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* output: renamed typedef AudioOutput to struct audio_outputMax Kellermann2008-09-071-7/+8
| | | | | Also rename AudioOutputPlugin to struct audio_output_plugin, and use forward declarations to reduce include dependencies.
* output: added output_api.hMax Kellermann2008-09-071-3/+2
| | | | | Just like decoder_api.h, output_api.h provides the audio output API which is used by the plugins.
* audio_format: converted typedef AudioFormat to struct audio_formatMax Kellermann2008-09-071-1/+1
| | | | | Get rid of CamelCase, and don't use a typedef, so we can forward-declare it, and unclutter the include dependencies.
* use size_t and constant pointer in ao pluginsMax Kellermann2008-04-121-1/+2
| | | | | | | 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
* whitespace cleanupMax Kellermann2008-04-121-7/+7
| | | | | Clean up some space indentations, replace with tabs. git-svn-id: https://svn.musicpd.org/mpd/trunk@7239 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* explicitly downcastMax Kellermann2008-03-261-1/+1
| | | | | | | | 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
* Cleanup #includes of standard system headers and put them in one placeEric Wong2008-01-031-4/+1
| | | | | | | | | | | | | 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
* Know about SND_PCM_STATE_RUNNING, might fix some bugsQball Cow2007-12-161-0/+4
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@7077 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* conf: use getBoolBlockParam for block params, tooEric Wong2007-09-051-3/+3
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@6858 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* dmix fix, don't call snd_pcm_drain unless we're already in the RUNNINGWarren Dukes2007-06-081-1/+3
| | | | | | | 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
* Don't allow "true" as a value for use_mmap for consistency with other "yesJ. Alexander Treuman2007-04-091-2/+1
| | | | | or no" parameters. git-svn-id: https://svn.musicpd.org/mpd/trunk@5896 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* The massive copyright updateAvuton Olrich2007-04-051-1/+1
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@5834 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 Dukes2006-10-181-3/+0
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4913 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Replace strdup and {c,re,m}alloc with x* variants to check for OOM errorsEric Wong2006-08-261-3/+3
| | | | | | | | | | | | | | | | | | | 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
* trivial: labels should be on the left-most column, no tabbingEric Wong2006-08-201-2/+2
| | | | | | | | | | | | 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
* 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
* 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
* 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
* Add mpd-indent.shAvuton Olrich2006-07-201-3/+3
| | | | | | 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 Olrich2006-07-201-117/+134
| | | | | | 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
* 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 Wong2006-07-171-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* 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
* Change shank's email addressJ. Alexander Treuman2006-07-141-1/+1
| | | 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 Treuman2006-07-141-14/+2
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4321 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Huge header update, update the copyright and addAvuton Olrich2006-07-131-1/+1
| | | | | the GPL header where necessary git-svn-id: https://svn.musicpd.org/mpd/trunk@4317 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
* 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
* gcc 2.95 fixesEric Wong2005-11-191-2/+3
| | | | | | | | | | 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
* print out bits in debug message output for OSS and ALSAWarren Dukes2005-03-191-2/+3
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3104 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* fix for dmixWarren Dukes2005-03-171-2/+1
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3094 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* slight changes to alsa errorsWarren Dukes2005-03-131-3/+32
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3072 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* if no audioOutput specified, we no attempt to detect if there exists a ↵Warren Dukes2005-03-121-1/+23
| | | | | usable oss or alsa device git-svn-id: https://svn.musicpd.org/mpd/trunk@3057 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* we ne allow audioOutput plugins to set the final outAudioFormat that will be ↵Warren Dukes2005-03-081-4/+10
| | | | | used. we now use alsa's _near functions to detect what to use. git-svn-id: https://svn.musicpd.org/mpd/trunk@3038 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* now player and decoder processes should only exit() when receiving term ↵Warren Dukes2005-03-081-1/+1
| | | | | signal from their respective parent processes git-svn-id: https://svn.musicpd.org/mpd/trunk@3034 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* don't need the extra snd_pcm_prepare after _dropWarren Dukes2005-03-051-1/+0
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3020 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* more alsa workWarren Dukes2005-03-051-41/+57
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3019 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* this seemed to help a bit with the blip's on nextWarren Dukes2005-03-051-0/+1
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3015 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* implemented dropping of current buffered audio, works for oss, but there seemsWarren Dukes2005-03-051-10/+29
| | | | | to be a "blip" for alsa devices, needs more work git-svn-id: https://svn.musicpd.org/mpd/trunk@3011 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* now alsa plugin should workWarren Dukes2005-03-051-7/+13
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3009 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* implemented alsa audioOutput plugin, now it needs testingWarren Dukes2005-03-051-0/+293
git-svn-id: https://svn.musicpd.org/mpd/trunk@3008 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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