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webrtc/modules/audio_coding
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Alessio Bazzica d428ddd8f1 iSAC fixed point: fix int overflows 
Bug: webrtc:11137
Change-Id: If9276457b39285191ee2d9a0fbcb7e0a7a379be8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168523
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30513}
2020-02-13 11:14:41 +00:00
..
acm2 Disable opus tests to allow upgrade to opus 1.3 2020-01-30 14:57:15 +00:00
audio_network_adaptor Use newer version of TimeDelta and TimeStamp factories in modules/ 2020-02-10 11:49:57 +00:00
codecs iSAC fixed point: fix int overflows 2020-02-13 11:14:41 +00:00
include Reland "Send absolute capture time through audio coding module." 2020-01-27 13:18:27 +00:00
neteq NetEQ audio decoder unit test: use ParsePayload 2020-02-13 09:05:55 +00:00
test Reland "Send absolute capture time through audio coding module." 2020-01-27 13:18:27 +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 Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED 2020-01-16 15:20:35 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Remove ossu@ from audio/ and audio_coding/ OWNERS 2020-02-10 11:05:27 +00:00