This patch tryes to introduce pluggable mixer (struct mixer_plugin) along with some basic infrastructure (mixer_* functions). Instance of mixer (struct mixer) is used in
alsa and oss output plugin

All what's left in pcm_utils.h is the pcm_range() utility function,
which is only used internally by pcm_volume and pcm_mix.

Use GLib's G_GNUC_UNUSED instead of gcc.h's mpd_unused.

Commit dd7711d8 removed MPD's default ALSA buffer_time.  The result
was a buffer size which was way too small for playing streams on some
sound hardware, and caused skips and distorted sound.  Revert the
default to 500 ms.

When using autodetection, AlsaData.device wasn't properly initialized
with NULL.  This broke autodetection randomly.

Make the pointers "device" and "name" non-const, so we don't need the
xfree() hack.  The default value is expressed as NULL.

When an output plugin fails to play a chunk, close it.  This replaces
various manual close() calls in nearly all plugins.

Don't return 0/-1 on success/error, but true/false.  Instead of int,
use bool for storing flags.

Don't compile the sources of disabled output plugins at all.

Again, no CamelCase in the directory name.

These macros are not available in older libasound versions (1.0.13
fails, 1.0.16 is ok).  Ignore the configuration if the constants are
not defined.

Added mpd.conf options for disabling automatic resamling, sample
format and channel conversion.  This way, users may choose to override
ALSA's automatic resampling, and use libsamplerate instead.

If the sample format isn't supported by the device (i.e. 24 bit on
low-end sound chips), fall back to 16 bit output.  There is code in
pcm_utils.c which converts PCM data to 16 bit.

Move code which loads configuration to alsa_configure().  This removes
one indent level.

ALSA does a good job measuring its buffer_time and period_time.  Don't
override its defaults, unless the user demands it.

Revert e4f5d6bd "re-enable-nonblocking, but sleep if busy".
Non-blocking mode with manual sleeping doesn't help at all (by the
way, the patch should have used snd_pcm_wait() instead of
my_usleep()).  ALSA knows much more about the hardware quirks, so we
just let it do the job.

A frame contains one sample per channel, thus it is sample_size *
channels.  This patch includes some cleanup for various locations
where the sample size for 24 bit audio was still 3 bytes (instead of
4).

The last bit of CamelCase in audio_format.h.  Additionally, rename a
bunch of local variables.

"bits" and "channels" cannot be negative.

The old struct initializers are error prone and don't allow moving
elements around.  Since we are going to overhaul some of the APIs
soon, it's easier to have all implementations use C99 initializers.

Seeing the "mpd_" prefix _everywhere_ is mind-numbing as the
mind needs to retrain itself to skip over the first 4 tokens of
a type to get to its meaning.  So avoid having extra characters
on my terminal to make it easier to follow code at 2:30 am in
the morning.

Please report any new issues you may come across on Free
toolchains.  I realize how difficult it can be to build/maintain
cross-compiling toolchains and I have no intention of forcing
people to upgrade their toolchains to build mpd.

Tested with gcc 2.95.4 and and gcc 4.3.1 on x86-32.

We have eliminated direct accesses to the audio_output struct from
the all output plugins.  Make it opaque for them, and move its real
declaration to output_internal.h, similar to decoder_internal.h.

Pass the opaque structure to plugin.init() only, which will return the
plugin's data pointer on success, and NULL on failure.  This data
pointer will be passed to all other methods instead of the
audio_output struct.

Since the output plugin returns a value indicating success or error,
we can have the output core code assign the "open" flag.

Pass the globally configured audio_format as a const pointer to
plugin.init().  plugin.open() gets a writable pointer which contains
the audio_format requested by the plugin.  Its initial value is either
the configured audio_format or the input file's audio_format.

The inline function audio_format_sample_size() calculates how many
bytes each sample consumes.  This function already takes into account
that 24 bit samples are 4 bytes long, not 3.

Instead of letting ALSA block for us (and potentially allowing
something stupid on certain hardware or drivers), we do the
sleeping ourselves.  We calculate the sleep to be a fraction of
period_time to avoid oversleeping (and thus audible skipping).

The way we used non-blocking mode was HORRIBLE.

It was non-blocking to ALSA, but we end up blocking in a busy
loop that does absolutely NOTHING but retry.  We don't check
for playback cancellation (like we do in decoders) or anything.

This is seriously broken and I can imagine it affects people on
fast CPUs more because we do asynchronous output buffering and
our ALSA device will always have data ready.

Lets not use deprecated functions. It's apparently
possible to not care about the sw_params stuff at all!

This is safer than the patch in
  http://www.musicpd.org/mantis/view.php?id=1542
with multiple audio outputs enabled.

Sadly, I only noticed that patch/problem when I googled for
"snd_config_update_free_global"

We never use it for anything anyways as we release the device
entirely on pause.

That's the name of this project.

Apparently snd_pcm_hw_params_can_resume() can return false even
though my hardware does in fact support resuming.  So stop
carrying that value in the canResume flag and just try to resume
when we're in the suspended state; falling back to
snd_pcm_prepare only if resuming fails.  libao does something
similar on resume, too.

While we're at it, use the E() macro which will enable us to
have better error reporting.

[mk: remove the E() macro stuff]

Since the plugin struct is never modified, we should store it in
constant locations.

Also rename AudioOutputPlugin to struct audio_output_plugin, and use
forward declarations to reduce include dependencies.