Jakob Ivarsson e9a2ee2631 Improve NetEq network adaptation in the beginning of the call. ... Change the way the forget factor converge to the steady state so that we don't overemphasize the first packets received. The logic is controlled by the delay histogram field trial which has an added parameter to control if emphasis should be even (c=1, default) or put on later packets (c>1) until we reach our steady state forget factor. Bug: webrtc:10411 Change-Id: Ia5d46c22d1a4a66994652f71c8cde664362bfacb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137050 Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28039}		2019-05-23 14:19:30 +00:00
..
acm2	Audio coding: Don't choke when RTP timestamp rate > sample rate	2019-05-21 03:10:49 +00:00
audio_network_adaptor	Stop DCHECK which occurs in ANA BitrateController when overhead is zero.	2019-04-27 00:20:37 +00:00
codecs	WebRTC Opus C interface: Add support for non-48 kHz encode sample rate	2019-05-22 22:56:58 +00:00
include	Expose new audio stats on the API	2019-05-03 10:10:15 +00:00
neteq	Improve NetEq network adaptation in the beginning of the call.	2019-05-23 14:19:30 +00:00
test	WebRTC Opus C interface: Add support for non-48 kHz encode sample rate	2019-05-22 22:56:58 +00:00
audio_coding.gni	Don't select audio codecs depending on GN vars build_with_{chromium|mozilla}	2017-11-01 18:59:27 +00:00
BUILD.gn	Encoder side of Multistream Opus.	2019-04-25 15:07:38 +00:00
DEPS	Fixing WebRTC after moving from src/webrtc to src/	2017-09-15 05:02:56 +00:00
OWNERS	Make ivoc owner of audio_coding.	2018-10-15 15:08:28 +00:00