games/semicongine: semiconginev2/audio/stb

comparison semiconginev2/audio/stb_vorbis.c @ 1225:27cd1c21290e compiletime-tests

did: refactor resources

author	sam <sam@basx.dev>
date	Wed, 17 Jul 2024 22:20:59 +0700
parents	semiconginev2/old/resources/stb_vorbis.c@56781cc0fc7c
children

comparison

equal deleted inserted replaced

-:a3fa15c25026
+:27cd1c21290e
+// Ogg Vorbis audio decoder - v1.22 - public domain
+// http://nothings.org/stb_vorbis/
+//
+// Original version written by Sean Barrett in 2007.
+//
+// Originally sponsored by RAD Game Tools. Seeking implementation
+// sponsored by Phillip Bennefall, Marc Andersen, Aaron Baker,
+// Elias Software, Aras Pranckevicius, and Sean Barrett.
+//
+// LICENSE
+//
+//   See end of file for license information.
+//
+// Limitations:
+//
+//   - floor 0 not supported (used in old ogg vorbis files pre-2004)
+//   - lossless sample-truncation at beginning ignored
+//   - cannot concatenate multiple vorbis streams
+//   - sample positions are 32-bit, limiting seekable 192Khz
+//       files to around 6 hours (Ogg supports 64-bit)
+//
+// Feature contributors:
+//    Dougall Johnson (sample-exact seeking)
+//
+// Bugfix/warning contributors:
+//    Terje Mathisen     Niklas Frykholm     Andy Hill
+//    Casey Muratori     John Bolton         Gargaj
+//    Laurent Gomila     Marc LeBlanc        Ronny Chevalier
+//    Bernhard Wodo      Evan Balster        github:alxprd
+//    Tom Beaumont       Ingo Leitgeb        Nicolas Guillemot
+//    Phillip Bennefall  Rohit               Thiago Goulart
+//    github:manxorist   Saga Musix          github:infatum
+//    Timur Gagiev       Maxwell Koo         Peter Waller
+//    github:audinowho   Dougall Johnson     David Reid
+//    github:Clownacy    Pedro J. Estebanez  Remi Verschelde
+//    AnthoFoxo          github:morlat       Gabriel Ravier
+//
+// Partial history:
+//    1.22    - 2021-07-11 - various small fixes
+//    1.21    - 2021-07-02 - fix bug for files with no comments
+//    1.20    - 2020-07-11 - several small fixes
+//    1.19    - 2020-02-05 - warnings
+//    1.18    - 2020-02-02 - fix seek bugs; parse header comments; misc warnings etc.
+//    1.17    - 2019-07-08 - fix CVE-2019-13217..CVE-2019-13223 (by ForAllSecure)
+//    1.16    - 2019-03-04 - fix warnings
+//    1.15    - 2019-02-07 - explicit failure if Ogg Skeleton data is found
+//    1.14    - 2018-02-11 - delete bogus dealloca usage
+//    1.13    - 2018-01-29 - fix truncation of last frame (hopefully)
+//    1.12    - 2017-11-21 - limit residue begin/end to blocksize/2 to avoid large temp allocs in bad/corrupt files
+//    1.11    - 2017-07-23 - fix MinGW compilation
+//    1.10    - 2017-03-03 - more robust seeking; fix negative ilog(); clear error in open_memory
+//    1.09    - 2016-04-04 - back out 'truncation of last frame' fix from previous version
+//    1.08    - 2016-04-02 - warnings; setup memory leaks; truncation of last frame
+//    1.07    - 2015-01-16 - fixes for crashes on invalid files; warning fixes; const
+//    1.06    - 2015-08-31 - full, correct support for seeking API (Dougall Johnson)
+//                           some crash fixes when out of memory or with corrupt files
+//                           fix some inappropriately signed shifts
+//    1.05    - 2015-04-19 - don't define __forceinline if it's redundant
+//    1.04    - 2014-08-27 - fix missing const-correct case in API
+//    1.03    - 2014-08-07 - warning fixes
+//    1.02    - 2014-07-09 - declare qsort comparison as explicitly _cdecl in Windows
+//    1.01    - 2014-06-18 - fix stb_vorbis_get_samples_float (interleaved was correct)
+//    1.0     - 2014-05-26 - fix memory leaks; fix warnings; fix bugs in >2-channel;
+//                           (API change) report sample rate for decode-full-file funcs
+//
+// See end of file for full version history.
+//////////////////////////////////////////////////////////////////////////////
+//
+//  HEADER BEGINS HERE
+//
+#ifndef STB_VORBIS_INCLUDE_STB_VORBIS_H
+#define STB_VORBIS_INCLUDE_STB_VORBIS_H
+#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
+#define STB_VORBIS_NO_STDIO 1
+#endif
+#ifndef STB_VORBIS_NO_STDIO
+#include <stdio.h>
+#endif
+#ifdef __cplusplus
+extern "C" {
+#endif
+///////////   THREAD SAFETY
+// Individual stb_vorbis* handles are not thread-safe; you cannot decode from
+// them from multiple threads at the same time. However, you can have multiple
+// stb_vorbis* handles and decode from them independently in multiple thrads.
+///////////   MEMORY ALLOCATION
+// normally stb_vorbis uses malloc() to allocate memory at startup,
+// and alloca() to allocate temporary memory during a frame on the
+// stack. (Memory consumption will depend on the amount of setup
+// data in the file and how you set the compile flags for speed
+// vs. size. In my test files the maximal-size usage is ~150KB.)
+//
+// You can modify the wrapper functions in the source (setup_malloc,
+// setup_temp_malloc, temp_malloc) to change this behavior, or you
+// can use a simpler allocation model: you pass in a buffer from
+// which stb_vorbis will allocate _all_ its memory (including the
+// temp memory). "open" may fail with a VORBIS_outofmem if you
+// do not pass in enough data; there is no way to determine how
+// much you do need except to succeed (at which point you can
+// query get_info to find the exact amount required. yes I know
+// this is lame).
+//
+// If you pass in a non-NULL buffer of the type below, allocation
+// will occur from it as described above. Otherwise just pass NULL
+// to use malloc()/alloca()
+typedef struct
+{
+char *alloc_buffer;
+int   alloc_buffer_length_in_bytes;
+} stb_vorbis_alloc;
+///////////   FUNCTIONS USEABLE WITH ALL INPUT MODES
+typedef struct stb_vorbis stb_vorbis;
+typedef struct
+{
+unsigned int sample_rate;
+int channels;
+unsigned int setup_memory_required;
+unsigned int setup_temp_memory_required;
+unsigned int temp_memory_required;
+int max_frame_size;
+} stb_vorbis_info;
+typedef struct
+{
+char *vendor;
+int comment_list_length;
+char **comment_list;
+} stb_vorbis_comment;
+// get general information about the file
+extern stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f);
+// get ogg comments
+extern stb_vorbis_comment stb_vorbis_get_comment(stb_vorbis *f);
+// get the last error detected (clears it, too)
+extern int stb_vorbis_get_error(stb_vorbis *f);
+// close an ogg vorbis file and free all memory in use
+extern void stb_vorbis_close(stb_vorbis *f);
+// this function returns the offset (in samples) from the beginning of the
+// file that will be returned by the next decode, if it is known, or -1
+// otherwise. after a flush_pushdata() call, this may take a while before
+// it becomes valid again.
+// NOT WORKING YET after a seek with PULLDATA API
+extern int stb_vorbis_get_sample_offset(stb_vorbis *f);
+// returns the current seek point within the file, or offset from the beginning
+// of the memory buffer. In pushdata mode it returns 0.
+extern unsigned int stb_vorbis_get_file_offset(stb_vorbis *f);
+///////////   PUSHDATA API
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+// this API allows you to get blocks of data from any source and hand
+// them to stb_vorbis. you have to buffer them; stb_vorbis will tell
+// you how much it used, and you have to give it the rest next time;
+// and stb_vorbis may not have enough data to work with and you will
+// need to give it the same data again PLUS more. Note that the Vorbis
+// specification does not bound the size of an individual frame.
+extern stb_vorbis *stb_vorbis_open_pushdata(
+const unsigned char * datablock, int datablock_length_in_bytes,
+int *datablock_memory_consumed_in_bytes,
+int *error,
+const stb_vorbis_alloc *alloc_buffer);
+// create a vorbis decoder by passing in the initial data block containing
+//    the ogg&vorbis headers (you don't need to do parse them, just provide
+//    the first N bytes of the file--you're told if it's not enough, see below)
+// on success, returns an stb_vorbis *, does not set error, returns the amount of
+//    data parsed/consumed on this call in *datablock_memory_consumed_in_bytes;
+// on failure, returns NULL on error and sets *error, does not change *datablock_memory_consumed
+// if returns NULL and *error is VORBIS_need_more_data, then the input block was
+//       incomplete and you need to pass in a larger block from the start of the file
+extern int stb_vorbis_decode_frame_pushdata(
+stb_vorbis *f,
+const unsigned char *datablock, int datablock_length_in_bytes,
+int *channels,             // place to write number of float * buffers
+float ***output,           // place to write float ** array of float * buffers
+int *samples               // place to write number of output samples
+);
+// decode a frame of audio sample data if possible from the passed-in data block
+//
+// return value: number of bytes we used from datablock
+//
+// possible cases:
+//     0 bytes used, 0 samples output (need more data)
+//     N bytes used, 0 samples output (resynching the stream, keep going)
+//     N bytes used, M samples output (one frame of data)
+// note that after opening a file, you will ALWAYS get one N-bytes,0-sample
+// frame, because Vorbis always "discards" the first frame.
+//
+// Note that on resynch, stb_vorbis will rarely consume all of the buffer,
+// instead only datablock_length_in_bytes-3 or less. This is because it wants
+// to avoid missing parts of a page header if they cross a datablock boundary,
+// without writing state-machiney code to record a partial detection.
+//
+// The number of channels returned are stored in *channels (which can be
+// NULL--it is always the same as the number of channels reported by
+// get_info). *output will contain an array of float* buffers, one per
+// channel. In other words, (*output)[0][0] contains the first sample from
+// the first channel, and (*output)[1][0] contains the first sample from
+// the second channel.
+//
+// *output points into stb_vorbis's internal output buffer storage; these
+// buffers are owned by stb_vorbis and application code should not free
+// them or modify their contents. They are transient and will be overwritten
+// once you ask for more data to get decoded, so be sure to grab any data
+// you need before then.
+extern void stb_vorbis_flush_pushdata(stb_vorbis *f);
+// inform stb_vorbis that your next datablock will not be contiguous with
+// previous ones (e.g. you've seeked in the data); future attempts to decode
+// frames will cause stb_vorbis to resynchronize (as noted above), and
+// once it sees a valid Ogg page (typically 4-8KB, as large as 64KB), it
+// will begin decoding the _next_ frame.
+//
+// if you want to seek using pushdata, you need to seek in your file, then
+// call stb_vorbis_flush_pushdata(), then start calling decoding, then once
+// decoding is returning you data, call stb_vorbis_get_sample_offset, and
+// if you don't like the result, seek your file again and repeat.
+#endif
+//////////   PULLING INPUT API
+#ifndef STB_VORBIS_NO_PULLDATA_API
+// This API assumes stb_vorbis is allowed to pull data from a source--
+// either a block of memory containing the _entire_ vorbis stream, or a
+// FILE * that you or it create, or possibly some other reading mechanism
+// if you go modify the source to replace the FILE * case with some kind
+// of callback to your code. (But if you don't support seeking, you may
+// just want to go ahead and use pushdata.)
+#if !defined(STB_VORBIS_NO_STDIO) && !defined(STB_VORBIS_NO_INTEGER_CONVERSION)
+extern int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output);
+#endif
+#if !defined(STB_VORBIS_NO_INTEGER_CONVERSION)
+extern int stb_vorbis_decode_memory(const unsigned char *mem, int len, int *channels, int *sample_rate, short **output);
+#endif
+// decode an entire file and output the data interleaved into a malloc()ed
+// buffer stored in *output. The return value is the number of samples
+// decoded, or -1 if the file could not be opened or was not an ogg vorbis file.
+// When you're done with it, just free() the pointer returned in *output.
+extern stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len,
+int *error, const stb_vorbis_alloc *alloc_buffer);
+// create an ogg vorbis decoder from an ogg vorbis stream in memory (note
+// this must be the entire stream!). on failure, returns NULL and sets *error
+#ifndef STB_VORBIS_NO_STDIO
+extern stb_vorbis * stb_vorbis_open_filename(const char *filename,
+int *error, const stb_vorbis_alloc *alloc_buffer);
+// create an ogg vorbis decoder from a filename via fopen(). on failure,
+// returns NULL and sets *error (possibly to VORBIS_file_open_failure).
+extern stb_vorbis * stb_vorbis_open_file(FILE *f, int close_handle_on_close,
+int *error, const stb_vorbis_alloc *alloc_buffer);
+// create an ogg vorbis decoder from an open FILE *, looking for a stream at
+// the _current_ seek point (ftell). on failure, returns NULL and sets *error.
+// note that stb_vorbis must "own" this stream; if you seek it in between
+// calls to stb_vorbis, it will become confused. Moreover, if you attempt to
+// perform stb_vorbis_seek_*() operations on this file, it will assume it
+// owns the _entire_ rest of the file after the start point. Use the next
+// function, stb_vorbis_open_file_section(), to limit it.
+extern stb_vorbis * stb_vorbis_open_file_section(FILE *f, int close_handle_on_close,
+int *error, const stb_vorbis_alloc *alloc_buffer, unsigned int len);
+// create an ogg vorbis decoder from an open FILE *, looking for a stream at
+// the _current_ seek point (ftell); the stream will be of length 'len' bytes.
+// on failure, returns NULL and sets *error. note that stb_vorbis must "own"
+// this stream; if you seek it in between calls to stb_vorbis, it will become
+// confused.
+#endif
+extern int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number);
+extern int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number);
+// these functions seek in the Vorbis file to (approximately) 'sample_number'.
+// after calling seek_frame(), the next call to get_frame_*() will include
+// the specified sample. after calling stb_vorbis_seek(), the next call to
+// stb_vorbis_get_samples_* will start with the specified sample. If you
+// do not need to seek to EXACTLY the target sample when using get_samples_*,
+// you can also use seek_frame().
+extern int stb_vorbis_seek_start(stb_vorbis *f);
+// this function is equivalent to stb_vorbis_seek(f,0)
+extern unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f);
+extern float        stb_vorbis_stream_length_in_seconds(stb_vorbis *f);
+// these functions return the total length of the vorbis stream
+extern int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output);
+// decode the next frame and return the number of samples. the number of
+// channels returned are stored in *channels (which can be NULL--it is always
+// the same as the number of channels reported by get_info). *output will
+// contain an array of float* buffers, one per channel. These outputs will
+// be overwritten on the next call to stb_vorbis_get_frame_*.
+//
+// You generally should not intermix calls to stb_vorbis_get_frame_*()
+// and stb_vorbis_get_samples_*(), since the latter calls the former.
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+extern int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts);
+extern int stb_vorbis_get_frame_short            (stb_vorbis *f, int num_c, short **buffer, int num_samples);
+#endif
+// decode the next frame and return the number of *samples* per channel.
+// Note that for interleaved data, you pass in the number of shorts (the
+// size of your array), but the return value is the number of samples per
+// channel, not the total number of samples.
+//
+// The data is coerced to the number of channels you request according to the
+// channel coercion rules (see below). You must pass in the size of your
+// buffer(s) so that stb_vorbis will not overwrite the end of the buffer.
+// The maximum buffer size needed can be gotten from get_info(); however,
+// the Vorbis I specification implies an absolute maximum of 4096 samples
+// per channel.
+// Channel coercion rules:
+//    Let M be the number of channels requested, and N the number of channels present,
+//    and Cn be the nth channel; let stereo L be the sum of all L and center channels,
+//    and stereo R be the sum of all R and center channels (channel assignment from the
+//    vorbis spec).
+//        M    N       output
+//        1    k      sum(Ck) for all k
+//        2    *      stereo L, stereo R
+//        k    l      k > l, the first l channels, then 0s
+//        k    l      k <= l, the first k channels
+//    Note that this is not _good_ surround etc. mixing at all! It's just so
+//    you get something useful.
+extern int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats);
+extern int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples);
+// gets num_samples samples, not necessarily on a frame boundary--this requires
+// buffering so you have to supply the buffers. DOES NOT APPLY THE COERCION RULES.
+// Returns the number of samples stored per channel; it may be less than requested
+// at the end of the file. If there are no more samples in the file, returns 0.
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+extern int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts);
+extern int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int num_samples);
+#endif
+// gets num_samples samples, not necessarily on a frame boundary--this requires
+// buffering so you have to supply the buffers. Applies the coercion rules above
+// to produce 'channels' channels. Returns the number of samples stored per channel;
+// it may be less than requested at the end of the file. If there are no more
+// samples in the file, returns 0.
+#endif
+////////   ERROR CODES
+enum STBVorbisError
+{
+VORBIS__no_error,
+VORBIS_need_more_data=1,             // not a real error
+VORBIS_invalid_api_mixing,           // can't mix API modes
+VORBIS_outofmem,                     // not enough memory
+VORBIS_feature_not_supported,        // uses floor 0
+VORBIS_too_many_channels,            // STB_VORBIS_MAX_CHANNELS is too small
+VORBIS_file_open_failure,            // fopen() failed
+VORBIS_seek_without_length,          // can't seek in unknown-length file
+VORBIS_unexpected_eof=10,            // file is truncated?
+VORBIS_seek_invalid,                 // seek past EOF
+// decoding errors (corrupt/invalid stream) -- you probably
+// don't care about the exact details of these
+// vorbis errors:
+VORBIS_invalid_setup=20,
+VORBIS_invalid_stream,
+// ogg errors:
+VORBIS_missing_capture_pattern=30,
+VORBIS_invalid_stream_structure_version,
+VORBIS_continued_packet_flag_invalid,
+VORBIS_incorrect_stream_serial_number,
+VORBIS_invalid_first_page,
+VORBIS_bad_packet_type,
+VORBIS_cant_find_last_page,
+VORBIS_seek_failed,
+VORBIS_ogg_skeleton_not_supported
+};
+#ifdef __cplusplus
+}
+#endif
+#endif // STB_VORBIS_INCLUDE_STB_VORBIS_H
+//
+//  HEADER ENDS HERE
+//
+//////////////////////////////////////////////////////////////////////////////
+#ifndef STB_VORBIS_HEADER_ONLY
+// global configuration settings (e.g. set these in the project/makefile),
+// or just set them in this file at the top (although ideally the first few
+// should be visible when the header file is compiled too, although it's not
+// crucial)
+// STB_VORBIS_NO_PUSHDATA_API
+//     does not compile the code for the various stb_vorbis_*_pushdata()
+//     functions
+// #define STB_VORBIS_NO_PUSHDATA_API
+// STB_VORBIS_NO_PULLDATA_API
+//     does not compile the code for the non-pushdata APIs
+// #define STB_VORBIS_NO_PULLDATA_API
+// STB_VORBIS_NO_STDIO
+//     does not compile the code for the APIs that use FILE *s internally
+//     or externally (implied by STB_VORBIS_NO_PULLDATA_API)
+// #define STB_VORBIS_NO_STDIO
+// STB_VORBIS_NO_INTEGER_CONVERSION
+//     does not compile the code for converting audio sample data from
+//     float to integer (implied by STB_VORBIS_NO_PULLDATA_API)
+// #define STB_VORBIS_NO_INTEGER_CONVERSION
+// STB_VORBIS_NO_FAST_SCALED_FLOAT
+//      does not use a fast float-to-int trick to accelerate float-to-int on
+//      most platforms which requires endianness be defined correctly.
+//#define STB_VORBIS_NO_FAST_SCALED_FLOAT
+// STB_VORBIS_MAX_CHANNELS [number]
+//     globally define this to the maximum number of channels you need.
+//     The spec does not put a restriction on channels except that
+//     the count is stored in a byte, so 255 is the hard limit.
+//     Reducing this saves about 16 bytes per value, so using 16 saves
+//     (255-16)*16 or around 4KB. Plus anything other memory usage
+//     I forgot to account for. Can probably go as low as 8 (7.1 audio),
+//     6 (5.1 audio), or 2 (stereo only).
+#ifndef STB_VORBIS_MAX_CHANNELS
+#define STB_VORBIS_MAX_CHANNELS    16  // enough for anyone?
+#endif
+// STB_VORBIS_PUSHDATA_CRC_COUNT [number]
+//     after a flush_pushdata(), stb_vorbis begins scanning for the
+//     next valid page, without backtracking. when it finds something
+//     that looks like a page, it streams through it and verifies its
+//     CRC32. Should that validation fail, it keeps scanning. But it's
+//     possible that _while_ streaming through to check the CRC32 of
+//     one candidate page, it sees another candidate page. This #define
+//     determines how many "overlapping" candidate pages it can search
+//     at once. Note that "real" pages are typically ~4KB to ~8KB, whereas
+//     garbage pages could be as big as 64KB, but probably average ~16KB.
+//     So don't hose ourselves by scanning an apparent 64KB page and
+//     missing a ton of real ones in the interim; so minimum of 2
+#ifndef STB_VORBIS_PUSHDATA_CRC_COUNT
+#define STB_VORBIS_PUSHDATA_CRC_COUNT  4
+#endif
+// STB_VORBIS_FAST_HUFFMAN_LENGTH [number]
+//     sets the log size of the huffman-acceleration table.  Maximum
+//     supported value is 24. with larger numbers, more decodings are O(1),
+//     but the table size is larger so worse cache missing, so you'll have
+//     to probe (and try multiple ogg vorbis files) to find the sweet spot.
+#ifndef STB_VORBIS_FAST_HUFFMAN_LENGTH
+#define STB_VORBIS_FAST_HUFFMAN_LENGTH   10
+#endif
+// STB_VORBIS_FAST_BINARY_LENGTH [number]
+//     sets the log size of the binary-search acceleration table. this
+//     is used in similar fashion to the fast-huffman size to set initial
+//     parameters for the binary search
+// STB_VORBIS_FAST_HUFFMAN_INT
+//     The fast huffman tables are much more efficient if they can be
+//     stored as 16-bit results instead of 32-bit results. This restricts
+//     the codebooks to having only 65535 possible outcomes, though.
+//     (At least, accelerated by the huffman table.)
+#ifndef STB_VORBIS_FAST_HUFFMAN_INT
+#define STB_VORBIS_FAST_HUFFMAN_SHORT
+#endif
+// STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+//     If the 'fast huffman' search doesn't succeed, then stb_vorbis falls
+//     back on binary searching for the correct one. This requires storing
+//     extra tables with the huffman codes in sorted order. Defining this
+//     symbol trades off space for speed by forcing a linear search in the
+//     non-fast case, except for "sparse" codebooks.
+// #define STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+// STB_VORBIS_DIVIDES_IN_RESIDUE
+//     stb_vorbis precomputes the result of the scalar residue decoding
+//     that would otherwise require a divide per chunk. you can trade off
+//     space for time by defining this symbol.
+// #define STB_VORBIS_DIVIDES_IN_RESIDUE
+// STB_VORBIS_DIVIDES_IN_CODEBOOK
+//     vorbis VQ codebooks can be encoded two ways: with every case explicitly
+//     stored, or with all elements being chosen from a small range of values,
+//     and all values possible in all elements. By default, stb_vorbis expands
+//     this latter kind out to look like the former kind for ease of decoding,
+//     because otherwise an integer divide-per-vector-element is required to
+//     unpack the index. If you define STB_VORBIS_DIVIDES_IN_CODEBOOK, you can
+//     trade off storage for speed.
+//#define STB_VORBIS_DIVIDES_IN_CODEBOOK
+#ifdef STB_VORBIS_CODEBOOK_SHORTS
+#error "STB_VORBIS_CODEBOOK_SHORTS is no longer supported as it produced incorrect results for some input formats"
+#endif
+// STB_VORBIS_DIVIDE_TABLE
+//     this replaces small integer divides in the floor decode loop with
+//     table lookups. made less than 1% difference, so disabled by default.
+// STB_VORBIS_NO_INLINE_DECODE
+//     disables the inlining of the scalar codebook fast-huffman decode.
+//     might save a little codespace; useful for debugging
+// #define STB_VORBIS_NO_INLINE_DECODE
+// STB_VORBIS_NO_DEFER_FLOOR
+//     Normally we only decode the floor without synthesizing the actual
+//     full curve. We can instead synthesize the curve immediately. This
+//     requires more memory and is very likely slower, so I don't think
+//     you'd ever want to do it except for debugging.
