Revert of Move delay_based_bwe_ into CongestionController

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/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/rate_statistics.h"
 #include "webrtc/call/audio_receive_stream.h"
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/call.h"
 #include "webrtc/common_types.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/include/module_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/rtcp_packet/common_header.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.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"
@@ skipping 1027 matching lines @@
   RateStatistics acked_bitrate(250, 8000);
 
   int64_t time_us = std::min(NextRtpTime(), NextRtcpTime());
   int64_t last_update_us = 0;
   while (time_us != std::numeric_limits<int64_t>::max()) {
     clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
     if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
       const LoggedRtcpPacket& rtcp = *rtcp_iterator->second;
       if (rtcp.type == kRtcpTransportFeedback) {
-        cc.OnTransportFeedback(
-            *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get()));
-        std::vector<PacketFeedback> feedback = cc.GetTransportFeedbackVector();
+        TransportFeedbackObserver* observer = cc.GetTransportFeedbackObserver();
+        observer->OnTransportFeedback(*static_cast<rtcp::TransportFeedback*>(
+            rtcp.packet.get()));
+        std::vector<PacketFeedback> feedback =
+            observer->GetTransportFeedbackVector();
         SortPacketFeedbackVector(&feedback);
         rtc::Optional<uint32_t> bitrate_bps;
         if (!feedback.empty()) {
           for (const PacketFeedback& packet : feedback)
             acked_bitrate.Update(packet.payload_size, packet.arrival_time_ms);
           bitrate_bps = acked_bitrate.Rate(feedback.back().arrival_time_ms);
         }
         uint32_t y = 0;
         if (bitrate_bps)
           y = *bitrate_bps / 1000;
         float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
                   1000000;
         acked_time_series.points.emplace_back(x, y);
       }
       ++rtcp_iterator;
     }
     if (clock.TimeInMicroseconds() >= NextRtpTime()) {
       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
       const LoggedRtpPacket& rtp = *rtp_iterator->second;
       if (rtp.header.extension.hasTransportSequenceNumber) {
         RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
-        cc.AddPacket(rtp.header.extension.transportSequenceNumber,
-                     rtp.total_length, PacedPacketInfo());
+        cc.GetTransportFeedbackObserver()->AddPacket(
+            rtp.header.extension.transportSequenceNumber, rtp.total_length,
+            PacedPacketInfo());
         rtc::SentPacket sent_packet(
             rtp.header.extension.transportSequenceNumber, rtp.timestamp / 1000);
         cc.OnSentPacket(sent_packet);
       }
       ++rtp_iterator;
     }
     if (clock.TimeInMicroseconds() >= NextProcessTime()) {
       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
       cc.Process();
     }
     if (observer.GetAndResetBitrateUpdated() ||
         time_us - last_update_us >= 1e6) {
       uint32_t y = observer.last_bitrate_bps() / 1000;
       float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
                 1000000;
       time_series.points.emplace_back(x, y);
       last_update_us = time_us;
     }
     time_us = std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
   }
   // Add the data set to the plot.
   plot->series_list_.push_back(std::move(time_series));
   plot->series_list_.push_back(std::move(acked_time_series));
 
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
   plot->SetTitle("Simulated BWE behavior");
 }
 
+// TODO(holmer): Remove once TransportFeedbackAdapter no longer needs a
+// BitrateController.
+class NullBitrateController : public BitrateController {
+ public:
+  ~NullBitrateController() override {}
+  RtcpBandwidthObserver* CreateRtcpBandwidthObserver() override {
+    return nullptr;
+  }
+  void SetStartBitrate(int start_bitrate_bps) override {}
+  void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) override {}
+  void SetBitrates(int start_bitrate_bps,
+                   int min_bitrate_bps,
+                   int max_bitrate_bps) override {}
+  void ResetBitrates(int bitrate_bps,
+                     int min_bitrate_bps,
+                     int max_bitrate_bps) override {}
+  void OnDelayBasedBweResult(const DelayBasedBwe::Result& result) override {}
+  bool AvailableBandwidth(uint32_t* bandwidth) const override { return false; }
+  void SetReservedBitrate(uint32_t reserved_bitrate_bps) override {}
+  bool GetNetworkParameters(uint32_t* bitrate,
+                            uint8_t* fraction_loss,
+                            int64_t* rtt) override {
+    return false;
+  }
+  int64_t TimeUntilNextProcess() override { return 0; }
+  void Process() override {}
+};
+
 void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(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);
-  TransportFeedbackAdapter feedback_adapter(&clock);
+  NullBitrateController null_controller;
+  TransportFeedbackAdapter feedback_adapter(nullptr, &clock, &null_controller);
+  feedback_adapter.InitBwe();
 
   TimeSeries time_series;
   time_series.label = "Network Delay Change";
   time_series.style = LINE_DOT_GRAPH;
   int64_t estimated_base_delay_ms = std::numeric_limits<int64_t>::max();
 
   auto rtp_iterator = outgoing_rtp.begin();
   auto rtcp_iterator = incoming_rtcp.begin();
 
   auto NextRtpTime = [&]() {
@@ skipping 209 matching lines @@
         return rtc::Optional<float>();
       });
   plot->series_list_.back().label = "Audio encoder number of channels";
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))",
                           kBottomMargin, kTopMargin);
   plot->SetTitle("Reported audio encoder number of channels");
 }
 }  // namespace plotting
 }  // namespace webrtc