Per Åhgren d82a02c837 ACM: Corrected temporary buffer size ... This CL corrects the temporary buffers size in the pre-processing of the capture audio before encoding. As part of this it removes the ACM-specific hardcoding of the size and instead ensures that the size of the temporary buffer matches that of the AudioFrame. Bug: webrtc:11242 Change-Id: I56dd6cadfd4e140e8e159966c33d1027383ea9fa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170340 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30775}		2020-03-12 12:23:20 +00:00
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acm2	ACM: Corrected temporary buffer size	2020-03-12 12:23:20 +00:00
audio_network_adaptor	Use newer version of TimeDelta and TimeStamp factories in modules/	2020-02-10 11:49:57 +00:00
codecs	Write protos as binary.	2020-03-12 09:43:57 +00:00
include	ACM: Corrected temporary buffer size	2020-03-12 12:23:20 +00:00
neteq	Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API	2020-03-11 12:08:32 +00:00
test	ACM: Corrected temporary buffer size	2020-03-12 12:23:20 +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	iSAC unit test: test encode/decode via API wrapper	2020-02-13 11:29:01 +00:00
DEPS	Fixing WebRTC after moving from src/webrtc to src/	2017-09-15 05:02:56 +00:00
OWNERS	Remove wildcard ownership for build files.	2020-02-19 14:05:46 +00:00