| Commit message (Collapse) | Author | Age | Files | Lines |
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When af_scaletempo2.c:process() detects a format change, it goes back
through mp_scaletempo2_init() to reinitialize everything. However,
mp_scaletempo2.input_buffer is not necessarily reallocated due to a
check in af_scaletempo2_internals.c:resize_input_buffer(). This is a
problem if the number of audio channels increases, since without
reallocating, the buffer for the new channel(s) will at best point to
NULL, and at worst uninitialized memory.
Since resize_input_buffer() is only called from two places, pull size
check out into mp_scaletempo2_fill_input_buffer(). This allows each
caller to decide whether they want to resize or not. We could be
smarter about when to reallocate, but that would add a lot of machinery
for a case I don't expect to be hit often in practice.
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This brings my scaletempo2 benchmark down from ~22s to ~7s on my machine
(-march=native), and down to ~11s with a generic compile.
Guarded behind an appropriate #ifdef to avoid being ableist against
people who have the clinical need to run obscure platforms.
Closes #8848
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The input buffer size was fixed, but the required size depends on the
speed. Now the buffer will be resized dynamically.
Fixes #8081
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I forgot why/how (C99?), but other code also uses it.
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scaletempo2 is a new audio filter for playing back
audio at modified speed and is based on chromium
commit 51ed77e3f37a9a9b80d6d0a8259e84a8ca635259.
It sounds subjectively better than the existing
implementions scaletempo and rubberband.
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