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* Don't initialize globals to zero (or NULL)Eric Wong2007-01-142-2/+2
| | | | | | | | 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
* jack: fix double-free in finish routineEric Wong2007-01-141-3/+12
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@5252 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* jack: fix potential segfaults in parsing bad configs for 'ports'Eric Wong2007-01-141-5/+21
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@5251 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* jack: fix multiple segfaults when jackd hasn't been startedEric Wong2007-01-141-8/+17
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@5250 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* jack: strdup => xstrdup for error checkingEric Wong2007-01-141-4/+4
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@5249 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* jack: declare global variables as staticEric Wong2007-01-141-3/+3
| | | | | | | 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
* And a fix for the warnings in the last patch for JACKAvuton Olrich2006-12-301-4/+5
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@5192 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* A JACK patch to fix some old issues, from author.Avuton Olrich2006-12-301-170/+189
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@5191 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* audioOutput_alsa: print out the bitrate we wanted to setEric Wong2006-11-071-1/+1
| | | | | ..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
* remove some unneccesary includes from the audioOutput'sWarren Dukes2006-10-187-21/+0
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4913 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* jack patch from anarch (and some type fixes for mp4 and acc plugins)Warren Dukes2006-10-181-0/+380
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4912 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* audioOutput_shout: use shout_set_nonblockingEric Wong2006-10-111-3/+17
| | | | | | | | | | | | 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
* Allow an ogg quality of -1 to be specified.J. Alexander Treuman2006-10-101-5/+5
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4893 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Hopefully fix endian problem mac x86/ppcQball Cow2006-09-091-2/+5
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4750 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Replace strdup and {c,re,m}alloc with x* variants to check for OOM errorsEric Wong2006-08-267-18/+18
| | | | | | | | | | | | | | | | | | | 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
* audioOutputs_oss: [trivial] make loop interation consistent with othersEric Wong2006-08-261-1/+1
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4689 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* trivial: labels should be on the left-most column, no tabbingEric Wong2006-08-202-3/+3
| | | | | | | | | | | | 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
* audioOutput_alsa.c: avoid changing our internal period and buffer time valuesEric Wong2006-08-121-6/+9
| | | | | | | | 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
* Spelling & GrammarAvuton Olrich2006-08-112-4/+4
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4612 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* audioOutput_pulse: ansi-fy function declarations (sparse)Eric Wong2006-08-081-2/+2
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4599 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* more sparse cleanupsEric Wong2006-08-082-3/+3
| | | | | | | | | * 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
* audioOutput_oss: cleanups (stolen from -ke)Eric Wong2006-08-011-73/+58
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4518 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Remove the fifo plugin. It's currently useless for the average user, and ↵J. Alexander Treuman2006-07-261-251/+0
| | | | | 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
* audioOutput_alsa: better period_size auto-configurationEric Wong2006-07-241-13/+20
| | | | | | | | | | | | | | | | | 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
* Renaming stat struct to st, for consistency with the rest of mpdJ. Alexander Treuman2006-07-231-5/+5
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4435 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* chang the default period_time to 50ms. On my setup, setting the period_time ↵Warren Dukes2006-07-231-1/+4
| | | | | 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
* Adding fifo output pluginJ. Alexander Treuman2006-07-221-0/+251
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4423 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* audioOutput_alsa: oops, I broke autodetection in r4363, fixedEric Wong2006-07-211-1/+2
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4416 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* #include <string.h> in PulseAudio output for correctnessJ. Alexander Treuman2006-07-201-0/+1
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4412 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Add mpd-indent.shAvuton Olrich2006-07-206-18/+18
| | | | | | 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
* Add mpd-indent.shAvuton Olrich2006-07-206-716/+796
| | | | | | 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
* s/ad/pd/ in the PluseAudio plugin (I forgot to rename when copying from alsa)J. Alexander Treuman2006-07-191-31/+31
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4404 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Throttle PuleAudio connection attempts so we don't spam the error logJ. Alexander Treuman2006-07-191-5/+24
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4403 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* audioOutput_mvp: remove unused variableEric Wong2006-07-171-1/+0
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4383 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* alsa: fix memory leaks from snd_*_open*()Eric Wong2006-07-171-0/+2
| | | | | | | | | 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
* sparse: replace 0 (integer) usage with NULL where appropriateEric Wong2006-07-171-4/+5
| | | | | | 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
* sparse: ANSI-fy function declarationsEric Wong2006-07-173-5/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* OSS: handle device disconnects and reconnects (w/o needing a mpd restart)Eric Wong2006-07-161-10/+11
| | | | | | | 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
* audio: attempt to gracefully handle disconnected/reconnected devicesEric Wong2006-07-161-0/+5
| | | | | | | | | | | 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
* audioOutput_alsa: add use_mmap, period_time, buffer_time optionsEric Wong2006-07-161-15/+23
| | | | | | | 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
* audioOutput_alsa: calculate period size from sample rateEric Wong2006-07-161-2/+5
| | | | | | | | | | | | | ... 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
* De-inline non-trivial, non-performance-critical functionsEric Wong2006-07-151-1/+1
| | | | | | 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
* [CLEANUP] Fix indentation to be like the rest ofAvuton Olrich2006-07-151-126/+130
| | | | | the repository git-svn-id: https://svn.musicpd.org/mpd/trunk@4348 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Use audio_output { name } for the stream name in PulseAudio, but do it The ↵J. Alexander Treuman2006-07-141-7/+2
| | | | | Right Way git-svn-id: https://svn.musicpd.org/mpd/trunk@4342 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Use audio_output { name } for the stream name in PulseAudioJ. Alexander Treuman2006-07-141-4/+8
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4340 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Change shank's email addressJ. Alexander Treuman2006-07-147-7/+7
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4333 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Use a macro to declare disabled audio output pluginsJ. Alexander Treuman2006-07-147-93/+13
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4321 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Huge header update, update the copyright and addAvuton Olrich2006-07-137-6/+24
| | | | | the GPL header where necessary git-svn-id: https://svn.musicpd.org/mpd/trunk@4317 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Add PulseAudio supportJ. Alexander Treuman2006-07-131-0/+193
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4316 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* OSS: correctly check for the device in oss_testDefault()Eric Wong2006-07-121-3/+4
| | | | | | | | | open(2) returns -1 on error (if the device does not exist), and -1 is true. Also, put shank's name in the copyright header since half the code is his (including this bug :P). git-svn-id: https://svn.musicpd.org/mpd/trunk@4310 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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