ad_lavc: use fmt-conversion to map sample formats
af: uncrustify
audio/filter: replace pointless memcpys with assignments

The change in af_scaletempo actually fixes a memory leak. af->data
contained a pointer to an allocated buffer, which was overwritten
during format negotiation. Set the format explicitly instead.
af_lavrresample: allow other ffmpeg sample formats for input/output

The format was locked to s16. Extend it to accept all other ffmpeg
sample formats, and even allow different in- and output formats. The
generic filter code will still insert af_format on format mismatches,
though.
af: remove automatically inserted filters on full reinit

Make sure automatically inserted filters are removed on full reinit
(they are re-added later if they are really needed). Automatically
inserted filters were never explicitly removed, instead, it was
expected that redundant conversion filters detach themselves. This
didn't work if there were several chained format conversion filters,
e.g. s16le->floatle->s16le, which could result from repeated filter
insertion and removal. (format filters detach only if input format and
output format are the same.)

Further, the dummy filter (which exists only because af.c can't handle
an empty filter chain for some reason) could introduce bad conversions
due to how the format negotiation works. Change the code so that the
dummy filter never takes part on format negotiation. (It would be better
to fix format negotiation, but that would be much more complicated and
would involving fixing all filters.)

Simplify af_reinit() and remove the start audio filter parameter. This
means format negotiation and filter initialization is run more often,
but should be harmless.
af: use af_lavrresample for format conversions, if possible

Refactor to remove the duplicated format filter insertion code. Allow
other format converting filters to be inserted on format mismatches.
af_info.test_conversion checks whether conversion between two formats
would work with the given filter; do this to avoid having to insert
multiple conversion filters at once and such things. (Although this
isn't ideal: what if we want to avoid af_format for some conversions?
What if we want to split af_format in endian-swapping filters etc.?)

Prefer af_lavrresample for conversions that it supports natively,
otherwise let af_format handle the full conversion.
af: factor channel filter insertion

Do this just like it has been done for the format filter.
af: simplification

If format negotiation fails, and additional filters are inserted to fix
this, don't try to reinitialize the filter immediately. Instead, correct
the audio format, and let the caller retry.

Add a retry counter to af_reinit() to ensure that misbehaving filters
can't put the format negotiation into an endless loop.
ao_alsa: reorder channels from internal to alsa order

Currently, internal and alsa order are exactly the same, so this will do
absolutely nothing.
audio: switch to libavcodec channel order, use libavresample for mixing

Switch the internal channel order to libavcodec's. If the channel number
mismatches at some point, use libavresample for up- or downmixing.
Remove the old af_pan automatic downmixing.

The libavcodec channel order should be equivalent to WAVEFORMATEX order,
at least nowadays. reorder_ch.h assumes that WAVEFORMATEX and libavcodec
might be different, but all defined channels have the same mappings.

Remove the downmixing with af_pan as well as the channel conversion with
af_channels from af.c, and prefer af_lavrresample for this. The
automatic downmixing behavior should be the same as before (if the
--channels option is set to 2, which is the default, the audio output
is forced to 2 channels, and libavresample does all downmixing).

Note that mpv still can't do channel layouts. It will pick the default
channel layout according to the channel count. This will be fixed later
by passing down the channel layout as well.

af_hrtf depends on the order of the input channels, so reorder to ALSA
(for which this code was written). This is better than changing the
filter code, which is more risky.

ao_pulse can accept waveext order directly, so set that as channel
mapping.
audio: remove float processing option

The only thing this option did was changing the behavior of af_volume.
The option decided what sample format af_volume would use, but only if
the sample format was not already float. If the option was set, it would
default to float, otherwise to S16.

Remove use of the option and all associated code, and make af_volume
always use float (unless a af_volume specific sub-option is set).

Silence maximum value tracking. This message is printed when the filter
is destroyed, and it's slightly annoying. Was enabled due to enabling
float by default.
af: remove force option

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).
af: remove accuracy option

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.
options: remove --af-adv

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.
af_lavrresample: add no-detach suboption

Normally, af_lavrresample detaches itself immediately if the input and
output audio parameters are the same. no-detach prevents this.
manpage: update af_lavrresample entry

Reflects the changes over the last few commits.
af: streamline format negotiation

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).
af: fix negotiation endless loop

Yeah... ok.

Can be reproduced by having AF_CONTROL_CHANNELS not really set the
correct channel map.
options: add option to prevent decoder audio downmixing

Also rename --a52drc to --ad-lavc-ac3drc, and add --ad-lavc-o.
For now, disable autodetection of sdl/sdl2

This is done because statically linked SDL libraries are incompatible
with direct X11 function use (e.g. vo_x11, vo_gl etc.) because of
clashing symbol names.

http://bugzilla.libsdl.org/show_bug.cgi?id=1828