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Unified Diff: webrtc/tools/event_log_visualizer/analyzer.cc

Issue 2147453002: Revert of Visualization tool for WebrtcEventLogs (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 5 months ago
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Index: webrtc/tools/event_log_visualizer/analyzer.cc
diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc
deleted file mode 100644
index 05d94ee0633dc316ff3f7ede094b5c1e62a7314e..0000000000000000000000000000000000000000
--- a/webrtc/tools/event_log_visualizer/analyzer.cc
+++ /dev/null
@@ -1,710 +0,0 @@
-/*
- * 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/call.h"
-#include "webrtc/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/rtp_utility.h"
-#include "webrtc/video_receive_stream.h"
-#include "webrtc/video_send_stream.h"
-
-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;
-}
-
-class StreamId {
- public:
- StreamId(uint32_t ssrc,
- webrtc::PacketDirection direction,
- webrtc::MediaType media_type)
- : ssrc_(ssrc), direction_(direction), media_type_(media_type) {}
-
- bool operator<(const StreamId& other) const {
- if (ssrc_ < other.ssrc_) {
- return true;
- }
- if (ssrc_ == other.ssrc_) {
- if (media_type_ < other.media_type_) {
- return true;
- }
- if (media_type_ == other.media_type_) {
- if (direction_ < other.direction_) {
- return true;
- }
- }
- }
- return false;
- }
-
- bool operator==(const StreamId& other) const {
- return ssrc_ == other.ssrc_ && direction_ == other.direction_ &&
- media_type_ == other.media_type_;
- }
-
- uint32_t GetSsrc() const { return ssrc_; }
-
- private:
- uint32_t ssrc_;
- webrtc::PacketDirection direction_;
- webrtc::MediaType media_type_;
-};
-
-const double kXMargin = 1.02;
-const double kYMargin = 1.1;
-const double kDefaultXMin = -1;
-const double kDefaultYMin = -1;
-
-} // namespace
-
-namespace webrtc {
-namespace plotting {
-
-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();
- for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
- ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
- if (event_type == ParsedRtcEventLog::EventType::VIDEO_RECEIVER_CONFIG_EVENT)
- continue;
- if (event_type == ParsedRtcEventLog::EventType::VIDEO_SENDER_CONFIG_EVENT)
- continue;
- if (event_type == ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT)
- continue;
- if (event_type == ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT)
- continue;
- uint64_t timestamp = parsed_log_.GetTimestamp(i);
- first_timestamp = std::min(first_timestamp, timestamp);
- last_timestamp = std::max(last_timestamp, timestamp);
- }
- if (last_timestamp < first_timestamp) {
- // No useful events in the log.
- first_timestamp = last_timestamp = 0;
- }
- begin_time_ = first_timestamp;
- end_time_ = last_timestamp;
-}
-
-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;
- float max_y = 0;
-
- 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;
- max_y = std::max(max_y, y);
- 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.push_back(std::move(kv.second));
- }
-
- plot->xaxis_min = kDefaultXMin;
- plot->xaxis_max = (end_time_ - begin_time_) / 1000000 * kXMargin;
- plot->xaxis_label = "Time (s)";
- plot->yaxis_min = kDefaultYMin;
- plot->yaxis_max = max_y * kYMargin;
- plot->yaxis_label = "Packet size (bytes)";
- if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
- plot->title = "Incoming RTP packets";
- } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
- plot->title = "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;
- float max_y = 0;
-
- 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;
- }
- max_y = std::max(max_y, y);
- 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.push_back(std::move(kv.second));
- }
-
- plot->xaxis_min = kDefaultXMin;
- plot->xaxis_max = (end_time_ - begin_time_) / 1000000 * kXMargin;
- plot->xaxis_label = "Time (s)";
- plot->yaxis_min = kDefaultYMin;
- plot->yaxis_max = max_y * kYMargin;
