Move PacedSender ownership to RtpTransportControllerSend.

webrtc/rtc_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.
  */
 
@@ skipping 11 matching lines @@
 #include "webrtc/call/call.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
 #include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_test.h"
 #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"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/rtc_base/checks.h"
@@ skipping 1081 matching lines @@
       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;
   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);
 
   TimeSeries time_series("Delay-based estimate", LINE_DOT_GRAPH);
   TimeSeries acked_time_series("Acked bitrate", LINE_DOT_GRAPH);
 
   auto rtp_iterator = outgoing_rtp.begin();
   auto rtcp_iterator = incoming_rtcp.begin();
 
@@ skipping 578 matching lines @@
     plot->AppendTimeSeries(std::move(series.second));
   }
 
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetYAxis(min_y_axis, max_y_axis, "Relative delay (ms)", kBottomMargin,
                  kTopMargin);
   plot->SetTitle("NetEq timing");
 }
 }  // namespace plotting
 }  // namespace webrtc