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* 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
* Remove unnecessary include, has always been pulledAvuton Olrich2006-07-051-1/+0
| | | | | in by vorbisenc.h git-svn-id: https://svn.musicpd.org/mpd/trunk@4298 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Reverting patch to "fix" the alsa plugin when used with dmix. It ended up ↵J. Alexander Treuman2006-06-111-1/+2
| | | | | 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
* src/audioOutputs/audioOutput_oss.c: fix for big-endian machinesEric Wong2006-03-251-15/+19
| | | | | | | Patch by Qball. Signed-off-by: Eric Wong <normalperson@yhbt.net> git-svn-id: https://svn.musicpd.org/mpd/trunk@3935 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Hopefully the last of the spelling fixes :>Eric Wong2006-03-161-2/+2
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3923 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* potential fix for bug #466Warren Dukes2005-12-121-6/+6
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3726 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* remove C++ style commentsEric Wong2005-11-193-19/+21
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3689 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* gcc 2.95 fixesEric Wong2005-11-194-5/+8
| | | | | | | | | | 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
* Fixed (tested in Fink and DarwinPorts) osX output pluginQball Cow2005-10-021-2/+2
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3500 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* patch from Oliver Logghe for Hauppage Media MVP supportWarren Dukes2005-08-111-0/+296
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3432 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* fix warning reported gcc 4.0: x == y == z does not work as intendedEric Wong2005-08-071-1/+2
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3422 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* don't check errno unless we have an error (ret<0) from write()Eric Wong2005-07-031-1/+1
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3382 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* fix OSS audioOutput plugin, so that when dropAudioBuffer is called and the ↵Warren Dukes2005-04-061-5/+10
| | | | | oss device is closed, a subsequent call to playAudio will reopen the device git-svn-id: https://svn.musicpd.org/mpd/trunk@3198 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* destroy mutex's and condition'sWarren Dukes2005-03-241-0/+2
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3132 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* revert to r3105Warren Dukes2005-03-241-35/+59
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3131 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* fix compilation error in os x pluginWarren Dukes2005-03-231-3/+3
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3115 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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