- plot->yaxis_label = "Time since last playout (ms)";
- plot->title = "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;
-
- int max_y = 1;
- int min_y = 0;
-
- 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;
- }
- max_y = std::max(max_y, y);
- min_y = std::min(min_y, y);
- 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.push_back(std::move(kv.second));
- }
-
- plot->xaxis_min = kDefaultXMin;
- plot->xaxis_max = (end_time_ - begin_time_) / 1000000 * kXMargin;
- plot->xaxis_label = "Time (s)";
- plot->yaxis_min = min_y - (kYMargin - 1) / 2 * (max_y - min_y);
- plot->yaxis_max = max_y + (kYMargin - 1) / 2 * (max_y - min_y);
- plot->yaxis_label = "Difference since last packet";
- plot->title = "Sequence number";
-}
-
-void EventLogAnalyzer::CreateDelayChangeGraph(Plot* plot) {
- // Maps a stream identifier consisting of ssrc, direction and MediaType
- // to the header extensions used by that stream,
- std::map<StreamId, RtpHeaderExtensionMap> extension_maps;
-
- struct SendReceiveTime {
- SendReceiveTime() = default;
- SendReceiveTime(uint32_t send_time, uint64_t recv_time)
- : absolute_send_time(send_time), receive_timestamp(recv_time) {}
- uint32_t absolute_send_time; // 24-bit value in units of 2^-18 seconds.
- uint64_t receive_timestamp; // In microseconds.
- };
- std::map<StreamId, SendReceiveTime> last_packet;
- std::map<StreamId, TimeSeries> time_series;
-
- PacketDirection direction;
- MediaType media_type;
- uint8_t header[IP_PACKET_SIZE];
- size_t header_length, total_length;
-
- double max_y = 10;
- double min_y = 0;
-
- 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) {
- VideoReceiveStream::Config config(nullptr);
- parsed_log_.GetVideoReceiveConfig(i, &config);
- StreamId stream(config.rtp.remote_ssrc, kIncomingPacket,
- MediaType::VIDEO);
- extension_maps[stream].Erase();
- for (size_t j = 0; j < config.rtp.extensions.size(); ++j) {
- const std::string& extension = config.rtp.extensions[j].uri;
- int id = config.rtp.extensions[j].id;
- extension_maps[stream].Register(StringToRtpExtensionType(extension),
- id);
- }
- } else if (event_type == ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
- VideoSendStream::Config config(nullptr);
- parsed_log_.GetVideoSendConfig(i, &config);
- for (auto ssrc : config.rtp.ssrcs) {
- StreamId stream(ssrc, kIncomingPacket, MediaType::VIDEO);
- extension_maps[stream].Erase();
- for (size_t j = 0; j < config.rtp.extensions.size(); ++j) {
- const std::string& extension = config.rtp.extensions[j].uri;
- int id = config.rtp.extensions[j].id;
- extension_maps[stream].Register(StringToRtpExtensionType(extension),
- id);
- }
- }
- } else if (event_type == ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
- AudioReceiveStream::Config config;
- // TODO(terelius): Parse the audio configs once we have them
- } else if (event_type == ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
- AudioSendStream::Config config(nullptr);
- // TODO(terelius): Parse the audio configs once we have them
- } else if (event_type == ParsedRtcEventLog::RTP_EVENT) {
- parsed_log_.GetRtpHeader(i, &direction, &media_type, header,
- &header_length, &total_length);
- if (direction == 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_)) {
- StreamId stream(parsed_header.ssrc, direction, media_type);
- // 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);
- if (parsed_header.extension.hasAbsoluteSendTime) {
- uint64_t timestamp = parsed_log_.GetTimestamp(i);
- int64_t send_time_diff = WrappingDifference(
- parsed_header.extension.absoluteSendTime,
- last_packet[stream].absolute_send_time, 1ul << 24);
- int64_t recv_time_diff =
- timestamp - last_packet[stream].receive_timestamp;
-
- float x = static_cast<float>(timestamp - begin_time_) / 1000000;
- double y = static_cast<double>(
- recv_time_diff -
- AbsSendTimeToMicroseconds(send_time_diff)) /
- 1000;
- if (time_series[stream].points.size() == 0) {
- // There were no previusly logged playout for this SSRC.
