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

webrtc/modules/remote_bitrate_estimator/send_time_history.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/remote_bitrate_estimator/include/send_time_history.h"
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
 SendTimeHistory::SendTimeHistory(const Clock* clock,
                                  int64_t packet_age_limit_ms)
-    : clock_(clock), packet_age_limit_ms_(packet_age_limit_ms) {}
+    : clock_(clock),
+      packet_age_limit_ms_(packet_age_limit_ms),
+      latest_acked_seq_num_(-1) {}
 
 SendTimeHistory::~SendTimeHistory() {}
 
 void SendTimeHistory::AddAndRemoveOld(const PacketFeedback& packet) {
   int64_t now_ms = clock_->TimeInMilliseconds();
   // Remove old.
   while (!history_.empty() &&
          now_ms - history_.begin()->second.creation_time_ms >
              packet_age_limit_ms_) {
     // TODO(sprang): Warn if erasing (too many) old items?
@@ skipping 13 matching lines @@
     return false;
   it->second.send_time_ms = send_time_ms;
   return true;
 }
 
 bool SendTimeHistory::GetFeedback(PacketFeedback* packet_feedback,
                                   bool remove) {
   RTC_DCHECK(packet_feedback);
   int64_t unwrapped_seq_num =
       seq_num_unwrapper_.Unwrap(packet_feedback->sequence_number);
+  latest_acked_seq_num_ = std::max(unwrapped_seq_num, latest_acked_seq_num_);
+  RTC_DCHECK_GE(latest_acked_seq_num_, 0);
   auto it = history_.find(unwrapped_seq_num);
   if (it == history_.end())
     return false;
 
   // Save arrival_time not to overwrite it.
   int64_t arrival_time_ms = packet_feedback->arrival_time_ms;
   *packet_feedback = it->second;
   packet_feedback->arrival_time_ms = arrival_time_ms;
 
   if (remove)
     history_.erase(it);
   return true;
 }
 
+size_t SendTimeHistory::GetOutstandingBytes(uint16_t local_net_id,
+                                            uint16_t remote_net_id) const {
+  size_t outstanding_bytes = 0;
+  auto unacked_it = history_.begin();
+  if (latest_acked_seq_num_ >= 0) {
+    unacked_it = history_.lower_bound(latest_acked_seq_num_);
+  }
+  for (; unacked_it != history_.end(); ++unacked_it) {
+    if (unacked_it->second.local_net_id == local_net_id &&
+        unacked_it->second.remote_net_id == remote_net_id &&
+        unacked_it->second.send_time_ms >= 0) {
+      outstanding_bytes += unacked_it->second.payload_size;
+    }
+  }
+  return outstanding_bytes;
+}
+
 }  // namespace webrtc