_adapter.GetPacketFeedbackVector( |
| *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get())); |
| for (const PacketInfo& 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.extension.transportSequenceNumber, |
| rtp.total_length, 0); |
| 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->series_list_.push_back(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."); |
| } |
| } // namespace plotting |
| } // namespace webrtc |
