webrtc/modules/audio_coding
Ivo Creusen d2d2ecb4a8 Add command-line flag for setting the max number of packets in the buffer.
There is currently no way to set this for simulations in neteq_rtpplay.

Bug: webrtc:9667
Change-Id: I34f34565538bd3c378cdb9d355f5173c3517d59a
Reviewed-on: https://webrtc-review.googlesource.com/c/105982
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25171}
2018-10-15 14:10:24 +00:00
..
acm2 Delete unused includes of assert.h 2018-10-04 14:01:44 +00:00
audio_network_adaptor Fixing py lint errors 2018-07-23 15:28:48 +00:00
codecs Add field trials for configuring Opus encoder packet loss rate. 2018-10-15 08:59:43 +00:00
include Delete unused method AudioCodingModuleImpl::SetOpusApplication. 2018-10-04 13:46:31 +00:00
neteq Add command-line flag for setting the max number of packets in the buffer. 2018-10-15 14:10:24 +00:00
test Delete unused includes of assert.h 2018-10-04 14:01:44 +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 RtcEventLogSource no longer uses deprecated parsing functions. 2018-10-11 16:13:17 +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