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* aac: check buffer lengthsMax Kellermann2008-08-261-2/+3
| | | | | The AAC plugin sometimes does not check the length of available data when checking for magic prefixes. Add length checks.
* aac: use fillAacBuffer() instead of manual readingMax Kellermann2008-08-261-16/+4
| | | | Eliminate some duplicated code by using fillAacBuffer().
* find AAC framesMax Kellermann2008-08-261-1/+35
| | | | | Find AAC frames in the input and skip invalid data. This prepares AAC streaming.
* aac: moved code to adts_check_frame()Max Kellermann2008-08-261-11/+20
| | | | | adts_check_frame() checks whether the buffer head is an AAC frame, and returns the frame length.
* aac: moved code to aac_buffer_shift()Max Kellermann2008-08-261-7/+14
| | | | | | Shifting from the buffer queue is a common operation, and should be provided as a separate function. Move code to aac_buffer_shift() and add a bunch of assertions.
* aac: use inputStreamAtEOF()Max Kellermann2008-08-261-5/+4
| | | | | | | | | When checking for EOF, we should not check whether the read request has been fully satisified. The InputStream API does not guarantee that readFromInputStream() always fills the whole buffer, if EOF is not reached. Since there is the function inputStreamAtEOF() dedicated for this purpose, we should use it for EOF checking after readFromInputStream()==0.
* aac: don't depend on consumed data in fillAacBuffer()Max Kellermann2008-08-261-6/+10
| | | | | | Fill the AacBuffer even when nothing has been consumed yet. The function should not check for consumed data, but for free space at the end of the buffer.
* aac: simplified fillAacBuffer()Max Kellermann2008-08-261-33/+25
| | | | | Return instead of putting all the code into a if-closure. That saves one level of indentation.
* aac: make adtsParse() voidMax Kellermann2008-08-261-3/+1
| | | | | adtsParse() always returns 1, and its caller does not use the return value.
* aac: use size_tMax Kellermann2008-08-261-6/+6
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* aac: removed unused initAacBuffer() parametersMax Kellermann2008-08-261-9/+3
| | | | | Since we eliminated the parameters retFileread and retTagsize in all callers, we can now safely remove it from the function prototype.
* eliminate unused variables in the AAC decoderMax Kellermann2008-08-261-10/+2
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* renamed InputPlugin to struct decoder_pluginMax Kellermann2008-08-261-2/+2
| | | | | | "decoder plugin" is a better name than "input plugin", since the plugin does not actually do the input - InputStream does. Also don't use typedef, so we can forward-declare it if required.
* use a local "initialized" flag instead of dc->stateMax Kellermann2008-08-261-2/+4
| | | | | | Since we want to hide mpd internals from the decoder plugins, the plugins should not check dc->state whether they have already called decoder_initialized(). Use a local variable to track that.
* added decoder_seek_where() and decoder_seek_error()Max Kellermann2008-08-261-7/+5
| | | | | Provide access to seeking for the decoder plugins; they have to know where to seek, and they need a way to tell us that seeking has failed.
* added decoder_command_finished() to decoder_api.hMax Kellermann2008-08-261-2/+2
| | | | | | | Some decoder commands are implemented in the decoder plugins, thus they need to have an API call to signal that their current command has been finished. Let them use the new decoder_command_finished() instead of the internal dc_command_finished().
* added decoder_get_command()Max Kellermann2008-08-261-3/+3
| | | | | | Another big patch which hides internal mpd APIs from decoder plugins: decoder plugins regularly poll dc->command; expose it with a decoder_api.h function.
* removed local variable "eof" because it is unusedMax Kellermann2008-08-261-9/+4
| | | | | "break" is so much easier than "eof=1; continue;", when "!eof" is the loop condition.
* added parameter total_time to decoder_initialized()Max Kellermann2008-08-261-3/+2
| | | | | Similar to the previous patch: pass total_time instead of manipulating dc->totalTime directly.
* added audio_format parameter to decoder_initialized()Max Kellermann2008-08-261-6/+5
| | | | | | dc->audioFormat is set once by the decoder plugins before invoking decoder_initialized(); hide dc->audioFormat and let the decoder pass an AudioFormat pointer to decoder_initialized().
* added decoder_clear() and decoder_flush()Max Kellermann2008-08-261-1/+1
| | | | | | We are now beginning to remove direct structure accesses from the decoder plugins. decoder_clear() and decoder_flush() mask two very common buffer functions.
