diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index f151f10..58f69a1 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -358,7 +358,7 @@
     configs += [ ":common_config" ]
     public_configs = [ ":common_inherited_config" ]
 
-    deps = [
+    public_deps = [
       ":rtc_event_log_proto",
       ":webrtc_common",
     ]
diff --git a/webrtc/tools/BUILD.gn b/webrtc/tools/BUILD.gn
index 2d09563..abf0ca0 100644
--- a/webrtc/tools/BUILD.gn
+++ b/webrtc/tools/BUILD.gn
@@ -171,6 +171,36 @@
 
 # Exclude tools depending on gflags since that's not available in Chromium.
 if (rtc_include_tests) {
+  if (rtc_enable_protobuf) {
+    executable("event_log_visualizer") {
+      sources = [
+        "event_log_visualizer/analyzer.cc",
+        "event_log_visualizer/analyzer.h",
+        "event_log_visualizer/generate_timeseries.cc",
+        "event_log_visualizer/plot_base.h",
+        "event_log_visualizer/plot_python.cc",
+        "event_log_visualizer/plot_python.h",
+      ]
+
+      configs += [ "..:common_config" ]
+      public_configs = [ "..:common_inherited_config" ]
+
+      if (is_clang && !is_nacl) {
+        # Suppress warnings from the Chromium Clang plugin.
+        # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+        configs -= [ "//build/config/clang:find_bad_constructs" ]
+      }
+
+      defines = [ "ENABLE_RTC_EVENT_LOG" ]
+      deps = [
+        "../:rtc_event_log_parser",
+        "../modules/rtp_rtcp:rtp_rtcp",
+        "../system_wrappers:system_wrappers_default",
+        "//third_party/gflags",
+      ]
+    }
+  }
+
   executable("agc_harness") {
     testonly = true
     sources = [
diff --git a/webrtc/tools/DEPS b/webrtc/tools/DEPS
index 73073f0..33df89f 100644
--- a/webrtc/tools/DEPS
+++ b/webrtc/tools/DEPS
@@ -1,7 +1,9 @@
 include_rules = [
   "+webrtc/base",
+  "+webrtc/call",
   "+webrtc/common_video",
   "+webrtc/modules/audio_processing",
+  "+webrtc/modules/rtp_rtcp",
   "+webrtc/system_wrappers",
   "+webrtc/voice_engine",
 ]
diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc
new file mode 100644
index 0000000..05d94ee
--- /dev/null
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc
@@ -0,0 +1,710 @@
+/*
+ *  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
diff --git a/webrtc/tools/event_log_visualizer/analyzer.h b/webrtc/tools/event_log_visualizer/analyzer.h
new file mode 100644
index 0000000..08fcd5c
--- /dev/null
+++ b/webrtc/tools/event_log_visualizer/analyzer.h
@@ -0,0 +1,65 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
+#define WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
+
+#include <vector>
+
+#include "webrtc/call/rtc_event_log_parser.h"
+#include "webrtc/tools/event_log_visualizer/plot_base.h"
+
+namespace webrtc {
+namespace plotting {
+
+class EventLogAnalyzer {
+ public:
+  // The EventLogAnalyzer keeps a reference to the ParsedRtcEventLog for the
+  // duration of its lifetime. The ParsedRtcEventLog must not be destroyed or
+  // modified while the EventLogAnalyzer is being used.
+  explicit EventLogAnalyzer(const ParsedRtcEventLog& log);
+
+  void CreatePacketGraph(PacketDirection desired_direction, Plot* plot);
+
+  void CreatePlayoutGraph(Plot* plot);
+
+  void CreateSequenceNumberGraph(Plot* plot);
+
+  void CreateDelayChangeGraph(Plot* plot);
+
+  void CreateAccumulatedDelayChangeGraph(Plot* plot);
+
+  void CreateTotalBitrateGraph(PacketDirection desired_direction, Plot* plot);
+
+  void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot);
+
+ private:
+  const ParsedRtcEventLog& parsed_log_;
+
+  // A list of SSRCs we are interested in analysing.
+  // If left empty, all SSRCs will be considered relevant.
+  std::vector<uint32_t> desired_ssrc_;
+
+  // Window and step size used for calculating moving averages, e.g. bitrate.
+  // The generated data points will be |step_| microseconds apart.
+  // Only events occuring at most |window_duration_| microseconds before the
+  // current data point will be part of the average.
