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* | Merge branch 'ew/deconst'Eric Wong2008-09-016-55/+26
|\ \ | | | | | | | | | | | | | | | * ew/deconst: use deconst_ptr instead of duplicating deconst logic provide a generic deconst_ptr function
| * | use deconst_ptr instead of duplicating deconst logicEric Wong2008-09-015-55/+15
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| * | provide a generic deconst_ptr functionEric Wong2008-09-011-0/+11
| |/ | | | | | | | | | | | | | | | | This is generic enough to be used for various purposes. It will only deconst their argument to work around various braindead APIs without having to write a new wrapper each time we use one of those braindead APIs. It does not cast nor do do anything other than quietly remove the const qualifier for those braindead APIs.
* / playlist: fix shuffle/random distributionEric Wong2008-09-011-6/+19
|/ | | | | | | | | | Previously we were using a naive randomization algorithm that could shuffle already shuffled songs. Now we attempt to correctly[1] implement the Fisher-Yates shuffle. [1] Note: I absolutely suck at basic arithmetic, so there could be off-by-one errors in here, too. I've added assertions in swapSongs and swapOrder functions to more quickly detect them.
* storedPlaylist: correctly expand path when writingEric Wong2008-09-011-6/+7
| | | | | Otherwise we'd be writing to whatever directory that mpd is running in.
* Merge branch 'mk/cleanups'Eric Wong2008-09-0165-778/+996
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * mk/cleanups: (60 commits) pass constant pointers const pointers unsigned integers and size_t oggflac: fix GCC warnings include cleanup protect locate.h from double inclusion playlist: eliminate unused fd parameters jack: made "sample_size" static const moved jack configuration to the JackData struct jack: removed unused macros jack: don't set audioOutput->data=NULL jack: initialize JackData in jack_initDriver() jack: added freeJackClient() jack: initialize jd->client after !jd check jack: eliminate superfluous freeJackData() calls mp3: converted the MUTEFRAME_ macros to an enum mp3: converted the DECODE_ constants to an enum wavpack: don't use "isp" before initialization wavpack: moved code to wavpack_open_wvc() simplified code in the ogg decoder plugin ...
| * pass constant pointersMax Kellermann2008-08-3110-16/+17
| | | | | | | | And again, convert arguments to const.
| * const pointersMax Kellermann2008-08-316-16/+16
| | | | | | | | The usual bunch of pointer arguments which should be const.
| * unsigned integers and size_tMax Kellermann2008-08-313-11/+11
| | | | | | | | | | Use "unsigned int" whenever negative values are not meaningful. Use size_t whenever we are going to describe buffer sizes.
| * oggflac: fix GCC warningsMax Kellermann2008-08-311-9/+9
| | | | | | | | | | | | Fix lots of "unused parameter" warnings in the OggFLAC decoder plugin. Not sure if anybody uses it anymore, since newer libflac obsoletes it.
| * include cleanupMax Kellermann2008-08-314-2/+3
| | | | | | | | Only include headers which are really needed.
| * protect locate.h from double inclusionMax Kellermann2008-08-311-0/+5
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| * playlist: eliminate unused fd parametersMax Kellermann2008-08-313-34/+34
| | | | | | | | | | Again, remove file descriptor parameters, which are not actually used. These functions can also be converted to return void.
| * jack: made "sample_size" static constMax Kellermann2008-08-311-1/+1
| | | | | | | | | | sample_size is a variable which is computed at compile time. Declare it "static const", so the compiler can optimize it away.
| * moved jack configuration to the JackData structMax Kellermann2008-08-311-49/+64
| | | | | | | | | | | | Storing local configuration in global (static) variables is obviously a bad idea. Move all those variables into the JackData struct, including the locks.
| * jack: removed unused macrosMax Kellermann2008-08-311-10/+0
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| * jack: don't set audioOutput->data=NULLMax Kellermann2008-08-311-5/+5
| | | | | | | | | | | | | | There is only one caller of freeJackData() left: jack_finishDriver(). This function is called by the mpd core, and is called exactly once for every successful jack_initDriver(). We do not need to clear audioOutput->data, since this variable is invalidated anyway.
| * jack: initialize JackData in jack_initDriver()Max Kellermann2008-08-311-6/+2
| | | | | | | | | | | | | | | | Over the lifetime of the jack AudioOutput object, we want a single valid JackData object, so we can persistently store data there (configuration etc.). Allocate JackData in jack_initDriver(). After that, we can safely remove all audioOutput->data==NULL checks (and replace them with assertions).
| * jack: added freeJackClient()Max Kellermann2008-08-311-13/+25
| | | | | | | | | | | | | | No need to destroy the JackData object when an error occurs, since jack_finishDriver() already frees it. Only deinitialize the jack library, introduce freeJackClient() for that, and move code from freeJackData().
| * jack: initialize jd->client after !jd checkMax Kellermann2008-08-311-5/+5
| | | | | | | | | | | | Prepare the next patch: make the "!jd" check independent of the jd->client initialization. This way we can change the "jd" initialization semantics later.
| * jack: eliminate superfluous freeJackData() callsMax Kellermann2008-08-311-6/+0
| | | | | | | | | | | | | | connect_jack() invokes freeJackData() in every error handler, although its caller also invokes this function after a failure. We can save a lot of lines in connect_jack() by removing these redundant freeJackData() invocations.
| * mp3: converted the MUTEFRAME_ macros to an enumMax Kellermann2008-08-311-9/+12
| | | | | | | | Also introduce MUTEFRAME_NONE; previously, the code used "0".
