Reduce the BWE with 50% when feedback is received too late.

webrtc/modules/congestion_controller/transport_feedback_adapter.cc

 /*
  *  Copyright (c) 2015 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/modules/congestion_controller/transport_feedback_adapter.h"
 
 #include <algorithm>
 #include <limits>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/delay_based_bwe.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
 const int64_t kNoTimestamp = -1;
-const int64_t kSendTimeHistoryWindowMs = 10000;
+const int64_t kSendTimeHistoryWindowMs = 60000;
 const int64_t kBaseTimestampScaleFactor =
     rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8);
 const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);
 
 class PacketInfoComparator {
  public:
   inline bool operator()(const PacketInfo& lhs, const PacketInfo& rhs) {
     if (lhs.arrival_time_ms != rhs.arrival_time_ms)
       return lhs.arrival_time_ms < rhs.arrival_time_ms;
     if (lhs.send_time_ms != rhs.send_time_ms)
       return lhs.send_time_ms < rhs.send_time_ms;
     return lhs.sequence_number < rhs.sequence_number;
   }
 };
 
 TransportFeedbackAdapter::TransportFeedbackAdapter(
     Clock* clock,
     BitrateController* bitrate_controller)
     : send_side_bwe_with_overhead_(webrtc::field_trial::FindFullName(
-          "WebRTC-SendSideBwe-WithOverhead") == "Enabled"),
+                                       "WebRTC-SendSideBwe-WithOverhead") ==
+                                   "Enabled"),
       transport_overhead_bytes_per_packet_(0),
       send_time_history_(clock, kSendTimeHistoryWindowMs),
       clock_(clock),
       current_offset_ms_(kNoTimestamp),
       last_timestamp_us_(kNoTimestamp),
       bitrate_controller_(bitrate_controller) {}
 
 TransportFeedbackAdapter::~TransportFeedbackAdapter() {}
 
 void TransportFeedbackAdapter::InitBwe() {
@@ skipping 53 matching lines @@
     if (std::abs(delta - kBaseTimestampRangeSizeUs) < std::abs(delta)) {
       delta -= kBaseTimestampRangeSizeUs;  // Wrap backwards.
     } else if (std::abs(delta + kBaseTimestampRangeSizeUs) < std::abs(delta)) {
       delta += kBaseTimestampRangeSizeUs;  // Wrap forwards.
     }
 
     current_offset_ms_ += delta / 1000;
   }
   last_timestamp_us_ = timestamp_us;
 
+  auto received_packets = feedback.GetReceivedPackets();
   std::vector<PacketInfo> packet_feedback_vector;
-  packet_feedback_vector.reserve(feedback.GetReceivedPackets().size());
+  packet_feedback_vector.reserve(received_packets.size());
+  if (received_packets.empty()) {
+    LOG(LS_INFO) << "Empty transport feedback packet received.";
+    return packet_feedback_vector;
+  }
   {
     rtc::CritScope cs(&lock_);
     size_t failed_lookups = 0;
     int64_t offset_us = 0;
+    int64_t timestamp_ms = 0;
     for (const auto& packet : feedback.GetReceivedPackets()) {
       offset_us += packet.delta_us();
-      int64_t timestamp_ms = current_offset_ms_ + (offset_us / 1000);
+      timestamp_ms = current_offset_ms_ + (offset_us / 1000);
       PacketInfo info(timestamp_ms, packet.sequence_number());
-      if (send_time_history_.GetInfo(&info, true) && info.send_time_ms >= 0) {
-        packet_feedback_vector.push_back(info);
-      } else {
+      if (!send_time_history_.GetInfo(&info, true))

[stefan-webrtc, 2017/02/10 14:15:11]

With this change we will add also acks which have timed out at the sender, but
they will have send_time_ms set to -1, etc.

         ++failed_lookups;
-      }
+      packet_feedback_vector.push_back(info);
     }
     std::sort(packet_feedback_vector.begin(), packet_feedback_vector.end(),
               PacketInfoComparator());
     if (failed_lookups > 0) {
       LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups
                       << " packet" << (failed_lookups > 1 ? "s" : "")
                       << ". Send time history too small?";
     }
   }
   return packet_feedback_vector;
 }
 
 void TransportFeedbackAdapter::OnTransportFeedback(
     const rtcp::TransportFeedback& feedback) {
   last_packet_feedback_vector_ = GetPacketFeedbackVector(feedback);
   DelayBasedBwe::Result result;
   {
     rtc::CritScope cs(&bwe_lock_);
     result = delay_based_bwe_->IncomingPacketFeedbackVector(
         last_packet_feedback_vector_);
   }
-  if (result.updated)
+  if (result.updated) {
     bitrate_controller_->OnDelayBasedBweResult(result);
+  }
 }
 
 std::vector<PacketInfo> TransportFeedbackAdapter::GetTransportFeedbackVector()
     const {
   return last_packet_feedback_vector_;
 }
 
 void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms,
                                            int64_t max_rtt_ms) {
   rtc::CritScope cs(&bwe_lock_);
   delay_based_bwe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
 }
 
 }  // namespace webrtc