Revert of Add BWE plot to event log analyzer.

webrtc/tools/event_log_visualizer/analyzer.cc

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
index c15de6d7fbd9d59744afc3c57881185b263814d5..e6dd35b6c6fa10e1d62f23cda08b2b36bf9ea792 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.cc
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc
@@ -22,12 +22,9 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/call.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
@@ -95,15 +92,21 @@
     return true;
   }
   if (ssrc_ == other.ssrc_) {
-    if (direction_ < other.direction_) {
+    if (media_type_ < other.media_type_) {
       return true;
     }
+    if (media_type_ == other.media_type_) {
+      if (direction_ < other.direction_) {
+        return true;
+      }
+    }
   }
   return false;
 }
 
 bool EventLogAnalyzer::StreamId::operator==(const StreamId& other) const {
-  return ssrc_ == other.ssrc_ && direction_ == other.direction_;
+  return ssrc_ == other.ssrc_ && direction_ == other.direction_ &&
+         media_type_ == other.media_type_;
 }
 
 
@@ -112,11 +115,12 @@
   uint64_t first_timestamp = std::numeric_limits<uint64_t>::max();
   uint64_t last_timestamp = std::numeric_limits<uint64_t>::min();
 
-  // Maps a stream identifier consisting of ssrc and direction
+  // Maps a stream identifier consisting of ssrc, direction and MediaType
   // to the header extensions used by that stream,
   std::map<StreamId, RtpHeaderExtensionMap> extension_maps;
 
   PacketDirection direction;
+  MediaType media_type;
   uint8_t header[IP_PACKET_SIZE];
   size_t header_length;
   size_t total_length;
@@ -136,7 +140,8 @@
       case ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: {
         VideoReceiveStream::Config config(nullptr);
         parsed_log_.GetVideoReceiveConfig(i, &config);
-        StreamId stream(config.rtp.remote_ssrc, kIncomingPacket);
+        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;
@@ -150,7 +155,7 @@
         VideoSendStream::Config config(nullptr);
         parsed_log_.GetVideoSendConfig(i, &config);
         for (auto ssrc : config.rtp.ssrcs) {
-          StreamId stream(ssrc, kOutgoingPacket);
+          StreamId stream(ssrc, kOutgoingPacket, 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;
@@ -172,14 +177,13 @@
         break;
       }
       case ParsedRtcEventLog::RTP_EVENT: {
-        MediaType media_type;
         parsed_log_.GetRtpHeader(i, &direction, &media_type, header,
                                  &header_length, &total_length);
         // Parse header to get SSRC.
         RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
         RTPHeader parsed_header;
         rtp_parser.Parse(&parsed_header);
-        StreamId stream(parsed_header.ssrc, direction);
+        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];
@@ -187,45 +191,10 @@
         }
         uint64_t timestamp = parsed_log_.GetTimestamp(i);
         rtp_packets_[stream].push_back(
-            LoggedRtpPacket(timestamp, parsed_header, total_length));
+            LoggedRtpPacket(timestamp, parsed_header));
         break;
       }
       case ParsedRtcEventLog::RTCP_EVENT: {
-        uint8_t packet[IP_PACKET_SIZE];
-        MediaType media_type;
-        parsed_log_.GetRtcpPacket(i, &direction, &media_type, packet,
-                                  &total_length);
-
-        RtpUtility::RtpHeaderParser rtp_parser(packet, total_length);
-        RTPHeader parsed_header;
-        RTC_CHECK(rtp_parser.ParseRtcp(&parsed_header));
-        uint32_t ssrc = parsed_header.ssrc;
-
-        RTCPUtility::RTCPParserV2 rtcp_parser(packet, total_length, true);
-        RTC_CHECK(rtcp_parser.IsValid());
-
-        RTCPUtility::RTCPPacketTypes packet_type = rtcp_parser.Begin();
-        while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
-          switch (packet_type) {
-            case RTCPUtility::RTCPPacketTypes::kTransportFeedback: {
-              // Currently feedback is logged twice, both for audio and video.
-              // Only act on one of them.
-              if (media_type == MediaType::VIDEO) {
-                std::unique_ptr<rtcp::RtcpPacket> rtcp_packet(
-                    rtcp_parser.ReleaseRtcpPacket());
-                StreamId stream(ssrc, direction);
-                uint64_t timestamp = parsed_log_.GetTimestamp(i);
-                rtcp_packets_[stream].push_back(LoggedRtcpPacket(
-                    timestamp, kRtcpTransportFeedback, std::move(rtcp_packet)));
-              }
-              break;
-            }
-            default:
-              break;
-          }
-          rtcp_parser.Iterate();
-          packet_type = rtcp_parser.PacketType();
-        }
         break;
       }
       case ParsedRtcEventLog::LOG_START: {
@@ -262,33 +231,6 @@
   begin_time_ = first_timestamp;
   end_time_ = last_timestamp;
 }
-
-class BitrateObserver : public CongestionController::Observer,
-                        public RemoteBitrateObserver {
- public:
-  BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}
-
-  void OnNetworkChanged(uint32_t bitrate_bps,
-                        uint8_t fraction_loss,
-                        int64_t rtt_ms) override {
-    last_bitrate_bps_ = bitrate_bps;
-    bitrate_updated_ = true;
-  }
-
-  void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
-                               uint32_t bitrate) override {}
-
-  uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
-  bool GetAndResetBitrateUpdated() {
-    bool bitrate_updated = bitrate_updated_;
-    bitrate_updated_ = false;
-    return bitrate_updated;
-  }
-
- private:
-  uint32_t last_bitrate_bps_;
-  bool bitrate_updated_;
-};
 
 void EventLogAnalyzer::CreatePacketGraph(PacketDirection desired_direction,
                                          Plot* plot) {
@@ -733,113 +675,5 @@
   }
 }
 
