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* alsa: moved code to alsa_configure()Max Kellermann2008-10-121-13/+18
| | | | | Move code which loads configuration to alsa_configure(). This removes one indent level.
* alsa: don't override libasound's buffer_time and period_timeMax Kellermann2008-10-111-20/+19
| | | | | ALSA does a good job measuring its buffer_time and period_time. Don't override its defaults, unless the user demands it.
* alsa: re-enable blocking modeMax Kellermann2008-10-111-15/+2
| | | | | | | | Revert e4f5d6bd "re-enable-nonblocking, but sleep if busy". Non-blocking mode with manual sleeping doesn't help at all (by the way, the patch should have used snd_pcm_wait() instead of my_usleep()). ALSA knows much more about the hardware quirks, so we just let it do the job.
* audio_format: added audio_format_frame_size()Max Kellermann2008-10-101-1/+1
| | | | | | | 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).
* audio_format: renamed sampleRate to sample_rateMax Kellermann2008-10-101-8/+8
| | | | | The last bit of CamelCase in audio_format.h. Additionally, rename a bunch of local variables.
* audio_format: unsigned integersMax Kellermann2008-10-101-4/+4
| | | | "bits" and "channels" cannot be negative.
* use C99 struct initializersMax Kellermann2008-09-291-9/+8
| | | | | | The old struct initializers are error prone and don't allow moving elements around. Since we are going to overhaul some of the APIs soon, it's easier to have all implementations use C99 initializers.
* Switch to C99 types (retaining compat with old compilers)Eric Wong2008-09-291-1/+1
| | | | | | | | | | | | | | | 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.
* output: make "struct audio_output" opaque for output pluginsMax Kellermann2008-09-241-18/+15
| | | | | | | | | | | 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.
* output: set audio_output->open=1 in audio_output_task()Max Kellermann2008-09-241-5/+0
| | | | | Since the output plugin returns a value indicating success or error, we can have the output core code assign the "open" flag.
* output: pass audio_format to plugin.init() and plugin.open()Max Kellermann2008-09-241-2/+3
| | | | | | | Pass the globally configured audio_format as a const pointer to plugin.init(). plugin.open() gets a writable pointer which contains the audio_format requested by the plugin. Its initial value is either the configured audio_format or the input file's audio_format.
* audio_format: added audio_format_sample_size()Max Kellermann2008-09-231-1/+1
| | | | | | The inline function audio_format_sample_size() calculates how many bytes each sample consumes. This function already takes into account that 24 bit samples are 4 bytes long, not 3.
* alsa: re-enable-nonblocking, but sleep if busyEric Wong2008-09-231-7/+10
| | | | | | | Instead of letting ALSA block for us (and potentially allowing something stupid on certain hardware or drivers), we do the sleeping ourselves. We calculate the sleep to be a fraction of period_time to avoid oversleeping (and thus audible skipping).
* alsa: use blocking instead of non-blocking writeEric Wong2008-09-091-1/+6
| | | | | | | | | | | | The way we used non-blocking mode was HORRIBLE. It was non-blocking to ALSA, but we end up blocking in a busy loop that does absolutely NOTHING but retry. We don't check for playback cancellation (like we do in decoders) or anything. This is seriously broken and I can imagine it affects people on fast CPUs more because we do asynchronous output buffering and our ALSA device will always have data ready.
* alsa: snd_pcm_sw_params_set_xfer_align is deprecatedEric Wong2008-09-081-5/+0
| | | | | Lets not use deprecated functions. It's apparently possible to not care about the sw_params stuff at all!
* alsa: only run snd_config_update_free_global once atexitEric Wong2008-09-081-3/+7
| | | | | | | | | This is safer than the patch in http://www.musicpd.org/mantis/view.php?id=1542 with multiple audio outputs enabled. Sadly, I only noticed that patch/problem when I googled for "snd_config_update_free_global"
* alsa: move bitformat reading code out of the wayEric Wong2008-09-081-16/+12
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* alsa: avoid unnecessary heap usage if we don't set a device nameEric Wong2008-09-081-11/+12
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* alsa: get rid of the needless canPause flagEric Wong2008-09-081-3/+0
| | | | | We never use it for anything anyways as we release the device entirely on pause.
* alsa: capitalize "ALSA" consistently in messagesEric Wong2008-09-081-8/+8
| | | | That's the name of this project.
* alsa: optimistically try resuming from suspendEric Wong2008-09-081-6/+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]
* output: const plugin structuresMax Kellermann2008-09-081-1/+1
| | | | | Since the plugin struct is never modified, we should store it in constant locations.
* 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
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