+// #define STB_VORBIS_NO_DEFER_FLOOR
+//////////////////////////////////////////////////////////////////////////////
+#ifdef STB_VORBIS_NO_PULLDATA_API
+#define STB_VORBIS_NO_INTEGER_CONVERSION
+#define STB_VORBIS_NO_STDIO
+#endif
+#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
+#define STB_VORBIS_NO_STDIO 1
+#endif
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT
+// only need endianness for fast-float-to-int, which we don't
+// use for pushdata
+#ifndef STB_VORBIS_BIG_ENDIAN
+#define STB_VORBIS_ENDIAN  0
+#else
+#define STB_VORBIS_ENDIAN  1
+#endif
+#endif
+#endif
+#ifndef STB_VORBIS_NO_STDIO
+#include <stdio.h>
+#endif
+#ifndef STB_VORBIS_NO_CRT
+#include <stdlib.h>
+#include <string.h>
+#include <assert.h>
+#include <math.h>
+// find definition of alloca if it's not in stdlib.h:
+#if defined(_MSC_VER) || defined(__MINGW32__)
+#include <malloc.h>
+#endif
+#if defined(__linux__) || defined(__linux) || defined(__sun__) || defined(__EMSCRIPTEN__) || defined(__NEWLIB__)
+#include <alloca.h>
+#endif
+#else // STB_VORBIS_NO_CRT
+#define NULL 0
+#define malloc(s)   0
+#define free(s)     ((void) 0)
+#define realloc(s)  0
+#endif // STB_VORBIS_NO_CRT
+#include <limits.h>
+#ifdef __MINGW32__
+// eff you mingw:
+//     "fixed":
+//         http://sourceforge.net/p/mingw-w64/mailman/message/32882927/
+//     "no that broke the build, reverted, who cares about C":
+//         http://sourceforge.net/p/mingw-w64/mailman/message/32890381/
+#ifdef __forceinline
+#undef __forceinline
+#endif
+#define __forceinline
+#ifndef alloca
+#define alloca __builtin_alloca
+#endif
+#elif !defined(_MSC_VER)
+#if __GNUC__
+#define __forceinline inline
+#else
+#define __forceinline
+#endif
+#endif
+#if STB_VORBIS_MAX_CHANNELS > 256
+#error "Value of STB_VORBIS_MAX_CHANNELS outside of allowed range"
+#endif
+#if STB_VORBIS_FAST_HUFFMAN_LENGTH > 24
+#error "Value of STB_VORBIS_FAST_HUFFMAN_LENGTH outside of allowed range"
+#endif
+#if 0
+#include <crtdbg.h>
+#define CHECK(f)   _CrtIsValidHeapPointer(f->channel_buffers[1])
+#else
+#define CHECK(f)   ((void) 0)
+#endif
+#define MAX_BLOCKSIZE_LOG  13   // from specification
+#define MAX_BLOCKSIZE      (1 << MAX_BLOCKSIZE_LOG)
+typedef unsigned char  uint8;
+typedef   signed char   int8;
+typedef unsigned short uint16;
+typedef   signed short  int16;
+typedef unsigned int   uint32;
+typedef   signed int    int32;
+#ifndef TRUE
+#define TRUE 1
+#define FALSE 0
+#endif
+typedef float codetype;
+#ifdef _MSC_VER
+#define STBV_NOTUSED(v)  (void)(v)
+#else
+#define STBV_NOTUSED(v)  (void)sizeof(v)
+#endif
+// @NOTE
+//
+// Some arrays below are tagged "//varies", which means it's actually
+// a variable-sized piece of data, but rather than malloc I assume it's
+// small enough it's better to just allocate it all together with the
+// main thing
+//
+// Most of the variables are specified with the smallest size I could pack
+// them into. It might give better performance to make them all full-sized
+// integers. It should be safe to freely rearrange the structures or change
+// the sizes larger--nothing relies on silently truncating etc., nor the
+// order of variables.
+#define FAST_HUFFMAN_TABLE_SIZE   (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH)
+#define FAST_HUFFMAN_TABLE_MASK   (FAST_HUFFMAN_TABLE_SIZE - 1)
+typedef struct
+{
+int dimensions, entries;
+uint8 *codeword_lengths;
+float  minimum_value;
+float  delta_value;
+uint8  value_bits;
+uint8  lookup_type;
+uint8  sequence_p;
+uint8  sparse;
+uint32 lookup_values;
+codetype *multiplicands;
+uint32 *codewords;
+#ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
+int16  fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
+#else
+int32  fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
+#endif
+uint32 *sorted_codewords;
+int    *sorted_values;
+int     sorted_entries;
+} Codebook;
+typedef struct
+{
+uint8 order;
+uint16 rate;
+uint16 bark_map_size;
+uint8 amplitude_bits;
+uint8 amplitude_offset;
+uint8 number_of_books;
+uint8 book_list[16]; // varies
+} Floor0;
+typedef struct
+{
+uint8 partitions;
+uint8 partition_class_list[32]; // varies
+uint8 class_dimensions[16]; // varies
+uint8 class_subclasses[16]; // varies
+uint8 class_masterbooks[16]; // varies
+int16 subclass_books[16][8]; // varies
+uint16 Xlist[31*8+2]; // varies
+uint8 sorted_order[31*8+2];
+uint8 neighbors[31*8+2][2];
+uint8 floor1_multiplier;
+uint8 rangebits;
+int values;
+} Floor1;
+typedef union
+{
+Floor0 floor0;
+Floor1 floor1;
+} Floor;
+typedef struct
+{
+uint32 begin, end;
+uint32 part_size;
+uint8 classifications;
+uint8 classbook;
+uint8 **classdata;
+int16 (*residue_books)[8];
+} Residue;
+typedef struct
+{
+uint8 magnitude;
+uint8 angle;
+uint8 mux;
+} MappingChannel;
+typedef struct
+{
+uint16 coupling_steps;
+MappingChannel *chan;
+uint8  submaps;
+uint8  submap_floor[15]; // varies
+uint8  submap_residue[15]; // varies
+} Mapping;
+typedef struct
+{
+uint8 blockflag;
+uint8 mapping;
+uint16 windowtype;
+uint16 transformtype;
+} Mode;
+typedef struct
+{
+uint32  goal_crc;    // expected crc if match
+int     bytes_left;  // bytes left in packet
+uint32  crc_so_far;  // running crc
+int     bytes_done;  // bytes processed in _current_ chunk
+uint32  sample_loc;  // granule pos encoded in page
+} CRCscan;
+typedef struct
+{
+uint32 page_start, page_end;
+uint32 last_decoded_sample;
+} ProbedPage;
+struct stb_vorbis
+{
+// user-accessible info
+unsigned int sample_rate;
+int channels;
+unsigned int setup_memory_required;
+unsigned int temp_memory_required;
+unsigned int setup_temp_memory_required;
+char *vendor;
+int comment_list_length;
+char **comment_list;
+// input config
+#ifndef STB_VORBIS_NO_STDIO
+FILE *f;
+uint32 f_start;
+int close_on_free;
+#endif
+uint8 *stream;
+uint8 *stream_start;
+uint8 *stream_end;
+uint32 stream_len;
+uint8  push_mode;
+// the page to seek to when seeking to start, may be zero
+uint32 first_audio_page_offset;
+// p_first is the page on which the first audio packet ends
+// (but not necessarily the page on which it starts)
+ProbedPage p_first, p_last;
+// memory management
+stb_vorbis_alloc alloc;
+int setup_offset;
+int temp_offset;
+// run-time results
+int eof;
+enum STBVorbisError error;
+// user-useful data
+// header info
+int blocksize[2];
+int blocksize_0, blocksize_1;
+int codebook_count;
+Codebook *codebooks;
+int floor_count;
+uint16 floor_types[64]; // varies
+Floor *floor_config;
+int residue_count;
+uint16 residue_types[64]; // varies
+Residue *residue_config;
+int mapping_count;
+Mapping *mapping;
+int mode_count;
+Mode mode_config[64];  // varies
+uint32 total_samples;
+// decode buffer
+float *channel_buffers[STB_VORBIS_MAX_CHANNELS];
+float *outputs        [STB_VORBIS_MAX_CHANNELS];
+float *previous_window[STB_VORBIS_MAX_CHANNELS];
+int previous_length;
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+int16 *finalY[STB_VORBIS_MAX_CHANNELS];
+#else
+float *floor_buffers[STB_VORBIS_MAX_CHANNELS];
+#endif
+uint32 current_loc; // sample location of next frame to decode
+int    current_loc_valid;
+// per-blocksize precomputed data
+// twiddle factors
+float *A[2],*B[2],*C[2];
+float *window[2];
+uint16 *bit_reverse[2];
+// current page/packet/segment streaming info
+uint32 serial; // stream serial number for verification
+int last_page;
+int segment_count;
+uint8 segments[255];
+uint8 page_flag;
+uint8 bytes_in_seg;
+uint8 first_decode;
+int next_seg;
+int last_seg;  // flag that we're on the last segment
+int last_seg_which; // what was the segment number of the last seg?
+uint32 acc;
+int valid_bits;
+int packet_bytes;
+int end_seg_with_known_loc;
+uint32 known_loc_for_packet;
+int discard_samples_deferred;
+uint32 samples_output;
+// push mode scanning
+int page_crc_tests; // only in push_mode: number of tests active; -1 if not searching
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+CRCscan scan[STB_VORBIS_PUSHDATA_CRC_COUNT];
+#endif
+// sample-access
+int channel_buffer_start;
+int channel_buffer_end;
+};
+#if defined(STB_VORBIS_NO_PUSHDATA_API)
+#define IS_PUSH_MODE(f)   FALSE
+#elif defined(STB_VORBIS_NO_PULLDATA_API)
+#define IS_PUSH_MODE(f)   TRUE
+#else
+#define IS_PUSH_MODE(f)   ((f)->push_mode)
+#endif
+typedef struct stb_vorbis vorb;
+static int error(vorb *f, enum STBVorbisError e)
+{
+f->error = e;
+if (!f->eof && e != VORBIS_need_more_data) {
+f->error=e; // breakpoint for debugging
+}
+return 0;
+}
+// these functions are used for allocating temporary memory
+// while decoding. if you can afford the stack space, use
+// alloca(); otherwise, provide a temp buffer and it will
+// allocate out of those.
+#define array_size_required(count,size)  (count*(sizeof(void *)+(size)))
+#define temp_alloc(f,size)              (f->alloc.alloc_buffer ? setup_temp_malloc(f,size) : alloca(size))
+#define temp_free(f,p)                  (void)0
+#define temp_alloc_save(f)              ((f)->temp_offset)
+#define temp_alloc_restore(f,p)         ((f)->temp_offset = (p))
+#define temp_block_array(f,count,size)  make_block_array(temp_alloc(f,array_size_required(count,size)), count, size)
+// given a sufficiently large block of memory, make an array of pointers to subblocks of it
+static void *make_block_array(void *mem, int count, int size)
+{
+int i;
+void ** p = (void **) mem;
+char *q = (char *) (p + count);
+for (i=0; i < count; ++i) {
+p[i] = q;
+q += size;
+}
+return p;
+}
+static void *setup_malloc(vorb *f, int sz)
+{
+sz = (sz+7) & ~7; // round up to nearest 8 for alignment of future allocs.
+f->setup_memory_required += sz;
+if (f->alloc.alloc_buffer) {
+void *p = (char *) f->alloc.alloc_buffer + f->setup_offset;
+if (f->setup_offset + sz > f->temp_offset) return NULL;
+f->setup_offset += sz;
+return p;
+}
+return sz ? malloc(sz) : NULL;
+}
+static void setup_free(vorb *f, void *p)
+{
+if (f->alloc.alloc_buffer) return; // do nothing; setup mem is a stack
+free(p);
+}
+static void *setup_temp_malloc(vorb *f, int sz)
+{
+sz = (sz+7) & ~7; // round up to nearest 8 for alignment of future allocs.
+if (f->alloc.alloc_buffer) {
+if (f->temp_offset - sz < f->setup_offset) return NULL;
+f->temp_offset -= sz;
+return (char *) f->alloc.alloc_buffer + f->temp_offset;
+}
+return malloc(sz);
+}
+static void setup_temp_free(vorb *f, void *p, int sz)
+{
+if (f->alloc.alloc_buffer) {
+f->temp_offset += (sz+7)&~7;
+return;
+}
+free(p);
+}
+#define CRC32_POLY    0x04c11db7   // from spec
+static uint32 crc_table[256];
+static void crc32_init(void)
+{
+int i,j;
+uint32 s;
+for(i=0; i < 256; i++) {
+for (s=(uint32) i << 24, j=0; j < 8; ++j)
+s = (s << 1) ^ (s >= (1U<<31) ? CRC32_POLY : 0);
+crc_table[i] = s;
+}
+}
+static __forceinline uint32 crc32_update(uint32 crc, uint8 byte)
+{
+return (crc << 8) ^ crc_table[byte ^ (crc >> 24)];
+}
+// used in setup, and for huffman that doesn't go fast path
+static unsigned int bit_reverse(unsigned int n)
+{
+n = ((n & 0xAAAAAAAA) >>  1) | ((n & 0x55555555) << 1);
+n = ((n & 0xCCCCCCCC) >>  2) | ((n & 0x33333333) << 2);
+n = ((n & 0xF0F0F0F0) >>  4) | ((n & 0x0F0F0F0F) << 4);
+n = ((n & 0xFF00FF00) >>  8) | ((n & 0x00FF00FF) << 8);
+return (n >> 16) | (n << 16);
+}
+static float square(float x)
+{
+return x*x;
+}
+// this is a weird definition of log2() for which log2(1) = 1, log2(2) = 2, log2(4) = 3
+// as required by the specification. fast(?) implementation from stb.h
+// @OPTIMIZE: called multiple times per-packet with "constants"; move to setup
+static int ilog(int32 n)
+{
+static signed char log2_4[16] = { 0,1,2,2,3,3,3,3,4,4,4,4,4,4,4,4 };
+if (n < 0) return 0; // signed n returns 0
+// 2 compares if n < 16, 3 compares otherwise (4 if signed or n > 1<<29)
+if (n < (1 << 14))
+if (n < (1 <<  4))            return  0 + log2_4[n      ];
+else if (n < (1 <<  9))       return  5 + log2_4[n >>  5];
+else                     return 10 + log2_4[n >> 10];
+else if (n < (1 << 24))
+if (n < (1 << 19))       return 15 + log2_4[n >> 15];
+else                     return 20 + log2_4[n >> 20];
+else if (n < (1 << 29))       return 25 + log2_4[n >> 25];
+else                     return 30 + log2_4[n >> 30];
+}
+#ifndef M_PI
+#define M_PI  3.14159265358979323846264f  // from CRC
+#endif
+// code length assigned to a value with no huffman encoding
+#define NO_CODE   255
+/////////////////////// LEAF SETUP FUNCTIONS //////////////////////////
+//
+// these functions are only called at setup, and only a few times
+// per file
+static float float32_unpack(uint32 x)
+{
+// from the specification
+uint32 mantissa = x & 0x1fffff;
+uint32 sign = x & 0x80000000;
+uint32 exp = (x & 0x7fe00000) >> 21;
+double res = sign ? -(double)mantissa : (double)mantissa;
+return (float) ldexp((float)res, (int)exp-788);
+}
+// zlib & jpeg huffman tables assume that the output symbols
+// can either be arbitrarily arranged, or have monotonically
+// increasing frequencies--they rely on the lengths being sorted;
+// this makes for a very simple generation algorithm.
+// vorbis allows a huffman table with non-sorted lengths. This
+// requires a more sophisticated construction, since symbols in
+// order do not map to huffman codes "in order".
+static void add_entry(Codebook *c, uint32 huff_code, int symbol, int count, int len, uint32 *values)
+{
+if (!c->sparse) {
+c->codewords      [symbol] = huff_code;
+} else {
+c->codewords       [count] = huff_code;
+c->codeword_lengths[count] = len;
+values             [count] = symbol;
+}
+}
+static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values)
+{
+int i,k,m=0;
+uint32 available[32];
+memset(available, 0, sizeof(available));
+// find the first entry
+for (k=0; k < n; ++k) if (len[k] < NO_CODE) break;
+if (k == n) { assert(c->sorted_entries == 0); return TRUE; }
+assert(len[k] < 32); // no error return required, code reading lens checks this
+// add to the list
+add_entry(c, 0, k, m++, len[k], values);
+// add all available leaves
+for (i=1; i <= len[k]; ++i)
+available[i] = 1U << (32-i);
+// note that the above code treats the first case specially,
+// but it's really the same as the following code, so they
+// could probably be combined (except the initial code is 0,
+// and I use 0 in available[] to mean 'empty')
+for (i=k+1; i < n; ++i) {
+uint32 res;
+int z = len[i], y;
+if (z == NO_CODE) continue;
+assert(z < 32); // no error return required, code reading lens checks this
+// find lowest available leaf (should always be earliest,
+// which is what the specification calls for)
+// note that this property, and the fact we can never have
+// more than one free leaf at a given level, isn't totally
+// trivial to prove, but it seems true and the assert never
+// fires, so!
+while (z > 0 && !available[z]) --z;
+if (z == 0) { return FALSE; }
+res = available[z];
+available[z] = 0;
+add_entry(c, bit_reverse(res), i, m++, len[i], values);
+// propagate availability up the tree
+if (z != len[i]) {
+for (y=len[i]; y > z; --y) {
+assert(available[y] == 0);
+available[y] = res + (1 << (32-y));
+}
+}
+}
+return TRUE;
+}
+// accelerated huffman table allows fast O(1) match of all symbols
+// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH
+static void compute_accelerated_huffman(Codebook *c)
+{
+int i, len;
+for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i)
+c->fast_huffman[i] = -1;
+len = c->sparse ? c->sorted_entries : c->entries;
+#ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
+if (len > 32767) len = 32767; // largest possible value we can encode!
+#endif
+for (i=0; i < len; ++i) {
+if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) {
+uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i];
+// set table entries for all bit combinations in the higher bits
+while (z < FAST_HUFFMAN_TABLE_SIZE) {
+c->fast_huffman[z] = i;
+z += 1 << c->codeword_lengths[i];
+}
+}
+}
+}
+#ifdef _MSC_VER
+#define STBV_CDECL __cdecl
+#else
+#define STBV_CDECL
+#endif
+static int STBV_CDECL uint32_compare(const void *p, const void *q)
+{
+uint32 x = * (uint32 *) p;
+uint32 y = * (uint32 *) q;
+return x < y ? -1 : x > y;
+}
+static int include_in_sort(Codebook *c, uint8 len)
+{
+if (c->sparse) { assert(len != NO_CODE); return TRUE; }
+if (len == NO_CODE) return FALSE;
+if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE;
+return FALSE;
+}
+// if the fast table above doesn't work, we want to binary
+// search them... need to reverse the bits
+static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values)
+{
+int i, len;
+// build a list of all the entries
+// OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN.
+// this is kind of a frivolous optimization--I don't see any performance improvement,
+// but it's like 4 extra lines of code, so.
+if (!c->sparse) {
+int k = 0;
+for (i=0; i < c->entries; ++i)
+if (include_in_sort(c, lengths[i]))
+c->sorted_codewords[k++] = bit_reverse(c->codewords[i]);
+assert(k == c->sorted_entries);
+} else {
+for (i=0; i < c->sorted_entries; ++i)
+c->sorted_codewords[i] = bit_reverse(c->codewords[i]);
+}
+qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare);
+c->sorted_codewords[c->sorted_entries] = 0xffffffff;
+len = c->sparse ? c->sorted_entries : c->entries;
+// now we need to indicate how they correspond; we could either
+//   #1: sort a different data structure that says who they correspond to
+//   #2: for each sorted entry, search the original list to find who corresponds
+//   #3: for each original entry, find the sorted entry
+// #1 requires extra storage, #2 is slow, #3 can use binary search!
+for (i=0; i < len; ++i) {
+int huff_len = c->sparse ? lengths[values[i]] : lengths[i];
+if (include_in_sort(c,huff_len)) {
+uint32 code = bit_reverse(c->codewords[i]);
+int x=0, n=c->sorted_entries;
+while (n > 1) {
+// invariant: sc[x] <= code < sc[x+n]
+int m = x + (n >> 1);
+if (c->sorted_codewords[m] <= code) {
+x = m;
+n -= (n>>1);
+} else {
+n >>= 1;
+}
+}
+assert(c->sorted_codewords[x] == code);
+if (c->sparse) {
+c->sorted_values[x] = values[i];
+c->codeword_lengths[x] = huff_len;
+} else {
+c->sorted_values[x] = i;
+}
+}
+}
+}
+// only run while parsing the header (3 times)
+static int vorbis_validate(uint8 *data)
+{
+static uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' };
+return memcmp(data, vorbis, 6) == 0;
+}
+// called from setup only, once per code book
+// (formula implied by specification)
+static int lookup1_values(int entries, int dim)
+{
+int r = (int) floor(exp((float) log((float) entries) / dim));
+if ((int) floor(pow((float) r+1, dim)) <= entries)   // (int) cast for MinGW warning;
+++r;                                              // floor() to avoid _ftol() when non-CRT
+if (pow((float) r+1, dim) <= entries)
+return -1;
+if ((int) floor(pow((float) r, dim)) > entries)
+return -1;
+return r;
+}
+// called twice per file
+static void compute_twiddle_factors(int n, float *A, float *B, float *C)
+{
+int n4 = n >> 2, n8 = n >> 3;
+int k,k2;
+for (k=k2=0; k < n4; ++k,k2+=2) {
+A[k2  ] = (float)  cos(4*k*M_PI/n);
+A[k2+1] = (float) -sin(4*k*M_PI/n);
+B[k2  ] = (float)  cos((k2+1)*M_PI/n/2) * 0.5f;
+B[k2+1] = (float)  sin((k2+1)*M_PI/n/2) * 0.5f;
+}
+for (k=k2=0; k < n8; ++k,k2+=2) {
+C[k2  ] = (float)  cos(2*(k2+1)*M_PI/n);
+C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n);
+}
+}
+static void compute_window(int n, float *window)
+{
+int n2 = n >> 1, i;
+for (i=0; i < n2; ++i)
+window[i] = (float) sin(0.5 * M_PI * square((float) sin((i - 0 + 0.5) / n2 * 0.5 * M_PI)));
+}
+static void compute_bitreverse(int n, uint16 *rev)
+{
+int ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
+int i, n8 = n >> 3;
+for (i=0; i < n8; ++i)
+rev[i] = (bit_reverse(i) >> (32-ld+3)) << 2;
+}
+static int init_blocksize(vorb *f, int b, int n)
+{
+int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3;
+f->A[b] = (float *) setup_malloc(f, sizeof(float) * n2);
+f->B[b] = (float *) setup_malloc(f, sizeof(float) * n2);
+f->C[b] = (float *) setup_malloc(f, sizeof(float) * n4);
+if (!f->A[b] || !f->B[b] || !f->C[b]) return error(f, VORBIS_outofmem);
+compute_twiddle_factors(n, f->A[b], f->B[b], f->C[b]);
+f->window[b] = (float *) setup_malloc(f, sizeof(float) * n2);
+if (!f->window[b]) return error(f, VORBIS_outofmem);
+compute_window(n, f->window[b]);
+f->bit_reverse[b] = (uint16 *) setup_malloc(f, sizeof(uint16) * n8);
+if (!f->bit_reverse[b]) return error(f, VORBIS_outofmem);
+compute_bitreverse(n, f->bit_reverse[b]);
+return TRUE;
+}
+static void neighbors(uint16 *x, int n, int *plow, int *phigh)
+{
+int low = -1;
+int high = 65536;
+int i;
+for (i=0; i < n; ++i) {
+if (x[i] > low  && x[i] < x[n]) { *plow  = i; low = x[i]; }
+if (x[i] < high && x[i] > x[n]) { *phigh = i; high = x[i]; }
+}
+}
+// this has been repurposed so y is now the original index instead of y
+typedef struct
+{
+uint16 x,id;
+} stbv__floor_ordering;
+static int STBV_CDECL point_compare(const void *p, const void *q)
+{
+stbv__floor_ordering *a = (stbv__floor_ordering *) p;
+stbv__floor_ordering *b = (stbv__floor_ordering *) q;
+return a->x < b->x ? -1 : a->x > b->x;
+}
+//
+/////////////////////// END LEAF SETUP FUNCTIONS //////////////////////////
+#if defined(STB_VORBIS_NO_STDIO)
+#define USE_MEMORY(z)    TRUE
+#else
+#define USE_MEMORY(z)    ((z)->stream)
+#endif
+static uint8 get8(vorb *z)
+{
+if (USE_MEMORY(z)) {
+if (z->stream >= z->stream_end) { z->eof = TRUE; return 0; }
+return *z->stream++;
+}
+#ifndef STB_VORBIS_NO_STDIO
+{
+int c = fgetc(z->f);
+if (c == EOF) { z->eof = TRUE; return 0; }
+return c;
+}
+#endif
+}
+static uint32 get32(vorb *f)
+{
+uint32 x;
+x = get8(f);
+x += get8(f) << 8;
+x += get8(f) << 16;
+x += (uint32) get8(f) << 24;
+return x;
+}
+static int getn(vorb *z, uint8 *data, int n)
+{
+if (USE_MEMORY(z)) {
+if (z->stream+n > z->stream_end) { z->eof = 1; return 0; }
+memcpy(data, z->stream, n);
+z->stream += n;
+return 1;
+}
+#ifndef STB_VORBIS_NO_STDIO
+if (fread(data, n, 1, z->f) == 1)
+return 1;
+else {
+z->eof = 1;
+return 0;
+}
+#endif
+}
+static void skip(vorb *z, int n)
+{
+if (USE_MEMORY(z)) {
+z->stream += n;
+if (z->stream >= z->stream_end) z->eof = 1;
+return;
+}
+#ifndef STB_VORBIS_NO_STDIO
+{
+long x = ftell(z->f);
+fseek(z->f, x+n, SEEK_SET);
+}
+#endif
+}
+static int set_file_offset(stb_vorbis *f, unsigned int loc)
+{
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+if (f->push_mode) return 0;
+#endif
+f->eof = 0;
+if (USE_MEMORY(f)) {
+if (f->stream_start + loc >= f->stream_end || f->stream_start + loc < f->stream_start) {
+f->stream = f->stream_end;
+f->eof = 1;
+return 0;
+} else {
+f->stream = f->stream_start + loc;
+return 1;
+}
+}
+#ifndef STB_VORBIS_NO_STDIO
+if (loc + f->f_start < loc || loc >= 0x80000000) {
+loc = 0x7fffffff;
+f->eof = 1;
+} else {
+loc += f->f_start;
+}
+if (!fseek(f->f, loc, SEEK_SET))
+return 1;
+f->eof = 1;
+fseek(f->f, f->f_start, SEEK_END);
+return 0;
+#endif
+}
+static uint8 ogg_page_header[4] = { 0x4f, 0x67, 0x67, 0x53 };
+static int capture_pattern(vorb *f)
+{
+if (0x4f != get8(f)) return FALSE;
+if (0x67 != get8(f)) return FALSE;
+if (0x67 != get8(f)) return FALSE;
+if (0x53 != get8(f)) return FALSE;
+return TRUE;
+}
+#define PAGEFLAG_continued_packet   1
+#define PAGEFLAG_first_page         2
+#define PAGEFLAG_last_page          4
+static int start_page_no_capturepattern(vorb *f)
+{
+uint32 loc0,loc1,n;
+if (f->first_decode && !IS_PUSH_MODE(f)) {
+f->p_first.page_start = stb_vorbis_get_file_offset(f) - 4;
+}
+// stream structure version
+if (0 != get8(f)) return error(f, VORBIS_invalid_stream_structure_version);
+// header flag
+f->page_flag = get8(f);
+// absolute granule position
+loc0 = get32(f);
+loc1 = get32(f);
+// @TODO: validate loc0,loc1 as valid positions?