+  uint64_t window_duration_;
+  uint64_t step_;
+
+  // First and last events of the log.
+  uint64_t begin_time_;
+  uint64_t end_time_;
+};
+
+}  // namespace plotting
+}  // namespace webrtc
+
+#endif  // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
diff --git a/webrtc/tools/event_log_visualizer/generate_timeseries.cc b/webrtc/tools/event_log_visualizer/generate_timeseries.cc
new file mode 100644
index 0000000..d213947
--- /dev/null
+++ b/webrtc/tools/event_log_visualizer/generate_timeseries.cc
@@ -0,0 +1,138 @@
+/*
+ *  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 <iostream>
+
+#include "gflags/gflags.h"
+#include "webrtc/call/rtc_event_log_parser.h"
+#include "webrtc/tools/event_log_visualizer/analyzer.h"
+#include "webrtc/tools/event_log_visualizer/plot_base.h"
+#include "webrtc/tools/event_log_visualizer/plot_python.h"
+
+DEFINE_bool(incoming, true, "Plot statistics for incoming packets.");
+DEFINE_bool(outgoing, true, "Plot statistics for outgoing packets.");
+DEFINE_bool(plot_all, true, "Plot all different data types.");
+DEFINE_bool(plot_packets,
+            false,
+            "Plot bar graph showing the size of each packet.");
+DEFINE_bool(plot_audio_playout,
+            false,
+            "Plot bar graph showing the time between each audio playout.");
+DEFINE_bool(
+    plot_sequence_number,
+    false,
+    "Plot the difference in sequence number between consecutive packets.");
+DEFINE_bool(
+    plot_delay_change,
+    false,
+    "Plot the difference in 1-way path delay between consecutive packets.");
+DEFINE_bool(plot_accumulated_delay_change,
+            false,
+            "Plot the accumulated 1-way path delay change, or the path delay "
+            "change compared to the first packet.");
+DEFINE_bool(plot_total_bitrate,
+            false,
+            "Plot the total bitrate used by all streams.");
+DEFINE_bool(plot_stream_bitrate,
+            false,
+            "Plot the bitrate used by each stream.");
+
+int main(int argc, char* argv[]) {
+  std::string program_name = argv[0];
+  std::string usage =
+      "A tool for visualizing WebRTC event logs.\n"
+      "Example usage:\n" +
+      program_name + " <logfile> | python\n" + "Run " + program_name +
+      " --help for a list of command line options\n";
+  google::SetUsageMessage(usage);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+
+  if (argc != 2) {
+    // Print usage information.
+    std::cout << google::ProgramUsage();
+    return 0;
+  }
+
+  std::string filename = argv[1];
+
+  webrtc::ParsedRtcEventLog parsed_log;
+
+  if (!parsed_log.ParseFile(filename)) {
+    std::cerr << "Could not parse the entire log file." << std::endl;
+    std::cerr << "Proceeding to analyze the first "
+              << parsed_log.GetNumberOfEvents() << " events in the file."
+              << std::endl;
+  }
+
+  webrtc::plotting::EventLogAnalyzer analyzer(parsed_log);
+  std::unique_ptr<webrtc::plotting::PlotCollection> collection(
+      new webrtc::plotting::PythonPlotCollection());
+
+  if (FLAGS_plot_all || FLAGS_plot_packets) {
+    if (FLAGS_incoming) {
+      analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket,
+                                 collection->append_new_plot());
+    }
+    if (FLAGS_outgoing) {
+      analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket,
+                                 collection->append_new_plot());
+    }
+  }
+
+  if (FLAGS_plot_all || FLAGS_plot_audio_playout) {
+    analyzer.CreatePlayoutGraph(collection->append_new_plot());
+  }
+
+  if (FLAGS_plot_all || FLAGS_plot_sequence_number) {
+    if (FLAGS_incoming) {
+      analyzer.CreateSequenceNumberGraph(collection->append_new_plot());
+    }
+  }
+
+  if (FLAGS_plot_all || FLAGS_plot_delay_change) {
+    if (FLAGS_incoming) {
+      analyzer.CreateDelayChangeGraph(collection->append_new_plot());
+    }
+  }
+
+  if (FLAGS_plot_all || FLAGS_plot_accumulated_delay_change) {
+    if (FLAGS_incoming) {
+      analyzer.CreateAccumulatedDelayChangeGraph(collection->append_new_plot());
+    }
+  }
+
+  if (FLAGS_plot_all || FLAGS_plot_total_bitrate) {
+    if (FLAGS_incoming) {
+      analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
+                                       collection->append_new_plot());
+    }
+    if (FLAGS_outgoing) {
+      analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
+                                       collection->append_new_plot());
+    }
+  }
+
+  if (FLAGS_plot_all || FLAGS_plot_stream_bitrate) {
+    if (FLAGS_incoming) {
+      analyzer.CreateStreamBitrateGraph(
+          webrtc::PacketDirection::kIncomingPacket,
+          collection->append_new_plot());
+    }
+    if (FLAGS_outgoing) {
+      analyzer.CreateStreamBitrateGraph(
+          webrtc::PacketDirection::kOutgoingPacket,
+          collection->append_new_plot());
+    }
+  }
+
+  collection->draw();
+
+  return 0;
+}
diff --git a/webrtc/tools/event_log_visualizer/plot_base.h b/webrtc/tools/event_log_visualizer/plot_base.h
new file mode 100644
index 0000000..925dcec
--- /dev/null
+++ b/webrtc/tools/event_log_visualizer/plot_base.h
@@ -0,0 +1,78 @@
+/*
+ *  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.
