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

webrtc/modules/congestion_controller/transport_feedback_adapter.cc

Index: webrtc/modules/congestion_controller/transport_feedback_adapter.cc
diff --git a/webrtc/modules/congestion_controller/transport_feedback_adapter.cc b/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
index 7f3f744ecf70a811610fc7fd05a6d87811b07ffb..918b9c5b0e11aa914a345f0af94270d1ef38b8ab 100644
--- a/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
+++ b/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
@@ -10,6 +10,8 @@
 
 #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"
@@ -103,11 +105,12 @@ void TransportFeedbackAdapter::SetNetworkIds(uint16_t local_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_;
 
@@ -128,7 +131,7 @@ std::vector<PacketFeedback> TransportFeedbackAdapter::GetPacketFeedbackVector(
     return packet_feedback_vector;
   }
   packet_feedback_vector.reserve(feedback.GetPacketStatusCount());
-
+  int64_t feedback_rtt = -1;
   {
     rtc::CritScope cs(&lock_);
     size_t failed_lookups = 0;
@@ -158,6 +161,12 @@ std::vector<PacketFeedback> TransportFeedbackAdapter::GetPacketFeedbackVector(
         ++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;
+          // max() is used to account for feedback being delayed by the
+          // receiver.
+          feedback_rtt = std::max(rtt, feedback_rtt);
+        }
         packet_feedback_vector.push_back(packet_feedback);
       }
 
@@ -169,6 +178,14 @@ std::vector<PacketFeedback> TransportFeedbackAdapter::GetPacketFeedbackVector(
                       << " 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;
 }
@@ -188,4 +205,14 @@ 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