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webrtc/modules/audio_coding
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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