+// stream serial number -- vorbis doesn't interleave, so discard
+get32(f);
+//if (f->serial != get32(f)) return error(f, VORBIS_incorrect_stream_serial_number);
+// page sequence number
+n = get32(f);
+f->last_page = n;
+// CRC32
+get32(f);
+// page_segments
+f->segment_count = get8(f);
+if (!getn(f, f->segments, f->segment_count))
+return error(f, VORBIS_unexpected_eof);
+// assume we _don't_ know any the sample position of any segments
+f->end_seg_with_known_loc = -2;
+if (loc0 != ~0U || loc1 != ~0U) {
+int i;
+// determine which packet is the last one that will complete
+for (i=f->segment_count-1; i >= 0; --i)
+if (f->segments[i] < 255)
+break;
+// 'i' is now the index of the _last_ segment of a packet that ends
+if (i >= 0) {
+f->end_seg_with_known_loc = i;
+f->known_loc_for_packet   = loc0;
+}
+}
+if (f->first_decode) {
+int i,len;
+len = 0;
+for (i=0; i < f->segment_count; ++i)
+len += f->segments[i];
+len += 27 + f->segment_count;
+f->p_first.page_end = f->p_first.page_start + len;
+f->p_first.last_decoded_sample = loc0;
+}
+f->next_seg = 0;
+return TRUE;
+}
+static int start_page(vorb *f)
+{
+if (!capture_pattern(f)) return error(f, VORBIS_missing_capture_pattern);
+return start_page_no_capturepattern(f);
+}
+static int start_packet(vorb *f)
+{
+while (f->next_seg == -1) {
+if (!start_page(f)) return FALSE;
+if (f->page_flag & PAGEFLAG_continued_packet)
+return error(f, VORBIS_continued_packet_flag_invalid);
+}
+f->last_seg = FALSE;
+f->valid_bits = 0;
+f->packet_bytes = 0;
+f->bytes_in_seg = 0;
+// f->next_seg is now valid
+return TRUE;
+}
+static int maybe_start_packet(vorb *f)
+{
+if (f->next_seg == -1) {
+int x = get8(f);
+if (f->eof) return FALSE; // EOF at page boundary is not an error!
+if (0x4f != x      ) return error(f, VORBIS_missing_capture_pattern);
+if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+if (0x53 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+if (!start_page_no_capturepattern(f)) return FALSE;
+if (f->page_flag & PAGEFLAG_continued_packet) {
+// set up enough state that we can read this packet if we want,
+// e.g. during recovery
+f->last_seg = FALSE;
+f->bytes_in_seg = 0;
+return error(f, VORBIS_continued_packet_flag_invalid);
+}
+}
+return start_packet(f);
+}
+static int next_segment(vorb *f)
+{
+int len;
+if (f->last_seg) return 0;
+if (f->next_seg == -1) {
+f->last_seg_which = f->segment_count-1; // in case start_page fails
+if (!start_page(f)) { f->last_seg = 1; return 0; }
+if (!(f->page_flag & PAGEFLAG_continued_packet)) return error(f, VORBIS_continued_packet_flag_invalid);
+}
+len = f->segments[f->next_seg++];
+if (len < 255) {
+f->last_seg = TRUE;
+f->last_seg_which = f->next_seg-1;
+}
+if (f->next_seg >= f->segment_count)
+f->next_seg = -1;
+assert(f->bytes_in_seg == 0);
+f->bytes_in_seg = len;
+return len;
+}
+#define EOP    (-1)
+#define INVALID_BITS  (-1)
+static int get8_packet_raw(vorb *f)
+{
+if (!f->bytes_in_seg) {  // CLANG!
+if (f->last_seg) return EOP;
+else if (!next_segment(f)) return EOP;
+}
+assert(f->bytes_in_seg > 0);
+--f->bytes_in_seg;
+++f->packet_bytes;
+return get8(f);
+}
+static int get8_packet(vorb *f)
+{
+int x = get8_packet_raw(f);
+f->valid_bits = 0;
+return x;
+}
+static int get32_packet(vorb *f)
+{
+uint32 x;
+x = get8_packet(f);
+x += get8_packet(f) << 8;
+x += get8_packet(f) << 16;
+x += (uint32) get8_packet(f) << 24;
+return x;
+}
+static void flush_packet(vorb *f)
+{
+while (get8_packet_raw(f) != EOP);
+}
+// @OPTIMIZE: this is the secondary bit decoder, so it's probably not as important
+// as the huffman decoder?
+static uint32 get_bits(vorb *f, int n)
+{
+uint32 z;
+if (f->valid_bits < 0) return 0;
+if (f->valid_bits < n) {
+if (n > 24) {
+// the accumulator technique below would not work correctly in this case
+z = get_bits(f, 24);
+z += get_bits(f, n-24) << 24;
+return z;
+}
+if (f->valid_bits == 0) f->acc = 0;
+while (f->valid_bits < n) {
+int z = get8_packet_raw(f);
+if (z == EOP) {
+f->valid_bits = INVALID_BITS;
+return 0;
+}
+f->acc += z << f->valid_bits;
+f->valid_bits += 8;
+}
+}
+assert(f->valid_bits >= n);
+z = f->acc & ((1 << n)-1);
+f->acc >>= n;
+f->valid_bits -= n;
+return z;
+}
+// @OPTIMIZE: primary accumulator for huffman
+// expand the buffer to as many bits as possible without reading off end of packet
+// it might be nice to allow f->valid_bits and f->acc to be stored in registers,
+// e.g. cache them locally and decode locally
+static __forceinline void prep_huffman(vorb *f)
+{
+if (f->valid_bits <= 24) {
+if (f->valid_bits == 0) f->acc = 0;
+do {
+int z;
+if (f->last_seg && !f->bytes_in_seg) return;
+z = get8_packet_raw(f);
+if (z == EOP) return;
+f->acc += (unsigned) z << f->valid_bits;
+f->valid_bits += 8;
+} while (f->valid_bits <= 24);
+}
+}
+enum
+{
+VORBIS_packet_id = 1,
+VORBIS_packet_comment = 3,
+VORBIS_packet_setup = 5
+};
+static int codebook_decode_scalar_raw(vorb *f, Codebook *c)
+{
+int i;
+prep_huffman(f);
+if (c->codewords == NULL && c->sorted_codewords == NULL)
+return -1;
+// cases to use binary search: sorted_codewords && !c->codewords
+//                             sorted_codewords && c->entries > 8
+if (c->entries > 8 ? c->sorted_codewords!=NULL : !c->codewords) {
+// binary search
+uint32 code = bit_reverse(f->acc);
+int x=0, n=c->sorted_entries, len;
+while (n > 1) {
+// invariant: sc[x] <= code < sc[x+n]
+int m = x + (n >> 1);
+if (c->sorted_codewords[m] <= code) {
+x = m;
+n -= (n>>1);
+} else {
+n >>= 1;
+}
+}
+// x is now the sorted index
+if (!c->sparse) x = c->sorted_values[x];
+// x is now sorted index if sparse, or symbol otherwise
+len = c->codeword_lengths[x];
+if (f->valid_bits >= len) {
+f->acc >>= len;
+f->valid_bits -= len;
+return x;
+}
+f->valid_bits = 0;
+return -1;
+}
+// if small, linear search
+assert(!c->sparse);
+for (i=0; i < c->entries; ++i) {
+if (c->codeword_lengths[i] == NO_CODE) continue;
+if (c->codewords[i] == (f->acc & ((1 << c->codeword_lengths[i])-1))) {
+if (f->valid_bits >= c->codeword_lengths[i]) {
+f->acc >>= c->codeword_lengths[i];
+f->valid_bits -= c->codeword_lengths[i];
+return i;
+}
+f->valid_bits = 0;
+return -1;
+}
+}
+error(f, VORBIS_invalid_stream);
+f->valid_bits = 0;
+return -1;
+}
+#ifndef STB_VORBIS_NO_INLINE_DECODE
+#define DECODE_RAW(var, f,c)                                  \
+if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH)        \
+prep_huffman(f);                                        \
+var = f->acc & FAST_HUFFMAN_TABLE_MASK;                    \
+var = c->fast_huffman[var];                                \
+if (var >= 0) {                                            \
+int n = c->codeword_lengths[var];                       \
+f->acc >>= n;                                           \
+f->valid_bits -= n;                                     \
+if (f->valid_bits < 0) { f->valid_bits = 0; var = -1; } \
+} else {                                                   \
+var = codebook_decode_scalar_raw(f,c);                  \
+}
+#else
+static int codebook_decode_scalar(vorb *f, Codebook *c)
+{
+int i;
+if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH)
+prep_huffman(f);
+// fast huffman table lookup
+i = f->acc & FAST_HUFFMAN_TABLE_MASK;
+i = c->fast_huffman[i];
+if (i >= 0) {
+f->acc >>= c->codeword_lengths[i];
+f->valid_bits -= c->codeword_lengths[i];
+if (f->valid_bits < 0) { f->valid_bits = 0; return -1; }
+return i;
+}
+return codebook_decode_scalar_raw(f,c);
+}
+#define DECODE_RAW(var,f,c)    var = codebook_decode_scalar(f,c);
+#endif
+#define DECODE(var,f,c)                                       \
+DECODE_RAW(var,f,c)                                        \
+if (c->sparse) var = c->sorted_values[var];
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+#define DECODE_VQ(var,f,c)   DECODE_RAW(var,f,c)
+#else
+#define DECODE_VQ(var,f,c)   DECODE(var,f,c)
+#endif
+// CODEBOOK_ELEMENT_FAST is an optimization for the CODEBOOK_FLOATS case
+// where we avoid one addition
+#define CODEBOOK_ELEMENT(c,off)          (c->multiplicands[off])
+#define CODEBOOK_ELEMENT_FAST(c,off)     (c->multiplicands[off])
+#define CODEBOOK_ELEMENT_BASE(c)         (0)
+static int codebook_decode_start(vorb *f, Codebook *c)
+{
+int z = -1;
+// type 0 is only legal in a scalar context
+if (c->lookup_type == 0)
+error(f, VORBIS_invalid_stream);
+else {
+DECODE_VQ(z,f,c);
+if (c->sparse) assert(z < c->sorted_entries);
+if (z < 0) {  // check for EOP
+if (!f->bytes_in_seg)
+if (f->last_seg)
+return z;
+error(f, VORBIS_invalid_stream);
+}
+}
+return z;
+}
+static int codebook_decode(vorb *f, Codebook *c, float *output, int len)
+{
+int i,z = codebook_decode_start(f,c);
+if (z < 0) return FALSE;
+if (len > c->dimensions) len = c->dimensions;
+#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+if (c->lookup_type == 1) {
+float last = CODEBOOK_ELEMENT_BASE(c);
+int div = 1;
+for (i=0; i < len; ++i) {
+int off = (z / div) % c->lookup_values;
+float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
+output[i] += val;
+if (c->sequence_p) last = val + c->minimum_value;
+div *= c->lookup_values;
+}
+return TRUE;
+}
+#endif
+z *= c->dimensions;
+if (c->sequence_p) {
+float last = CODEBOOK_ELEMENT_BASE(c);
+for (i=0; i < len; ++i) {
+float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+output[i] += val;
+last = val + c->minimum_value;
+}
+} else {
+float last = CODEBOOK_ELEMENT_BASE(c);
+for (i=0; i < len; ++i) {
+output[i] += CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+}
+}
+return TRUE;
+}
+static int codebook_decode_step(vorb *f, Codebook *c, float *output, int len, int step)
+{
+int i,z = codebook_decode_start(f,c);
+float last = CODEBOOK_ELEMENT_BASE(c);
+if (z < 0) return FALSE;
+if (len > c->dimensions) len = c->dimensions;
+#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+if (c->lookup_type == 1) {
+int div = 1;
+for (i=0; i < len; ++i) {
+int off = (z / div) % c->lookup_values;
+float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
+output[i*step] += val;
+if (c->sequence_p) last = val;
+div *= c->lookup_values;
+}
+return TRUE;
+}
+#endif
+z *= c->dimensions;
+for (i=0; i < len; ++i) {
+float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+output[i*step] += val;
+if (c->sequence_p) last = val;
+}
+return TRUE;
+}
+static int codebook_decode_deinterleave_repeat(vorb *f, Codebook *c, float **outputs, int ch, int *c_inter_p, int *p_inter_p, int len, int total_decode)
+{
+int c_inter = *c_inter_p;
+int p_inter = *p_inter_p;
+int i,z, effective = c->dimensions;
+// type 0 is only legal in a scalar context
+if (c->lookup_type == 0)   return error(f, VORBIS_invalid_stream);
+while (total_decode > 0) {
+float last = CODEBOOK_ELEMENT_BASE(c);
+DECODE_VQ(z,f,c);
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+assert(!c->sparse || z < c->sorted_entries);
+#endif
+if (z < 0) {
+if (!f->bytes_in_seg)
+if (f->last_seg) return FALSE;
+return error(f, VORBIS_invalid_stream);
+}
+// if this will take us off the end of the buffers, stop short!
+// we check by computing the length of the virtual interleaved
+// buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter),
+// and the length we'll be using (effective)
+if (c_inter + p_inter*ch + effective > len * ch) {
+effective = len*ch - (p_inter*ch - c_inter);
+}
+#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+if (c->lookup_type == 1) {
+int div = 1;
+for (i=0; i < effective; ++i) {
+int off = (z / div) % c->lookup_values;
+float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
+if (outputs[c_inter])
+outputs[c_inter][p_inter] += val;
+if (++c_inter == ch) { c_inter = 0; ++p_inter; }
+if (c->sequence_p) last = val;
+div *= c->lookup_values;
+}
+} else
+#endif
+{
+z *= c->dimensions;
+if (c->sequence_p) {
+for (i=0; i < effective; ++i) {
+float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+if (outputs[c_inter])
+outputs[c_inter][p_inter] += val;
+if (++c_inter == ch) { c_inter = 0; ++p_inter; }
+last = val;
+}
+} else {
+for (i=0; i < effective; ++i) {
+float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+if (outputs[c_inter])
+outputs[c_inter][p_inter] += val;
+if (++c_inter == ch) { c_inter = 0; ++p_inter; }
+}
+}
+}
+total_decode -= effective;
+}
+*c_inter_p = c_inter;
+*p_inter_p = p_inter;
+return TRUE;
+}
+static int predict_point(int x, int x0, int x1, int y0, int y1)
+{
+int dy = y1 - y0;
+int adx = x1 - x0;
+// @OPTIMIZE: force int division to round in the right direction... is this necessary on x86?
+int err = abs(dy) * (x - x0);
+int off = err / adx;
+return dy < 0 ? y0 - off : y0 + off;
+}
+// the following table is block-copied from the specification
+static float inverse_db_table[256] =
+{
+1.0649863e-07f, 1.1341951e-07f, 1.2079015e-07f, 1.2863978e-07f,
+1.3699951e-07f, 1.4590251e-07f, 1.5538408e-07f, 1.6548181e-07f,
+1.7623575e-07f, 1.8768855e-07f, 1.9988561e-07f, 2.1287530e-07f,
+2.2670913e-07f, 2.4144197e-07f, 2.5713223e-07f, 2.7384213e-07f,
+2.9163793e-07f, 3.1059021e-07f, 3.3077411e-07f, 3.5226968e-07f,
+3.7516214e-07f, 3.9954229e-07f, 4.2550680e-07f, 4.5315863e-07f,
+4.8260743e-07f, 5.1396998e-07f, 5.4737065e-07f, 5.8294187e-07f,
+6.2082472e-07f, 6.6116941e-07f, 7.0413592e-07f, 7.4989464e-07f,
+7.9862701e-07f, 8.5052630e-07f, 9.0579828e-07f, 9.6466216e-07f,
+1.0273513e-06f, 1.0941144e-06f, 1.1652161e-06f, 1.2409384e-06f,
+1.3215816e-06f, 1.4074654e-06f, 1.4989305e-06f, 1.5963394e-06f,
+1.7000785e-06f, 1.8105592e-06f, 1.9282195e-06f, 2.0535261e-06f,
+2.1869758e-06f, 2.3290978e-06f, 2.4804557e-06f, 2.6416497e-06f,
+2.8133190e-06f, 2.9961443e-06f, 3.1908506e-06f, 3.3982101e-06f,
+3.6190449e-06f, 3.8542308e-06f, 4.1047004e-06f, 4.3714470e-06f,
+4.6555282e-06f, 4.9580707e-06f, 5.2802740e-06f, 5.6234160e-06f,
+5.9888572e-06f, 6.3780469e-06f, 6.7925283e-06f, 7.2339451e-06f,
+7.7040476e-06f, 8.2047000e-06f, 8.7378876e-06f, 9.3057248e-06f,
+9.9104632e-06f, 1.0554501e-05f, 1.1240392e-05f, 1.1970856e-05f,
+1.2748789e-05f, 1.3577278e-05f, 1.4459606e-05f, 1.5399272e-05f,
+1.6400004e-05f, 1.7465768e-05f, 1.8600792e-05f, 1.9809576e-05f,
+2.1096914e-05f, 2.2467911e-05f, 2.3928002e-05f, 2.5482978e-05f,
+2.7139006e-05f, 2.8902651e-05f, 3.0780908e-05f, 3.2781225e-05f,
+3.4911534e-05f, 3.7180282e-05f, 3.9596466e-05f, 4.2169667e-05f,
+4.4910090e-05f, 4.7828601e-05f, 5.0936773e-05f, 5.4246931e-05f,
+5.7772202e-05f, 6.1526565e-05f, 6.5524908e-05f, 6.9783085e-05f,
+7.4317983e-05f, 7.9147585e-05f, 8.4291040e-05f, 8.9768747e-05f,
+9.5602426e-05f, 0.00010181521f, 0.00010843174f, 0.00011547824f,
+0.00012298267f, 0.00013097477f, 0.00013948625f, 0.00014855085f,
+0.00015820453f, 0.00016848555f, 0.00017943469f, 0.00019109536f,
+0.00020351382f, 0.00021673929f, 0.00023082423f, 0.00024582449f,
+0.00026179955f, 0.00027881276f, 0.00029693158f, 0.00031622787f,
+0.00033677814f, 0.00035866388f, 0.00038197188f, 0.00040679456f,
+0.00043323036f, 0.00046138411f, 0.00049136745f, 0.00052329927f,
+0.00055730621f, 0.00059352311f, 0.00063209358f, 0.00067317058f,
+0.00071691700f, 0.00076350630f, 0.00081312324f, 0.00086596457f,
+0.00092223983f, 0.00098217216f, 0.0010459992f,  0.0011139742f,
+0.0011863665f,  0.0012634633f,  0.0013455702f,  0.0014330129f,
+0.0015261382f,  0.0016253153f,  0.0017309374f,  0.0018434235f,
+0.0019632195f,  0.0020908006f,  0.0022266726f,  0.0023713743f,
+0.0025254795f,  0.0026895994f,  0.0028643847f,  0.0030505286f,
+0.0032487691f,  0.0034598925f,  0.0036847358f,  0.0039241906f,
+0.0041792066f,  0.0044507950f,  0.0047400328f,  0.0050480668f,
+0.0053761186f,  0.0057254891f,  0.0060975636f,  0.0064938176f,
+0.0069158225f,  0.0073652516f,  0.0078438871f,  0.0083536271f,
+0.0088964928f,  0.009474637f,   0.010090352f,   0.010746080f,
+0.011444421f,   0.012188144f,   0.012980198f,   0.013823725f,
+0.014722068f,   0.015678791f,   0.016697687f,   0.017782797f,
+0.018938423f,   0.020169149f,   0.021479854f,   0.022875735f,
+0.024362330f,   0.025945531f,   0.027631618f,   0.029427276f,
+0.031339626f,   0.033376252f,   0.035545228f,   0.037855157f,
+0.040315199f,   0.042935108f,   0.045725273f,   0.048696758f,
+0.051861348f,   0.055231591f,   0.058820850f,   0.062643361f,
+0.066714279f,   0.071049749f,   0.075666962f,   0.080584227f,
+0.085821044f,   0.091398179f,   0.097337747f,   0.10366330f,
+0.11039993f,    0.11757434f,    0.12521498f,    0.13335215f,
+0.14201813f,    0.15124727f,    0.16107617f,    0.17154380f,
+0.18269168f,    0.19456402f,    0.20720788f,    0.22067342f,
+0.23501402f,    0.25028656f,    0.26655159f,    0.28387361f,
+0.30232132f,    0.32196786f,    0.34289114f,    0.36517414f,
+0.38890521f,    0.41417847f,    0.44109412f,    0.46975890f,
+0.50028648f,    0.53279791f,    0.56742212f,    0.60429640f,
+0.64356699f,    0.68538959f,    0.72993007f,    0.77736504f,
+0.82788260f,    0.88168307f,    0.9389798f,     1.0f
+};
+// @OPTIMIZE: if you want to replace this bresenham line-drawing routine,
+// note that you must produce bit-identical output to decode correctly;
+// this specific sequence of operations is specified in the spec (it's
+// drawing integer-quantized frequency-space lines that the encoder
+// expects to be exactly the same)
+//     ... also, isn't the whole point of Bresenham's algorithm to NOT
+// have to divide in the setup? sigh.
