| Commit message (Collapse) | Author | Age | Files | Lines |
|---|
| |
|
|
|
|
|
|
| |
The AF control commands used an elaborate and unnecessary organization
for the command constants. Get rid of all that and convert the
definitions to a simple enum. Also remove the control commands that
were not really needed, because they were not used outside of the
filters that implemented them.
|
| |
|
|
|
|
| |
You get the native format by not appending any suffix to the format.
This change includes user-facing names, e.g. for the --format option.
|
| |
|
|
|
|
|
|
|
|
|
|
|
| |
Before this commit, the af_instance->mul/delay values were in bytes.
Using bytes is confusing for non-interleaved audio, so switch mul to
samples, and delay to seconds. For delay, seconds are more intuitive
than bytes or samples, because it's used for the latency calculation.
We also might want to replace the delay mechanism with real PTS
tracking inside the filter chain some time in the future, and PTS
will also require time-adjustments to be done in seconds.
For most filters, we just remove the redundant mul=1 initialization.
(Setting this used to be required, but not anymore.)
|
| |
|
|
|
|
|
|
|
|
|
|
|
| |
Allocate af_instance->data in generic code before filter initialization.
Every filter needs af->data (since it contains the output
configuration), so there's no reason why every filter should allocate
and free it.
Remove RESIZE_LOCAL_BUFFER(), and replace it with mp_audio_realloc_min().
Interestingly, most code becomes simpler, because the new function takes
the size in samples, and not in bytes. There are larger change in
af_scaletempo.c and af_lavcac3enc.c, because these had copied and
modified versions of the RESIZE_LOCAL_BUFFER macro/function.
|
| |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
Based on earlier work by Stefano Pigozzi.
There are 2 changes:
1. Instead of mp_audio.audio, mp_audio.planes[0] must be used.
2. mp_audio.len used to contain the size of the audio in bytes. Now
mp_audio.samples must be used. (Where 1 sample is the smallest unit
of audio that covers all channels.)
Also, some filters need changes to reject non-interleaved formats
properly.
Nothing uses the non-interleaved features yet, but this is needed so
that things don't just break when doing so.
|
| |
|
|
|
| |
The conversion works for native endian only. The correct check lists
supported format combination explicitly, but is also much simpler.
|
| |
|
|
|
|
|
| |
Drop the author and comment fields. They were completely unused - not
even printed in verbose mode, just dead weight.
Also use designated initializers and drop redundant flags.
|
|
|
af_format is the old audio conversion filter. It could do all possible
conversions supported by the audio chain. However, ever since the
addition of af_lavrresample, most conversions are done by
libav/swresample, and af_format is used as fallback.
Separate out the fallback cases and remove af_format. af_convert24 does
24 bit <-> 32 bit conversions, while af_convertsignendian does sign and
endian conversions. Maybe the way the conversions are split sounds a bit
odd. But the former changes the size of the audio data, while the latter
is fully in-place, so there's at least different buffer management.
This requires a quite complicated algorithm to make sure all these
"partial" conversion filters can actually get from one format to
another. E.g. s24le->s32be always requires convertsignendian and
convert24, but af.c has no idea what the intermediate format should
be. So I added a graph search (trying every possible format and
filter) to determine required format and filter. When I wrote this,
it seemed this was still better than messing everything into
af_lavrresample, but maybe this is overkill and I'll change my
opinion. For now, it seems nice to get rid of af_format though.
The AC3->IEC61937 conversion isn't supported anymore, but I don't think
this is needed anywhere. Most AOs test all formats explicitly, or use
the AF_FORMAT_IS_IEC61937() macro (which includes AC3).
One positive consequence of this change is that conversions always
include dithering (done by libav/swresample), instead of possibly going
through af_format, which doesn't do anything fancy.
Rename af_force to af_format. It's essentially compatible with command
line uses of af_format. We retain a compatibility alias for af_force.
|
