Mirko Bonadei 9621377730 Remove WebRTC-Audio-NewOpusPacketLossRateOptimization. ... This field trial is unused. Bug: webrtc:11503 Change-Id: I34262ea4ab169479ceded820c1aa309981731f1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173338 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31059}		2020-04-14 10:02:52 +00:00
..
acm2	ACM: Corrected temporary buffer size	2020-03-12 12:23:20 +00:00
audio_network_adaptor	Use newer version of TimeDelta and TimeStamp factories in modules/	2020-02-10 11:49:57 +00:00
codecs	Remove WebRTC-Audio-NewOpusPacketLossRateOptimization.	2020-04-14 10:02:52 +00:00
include	ACM: Corrected temporary buffer size	2020-03-12 12:23:20 +00:00
neteq	Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API	2020-03-11 12:08:32 +00:00
test	ACM: Corrected temporary buffer size	2020-03-12 12:23:20 +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	Implement AudioEncoder::GetFrameLengthRange() for all audio encoders.	2020-03-25 22:19:21 +00:00
DEPS	Fixing WebRTC after moving from src/webrtc to src/	2017-09-15 05:02:56 +00:00
OWNERS	Remove wildcard ownership for build files.	2020-02-19 14:05:46 +00:00