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+#define LINE_OP(a,b)   a *= b
+#else
+#define LINE_OP(a,b)   a = b
+#endif
+#ifdef STB_VORBIS_DIVIDE_TABLE
+#define DIVTAB_NUMER   32
+#define DIVTAB_DENOM   64
+int8 integer_divide_table[DIVTAB_NUMER][DIVTAB_DENOM]; // 2KB
+#endif
+static __forceinline void draw_line(float *output, int x0, int y0, int x1, int y1, int n)
+{
+int dy = y1 - y0;
+int adx = x1 - x0;
+int ady = abs(dy);
+int base;
+int x=x0,y=y0;
+int err = 0;
+int sy;
+#ifdef STB_VORBIS_DIVIDE_TABLE
+if (adx < DIVTAB_DENOM && ady < DIVTAB_NUMER) {
+if (dy < 0) {
+base = -integer_divide_table[ady][adx];
+sy = base-1;
+} else {
+base =  integer_divide_table[ady][adx];
+sy = base+1;
+}
+} else {
+base = dy / adx;
+if (dy < 0)
+sy = base - 1;
+else
+sy = base+1;
+}
+#else
+base = dy / adx;
+if (dy < 0)
+sy = base - 1;
+else
+sy = base+1;
+#endif
+ady -= abs(base) * adx;
+if (x1 > n) x1 = n;
+if (x < x1) {
+LINE_OP(output[x], inverse_db_table[y&255]);
+for (++x; x < x1; ++x) {
+err += ady;
+if (err >= adx) {
+err -= adx;
+y += sy;
+} else
+y += base;
+LINE_OP(output[x], inverse_db_table[y&255]);
+}
+}
+}
+static int residue_decode(vorb *f, Codebook *book, float *target, int offset, int n, int rtype)
+{
+int k;
+if (rtype == 0) {
+int step = n / book->dimensions;
+for (k=0; k < step; ++k)
+if (!codebook_decode_step(f, book, target+offset+k, n-offset-k, step))
+return FALSE;
+} else {
+for (k=0; k < n; ) {
+if (!codebook_decode(f, book, target+offset, n-k))
+return FALSE;
+k += book->dimensions;
+offset += book->dimensions;
+}
+}
+return TRUE;
+}
+// n is 1/2 of the blocksize --
+// specification: "Correct per-vector decode length is [n]/2"
+static void decode_residue(vorb *f, float *residue_buffers[], int ch, int n, int rn, uint8 *do_not_decode)
+{
+int i,j,pass;
+Residue *r = f->residue_config + rn;
+int rtype = f->residue_types[rn];
+int c = r->classbook;
+int classwords = f->codebooks[c].dimensions;
+unsigned int actual_size = rtype == 2 ? n*2 : n;
+unsigned int limit_r_begin = (r->begin < actual_size ? r->begin : actual_size);
+unsigned int limit_r_end   = (r->end   < actual_size ? r->end   : actual_size);
+int n_read = limit_r_end - limit_r_begin;
+int part_read = n_read / r->part_size;
+int temp_alloc_point = temp_alloc_save(f);
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+uint8 ***part_classdata = (uint8 ***) temp_block_array(f,f->channels, part_read * sizeof(**part_classdata));
+#else
+int **classifications = (int **) temp_block_array(f,f->channels, part_read * sizeof(**classifications));
+#endif
+CHECK(f);
+for (i=0; i < ch; ++i)
+if (!do_not_decode[i])
+memset(residue_buffers[i], 0, sizeof(float) * n);
+if (rtype == 2 && ch != 1) {
+for (j=0; j < ch; ++j)
+if (!do_not_decode[j])
+break;
+if (j == ch)
+goto done;
+for (pass=0; pass < 8; ++pass) {
+int pcount = 0, class_set = 0;
+if (ch == 2) {
+while (pcount < part_read) {
+int z = r->begin + pcount*r->part_size;
+int c_inter = (z & 1), p_inter = z>>1;
+if (pass == 0) {
+Codebook *c = f->codebooks+r->classbook;
+int q;
+DECODE(q,f,c);
+if (q == EOP) goto done;
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+part_classdata[0][class_set] = r->classdata[q];
+#else
+for (i=classwords-1; i >= 0; --i) {
+classifications[0][i+pcount] = q % r->classifications;
+q /= r->classifications;
+}
+#endif
+}
+for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
+int z = r->begin + pcount*r->part_size;
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+int c = part_classdata[0][class_set][i];
+#else
+int c = classifications[0][pcount];
+#endif
+int b = r->residue_books[c][pass];
+if (b >= 0) {
+Codebook *book = f->codebooks + b;
+#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+goto done;
+#else
+// saves 1%
+if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+goto done;
+#endif
+} else {
+z += r->part_size;
+c_inter = z & 1;
+p_inter = z >> 1;
+}
+}
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+++class_set;
+#endif
+}
+} else if (ch > 2) {
+while (pcount < part_read) {
+int z = r->begin + pcount*r->part_size;
+int c_inter = z % ch, p_inter = z/ch;
+if (pass == 0) {
+Codebook *c = f->codebooks+r->classbook;
+int q;
+DECODE(q,f,c);
+if (q == EOP) goto done;
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+part_classdata[0][class_set] = r->classdata[q];
+#else
+for (i=classwords-1; i >= 0; --i) {
+classifications[0][i+pcount] = q % r->classifications;
+q /= r->classifications;
+}
+#endif
+}
+for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
+int z = r->begin + pcount*r->part_size;
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+int c = part_classdata[0][class_set][i];
+#else
+int c = classifications[0][pcount];
+#endif
+int b = r->residue_books[c][pass];
+if (b >= 0) {
+Codebook *book = f->codebooks + b;
+if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+goto done;
+} else {
+z += r->part_size;
+c_inter = z % ch;
+p_inter = z / ch;
+}
+}
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+++class_set;
+#endif
+}
+}
+}
+goto done;
+}
+CHECK(f);
+for (pass=0; pass < 8; ++pass) {
+int pcount = 0, class_set=0;
+while (pcount < part_read) {
+if (pass == 0) {
+for (j=0; j < ch; ++j) {
+if (!do_not_decode[j]) {
+Codebook *c = f->codebooks+r->classbook;
+int temp;
+DECODE(temp,f,c);
+if (temp == EOP) goto done;
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+part_classdata[j][class_set] = r->classdata[temp];
+#else
+for (i=classwords-1; i >= 0; --i) {
+classifications[j][i+pcount] = temp % r->classifications;
+temp /= r->classifications;
+}
+#endif
+}
+}
+}
+for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
+for (j=0; j < ch; ++j) {
+if (!do_not_decode[j]) {
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+int c = part_classdata[j][class_set][i];
+#else
+int c = classifications[j][pcount];
+#endif
+int b = r->residue_books[c][pass];
+if (b >= 0) {
+float *target = residue_buffers[j];
+int offset = r->begin + pcount * r->part_size;
+int n = r->part_size;
+Codebook *book = f->codebooks + b;
+if (!residue_decode(f, book, target, offset, n, rtype))
+goto done;
+}
+}
+}
+}
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+++class_set;
+#endif
+}
+}
+done:
+CHECK(f);
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+temp_free(f,part_classdata);
+#else
+temp_free(f,classifications);
+#endif
+temp_alloc_restore(f,temp_alloc_point);
+}
+#if 0
+// slow way for debugging
+void inverse_mdct_slow(float *buffer, int n)
+{
+int i,j;
+int n2 = n >> 1;
+float *x = (float *) malloc(sizeof(*x) * n2);
+memcpy(x, buffer, sizeof(*x) * n2);
+for (i=0; i < n; ++i) {
+float acc = 0;
+for (j=0; j < n2; ++j)
+// formula from paper:
+//acc += n/4.0f * x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1));
+// formula from wikipedia
+//acc += 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5));
+// these are equivalent, except the formula from the paper inverts the multiplier!
+// however, what actually works is NO MULTIPLIER!?!
+//acc += 64 * 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5));
+acc += x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1));
+buffer[i] = acc;
+}
+free(x);
+}
+#elif 0
+// same as above, but just barely able to run in real time on modern machines
+void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype)
+{
+float mcos[16384];
+int i,j;
+int n2 = n >> 1, nmask = (n << 2) -1;
+float *x = (float *) malloc(sizeof(*x) * n2);
+memcpy(x, buffer, sizeof(*x) * n2);
+for (i=0; i < 4*n; ++i)
+mcos[i] = (float) cos(M_PI / 2 * i / n);
+for (i=0; i < n; ++i) {
+float acc = 0;
+for (j=0; j < n2; ++j)
+acc += x[j] * mcos[(2 * i + 1 + n2)*(2*j+1) & nmask];
+buffer[i] = acc;
+}
+free(x);
+}
+#elif 0
+// transform to use a slow dct-iv; this is STILL basically trivial,
+// but only requires half as many ops
+void dct_iv_slow(float *buffer, int n)
+{
+float mcos[16384];
+float x[2048];
+int i,j;
+int n2 = n >> 1, nmask = (n << 3) - 1;
+memcpy(x, buffer, sizeof(*x) * n);
+for (i=0; i < 8*n; ++i)
+mcos[i] = (float) cos(M_PI / 4 * i / n);
+for (i=0; i < n; ++i) {
+float acc = 0;
+for (j=0; j < n; ++j)
+acc += x[j] * mcos[((2 * i + 1)*(2*j+1)) & nmask];
+buffer[i] = acc;
+}
+}
+void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype)
+{
+int i, n4 = n >> 2, n2 = n >> 1, n3_4 = n - n4;
+float temp[4096];
+memcpy(temp, buffer, n2 * sizeof(float));
+dct_iv_slow(temp, n2);  // returns -c'-d, a-b'
+for (i=0; i < n4  ; ++i) buffer[i] = temp[i+n4];            // a-b'
+for (   ; i < n3_4; ++i) buffer[i] = -temp[n3_4 - i - 1];   // b-a', c+d'
+for (   ; i < n   ; ++i) buffer[i] = -temp[i - n3_4];       // c'+d
+}
+#endif
+#ifndef LIBVORBIS_MDCT
+#define LIBVORBIS_MDCT 0
+#endif
+#if LIBVORBIS_MDCT
+// directly call the vorbis MDCT using an interface documented
+// by Jeff Roberts... useful for performance comparison
+typedef struct
+{
+int n;
+int log2n;
+float *trig;
+int   *bitrev;
+float scale;
+} mdct_lookup;
+extern void mdct_init(mdct_lookup *lookup, int n);
+extern void mdct_clear(mdct_lookup *l);
+extern void mdct_backward(mdct_lookup *init, float *in, float *out);
+mdct_lookup M1,M2;
+void inverse_mdct(float *buffer, int n, vorb *f, int blocktype)
+{
+mdct_lookup *M;
+if (M1.n == n) M = &M1;
+else if (M2.n == n) M = &M2;
+else if (M1.n == 0) { mdct_init(&M1, n); M = &M1; }
+else {
+if (M2.n) __asm int 3;
+mdct_init(&M2, n);
+M = &M2;
+}
+mdct_backward(M, buffer, buffer);
+}
+#endif
+// the following were split out into separate functions while optimizing;
+// they could be pushed back up but eh. __forceinline showed no change;
+// they're probably already being inlined.
+static void imdct_step3_iter0_loop(int n, float *e, int i_off, int k_off, float *A)
+{
+float *ee0 = e + i_off;
+float *ee2 = ee0 + k_off;
+int i;
+assert((n & 3) == 0);
+for (i=(n>>2); i > 0; --i) {
+float k00_20, k01_21;
+k00_20  = ee0[ 0] - ee2[ 0];
+k01_21  = ee0[-1] - ee2[-1];
+ee0[ 0] += ee2[ 0];//ee0[ 0] = ee0[ 0] + ee2[ 0];
+ee0[-1] += ee2[-1];//ee0[-1] = ee0[-1] + ee2[-1];
+ee2[ 0] = k00_20 * A[0] - k01_21 * A[1];
+ee2[-1] = k01_21 * A[0] + k00_20 * A[1];
+A += 8;
+k00_20  = ee0[-2] - ee2[-2];
+k01_21  = ee0[-3] - ee2[-3];
+ee0[-2] += ee2[-2];//ee0[-2] = ee0[-2] + ee2[-2];
+ee0[-3] += ee2[-3];//ee0[-3] = ee0[-3] + ee2[-3];
+ee2[-2] = k00_20 * A[0] - k01_21 * A[1];
+ee2[-3] = k01_21 * A[0] + k00_20 * A[1];
+A += 8;
+k00_20  = ee0[-4] - ee2[-4];
+k01_21  = ee0[-5] - ee2[-5];
+ee0[-4] += ee2[-4];//ee0[-4] = ee0[-4] + ee2[-4];
+ee0[-5] += ee2[-5];//ee0[-5] = ee0[-5] + ee2[-5];
+ee2[-4] = k00_20 * A[0] - k01_21 * A[1];
+ee2[-5] = k01_21 * A[0] + k00_20 * A[1];
+A += 8;
+k00_20  = ee0[-6] - ee2[-6];
+k01_21  = ee0[-7] - ee2[-7];
+ee0[-6] += ee2[-6];//ee0[-6] = ee0[-6] + ee2[-6];
+ee0[-7] += ee2[-7];//ee0[-7] = ee0[-7] + ee2[-7];
+ee2[-6] = k00_20 * A[0] - k01_21 * A[1];
+ee2[-7] = k01_21 * A[0] + k00_20 * A[1];
+A += 8;
+ee0 -= 8;
+ee2 -= 8;
+}
+}
+static void imdct_step3_inner_r_loop(int lim, float *e, int d0, int k_off, float *A, int k1)
+{
+int i;
+float k00_20, k01_21;
+float *e0 = e + d0;
+float *e2 = e0 + k_off;
+for (i=lim >> 2; i > 0; --i) {
+k00_20 = e0[-0] - e2[-0];
+k01_21 = e0[-1] - e2[-1];
+e0[-0] += e2[-0];//e0[-0] = e0[-0] + e2[-0];
+e0[-1] += e2[-1];//e0[-1] = e0[-1] + e2[-1];
+e2[-0] = (k00_20)*A[0] - (k01_21) * A[1];
+e2[-1] = (k01_21)*A[0] + (k00_20) * A[1];
+A += k1;
+k00_20 = e0[-2] - e2[-2];
+k01_21 = e0[-3] - e2[-3];
+e0[-2] += e2[-2];//e0[-2] = e0[-2] + e2[-2];
+e0[-3] += e2[-3];//e0[-3] = e0[-3] + e2[-3];
+e2[-2] = (k00_20)*A[0] - (k01_21) * A[1];
+e2[-3] = (k01_21)*A[0] + (k00_20) * A[1];
+A += k1;
+k00_20 = e0[-4] - e2[-4];
+k01_21 = e0[-5] - e2[-5];
+e0[-4] += e2[-4];//e0[-4] = e0[-4] + e2[-4];
+e0[-5] += e2[-5];//e0[-5] = e0[-5] + e2[-5];
+e2[-4] = (k00_20)*A[0] - (k01_21) * A[1];
+e2[-5] = (k01_21)*A[0] + (k00_20) * A[1];
+A += k1;
+k00_20 = e0[-6] - e2[-6];
+k01_21 = e0[-7] - e2[-7];
+e0[-6] += e2[-6];//e0[-6] = e0[-6] + e2[-6];
+e0[-7] += e2[-7];//e0[-7] = e0[-7] + e2[-7];
+e2[-6] = (k00_20)*A[0] - (k01_21) * A[1];
+e2[-7] = (k01_21)*A[0] + (k00_20) * A[1];
+e0 -= 8;
+e2 -= 8;
+A += k1;
+}
+}
+static void imdct_step3_inner_s_loop(int n, float *e, int i_off, int k_off, float *A, int a_off, int k0)
+{
+int i;
+float A0 = A[0];
+float A1 = A[0+1];
+float A2 = A[0+a_off];
+float A3 = A[0+a_off+1];
+float A4 = A[0+a_off*2+0];
+float A5 = A[0+a_off*2+1];
+float A6 = A[0+a_off*3+0];
+float A7 = A[0+a_off*3+1];
+float k00,k11;
+float *ee0 = e  +i_off;
+float *ee2 = ee0+k_off;
+for (i=n; i > 0; --i) {
+k00     = ee0[ 0] - ee2[ 0];
+k11     = ee0[-1] - ee2[-1];
+ee0[ 0] =  ee0[ 0] + ee2[ 0];
+ee0[-1] =  ee0[-1] + ee2[-1];
+ee2[ 0] = (k00) * A0 - (k11) * A1;
+ee2[-1] = (k11) * A0 + (k00) * A1;
+k00     = ee0[-2] - ee2[-2];
+k11     = ee0[-3] - ee2[-3];
+ee0[-2] =  ee0[-2] + ee2[-2];
+ee0[-3] =  ee0[-3] + ee2[-3];
+ee2[-2] = (k00) * A2 - (k11) * A3;
+ee2[-3] = (k11) * A2 + (k00) * A3;
+k00     = ee0[-4] - ee2[-4];
+k11     = ee0[-5] - ee2[-5];
+ee0[-4] =  ee0[-4] + ee2[-4];
+ee0[-5] =  ee0[-5] + ee2[-5];
+ee2[-4] = (k00) * A4 - (k11) * A5;
+ee2[-5] = (k11) * A4 + (k00) * A5;
+k00     = ee0[-6] - ee2[-6];
+k11     = ee0[-7] - ee2[-7];
+ee0[-6] =  ee0[-6] + ee2[-6];
+ee0[-7] =  ee0[-7] + ee2[-7];
+ee2[-6] = (k00) * A6 - (k11) * A7;
+ee2[-7] = (k11) * A6 + (k00) * A7;
+ee0 -= k0;
+ee2 -= k0;
+}
+}
+static __forceinline void iter_54(float *z)
+{
+float k00,k11,k22,k33;
+float y0,y1,y2,y3;
+k00  = z[ 0] - z[-4];
+y0   = z[ 0] + z[-4];
+y2   = z[-2] + z[-6];
+k22  = z[-2] - z[-6];
+z[-0] = y0 + y2;      // z0 + z4 + z2 + z6
+z[-2] = y0 - y2;      // z0 + z4 - z2 - z6
+// done with y0,y2
+k33  = z[-3] - z[-7];
+z[-4] = k00 + k33;    // z0 - z4 + z3 - z7
+z[-6] = k00 - k33;    // z0 - z4 - z3 + z7
+// done with k33
+k11  = z[-1] - z[-5];
+y1   = z[-1] + z[-5];
+y3   = z[-3] + z[-7];
+z[-1] = y1 + y3;      // z1 + z5 + z3 + z7
+z[-3] = y1 - y3;      // z1 + z5 - z3 - z7
+z[-5] = k11 - k22;    // z1 - z5 + z2 - z6
+z[-7] = k11 + k22;    // z1 - z5 - z2 + z6
+}
+static void imdct_step3_inner_s_loop_ld654(int n, float *e, int i_off, float *A, int base_n)
+{
+int a_off = base_n >> 3;
+float A2 = A[0+a_off];
+float *z = e + i_off;
+float *base = z - 16 * n;
+while (z > base) {
+float k00,k11;
+float l00,l11;
+k00    = z[-0] - z[ -8];
+k11    = z[-1] - z[ -9];
+l00    = z[-2] - z[-10];
+l11    = z[-3] - z[-11];
+z[ -0] = z[-0] + z[ -8];
+z[ -1] = z[-1] + z[ -9];
+z[ -2] = z[-2] + z[-10];
+z[ -3] = z[-3] + z[-11];
+z[ -8] = k00;
+z[ -9] = k11;
+z[-10] = (l00+l11) * A2;
+z[-11] = (l11-l00) * A2;
+k00    = z[ -4] - z[-12];
+k11    = z[ -5] - z[-13];
+l00    = z[ -6] - z[-14];
+l11    = z[ -7] - z[-15];
+z[ -4] = z[ -4] + z[-12];
+z[ -5] = z[ -5] + z[-13];
+z[ -6] = z[ -6] + z[-14];
+z[ -7] = z[ -7] + z[-15];
+z[-12] = k11;
+z[-13] = -k00;
+z[-14] = (l11-l00) * A2;
+z[-15] = (l00+l11) * -A2;
+iter_54(z);
+iter_54(z-8);
+z -= 16;
+}
+}
+static void inverse_mdct(float *buffer, int n, vorb *f, int blocktype)
+{
+int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l;
+int ld;
+// @OPTIMIZE: reduce register pressure by using fewer variables?
+int save_point = temp_alloc_save(f);
+float *buf2 = (float *) temp_alloc(f, n2 * sizeof(*buf2));
+float *u=NULL,*v=NULL;
+// twiddle factors
+float *A = f->A[blocktype];
+// IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio"
+// See notes about bugs in that paper in less-optimal implementation 'inverse_mdct_old' after this function.
+// kernel from paper
+// merged:
+//   copy and reflect spectral data
+//   step 0
+// note that it turns out that the items added together during
+// this step are, in fact, being added to themselves (as reflected
+// by step 0). inexplicable inefficiency! this became obvious
+// once I combined the passes.
+// so there's a missing 'times 2' here (for adding X to itself).
+// this propagates through linearly to the end, where the numbers
+// are 1/2 too small, and need to be compensated for.
+{
+float *d,*e, *AA, *e_stop;
+d = &buf2[n2-2];
+AA = A;
+e = &buffer[0];
+e_stop = &buffer[n2];
+while (e != e_stop) {
+d[1] = (e[0] * AA[0] - e[2]*AA[1]);
+d[0] = (e[0] * AA[1] + e[2]*AA[0]);
+d -= 2;
+AA += 2;
+e += 4;
+}
+e = &buffer[n2-3];
+while (d >= buf2) {
+d[1] = (-e[2] * AA[0] - -e[0]*AA[1]);
+d[0] = (-e[2] * AA[1] + -e[0]*AA[0]);
+d -= 2;
+AA += 2;
+e -= 4;
+}
+}
+// now we use symbolic names for these, so that we can
+// possibly swap their meaning as we change which operations
+// are in place
+u = buffer;
+v = buf2;
+// step 2    (paper output is w, now u)
+// this could be in place, but the data ends up in the wrong
+// place... _somebody_'s got to swap it, so this is nominated
+{
+float *AA = &A[n2-8];
+float *d0,*d1, *e0, *e1;
+e0 = &v[n4];
+e1 = &v[0];
+d0 = &u[n4];
+d1 = &u[0];
+while (AA >= A) {
+float v40_20, v41_21;
+v41_21 = e0[1] - e1[1];
+v40_20 = e0[0] - e1[0];
+d0[1]  = e0[1] + e1[1];
+d0[0]  = e0[0] + e1[0];
+d1[1]  = v41_21*AA[4] - v40_20*AA[5];
+d1[0]  = v40_20*AA[4] + v41_21*AA[5];
+v41_21 = e0[3] - e1[3];
+v40_20 = e0[2] - e1[2];
+d0[3]  = e0[3] + e1[3];
+d0[2]  = e0[2] + e1[2];
+d1[3]  = v41_21*AA[0] - v40_20*AA[1];
+d1[2]  = v40_20*AA[0] + v41_21*AA[1];
+AA -= 8;
+d0 += 4;
+d1 += 4;
+e0 += 4;
+e1 += 4;
+}
+}
+// step 3
+ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
+// optimized step 3:
+// the original step3 loop can be nested r inside s or s inside r;
+// it's written originally as s inside r, but this is dumb when r
+// iterates many times, and s few. So I have two copies of it and
+// switch between them halfway.
+// this is iteration 0 of step 3
+imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*0, -(n >> 3), A);
+imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*1, -(n >> 3), A);
+// this is iteration 1 of step 3
+imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*0, -(n >> 4), A, 16);
+imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*1, -(n >> 4), A, 16);
+imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*2, -(n >> 4), A, 16);
+imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*3, -(n >> 4), A, 16);
+l=2;
+for (; l < (ld-3)>>1; ++l) {
+int k0 = n >> (l+2), k0_2 = k0>>1;
+int lim = 1 << (l+1);
+int i;
+for (i=0; i < lim; ++i)
+imdct_step3_inner_r_loop(n >> (l+4), u, n2-1 - k0*i, -k0_2, A, 1 << (l+3));
+}
+for (; l < ld-6; ++l) {
+int k0 = n >> (l+2), k1 = 1 << (l+3), k0_2 = k0>>1;
+int rlim = n >> (l+6), r;
+int lim = 1 << (l+1);
+int i_off;
+float *A0 = A;
+i_off = n2-1;
+for (r=rlim; r > 0; --r) {
+imdct_step3_inner_s_loop(lim, u, i_off, -k0_2, A0, k1, k0);
+A0 += k1*4;
+i_off -= 8;
+}
+}
+// iterations with count:
+//   ld-6,-5,-4 all interleaved together
+//       the big win comes from getting rid of needless flops
+//         due to the constants on pass 5 & 4 being all 1 and 0;
+//       combining them to be simultaneous to improve cache made little difference
+imdct_step3_inner_s_loop_ld654(n >> 5, u, n2-1, A, n);
+// output is u
+// step 4, 5, and 6
+// cannot be in-place because of step 5
+{
+uint16 *bitrev = f->bit_reverse[blocktype];
+// weirdly, I'd have thought reading sequentially and writing
+// erratically would have been better than vice-versa, but in
+// fact that's not what my testing showed. (That is, with
+// j = bitreverse(i), do you read i and write j, or read j and write i.)
