Prioritize video packets when sending padding or preemptive retransmits.

webrtc/modules/pacing/packet_router.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.
  */
 
@@ skipping 12 matching lines @@
 }
 
 PacketRouter::~PacketRouter() {
   RTC_DCHECK(rtp_modules_.empty());
 }
 
 void PacketRouter::AddRtpModule(RtpRtcp* rtp_module) {
   rtc::CritScope cs(&modules_crit_);
   RTC_DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) ==
              rtp_modules_.end());
-  rtp_modules_.push_back(rtp_module);
+  // Put modules which can use regular payload packets (over rtx) instead of
+  // padding first as it's less of a waste
+  if ((rtp_module->RtxSendStatus() & kRtxRedundantPayloads) > 0) {
+    rtp_modules_.push_front(rtp_module);
+  } else {
+    rtp_modules_.push_back(rtp_module);
+  }
 }
 
 void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module) {
   rtc::CritScope cs(&modules_crit_);
   RTC_DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) !=
              rtp_modules_.end());
   rtp_modules_.remove(rtp_module);
 }
 
 bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
@@ skipping 13 matching lines @@
     }
   }
   return true;
 }
 
 size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send,
                                        int probe_cluster_id) {
   RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
   size_t total_bytes_sent = 0;
   rtc::CritScope cs(&modules_crit_);
+  // Rtp modules are ordered by which stream can most benefit from padding.
   for (RtpRtcp* module : rtp_modules_) {
     if (module->SendingMedia()) {
       size_t bytes_sent = module->TimeToSendPadding(
           bytes_to_send - total_bytes_sent, probe_cluster_id);
       total_bytes_sent += bytes_sent;
       if (total_bytes_sent >= bytes_to_send)
         break;
     }
   }
   return total_bytes_sent;
@@ skipping 25 matching lines @@
   rtc::CritScope cs(&modules_crit_);
   for (auto* rtp_module : rtp_modules_) {
     packet->SetSenderSsrc(rtp_module->SSRC());
     if (rtp_module->SendFeedbackPacket(*packet))
       return true;
   }
   return false;
 }
 
 }  // namespace webrtc