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webrtc/modules/audio_coding
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Per Åhgren 85c2dafbf3 [Merge M81] - 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.

(cherry picked from commit d82a02c837)

No-Try: True
TBR: kwiberg@webrtc.org
Bug: webrtc:11242, chromium:1060647
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-Original-Commit-Position: refs/heads/master@{#30775}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170780
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/branch-heads/4044@{#10}
Cr-Branched-From: be99ee8f17f93e06c81e3deb4897dfa8253d3211-refs/heads/master@{#30432}
2020-03-18 20:22:23 +00:00
..
acm2 [Merge M81] - ACM: Corrected temporary buffer size 2020-03-18 20:22:23 +00:00
audio_network_adaptor Removes RPLR based FEC controller. 2019-10-31 13:56:44 +00:00
codecs Disable opus tests to allow upgrade to opus 1.3 2020-01-30 14:57:15 +00:00
include [Merge M81] - ACM: Corrected temporary buffer size 2020-03-18 20:22:23 +00:00
neteq Disable opus tests to allow upgrade to opus 1.3 2020-01-30 14:57:15 +00:00
test [Merge M81] - ACM: Corrected temporary buffer size 2020-03-18 20:22:23 +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 Make ivoc owner of audio_coding. 2018-10-15 15:08:28 +00:00