+float *d0 = &v[n4-4];
+float *d1 = &v[n2-4];
+while (d0 >= v) {
+int k4;
+k4 = bitrev[0];
+d1[3] = u[k4+0];
+d1[2] = u[k4+1];
+d0[3] = u[k4+2];
+d0[2] = u[k4+3];
+k4 = bitrev[1];
+d1[1] = u[k4+0];
+d1[0] = u[k4+1];
+d0[1] = u[k4+2];
+d0[0] = u[k4+3];
+d0 -= 4;
+d1 -= 4;
+bitrev += 2;
+}
+}
+// (paper output is u, now v)
+// data must be in buf2
+assert(v == buf2);
+// step 7   (paper output is v, now v)
+// this is now in place
+{
+float *C = f->C[blocktype];
+float *d, *e;
+d = v;
+e = v + n2 - 4;
+while (d < e) {
+float a02,a11,b0,b1,b2,b3;
+a02 = d[0] - e[2];
+a11 = d[1] + e[3];
+b0 = C[1]*a02 + C[0]*a11;
+b1 = C[1]*a11 - C[0]*a02;
+b2 = d[0] + e[ 2];
+b3 = d[1] - e[ 3];
+d[0] = b2 + b0;
+d[1] = b3 + b1;
+e[2] = b2 - b0;
+e[3] = b1 - b3;
+a02 = d[2] - e[0];
+a11 = d[3] + e[1];
+b0 = C[3]*a02 + C[2]*a11;
+b1 = C[3]*a11 - C[2]*a02;
+b2 = d[2] + e[ 0];
+b3 = d[3] - e[ 1];
+d[2] = b2 + b0;
+d[3] = b3 + b1;
+e[0] = b2 - b0;
+e[1] = b1 - b3;
+C += 4;
+d += 4;
+e -= 4;
+}
+}
+// data must be in buf2
+// step 8+decode   (paper output is X, now buffer)
+// this generates pairs of data a la 8 and pushes them directly through
+// the decode kernel (pushing rather than pulling) to avoid having
+// to make another pass later
+// this cannot POSSIBLY be in place, so we refer to the buffers directly
+{
+float *d0,*d1,*d2,*d3;
+float *B = f->B[blocktype] + n2 - 8;
+float *e = buf2 + n2 - 8;
+d0 = &buffer[0];
+d1 = &buffer[n2-4];
+d2 = &buffer[n2];
+d3 = &buffer[n-4];
+while (e >= v) {
+float p0,p1,p2,p3;
+p3 =  e[6]*B[7] - e[7]*B[6];
+p2 = -e[6]*B[6] - e[7]*B[7];
+d0[0] =   p3;
+d1[3] = - p3;
+d2[0] =   p2;
+d3[3] =   p2;
+p1 =  e[4]*B[5] - e[5]*B[4];
+p0 = -e[4]*B[4] - e[5]*B[5];
+d0[1] =   p1;
+d1[2] = - p1;
+d2[1] =   p0;
+d3[2] =   p0;
+p3 =  e[2]*B[3] - e[3]*B[2];
+p2 = -e[2]*B[2] - e[3]*B[3];
+d0[2] =   p3;
+d1[1] = - p3;
+d2[2] =   p2;
+d3[1] =   p2;
+p1 =  e[0]*B[1] - e[1]*B[0];
+p0 = -e[0]*B[0] - e[1]*B[1];
+d0[3] =   p1;
+d1[0] = - p1;
+d2[3] =   p0;
+d3[0] =   p0;
+B -= 8;
+e -= 8;
+d0 += 4;
+d2 += 4;
+d1 -= 4;
+d3 -= 4;
+}
+}
+temp_free(f,buf2);
+temp_alloc_restore(f,save_point);
+}
+#if 0
+// this is the original version of the above code, if you want to optimize it from scratch
+void inverse_mdct_naive(float *buffer, int n)
+{
+float s;
+float A[1 << 12], B[1 << 12], C[1 << 11];
+int i,k,k2,k4, n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l;
+int n3_4 = n - n4, ld;
+// how can they claim this only uses N words?!
+// oh, because they're only used sparsely, whoops
+float u[1 << 13], X[1 << 13], v[1 << 13], w[1 << 13];
+// set up twiddle factors
+for (k=k2=0; k < n4; ++k,k2+=2) {
+A[k2  ] = (float)  cos(4*k*M_PI/n);
+A[k2+1] = (float) -sin(4*k*M_PI/n);
+B[k2  ] = (float)  cos((k2+1)*M_PI/n/2);
+B[k2+1] = (float)  sin((k2+1)*M_PI/n/2);
+}
+for (k=k2=0; k < n8; ++k,k2+=2) {
+C[k2  ] = (float)  cos(2*(k2+1)*M_PI/n);
+C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n);
+}
+// IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio"
+// Note there are bugs in that pseudocode, presumably due to them attempting
+// to rename the arrays nicely rather than representing the way their actual
+// implementation bounces buffers back and forth. As a result, even in the
+// "some formulars corrected" version, a direct implementation fails. These
+// are noted below as "paper bug".
+// copy and reflect spectral data
+for (k=0; k < n2; ++k) u[k] = buffer[k];
+for (   ; k < n ; ++k) u[k] = -buffer[n - k - 1];
+// kernel from paper
+// step 1
+for (k=k2=k4=0; k < n4; k+=1, k2+=2, k4+=4) {
+v[n-k4-1] = (u[k4] - u[n-k4-1]) * A[k2]   - (u[k4+2] - u[n-k4-3])*A[k2+1];
+v[n-k4-3] = (u[k4] - u[n-k4-1]) * A[k2+1] + (u[k4+2] - u[n-k4-3])*A[k2];
+}
+// step 2
+for (k=k4=0; k < n8; k+=1, k4+=4) {
+w[n2+3+k4] = v[n2+3+k4] + v[k4+3];
+w[n2+1+k4] = v[n2+1+k4] + v[k4+1];
+w[k4+3]    = (v[n2+3+k4] - v[k4+3])*A[n2-4-k4] - (v[n2+1+k4]-v[k4+1])*A[n2-3-k4];
+w[k4+1]    = (v[n2+1+k4] - v[k4+1])*A[n2-4-k4] + (v[n2+3+k4]-v[k4+3])*A[n2-3-k4];
+}
+// step 3
+ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
+for (l=0; l < ld-3; ++l) {
+int k0 = n >> (l+2), k1 = 1 << (l+3);
+int rlim = n >> (l+4), r4, r;
+int s2lim = 1 << (l+2), s2;
+for (r=r4=0; r < rlim; r4+=4,++r) {
+for (s2=0; s2 < s2lim; s2+=2) {
+u[n-1-k0*s2-r4] = w[n-1-k0*s2-r4] + w[n-1-k0*(s2+1)-r4];
+u[n-3-k0*s2-r4] = w[n-3-k0*s2-r4] + w[n-3-k0*(s2+1)-r4];
+u[n-1-k0*(s2+1)-r4] = (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1]
+- (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1+1];
+u[n-3-k0*(s2+1)-r4] = (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1]
++ (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1+1];
+}
+}
+if (l+1 < ld-3) {
+// paper bug: ping-ponging of u&w here is omitted
+memcpy(w, u, sizeof(u));
+}
+}
+// step 4
+for (i=0; i < n8; ++i) {
+int j = bit_reverse(i) >> (32-ld+3);
+assert(j < n8);
+if (i == j) {
+// paper bug: original code probably swapped in place; if copying,
+//            need to directly copy in this case
+int i8 = i << 3;
+v[i8+1] = u[i8+1];
+v[i8+3] = u[i8+3];
+v[i8+5] = u[i8+5];
+v[i8+7] = u[i8+7];
+} else if (i < j) {
+int i8 = i << 3, j8 = j << 3;
+v[j8+1] = u[i8+1], v[i8+1] = u[j8 + 1];
+v[j8+3] = u[i8+3], v[i8+3] = u[j8 + 3];
+v[j8+5] = u[i8+5], v[i8+5] = u[j8 + 5];
+v[j8+7] = u[i8+7], v[i8+7] = u[j8 + 7];
+}
+}
+// step 5
+for (k=0; k < n2; ++k) {
+w[k] = v[k*2+1];
+}
+// step 6
+for (k=k2=k4=0; k < n8; ++k, k2 += 2, k4 += 4) {
+u[n-1-k2] = w[k4];
+u[n-2-k2] = w[k4+1];
+u[n3_4 - 1 - k2] = w[k4+2];
+u[n3_4 - 2 - k2] = w[k4+3];
+}
+// step 7
+for (k=k2=0; k < n8; ++k, k2 += 2) {
+v[n2 + k2 ] = ( u[n2 + k2] + u[n-2-k2] + C[k2+1]*(u[n2+k2]-u[n-2-k2]) + C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2;
+v[n-2 - k2] = ( u[n2 + k2] + u[n-2-k2] - C[k2+1]*(u[n2+k2]-u[n-2-k2]) - C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2;
+v[n2+1+ k2] = ( u[n2+1+k2] - u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2;
+v[n-1 - k2] = (-u[n2+1+k2] + u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2;
+}
+// step 8
+for (k=k2=0; k < n4; ++k,k2 += 2) {
+X[k]      = v[k2+n2]*B[k2  ] + v[k2+1+n2]*B[k2+1];
+X[n2-1-k] = v[k2+n2]*B[k2+1] - v[k2+1+n2]*B[k2  ];
+}
+// decode kernel to output
+// determined the following value experimentally
+// (by first figuring out what made inverse_mdct_slow work); then matching that here
+// (probably vorbis encoder premultiplies by n or n/2, to save it on the decoder?)
+s = 0.5; // theoretically would be n4
+// [[[ note! the s value of 0.5 is compensated for by the B[] in the current code,
+//     so it needs to use the "old" B values to behave correctly, or else
+//     set s to 1.0 ]]]
+for (i=0; i < n4  ; ++i) buffer[i] = s * X[i+n4];
+for (   ; i < n3_4; ++i) buffer[i] = -s * X[n3_4 - i - 1];
+for (   ; i < n   ; ++i) buffer[i] = -s * X[i - n3_4];
+}
+#endif
+static float *get_window(vorb *f, int len)
+{
+len <<= 1;
+if (len == f->blocksize_0) return f->window[0];
+if (len == f->blocksize_1) return f->window[1];
+return NULL;
+}
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+typedef int16 YTYPE;
+#else
+typedef int YTYPE;
+#endif
+static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *finalY, uint8 *step2_flag)
+{
+int n2 = n >> 1;
+int s = map->chan[i].mux, floor;
+floor = map->submap_floor[s];
+if (f->floor_types[floor] == 0) {
+return error(f, VORBIS_invalid_stream);
+} else {
+Floor1 *g = &f->floor_config[floor].floor1;
+int j,q;
+int lx = 0, ly = finalY[0] * g->floor1_multiplier;
+for (q=1; q < g->values; ++q) {
+j = g->sorted_order[q];
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+STBV_NOTUSED(step2_flag);
+if (finalY[j] >= 0)
+#else
+if (step2_flag[j])
+#endif
+{
+int hy = finalY[j] * g->floor1_multiplier;
+int hx = g->Xlist[j];
+if (lx != hx)
+draw_line(target, lx,ly, hx,hy, n2);
+CHECK(f);
+lx = hx, ly = hy;
+}
+}
+if (lx < n2) {
+// optimization of: draw_line(target, lx,ly, n,ly, n2);
+for (j=lx; j < n2; ++j)
+LINE_OP(target[j], inverse_db_table[ly]);
+CHECK(f);
+}
+}
+return TRUE;
+}
+// The meaning of "left" and "right"
+//
+// For a given frame:
+//     we compute samples from 0..n
+//     window_center is n/2
+//     we'll window and mix the samples from left_start to left_end with data from the previous frame
+//     all of the samples from left_end to right_start can be output without mixing; however,
+//        this interval is 0-length except when transitioning between short and long frames
+//     all of the samples from right_start to right_end need to be mixed with the next frame,
+//        which we don't have, so those get saved in a buffer
+//     frame N's right_end-right_start, the number of samples to mix with the next frame,
+//        has to be the same as frame N+1's left_end-left_start (which they are by
+//        construction)
+static int vorbis_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode)
+{
+Mode *m;
+int i, n, prev, next, window_center;
+f->channel_buffer_start = f->channel_buffer_end = 0;
+retry:
+if (f->eof) return FALSE;
+if (!maybe_start_packet(f))
+return FALSE;
+// check packet type
+if (get_bits(f,1) != 0) {
+if (IS_PUSH_MODE(f))
+return error(f,VORBIS_bad_packet_type);
+while (EOP != get8_packet(f));
+goto retry;
+}
+if (f->alloc.alloc_buffer)
+assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+i = get_bits(f, ilog(f->mode_count-1));
+if (i == EOP) return FALSE;
+if (i >= f->mode_count) return FALSE;
+*mode = i;
+m = f->mode_config + i;
+if (m->blockflag) {
+n = f->blocksize_1;
+prev = get_bits(f,1);
+next = get_bits(f,1);
+} else {
+prev = next = 0;
+n = f->blocksize_0;
+}
+// WINDOWING
+window_center = n >> 1;
+if (m->blockflag && !prev) {
+*p_left_start = (n - f->blocksize_0) >> 2;
+*p_left_end   = (n + f->blocksize_0) >> 2;
+} else {
+*p_left_start = 0;
+*p_left_end   = window_center;
+}
+if (m->blockflag && !next) {
+*p_right_start = (n*3 - f->blocksize_0) >> 2;
+*p_right_end   = (n*3 + f->blocksize_0) >> 2;
+} else {
+*p_right_start = window_center;
+*p_right_end   = n;
+}
+return TRUE;
+}
+static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start, int left_end, int right_start, int right_end, int *p_left)
+{
+Mapping *map;
+int i,j,k,n,n2;
+int zero_channel[256];
+int really_zero_channel[256];
+// WINDOWING
+STBV_NOTUSED(left_end);
+n = f->blocksize[m->blockflag];
+map = &f->mapping[m->mapping];
+// FLOORS
+n2 = n >> 1;
+CHECK(f);
+for (i=0; i < f->channels; ++i) {
+int s = map->chan[i].mux, floor;
+zero_channel[i] = FALSE;
+floor = map->submap_floor[s];
+if (f->floor_types[floor] == 0) {
+return error(f, VORBIS_invalid_stream);
+} else {
+Floor1 *g = &f->floor_config[floor].floor1;
+if (get_bits(f, 1)) {
+short *finalY;
+uint8 step2_flag[256];
+static int range_list[4] = { 256, 128, 86, 64 };
+int range = range_list[g->floor1_multiplier-1];
+int offset = 2;
+finalY = f->finalY[i];
+finalY[0] = get_bits(f, ilog(range)-1);
+finalY[1] = get_bits(f, ilog(range)-1);
+for (j=0; j < g->partitions; ++j) {
+int pclass = g->partition_class_list[j];
+int cdim = g->class_dimensions[pclass];
+int cbits = g->class_subclasses[pclass];
+int csub = (1 << cbits)-1;
+int cval = 0;
+if (cbits) {
+Codebook *c = f->codebooks + g->class_masterbooks[pclass];
+DECODE(cval,f,c);
+}
+for (k=0; k < cdim; ++k) {
+int book = g->subclass_books[pclass][cval & csub];
+cval = cval >> cbits;
+if (book >= 0) {
+int temp;
+Codebook *c = f->codebooks + book;
+DECODE(temp,f,c);
+finalY[offset++] = temp;
+} else
+finalY[offset++] = 0;
+}
+}
+if (f->valid_bits == INVALID_BITS) goto error; // behavior according to spec
+step2_flag[0] = step2_flag[1] = 1;
+for (j=2; j < g->values; ++j) {
+int low, high, pred, highroom, lowroom, room, val;
+low = g->neighbors[j][0];
+high = g->neighbors[j][1];
+//neighbors(g->Xlist, j, &low, &high);
+pred = predict_point(g->Xlist[j], g->Xlist[low], g->Xlist[high], finalY[low], finalY[high]);
+val = finalY[j];
+highroom = range - pred;
+lowroom = pred;
+if (highroom < lowroom)
+room = highroom * 2;
+else
+room = lowroom * 2;
+if (val) {
+step2_flag[low] = step2_flag[high] = 1;
+step2_flag[j] = 1;
+if (val >= room)
+if (highroom > lowroom)
+finalY[j] = val - lowroom + pred;
+else
+finalY[j] = pred - val + highroom - 1;
+else
+if (val & 1)
+finalY[j] = pred - ((val+1)>>1);
+else
+finalY[j] = pred + (val>>1);
+} else {
+step2_flag[j] = 0;
+finalY[j] = pred;
+}
+}
+#ifdef STB_VORBIS_NO_DEFER_FLOOR
+do_floor(f, map, i, n, f->floor_buffers[i], finalY, step2_flag);
+#else
+// defer final floor computation until _after_ residue
+for (j=0; j < g->values; ++j) {
+if (!step2_flag[j])
+finalY[j] = -1;
+}
+#endif
+} else {
+error:
+zero_channel[i] = TRUE;
+}
+// So we just defer everything else to later
+// at this point we've decoded the floor into buffer
+}
+}
+CHECK(f);
+// at this point we've decoded all floors
+if (f->alloc.alloc_buffer)
+assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+// re-enable coupled channels if necessary
+memcpy(really_zero_channel, zero_channel, sizeof(really_zero_channel[0]) * f->channels);
+for (i=0; i < map->coupling_steps; ++i)
+if (!zero_channel[map->chan[i].magnitude] || !zero_channel[map->chan[i].angle]) {
+zero_channel[map->chan[i].magnitude] = zero_channel[map->chan[i].angle] = FALSE;
+}
+CHECK(f);
+// RESIDUE DECODE
+for (i=0; i < map->submaps; ++i) {
+float *residue_buffers[STB_VORBIS_MAX_CHANNELS];
+int r;
+uint8 do_not_decode[256];
+int ch = 0;
+for (j=0; j < f->channels; ++j) {
+if (map->chan[j].mux == i) {
+if (zero_channel[j]) {
+do_not_decode[ch] = TRUE;
+residue_buffers[ch] = NULL;
+} else {
+do_not_decode[ch] = FALSE;
+residue_buffers[ch] = f->channel_buffers[j];
+}
+++ch;
+}
+}
+r = map->submap_residue[i];
+decode_residue(f, residue_buffers, ch, n2, r, do_not_decode);
+}
+if (f->alloc.alloc_buffer)
+assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+CHECK(f);
+// INVERSE COUPLING
+for (i = map->coupling_steps-1; i >= 0; --i) {
+int n2 = n >> 1;
+float *m = f->channel_buffers[map->chan[i].magnitude];
+float *a = f->channel_buffers[map->chan[i].angle    ];
+for (j=0; j < n2; ++j) {
+float a2,m2;
+if (m[j] > 0)
+if (a[j] > 0)
+m2 = m[j], a2 = m[j] - a[j];
+else
+a2 = m[j], m2 = m[j] + a[j];
+else
+if (a[j] > 0)
+m2 = m[j], a2 = m[j] + a[j];
+else
+a2 = m[j], m2 = m[j] - a[j];
+m[j] = m2;
+a[j] = a2;
+}
+}
+CHECK(f);
+// finish decoding the floors
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+for (i=0; i < f->channels; ++i) {
+if (really_zero_channel[i]) {
+memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2);
+} else {
+do_floor(f, map, i, n, f->channel_buffers[i], f->finalY[i], NULL);
+}
+}
+#else
+for (i=0; i < f->channels; ++i) {
+if (really_zero_channel[i]) {
+memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2);
+} else {
+for (j=0; j < n2; ++j)
+f->channel_buffers[i][j] *= f->floor_buffers[i][j];
+}
+}
+#endif
+// INVERSE MDCT
+CHECK(f);
+for (i=0; i < f->channels; ++i)
+inverse_mdct(f->channel_buffers[i], n, f, m->blockflag);
+CHECK(f);
+// this shouldn't be necessary, unless we exited on an error
+// and want to flush to get to the next packet
+flush_packet(f);
+if (f->first_decode) {
+// assume we start so first non-discarded sample is sample 0
+// this isn't to spec, but spec would require us to read ahead
+// and decode the size of all current frames--could be done,
+// but presumably it's not a commonly used feature
+f->current_loc = 0u - n2; // start of first frame is positioned for discard (NB this is an intentional unsigned overflow/wrap-around)
+// we might have to discard samples "from" the next frame too,
+// if we're lapping a large block then a small at the start?
+f->discard_samples_deferred = n - right_end;
+f->current_loc_valid = TRUE;
+f->first_decode = FALSE;
+} else if (f->discard_samples_deferred) {
+if (f->discard_samples_deferred >= right_start - left_start) {
+f->discard_samples_deferred -= (right_start - left_start);
+left_start = right_start;
+*p_left = left_start;
+} else {
+left_start += f->discard_samples_deferred;
+*p_left = left_start;
+f->discard_samples_deferred = 0;
+}
+} else if (f->previous_length == 0 && f->current_loc_valid) {
+// we're recovering from a seek... that means we're going to discard
+// the samples from this packet even though we know our position from
+// the last page header, so we need to update the position based on
+// the discarded samples here
+// but wait, the code below is going to add this in itself even
+// on a discard, so we don't need to do it here...
+}
+// check if we have ogg information about the sample # for this packet
+if (f->last_seg_which == f->end_seg_with_known_loc) {
+// if we have a valid current loc, and this is final:
+if (f->current_loc_valid && (f->page_flag & PAGEFLAG_last_page)) {
+uint32 current_end = f->known_loc_for_packet;
+// then let's infer the size of the (probably) short final frame
+if (current_end < f->current_loc + (right_end-left_start)) {
+if (current_end < f->current_loc) {
+// negative truncation, that's impossible!
+*len = 0;
+} else {
+*len = current_end - f->current_loc;
+}
+*len += left_start; // this doesn't seem right, but has no ill effect on my test files
+if (*len > right_end) *len = right_end; // this should never happen
+f->current_loc += *len;
+return TRUE;
+}
+}
+// otherwise, just set our sample loc
+// guess that the ogg granule pos refers to the _middle_ of the
+// last frame?
+// set f->current_loc to the position of left_start
+f->current_loc = f->known_loc_for_packet - (n2-left_start);
+f->current_loc_valid = TRUE;
+}
+if (f->current_loc_valid)
+f->current_loc += (right_start - left_start);
+if (f->alloc.alloc_buffer)
+assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+*len = right_end;  // ignore samples after the window goes to 0
+CHECK(f);
+return TRUE;
+}
+static int vorbis_decode_packet(vorb *f, int *len, int *p_left, int *p_right)
+{
+int mode, left_end, right_end;
+if (!vorbis_decode_initial(f, p_left, &left_end, p_right, &right_end, &mode)) return 0;
+return vorbis_decode_packet_rest(f, len, f->mode_config + mode, *p_left, left_end, *p_right, right_end, p_left);
+}
+static int vorbis_finish_frame(stb_vorbis *f, int len, int left, int right)
+{
+int prev,i,j;
+// we use right&left (the start of the right- and left-window sin()-regions)
+// to determine how much to return, rather than inferring from the rules
+// (same result, clearer code); 'left' indicates where our sin() window
+// starts, therefore where the previous window's right edge starts, and
+// therefore where to start mixing from the previous buffer. 'right'
+// indicates where our sin() ending-window starts, therefore that's where
+// we start saving, and where our returned-data ends.
+// mixin from previous window
+if (f->previous_length) {
+int i,j, n = f->previous_length;
+float *w = get_window(f, n);
+if (w == NULL) return 0;
+for (i=0; i < f->channels; ++i) {
+for (j=0; j < n; ++j)
+f->channel_buffers[i][left+j] =
+f->channel_buffers[i][left+j]*w[    j] +
+f->previous_window[i][     j]*w[n-1-j];
+}
+}
+prev = f->previous_length;
+// last half of this data becomes previous window
+f->previous_length = len - right;
+// @OPTIMIZE: could avoid this copy by double-buffering the
+// output (flipping previous_window with channel_buffers), but
+// then previous_window would have to be 2x as large, and
+// channel_buffers couldn't be temp mem (although they're NOT
+// currently temp mem, they could be (unless we want to level
+// performance by spreading out the computation))
+for (i=0; i < f->channels; ++i)
+for (j=0; right+j < len; ++j)
+f->previous_window[i][j] = f->channel_buffers[i][right+j];
+if (!prev)
+// there was no previous packet, so this data isn't valid...
+// this isn't entirely true, only the would-have-overlapped data
+// isn't valid, but this seems to be what the spec requires
+return 0;
+// truncate a short frame
+if (len < right) right = len;
+f->samples_output += right-left;
+return right - left;
+}
+static int vorbis_pump_first_frame(stb_vorbis *f)
+{
+int len, right, left, res;
+res = vorbis_decode_packet(f, &len, &left, &right);
+if (res)
+vorbis_finish_frame(f, len, left, right);
+return res;
+}
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+static int is_whole_packet_present(stb_vorbis *f)
+{
+// make sure that we have the packet available before continuing...
