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
Functions that should stay inlined should have an explanation
attached to them.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4355 09075e82-0dd4-0310-85a5-a0d7c8717e4f
In the words of the original author, it was 'crappy'. I tend to
agree :)
The code has also been broken for at least the past few months,
and nobody bothered fixing it
The previous format it was overly complex: 5 lines to describe
each device. The new format is one-line per-device:
audio_device_state:%d:%s
%d - 0 for disabled, any integer for enabled
%s - name of the device as specified in the config file,
whitespace and all
Incompatibilities:
* Output names are now _required_ to be unique.
This is required because the new format relies solely on the
name of the audio device.
Relying on the device IDs internal to MPD was a bad idea
anyways since the user usually has none or very little idea
how they're generated, and adding a new device or removing
one from a config would throw things off completely.
This is also just a Good Idea(TM) because it makes things
less confusing to users when they see it in their clients.
* Output states are not preserved from the previous format.
Not a big deal, since the previous code was never officially
released. Also, it's been broken for months now, so I doubt
anybody would notice :)
git-svn-id: https://svn.musicpd.org/mpd/trunk@3928 09075e82-0dd4-0310-85a5-a0d7c8717e4f
now format can be specified for each different audioOutput device
git-svn-id: https://svn.musicpd.org/mpd/trunk@2474 09075e82-0dd4-0310-85a5-a0d7c8717e4f
warnings are buffered until the error log is opened, and then flushed to the
error log.
git-svn-id: https://svn.musicpd.org/mpd/trunk@1442 09075e82-0dd4-0310-85a5-a0d7c8717e4f
right conversion routines for bit conversion and channel conversion
git-svn-id: https://svn.musicpd.org/mpd/trunk@971 09075e82-0dd4-0310-85a5-a0d7c8717e4f
J. Alexander Treuman