- // Generate a point, but place it on the x-axis.
- y = 0;
- }
- max_y = std::max(max_y, y);
- min_y = std::min(min_y, y);
- time_series[stream].points.push_back(TimeSeriesPoint(x, y));
- last_packet[stream] = SendReceiveTime(
- parsed_header.extension.absoluteSendTime, timestamp);
- }
- }
- }
- }
- }
- }
-
- // Set labels and put in graph.
- for (auto& kv : time_series) {
- kv.second.label = SsrcToString(kv.first.GetSsrc());
- kv.second.style = BAR_GRAPH;
- plot->series.push_back(std::move(kv.second));
- }
-
- plot->xaxis_min = kDefaultXMin;
- plot->xaxis_max = (end_time_ - begin_time_) / 1000000 * kXMargin;
- plot->xaxis_label = "Time (s)";
- plot->yaxis_min = min_y - (kYMargin - 1) / 2 * (max_y - min_y);
- plot->yaxis_max = max_y + (kYMargin - 1) / 2 * (max_y - min_y);
- plot->yaxis_label = "Latency change (ms)";
- plot->title = "Network latency change between consecutive packets";
-}
-
-void EventLogAnalyzer::CreateAccumulatedDelayChangeGraph(Plot* plot) {
- // TODO(terelius): Refactor
-
- // Maps a stream identifier consisting of ssrc, direction and MediaType
- // to the header extensions used by that stream.
- std::map<StreamId, RtpHeaderExtensionMap> extension_maps;
-
- struct SendReceiveTime {
- SendReceiveTime() = default;
- SendReceiveTime(uint32_t send_time, uint64_t recv_time, double accumulated)
- : absolute_send_time(send_time),
- receive_timestamp(recv_time),
- accumulated_delay(accumulated) {}
- uint32_t absolute_send_time; // 24-bit value in units of 2^-18 seconds.
- uint64_t receive_timestamp; // In microseconds.
- double accumulated_delay; // In milliseconds.
- };
- std::map<StreamId, SendReceiveTime> last_packet;
- std::map<StreamId, TimeSeries> time_series;
-
- PacketDirection direction;
- MediaType media_type;
- uint8_t header[IP_PACKET_SIZE];
- size_t header_length, total_length;
-
- double max_y = 10;
- double min_y = 0;
-
- 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) {
- VideoReceiveStream::Config config(nullptr);
- parsed_log_.GetVideoReceiveConfig(i, &config);
- StreamId stream(config.rtp.remote_ssrc, kIncomingPacket,
- MediaType::VIDEO);
- extension_maps[stream].Erase();
- for (size_t j = 0; j < config.rtp.extensions.size(); ++j) {
- const std::string& extension = config.rtp.extensions[j].uri;
- int id = config.rtp.extensions[j].id;
- extension_maps[stream].Register(StringToRtpExtensionType(extension),
- id);
- }
- } else if (event_type == ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
- VideoSendStream::Config config(nullptr);
- parsed_log_.GetVideoSendConfig(i, &config);
- for (auto ssrc : config.rtp.ssrcs) {
- StreamId stream(ssrc, kIncomingPacket, MediaType::VIDEO);
- extension_maps[stream].Erase();
- for (size_t j = 0; j < config.rtp.extensions.size(); ++j) {
- const std::string& extension = config.rtp.extensions[j].uri;
- int id = config.rtp.extensions[j].id;
- extension_maps[stream].Register(StringToRtpExtensionType(extension),
- id);
- }
- }
- } else if (event_type == ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
- AudioReceiveStream::Config config;
- // TODO(terelius): Parse the audio configs once we have them
- } else if (event_type == ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
- AudioSendStream::Config config(nullptr);
- // TODO(terelius): Parse the audio configs once we have them
- } else if (event_type == ParsedRtcEventLog::RTP_EVENT) {
- parsed_log_.GetRtpHeader(i, &direction, &media_type, header,
- &header_length, &total_length);
- if (direction == 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_)) {
- StreamId stream(parsed_header.ssrc, direction, media_type);
- // 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);
- if (parsed_header.extension.hasAbsoluteSendTime) {
- uint64_t timestamp = parsed_log_.GetTimestamp(i);
- int64_t send_time_diff = WrappingDifference(
- parsed_header.extension.absoluteSendTime,
- last_packet[stream].absolute_send_time, 1ul << 24);
- int64_t recv_time_diff =
- timestamp - last_packet[stream].receive_timestamp;
-
- float x = static_cast<float>(timestamp - begin_time_) / 1000000;
- double y = last_packet[stream].accumulated_delay +
- static_cast<double>(
- recv_time_diff -
- AbsSendTimeToMicroseconds(send_time_diff)) /
- 1000;
- if (time_series[stream].points.size() == 0) {
- // There were no previusly logged playout for this SSRC.