+// this requires a full ogg parse, but we know we can fetch from f->stream
+// instead of coding this out explicitly, we could save the current read state,
+// read the next packet with get8() until end-of-packet, check f->eof, then
+// reset the state? but that would be slower, esp. since we'd have over 256 bytes
+// of state to restore (primarily the page segment table)
+int s = f->next_seg, first = TRUE;
+uint8 *p = f->stream;
+if (s != -1) { // if we're not starting the packet with a 'continue on next page' flag
+for (; s < f->segment_count; ++s) {
+p += f->segments[s];
+if (f->segments[s] < 255)               // stop at first short segment
+break;
+}
+// either this continues, or it ends it...
+if (s == f->segment_count)
+s = -1; // set 'crosses page' flag
+if (p > f->stream_end)                     return error(f, VORBIS_need_more_data);
+first = FALSE;
+}
+for (; s == -1;) {
+uint8 *q;
+int n;
+// check that we have the page header ready
+if (p + 26 >= f->stream_end)               return error(f, VORBIS_need_more_data);
+// validate the page
+if (memcmp(p, ogg_page_header, 4))         return error(f, VORBIS_invalid_stream);
+if (p[4] != 0)                             return error(f, VORBIS_invalid_stream);
+if (first) { // the first segment must NOT have 'continued_packet', later ones MUST
+if (f->previous_length)
+if ((p[5] & PAGEFLAG_continued_packet))  return error(f, VORBIS_invalid_stream);
+// if no previous length, we're resynching, so we can come in on a continued-packet,
+// which we'll just drop
+} else {
+if (!(p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream);
+}
+n = p[26]; // segment counts
+q = p+27;  // q points to segment table
+p = q + n; // advance past header
+// make sure we've read the segment table
+if (p > f->stream_end)                     return error(f, VORBIS_need_more_data);
+for (s=0; s < n; ++s) {
+p += q[s];
+if (q[s] < 255)
+break;
+}
+if (s == n)
+s = -1; // set 'crosses page' flag
+if (p > f->stream_end)                     return error(f, VORBIS_need_more_data);
+first = FALSE;
+}
+return TRUE;
+}
+#endif // !STB_VORBIS_NO_PUSHDATA_API
+static int start_decoder(vorb *f)
+{
+uint8 header[6], x,y;
+int len,i,j,k, max_submaps = 0;
+int longest_floorlist=0;
+// first page, first packet
+f->first_decode = TRUE;
+if (!start_page(f))                              return FALSE;
+// validate page flag
+if (!(f->page_flag & PAGEFLAG_first_page))       return error(f, VORBIS_invalid_first_page);
+if (f->page_flag & PAGEFLAG_last_page)           return error(f, VORBIS_invalid_first_page);
+if (f->page_flag & PAGEFLAG_continued_packet)    return error(f, VORBIS_invalid_first_page);
+// check for expected packet length
+if (f->segment_count != 1)                       return error(f, VORBIS_invalid_first_page);
+if (f->segments[0] != 30) {
+// check for the Ogg skeleton fishead identifying header to refine our error
+if (f->segments[0] == 64 &&
+getn(f, header, 6) &&
+header[0] == 'f' &&
+header[1] == 'i' &&
+header[2] == 's' &&
+header[3] == 'h' &&
+header[4] == 'e' &&
+header[5] == 'a' &&
+get8(f)   == 'd' &&
+get8(f)   == '\0')                        return error(f, VORBIS_ogg_skeleton_not_supported);
+else
+return error(f, VORBIS_invalid_first_page);
+}
+// read packet
+// check packet header
+if (get8(f) != VORBIS_packet_id)                 return error(f, VORBIS_invalid_first_page);
+if (!getn(f, header, 6))                         return error(f, VORBIS_unexpected_eof);
+if (!vorbis_validate(header))                    return error(f, VORBIS_invalid_first_page);
+// vorbis_version
+if (get32(f) != 0)                               return error(f, VORBIS_invalid_first_page);
+f->channels = get8(f); if (!f->channels)         return error(f, VORBIS_invalid_first_page);
+if (f->channels > STB_VORBIS_MAX_CHANNELS)       return error(f, VORBIS_too_many_channels);
+f->sample_rate = get32(f); if (!f->sample_rate)  return error(f, VORBIS_invalid_first_page);
+get32(f); // bitrate_maximum
+get32(f); // bitrate_nominal
+get32(f); // bitrate_minimum
+x = get8(f);
+{
+int log0,log1;
+log0 = x & 15;
+log1 = x >> 4;
+f->blocksize_0 = 1 << log0;
+f->blocksize_1 = 1 << log1;
+if (log0 < 6 || log0 > 13)                       return error(f, VORBIS_invalid_setup);
+if (log1 < 6 || log1 > 13)                       return error(f, VORBIS_invalid_setup);
+if (log0 > log1)                                 return error(f, VORBIS_invalid_setup);
+}
+// framing_flag
+x = get8(f);
+if (!(x & 1))                                    return error(f, VORBIS_invalid_first_page);
+// second packet!
+if (!start_page(f))                              return FALSE;
+if (!start_packet(f))                            return FALSE;
+if (!next_segment(f))                            return FALSE;
+if (get8_packet(f) != VORBIS_packet_comment)            return error(f, VORBIS_invalid_setup);
+for (i=0; i < 6; ++i) header[i] = get8_packet(f);
+if (!vorbis_validate(header))                    return error(f, VORBIS_invalid_setup);
+//file vendor
+len = get32_packet(f);
+f->vendor = (char*)setup_malloc(f, sizeof(char) * (len+1));
+if (f->vendor == NULL)                           return error(f, VORBIS_outofmem);
+for(i=0; i < len; ++i) {
+f->vendor[i] = get8_packet(f);
+}
+f->vendor[len] = (char)'\0';
+//user comments
+f->comment_list_length = get32_packet(f);
+f->comment_list = NULL;
+if (f->comment_list_length > 0)
+{
+f->comment_list = (char**) setup_malloc(f, sizeof(char*) * (f->comment_list_length));
+if (f->comment_list == NULL)                  return error(f, VORBIS_outofmem);
+}
+for(i=0; i < f->comment_list_length; ++i) {
+len = get32_packet(f);
+f->comment_list[i] = (char*)setup_malloc(f, sizeof(char) * (len+1));
+if (f->comment_list[i] == NULL)               return error(f, VORBIS_outofmem);
+for(j=0; j < len; ++j) {
+f->comment_list[i][j] = get8_packet(f);
+}
+f->comment_list[i][len] = (char)'\0';
+}
+// framing_flag
+x = get8_packet(f);
+if (!(x & 1))                                    return error(f, VORBIS_invalid_setup);
+skip(f, f->bytes_in_seg);
+f->bytes_in_seg = 0;
+do {
+len = next_segment(f);
+skip(f, len);
+f->bytes_in_seg = 0;
+} while (len);
+// third packet!
+if (!start_packet(f))                            return FALSE;
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+if (IS_PUSH_MODE(f)) {
+if (!is_whole_packet_present(f)) {
+// convert error in ogg header to write type
+if (f->error == VORBIS_invalid_stream)
+f->error = VORBIS_invalid_setup;
+return FALSE;
+}
+}
+#endif
+crc32_init(); // always init it, to avoid multithread race conditions
+if (get8_packet(f) != VORBIS_packet_setup)       return error(f, VORBIS_invalid_setup);
+for (i=0; i < 6; ++i) header[i] = get8_packet(f);
+if (!vorbis_validate(header))                    return error(f, VORBIS_invalid_setup);
+// codebooks
+f->codebook_count = get_bits(f,8) + 1;
+f->codebooks = (Codebook *) setup_malloc(f, sizeof(*f->codebooks) * f->codebook_count);
+if (f->codebooks == NULL)                        return error(f, VORBIS_outofmem);
+memset(f->codebooks, 0, sizeof(*f->codebooks) * f->codebook_count);
+for (i=0; i < f->codebook_count; ++i) {
+uint32 *values;
+int ordered, sorted_count;
+int total=0;
+uint8 *lengths;
+Codebook *c = f->codebooks+i;
+CHECK(f);
+x = get_bits(f, 8); if (x != 0x42)            return error(f, VORBIS_invalid_setup);
+x = get_bits(f, 8); if (x != 0x43)            return error(f, VORBIS_invalid_setup);
+x = get_bits(f, 8); if (x != 0x56)            return error(f, VORBIS_invalid_setup);
+x = get_bits(f, 8);
+c->dimensions = (get_bits(f, 8)<<8) + x;
+x = get_bits(f, 8);
+y = get_bits(f, 8);
+c->entries = (get_bits(f, 8)<<16) + (y<<8) + x;
+ordered = get_bits(f,1);
+c->sparse = ordered ? 0 : get_bits(f,1);
+if (c->dimensions == 0 && c->entries != 0)    return error(f, VORBIS_invalid_setup);
+if (c->sparse)
+lengths = (uint8 *) setup_temp_malloc(f, c->entries);
+else
+lengths = c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries);
+if (!lengths) return error(f, VORBIS_outofmem);
+if (ordered) {
+int current_entry = 0;
+int current_length = get_bits(f,5) + 1;
+while (current_entry < c->entries) {
+int limit = c->entries - current_entry;
+int n = get_bits(f, ilog(limit));
+if (current_length >= 32) return error(f, VORBIS_invalid_setup);
+if (current_entry + n > (int) c->entries) { return error(f, VORBIS_invalid_setup); }
+memset(lengths + current_entry, current_length, n);
+current_entry += n;
+++current_length;
+}
+} else {
+for (j=0; j < c->entries; ++j) {
+int present = c->sparse ? get_bits(f,1) : 1;
+if (present) {
+lengths[j] = get_bits(f, 5) + 1;
+++total;
+if (lengths[j] == 32)
+return error(f, VORBIS_invalid_setup);
+} else {
+lengths[j] = NO_CODE;
+}
+}
+}
+if (c->sparse && total >= c->entries >> 2) {
+// convert sparse items to non-sparse!
+if (c->entries > (int) f->setup_temp_memory_required)
+f->setup_temp_memory_required = c->entries;
+c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries);
+if (c->codeword_lengths == NULL) return error(f, VORBIS_outofmem);
+memcpy(c->codeword_lengths, lengths, c->entries);
+setup_temp_free(f, lengths, c->entries); // note this is only safe if there have been no intervening temp mallocs!
+lengths = c->codeword_lengths;
+c->sparse = 0;
+}
+// compute the size of the sorted tables
+if (c->sparse) {
+sorted_count = total;
+} else {
+sorted_count = 0;
+#ifndef STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+for (j=0; j < c->entries; ++j)
+if (lengths[j] > STB_VORBIS_FAST_HUFFMAN_LENGTH && lengths[j] != NO_CODE)
+++sorted_count;
+#endif
+}
+c->sorted_entries = sorted_count;
+values = NULL;
+CHECK(f);
+if (!c->sparse) {
+c->codewords = (uint32 *) setup_malloc(f, sizeof(c->codewords[0]) * c->entries);
+if (!c->codewords)                  return error(f, VORBIS_outofmem);
+} else {
+unsigned int size;
+if (c->sorted_entries) {
+c->codeword_lengths = (uint8 *) setup_malloc(f, c->sorted_entries);
+if (!c->codeword_lengths)           return error(f, VORBIS_outofmem);
+c->codewords = (uint32 *) setup_temp_malloc(f, sizeof(*c->codewords) * c->sorted_entries);
+if (!c->codewords)                  return error(f, VORBIS_outofmem);
+values = (uint32 *) setup_temp_malloc(f, sizeof(*values) * c->sorted_entries);
+if (!values)                        return error(f, VORBIS_outofmem);
+}
+size = c->entries + (sizeof(*c->codewords) + sizeof(*values)) * c->sorted_entries;
+if (size > f->setup_temp_memory_required)
+f->setup_temp_memory_required = size;
+}
+if (!compute_codewords(c, lengths, c->entries, values)) {
+if (c->sparse) setup_temp_free(f, values, 0);
+return error(f, VORBIS_invalid_setup);
+}
+if (c->sorted_entries) {
+// allocate an extra slot for sentinels
+c->sorted_codewords = (uint32 *) setup_malloc(f, sizeof(*c->sorted_codewords) * (c->sorted_entries+1));
+if (c->sorted_codewords == NULL) return error(f, VORBIS_outofmem);
+// allocate an extra slot at the front so that c->sorted_values[-1] is defined
+// so that we can catch that case without an extra if
+c->sorted_values    = ( int   *) setup_malloc(f, sizeof(*c->sorted_values   ) * (c->sorted_entries+1));
+if (c->sorted_values == NULL) return error(f, VORBIS_outofmem);
+++c->sorted_values;
+c->sorted_values[-1] = -1;
+compute_sorted_huffman(c, lengths, values);
+}
+if (c->sparse) {
+setup_temp_free(f, values, sizeof(*values)*c->sorted_entries);
+setup_temp_free(f, c->codewords, sizeof(*c->codewords)*c->sorted_entries);
+setup_temp_free(f, lengths, c->entries);
+c->codewords = NULL;
+}
+compute_accelerated_huffman(c);
+CHECK(f);
+c->lookup_type = get_bits(f, 4);
+if (c->lookup_type > 2) return error(f, VORBIS_invalid_setup);
+if (c->lookup_type > 0) {
+uint16 *mults;
+c->minimum_value = float32_unpack(get_bits(f, 32));
+c->delta_value = float32_unpack(get_bits(f, 32));
+c->value_bits = get_bits(f, 4)+1;
+c->sequence_p = get_bits(f,1);
+if (c->lookup_type == 1) {
+int values = lookup1_values(c->entries, c->dimensions);
+if (values < 0) return error(f, VORBIS_invalid_setup);
+c->lookup_values = (uint32) values;
+} else {
+c->lookup_values = c->entries * c->dimensions;
+}
+if (c->lookup_values == 0) return error(f, VORBIS_invalid_setup);
+mults = (uint16 *) setup_temp_malloc(f, sizeof(mults[0]) * c->lookup_values);
+if (mults == NULL) return error(f, VORBIS_outofmem);
+for (j=0; j < (int) c->lookup_values; ++j) {
+int q = get_bits(f, c->value_bits);
+if (q == EOP) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_invalid_setup); }
+mults[j] = q;
+}
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+if (c->lookup_type == 1) {
+int len, sparse = c->sparse;
+float last=0;
+// pre-expand the lookup1-style multiplicands, to avoid a divide in the inner loop
+if (sparse) {
+if (c->sorted_entries == 0) goto skip;
+c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->sorted_entries * c->dimensions);
+} else
+c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->entries        * c->dimensions);
+if (c->multiplicands == NULL) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); }
+len = sparse ? c->sorted_entries : c->entries;
+for (j=0; j < len; ++j) {
+unsigned int z = sparse ? c->sorted_values[j] : j;
+unsigned int div=1;
+for (k=0; k < c->dimensions; ++k) {
+int off = (z / div) % c->lookup_values;
+float val = mults[off]*c->delta_value + c->minimum_value + last;
+c->multiplicands[j*c->dimensions + k] = val;
+if (c->sequence_p)
+last = val;
+if (k+1 < c->dimensions) {
+if (div > UINT_MAX / (unsigned int) c->lookup_values) {
+setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values);
+return error(f, VORBIS_invalid_setup);
+}
+div *= c->lookup_values;
+}
+}
+}
+c->lookup_type = 2;
+}
+else
+#endif
+{
+float last=0;
+CHECK(f);
+c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values);
+if (c->multiplicands == NULL) { setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); }
+for (j=0; j < (int) c->lookup_values; ++j) {
+float val = mults[j] * c->delta_value + c->minimum_value + last;
+c->multiplicands[j] = val;
+if (c->sequence_p)
+last = val;
+}
+}
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+skip:;
+#endif
+setup_temp_free(f, mults, sizeof(mults[0])*c->lookup_values);
+CHECK(f);
+}
+CHECK(f);
+}
+// time domain transfers (notused)
+x = get_bits(f, 6) + 1;
+for (i=0; i < x; ++i) {
+uint32 z = get_bits(f, 16);
+if (z != 0) return error(f, VORBIS_invalid_setup);
+}
+// Floors
+f->floor_count = get_bits(f, 6)+1;
+f->floor_config = (Floor *)  setup_malloc(f, f->floor_count * sizeof(*f->floor_config));
+if (f->floor_config == NULL) return error(f, VORBIS_outofmem);
+for (i=0; i < f->floor_count; ++i) {
+f->floor_types[i] = get_bits(f, 16);
+if (f->floor_types[i] > 1) return error(f, VORBIS_invalid_setup);
+if (f->floor_types[i] == 0) {
+Floor0 *g = &f->floor_config[i].floor0;
+g->order = get_bits(f,8);
+g->rate = get_bits(f,16);
+g->bark_map_size = get_bits(f,16);
+g->amplitude_bits = get_bits(f,6);
+g->amplitude_offset = get_bits(f,8);
+g->number_of_books = get_bits(f,4) + 1;
+for (j=0; j < g->number_of_books; ++j)
+g->book_list[j] = get_bits(f,8);
+return error(f, VORBIS_feature_not_supported);
+} else {
+stbv__floor_ordering p[31*8+2];
+Floor1 *g = &f->floor_config[i].floor1;
+int max_class = -1;
+g->partitions = get_bits(f, 5);
+for (j=0; j < g->partitions; ++j) {
+g->partition_class_list[j] = get_bits(f, 4);
+if (g->partition_class_list[j] > max_class)
+max_class = g->partition_class_list[j];
+}
+for (j=0; j <= max_class; ++j) {
+g->class_dimensions[j] = get_bits(f, 3)+1;
+g->class_subclasses[j] = get_bits(f, 2);
+if (g->class_subclasses[j]) {
+g->class_masterbooks[j] = get_bits(f, 8);
+if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+}
+for (k=0; k < 1 << g->class_subclasses[j]; ++k) {
+g->subclass_books[j][k] = (int16)get_bits(f,8)-1;
+if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+}
+}
+g->floor1_multiplier = get_bits(f,2)+1;
+g->rangebits = get_bits(f,4);
+g->Xlist[0] = 0;
+g->Xlist[1] = 1 << g->rangebits;
+g->values = 2;
+for (j=0; j < g->partitions; ++j) {
+int c = g->partition_class_list[j];
+for (k=0; k < g->class_dimensions[c]; ++k) {
+g->Xlist[g->values] = get_bits(f, g->rangebits);
+++g->values;
+}
+}
+// precompute the sorting
+for (j=0; j < g->values; ++j) {
+p[j].x = g->Xlist[j];
+p[j].id = j;
+}
+qsort(p, g->values, sizeof(p[0]), point_compare);
+for (j=0; j < g->values-1; ++j)
+if (p[j].x == p[j+1].x)
+return error(f, VORBIS_invalid_setup);
+for (j=0; j < g->values; ++j)
+g->sorted_order[j] = (uint8) p[j].id;
+// precompute the neighbors
+for (j=2; j < g->values; ++j) {
+int low = 0,hi = 0;
+neighbors(g->Xlist, j, &low,&hi);
+g->neighbors[j][0] = low;
+g->neighbors[j][1] = hi;
+}
+if (g->values > longest_floorlist)
+longest_floorlist = g->values;
+}
+}
+// Residue
+f->residue_count = get_bits(f, 6)+1;
+f->residue_config = (Residue *) setup_malloc(f, f->residue_count * sizeof(f->residue_config[0]));
+if (f->residue_config == NULL) return error(f, VORBIS_outofmem);
+memset(f->residue_config, 0, f->residue_count * sizeof(f->residue_config[0]));
+for (i=0; i < f->residue_count; ++i) {
+uint8 residue_cascade[64];
+Residue *r = f->residue_config+i;
+f->residue_types[i] = get_bits(f, 16);
+if (f->residue_types[i] > 2) return error(f, VORBIS_invalid_setup);
+r->begin = get_bits(f, 24);
+r->end = get_bits(f, 24);
+if (r->end < r->begin) return error(f, VORBIS_invalid_setup);
+r->part_size = get_bits(f,24)+1;
+r->classifications = get_bits(f,6)+1;
+r->classbook = get_bits(f,8);
+if (r->classbook >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+for (j=0; j < r->classifications; ++j) {
+uint8 high_bits=0;
+uint8 low_bits=get_bits(f,3);
+if (get_bits(f,1))
+high_bits = get_bits(f,5);
+residue_cascade[j] = high_bits*8 + low_bits;
+}
+r->residue_books = (short (*)[8]) setup_malloc(f, sizeof(r->residue_books[0]) * r->classifications);
+if (r->residue_books == NULL) return error(f, VORBIS_outofmem);
+for (j=0; j < r->classifications; ++j) {
+for (k=0; k < 8; ++k) {
+if (residue_cascade[j] & (1 << k)) {
+r->residue_books[j][k] = get_bits(f, 8);
+if (r->residue_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+} else {
+r->residue_books[j][k] = -1;
+}
+}
+}
+// precompute the classifications[] array to avoid inner-loop mod/divide
+// call it 'classdata' since we already have r->classifications
+r->classdata = (uint8 **) setup_malloc(f, sizeof(*r->classdata) * f->codebooks[r->classbook].entries);
+if (!r->classdata) return error(f, VORBIS_outofmem);
+memset(r->classdata, 0, sizeof(*r->classdata) * f->codebooks[r->classbook].entries);
+for (j=0; j < f->codebooks[r->classbook].entries; ++j) {
+int classwords = f->codebooks[r->classbook].dimensions;
+int temp = j;
+r->classdata[j] = (uint8 *) setup_malloc(f, sizeof(r->classdata[j][0]) * classwords);
+if (r->classdata[j] == NULL) return error(f, VORBIS_outofmem);
+for (k=classwords-1; k >= 0; --k) {
+r->classdata[j][k] = temp % r->classifications;
+temp /= r->classifications;
+}
+}
+}
+f->mapping_count = get_bits(f,6)+1;
+f->mapping = (Mapping *) setup_malloc(f, f->mapping_count * sizeof(*f->mapping));
+if (f->mapping == NULL) return error(f, VORBIS_outofmem);
+memset(f->mapping, 0, f->mapping_count * sizeof(*f->mapping));
+for (i=0; i < f->mapping_count; ++i) {
+Mapping *m = f->mapping + i;
+int mapping_type = get_bits(f,16);
+if (mapping_type != 0) return error(f, VORBIS_invalid_setup);
+m->chan = (MappingChannel *) setup_malloc(f, f->channels * sizeof(*m->chan));
+if (m->chan == NULL) return error(f, VORBIS_outofmem);
+if (get_bits(f,1))
+m->submaps = get_bits(f,4)+1;
+else
+m->submaps = 1;
+if (m->submaps > max_submaps)
+max_submaps = m->submaps;
+if (get_bits(f,1)) {
+m->coupling_steps = get_bits(f,8)+1;
+if (m->coupling_steps > f->channels) return error(f, VORBIS_invalid_setup);
+for (k=0; k < m->coupling_steps; ++k) {
+m->chan[k].magnitude = get_bits(f, ilog(f->channels-1));
+m->chan[k].angle = get_bits(f, ilog(f->channels-1));
+if (m->chan[k].magnitude >= f->channels)        return error(f, VORBIS_invalid_setup);
+if (m->chan[k].angle     >= f->channels)        return error(f, VORBIS_invalid_setup);
+if (m->chan[k].magnitude == m->chan[k].angle)   return error(f, VORBIS_invalid_setup);
+}
+} else
+m->coupling_steps = 0;
+// reserved field
+if (get_bits(f,2)) return error(f, VORBIS_invalid_setup);
+if (m->submaps > 1) {
+for (j=0; j < f->channels; ++j) {
+m->chan[j].mux = get_bits(f, 4);
+if (m->chan[j].mux >= m->submaps)                return error(f, VORBIS_invalid_setup);
+}
+} else
+// @SPECIFICATION: this case is missing from the spec
+for (j=0; j < f->channels; ++j)
+m->chan[j].mux = 0;
+for (j=0; j < m->submaps; ++j) {
+get_bits(f,8); // discard
+m->submap_floor[j] = get_bits(f,8);
+m->submap_residue[j] = get_bits(f,8);
+if (m->submap_floor[j] >= f->floor_count)      return error(f, VORBIS_invalid_setup);
+if (m->submap_residue[j] >= f->residue_count)  return error(f, VORBIS_invalid_setup);
+}
+}
+// Modes
+f->mode_count = get_bits(f, 6)+1;
+for (i=0; i < f->mode_count; ++i) {
+Mode *m = f->mode_config+i;
+m->blockflag = get_bits(f,1);
+m->windowtype = get_bits(f,16);
+m->transformtype = get_bits(f,16);
+m->mapping = get_bits(f,8);
+if (m->windowtype != 0)                 return error(f, VORBIS_invalid_setup);
+if (m->transformtype != 0)              return error(f, VORBIS_invalid_setup);
+if (m->mapping >= f->mapping_count)     return error(f, VORBIS_invalid_setup);
+}
+flush_packet(f);
+f->previous_length = 0;
+for (i=0; i < f->channels; ++i) {
+f->channel_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1);
+f->previous_window[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2);
+f->finalY[i]          = (int16 *) setup_malloc(f, sizeof(int16) * longest_floorlist);
+if (f->channel_buffers[i] == NULL || f->previous_window[i] == NULL || f->finalY[i] == NULL) return error(f, VORBIS_outofmem);
+memset(f->channel_buffers[i], 0, sizeof(float) * f->blocksize_1);
+#ifdef STB_VORBIS_NO_DEFER_FLOOR
+f->floor_buffers[i]   = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2);
+if (f->floor_buffers[i] == NULL) return error(f, VORBIS_outofmem);
+#endif
+}
+if (!init_blocksize(f, 0, f->blocksize_0)) return FALSE;
+if (!init_blocksize(f, 1, f->blocksize_1)) return FALSE;
+f->blocksize[0] = f->blocksize_0;
+f->blocksize[1] = f->blocksize_1;
+#ifdef STB_VORBIS_DIVIDE_TABLE
+if (integer_divide_table[1][1]==0)
+for (i=0; i < DIVTAB_NUMER; ++i)
+for (j=1; j < DIVTAB_DENOM; ++j)
+integer_divide_table[i][j] = i / j;
+#endif
+// compute how much temporary memory is needed
+// 1.
