Henrik Lundin 80b2806250 Fixing a buffer overflow in Merge::Downsample ... In the unlikely event that the decoded audio is really short, the downsampling would read outside of the decoded audio vector. This CL fixes that, and adds a unit test that verifies the fix (when running with ASan). Bug: chromium:1016506 Change-Id: Ifb8071ce0550111cd66e7f7c1bed7f17b33f93c5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160304 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29898}		2019-11-25 12:16:30 +00:00
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acm2	ACM: Adding unittests for the remixing functionality	2019-11-20 06:20:22 +00:00
audio_network_adaptor	Removes RPLR based FEC controller.	2019-10-31 13:56:44 +00:00
codecs	Adding GetInDtx to WebRTC Opus Interface.	2019-11-19 14:14:06 +00:00
include	Enable injection of a custom NetEqFactory into PeerConnectionFactory.	2019-11-01 11:30:36 +00:00
neteq	Fixing a buffer overflow in Merge::Downsample	2019-11-25 12:16:30 +00:00
test	Make Opus PLC always output 10ms audio.	2019-11-07 21:15:58 +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	ACM: Adding unittests for the remixing functionality	2019-11-20 06:20:22 +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