- // Generate a point, but place it on the x-axis.
- y = 0;
- }
- max_y = std::max(max_y, y);
- min_y = std::min(min_y, y);
- time_series[stream].points.push_back(TimeSeriesPoint(x, y));
- last_packet[stream] = SendReceiveTime(
- parsed_header.extension.absoluteSendTime, timestamp, y);
- }
- }
- }
- }
- }
- }
-
- // Set labels and put in graph.
- for (auto& kv : time_series) {
- kv.second.label = SsrcToString(kv.first.GetSsrc());
- kv.second.style = LINE_GRAPH;
- plot->series.push_back(std::move(kv.second));
- }
-
- plot->xaxis_min = kDefaultXMin;
- plot->xaxis_max = (end_time_ - begin_time_) / 1000000 * kXMargin;
- plot->xaxis_label = "Time (s)";
- plot->yaxis_min = min_y - (kYMargin - 1) / 2 * (max_y - min_y);
- plot->yaxis_max = max_y + (kYMargin - 1) / 2 * (max_y - min_y);
- plot->yaxis_label = "Latency change (ms)";
- plot->title = "Accumulated network latency change";
-}
-
-// 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;
- float max_y = 0;
-
- // Calculate a moving average of the bitrate and store in a TimeSeries.
- plot->series.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;
- max_y = std::max(max_y, y);
- plot->series.back().points.push_back(TimeSeriesPoint(x, y));
- }
-
- // Set labels.
- if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
- plot->series.back().label = "Incoming bitrate";
- } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
- plot->series.back().label = "Outgoing bitrate";
- }
- plot->series.back().style = LINE_GRAPH;
-
- plot->xaxis_min = kDefaultXMin;
- plot->xaxis_max = (end_time_ - begin_time_) / 1000000 * kXMargin;
- plot->xaxis_label = "Time (s)";
- plot->yaxis_min = kDefaultYMin;
- plot->yaxis_max = max_y * kYMargin;
- plot->yaxis_label = "Bitrate (kbps)";
- if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
- plot->title = "Incoming RTP bitrate";
- } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
- plot->title = "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));
- }
- }
- }
- }
-
- float max_y = 0;
-
- 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.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;
- max_y = std::max(max_y, y);
- plot->series.back().points.push_back(TimeSeriesPoint(x, y));
- }
-
- // Set labels.
- plot->series.back().label = SsrcToString(kv.first);
- plot->series.back().style = LINE_GRAPH;
- }
-
- plot->xaxis_min = kDefaultXMin;
- plot->xaxis_max = (end_time_ - begin_time_) / 1000000 * kXMargin;
- plot->xaxis_label = "Time (s)";
- plot->yaxis_min = kDefaultYMin;
- plot->yaxis_max = max_y * kYMargin;
- plot->yaxis_label = "Bitrate (kbps)";
- if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
- plot->title = "Incoming bitrate per stream";
- } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
- plot->title = "Outgoing bitrate per stream";
- }
-}
-
-} // namespace plotting
-} // namespace webrtc
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