The only caller of deletePlaylist() appends PLAYLIST_FILE_SUFFIX, so
we can be sure it's already there.  We don't need to stat the file,
since unlink() does all the checking.

Commit 80a2c937 broke resume after pause: it cleared the
input_audio_format when it attempted to simplify a complicated
expression.  Don't clear it, just assign input_audio_format if a new
format was specified.

The inline functions directory_is_empty() and directory_get_path()
don't modify the object - pass constant object pointers to them.

We only need to lock sv->nr changes to prevent traversals ( why
it's called "nr_lock").  free(3) is a "slow" function on my
system; so we can avoid unnecessarily holding a lock long for
longer than needed.

Previously only updates with subdirectories being
specified could be queued.  No harm in queueing
full updates.

This way we avoid unnecessary heap allocations.

Not all compilers support struct packing, and those that don't
shouldn't be punished for it.

Added configuration parameter "protocol" which lets the user choose
from 3 shout protocols.  This adds support for real shoutcast servers.

Case insensitivity isn't helpful, and comparing only the first 3 bytes
of a configured value may encourage users to supply wrong or
misleading values.

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.

When calculating the conversion buffer size, don't hard-code the
formulas for only mono<->stereo.

Convert any number of channels to stereo.  In fact, this isn't really
stereo, it's rater mono blown up to stereo.  This patch should only
make it possible to play 5.1 files at all; "real" conversion to stereo
should be implemented, but for now, this is better than nothing.

In order to be able to deal with non-trivial conversions,
pcm_convertChannels() needs to know both the input and the output
channel count.  Simplify buffer allocation in that function.

Moved code from pcm_convertChannels() to pcm_convert_channels_1_to_2()
and pcm_convert_channels_2_to_1().  Improved the quality of
pcm_convert_channels_2_to_1() by calculating the arithmetic mean value
of both samples.

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

Pass int16_t pointers instead of char pointers to functions which can
deal with 16 bit audio only.

buffered_before_play was copied to struct player because it was used
to disable buffering when seeking.  Instead of mainaining a copy of
this number, move just the flag to the player struct.

Renamed audio_configFormat to configured_audio_format.  Renamed
audio_buffer.format to input_audio_format.  Simplified its
initialization in openAudioDevice().

All what's left from ad_state is a simple boolean flag (ENABLE /
DISABLE).  Convert it to a C99 bool.

audio.c maintained one of MPD's many layers of audio buffers.  It was
without any benefit, since playAudio() can simply send the source
buffer directly to the audio output plugin.

The function ob_is_empty() is called very often.  It's worth it to
convert it to an inline function.

QUEUE adds a new song to the player's queue.  CANCEL clears the queue.
These two commands replace the old and complex queueState and
queueLockState code.

Simplify and merge several if clauses before the clearPlayerQueue()
invocation.  Call clearPlayerQueue() only if a song is actually
queued; add an assertion for that in clearPlayerQueue().

The player struct holds the local variables which used to be passed to
all those helper functions in player_thread.c.

set_current_song() is an internal function and consists of only one
assignment.  Eliminate it.

This variable is superfluous, it is only used to copy its value to
player_control.totalTime.  Since the original source of this value
(song->tag->time) will still be available at this point, we can safely
remove fileTime.

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.

Leftover from the output API changes: oss_open_default() was changed
to return a void*, but it still returned "0" to report success.
Report the OssData pointer instead.

The decoder was woken up after each chunk which had been played.  That
caused a lot of superfluous context switches.  Wake up the decoder
only when a certain amount of the buffer has been consumed.  This
formula is somewhat arbitrary, and has to be proven experimentally.

The mp3 plugin did not use the MAD_NCHANNELS() value correctly: when a
stream was not stereo, it was assumed to be mono, although the correct
number was passed to MPD.  libmad doesn't support more than 2
channels, but this change allows gcc to optimize its inlining
strategy.

The dithering function audio_linear_dither() worked for signed 16 bits
only anyway, having a variable "bits" just disables important gcc
optimizations.

player_get_audio_format() replaces getPlayerSampleRate(),
getPlayerBits(), getPlayerChannels().

This replaces the attributes bits, channels, sampleRate.

Fix some gcc warnings by using unsigned where appropriate.  Declare
numfrequencies as "const", and replaced the magic number 12 with a
sizeof.

Again, a plugin which was disabled in my test environment and suffered
from compilation errors.

This replaces the separate properties channels, sampleRate, bits.

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).