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webrtc/modules/audio_coding
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Danil Chapovalov a86cef7e2c Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in audio_coding 
Bug: webrtc:12336
Change-Id: Icae229b957c2bfcc410788179a504c576cfde151
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201736
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32995}
2021-01-15 10:58:20 +00:00
..
acm2 Delete legacy stats minWaitingTimeMs and medianWaitingTimeMs from ACM. 2020-11-03 11:06:44 +00:00
audio_network_adaptor Remove check for WebRTC-SendSideBwe-WithOverhead in bitrate controller. 2020-11-09 23:11:26 +00:00
codecs Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in audio_coding 2021-01-15 10:58:20 +00:00
include Delete legacy stats minWaitingTimeMs and medianWaitingTimeMs from ACM. 2020-11-03 11:06:44 +00:00
neteq Correction for the calculation of the abs max value 2021-01-12 16:28:00 +00:00
test Revert "opus: take SILK vad result into account for voice detection" 2020-11-04 07:29:48 +00:00
audio_coding.gni build: remove WEBRTC_CODEC_RED 2020-05-26 11:01:26 +00:00
BUILD.gn Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in audio_coding 2021-01-15 10:58:20 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Remove kwiberg@webrtc.org from OWNERS files 2020-12-04 15:11:26 +00:00