webrtc/modules/audio_coding
Karl Wiberg 338f58d95c iSAC decoder: Don't read past the end of the buffer of encoded bytes
Bug: chromium:825524
Change-Id: Iff40a9fd62a34474af71b51dd3831a16412fbf3b
Reviewed-on: https://webrtc-review.googlesource.com/66361
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22748}
2018-04-05 13:22:53 +00:00
..
acm2 This CL removes all usages of our custom ostream << overloads. 2018-04-03 12:51:00 +00:00
audio_network_adaptor Move file_wrapper.h to rtc_base/system/ 2018-03-23 11:17:15 +00:00
codecs iSAC decoder: Don't read past the end of the buffer of encoded bytes 2018-04-05 13:22:53 +00:00
include Add new UMA metric for NetEq target buffer delay 2017-11-29 08:56:29 +00:00
neteq This CL removes all usages of our custom ostream << overloads. 2018-04-03 12:51:00 +00:00
test Move rw_lock_wrapper.h to rtc_base/synchronization/ 2018-03-23 19:47:08 +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 This CL removes all usages of our custom ostream << overloads. 2018-04-03 12:51:00 +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