Add PacketRouterTest.Sanity_NoModuleRegistered_*

webrtc/modules/pacing/paced_sender.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 451 matching lines @@
   const bool success = packet_sender_->TimeToSendPacket(
       packet.ssrc, packet.sequence_number, packet.capture_time_ms,
       packet.retransmission, pacing_info);
   critsect_.Enter();
 
   if (success) {
     // TODO(holmer): High priority packets should only be accounted for if we
     // are allocating bandwidth for audio.
     if (packet.priority != kHighPriority) {
       // Update media bytes sent.
+      // TODO(eladalon): TimeToSendPacket() can also return |true| in some
+      // situations where nothing actually ended up being sent to the network,
+      // and we probably don't want to update the budget in such cases.
+      // https://bugs.chromium.org/p/webrtc/issues/detail?id=8052
       UpdateBudgetWithBytesSent(packet.bytes);
     }
   }
 
   return success;
 }
 
 size_t PacedSender::SendPadding(size_t padding_needed,
                                 const PacedPacketInfo& pacing_info) {
   critsect_.Leave();
@@ skipping 21 matching lines @@
   rtc::CritScope cs(&critsect_);
   pacing_factor_ = pacing_factor;
 }
 
 void PacedSender::SetQueueTimeLimit(int limit_ms) {
   rtc::CritScope cs(&critsect_);
   queue_time_limit = limit_ms;
 }
 
 }  // namespace webrtc