aboutsummaryrefslogtreecommitdiffstats
path: root/src/inputPlugins (follow)
Commit message (Collapse)AuthorAgeFilesLines
* wavpack: remove C99 initializersEric Wong2008-10-051-12/+16
| | | | | More power to those who can't afford newer computers capable of installing/building/running newer toolchains.
* Revert "Start using song pointers in core data structures"Eric Wong2008-10-021-4/+2
| | | | | | | This actually opened us up to making lock dependencies more difficult than they needed to be now that we have threaded updates. We would always use the memory anyways, just in the stack instead of bss.
* Switch to C99 types (retaining compat with old compilers)Eric Wong2008-09-296-19/+19
| | | | | | | | | | | | | | | Seeing the "mpd_" prefix _everywhere_ is mind-numbing as the mind needs to retrain itself to skip over the first 4 tokens of a type to get to its meaning. So avoid having extra characters on my terminal to make it easier to follow code at 2:30 am in the morning. Please report any new issues you may come across on Free toolchains. I realize how difficult it can be to build/maintain cross-compiling toolchains and I have no intention of forcing people to upgrade their toolchains to build mpd. Tested with gcc 2.95.4 and and gcc 4.3.1 on x86-32.
* flac: removed FlacData.chunk_lengthMax Kellermann2008-09-294-9/+3
| | | | | chunk_length can be converted to a local variable, because it is always reset to 0 after it was used.
* flac: merged flacSendChunk() into flac_common_write()Max Kellermann2008-09-291-11/+6
| | | | | | | | | | | Since flacSendChunk() is a trivial function and is only used in one location, move the code there. The advantage is that calling decoder_data() directly returns the decoder_command value, so we can eliminate one decoder_get_command() call. [ew: using the terser API in the main branch, it always eliminated the dc_intr()/dc_seek() call, but I agree that killing a camel, err.. camelCase function is a good thing]
* flac: removed generic sample size supportMax Kellermann2008-09-291-31/+27
| | | | | | | | Support for bit rates except 16 bits (and 8 bits on little endian) has always been broken. Since we added optimized functions for 8, 16, 24/32 bits, we can remove the generic flac_convert() function. Instead of removing it, convert it to a wrapper function for flac_convert_*().
* flac: added special functions for 8 and 32 bitMax Kellermann2008-09-291-0/+37
| | | | | Same optimization for 8 and 32 bit files, like the previous patch for 16 bit. Along the way, this patch adds 24 bit FLAC support!
* flac: added optimized converter for 16 bitMax Kellermann2008-09-291-0/+17
| | | | | | flac_convert_16() runs a lot faster than the generic (and quite buggy) function flac_convert(). flac_convert_16() is only used for non-stereo files, since there is already flac_convert_stereo16().
* flac: use signed integers in flac_convert_stereo16()Max Kellermann2008-09-291-6/+4
| | | | | | By mistake, I casted the sample value to uint16_t, which is wrong. This patch simplifies the code by using a int16_t pointer instead of casting to int16_t* every time.
* flac: moved code from flacWrite() to _flac_common.cMax Kellermann2008-09-294-120/+90
| | | | | | | 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 Kellermann2008-09-291-6/+2
| | | | | The previous patch on this topic was incomplete: it still added data->chunk_length when calling flac_convert(). Remove this, too.
* start using prefixcmp()Eric Wong2008-09-201-1/+1
| | | | | LOC reduction and less noise makes things easier for tired old folks to follow.
* mp3: fix long line, I can't read past 80 colsEric Wong2008-09-171-1/+2
|
* mp3: fix buffer overflow when max_frames is too largeMax Kellermann2008-09-171-0/+5
| | | | | | | The function decodeFirstFrame() allocates memory based on data from the mp3 header. This can make the buffer size allocation overflow, or lead to a DoS attack with a very large buffer. Cap this buffer at 8 million frames, which should really be enough for reasonable files.
* mp4: fix potential integer overflow bug in the mp4_decode() functionTerry2008-09-121-0/+7
| | | | | | | | | | A crafted mp4 file could cause an integer overflow in mp4_decode function in src/inputPlugins/mp4_plugin.c. mp4ff_num_samples() function returns some tainted value. sizeof(float) * numSamples is an integer overflow operation if numSamples is too huge, so xmalloc will allocate a small memory region. I constructe a mp4 file, and use faad2 to open the file. mp4ff_num_samples() returns -1. So I think mpd bears from the same problem.
* fix -Wcast-qual -Wwrite-strings warningsMax Kellermann2008-09-092-8/+21
| | | | | | | | | The previous patch enabled these warnings. In Eric's branch, they were worked around with a generic deconst_ptr() function. There are several places where we can add "const" to pointers, and in others, libraries want non-const strings. In the latter, convert string literals to "static char[]" variables - this takes the same space, and seems safer than deconsting a string literal.
* tag: fix the shout and oggflac pluginsMax Kellermann2008-09-021-1/+3
| | | | | | During the tag library refactoring, the shout plugin was disabled, and I forgot about adapting it to the new API. Apply the same fixes to the oggflac decoder plugin.
* wavpack: tag_new() cannot failMax Kellermann2008-09-021-5/+0
| | | | | Since tag_new() uses xmalloc(), it cannot fail - if we're really out of memory, the process will abort.
* tag: renamed functions, no CamelCaseMax Kellermann2008-09-0210-53/+52
|
* tag: renamed MpdTag and MpdTagItem to struct mpd_tag, struct tag_itemMax Kellermann2008-09-0212-44/+44
| | | | | Getting rid of CamelCase; not having typedefs also allows us to forward-declare the structures.
* Merge branch 'mk/cleanups'Eric Wong2008-09-0113-418/+509
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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 ...
| * 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.
| * 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
| |
| * 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
| |
| * 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.
| * 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.
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-05 03:39:12 +0000