webrtc/modules/audio_coding
Henrik Lundin f1061c2d90 rtp_encode: Unify the encoder configs somewhat
For uniformity. Uniformity is nice.

Bug: webrtc:2692
Change-Id: Id85e54fa31bf3cc79e73a72805e57d5e3164252f
Reviewed-on: https://webrtc-review.googlesource.com/27400
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21135}
2017-12-07 09:43:17 +00:00
..
acm2 Add new UMA metric for NetEq target buffer delay 2017-11-29 08:56:29 +00:00
audio_network_adaptor Stop using ByteSize (deprecated) to get the size of a proto message. 2017-11-30 14:27:50 +00:00
codecs Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
include Add new UMA metric for NetEq target buffer delay 2017-11-29 08:56:29 +00:00
neteq rtp_encode: Unify the encoder configs somewhat 2017-12-07 09:43:17 +00:00
test Avoid flagging Opus DTX frames as speech. 2017-11-20 14:53:40 +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 Stop using public_deps in system_wrappers. 2017-12-06 08:56:52 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00