| Commit message (Collapse) | Author | Age | Files | Lines |
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5267 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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just casting to int because it's the simplest (%z is not
well-supported)
Noticed-by: avuton on a 64-bit machine
git-svn-id: https://svn.musicpd.org/mpd/trunk@5257 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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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
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5252 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5251 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5250 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5249 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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There's no reason they shouldn't be static. Additionally,
output_ports doesn't need to be initialized to NULLs; that is
(and has always been) implicit (for all global variables)
git-svn-id: https://svn.musicpd.org/mpd/trunk@5247 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5192 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5191 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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..and not the enum value that corresponds to that bitrate
git-svn-id: https://svn.musicpd.org/mpd/trunk@5030 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4913 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4912 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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This patch should continue to allow mpd to play as well as
possible to icecast servers while avoiding stalls on local
devices. This has eliminated ALSA underrun errors
for me while streaming to a remote host while the network
connection was bad.
Of course, this makes opening a connection non-blocking, too,
so myShout_openShoutConn is a bit more complex.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4898 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4893 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4750 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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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
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4689 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Unfortunately there doesn't seem to be an indent switch for this,
but we have find + perl:
find src -name '*.[ch]' | xargs perl -i -p -e \
's/^\s+(\w+):/$1:/g unless /^\s+default:/'
This is a followup to r4605, and there are no actual code
changes in this.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4661 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Passing a ref to snd_pcm_hw_params_set_{buffer,period}_time_near
can modify our internal {period,buffer}_time members inside the
AlsaData structure, making re-initializing the device across
sample/bit rate and channel changes non-idempotent.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4616 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4612 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4599 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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* less-commonly compiled things like ao/mvp outputs
* Adding -Wno-transparent-union to SPARSE_FLAGS makes it check
inside decode.c, directory.c, player.c, and sig_handlers.c
* remove unused variables leftover from the master process
in sig_handlers.c
git-svn-id: https://svn.musicpd.org/mpd/trunk@4598 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4518 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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making it more presentable isn't something I'm willing to do before 0.12. It will likely be added back after 0.12, along with some very experimental stuff to make it more usable.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4472 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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We'll try setting an initial value of 50ms, and halve it each
time snd_pcm_hw_params fails with -EPIPE.
This way we'll can use a larger (50ms) period_size whenever a device
supports it, and automatically pick smaller ones if we can't set
larger ones.
This removes the calculation borrowed from libao (svn) as well.
Other minor things:
"Alsa" => "ALSA" in error messages
_US appended to *_TIME constants so we won't get confused
(shank's request)
git-svn-id: https://svn.musicpd.org/mpd/trunk@4438 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4435 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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to 0ms sounds like complete crap. 50ms is the default that xmms has used for years, so lets just stick with that.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4433 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4423 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4416 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4412 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Add a few new options for indent to try to make
things a bit cleaner
git-svn-id: https://svn.musicpd.org/mpd/trunk@4411 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Indent the entire tree, hopefully we can keep
it indented.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4410 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4404 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4403 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4383 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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ALSA uses a global config structure that's overwritten (and not
free'd) every time one of those functions is called, so we have
to manually call snd_config_update_free_global() to release it.
Hint taken from MEMORY-LEAK in the ALSA source code
git-svn-id: https://svn.musicpd.org/mpd/trunk@4381 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Probably pedantic, but yes, might as well in case we run into
strange platforms where NULL is something strange.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4380 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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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
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Like the ALSA patches, this allows OSS devices to be disconnected
during playback and MPD will be able to reopen and reuse them without
restarting.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4366 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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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
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ALSA support in libao supports configuring of these variables,
and some hardware setups may benefit from having these things
as tweakable.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4363 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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... instead of hard-coding it to a ridiculously high value that
makes bandwidth-starved devices unhappy.
libao (in SVN) does the same thing, and this calculation was indeed
taken from it.
Low-bandwidth USB (1.1) sound devices seem to need this to prevent
underrun / broken pipe errors (during hw setup, no less) from being
triggered.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4362 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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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
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the repository
git-svn-id: https://svn.musicpd.org/mpd/trunk@4348 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Right Way
git-svn-id: https://svn.musicpd.org/mpd/trunk@4342 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4340 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4333 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4321 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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the GPL header where necessary
git-svn-id: https://svn.musicpd.org/mpd/trunk@4317 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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