Karl Wiberg 126f2b37ac AudioEncoderOpus: Add support for 16 kHz input sample rate ... In addition to the 48 kHz that we've always used. Bug: webrtc:10631 Change-Id: I5e4f6600e39a463d20d3988db098c7e38281f4a0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138264 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28074}		2019-05-27 13:01:04 +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: Add support for 16 kHz input sample rate	2019-05-27 13:01:04 +00:00
include	Expose new audio stats on the API	2019-05-03 10:10:15 +00:00
neteq	Improve NetEq network adaptation in the beginning of the call.	2019-05-23 14:19:30 +00:00
test	WebRTC Opus C interface: Add support for non-48 kHz encode sample rate	2019-05-22 22:56:58 +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