* added decoder_data()Max Kellermann2008-08-261-3/+3
| | | | | Moved all of the player-waiting code to decoder_data(), to make OutputBuffer more generic.
* added decoder_initialized()Max Kellermann2008-08-261-3/+3
| | | | | | | decoder_initialized() sets the state to DECODE_STATE_DECODE and wakes up the player thread. It is called by the decoder plugin after its internal initialization is finished. More arguments will be added later to prevent direct accesses to the DecoderControl struct.
* added struct decoderMax Kellermann2008-08-261-1/+1
| | | | | | The decoder struct should later be made opaque to the decoder plugin, because maintaining a stable struct ABI is quite difficult. The ABI should only consist of a small number of stable functions.
* added dc_command_finished()Max Kellermann2008-08-261-4/+2
| | | | | | | | dc_command_finished() is invoked by the decoder thread when it has finished a command (sent by the player thread). It resets dc.command and wakes up the player thread. This combination was used at a lot of places, and by introducing this function, the code will be more readable.
* merged start, stop, seek into DecoderControl.commandMax Kellermann2008-08-261-7/+7
| | | | | | | Much of the existing code queries all three variables sequentially. Since only one of them can be set at a time, this can be optimized and unified by merging all of them into one enum variable. Later, the "command" checks can be expressed in a "switch" statement.
* clean up CPP includesMax Kellermann2008-08-261-4/+0
| | | | | Include only headers which are really required. This speeds up compilation and helps detect cross-layer accesses.
* Make the OutputBuffer API more consistentEric Wong2008-04-131-3/+3
| | | | | | | | | | We had functions names varied between outputBufferFoo, fooOutputBuffer, and output_buffer_foo That was too confusing for my little brain to handle. And the global variable was somehow named 'cb' instead of the more obvious 'ob'... git-svn-id: https://svn.musicpd.org/mpd/trunk@7355 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Stop passing our single OutputBuffer object everywhereEric Wong2008-04-131-4/+4
| | | | | | | All of our main singleton data structures are implicitly shared, so there's no reason to keep passing them around and around in the stack and making our internal API harder to deal with. git-svn-id: https://svn.musicpd.org/mpd/trunk@7354 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Stop passing our single DecoderControl object everywhereEric Wong2008-04-131-17/+17
| | | | | | | This at least makes the argument list to a lot of our plugin functions shorter and removes a good amount of line nois^W^Wcode, hopefully making things easier to read and follow. git-svn-id: https://svn.musicpd.org/mpd/trunk@7353 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Fix a few more warnings from -WshadowEric Wong2008-04-121-12/+11
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@7300 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* constant pointersMax Kellermann2008-04-121-2/+2
| | | | | | There were some const pointers missing in the previous const-cleanup patch. git-svn-id: https://svn.musicpd.org/mpd/trunk@7290 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* fix parameter types in the faad callsMax Kellermann2008-04-121-2/+2
| | | | | | libfaad wants uint32_t pointers. Passing a long pointer is bugged on amd64. git-svn-id: https://svn.musicpd.org/mpd/trunk@7289 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Initial cut of fork() => pthreads() for decoder and playerEric Wong2008-04-121-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* 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
* Simplify decode cleanup logic a bitEric Wong2008-01-011-8/+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
* Add MIME types for the aac and mp4 input plugins. Note that these won'tJ. Alexander Treuman2007-06-041-3/+4
| | | | | | have any effect until the aac and mp4 input plugins actually support a stream decoding API. git-svn-id: https://svn.musicpd.org/mpd/trunk@6481 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
* jack patch from anarch (and some type fixes for mp4 and acc plugins)Warren Dukes2006-10-181-2/+2
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4912 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Replace strdup and {c,re,m}alloc with x* variants to check for OOM errorsEric Wong2006-08-261-1/+1
| | | | | | | | | | | | | | | | | | | 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
* Fix an esoteric gcc warningJ. Alexander Treuman2006-08-251-3/+4
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4684 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* This fixes 5 potential bugs where the conditional would always be true.Avuton Olrich2006-08-201-2/+2
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@4659 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 Olrich2006-07-201-165/+203
| | | | | | 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
* inputPlugins/*_plugin.c: static-ficationEric Wong2006-07-171-9/+9
| | | | | | 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
* 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
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