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Instead of letting ALSA block for us (and potentially allowing
something stupid on certain hardware or drivers), we do the
sleeping ourselves. We calculate the sleep to be a fraction of
period_time to avoid oversleeping (and thus audible skipping).
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The way we used non-blocking mode was HORRIBLE.
It was non-blocking to ALSA, but we end up blocking in a busy
loop that does absolutely NOTHING but retry. We don't check
for playback cancellation (like we do in decoders) or anything.
This is seriously broken and I can imagine it affects people on
fast CPUs more because we do asynchronous output buffering and
our ALSA device will always have data ready.
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Print out {buffer,period}_{size,time}. Not sure if this
is going to help. I've been searching everywhere looking
for a possible clue as to what's causing the high CPU
usage problems...
Also, add device information to some messages I missed earlier.
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Lets not use deprecated functions. It's apparently
possible to not care about the sw_params stuff at all!
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This is safer than the patch in
http://www.musicpd.org/mantis/view.php?id=1542
with multiple audio outputs enabled.
Sadly, I only noticed that patch/problem when I googled for
"snd_config_update_free_global"
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We never use it for anything anyways as we release the device
entirely on pause.
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This saves me precious terminal space
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That's the name of this project.
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Apparently snd_pcm_hw_params_can_resume() can return false even
though my hardware does in fact support resuming. So stop
carrying that value in the canResume flag and just try to resume
when we're in the suspended state; falling back to
snd_pcm_prepare only if resuming fails. libao does something
similar on resume, too.
While we're at it, use the E() macro which will enable us to
have better error reporting.
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Hibernating my laptop while MPD is playing results in ugliness
about "alsa device foo was suspend" constantly printed to the
logs.
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Given the length of the ALSA command names, I only want
to see them once per-section of code, if at all...
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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
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Clean up some space indentations, replace with tabs.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7239 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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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
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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
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git-svn-id: https://svn.musicpd.org/mpd/trunk@7077 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@6858 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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state (when users press stop, previous snd_pcm_drop(), then
snd_pcm_drain() was called. this would lockup dmix)
git-svn-id: https://svn.musicpd.org/mpd/trunk@6517 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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or no" parameters.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5896 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5834 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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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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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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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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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@4416 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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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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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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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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breaking the alsa rate plugin, and dmix seems to work fine without it. Thanks to Skee from #mpd for testing.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4269 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@3726 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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audioOutput_osx.c, aac_decode.c, mp4_decode.c have NOT been thoroughly
checked, but I nevertheless managed to eyeball and fix one
incompatibility in audioOutput_osx.c
All other files have been build successfully with gcc 2.95
git-svn-id: https://svn.musicpd.org/mpd/trunk@3688 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@3104 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@3094 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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