+ */
+#ifndef WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_BASE_H_
+#define WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_BASE_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+namespace webrtc {
+namespace plotting {
+
+enum PlotStyle { LINE_GRAPH, BAR_GRAPH };
+
+struct TimeSeriesPoint {
+  TimeSeriesPoint(float x, float y) : x(x), y(y) {}
+  float x;
+  float y;
+};
+
+struct TimeSeries {
+  TimeSeries() = default;
+  TimeSeries(TimeSeries&& other)
+      : label(std::move(other.label)),
+        style(other.style),
+        points(std::move(other.points)) {}
+  TimeSeries& operator=(TimeSeries&& other) {
+    label = std::move(other.label);
+    style = other.style;
+    points = std::move(other.points);
+    return *this;
+  }
+
+  std::string label;
+  PlotStyle style;
+  std::vector<TimeSeriesPoint> points;
+};
+
+// This is basically a struct that represents of a general graph, with axes,
+// title and one or more data series. We make it a class only to document that
+// it also specifies an interface for the draw()ing objects.
+class Plot {
+ public:
+  virtual ~Plot() {}
+  virtual void draw() = 0;
+
+  float xaxis_min;
+  float xaxis_max;
+  std::string xaxis_label;
+  float yaxis_min;
+  float yaxis_max;
+  std::string yaxis_label;
+  std::vector<TimeSeries> series;
+  std::string title;
+};
+
+class PlotCollection {
+ public:
+  virtual ~PlotCollection() {}
+  virtual void draw() = 0;
+  virtual Plot* append_new_plot() = 0;
+
+ protected:
+  std::vector<std::unique_ptr<Plot> > plots;
+};
+
+}  // namespace plotting
+}  // namespace webrtc
+
+#endif  // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_BASE_H_
diff --git a/webrtc/tools/event_log_visualizer/plot_python.cc b/webrtc/tools/event_log_visualizer/plot_python.cc
new file mode 100644
index 0000000..916bc2e
--- /dev/null
+++ b/webrtc/tools/event_log_visualizer/plot_python.cc
@@ -0,0 +1,98 @@
+/*
+ *  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/plot_python.h"
+
+#include <stdio.h>
+
+namespace webrtc {
+namespace plotting {
+
+PythonPlot::PythonPlot() {}
+
+PythonPlot::~PythonPlot() {}
+
+void PythonPlot::draw() {
+  // Write python commands to stdout. Intended program usage is
+  // ./event_log_visualizer event_log160330.dump | python
+
+  if (!series.empty()) {
+    printf("color_count = %zu\n", series.size());
+    printf(
+        "hls_colors = [(i*1.0/color_count, 0.25+i*0.5/color_count, 0.8) for i "
+        "in range(color_count)]\n");
+    printf("rgb_colors = [colorsys.hls_to_rgb(*hls) for hls in hls_colors]\n");
+
+    for (size_t i = 0; i < series.size(); i++) {
+      // List x coordinates
+      printf("x%zu = [", i);
+      if (series[i].points.size() > 0)
+        printf("%G", series[i].points[0].x);
+      for (size_t j = 1; j < series[i].points.size(); j++)
+        printf(", %G", series[i].points[j].x);
+      printf("]\n");
+
+      // List y coordinates
+      printf("y%zu = [", i);
+      if (series[i].points.size() > 0)
+        printf("%G", series[i].points[0].y);
+      for (size_t j = 1; j < series[i].points.size(); j++)
+        printf(", %G", series[i].points[j].y);
+      printf("]\n");
+
+      if (series[i].style == BAR_GRAPH) {
+        // There is a plt.bar function that draws bar plots,
+        // but it is *way* too slow to be useful.
