Nobody should call playAudio() with an empty chunk.  Add some
assertions on that.

Since open() and play() close the device on error, we can simply check
audio_output.open instead of audio_output.result after a call.

When one of several output devices failed, MPD tried to reopen it
quite often, wasting a lot of resources.  This patch adds a delay:
wait 10 seconds before retrying.  This might be changed to exponential
delays later, but for now, it makes the problem go away.

Moved code from syncAudioDeviceStates() to audio_output_update().

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

When there are standardized headers, use these instead of the bloated
os_compat.h.

pause() puts the audio output into pause mode: if supported, it may
perform a special action, which keeps the device open, but does not
play anything.  Output plugins like "shout" might want to play silence
during pause, so their clients won't be disconnected.  Plugins which
do not support pausing will simply be closed, and have to be reopened
when unpaused.

This pach includes an implementation for the shout plugin, which
sends silence chunks.

During debugging, I found a deadlock between flushAudioBuffer() and
the audio_output_task(): audio_output_task() didn't notice that there
is a command, and flushAudioBuffer() waited forever in notify_wait().
I am not sure yet what is the real cause; work around this for now by
waking up non-finished audio outputs in every iteration.

Send an output buffer to all output plugins at the same time, instead
of waiting for each of them separately.  Make several functions
non-blocking, and introduce the new function audio_output_wait_all()
to synchronize with all audio output threads.

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.

To keep I/O nastiness and latencies away from the core, move the audio
output code to a separate thread, one per output.  The thread is
created on demand, and currently runs until mpd exits.

If the output device is already open, it may have modified
outAudioFormat; in this case, outAudioFormat is still valid, and does
not need an overwrite.

Eliminate sameInAndOutFormats and check with audio_format_equals()
each time it this information is needed.  Another 4 bytes saved.

Instead of checking convertAudioFormat, we can simply check if
reqAudioFormat is defined.  This saves 4 bytes in the struct.

Rename it to audio_format_equals() and return "true" if they are
equal.

The "!src" check in copyAudioFormat() used to hide bugs - one should
never pass NULL to it.  There is one caller which might pass NULL, add
a check in this caller.

Instead of doing mempcy(), we can simply assign the structures, which
looks more natural.

Getting rid of CamcelCase, again.

Similar to decoder_control.c, output_control.c will provide functions
for controlling the output thread (which will be implemented later).

No CamelCase.  Also don't declare typedefs for the methods.

Instead of having to register each output plugin, store them
statically in an array.  This eliminates the need for the List library
here, and saves some small allocations during startup.

Instead of copying all that stuff from the audio output plugin to the
audio output structure, store a pointer to the plugin.

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

Just like decoder_api.h, output_api.h provides the audio output API
which is used by the plugins.

Get rid of CamelCase, and don't use a typedef, so we can
forward-declare it, and unclutter the include dependencies.

Getting rid of CamelCase; not having typedefs also allows us to
forward-declare the structures.

And again, convert arguments to const.

We want to expose the AudioFormat structure to plugins; remove some
clutter by moving its declaration to a separate header file.

git-svn-id: https://svn.musicpd.org/mpd/trunk@7345 09075e82-0dd4-0310-85a5-a0d7c8717e4f

When growing the audioOutput->convBuffer, we can use free()+malloc()
instead of realloc(), which saves a memcpy().

git-svn-id: https://svn.musicpd.org/mpd/trunk@7295 09075e82-0dd4-0310-85a5-a0d7c8717e4f

git-svn-id: https://svn.musicpd.org/mpd/trunk@7294 09075e82-0dd4-0310-85a5-a0d7c8717e4f

The audio output plugins should get a constant pointer, because they
must not modify the buffer.  Since the size is a non-negative buffer
size in bytes, we should change its type to size_t.

git-svn-id: https://svn.musicpd.org/mpd/trunk@7293 09075e82-0dd4-0310-85a5-a0d7c8717e4f

In my previous patch set, I forgot to change the
pcm_sizeOfConvBuffer() invocation in convertAudioFormat() to also use
size_t.

git-svn-id: https://svn.musicpd.org/mpd/trunk@7292 09075e82-0dd4-0310-85a5-a0d7c8717e4f

[ew: cleaned up the dirty union hack a bit]
Signed-off-by: Eric Wong <normalperson@yhbt.net>

git-svn-id: https://svn.musicpd.org/mpd/trunk@7180 09075e82-0dd4-0310-85a5-a0d7c8717e4f

This will make refactoring features easier, especially now that
pthreads support and larger refactorings are on the horizon.

Hopefully, this will make porting to other platforms (even
non-UNIX-like ones for masochists) easier, too.

os_compat.h will house all the #includes for system headers
considered to be the "core" of MPD.  Headers for optional
features will be left to individual source files.

git-svn-id: https://svn.musicpd.org/mpd/trunk@7130 09075e82-0dd4-0310-85a5-a0d7c8717e4f

git-svn-id: https://svn.musicpd.org/mpd/trunk@6527 09075e82-0dd4-0310-85a5-a0d7c8717e4f

one now, and trying to call NULL was causing a segfault at exit.

git-svn-id: https://svn.musicpd.org/mpd/trunk@6398 09075e82-0dd4-0310-85a5-a0d7c8717e4f

call to FATAL().

git-svn-id: https://svn.musicpd.org/mpd/trunk@6276 09075e82-0dd4-0310-85a5-a0d7c8717e4f

because lsr may return less than the input buffer size, and the rest of the
audio code needs to know the new size.  This fixes the clicking that was
introduced with recent changes to the lsr code.  A huge thanks to remiss
for figuring this out.

git-svn-id: https://svn.musicpd.org/mpd/trunk@6273 09075e82-0dd4-0310-85a5-a0d7c8717e4f

audio at once, so it won't work for us.  The old full API code was still
heavily broken, as each call to pcm_convertSampleRate() used the same
state, even if it was processing two streams of audio.  The new code keeps
a separate state for each audio stream that's being converted.

git-svn-id: https://svn.musicpd.org/mpd/trunk@6255 09075e82-0dd4-0310-85a5-a0d7c8717e4f

and samplerate conversion.  This makes the code much easier to read, and
fixes a few bugs that were previously there.

git-svn-id: https://svn.musicpd.org/mpd/trunk@6224 09075e82-0dd4-0310-85a5-a0d7c8717e4f