-void EventLogAnalyzer::CreateBweGraph(Plot* plot) {
-  std::map<uint64_t, const LoggedRtpPacket*> outgoing_rtp;
-  std::map<uint64_t, const LoggedRtcpPacket*> incoming_rtcp;
-
-  for (const auto& kv : rtp_packets_) {
-    if (kv.first.GetDirection() == PacketDirection::kOutgoingPacket) {
-      for (const LoggedRtpPacket& rtp_packet : kv.second)
-        outgoing_rtp.insert(std::make_pair(rtp_packet.timestamp, &rtp_packet));
-    }
-  }
-
-  for (const auto& kv : rtcp_packets_) {
-    if (kv.first.GetDirection() == PacketDirection::kIncomingPacket) {
-      for (const LoggedRtcpPacket& rtcp_packet : kv.second)
-        incoming_rtcp.insert(
-            std::make_pair(rtcp_packet.timestamp, &rtcp_packet));
-    }
-  }
-
-  SimulatedClock clock(0);
-  BitrateObserver observer;
-  RtcEventLogNullImpl null_event_log;
-  CongestionController cc(&clock, &observer, &observer, &null_event_log);
-  // TODO(holmer): Log the call config and use that here instead.
-  static const uint32_t kDefaultStartBitrateBps = 300000;
-  cc.SetBweBitrates(0, kDefaultStartBitrateBps, -1);
-
-  TimeSeries time_series;
-  time_series.label = "BWE";
-  time_series.style = LINE_DOT_GRAPH;
-  uint32_t max_y = 10;
-  uint32_t min_y = 0;
-
-  auto rtp_iterator = outgoing_rtp.begin();
-  auto rtcp_iterator = incoming_rtcp.begin();
-
-  auto NextRtpTime = [&]() {
-    if (rtp_iterator != outgoing_rtp.end())
-      return static_cast<int64_t>(rtp_iterator->first);
-    return std::numeric_limits<int64_t>::max();
-  };
-
-  auto NextRtcpTime = [&]() {
-    if (rtcp_iterator != incoming_rtcp.end())
-      return static_cast<int64_t>(rtcp_iterator->first);
-    return std::numeric_limits<int64_t>::max();
-  };
-
-  auto NextProcessTime = [&]() {
-    if (rtcp_iterator != incoming_rtcp.end() ||
-        rtp_iterator != outgoing_rtp.end()) {
-      return clock.TimeInMicroseconds() +
-             std::max<int64_t>(cc.TimeUntilNextProcess() * 1000, 0);
-    }
-    return std::numeric_limits<int64_t>::max();
-  };
-
-  int64_t time_us = std::min(NextRtpTime(), NextRtcpTime());
-  while (time_us != std::numeric_limits<int64_t>::max()) {
-    clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
-    if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
-      clock.AdvanceTimeMilliseconds(rtcp_iterator->first / 1000 -
-                                    clock.TimeInMilliseconds());
-      const LoggedRtcpPacket& rtcp = *rtcp_iterator->second;
-      if (rtcp.type == kRtcpTransportFeedback) {
-        cc.GetTransportFeedbackObserver()->OnTransportFeedback(
-            *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get()));
-      }
-      ++rtcp_iterator;
-    }
-    if (clock.TimeInMicroseconds() >= NextRtpTime()) {
-      clock.AdvanceTimeMilliseconds(rtp_iterator->first / 1000 -
-                                    clock.TimeInMilliseconds());
-      const LoggedRtpPacket& rtp = *rtp_iterator->second;
-      if (rtp.header.extension.hasTransportSequenceNumber) {
-        RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
-        cc.GetTransportFeedbackObserver()->AddPacket(
-            rtp.header.extension.transportSequenceNumber, rtp.total_length, 0);
-        rtc::SentPacket sent_packet(
-            rtp.header.extension.transportSequenceNumber, rtp.timestamp / 1000);
-        cc.OnSentPacket(sent_packet);
-      }
-      ++rtp_iterator;
-    }
-    if (clock.TimeInMicroseconds() >= NextProcessTime())
-      cc.Process();
-    if (observer.GetAndResetBitrateUpdated()) {
-      uint32_t y = observer.last_bitrate_bps() / 1000;
-      max_y = std::max(max_y, y);
-      min_y = std::min(min_y, y);
-      float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
-                1000000;
-      time_series.points.emplace_back(x, y);
-    }
-    time_us = std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
-  }
-  // Add the data set to the plot.
-  plot->series.push_back(std::move(time_series));
-
-  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 = "Bitrate (kbps)";
-  plot->title = "BWE";
-}
-
 }  // namespace plotting
 }  // namespace webrtc