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* renamed src/inputPlugins/ to src/decoder/Max Kellermann2008-10-261-459/+0
| | | | | These plugins are not input plugins, they are decoder plugins. No CamelCase in the directory name.
* Makefile.am: don't compile disabled decoder pluginsMax Kellermann2008-10-171-36/+13
| | | | | Don't compile the sources of disabled decoder plugins at all, and don't attempt to register these.
* audio_format: renamed sampleRate to sample_rateMax Kellermann2008-10-101-2/+2
| | | | | The last bit of CamelCase in audio_format.h. Additionally, rename a bunch of local variables.
* use the "bool" data type instead of "int"Max Kellermann2008-10-081-2/+2
| | | | "bool" should be used in C99 programs for boolean values.
* don't include os_compat.hMax Kellermann2008-10-081-0/+2
| | | | | When there are standardized headers, use these instead of the bloated os_compat.h.
* use C99 struct initializersMax Kellermann2008-09-291-10/+7
| | | | | | 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.
* decoder: renamed plugin methodsMax Kellermann2008-09-291-3/+3
| | | | | Why have a "_func" prefix on all method names? Also don't typedef the methods, there is no advantage in that.
* flac: moved code from flacWrite() to _flac_common.cMax Kellermann2008-09-231-72/+2
| | | | | | | There is still a lot of duplicated code in flac_plugin.c and oggflac_plugin.c. Move code from flac_plugin.c to _flac_common.c, and use the new function flac_common_write() also in oggflac_plugin.c, porting lots of optimizations over to it.
* flac: assume the buffer is empty in flacWrite() IIMax Kellermann2008-09-231-7/+2
| | | | | The previous patch on this topic was incomplete: it still added data->chunk_length when calling flac_convert(). Remove this, too.
* audio_format: added audio_format_sample_size()Max Kellermann2008-09-231-1/+2
| | | | | | 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.
* tag: renamed functions, no CamelCaseMax Kellermann2008-08-291-4/+4
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* tag: renamed MpdTag and MpdTagItem to struct tag, struct mpd_tag_itemMax Kellermann2008-08-291-7/+7
| | | | | Getting rid of CamelCase; not having typedefs also allows us to forward-declare the structures.
* flac: decoder command means EOFMax Kellermann2008-08-261-8/+6
| | | | | | | It was possible for the decoder thread to go into an endless loop (flac and oggflac decoders): when a "STOP" command arrived, the Read() callback would return 0, but the EOF() callback returned false. Fix: when decoder_get_command()!=NONE, return EOF==true.
* mp3, flac: check for seek command after decoder_read()Max Kellermann2008-08-261-3/+6
| | | | | | | When we introduced decoder_read(), we added code which aborts the read operation when a decoder command arrives. Several plugins however did not expect that when they were converted to decoder_read(). Add proper checks to the mp3 and flac decoder plugins.
* check decoder_command!=NONE instead of decoder_command==STOPMax Kellermann2008-08-261-1/+1
| | | | | | The code said "decoder_command==STOP" because that was a conversion from the old "dc->stop" test. As we can now check for all commands in one test, we can simply rewrite that to decoder_command!=NONE.
* added decoder_read()Max Kellermann2008-08-261-8/+1
| | | | | | | | | On our way to stabilize the decoder API, we will one day remove the input stream functions. The most basic function, read() will be provided by decoder_api.h with this patch. It already contains a loop (still with manual polling), error/eof handling and decoder command checks. This kind of code used to be duplicated in all decoder plugins.
* added decoder_plugin_register()Max Kellermann2008-08-261-2/+1
| | | | | | | With the functions decoder_plugin_register() and decoder_plugin_unregister(), decoder plugins can register a "secondary" plugin, like the flac input plugin does this for "oggflac".
* renamed functions in decoder_list.hMax Kellermann2008-08-261-1/+1
| | | | InputPlugin to decoder_plugin, and no camelCase.
* no camel case in struct decoder_pluginMax Kellermann2008-08-261-5/+5
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* renamed inputPlugin.* to decoder_list.*Max Kellermann2008-08-261-1/+1
| | | | | Since inputPlugin.c manages the list of registered decoders, we should rename the source file.
* renamed InputPlugin to struct decoder_pluginMax Kellermann2008-08-261-3/+3
| | | | | | "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.
* added decoder_seek_where() and decoder_seek_error()Max Kellermann2008-08-261-3/+3
| | | | | 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 Kellermann2008-08-261-1/+1
| | | | | | | 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 Kellermann2008-08-261-6/+6
| | | | | | 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.
* moved InputPlugin to decoder_api.hMax Kellermann2008-08-261-0/+1
| | | | | | InputPlugin is the API which is implemented by a decoder plugin. This belongs to the public API/ABI, so move it to decoder_api.h. It will later be renamed to something like "decoder_plugin".
