webrtc/modules/audio_coding
Ivo Creusen 385b10bbaa Added experiment to improve handling of frame length changes in NetEq.
The field trial effects two things: after a frame length change the IAT
histogram is scaled to prevent an immediate change in target buffer
level. Also, the peak history in the delay peak detector is cleared, 
because the size of the peaks is stored in number of packets (which
will be incorrect after a frame length change).

Bug: webrtc:8381
Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8
Reviewed-on: https://webrtc-review.googlesource.com/8101
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20284}
2017-10-13 13:26:57 +00:00
..
acm2 Add explicit includes of refcountedobject.h where it is used. 2017-10-06 13:00:14 +00:00
audio_network_adaptor Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead 2017-10-03 15:26:56 +00:00
codecs Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
include Remove AudioCodingModule::IncomingPayload 2017-09-29 14:23:27 +00:00
neteq Added experiment to improve handling of frame length changes in NetEq. 2017-10-13 13:26:57 +00:00
test Reland "Reland "Remove WEBRTC_TRACE."" 2017-10-04 14:40:44 +00:00
audio_coding.gni Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
BUILD.gn Added experiment to improve handling of frame length changes in NetEq. 2017-10-13 13:26:57 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00