+{
+uint32 imdct_mem = (f->blocksize_1 * sizeof(float) >> 1);
+uint32 classify_mem;
+int i,max_part_read=0;
+for (i=0; i < f->residue_count; ++i) {
+Residue *r = f->residue_config + i;
+unsigned int actual_size = f->blocksize_1 / 2;
+unsigned int limit_r_begin = r->begin < actual_size ? r->begin : actual_size;
+unsigned int limit_r_end   = r->end   < actual_size ? r->end   : actual_size;
+int n_read = limit_r_end - limit_r_begin;
+int part_read = n_read / r->part_size;
+if (part_read > max_part_read)
+max_part_read = part_read;
+}
+#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(uint8 *));
+#else
+classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(int *));
+#endif
+// maximum reasonable partition size is f->blocksize_1
+f->temp_memory_required = classify_mem;
+if (imdct_mem > f->temp_memory_required)
+f->temp_memory_required = imdct_mem;
+}
+if (f->alloc.alloc_buffer) {
+assert(f->temp_offset == f->alloc.alloc_buffer_length_in_bytes);
+// check if there's enough temp memory so we don't error later
+if (f->setup_offset + sizeof(*f) + f->temp_memory_required > (unsigned) f->temp_offset)
+return error(f, VORBIS_outofmem);
+}
+// @TODO: stb_vorbis_seek_start expects first_audio_page_offset to point to a page
+// without PAGEFLAG_continued_packet, so this either points to the first page, or
+// the page after the end of the headers. It might be cleaner to point to a page
+// in the middle of the headers, when that's the page where the first audio packet
+// starts, but we'd have to also correctly skip the end of any continued packet in
+// stb_vorbis_seek_start.
+if (f->next_seg == -1) {
+f->first_audio_page_offset = stb_vorbis_get_file_offset(f);
+} else {
+f->first_audio_page_offset = 0;
+}
+return TRUE;
+}
+static void vorbis_deinit(stb_vorbis *p)
+{
+int i,j;
+setup_free(p, p->vendor);
+for (i=0; i < p->comment_list_length; ++i) {
+setup_free(p, p->comment_list[i]);
+}
+setup_free(p, p->comment_list);
+if (p->residue_config) {
+for (i=0; i < p->residue_count; ++i) {
+Residue *r = p->residue_config+i;
+if (r->classdata) {
+for (j=0; j < p->codebooks[r->classbook].entries; ++j)
+setup_free(p, r->classdata[j]);
+setup_free(p, r->classdata);
+}
+setup_free(p, r->residue_books);
+}
+}
+if (p->codebooks) {
+CHECK(p);
+for (i=0; i < p->codebook_count; ++i) {
+Codebook *c = p->codebooks + i;
+setup_free(p, c->codeword_lengths);
+setup_free(p, c->multiplicands);
+setup_free(p, c->codewords);
+setup_free(p, c->sorted_codewords);
+// c->sorted_values[-1] is the first entry in the array
+setup_free(p, c->sorted_values ? c->sorted_values-1 : NULL);
+}
+setup_free(p, p->codebooks);
+}
+setup_free(p, p->floor_config);
+setup_free(p, p->residue_config);
+if (p->mapping) {
+for (i=0; i < p->mapping_count; ++i)
+setup_free(p, p->mapping[i].chan);
+setup_free(p, p->mapping);
+}
+CHECK(p);
+for (i=0; i < p->channels && i < STB_VORBIS_MAX_CHANNELS; ++i) {
+setup_free(p, p->channel_buffers[i]);
+setup_free(p, p->previous_window[i]);
+#ifdef STB_VORBIS_NO_DEFER_FLOOR
+setup_free(p, p->floor_buffers[i]);
+#endif
+setup_free(p, p->finalY[i]);
+}
+for (i=0; i < 2; ++i) {
+setup_free(p, p->A[i]);
+setup_free(p, p->B[i]);
+setup_free(p, p->C[i]);
+setup_free(p, p->window[i]);
+setup_free(p, p->bit_reverse[i]);
+}
+#ifndef STB_VORBIS_NO_STDIO
+if (p->close_on_free) fclose(p->f);
+#endif
+}
+void stb_vorbis_close(stb_vorbis *p)
+{
+if (p == NULL) return;
+vorbis_deinit(p);
+setup_free(p,p);
+}
+static void vorbis_init(stb_vorbis *p, const stb_vorbis_alloc *z)
+{
+memset(p, 0, sizeof(*p)); // NULL out all malloc'd pointers to start
+if (z) {
+p->alloc = *z;
+p->alloc.alloc_buffer_length_in_bytes &= ~7;
+p->temp_offset = p->alloc.alloc_buffer_length_in_bytes;
+}
+p->eof = 0;
+p->error = VORBIS__no_error;
+p->stream = NULL;
+p->codebooks = NULL;
+p->page_crc_tests = -1;
+#ifndef STB_VORBIS_NO_STDIO
+p->close_on_free = FALSE;
+p->f = NULL;
+#endif
+}
+int stb_vorbis_get_sample_offset(stb_vorbis *f)
+{
+if (f->current_loc_valid)
+return f->current_loc;
+else
+return -1;
+}
+stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f)
+{
+stb_vorbis_info d;
+d.channels = f->channels;
+d.sample_rate = f->sample_rate;
+d.setup_memory_required = f->setup_memory_required;
+d.setup_temp_memory_required = f->setup_temp_memory_required;
+d.temp_memory_required = f->temp_memory_required;
+d.max_frame_size = f->blocksize_1 >> 1;
+return d;
+}
+stb_vorbis_comment stb_vorbis_get_comment(stb_vorbis *f)
+{
+stb_vorbis_comment d;
+d.vendor = f->vendor;
+d.comment_list_length = f->comment_list_length;
+d.comment_list = f->comment_list;
+return d;
+}
+int stb_vorbis_get_error(stb_vorbis *f)
+{
+int e = f->error;
+f->error = VORBIS__no_error;
+return e;
+}
+static stb_vorbis * vorbis_alloc(stb_vorbis *f)
+{
+stb_vorbis *p = (stb_vorbis *) setup_malloc(f, sizeof(*p));
+return p;
+}
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+void stb_vorbis_flush_pushdata(stb_vorbis *f)
+{
+f->previous_length = 0;
+f->page_crc_tests  = 0;
+f->discard_samples_deferred = 0;
+f->current_loc_valid = FALSE;
+f->first_decode = FALSE;
+f->samples_output = 0;
+f->channel_buffer_start = 0;
+f->channel_buffer_end = 0;
+}
+static int vorbis_search_for_page_pushdata(vorb *f, uint8 *data, int data_len)
+{
+int i,n;
+for (i=0; i < f->page_crc_tests; ++i)
+f->scan[i].bytes_done = 0;
+// if we have room for more scans, search for them first, because
+// they may cause us to stop early if their header is incomplete
+if (f->page_crc_tests < STB_VORBIS_PUSHDATA_CRC_COUNT) {
+if (data_len < 4) return 0;
+data_len -= 3; // need to look for 4-byte sequence, so don't miss
+// one that straddles a boundary
+for (i=0; i < data_len; ++i) {
+if (data[i] == 0x4f) {
+if (0==memcmp(data+i, ogg_page_header, 4)) {
+int j,len;
+uint32 crc;
+// make sure we have the whole page header
+if (i+26 >= data_len || i+27+data[i+26] >= data_len) {
+// only read up to this page start, so hopefully we'll
+// have the whole page header start next time
+data_len = i;
+break;
+}
+// ok, we have it all; compute the length of the page
+len = 27 + data[i+26];
+for (j=0; j < data[i+26]; ++j)
+len += data[i+27+j];
+// scan everything up to the embedded crc (which we must 0)
+crc = 0;
+for (j=0; j < 22; ++j)
+crc = crc32_update(crc, data[i+j]);
+// now process 4 0-bytes
+for (   ; j < 26; ++j)
+crc = crc32_update(crc, 0);
+// len is the total number of bytes we need to scan
+n = f->page_crc_tests++;
+f->scan[n].bytes_left = len-j;
+f->scan[n].crc_so_far = crc;
+f->scan[n].goal_crc = data[i+22] + (data[i+23] << 8) + (data[i+24]<<16) + (data[i+25]<<24);
+// if the last frame on a page is continued to the next, then
+// we can't recover the sample_loc immediately
+if (data[i+27+data[i+26]-1] == 255)
+f->scan[n].sample_loc = ~0;
+else
+f->scan[n].sample_loc = data[i+6] + (data[i+7] << 8) + (data[i+ 8]<<16) + (data[i+ 9]<<24);
+f->scan[n].bytes_done = i+j;
+if (f->page_crc_tests == STB_VORBIS_PUSHDATA_CRC_COUNT)
+break;
+// keep going if we still have room for more
+}
+}
+}
+}
+for (i=0; i < f->page_crc_tests;) {
+uint32 crc;
+int j;
+int n = f->scan[i].bytes_done;
+int m = f->scan[i].bytes_left;
+if (m > data_len - n) m = data_len - n;
+// m is the bytes to scan in the current chunk
+crc = f->scan[i].crc_so_far;
+for (j=0; j < m; ++j)
+crc = crc32_update(crc, data[n+j]);
+f->scan[i].bytes_left -= m;
+f->scan[i].crc_so_far = crc;
+if (f->scan[i].bytes_left == 0) {
+// does it match?
+if (f->scan[i].crc_so_far == f->scan[i].goal_crc) {
+// Houston, we have page
+data_len = n+m; // consumption amount is wherever that scan ended
+f->page_crc_tests = -1; // drop out of page scan mode
+f->previous_length = 0; // decode-but-don't-output one frame
+f->next_seg = -1;       // start a new page
+f->current_loc = f->scan[i].sample_loc; // set the current sample location
+// to the amount we'd have decoded had we decoded this page
+f->current_loc_valid = f->current_loc != ~0U;
+return data_len;
+}
+// delete entry
+f->scan[i] = f->scan[--f->page_crc_tests];
+} else {
+++i;
+}
+}
+return data_len;
+}
+// return value: number of bytes we used
+int stb_vorbis_decode_frame_pushdata(
+stb_vorbis *f,                   // the file we're decoding
+const uint8 *data, int data_len, // the memory available for decoding
+int *channels,                   // place to write number of float * buffers
+float ***output,                 // place to write float ** array of float * buffers
+int *samples                     // place to write number of output samples
+)
+{
+int i;
+int len,right,left;
+if (!IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+if (f->page_crc_tests >= 0) {
+*samples = 0;
+return vorbis_search_for_page_pushdata(f, (uint8 *) data, data_len);
+}
+f->stream     = (uint8 *) data;
+f->stream_end = (uint8 *) data + data_len;
+f->error      = VORBIS__no_error;
+// check that we have the entire packet in memory
+if (!is_whole_packet_present(f)) {
+*samples = 0;
+return 0;
+}
+if (!vorbis_decode_packet(f, &len, &left, &right)) {
+// save the actual error we encountered
+enum STBVorbisError error = f->error;
+if (error == VORBIS_bad_packet_type) {
+// flush and resynch
+f->error = VORBIS__no_error;
+while (get8_packet(f) != EOP)
+if (f->eof) break;
+*samples = 0;
+return (int) (f->stream - data);
+}
+if (error == VORBIS_continued_packet_flag_invalid) {
+if (f->previous_length == 0) {
+// we may be resynching, in which case it's ok to hit one
+// of these; just discard the packet
+f->error = VORBIS__no_error;
+while (get8_packet(f) != EOP)
+if (f->eof) break;
+*samples = 0;
+return (int) (f->stream - data);
+}
+}
+// if we get an error while parsing, what to do?
+// well, it DEFINITELY won't work to continue from where we are!
+stb_vorbis_flush_pushdata(f);
+// restore the error that actually made us bail
+f->error = error;
+*samples = 0;
+return 1;
+}
+// success!
+len = vorbis_finish_frame(f, len, left, right);
+for (i=0; i < f->channels; ++i)
+f->outputs[i] = f->channel_buffers[i] + left;
+if (channels) *channels = f->channels;
+*samples = len;
+*output = f->outputs;
+return (int) (f->stream - data);
+}
+stb_vorbis *stb_vorbis_open_pushdata(
+const unsigned char *data, int data_len, // the memory available for decoding
+int *data_used,              // only defined if result is not NULL
+int *error, const stb_vorbis_alloc *alloc)
+{
+stb_vorbis *f, p;
+vorbis_init(&p, alloc);
+p.stream     = (uint8 *) data;
+p.stream_end = (uint8 *) data + data_len;
+p.push_mode  = TRUE;
+if (!start_decoder(&p)) {
+if (p.eof)
+*error = VORBIS_need_more_data;
+else
+*error = p.error;
+vorbis_deinit(&p);
+return NULL;
+}
+f = vorbis_alloc(&p);
+if (f) {
+*f = p;
+*data_used = (int) (f->stream - data);
+*error = 0;
+return f;
+} else {
+vorbis_deinit(&p);
+return NULL;
+}
+}
+#endif // STB_VORBIS_NO_PUSHDATA_API
+unsigned int stb_vorbis_get_file_offset(stb_vorbis *f)
+{
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+if (f->push_mode) return 0;
+#endif
+if (USE_MEMORY(f)) return (unsigned int) (f->stream - f->stream_start);
+#ifndef STB_VORBIS_NO_STDIO
+return (unsigned int) (ftell(f->f) - f->f_start);
+#endif
+}
+#ifndef STB_VORBIS_NO_PULLDATA_API
+//
+// DATA-PULLING API
+//
+static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last)
+{
+for(;;) {
+int n;
+if (f->eof) return 0;
+n = get8(f);
+if (n == 0x4f) { // page header candidate
+unsigned int retry_loc = stb_vorbis_get_file_offset(f);
+int i;
+// check if we're off the end of a file_section stream
+if (retry_loc - 25 > f->stream_len)
+return 0;
+// check the rest of the header
+for (i=1; i < 4; ++i)
+if (get8(f) != ogg_page_header[i])
+break;
+if (f->eof) return 0;
+if (i == 4) {
+uint8 header[27];
+uint32 i, crc, goal, len;
+for (i=0; i < 4; ++i)
+header[i] = ogg_page_header[i];
+for (; i < 27; ++i)
+header[i] = get8(f);
+if (f->eof) return 0;
+if (header[4] != 0) goto invalid;
+goal = header[22] + (header[23] << 8) + (header[24]<<16) + ((uint32)header[25]<<24);
+for (i=22; i < 26; ++i)
+header[i] = 0;
+crc = 0;
+for (i=0; i < 27; ++i)
+crc = crc32_update(crc, header[i]);
+len = 0;
+for (i=0; i < header[26]; ++i) {
+int s = get8(f);
+crc = crc32_update(crc, s);
+len += s;
+}
+if (len && f->eof) return 0;
+for (i=0; i < len; ++i)
+crc = crc32_update(crc, get8(f));
+// finished parsing probable page
+if (crc == goal) {
+// we could now check that it's either got the last
+// page flag set, OR it's followed by the capture
+// pattern, but I guess TECHNICALLY you could have
+// a file with garbage between each ogg page and recover
+// from it automatically? So even though that paranoia
+// might decrease the chance of an invalid decode by
+// another 2^32, not worth it since it would hose those
+// invalid-but-useful files?
+if (end)
+*end = stb_vorbis_get_file_offset(f);
+if (last) {
+if (header[5] & 0x04)
+*last = 1;
+else
+*last = 0;
+}
+set_file_offset(f, retry_loc-1);
+return 1;
+}
+}
+invalid:
+// not a valid page, so rewind and look for next one
+set_file_offset(f, retry_loc);
+}
+}
+}
+#define SAMPLE_unknown  0xffffffff
+// seeking is implemented with a binary search, which narrows down the range to
+// 64K, before using a linear search (because finding the synchronization
+// pattern can be expensive, and the chance we'd find the end page again is
+// relatively high for small ranges)
+//
+// two initial interpolation-style probes are used at the start of the search
+// to try to bound either side of the binary search sensibly, while still
+// working in O(log n) time if they fail.
+static int get_seek_page_info(stb_vorbis *f, ProbedPage *z)
+{
+uint8 header[27], lacing[255];
+int i,len;
+// record where the page starts
+z->page_start = stb_vorbis_get_file_offset(f);
+// parse the header
+getn(f, header, 27);
+if (header[0] != 'O' || header[1] != 'g' || header[2] != 'g' || header[3] != 'S')
+return 0;
+getn(f, lacing, header[26]);
+// determine the length of the payload
+len = 0;
+for (i=0; i < header[26]; ++i)
+len += lacing[i];
+// this implies where the page ends
+z->page_end = z->page_start + 27 + header[26] + len;
+// read the last-decoded sample out of the data
+z->last_decoded_sample = header[6] + (header[7] << 8) + (header[8] << 16) + (header[9] << 24);
+// restore file state to where we were
+set_file_offset(f, z->page_start);
+return 1;
+}
+// rarely used function to seek back to the preceding page while finding the
+// start of a packet
+static int go_to_page_before(stb_vorbis *f, unsigned int limit_offset)
+{
+unsigned int previous_safe, end;
+// now we want to seek back 64K from the limit
+if (limit_offset >= 65536 && limit_offset-65536 >= f->first_audio_page_offset)
+previous_safe = limit_offset - 65536;
+else
+previous_safe = f->first_audio_page_offset;
+set_file_offset(f, previous_safe);
+while (vorbis_find_page(f, &end, NULL)) {
+if (end >= limit_offset && stb_vorbis_get_file_offset(f) < limit_offset)
+return 1;
+set_file_offset(f, end);
+}
+return 0;
+}
+// implements the search logic for finding a page and starting decoding. if
+// the function succeeds, current_loc_valid will be true and current_loc will
+// be less than or equal to the provided sample number (the closer the
+// better).
+static int seek_to_sample_coarse(stb_vorbis *f, uint32 sample_number)
+{
+ProbedPage left, right, mid;
+int i, start_seg_with_known_loc, end_pos, page_start;
+uint32 delta, stream_length, padding, last_sample_limit;
+double offset = 0.0, bytes_per_sample = 0.0;
+int probe = 0;
+// find the last page and validate the target sample
+stream_length = stb_vorbis_stream_length_in_samples(f);
+if (stream_length == 0)            return error(f, VORBIS_seek_without_length);
+if (sample_number > stream_length) return error(f, VORBIS_seek_invalid);
+// this is the maximum difference between the window-center (which is the
+// actual granule position value), and the right-start (which the spec
+// indicates should be the granule position (give or take one)).
+padding = ((f->blocksize_1 - f->blocksize_0) >> 2);
+if (sample_number < padding)
+last_sample_limit = 0;
+else
+last_sample_limit = sample_number - padding;
+left = f->p_first;
+while (left.last_decoded_sample == ~0U) {
+// (untested) the first page does not have a 'last_decoded_sample'
+set_file_offset(f, left.page_end);
+if (!get_seek_page_info(f, &left)) goto error;
+}
+right = f->p_last;
+assert(right.last_decoded_sample != ~0U);
+// starting from the start is handled differently
+if (last_sample_limit <= left.last_decoded_sample) {
+if (stb_vorbis_seek_start(f)) {
+if (f->current_loc > sample_number)
+return error(f, VORBIS_seek_failed);
+return 1;
+}
+return 0;
+}
+while (left.page_end != right.page_start) {
+assert(left.page_end < right.page_start);
+// search range in bytes
+delta = right.page_start - left.page_end;
+if (delta <= 65536) {
+// there's only 64K left to search - handle it linearly
+set_file_offset(f, left.page_end);
+} else {
+if (probe < 2) {
+if (probe == 0) {
+// first probe (interpolate)
+double data_bytes = right.page_end - left.page_start;
+bytes_per_sample = data_bytes / right.last_decoded_sample;
+offset = left.page_start + bytes_per_sample * (last_sample_limit - left.last_decoded_sample);
+} else {
+// second probe (try to bound the other side)
+double error = ((double) last_sample_limit - mid.last_decoded_sample) * bytes_per_sample;
+if (error >= 0 && error <  8000) error =  8000;
+if (error <  0 && error > -8000) error = -8000;
+offset += error * 2;
+}
+// ensure the offset is valid
+if (offset < left.page_end)
+offset = left.page_end;
+if (offset > right.page_start - 65536)
+offset = right.page_start - 65536;
+set_file_offset(f, (unsigned int) offset);
+} else {
+// binary search for large ranges (offset by 32K to ensure
+// we don't hit the right page)
+set_file_offset(f, left.page_end + (delta / 2) - 32768);
+}
+if (!vorbis_find_page(f, NULL, NULL)) goto error;
+}
+for (;;) {
+if (!get_seek_page_info(f, &mid)) goto error;
+if (mid.last_decoded_sample != ~0U) break;
+// (untested) no frames end on this page
+set_file_offset(f, mid.page_end);
+assert(mid.page_start < right.page_start);
+}
+// if we've just found the last page again then we're in a tricky file,
+// and we're close enough (if it wasn't an interpolation probe).
