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* flac: added optimized converter for 16 bitMax Kellermann2008-09-231-0/+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 Kellermann2008-09-231-6/+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 Kellermann2008-09-234-129/+97
| | | | | | | 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 Kellermann2008-09-232-3/+4
| | | | | | 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.
* start using prefixcmp()Eric Wong2008-09-231-1/+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 Wong2008-09-231-1/+2
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* 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.
* audio_format: converted typedef AudioFormat to struct audio_formatMax Kellermann2008-09-079-11/+11
| | | | | Get rid of CamelCase, and don't use a typedef, so we can forward-declare it, and unclutter the include dependencies.
* fix -Wcast-qual -Wwrite-strings warningsMax Kellermann2008-09-072-8/+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.
* oggflac: fix GCC warningsMax Kellermann2008-08-291-9/+9
| | | | | | Fix lots of "unused parameter" warnings in the OggFLAC decoder plugin. Not sure if anybody uses it anymore, since newer libflac obsoletes it.
* tag: fix the shout and oggflac pluginsMax Kellermann2008-08-291-2/+4
| | | | | | 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 Kellermann2008-08-291-5/+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 Kellermann2008-08-2910-58/+57
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* tag: renamed MpdTag and MpdTagItem to struct tag, struct mpd_tag_itemMax Kellermann2008-08-2911-43/+43
| | | | | Getting rid of CamelCase; not having typedefs also allows us to forward-declare the structures.
* moved global variable "ob" to outputBuffer.hMax Kellermann2008-08-262-0/+6
| | | | | | | This releases several include file dependencies. As a side effect, "CHUNK_SIZE" isn't defined by decoder_api.h anymore, so we have to define it directly in the plugins which need it. It just isn't worth it to add it to the decoder plugin API.
* flac: decoder command means EOFMax Kellermann2008-08-262-11/+9
| | | | | | | 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 Kellermann2008-08-262-4/+16
| | | | | | | 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 Kellermann2008-08-265-13/+14
| | | | | | 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.
* mp3: converted the MUTEFRAME_ macros to an enumMax Kellermann2008-08-261-9/+12
| | | | Also introduce MUTEFRAME_NONE; previously, the code used "0".
* mp3: converted the DECODE_ constants to an enumMax Kellermann2008-08-261-8/+13
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* wavpack: don't use "isp" before initializationMax Kellermann2008-08-261-4/+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 Kellermann2008-08-261-79/+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 Kellermann2008-08-261-25/+25
| | | | | Return early when the player thread sent us a command. This saves one level of indentation.
* added decoder_read()Max Kellermann2008-08-268-66/+17
| | | | | | | | | 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.
* wavpack: added InputStreamPlus.decoderMax Kellermann2008-08-261-4/+7
| | | | The "decoder" object reference will be used by another patch.
* oggvorbis: don't detect OGG header if stream is not seekableMax Kellermann2008-08-262-0/+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().
* added AacBuffer.decoderMax Kellermann2008-08-261-4/+7
| | | | | | We need the decoder object at several places in the AAC plugin. Add it to mp3DecodeData, so we don't have to pass it around in every function.
* mp3: added mp3DecodeData.decoderMax Kellermann2008-08-261-9/+13
| | | | | | We need the decoder object at several places in the mp3 plugin. Add it to mp3DecodeData, so we don't have to pass it around in every function.
* mp3: audio_linear_dither() returns mpd_sint16Max Kellermann2008-08-261-11/+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 Kellermann2008-08-261-3/+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 Kellermann2008-08-261-5/+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 Kellermann2008-08-261-14/+3
| | | | | | 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 Kellermann2008-08-261-17/+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 Kellermann2008-08-261-15/+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 Kellermann2008-08-261-3/+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 Kellermann2008-08-261-21/+18
| | | | | | | 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 Kellermann2008-08-261-2/+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 Kellermann2008-08-261-11/+11
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* mp3: removed double cmd==STOP checkMax Kellermann2008-08-261-3/+0
| | | | | cmd has already been checked before, it cannot have changed meanwhile because it is a local variable.
* mp3: moved code to dither_buffer()Max Kellermann2008-08-261-14/+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 Kellermann2008-08-261-7/+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 Kellermann2008-08-261-2/+137
| | | | | | 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 Kellermann2008-08-261-11/+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 Kellermann2008-08-261-2/+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.
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