| * mp3: converted the DECODE_ constants to an enumMax Kellermann2008-08-311-8/+13
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| * wavpack: don't use "isp" before initializationMax Kellermann2008-08-311-4/+1
| | | | | | | | | | | | The old code called can_seek() with the uninitialized pointer "isp.is". Has this ever worked? Anyway, initialize "isp" first, then call can_seek(&isp).
| * wavpack: moved code to wavpack_open_wvc()Max Kellermann2008-08-311-70/+66
| | | | | | | | | | | | Move everything related to finding and initializing the WVC stream to wavpack_open_wvc(). This greatly simplifies its error handling and the function wavpack_streamdecode().
| * simplified code in the ogg decoder pluginMax Kellermann2008-08-301-25/+25
| | | | | | | | | | Return early when the player thread sent us a command. This saves one level of indentation.
| * oggvorbis: don't detect OGG header if stream is not seekableMax Kellermann2008-08-302-0/+10
| | | | | | | | | | | | | | | | | | If the input stream is not seekable, the try_decode() function consumes valuable data, which is not available to the decode() function anymore. This means that the decode() function does not parse the header correctly. Better skip the detection if we cannot seek. Or implement better buffering, something like unread() or buffered rewind().
| * mp3: audio_linear_dither() returns mpd_sint16Max Kellermann2008-08-301-11/+9
| | | | | | | | | | | | The return value of audio_linear_dither() is always casted to mpd_sint16. Returning long does not make sense, and consumed 8 bytes on a 64 bit platform.
| * mp3: changed outputBuffer's type to mpd_sint16[]Max Kellermann2008-08-301-4/+3
| | | | | | | | | | The output buffer always contains mpd_sint16; declaring it with that type saves several casts.
| * mp3: moved num_samples calculation out of the loopMax Kellermann2008-08-301-6/+7
| | | | | | | | | | The previous patch removed all loop specific dependencies from the num_samples formula; we can now calculate it before entering the loop.
| * mp3: eliminated outputPtrMax Kellermann2008-08-301-17/+9
| | | | | | | | | | | | The output buffer is always flushed after being appended to, which allows us to assume it is always empty. Always start writing at outputBuffer, don't remember outputPtr.
| * mp3: don't do a second flush in mp3_decode()Max Kellermann2008-08-301-12/+1
| | | | | | | | | | | | The previous patch made mp3Read() flush the output buffer in every iteration, which means we can eliminate the flush check after invoking mp3Read().
| * mp3: always flush directly after decoding/ditheringMax Kellermann2008-08-301-17/+13
| | | | | | | | | | Since we try to fill the buffer in every iteration, we assume that we should flush the output buffer at the end of each iteration.
| * mp3: dither a whole block at a timeMax Kellermann2008-08-301-3/+9
| | | | | | | | | | | | Fill the whole output buffer at a time by using dither_buffer()'s ability to decode blocks. Calculate how many samples fit into the output buffer before each invocation.
| * mp3: moved dropSamplesAtEnd check out of the loopMax Kellermann2008-08-301-21/+19
| | | | | | | | | | | | | | Simplifying loops for performance: why check dropSamplesAtEnd in every iteration, when we could modify the loop boundary? The (writable) variable samplesLeft can be eliminated; add a write-once variable pcm_length instead, which is used for the loop condition.
| * mp3: make samplesPerFrame more localMax Kellermann2008-08-301-2/+1
| | | | | | | | | | | | The variable samplesPerFrame is used only in one single closure. Make it local to this closure. The compiler will probably convert it to a register anyway.
| * mp3: unsigned integersMax Kellermann2008-08-301-11/+11
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| * mp3: moved code to dither_buffer()Max Kellermann2008-08-301-14/+30
| | | | | | | | | | | | | | | | Preparing for simplifying and thus speeding up the dithering code: moved dithering to a separate function which contains a trivial loop. With this patch, only one sample is dithered at a time, but the following patches will allow us to dither a whole block at a time, without complicated buffer length checks.
| * mp3: don't check dropSamplesAtStart in the loopMax Kellermann2008-08-301-7/+14
| | | | | | | | | | Performance improvement by moving stuff out of a loop: skip part of the first frame before entering the loop.
| * assert song->url != NULLMax Kellermann2008-08-301-0/+3
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| * aac: support decoding AAC streamsMax Kellermann2008-08-301-2/+128
| | | | | | | | | | | | Copy some code from aac_decode() to aac_stream_decode() and apply necessary changes to allow streaming audio data. Both functions might be merged later.
| * aac: splitted aac_parse_header() from initAacBuffer()Max Kellermann2008-08-301-11/+16
| | | | | | | | | | | | | | initAacBuffer() should really only initialize the buffer; currently, it also reads data from the input stream and parses the header. All of the AAC buffer code should probably be moved to a separate library anyway.
| * aac: check buffer lengthsMax Kellermann2008-08-301-2/+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 Kellermann2008-08-301-16/+4
| | | | | | | | Eliminate some duplicated code by using fillAacBuffer().
| * find AAC framesMax Kellermann2008-08-301-1/+35
| | | | | | | | | | Find AAC frames in the input and skip invalid data. This prepares AAC streaming.
| * aac: moved code to adts_check_frame()Max Kellermann2008-08-301-11/+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 Kellermann2008-08-301-7/+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 Kellermann2008-08-301-5/+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 Kellermann2008-08-301-6/+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 Kellermann2008-08-301-33/+25
| | | | | | | | | | Return instead of putting all the code into a if-closure. That saves one level of indentation.
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