Visualization tool for WebrtcEventLogs

webrtc/tools/event_log_visualizer/analyzer.cc

+/*
+ *  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 HeaderToString(const webrtc::RTPHeader& parsed_header) {

[stefan-webrtc, 2016/06/29 11:13:05]

Should we make this a method on RTPHeader instead?

[terelius, 2016/07/01 16:54:45]

I have no opinion on this, but this function does
not serialize all fields of the header, only a few that
I though would be interesting to look at. If it was a
method on RTPHEader, I guess it should be more general.

[stefan-webrtc, 2016/07/05 08:58:11]

I think it's fine as is for now.

+  std::stringstream ss;
+  ss << "Marker=" << parsed_header.markerBit
+     << ", PType=" << parsed_header.payloadType
+     << ", SeqNum=" << parsed_header.sequenceNumber
+     << ", CaptureTime=" << parsed_header.timestamp
+     << ", SSRC=" << parsed_header.ssrc;
+  return ss.str();
+}
+
+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;
+}
+
+// There could conceivably be e.g. an incoming and an outgoing stream with
+// the same SSRC. To get a unique identifier for each stream we append some
+// other values to the SSRC.
+uint64_t GetStreamId(uint32_t ssrc,

[stefan-webrtc, 2016/06/29 11:13:04]

Wouldn't it be nicer to make stream id a comparable type instead?
class StreamId {
 public:
  operator<() {...}

 private:
  uint32_t ssrc_;
  webrtc::PacketDirection direction_;
  webrtc::MediaType media_type_;
};

Seems like a less hacky way of achieving the same thing

[terelius, 2016/07/01 16:54:45]

Done. However, the advantage of precomputing a 64-bit identifier is that we
would only need 1 comparison instead of 5 when we're comparing two streams.

+                     webrtc::PacketDirection direction,
+                     webrtc::MediaType media_type) {
+  uint64_t stream = ssrc;
+  stream = (stream << 8) +
+           static_cast<uint64_t>(direction == webrtc::kIncomingPacket);
+  stream = (stream << 8) + static_cast<uint64_t>(media_type);
+  return stream;
+}
+
+uint32_t GetSsrcFromStreamId(uint64_t stream) {
+  return stream >> 16;
+}
+
+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,
+                                   bool extra_info)
+    : parsed_log_(log), extra_point_info_(extra_info) {
+  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;
+  window_duration_ = 250000;
+  step_ = 10000;

[stefan-webrtc, 2016/06/29 11:13:04]

These two members should be set in the initializer list.

[terelius, 2016/07/01 16:54:45]

Done.

+}
+
+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);
+          std::string message;
+          if (extra_point_info_) {
+            message = HeaderToString(parsed_header);
+          }
+          time_series[parsed_header.ssrc].points.push_back(
+              TimeSeriesPoint(x, y, message));
+        }
+      }
+    }
+  }
+
+  // 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(TimeSeries());
+    plot->series.back().swap(kv.second);

[stefan-webrtc, 2016/06/29 11:13:04]

Why not simply push_back(kv.second)?

[terelius, 2016/07/01 16:54:45]

Because I don't want to copy all the data in the TimeSeries object.
I guess the better way to achieve the same effect is to
push_back(std::move(kv.second)) and overload the move constructor and move
assignment operator.

+  }
+
+  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(TimeSeries());
+    plot->series.back().swap(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(TimeSeries());
+    plot->series.back().swap(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<uint64_t, 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<uint64_t, SendReceiveTime> last_packet;
+  std::map<uint64_t, 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) {

[stefan-webrtc, 2016/06/29 11:13:04]

Handling of config events seems to happen in many Create...Graph() methods. Can
we break it out? I assume it can be handled similarly for all graphs?

[terelius, 2016/07/01 16:54:45]

Config events are used in CreateDelayChangeGraph and
CreateAccumulatedDelayChangeGraph to get the header extensions.

I have another CL that breaks out parsing of the RTP packets. That will parse
the packets once and cache a map from SSRC to RTPHeader. Do you want me to merge
that into this CL?

[stefan-webrtc, 2016/07/05 08:58:11]

If it becomes better in a follow up I think I'm ok with waiting for that.

+      VideoReceiveStream::Config config(nullptr);
+      parsed_log_.GetVideoReceiveConfig(i, &config);
+      uint64_t stream = GetStreamId(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) {
+        uint64_t stream = GetStreamId(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_)) {

[stefan-webrtc, 2016/06/29 11:13:05]

Maybe we could have a method similar to:
GetEvents(ssrc, direction, types)

which returns a list of events matching the filters we're interested in? That
way we would separate filtering from the plotting and computation code. WDYT?

[terelius, 2016/07/01 16:54:45]

The follow up CL that caches a map from streams to parsed headers would mostly
accomplish this. We could iterate over the streams and "continue;" for all
streams we are not interested in.

[stefan-webrtc, 2016/07/05 08:58:11]

Acknowledged.

+          uint64_t stream =
+              GetStreamId(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(GetSsrcFromStreamId(kv.first));
+    kv.second.style = BAR_GRAPH;
+    plot->series.push_back(TimeSeries());
+    plot->series.back().swap(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<uint64_t, 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<uint64_t, SendReceiveTime> last_packet;
+  std::map<uint64_t, 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);
+      uint64_t stream = GetStreamId(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) {
+        uint64_t stream = GetStreamId(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_)) {
+          uint64_t stream =
+              GetStreamId(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(GetSsrcFromStreamId(kv.first));
+    kv.second.style = LINE_GRAPH;
+    plot->series.push_back(TimeSeries());
+    plot->series.back().swap(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 bandwitch 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_) {
+      bytes_in_window -= packets[window_index_begin].size;
+      window_index_begin++;
+    }
+    RTC_DCHECK_LE(0ul, bytes_in_window);
+    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 bandwitch used by that stream.

[stefan-webrtc, 2016/06/29 11:13:04]

bandwidth

[terelius, 2016/07/01 16:54:45]

Done. :)

[stefan-webrtc, 2016/07/05 08:58:11]

Nope, still there... :)

[terelius, 2016/07/06 15:15:12]

Done.

+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_) {
+        bytes_in_window -= kv.second[window_index_begin].size;
+        window_index_begin++;
+      }
+      RTC_DCHECK_LE(0ul, bytes_in_window);
+      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