Try to only include headers which are really needed. We should
particularly check all "headers including other headers". The
long-term goal is to have a manageable, small API for plugins
(decoders, output) without so many mpd internals cluttering the
namespace.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7319 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Buffer sizes should be size_t. This is safe here, at least not
unsafer than without the patch. I have no idea why audioBufferSize
and audioBufferPos were explicitly declared as signed integer.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7296 09075e82-0dd4-0310-85a5-a0d7c8717e4f
we do not save anything by limiting a variable to an unsigned char,
since the compiler aligns it at machine word size anyway. however by
using the full machine word, we save one instruction, and we remove
the useless artificial limitation to 255.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7203 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Tools like "sparse" check for missing downcasts, since implicit cast
may be dangerous. Although that does not change the compiler result,
it may make the code more readable (IMHO), because you always see when
there may be data cut off.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7196 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
thread-safety work in preparation for rewrite to use pthreads
Expect no regressions against trunk (r7078), possibly minor
performance improvements in update (due to fewer heap
allocations), but increased stack usage.
Applied the following patches:
* maxpath_str for reentrancy (temporary fix, reverted)
* path: start working on thread-safe variants of these methods
* Re-entrancy work on path/character-set conversions
* directory.c: exploreDirectory() use reentrant functions here
* directory/update: more use of reentrant functions + cleanups
* string_toupper: a strdup-less version of strDupToUpper
* get_song_url: a static-variable-free version of getSongUrl()
* Use reentrant/thread-safe get_song_url everywhere
* replace rmp2amp with the reentrant version, rmp2amp_r
* Get rid of the non-reentrant/non-thread-safe rpp2app, too.
* buffer2array: assert strdup() returns a usable value in unit tests
* replace utf8ToFsCharset and fsCharsetToUtf8 with thread-safe variants
* fix storing playlists w/o absolute paths
* parent_path(), a reentrant version of parentPath()
* parentPath => parent_path for reentrancy and thread-safety
* allow "make test" to automatically run embedded unit tests
* remove convStrDup() and maxpath_str()
* use MPD_PATH_MAX everywhere instead of MAXPATHLEN
* path: get rid of appendSlash, pfx_path and just use pfx_dir
* get_song_url: fix the ability to play songs in the top-level music_directory
git-svn-id: https://svn.musicpd.org/mpd/trunk@7106 09075e82-0dd4-0310-85a5-a0d7c8717e4f
outputs, which is actually desired behaviour. This way if the shout server
takes a while to respond, the shout output can block until connected
without messing up other audio outputs.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6554 09075e82-0dd4-0310-85a5-a0d7c8717e4f
leave it in that state. Likewise, if an audio output is in state
DEVICE_ON, and reopening the device due to a format change fails, change it
to state DEVICE_ENABLE. This will prevent flushAudioBuffer from even
attempting to play audio on a closed device (even though it would fail
anyway).
git-svn-id: https://svn.musicpd.org/mpd/trunk@6529 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Some compilers and linkers aren't smart enough to optimize this,
as global variables are implictly initialized to zero. As a
result, binaries are a bit smaller as more goes in the .bss and
less in the text section.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5254 09075e82-0dd4-0310-85a5-a0d7c8717e4f
I'm checking for zero-size allocations and assert()-ing them,
so we can more easily get backtraces and debug problems, but we'll
also allow -DNDEBUG people to live on the edge if they wish.
We do not rely on errno when checking for OOM errors because
some implementations of malloc do not set it, and malloc
is commonly overridden by userspace wrappers.
I've spent some time looking through the source and didn't find any
obvious places where we would explicitly allocate 0 bytes, so we
shouldn't trip any of those assertions.
We also avoid allocating zero bytes because C libraries don't
handle this consistently (some return NULL, some not); and it's
dangerous either way.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4690 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Oops, I broke pause/resuming from a statefile r4514
Everything should be fixed out.
Also we now avoid opening the audio device until we have a
playable audio_format set. This is a long-standing bug that got
exposed more blatantly with the single array.
Thanks to MattD in #mpd for reporting my breakage.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4516 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Just malloc all of the audioOutput array in one shot
to avoid fragmentation and to improve cache locality
when iterating through the array.
We also know name and type members of the AudioOutput
struct won't change in the config, so there's no
need to strdup them.
newAudioOutput => initAudioOutput
git-svn-id: https://svn.musicpd.org/mpd/trunk@4515 09075e82-0dd4-0310-85a5-a0d7c8717e4f
It just made things more confusing. We'll just store
the states in playerData_pd->audioDevicesStates and be
done with it (it's a unsigned byte now).
git-svn-id: https://svn.musicpd.org/mpd/trunk@4514 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Some people have more than 8 devices (the old limit). It's
pretty easy to support as many as our hardware and OS allows
so we might as well.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4513 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This shaves another 5-6k because we've removed the paranoid
fflush() calls after every fprintf. Now we only fflush()
when we need to
git-svn-id: https://svn.musicpd.org/mpd/trunk@4493 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This patch massively reduces the amount of heap allocations at
the interface/command layer. Most commands with minimal output
should not allocate memory from the heap at all. Things like
repeatedly polling status, currentsong, and volume changes
should be faster as a result, and more importantly, not a source
of memory fragmentation.
These changes should be safe in that there's no way for a
remote-client to corrupt memory or otherwise do bad stuff to
MPD, but an extra set of eyes to review would be good. Of
course there's never any warranty :)
No longer do we use FILE * structures in the interface, which means
we don't have to allocate any new memory for most connections.
Now, before you go on about losing the buffering that FILE *
+implies+, remember that myfprintf() never took advantage of
any of the stdio buffering features.
To reduce the diff and make bugs easier to spot in the diff,
I've kept myfprintf in places where we write to files (and not
network interfaces). Expect myfprintf to go away entirely soon
(we'll use fprintf for writing regular files).
git-svn-id: https://svn.musicpd.org/mpd/trunk@4483 09075e82-0dd4-0310-85a5-a0d7c8717e4f
These are just warnings from sparse, but it makes the output
easier to read. I ran this through a quick perl script, but
of course verified the output by looking at the diff and making
sure the thing still compiles.
here's the quick perl script I wrote to generate this patch:
----------- 8< -----------
use Tie::File;
defined(my $pid = open my $fh, '-|') or die $!;
if (!$pid) {
open STDERR, '>&STDOUT' or die $!;
exec 'sparse', @ARGV or die $!;
}
my $na = 'warning: non-ANSI function declaration of function';
while (<$fh>) {
print STDERR $_;
if (/^(.+?\.[ch]):(\d+):(\d+): $na '(\w+)'/o) {
my ($f, $l, $pos, $func) = ($1, $2, $3, $4);
$l--;
tie my @x, 'Tie::File', $f or die "$!: $f";
print '-', $x[$l], "\n";
$x[$l] =~ s/\b($func\s*)\(\s*\)/$1(void)/;
print '+', $x[$l], "\n";
untie @x;
}
}
git-svn-id: https://svn.musicpd.org/mpd/trunk@4378 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Currently only ALSA is supported/tested, and only if the mixer
device is not on the audio device being disconnected (software
mixer).
This patch allows me to disconnect my Headroom Total Airhead USB
sound card, and resume playback (skips to the next song, which
should be fixed) when the device is plugged back in.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4364 09075e82-0dd4-0310-85a5-a0d7c8717e4f
