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* Initial cut of fork() => pthreads() for decoder and playerEric Wong2008-04-121-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | I initially started to do a heavy rewrite that changed the way processes communicated, but that was too much to do at once. So this change only focuses on replacing the player and decode processes with threads and using condition variables instead of polling in loops; so the changeset itself is quiet small. * The shared output buffer variables will still need locking to guard against race conditions. So in this effect, we're probably just as buggy as before. The reduced context-switching overhead of using threads instead of processes may even make bugs show up more or less often... * Basic functionality appears to be working for playing local (and NFS) audio, including: play, pause, stop, seek, previous, next, and main playlist editing * I haven't tested HTTP streams yet, they should work. * I've only tested ALSA and Icecast. ALSA works fine, Icecast metadata seems to get screwy at times and breaks song advancement in the playlist at times. * state file loading works, too (after some last-minute hacks with non-blocking wakeup functions) * The non-blocking (*_nb) variants of the task management functions are probably overused. They're more lenient and easier to use because much of our code is still based on our previous polling-based system. * It currently segfaults on exit. I haven't paid much attention to the exit/signal-handling routines other than ensuring it compiles. At least the state file seems to work. We don't do any cleanups of the threads on exit, yet. * Update is still done in a child process and not in a thread. To do this in a thread, we'll need to ensure it does proper locking and communication with the main thread; but should require less memory in the end because we'll be updating the database "in-place" rather than updating a copy and then bulk-loading when done. * We're more sensitive to bugs in 3rd party libraries now. My plan is to eventually use a master process which forks() and restarts the child when it dies: locking and communication with the main thread; but should require less memory in the end because we'll be updating the database "in-place" rather than updating a copy and then bulk-loading when done. * We're more sensitive to bugs in 3rd party libraries now. My plan is to eventually use a master process which forks() and restarts the child when it dies: master - just does waitpid() + fork() in a loop \- main thread \- decoder thread \- player thread At the beginning of every song, the main thread will set a dirty flag and update the state file. This way, if we encounter a song that triggers a segfault killing the main thread, the master will start the replacement main on the next song. * The main thread still wakes up every second on select() to check for signals; which affects power management. [merged r7138 from branches/ew] git-svn-id: https://svn.musicpd.org/mpd/trunk@7240 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* audiofile_plugin: fix nasty long lines introduced in previous commitEric Wong2008-03-261-2/+4
| | | | | Terminals are 80 columns and that's a hard limit, no exceptions git-svn-id: https://svn.musicpd.org/mpd/trunk@7207 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* explicitly downcastMax Kellermann2008-03-261-4/+4
| | | | | | | | 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
* fix -Wconst warningsMax Kellermann2008-02-051-1/+1
| | | | | | [ew: cleaned up the dirty union hack a bit] Signed-off-by: Eric Wong <normalperson@yhbt.net> git-svn-id: https://svn.musicpd.org/mpd/trunk@7180 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* fixed -Wshadow warningsMax Kellermann2008-01-261-6/+6
| | | | | Signed-off-by: Eric Wong <normalperson@yhbt.net> git-svn-id: https://svn.musicpd.org/mpd/trunk@7143 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Cleanup #includes of standard system headers and put them in one placeEric Wong2008-01-031-7/+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
* Simplify decode cleanup logic a bitEric Wong2008-01-011-11/+0
| | | | | | | | | | | DECODE_STATE_STOP is always set as dc->state, and dc->stop is always cleared. So handle it in decodeStart once rather than doing it in every plugin. While we're at it, fix a long-standing (but difficult to trigger) bug in mpc_decode where we failed to return if mpc_decoder_initialize() fails. git-svn-id: https://svn.musicpd.org/mpd/trunk@7122 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
* Don't initialize globals to zero (or NULL)Eric Wong2007-01-141-12/+1
| | | | | | | | 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
* audiofile_plugin: use afSetVirtualSampleFormat, tooEric Wong2006-08-241-0/+2
| | | | | | | | | | | | | | | | | | This finally fixes a bug from over two years ago playing a wave file (oprah.wav) with the following characteristics (from sfinfo): File Format Microsoft RIFF WAVE Format (wave) Data Format 8-bit integer (unsigned, little endian) Audio Data 986827 bytes begins at offset 58 (3a hex) 1 channel, 986827 frames Sampling Rate 22050.00 Hz Duration 44.754 seconds Of course, this has been regression tested with all the files that the previous commit got working. Thanks to Michael Pruett (audiofile author) for the hint and shame on me for forgetting about it for over two years :x git-svn-id: https://svn.musicpd.org/mpd/trunk@4682 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* audiofile_plugin: fix for playing mono .au files with 8000Hz sample rateEric Wong2006-08-241-3/+3
| | | | | | | | | | | | | | | | Use the 'Virtual' variants of afGetSampleFormat, afGetChannels, afGetVirtualFrameSize in the audiofile library, since it already does the necessary abstraction for us. Of course, I've regression tested these changes against my standard 44100Hz/2ch/16bit wave files and they continue to play fine. Files tested: english.au (Linus Torvalds pronouncing 'Linux' in English) B01.Red_Bright_Heart.au (Chinese opera, sounds correct to me even though I don't actually understand the words) git-svn-id: https://svn.musicpd.org/mpd/trunk@4681 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Add mpd-indent.shAvuton Olrich2006-07-201-1/+1
| | | | | | 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 Olrich2006-07-201-72/+79
| | | | | | 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
* some quick hacks to avoid signedness warnings with gcc4Warren Dukes2006-07-171-1/+1
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4387 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* inputPlugins/*_plugin.c: static-ficationEric Wong2006-07-171-4/+4
| | | | | | Nothing here is ever exported for linkage besides the InputPlugin structure, so mark them static to save a few bytes. git-svn-id: https://svn.musicpd.org/mpd/trunk@4382 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
* 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
* Add 'aif' as an extension with libaudiofile.Avuton Olrich2006-05-031-1/+1
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4132 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* merge with mpd/trunk up to r3925Eric Wong2006-03-161-0/+2
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@3926 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Patch to make the configure flag for mpd-mad and mpd-libid3tag more logic ↵Qball Cow2005-09-081-0/+3
| | | | | (from ticho) git-svn-id: https://svn.musicpd.org/mpd/trunk@3477 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* rewrite replaygain code, needs testingWarren Dukes2004-11-021-1/+2
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@2482 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* add "aiff" and "au" as suffixes that audiofile can handleWarren Dukes2004-06-011-1/+1
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@1275 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* some stream metadata fixesWarren Dukes2004-05-311-0/+4
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@1266 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* icynames are now copied to title of streamsWarren Dukes2004-05-311-6/+6
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@1258 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* audiofile_pluginWarren Dukes2004-05-311-0/+180
git-svn-id: https://svn.musicpd.org/mpd/trunk@1248 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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