Karl Wiberg 053c371552 Audio coding: Don't choke when RTP timestamp rate > sample rate ... Bug: webrtc:10631 Change-Id: If0422786172502f039acc2cac5e8c13b637af54c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137048 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27998}		2019-05-21 03:10:49 +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	AudioEncoderOpus: Don't mix up sample rate and RTP timestamp rate	2019-05-20 17:33:56 +00:00
include	Expose new audio stats on the API	2019-05-03 10:10:15 +00:00
neteq	Add DecelerationTargetLevelOffset Field Trial.	2019-05-17 08:08:12 +00:00
test	Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData	2019-04-26 12:58:14 +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