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author | wm4 <wm4@nowhere> | 2017-07-07 17:35:09 +0200 |
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committer | wm4 <wm4@nowhere> | 2017-07-07 17:54:05 +0200 |
commit | 90dd2298713d3414bad39b6e8648490cd9f52603 (patch) | |
tree | e1da2c0b5d04a2e2fd6bce752131dd3089e603f9 /appveyor.yml | |
parent | 7c1db05cbb386522017942a173cc9552637a660e (diff) | |
download | mpv-90dd2298713d3414bad39b6e8648490cd9f52603.tar.bz2 mpv-90dd2298713d3414bad39b6e8648490cd9f52603.tar.xz |
audio/out: add helper code to do 24 bit conversion in AO
I plan to remove the S24 sample formats in mpv. It seems like we should
still support this _somehow_ in AOs though. So the idea is to convert
the data to more obscure representations (that would not be useful for
filtering etc. anyway) within the AO.
This commit adds helper to enable this. ao_convert_fmt is meant to
provide mechanisms for this, rather than a generic audio format
description (as the latter leads only to overly generic misery). The
conversion also supports only cases which we think will be needed at
all.
The main advantage of this approach is that we get S24 out of sight,
and that we could support other crazy formats (like S20). The main
disadvantage is that usually S32 will be selected (if both S32 and S24
are available), and there's no user control to force S24. That doesn't
really matter though, and at worst makes testing harder or will lead
to unpleasant arguments with audiophiles (they'd be wrong anyway).
ao_convert_fmt.pad_lsb is ignored, although if we ever find a case in
which playing S32 with data in the LSBs breaks when playing it as padded
24 bit format. (For example, WAVEFORMATEXTENSIBLE recommends setting the
unused bits to 0 if wValidBitsPerSample implies LSB padding.)
Diffstat (limited to 'appveyor.yml')
0 files changed, 0 insertions, 0 deletions