+        printf(
+            "plt.vlines(x%zu, map(lambda t: min(t,0), y%zu), map(lambda t: "
+            "max(t,0), y%zu), color=rgb_colors[%zu], "
+            "label=\'%s\')\n",
+            i, i, i, i, series[i].label.c_str());
+      } else if (series[i].style == LINE_GRAPH) {
+        printf("plt.plot(x%zu, y%zu, color=rgb_colors[%zu], label=\'%s\')\n", i,
+               i, i, series[i].label.c_str());
+      } else {
+        printf("raise Exception(\"Unknown graph type\")\n");
+      }
+    }
+  }
+
+  printf("plt.xlim(%f, %f)\n", xaxis_min, xaxis_max);
+  printf("plt.ylim(%f, %f)\n", yaxis_min, yaxis_max);
+  printf("plt.xlabel(\'%s\')\n", xaxis_label.c_str());
+  printf("plt.ylabel(\'%s\')\n", yaxis_label.c_str());
+  printf("plt.title(\'%s\')\n", title.c_str());
+  if (!series.empty()) {
+    printf("plt.legend(loc=\'best\', fontsize=\'small\')\n");
+  }
+}
+
+PythonPlotCollection::PythonPlotCollection() {}
+
+PythonPlotCollection::~PythonPlotCollection() {}
+
+void PythonPlotCollection::draw() {
+  printf("import matplotlib.pyplot as plt\n");
+  printf("import colorsys\n");
+  for (size_t i = 0; i < plots.size(); i++) {
+    printf("plt.figure(%zu)\n", i);
+    plots[i]->draw();
+  }
+  printf("plt.show()\n");
+}
+
+Plot* PythonPlotCollection::append_new_plot() {
+  Plot* plot = new PythonPlot();
+  plots.push_back(std::unique_ptr<Plot>(plot));
+  return plot;
+}
+
+}  // namespace plotting
+}  // namespace webrtc
diff --git a/webrtc/tools/event_log_visualizer/plot_python.h b/webrtc/tools/event_log_visualizer/plot_python.h
new file mode 100644
index 0000000..8d63f95
--- /dev/null
+++ b/webrtc/tools/event_log_visualizer/plot_python.h
@@ -0,0 +1,36 @@
+/*
+ *  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.
+ */
+#ifndef WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_PYTHON_H_
+#define WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_PYTHON_H_
+
+#include "webrtc/tools/event_log_visualizer/plot_base.h"
+
+namespace webrtc {
+namespace plotting {
+
+class PythonPlot final : public Plot {
+ public:
+  PythonPlot();
+  ~PythonPlot() override;
+  void draw() override;
+};
+
+class PythonPlotCollection final : public PlotCollection {
+ public:
+  PythonPlotCollection();
+  ~PythonPlotCollection() override;
+  void draw() override;
+  Plot* append_new_plot() override;
+};
+
+}  // namespace plotting
+}  // namespace webrtc
+
+#endif  // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_PYTHON_H_
diff --git a/webrtc/tools/tools.gyp b/webrtc/tools/tools.gyp
index 998319e..fda6b05 100644
--- a/webrtc/tools/tools.gyp
+++ b/webrtc/tools/tools.gyp
@@ -99,6 +99,29 @@
     }, # force_mic_volume_max
   ],
   'conditions': [
+    ['enable_protobuf==1', {
+      'targets': [
+        {
+          # This target should only be built if enable_protobuf is set
+          'target_name': 'event_log_visualizer',
+          'type': 'executable',
+          'dependencies': [
+            '<(webrtc_root)/webrtc.gyp:rtc_event_log_parser',
+            '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+          ],
+          'sources': [
+            'event_log_visualizer/analyzer.cc',
+            'event_log_visualizer/analyzer.h',
+            'event_log_visualizer/generate_timeseries.cc',
+            'event_log_visualizer/plot_base.h',
+            'event_log_visualizer/plot_python.cc',
+            'event_log_visualizer/plot_python.h',
+          ],
+        },
+      ],
+    }],
     ['include_tests==1', {
       'targets' : [
         {