webrtc/modules/audio_coding
Ivo Creusen ed04912ccd Stop simulations when a LOG_END event is reached.
When a LOG_END event is reached, it makes no sense to continue simulating NetEq.

Bug: webrtc:9667
Change-Id: Ie4f6811cdec0d0632f6e7906059e0e74e9f10438
Reviewed-on: https://webrtc-review.googlesource.com/c/105643
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25176}
2018-10-15 16:06:40 +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 Stop simulations when a LOG_END event is reached. 2018-10-15 16:06:40 +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 Extract functionality of test_main into separate library. 2018-10-15 14:13:06 +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