Video/Screenshare loopback tool.

webrtc/test/fake_network_pipe.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 114 matching lines @@
       capacity_link_.size() >= config_.queue_length_packets) {
     // Too many packet on the link, drop this one.
     ++dropped_packets_;
     return;
   }
 
   int64_t time_now = clock_->TimeInMilliseconds();
 
   // Delay introduced by the link capacity.
   int64_t capacity_delay_ms = 0;
-  if (config_.link_capacity_kbps > 0)
-    capacity_delay_ms = data_length / (config_.link_capacity_kbps / 8);
+  if (config_.link_capacity_kbps > 0) {
+    int offset_bytes = config_.link_capacity_kbps / 8 / 2;
+    capacity_delay_ms = (data_length + capacity_delay_error_ + offset_bytes) /
+                        (config_.link_capacity_kbps / 8);
+    capacity_delay_error_ +=
+        data_length - capacity_delay_ms * (config_.link_capacity_kbps / 8);
+    // Print("Error: %ld", capacity_delay_error_);
+  }
   int64_t network_start_time = time_now;
 
   // Check if there already are packets on the link and change network start
   // time forward if there is.
   if (!capacity_link_.empty() &&
       network_start_time < capacity_link_.back()->arrival_time())
     network_start_time = capacity_link_.back()->arrival_time();
 
   int64_t arrival_time = network_start_time + capacity_delay_ms;
   NetworkPacket* packet = new NetworkPacket(data, data_length, time_now,
@@ skipping 91 matching lines @@
                            : time_now + kDefaultProcessIntervalMs;
 }
 
 int64_t FakeNetworkPipe::TimeUntilNextProcess() const {
   rtc::CritScope crit(&lock_);
   return std::max<int64_t>(next_process_time_ - clock_->TimeInMilliseconds(),
                            0);
 }
 
 }  // namespace webrtc