+if (mid.page_start == right.page_start) {
+if (probe >= 2 || delta <= 65536)
+break;
+} else {
+if (last_sample_limit < mid.last_decoded_sample)
+right = mid;
+else
+left = mid;
+}
+++probe;
+}
+// seek back to start of the last packet
+page_start = left.page_start;
+set_file_offset(f, page_start);
+if (!start_page(f)) return error(f, VORBIS_seek_failed);
+end_pos = f->end_seg_with_known_loc;
+assert(end_pos >= 0);
+for (;;) {
+for (i = end_pos; i > 0; --i)
+if (f->segments[i-1] != 255)
+break;
+start_seg_with_known_loc = i;
+if (start_seg_with_known_loc > 0 || !(f->page_flag & PAGEFLAG_continued_packet))
+break;
+// (untested) the final packet begins on an earlier page
+if (!go_to_page_before(f, page_start))
+goto error;
+page_start = stb_vorbis_get_file_offset(f);
+if (!start_page(f)) goto error;
+end_pos = f->segment_count - 1;
+}
+// prepare to start decoding
+f->current_loc_valid = FALSE;
+f->last_seg = FALSE;
+f->valid_bits = 0;
+f->packet_bytes = 0;
+f->bytes_in_seg = 0;
+f->previous_length = 0;
+f->next_seg = start_seg_with_known_loc;
+for (i = 0; i < start_seg_with_known_loc; i++)
+skip(f, f->segments[i]);
+// start decoding (optimizable - this frame is generally discarded)
+if (!vorbis_pump_first_frame(f))
+return 0;
+if (f->current_loc > sample_number)
+return error(f, VORBIS_seek_failed);
+return 1;
+error:
+// try to restore the file to a valid state
+stb_vorbis_seek_start(f);
+return error(f, VORBIS_seek_failed);
+}
+// the same as vorbis_decode_initial, but without advancing
+static int peek_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode)
+{
+int bits_read, bytes_read;
+if (!vorbis_decode_initial(f, p_left_start, p_left_end, p_right_start, p_right_end, mode))
+return 0;
+// either 1 or 2 bytes were read, figure out which so we can rewind
+bits_read = 1 + ilog(f->mode_count-1);
+if (f->mode_config[*mode].blockflag)
+bits_read += 2;
+bytes_read = (bits_read + 7) / 8;
+f->bytes_in_seg += bytes_read;
+f->packet_bytes -= bytes_read;
+skip(f, -bytes_read);
+if (f->next_seg == -1)
+f->next_seg = f->segment_count - 1;
+else
+f->next_seg--;
+f->valid_bits = 0;
+return 1;
+}
+int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number)
+{
+uint32 max_frame_samples;
+if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+// fast page-level search
+if (!seek_to_sample_coarse(f, sample_number))
+return 0;
+assert(f->current_loc_valid);
+assert(f->current_loc <= sample_number);
+// linear search for the relevant packet
+max_frame_samples = (f->blocksize_1*3 - f->blocksize_0) >> 2;
+while (f->current_loc < sample_number) {
+int left_start, left_end, right_start, right_end, mode, frame_samples;
+if (!peek_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode))
+return error(f, VORBIS_seek_failed);
+// calculate the number of samples returned by the next frame
+frame_samples = right_start - left_start;
+if (f->current_loc + frame_samples > sample_number) {
+return 1; // the next frame will contain the sample
+} else if (f->current_loc + frame_samples + max_frame_samples > sample_number) {
+// there's a chance the frame after this could contain the sample
+vorbis_pump_first_frame(f);
+} else {
+// this frame is too early to be relevant
+f->current_loc += frame_samples;
+f->previous_length = 0;
+maybe_start_packet(f);
+flush_packet(f);
+}
+}
+// the next frame should start with the sample
+if (f->current_loc != sample_number) return error(f, VORBIS_seek_failed);
+return 1;
+}
+int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number)
+{
+if (!stb_vorbis_seek_frame(f, sample_number))
+return 0;
+if (sample_number != f->current_loc) {
+int n;
+uint32 frame_start = f->current_loc;
+stb_vorbis_get_frame_float(f, &n, NULL);
+assert(sample_number > frame_start);
+assert(f->channel_buffer_start + (int) (sample_number-frame_start) <= f->channel_buffer_end);
+f->channel_buffer_start += (sample_number - frame_start);
+}
+return 1;
+}
+int stb_vorbis_seek_start(stb_vorbis *f)
+{
+if (IS_PUSH_MODE(f)) { return error(f, VORBIS_invalid_api_mixing); }
+set_file_offset(f, f->first_audio_page_offset);
+f->previous_length = 0;
+f->first_decode = TRUE;
+f->next_seg = -1;
+return vorbis_pump_first_frame(f);
+}
+unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f)
+{
+unsigned int restore_offset, previous_safe;
+unsigned int end, last_page_loc;
+if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+if (!f->total_samples) {
+unsigned int last;
+uint32 lo,hi;
+char header[6];
+// first, store the current decode position so we can restore it
+restore_offset = stb_vorbis_get_file_offset(f);
+// now we want to seek back 64K from the end (the last page must
+// be at most a little less than 64K, but let's allow a little slop)
+if (f->stream_len >= 65536 && f->stream_len-65536 >= f->first_audio_page_offset)
+previous_safe = f->stream_len - 65536;
+else
+previous_safe = f->first_audio_page_offset;
+set_file_offset(f, previous_safe);
+// previous_safe is now our candidate 'earliest known place that seeking
+// to will lead to the final page'
+if (!vorbis_find_page(f, &end, &last)) {
+// if we can't find a page, we're hosed!
+f->error = VORBIS_cant_find_last_page;
+f->total_samples = 0xffffffff;
+goto done;
+}
+// check if there are more pages
+last_page_loc = stb_vorbis_get_file_offset(f);
+// stop when the last_page flag is set, not when we reach eof;
+// this allows us to stop short of a 'file_section' end without
+// explicitly checking the length of the section
+while (!last) {
+set_file_offset(f, end);
+if (!vorbis_find_page(f, &end, &last)) {
+// the last page we found didn't have the 'last page' flag
+// set. whoops!
+break;
+}
+//previous_safe = last_page_loc+1; // NOTE: not used after this point, but note for debugging
+last_page_loc = stb_vorbis_get_file_offset(f);
+}
+set_file_offset(f, last_page_loc);
+// parse the header
+getn(f, (unsigned char *)header, 6);
+// extract the absolute granule position
+lo = get32(f);
+hi = get32(f);
+if (lo == 0xffffffff && hi == 0xffffffff) {
+f->error = VORBIS_cant_find_last_page;
+f->total_samples = SAMPLE_unknown;
+goto done;
+}
+if (hi)
+lo = 0xfffffffe; // saturate
+f->total_samples = lo;
+f->p_last.page_start = last_page_loc;
+f->p_last.page_end   = end;
+f->p_last.last_decoded_sample = lo;
+done:
+set_file_offset(f, restore_offset);
+}
+return f->total_samples == SAMPLE_unknown ? 0 : f->total_samples;
+}
+float stb_vorbis_stream_length_in_seconds(stb_vorbis *f)
+{
+return stb_vorbis_stream_length_in_samples(f) / (float) f->sample_rate;
+}
+int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output)
+{
+int len, right,left,i;
+if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+if (!vorbis_decode_packet(f, &len, &left, &right)) {
+f->channel_buffer_start = f->channel_buffer_end = 0;
+return 0;
+}
+len = vorbis_finish_frame(f, len, left, right);
+for (i=0; i < f->channels; ++i)
+f->outputs[i] = f->channel_buffers[i] + left;
+f->channel_buffer_start = left;
+f->channel_buffer_end   = left+len;
+if (channels) *channels = f->channels;
+if (output)   *output = f->outputs;
+return len;
+}
+#ifndef STB_VORBIS_NO_STDIO
+stb_vorbis * stb_vorbis_open_file_section(FILE *file, int close_on_free, int *error, const stb_vorbis_alloc *alloc, unsigned int length)
+{
+stb_vorbis *f, p;
+vorbis_init(&p, alloc);
+p.f = file;
+p.f_start = (uint32) ftell(file);
+p.stream_len   = length;
+p.close_on_free = close_on_free;
+if (start_decoder(&p)) {
+f = vorbis_alloc(&p);
+if (f) {
+*f = p;
+vorbis_pump_first_frame(f);
+return f;
+}
+}
+if (error) *error = p.error;
+vorbis_deinit(&p);
+return NULL;
+}
+stb_vorbis * stb_vorbis_open_file(FILE *file, int close_on_free, int *error, const stb_vorbis_alloc *alloc)
+{
+unsigned int len, start;
+start = (unsigned int) ftell(file);
+fseek(file, 0, SEEK_END);
+len = (unsigned int) (ftell(file) - start);
+fseek(file, start, SEEK_SET);
+return stb_vorbis_open_file_section(file, close_on_free, error, alloc, len);
+}
+stb_vorbis * stb_vorbis_open_filename(const char *filename, int *error, const stb_vorbis_alloc *alloc)
+{
+FILE *f;
+#if defined(_WIN32) && defined(__STDC_WANT_SECURE_LIB__)
+if (0 != fopen_s(&f, filename, "rb"))
+f = NULL;
+#else
+f = fopen(filename, "rb");
+#endif
+if (f)
+return stb_vorbis_open_file(f, TRUE, error, alloc);
+if (error) *error = VORBIS_file_open_failure;
+return NULL;
+}
+#endif // STB_VORBIS_NO_STDIO
+stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, int *error, const stb_vorbis_alloc *alloc)
+{
+stb_vorbis *f, p;
+if (!data) {
+if (error) *error = VORBIS_unexpected_eof;
+return NULL;
+}
+vorbis_init(&p, alloc);
+p.stream = (uint8 *) data;
+p.stream_end = (uint8 *) data + len;
+p.stream_start = (uint8 *) p.stream;
+p.stream_len = len;
+p.push_mode = FALSE;
+if (start_decoder(&p)) {
+f = vorbis_alloc(&p);
+if (f) {
+*f = p;
+vorbis_pump_first_frame(f);
+if (error) *error = VORBIS__no_error;
+return f;
+}
+}
+if (error) *error = p.error;
+vorbis_deinit(&p);
+return NULL;
+}
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+#define PLAYBACK_MONO     1
+#define PLAYBACK_LEFT     2
+#define PLAYBACK_RIGHT    4
+#define L  (PLAYBACK_LEFT  | PLAYBACK_MONO)
+#define C  (PLAYBACK_LEFT  | PLAYBACK_RIGHT | PLAYBACK_MONO)
+#define R  (PLAYBACK_RIGHT | PLAYBACK_MONO)
+static int8 channel_position[7][6] =
+{
+{ 0 },
+{ C },
+{ L, R },
+{ L, C, R },
+{ L, R, L, R },
+{ L, C, R, L, R },
+{ L, C, R, L, R, C },
+};
+#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT
+typedef union {
+float f;
+int i;
+} float_conv;
+typedef char stb_vorbis_float_size_test[sizeof(float)==4 && sizeof(int) == 4];
+#define FASTDEF(x) float_conv x
+// add (1<<23) to convert to int, then divide by 2^SHIFT, then add 0.5/2^SHIFT to round
+#define MAGIC(SHIFT) (1.5f * (1 << (23-SHIFT)) + 0.5f/(1 << SHIFT))
+#define ADDEND(SHIFT) (((150-SHIFT) << 23) + (1 << 22))
+#define FAST_SCALED_FLOAT_TO_INT(temp,x,s) (temp.f = (x) + MAGIC(s), temp.i - ADDEND(s))
+#define check_endianness()
+#else
+#define FAST_SCALED_FLOAT_TO_INT(temp,x,s) ((int) ((x) * (1 << (s))))
+#define check_endianness()
+#define FASTDEF(x)
+#endif
+static void copy_samples(short *dest, float *src, int len)
+{
+int i;
+check_endianness();
+for (i=0; i < len; ++i) {
+FASTDEF(temp);
+int v = FAST_SCALED_FLOAT_TO_INT(temp, src[i],15);
+if ((unsigned int) (v + 32768) > 65535)
+v = v < 0 ? -32768 : 32767;
+dest[i] = v;
+}
+}
+static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len)
+{
+#define STB_BUFFER_SIZE  32
+float buffer[STB_BUFFER_SIZE];
+int i,j,o,n = STB_BUFFER_SIZE;
+check_endianness();
+for (o = 0; o < len; o += STB_BUFFER_SIZE) {
+memset(buffer, 0, sizeof(buffer));
+if (o + n > len) n = len - o;
+for (j=0; j < num_c; ++j) {
+if (channel_position[num_c][j] & mask) {
+for (i=0; i < n; ++i)
+buffer[i] += data[j][d_offset+o+i];
+}
+}
+for (i=0; i < n; ++i) {
+FASTDEF(temp);
+int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15);
+if ((unsigned int) (v + 32768) > 65535)
+v = v < 0 ? -32768 : 32767;
+output[o+i] = v;
+}
+}
+#undef STB_BUFFER_SIZE
+}
+static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len)
+{
+#define STB_BUFFER_SIZE  32
+float buffer[STB_BUFFER_SIZE];
+int i,j,o,n = STB_BUFFER_SIZE >> 1;
+// o is the offset in the source data
+check_endianness();
+for (o = 0; o < len; o += STB_BUFFER_SIZE >> 1) {
+// o2 is the offset in the output data
+int o2 = o << 1;
+memset(buffer, 0, sizeof(buffer));
+if (o + n > len) n = len - o;
+for (j=0; j < num_c; ++j) {
+int m = channel_position[num_c][j] & (PLAYBACK_LEFT | PLAYBACK_RIGHT);
+if (m == (PLAYBACK_LEFT | PLAYBACK_RIGHT)) {
+for (i=0; i < n; ++i) {
+buffer[i*2+0] += data[j][d_offset+o+i];
+buffer[i*2+1] += data[j][d_offset+o+i];
+}
+} else if (m == PLAYBACK_LEFT) {
+for (i=0; i < n; ++i) {
+buffer[i*2+0] += data[j][d_offset+o+i];
+}
+} else if (m == PLAYBACK_RIGHT) {
+for (i=0; i < n; ++i) {
+buffer[i*2+1] += data[j][d_offset+o+i];
+}
+}
+}
+for (i=0; i < (n<<1); ++i) {
+FASTDEF(temp);
+int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15);
+if ((unsigned int) (v + 32768) > 65535)
+v = v < 0 ? -32768 : 32767;
+output[o2+i] = v;
+}
+}
+#undef STB_BUFFER_SIZE
+}
+static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples)
+{
+int i;
+if (buf_c != data_c && buf_c <= 2 && data_c <= 6) {
+static int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} };
+for (i=0; i < buf_c; ++i)
+compute_samples(channel_selector[buf_c][i], buffer[i]+b_offset, data_c, data, d_offset, samples);
+} else {
+int limit = buf_c < data_c ? buf_c : data_c;
+for (i=0; i < limit; ++i)
+copy_samples(buffer[i]+b_offset, data[i]+d_offset, samples);
+for (   ; i < buf_c; ++i)
+memset(buffer[i]+b_offset, 0, sizeof(short) * samples);
+}
+}
+int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples)
+{
+float **output = NULL;
+int len = stb_vorbis_get_frame_float(f, NULL, &output);
+if (len > num_samples) len = num_samples;
+if (len)
+convert_samples_short(num_c, buffer, 0, f->channels, output, 0, len);
+return len;
+}
+static void convert_channels_short_interleaved(int buf_c, short *buffer, int data_c, float **data, int d_offset, int len)
+{
+int i;
+check_endianness();
+if (buf_c != data_c && buf_c <= 2 && data_c <= 6) {
+assert(buf_c == 2);
+for (i=0; i < buf_c; ++i)
+compute_stereo_samples(buffer, data_c, data, d_offset, len);
+} else {
+int limit = buf_c < data_c ? buf_c : data_c;
+int j;
+for (j=0; j < len; ++j) {
+for (i=0; i < limit; ++i) {
+FASTDEF(temp);
+float f = data[i][d_offset+j];
+int v = FAST_SCALED_FLOAT_TO_INT(temp, f,15);//data[i][d_offset+j],15);
+if ((unsigned int) (v + 32768) > 65535)
+v = v < 0 ? -32768 : 32767;
+*buffer++ = v;
+}
+for (   ; i < buf_c; ++i)
+*buffer++ = 0;
+}
+}
+}
+int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts)
+{
+float **output;
+int len;
+if (num_c == 1) return stb_vorbis_get_frame_short(f,num_c,&buffer, num_shorts);
+len = stb_vorbis_get_frame_float(f, NULL, &output);
+if (len) {
+if (len*num_c > num_shorts) len = num_shorts / num_c;
+convert_channels_short_interleaved(num_c, buffer, f->channels, output, 0, len);
+}
+return len;
+}
+int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts)
+{
+float **outputs;
+int len = num_shorts / channels;
+int n=0;
+while (n < len) {
+int k = f->channel_buffer_end - f->channel_buffer_start;
+if (n+k >= len) k = len - n;
+if (k)
+convert_channels_short_interleaved(channels, buffer, f->channels, f->channel_buffers, f->channel_buffer_start, k);
+buffer += k*channels;
+n += k;
+f->channel_buffer_start += k;
+if (n == len) break;
+if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
+}
+return n;
+}
+int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int len)
+{
+float **outputs;
+int n=0;
+while (n < len) {
+int k = f->channel_buffer_end - f->channel_buffer_start;
+if (n+k >= len) k = len - n;
+if (k)
+convert_samples_short(channels, buffer, n, f->channels, f->channel_buffers, f->channel_buffer_start, k);
+n += k;
+f->channel_buffer_start += k;
+if (n == len) break;
+if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
+}
+return n;
+}
+#ifndef STB_VORBIS_NO_STDIO
+int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output)
+{
+int data_len, offset, total, limit, error;
+short *data;
+stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL);
+if (v == NULL) return -1;
+limit = v->channels * 4096;
+*channels = v->channels;
+if (sample_rate)
+*sample_rate = v->sample_rate;
+offset = data_len = 0;
+total = limit;
+data = (short *) malloc(total * sizeof(*data));
+if (data == NULL) {
+stb_vorbis_close(v);
+return -2;
+}
+for (;;) {
+int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset);
+if (n == 0) break;
+data_len += n;
+offset += n * v->channels;
+if (offset + limit > total) {
+short *data2;
+total *= 2;
+data2 = (short *) realloc(data, total * sizeof(*data));
+if (data2 == NULL) {
+free(data);
+stb_vorbis_close(v);
+return -2;
+}
+data = data2;
+}
+}
+*output = data;
+stb_vorbis_close(v);
+return data_len;
+}
+#endif // NO_STDIO
+int stb_vorbis_decode_memory(const uint8 *mem, int len, int *channels, int *sample_rate, short **output)
+{
+int data_len, offset, total, limit, error;
+short *data;
+stb_vorbis *v = stb_vorbis_open_memory(mem, len, &error, NULL);
+if (v == NULL) return -1;
+limit = v->channels * 4096;
+*channels = v->channels;
+if (sample_rate)
+*sample_rate = v->sample_rate;
+offset = data_len = 0;
+total = limit;
+data = (short *) malloc(total * sizeof(*data));
+if (data == NULL) {
+stb_vorbis_close(v);
+return -2;
+}
+for (;;) {
+int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset);
+if (n == 0) break;
+data_len += n;
+offset += n * v->channels;
+if (offset + limit > total) {
+short *data2;
+total *= 2;
+data2 = (short *) realloc(data, total * sizeof(*data));
+if (data2 == NULL) {
+free(data);
+stb_vorbis_close(v);
+return -2;
+}
+data = data2;
+}
+}
+*output = data;
+stb_vorbis_close(v);
+return data_len;
+}
+#endif // STB_VORBIS_NO_INTEGER_CONVERSION
+int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats)
+{
+float **outputs;
+int len = num_floats / channels;
+int n=0;
+int z = f->channels;
+if (z > channels) z = channels;
+while (n < len) {
+int i,j;
+int k = f->channel_buffer_end - f->channel_buffer_start;
+if (n+k >= len) k = len - n;
+for (j=0; j < k; ++j) {
+for (i=0; i < z; ++i)
+*buffer++ = f->channel_buffers[i][f->channel_buffer_start+j];
+for (   ; i < channels; ++i)
+*buffer++ = 0;
+}
+n += k;
+f->channel_buffer_start += k;
+if (n == len)
+break;
+if (!stb_vorbis_get_frame_float(f, NULL, &outputs))
+break;
+}
+return n;
+}
+int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples)
+{
+float **outputs;
+int n=0;
+int z = f->channels;
+if (z > channels) z = channels;
+while (n < num_samples) {
+int i;
+int k = f->channel_buffer_end - f->channel_buffer_start;
+if (n+k >= num_samples) k = num_samples - n;
+if (k) {
+for (i=0; i < z; ++i)
+memcpy(buffer[i]+n, f->channel_buffers[i]+f->channel_buffer_start, sizeof(float)*k);
+for (   ; i < channels; ++i)
+memset(buffer[i]+n, 0, sizeof(float) * k);
+}
+n += k;
+f->channel_buffer_start += k;
+if (n == num_samples)
+break;
+if (!stb_vorbis_get_frame_float(f, NULL, &outputs))
+break;
+}
+return n;
+}
+#endif // STB_VORBIS_NO_PULLDATA_API
+/* Version history
+1.17    - 2019-07-08 - fix CVE-2019-13217, -13218, -13219, -13220, -13221, -13222, -13223
+found with Mayhem by ForAllSecure
+1.16    - 2019-03-04 - fix warnings
+1.15    - 2019-02-07 - explicit failure if Ogg Skeleton data is found
+1.14    - 2018-02-11 - delete bogus dealloca usage
+1.13    - 2018-01-29 - fix truncation of last frame (hopefully)
+1.12    - 2017-11-21 - limit residue begin/end to blocksize/2 to avoid large temp allocs in bad/corrupt files
+1.11    - 2017-07-23 - fix MinGW compilation
+1.10    - 2017-03-03 - more robust seeking; fix negative ilog(); clear error in open_memory
+1.09    - 2016-04-04 - back out 'avoid discarding last frame' fix from previous version
+1.08    - 2016-04-02 - fixed multiple warnings; fix setup memory leaks;
+avoid discarding last frame of audio data
+1.07    - 2015-01-16 - fixed some warnings, fix mingw, const-correct API
+some more crash fixes when out of memory or with corrupt files
+1.06    - 2015-08-31 - full, correct support for seeking API (Dougall Johnson)
+some crash fixes when out of memory or with corrupt files
+1.05    - 2015-04-19 - don't define __forceinline if it's redundant
+1.04    - 2014-08-27 - fix missing const-correct case in API
+1.03    - 2014-08-07 - Warning fixes
+1.02    - 2014-07-09 - Declare qsort compare function _cdecl on windows
+1.01    - 2014-06-18 - fix stb_vorbis_get_samples_float
+1.0     - 2014-05-26 - fix memory leaks; fix warnings; fix bugs in multichannel
+(API change) report sample rate for decode-full-file funcs
+0.99996 - bracket #include <malloc.h> for macintosh compilation by Laurent Gomila
+0.99995 - use union instead of pointer-cast for fast-float-to-int to avoid alias-optimization problem
+0.99994 - change fast-float-to-int to work in single-precision FPU mode, remove endian-dependence
+0.99993 - remove assert that fired on legal files with empty tables
+0.99992 - rewind-to-start
+0.99991 - bugfix to stb_vorbis_get_samples_short by Bernhard Wodo
+0.9999 - (should have been 0.99990) fix no-CRT support, compiling as C++
+0.9998 - add a full-decode function with a memory source
+0.9997 - fix a bug in the read-from-FILE case in 0.9996 addition
+0.9996 - query length of vorbis stream in samples/seconds
+0.9995 - bugfix to another optimization that only happened in certain files
+0.9994 - bugfix to one of the optimizations that caused significant (but inaudible?) errors
+0.9993 - performance improvements; runs in 99% to 104% of time of reference implementation
+0.9992 - performance improvement of IMDCT; now performs close to reference implementation
+0.9991 - performance improvement of IMDCT
+0.999 - (should have been 0.9990) performance improvement of IMDCT
+0.998 - no-CRT support from Casey Muratori
+0.997 - bugfixes for bugs found by Terje Mathisen
+0.996 - bugfix: fast-huffman decode initialized incorrectly for sparse codebooks; fixing gives 10% speedup - found by Terje Mathisen
+0.995 - bugfix: fix to 'effective' overrun detection - found by Terje Mathisen
+0.994 - bugfix: garbage decode on final VQ symbol of a non-multiple - found by Terje Mathisen
+0.993 - bugfix: pushdata API required 1 extra byte for empty page (failed to consume final page if empty) - found by Terje Mathisen
+0.992 - fixes for MinGW warning
+0.991 - turn fast-float-conversion on by default
+0.990 - fix push-mode seek recovery if you seek into the headers
+0.98b - fix to bad release of 0.98
+0.98 - fix push-mode seek recovery; robustify float-to-int and support non-fast mode
+0.97 - builds under c++ (typecasting, don't use 'class' keyword)
+0.96 - somehow MY 0.95 was right, but the web one was wrong, so here's my 0.95 rereleased as 0.96, fixes a typo in the clamping code
+0.95 - clamping code for 16-bit functions
+0.94 - not publically released
+0.93 - fixed all-zero-floor case (was decoding garbage)
+0.92 - fixed a memory leak
+0.91 - conditional compiles to omit parts of the API and the infrastructure to support them: STB_VORBIS_NO_PULLDATA_API, STB_VORBIS_NO_PUSHDATA_API, STB_VORBIS_NO_STDIO, STB_VORBIS_NO_INTEGER_CONVERSION
+0.90 - first public release
+*/
+#endif // STB_VORBIS_HEADER_ONLY
+/*
+------------------------------------------------------------------------------
+This software is available under 2 licenses -- choose whichever you prefer.
+------------------------------------------------------------------------------
+ALTERNATIVE A - MIT License
+Copyright (c) 2017 Sean Barrett
+Permission is hereby granted, free of charge, to any person obtaining a copy of
+this software and associated documentation files (the "Software"), to deal in
+the Software without restriction, including without limitation the rights to
+use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+of the Software, and to permit persons to whom the Software is furnished to do
+so, subject to the following conditions:
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.
+------------------------------------------------------------------------------
+ALTERNATIVE B - Public Domain (www.unlicense.org)
+This is free and unencumbered software released into the public domain.
+Anyone is free to copy, modify, publish, use, compile, sell, or distribute this
+software, either in source code form or as a compiled binary, for any purpose,
+commercial or non-commercial, and by any means.
+In jurisdictions that recognize copyright laws, the author or authors of this
+software dedicate any and all copyright interest in the software to the public
+domain. We make this dedication for the benefit of the public at large and to
+the detriment of our heirs and successors. We intend this dedication to be an
+overt act of relinquishment in perpetuity of all present and future rights to
+this software under copyright law.
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN
+ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+------------------------------------------------------------------------------
+*/

Mercurial > games > semicongine

comparison semiconginev2/audio/stb_vorbis.c @ 1225:27cd1c21290e compiletime-tests