Move congestion controller processing to the pacer thread.

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 432 matching lines @@
     // do, timestamps get messed up.
     if (packet_counter_ > 0) {
       int padding_needed =
           static_cast<int>(is_probing ? (recommended_probe_size - bytes_sent)
                                       : padding_budget_->bytes_remaining());
 
       if (padding_needed > 0)
         bytes_sent += SendPadding(padding_needed, probe_cluster_id);
     }
   }
-  if (is_probing && bytes_sent > 0)
-    prober_->ProbeSent(clock_->TimeInMilliseconds(), bytes_sent);
-  alr_detector_->OnBytesSent(bytes_sent, now_us / 1000);
+  if (bytes_sent > 0) {
+    if (is_probing)
+      prober_->ProbeSent(clock_->TimeInMilliseconds(), bytes_sent);
+    alr_detector_->OnBytesSent(bytes_sent, now_us / 1000);
+  }
 }
 
 bool PacedSender::SendPacket(const paced_sender::Packet& packet,
                              int probe_cluster_id) {
   // TODO(holmer): Because of this bug issue 5307 we have to send audio
   // packets even when the pacer is paused. Here we assume audio packets are
   // always high priority and that they are the only high priority packets.
   if (packet.priority != kHighPriority) {
     if (paused_)
       return false;
@@ skipping 35 matching lines @@
 void PacedSender::UpdateBudgetWithElapsedTime(int64_t delta_time_ms) {
   media_budget_->IncreaseBudget(delta_time_ms);
   padding_budget_->IncreaseBudget(delta_time_ms);
 }
 
 void PacedSender::UpdateBudgetWithBytesSent(size_t bytes_sent) {
   media_budget_->UseBudget(bytes_sent);
   padding_budget_->UseBudget(bytes_sent);
 }
 }  // namespace webrtc