Prioritize video packets when sending padding or preemptive retransmits.

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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/video/video_send_stream.h"
@@ skipping 779 matching lines @@
 
   RTC_DCHECK(!config_->rtp.ssrcs.empty());
   RTC_DCHECK(call_stats_);
   RTC_DCHECK(congestion_controller_);
   RTC_DCHECK(remb_);
 
   congestion_controller_->EnablePeriodicAlrProbing(
       config_->periodic_alr_bandwidth_probing);
 
   // RTP/RTCP initialization.
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+
+  // We add the highest spatial layer first to ensure it'll be prioritized
+  // when sending padding, with the hope that the packet rate will be smaller,
+  // and that it's more important to protect than the lower layers.
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
     packet_router_->AddRtpModule(rtp_rtcp);
-  }
 
   for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) {
     const std::string& extension = config_->rtp.extensions[i].uri;
     int id = config_->rtp.extensions[i].id;
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     RTC_DCHECK_GE(id, 1);
     RTC_DCHECK_LE(id, 14);
     RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
     for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
       RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension(
@@ skipping 495 matching lines @@
       std::min(config_->rtp.max_packet_size,
                kPathMTU - transport_overhead_bytes_per_packet_);
 
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc