webrtc/modules/audio_coding
Mirko Bonadei b5728d9b0f Stop using public_deps in modules/rtp_rtcp.
TBR=phoglund@webrtc.org

Bug: webrtc:8603
Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/29780
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:37:52 +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 Fix circular dependency in rtc_event_log. 2017-11-29 10:46:19 +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 modules/rtp_rtcp. 2017-12-06 07:37:52 +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