Move PacedSender ownership to RtpTransportControllerSend.

webrtc/rtc_tools/event_log_visualizer/analyzer.cc

Index: webrtc/rtc_tools/event_log_visualizer/analyzer.cc
diff --git a/webrtc/rtc_tools/event_log_visualizer/analyzer.cc b/webrtc/rtc_tools/event_log_visualizer/analyzer.cc
index 593a3b8bf72def0728a32feb579d62ab953cd50e..b02b3f128bf132e8cdc642f0d8b042bc56121af4 100644
--- a/webrtc/rtc_tools/event_log_visualizer/analyzer.cc
+++ b/webrtc/rtc_tools/event_log_visualizer/analyzer.cc
@@ -29,6 +29,7 @@
 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/pacing/paced_sender.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/rtcp_packet/common_header.h"
@@ -1130,8 +1131,8 @@ void EventLogAnalyzer::CreateBweSimulationGraph(Plot* plot) {
   BitrateObserver observer;
   RtcEventLogNullImpl null_event_log;
   PacketRouter packet_router;
-  CongestionController cc(&clock, &observer, &observer, &null_event_log,
-                          &packet_router);
+  PacedSender pacer(&clock, &packet_router, &null_event_log);
+  SendSideCongestionController cc(&clock, &observer, &null_event_log, &pacer);
   // TODO(holmer): Log the call config and use that here instead.
   static const uint32_t kDefaultStartBitrateBps = 300000;
   cc.SetBweBitrates(0, kDefaultStartBitrateBps, -1);