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.
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).
pause() puts the audio output into pause mode: if supported, it may
perform a special action, which keeps the device open, but does not
play anything. Output plugins like "shout" might want to play silence
during pause, so their clients won't be disconnected. Plugins which
do not support pausing will simply be closed, and have to be reopened
when unpaused.
This pach includes an implementation for the shout plugin, which
sends silence chunks.
Seeing the "mpd_" prefix _everywhere_ is mind-numbing as the
mind needs to retrain itself to skip over the first 4 tokens of
a type to get to its meaning. So avoid having extra characters
on my terminal to make it easier to follow code at 2:30 am in
the morning.
Please report any new issues you may come across on Free
toolchains. I realize how difficult it can be to build/maintain
cross-compiling toolchains and I have no intention of forcing
people to upgrade their toolchains to build mpd.
Tested with gcc 2.95.4 and and gcc 4.3.1 on x86-32.
During debugging, I found a deadlock between flushAudioBuffer() and
the audio_output_task(): audio_output_task() didn't notice that there
is a command, and flushAudioBuffer() waited forever in notify_wait().
I am not sure yet what is the real cause; work around this for now by
waking up non-finished audio outputs in every iteration.
To check whether a device is really on or off, we should rather check
audio_output.open, instead of managing another variable. Wrap
audio_output.open in the inline function audio_output_is_open() and
use it instead of DEVICE_ON and DEVICE_OFF.
Send an output buffer to all output plugins at the same time, instead
of waiting for each of them separately. Make several functions
non-blocking, and introduce the new function audio_output_wait_all()
to synchronize with all audio output threads.
We have eliminated direct accesses to the audio_output struct from
the all output plugins. Make it opaque for them, and move its real
declaration to output_internal.h, similar to decoder_internal.h.
Pass the opaque structure to plugin.init() only, which will return the
plugin's data pointer on success, and NULL on failure. This data
pointer will be passed to all other methods instead of the
audio_output struct.
Save one allocation, since the whole audio_format struct is nearly the
same size as the pointer to it. Check audio_format_defined(af)
instead of af!=NULL.
free(NULL) isn't explicitly forbidden, but isn't exactly good style.
Check the rare case that the audio buffer isn't initialized yet in
closeAudioDevice(). In this case, we also don't have to call
flushAudioBuffer().
To make openAudioDevice() smaller and more readable, move code to a
static function. Also don't use realloc(), since the old value of the
buffer isn't needed anymore, saving a memcpy().
There are too many static variables in audio.c - organize all
properties of the audio buffer in a struct. The current audio format
is also a property of the buffer, since it describes the buffer's
data format.
audio_format_clear() sets an audio_format struct to an cleared
(undefined) state, which is both faster and smaller than memset(0).
audio_format_defined() checks if the audio_format struct actually has
a defined value (i.e. non-zero). Both can be used to avoid pointers
to audio_format, replacing the "NULL" value with an "undefined"
audio_format.
The "!src" check in copyAudioFormat() used to hide bugs - one should
never pass NULL to it. There is one caller which might pass NULL, add
a check in this caller.
Instead of doing mempcy(), we can simply assign the structures, which
looks more natural.
Instead of having to register each output plugin, store them
statically in an array. This eliminates the need for the List library
here, and saves some small allocations during startup.
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
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
Max Kellermann