webrtc/modules/audio_coding/include
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
..
audio_coding_module.h ACM: Corrected temporary buffer size 2020-03-12 12:23:20 +00:00
audio_coding_module_typedefs.h Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API 2020-03-11 12:08:32 +00:00