Add functionality which limits the number of bytes on the network.

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 "webrtc/modules/congestion_controller/delay_based_bwe.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/logging.h"
 #include "webrtc/rtc_base/mod_ops.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
 const int64_t kNoTimestamp = -1;
@@ skipping 73 matching lines @@
 void TransportFeedbackAdapter::SetNetworkIds(uint16_t local_id,
                                              uint16_t remote_id) {
   rtc::CritScope cs(&lock_);
   local_net_id_ = local_id;
   remote_net_id_ = remote_id;
 }
 
 std::vector<PacketFeedback> TransportFeedbackAdapter::GetPacketFeedbackVector(
     const rtcp::TransportFeedback& feedback) {
   int64_t timestamp_us = feedback.GetBaseTimeUs();
+  int64_t now_ms = clock_->TimeInMilliseconds();
   // Add timestamp deltas to a local time base selected on first packet arrival.
   // This won't be the true time base, but makes it easier to manually inspect
   // time stamps.
   if (last_timestamp_us_ == kNoTimestamp) {
-    current_offset_ms_ = clock_->TimeInMilliseconds();
+    current_offset_ms_ = now_ms;
   } else {
     int64_t delta = timestamp_us - last_timestamp_us_;
 
     // Detect and compensate for wrap-arounds in base time.
     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;
 
   std::vector<PacketFeedback> packet_feedback_vector;
   if (feedback.GetPacketStatusCount() == 0) {
     LOG(LS_INFO) << "Empty transport feedback packet received.";
     return packet_feedback_vector;
   }
   packet_feedback_vector.reserve(feedback.GetPacketStatusCount());
-
+  int64_t feedback_rtt = -1;
   {
     rtc::CritScope cs(&lock_);
     size_t failed_lookups = 0;
     int64_t offset_us = 0;
     int64_t timestamp_ms = 0;
     uint16_t seq_num = feedback.GetBaseSequence();
     for (const auto& packet : feedback.GetReceivedPackets()) {
       // Insert into the vector those unreceived packets which precede this
       // iteration's received packet.
       for (; seq_num != packet.sequence_number(); ++seq_num) {
         PacketFeedback packet_feedback(PacketFeedback::kNotReceived, seq_num);
         // Note: Element not removed from history because it might be reported
         // as received by another feedback.
         if (!send_time_history_.GetFeedback(&packet_feedback, false))
           ++failed_lookups;
         if (packet_feedback.local_net_id == local_net_id_ &&
             packet_feedback.remote_net_id == remote_net_id_) {
           packet_feedback_vector.push_back(packet_feedback);
         }
       }
 
       // Handle this iteration's received packet.
       offset_us += packet.delta_us();
       timestamp_ms = current_offset_ms_ + (offset_us / 1000);
       PacketFeedback packet_feedback(timestamp_ms, packet.sequence_number());
       if (!send_time_history_.GetFeedback(&packet_feedback, true))
         ++failed_lookups;
       if (packet_feedback.local_net_id == local_net_id_ &&
           packet_feedback.remote_net_id == remote_net_id_) {
+        if (packet_feedback.send_time_ms >= 0) {
+          int64_t rtt = now_ms - packet_feedback.send_time_ms;
+          if (feedback_rtt == -1) {

[philipel, 2017/08/09 11:58:43]

Nit: remove if/else statement.

[stefan-webrtc, 2017/08/09 12:06:12]

Done.

+            feedback_rtt = rtt;
+          } else {
+            feedback_rtt = std::max(rtt, feedback_rtt);

[philipel, 2017/08/09 11:58:43]

Nit: add a comment that you use max to account for feedback being delayed by the
receiver.

[stefan-webrtc, 2017/08/09 12:06:12]

Done.

+          }
+        }
         packet_feedback_vector.push_back(packet_feedback);
       }
 
       ++seq_num;
     }
 
     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?";
     }
+    if (feedback_rtt > -1) {
+      feedback_rtts_.push_back(feedback_rtt);
+      const size_t kFeedbackRttWindow = 32;
+      if (feedback_rtts_.size() > kFeedbackRttWindow)
+        feedback_rtts_.pop_front();
+      min_feedback_rtt_.emplace(
+          *std::min_element(feedback_rtts_.begin(), feedback_rtts_.end()));
+    }
   }
   return packet_feedback_vector;
 }
 
 void TransportFeedbackAdapter::OnTransportFeedback(
     const rtcp::TransportFeedback& feedback) {
   last_packet_feedback_vector_ = GetPacketFeedbackVector(feedback);
   {
     rtc::CritScope cs(&observers_lock_);
     for (auto observer : observers_) {
       observer->OnPacketFeedbackVector(last_packet_feedback_vector_);
     }
   }
 }
 
 std::vector<PacketFeedback>
 TransportFeedbackAdapter::GetTransportFeedbackVector() const {
   return last_packet_feedback_vector_;
 }
+
+rtc::Optional<int64_t> TransportFeedbackAdapter::GetMinFeedbackLoopRtt() const {
+  rtc::CritScope cs(&lock_);
+  return min_feedback_rtt_;
+}
+
+size_t TransportFeedbackAdapter::GetOutstandingBytes() const {
+  rtc::CritScope cs(&lock_);
+  return send_time_history_.GetOutstandingBytes(local_net_id_, remote_net_id_);
+}
 }  // namespace webrtc