* added parameter total_time to decoder_initialized()Max Kellermann2008-08-261-1/+1
| | | | | Similar to the previous patch: pass total_time instead of manipulating dc->totalTime directly.
* added audio_format parameter to decoder_initialized()Max Kellermann2008-08-261-5/+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 Kellermann2008-08-261-2/+2
| | | | | | 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_initialized()Max Kellermann2008-08-261-1/+1
| | | | | | | 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 Kellermann2008-08-261-6/+7
| | | | | | 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 Kellermann2008-08-261-2/+1
| | | | | | | | 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/+8
| | | | | | | 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 Kellermann2008-08-261-6/+0
| | | | | Include only headers which are really required. This speeds up compilation and helps detect cross-layer accesses.
* enable -Wpointer-arith, -Wstrict-prototypesMax Kellermann2008-08-261-7/+7
| | | | | | Also enable -Wunused-parameter - this forces us to add the gcc "unused" attribute to a lot of parameters (mostly library callback functions), but it's worth it during code refactorizations.
* Make the OutputBuffer API more consistentEric Wong2008-04-131-2/+2
| | | | | | | | | | 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 Wong2008-04-131-9/+8
| | | | | | | 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 Wong2008-04-131-23/+20
| | | | | | | 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
* replaced assertion with checkMax Kellermann2008-04-121-2/+1
| | | | | | | During my tests, it happened that data->position>newPosition. I have not yet fully understood why this can happen; for now, replace this with a run-time check. git-svn-id: https://svn.musicpd.org/mpd/trunk@7334 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* multiply num_samples with bytes_per_channelMax Kellermann2008-04-121-1/+1
| | | | | | | The patch "convert blocks until the buffer is full" did not update data->chunk_length correctly: it added the number of samples, not the number of bytes. Multiply that with bytes_per_channel git-svn-id: https://svn.musicpd.org/mpd/trunk@7332 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* missing num_channels check in previous patchMax Kellermann2008-04-121-1/+1
| | | | | | In the patch "special optimized case for 16bit stereo", the check for "num_channels==2" was missing. git-svn-id: https://svn.musicpd.org/mpd/trunk@7331 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* special optimized case for 16bit stereoMax Kellermann2008-04-121-3/+20
| | | | | | | Not having to loop for every sample byte (depending on a variable unknown at compile time) saves a lot of CPU cycles. We could consider reimplementing this function with liboil... git-svn-id: https://svn.musicpd.org/mpd/trunk@7330 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* read num_channels onceMax Kellermann2008-04-121-3/+4
| | | | | | Read frame->header.channels once, and pass only this integer to flac_convert(). git-svn-id: https://svn.musicpd.org/mpd/trunk@7329 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* assume the buffer is empty in flacWrite()Max Kellermann2008-04-121-4/+3
| | | | | | | flacWrite() is the only function which sets data->chunk_length. If we flush the buffer before we return, we can assume that it is always empty upon entering flacWrite(). git-svn-id: https://svn.musicpd.org/mpd/trunk@7328 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* convert blocks until the buffer is fullMax Kellermann2008-04-121-23/+43
| | | | | | | | | Move the inner loop which converts samples to flac_convert(). There it is isolated and easier to optimize. This function does not have to worry about buffer boundaries; the caller (i.e. flacWrite()) calculates how much is left and is responsible for flushing. That saves a lot of superfluous range checks within the loop. git-svn-id: https://svn.musicpd.org/mpd/trunk@7327 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* calculate bytes_per_channel, check for buffer flush onceMax Kellermann2008-04-121-11/+14
| | | | | | Check for flushing the chunk buffer only once per sample, before iterating over channels and bytes. This saves another 5% CPU cycles. git-svn-id: https://svn.musicpd.org/mpd/trunk@7326 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* don't calculate bytes per sample within the loopMax Kellermann2008-04-121-1/+2
| | | | | | | AudioFormat.bits is volatile, and to read it, 3 pointers had to be deferenced. Calculate this value once. This speeds up this function by 5%. git-svn-id: https://svn.musicpd.org/mpd/trunk@7325 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* use unsigned integers and size_t in the flac pluginMax Kellermann2008-04-121-2/+7
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@7324 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* 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
* use unsigned integers in the flac pluginMax Kellermann2008-04-121-1/+1
| | | | | | The counter variables c_samp and c_chan begin at zero and can never be negative. git-svn-id: https://svn.musicpd.org/mpd/trunk@7228 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* removed unused variableMax Kellermann2008-04-121-3/+3
| | | | | The local variable d_samp is initialized, but never actually used. git-svn-id: https://svn.musicpd.org/mpd/trunk@7227 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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