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2008-09-08alsa: optimistically try resuming from suspendEric Wong1-6/+4
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. [mk: remove the E() macro stuff]
2008-09-08output: const plugin structuresMax Kellermann1-1/+1
Since the plugin struct is never modified, we should store it in constant locations.
2008-09-07output: renamed typedef AudioOutput to struct audio_outputMax Kellermann1-7/+8
Also rename AudioOutputPlugin to struct audio_output_plugin, and use forward declarations to reduce include dependencies.
2008-09-07output: added output_api.hMax Kellermann1-3/+2
Just like decoder_api.h, output_api.h provides the audio output API which is used by the plugins.
2008-09-07audio_format: converted typedef AudioFormat to struct audio_formatMax Kellermann1-1/+1
Get rid of CamelCase, and don't use a typedef, so we can forward-declare it, and unclutter the include dependencies.
2008-04-12use size_t and constant pointer in ao pluginsMax Kellermann1-1/+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
2008-04-12whitespace cleanupMax Kellermann1-7/+7
Clean up some space indentations, replace with tabs. git-svn-id: https://svn.musicpd.org/mpd/trunk@7239 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2008-03-26explicitly downcastMax Kellermann1-1/+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
2008-01-03Cleanup #includes of standard system headers and put them in one placeEric Wong1-4/+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
2007-12-16Know about SND_PCM_STATE_RUNNING, might fix some bugsQball Cow1-0/+4
git-svn-id: https://svn.musicpd.org/mpd/trunk@7077 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-09-05conf: use getBoolBlockParam for block params, tooEric Wong1-3/+3
git-svn-id: https://svn.musicpd.org/mpd/trunk@6858 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-06-08dmix fix, don't call snd_pcm_drain unless we're already in the RUNNINGWarren Dukes1-1/+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
2007-04-09Don't allow "true" as a value for use_mmap for consistency with other "yesJ. Alexander Treuman1-2/+1
or no" parameters. git-svn-id: https://svn.musicpd.org/mpd/trunk@5896 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-04-05The massive copyright updateAvuton Olrich1-1/+1
git-svn-id: https://svn.musicpd.org/mpd/trunk@5834 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2006-11-07audioOutput_alsa: print out the bitrate we wanted to setEric Wong1-1/+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
2006-10-18remove some unneccesary includes from the audioOutput'sWarren Dukes1-3/+0
git-svn-id: https://svn.musicpd.org/mpd/trunk@4913 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2006-08-26Replace strdup and {c,re,m}alloc with x* variants to check for OOM errorsEric Wong1-3/+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
2006-08-20trivial: labels should be on the left-most column, no tabbingEric Wong1-2/+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
2006-08-12audioOutput_alsa.c: avoid changing our internal period and buffer time valuesEric Wong1-6/+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
2006-07-24audioOutput_alsa: better period_size auto-configurationEric Wong1-13/+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
2006-07-23chang the default period_time to 50ms. On my setup, setting the period_time ↵Warren Dukes1-1/+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
2006-07-21audioOutput_alsa: oops, I broke autodetection in r4363, fixedEric Wong1-1/+2
git-svn-id: https://svn.musicpd.org/mpd/trunk@4416 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2006-07-20Add mpd-indent.shAvuton Olrich1-3/+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
2006-07-20Add mpd-indent.shAvuton Olrich1-117/+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
2006-07-17alsa: fix memory leaks from snd_*_open*()Eric Wong1-0/+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
2006-07-17sparse: replace 0 (integer) usage with NULL where appropriateEric Wong1-4/+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
2006-07-17sparse: ANSI-fy function declarationsEric Wong1-2/+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
2006-07-16audio: attempt to gracefully handle disconnected/reconnected devicesEric Wong1-0/+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
2006-07-16audioOutput_alsa: add use_mmap, period_time, buffer_time optionsEric Wong1-15/+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
2006-07-16audioOutput_alsa: calculate period size from sample rateEric Wong1-2/+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
2006-07-15De-inline non-trivial, non-performance-critical functionsEric Wong1-1/+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
2006-07-14Change shank's email addressJ. Alexander Treuman1-1/+1
git-svn-id: https://svn.musicpd.org/mpd/trunk@4333 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2006-07-14Use a macro to declare disabled audio output pluginsJ. Alexander Treuman1-14/+2
git-svn-id: https://svn.musicpd.org/mpd/trunk@4321 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2006-07-13Huge header update, update the copyright and addAvuton Olrich1-1/+1
the GPL header where necessary git-svn-id: https://svn.musicpd.org/mpd/trunk@4317 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2006-06-11Reverting patch to "fix" the alsa plugin when used with dmix. It ended up ↵J. Alexander Treuman1-1/+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
2005-12-12potential fix for bug #466Warren Dukes1-6/+6
git-svn-id: https://svn.musicpd.org/mpd/trunk@3726 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2005-11-19gcc 2.95 fixesEric Wong1-2/+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
2005-03-19print out bits in debug message output for OSS and ALSAWarren Dukes1-2/+3
git-svn-id: https://svn.musicpd.org/mpd/trunk@3104 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2005-03-17fix for dmixWarren Dukes1-2/+1
git-svn-id: https://svn.musicpd.org/mpd/trunk@3094 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2005-03-13slight changes to alsa errorsWarren Dukes1-3/+32
git-svn-id: https://svn.musicpd.org/mpd/trunk@3072 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2005-03-12if no audioOutput specified, we no attempt to detect if there exists a ↵Warren Dukes1-1/+23
usable oss or alsa device git-svn-id: https://svn.musicpd.org/mpd/trunk@3057 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2005-03-08we ne allow audioOutput plugins to set the final outAudioFormat that will be ↵Warren Dukes1-4/+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
2005-03-08now player and decoder processes should only exit() when receiving term ↵Warren Dukes1-1/+1
signal from their respective parent processes git-svn-id: https://svn.musicpd.org/mpd/trunk@3034 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2005-03-05don't need the extra snd_pcm_prepare after _dropWarren Dukes1-1/+0
git-svn-id: https://svn.musicpd.org/mpd/trunk@3020 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2005-03-05more alsa workWarren Dukes1-41/+57
git-svn-id: https://svn.musicpd.org/mpd/trunk@3019 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2005-03-05this seemed to help a bit with the blip's on nextWarren Dukes1-0/+1
git-svn-id: https://svn.musicpd.org/mpd/trunk@3015 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2005-03-05implemented dropping of current buffered audio, works for oss, but there seemsWarren Dukes1-10/+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
2005-03-05now alsa plugin should workWarren Dukes1-7/+13
git-svn-id: https://svn.musicpd.org/mpd/trunk@3009 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2005-03-05implemented alsa audioOutput plugin, now it needs testingWarren Dukes1-0/+293
git-svn-id: https://svn.musicpd.org/mpd/trunk@3008 09075e82-0dd4-0310-85a5-a0d7c8717e4f
generated by cgit v1.2.3 (git 2.25.1) at 2024-08-14 15:29:20 +0000