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* 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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* 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 Kellermann2008-08-2610-21/+21
| | | | | | "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.
* eliminate OUTPUT_BUFFER_DC_STOP, OUTPUT_BUFFER_DC_SEEKMax Kellermann2008-08-262-4/+5
| | | | | (Ab)use the decoder_command enumeration, which has nearly the same values and the same meaning.
* added decoder_get_url()Max Kellermann2008-08-261-1/+1
| | | | | | The wavpack decoder plugin implements a hack, and it needs the song URL for that. This API (and the hack) should be revised later, but add that function for now.
* don't set dc->seekable in wavpack pluginMax Kellermann2008-08-261-2/+0
| | | | dc->seekable is already set by decodeStart().
* use a local "initialized" flag instead of dc->stateMax Kellermann2008-08-263-5/+11
| | | | | | 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 Kellermann2008-08-2610-40/+40
| | | | | 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 Kellermann2008-08-2610-15/+15
| | | | | | | 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 Kellermann2008-08-2611-59/+67
| | | | | | 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 Kellermann2008-08-264-4/+3
| | | | | | 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".
* remove one indent level from audiofile pluginMax Kellermann2008-08-261-30/+25
| | | | | Anonymous code blocks just to declare variables look ugly. Move the variable declarations up and disband the code block.
* use break instead of local variable "eof"Max Kellermann2008-08-261-16/+12
| | | | | Similar to previous patch: eliminate one variable by using "break". This also simplifies the code since we can remove one level of indent.
* removed local variable "eof" because it is unusedMax Kellermann2008-08-262-19/+9
| | | | | "break" is so much easier than "eof=1; continue;", when "!eof" is the loop condition.
* simplify several dc->command checksMax Kellermann2008-08-261-7/+3
| | | | | | Since we have merged dc->stop, dc->seek into one variable, we don't have to check both conditions at a time; we can replace "!stop && !seek" with "none".
* added parameter total_time to decoder_initialized()Max Kellermann2008-08-2612-27/+25
| | | | | Similar to the previous patch: pass total_time instead of manipulating dc->totalTime directly.
* added audio_format parameter to decoder_initialized()Max Kellermann2008-08-2612-81/+78
| | | | | | 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 Kellermann2008-08-2610-22/+21
| | | | | | 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 Kellermann2008-08-269-59/+58
| | | | | Moved all of the player-waiting code to decoder_data(), to make OutputBuffer more generic.
* added decoder_initialized()Max Kellermann2008-08-2611-31/+28
| | | | | | | 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 Kellermann2008-08-2612-23/+35
| | | | | | 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 Kellermann2008-08-2610-30/+15
| | | | | | | | 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 Kellermann2008-08-2611-80/+95
| | | | | | | 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 Kellermann2008-08-2613-64/+0
| | | | | Include only headers which are really required. This speeds up compilation and helps detect cross-layer accesses.
* enable -Wpointer-arith, -Wstrict-prototypesMax Kellermann2008-08-264-10/+11
| | | | | | 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.
* don't call seekInputStream(0) if r==0Max Kellermann2008-06-301-1/+2
| | | | | | If nothing has been read from the input stream, we don't have to rewind it. git-svn-id: https://svn.musicpd.org/mpd/trunk@7397 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* eliminated local variable "to_read"Max Kellermann2008-06-301-4/+3
| | | | | | | The variable "to_read" is never modified except in the last iteration of the while loop. This means the while condition will never become false, as the body will break before that may be checked. git-svn-id: https://svn.musicpd.org/mpd/trunk@7396 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* mod: fix crashing on modtracker filesHans de Goede2008-06-131-1/+1
| | | | | | | | | This patch was taken from http://bugzilla.livna.org/show_bug.cgi?id=1987 and addresses bug 0001693[1] [1] - http://musicpd.org/mantis/view.php?id=1693 git-svn-id: https://svn.musicpd.org/mpd/trunk@7374 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* oggflac_plugin: fix build with libOggFLAC lib (libFLAC <= 7)Eric Wong2008-06-011-1/+1
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@7370 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* use dc.current_song instead of pc.current_songMax Kellermann2008-04-151-1/+1
| | | | | | When we are in an input plugin, dc.current_song should already be set. Use it instead of pc.current_song. git-svn-id: https://svn.musicpd.org/mpd/trunk@7363 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* Make the OutputBuffer API more consistentEric Wong2008-04-1312-42/+42
| | | | | | | | | | 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 Wong2008-04-1312-86/+77
| | | | | | | 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 Wong2008-04-1312-244/+222
| | | | | | | 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
* Get rid of PlayerControl inside the PlayerData structEric Wong2008-04-131-1/+1
| | | | | | | | | | | | It actually increases our image size a small bit and may even hurt performance a very small bit, but makes the code less verbose and easier to manage. I don't see a reason for mpd to ever support playing multiple files at the same time (users can run multiple instances of mpd if they really want to play Zaireeka, but that's such an edge case it's not worth ever supporting in our code). git-svn-id: https://svn.musicpd.org/mpd/trunk@7352 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
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