Roughly follows MPlayer svn commits 36492 and 36493. We also remove
the volume peak reporting. (There are much better libavfilter filters
for this, I think.)
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.
Set the input/output format in filter init. This doesn't change anything
functionally, but it makes the forced format show up in the filter chain
init verbose output (which sometimes prints the filter chain before all
filters have been configured).
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.
Calling them separately doesn't really make sense, and all existing
calls to them usually combined them. One subtitle difference was that
af_init() didn't wipe the filter chain if initialization of the chain
itself failed, but that didn't really make sense anyway.
Also remove af_init() from the code for setting balance in mixer.c. The
mixer should be in the initialized state only if audio is fully
initialized, so the af_init() call made no sense.
Note that the filter "editing" code in command.c doesn't really do a
nice job of handling errors in case recreating an _old_ (known to work)
filter chain unexpectedly fails, and this obscure/rare case might be
differently handled after this change.
Softvol always used a linear multiplier for volume control. This was
converted to dB, and then back to linear in af_volume. Remove this non-
sense. We still try to keep the command line argument to af_volume in
dB, though.
It's quite unlikely, but functions like mp_find_user_config_file() can
return NULL, e.g. if $HOME is unset.
Fix all the code that didn't check for this correctly yet.
Having to use -1 for that is generally quite annoying.
Audio formats are created from bitmasks, and it can't be excluded that
0 is not a valid format. Fix this by adjusting AF_FORMAT_I so that it
is never 0. Along with AF_FORMAT_F and the special formats, all valid
formats are covered and guaranteed to be non-0.
It's possible that this commit will cause some regressions, as the
check for invalid audio formats changes a bit.
The --speed option and the speed property used float. Change them to
double.
Change the commands that manipulate the property (speed_mult/add) to
double as well. Since the cycle command shares code with the add
command, we change that as well.
The reason for this change is that this allows better control over
speed, such as stepping by semitones. Using floats is also just plain
unnecessary.
In general, this warning can hint to actual bugs. We don't enable it
yet, because it would conflict with some unmerged code, and we should
check with clang too (this commit was done by testing with gcc).
Make the VF/VO/AO option parser available to audio filters. No audio
filter uses this yet, but it's still a quite intrusive change.
In particular, the commands for manipulating filters at runtime
completely change. We delete the old code, and use the same
infrastructure as for video filters. (This forces complete
reinitialization of the filter chain, which hopefully isn't a problem
for any use cases. The old code forced reinitialization too, but it
could potentially allow a filter to cache things; e.g. consider loaded
ladspa plugins and such.)
This code is supposed to run if dynamic filter insertion (such as when
inserting a volume filter in mixer.c) fails. Then it removes all filters
and recreates the default list of filters. But the code just blew up and
entered an endless loop, because it removed even the sentinel in/out
filters. This could happen when trying to use softvol controls while
using spdif, but also other situations. Fix it by calling the correct
code.
Also remove these obnoxious yoda-conditions.
Mostly copied from vf_lavfi. The parts that could be shared are minor,
because most code is about setting up audio and video, which are too
different.
This won't work with Libav. I used ffplay.c as guide, and noticed too
late that their setup methods are incompatible with Libav's. Trying to
make it work with both would be too much effort. The configure test for
av_opt_set_int_list() should disable af_lavfi gracefully when compiling
with Libav.
Due to option parser chaos, you currently can't have a "," as part of
the filter graph string - not even with quoting or escaping. This will
probably be fixed later.
The audio filter chain is not PTS aware. So we have to do some hacks
to make up a fake PTS, and we have to map the output PTS back to the
filter chain's method of tracking PTS changes and buffering, by
adjusting af->delay.
The libavresample version of the current Libav stable release lacks the
avresample_set_channel_mapping() function. (FFmpeg's libswresample seems
to be fine, because they added swr_set_channel_mapping() first.)
Add a cheap/slow workaround to do channel reordering on our own. We
don't use the recently removed MPlayer code (see commit 586b75a),
because that is not generic enough.
The functionality should be the same as with full-featured
libavresample, and any differences are bugs. It's probably slower,
though.
af_reinit() is responsible for inserting automatic conversion filters
for channel remixing, format conversion, and resampling. We don't
require that a single filter can do all these (even though
af_lavrresample does nearly all of this, sometimes af_format has to be
used instead for format conversions). This makes setting up the chain
more complicated, and a way is needed to prevent endless appending of
conversion filters if a conversion is not possible.
Until now, this used a stupidly simple yet robust static retry limit to
detect failure. This is perfectly fine, and the limit (20) was good
enough to handle about ~5 filters. But with more filters, and if each
filter requires 3 additional conversion filters, this would fail. So
raise the limit to 4 retries per filter. This is still stupidly simple
and robust, but won't arbitrarily fail if the filter count is too large.
If one of the input or output is an unknown layout, but the other is
known, it can still happen that channels are remixed randomly. Avoid
this by forcing default layouts in this case. (Doesn't work if the
channel counts are different.)
This is done in af_lavrresample now, and as part of format negotiation.
Also remove the remaining reorder_channel calls. They were redundant
and did nothing.
mp_audio has some redundant fields. Setters like mp_audio_set_format()
initialize these properly.
Also move the mp_audio struct to a the file audio.c.
We can remove a mysterious line of code from af.c:
in.format |= af_bits2fmt(in.bps * 8);
I'm not sure if this was ever actually needed, or if it was some kind of
"make it work" quick-fix that works against the way things were supposed
to work. All filters etc. now set the format correctly, so if there ever
was a need for this code, it's definitely gone.
Add dummy input and output filters to remove special cases in the format
negotiation code (af_fix_format_conversion() etc.). The output of the
filter chain is now negotiated in exactly the same way as normal
filters.
Negotiate setting the sample rate in the same way as other audio
parameters. As a side effect, the resampler is inserted at the start of
the filter chain instead of the end, but that shouldn't matter much,
especially since conversion and channel mixing are conflated into the
same filter (due to libavresample's API).
Anything this option did has been removed in the preceding 3 commits.
Note that even though these options sounded like a good idea (like
setting accuracy vs. speed tradeoffs), they were not really properly
implemented.
All this option did was deciding whether the resample filter was to be
insert at the beginning or end of the filter chain. Always do what the
option set for accuracy did. I doubt it makes much of a difference.
libavresample does most things in just one go anyway, so it won't
matter.
Dangerous and misleading. If it turns out that this is actually needed
to make certain setups work right, it should be added back in a better
way (in a way it doesn't cause random crashes).
