tree bb6b440936696d2e0ed528fb37118209427410af
parent c979c46bdaf206805c62cb687be450c93dbfb533
author Danil Chapovalov <danilchap@webrtc.org> 1543943032 +0100
committer Commit Bot <commit-bot@chromium.org> 1544021543 +0000

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}