commit 2a977cf466369a01f2fdc076326886b4c67ba39e
author Danil Chapovalov <danilchap@webrtc.org> Tue Dec 04 17:03:52 2018
committer Commit Bot <commit-bot@chromium.org> Wed Dec 05 14:52:23 2018
tree bb6b440936696d2e0ed528fb37118209427410af
parent c979c46bdaf206805c62cb687be450c93dbfb533

For audio receive channel use default max reordering threshold instead of 0

setting max reordering recently has been fix to actually set it.
(https://webrtc-review.googlesource.com/c/src/+/111752)
Another recent change fix stats to skip counting large sequence number jumps as packet loss
(https://webrtc-review.googlesource.com/c/src/+/111962)

max reordering thresholds affects how packet loss is calculated.
Packet loss is then reported to remote sending participant in rtcp receiver reports.
Sender uses packet loss mostly for stats, but also e.g. for opus fec adjustment.

Setting threshold to zero de-facto imply all packets should be considered in order.
That bug was mitigated by two other bugs mentioned above

This change increase threshold to default 50 packets aligning it with Video receiver
and unblocks (re)landing 2nd fix

Bug: b/120482366
Change-Id: Iadda0c2148ed84dd83c01183cfe9285568db4e29
Reviewed-on: https://webrtc-review.googlesource.com/c/113064
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25905}

audio/channel_receive.cc

diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 483147f..f2d7c9d 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -33,6 +33,7 @@
 #include "modules/rtp_rtcp/source/contributing_sources.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "modules/utility/include/process_thread.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/criticalsection.h"
@@ -802,11 +803,14 @@
 void ChannelReceive::SetNACKStatus(bool enable, int max_packets) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   // None of these functions can fail.
-  rtp_receive_statistics_->SetMaxReorderingThreshold(max_packets);
-  if (enable)
+  if (enable) {
+    rtp_receive_statistics_->SetMaxReorderingThreshold(max_packets);
     audio_coding_->EnableNack(max_packets);
-  else
+  } else {
+    rtp_receive_statistics_->SetMaxReorderingThreshold(
+        kDefaultMaxReorderingThreshold);
     audio_coding_->DisableNack();
+  }
 }
 
 // Called when we are missing one or more packets.