Jakob Ivarsson 79890ef91f Remove sync buffer length from FilteredCurrentDelayMs. ... The sync buffer length is already added when the buffer level filter is updated. Bug: webrtc:10736 Change-Id: Icbd411d4fd7b16f31b800142d1b6a8de79365d91 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140950 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28261}		2019-06-13 09:38:22 +00:00
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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	AudioDecoderOpus: Add support for 16 kHz output sample rate	2019-05-29 12:42:38 +00:00
include	Expose new audio stats on the API	2019-05-03 10:10:15 +00:00
neteq	Remove sync buffer length from FilteredCurrentDelayMs.	2019-06-13 09:38:22 +00:00
test	WebRTC Opus C interface: Add support for non-48 kHz decode sample rate	2019-05-29 10:33:03 +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