polycode-engine/include/stb_vorbis.h

// Ogg Vorbis audio decoder - v1.09 - public domain
// http://nothings.org/stb_vorbis/
//
// Original version written by Sean Barrett in 2007.
//
// Originally sponsored by RAD Game Tools. Seeking sponsored
// by Phillip Bennefall, Marc Andersen, Aaron Baker, Elias Software,
// Aras Pranckevicius, and Sean Barrett.
//
// LICENSE
//
//   This software is dual-licensed to the public domain and under the following
//   license: you are granted a perpetual, irrevocable license to copy, modify,
//   publish, and distribute this file as you see fit.
//
// No warranty for any purpose is expressed or implied by the author (nor
// by RAD Game Tools). Report bugs and send enhancements to the author.
//
// 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        alxprd@github
//    Tom Beaumont       Ingo Leitgeb        Nicolas Guillemot
//    Phillip Bennefall  Rohit               Thiago Goulart
//    manxorist@github   saga musix
//
// Partial history:
//    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>
#include "polycode/core/PolyCore.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;

	// get general information about the file
	extern stb_vorbis_info stb_vorbis_get_info(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.

	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(Polycode::CoreFile *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. Morever, 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(Polycode::CoreFile *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 void 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
	};


#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>
#if !(defined(__APPLE__) || defined(MACOSX) || defined(macintosh) || defined(Macintosh))
#include <malloc.h>
#if defined(__linux__) || defined(__linux) || defined(__EMSCRIPTEN__)
#include <alloca.h>
#endif
#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
#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;

// @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;

	// 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;

	uint32 first_audio_page_offset;

	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))
#ifdef dealloca
#define temp_free(f,p)                  (f->alloc.alloc_buffer ? 0 : dealloca(size))
#else
#define temp_free(f,p)                  0
#endif
#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 + 3) & ~3;
	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 + 3) & ~3;
	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 + 3)&~3;
		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};

	// 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 if (n < (1 << 31)) return 30 + log2_4[n >> 30];
	else                return 0; // signed n returns 0
}

#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, 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; }
	// 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;
		// 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];
		assert(z >= 0 && z < 32);
		available[z] = 0;
		add_entry(c, bit_reverse(res), i, m++, len[i], values);
		// propogate availability up the tree
		if (z != len[i]) {
			assert(len[i] >= 0 && len[i] < 32);
			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
	assert(pow((float) r + 1, dim) > entries);
	assert((int) floor(pow((float) r, dim)) <= entries); // (int),floor() as above
	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, y;
} Point;

static int STBV_CDECL point_compare(const void *p, const void *q) {
	Point *a = (Point *) p;
	Point *b = (Point *) 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;
	// 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;
		ProbedPage p;
		len = 0;
		for (i = 0; i < f->segment_count; ++i)
			len += f->segments[i];
		len += 27 + f->segment_count;
		p.page_start = f->first_audio_page_offset;
		p.page_end = p.page_start + len;
		p.last_decoded_sample = loc0;
		f->p_first = p;
	}
	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 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;
		}
	}
	if (f->valid_bits < 0) return 0;
	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]);
		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]);
		}
	}
}

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;
}

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;
	int n_read = r->end - 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 == 1) {
				while (pcount < part_read) {
					int z = r->begin + pcount*r->part_size;
					int c_inter = 0, p_inter = z;
					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 = 0;
							p_inter = z;
						}
					}
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
					++class_set;
#endif
				}
			} else {
				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;

		k00 = z[-0] - z[-8];
		k11 = z[-1] - z[-9];
		z[-0] = z[-0] + z[-8];
		z[-1] = z[-1] + z[-9];
		z[-8] = k00;
		z[-9] = k11;

		k00 = z[-2] - z[-10];
		k11 = z[-3] - z[-11];
		z[-2] = z[-2] + z[-10];
		z[-3] = z[-3] + z[-11];
		z[-10] = (k00 + k11) * A2;
		z[-11] = (k11 - k00) * A2;

		k00 = z[-12] - z[-4];  // reverse to avoid a unary negation
		k11 = z[-5] - z[-13];
		z[-4] = z[-4] + z[-12];
		z[-5] = z[-5] + z[-13];
		z[-12] = k11;
		z[-13] = k00;

		k00 = z[-14] - z[-6];  // reverse to avoid a unary negation
		k11 = z[-7] - z[-15];
		z[-6] = z[-6] + z[-14];
		z[-7] = z[-7] + z[-15];
		z[-14] = (k00 + k11) * A2;
		z[-15] = (k00 - k11) * 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 propogates 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];
	assert(0);
	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
			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

	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 = -n2; // start of first frame is positioned for discard
							  // 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 - (n - right_end);
			// 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;
				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);
		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 void vorbis_pump_first_frame(stb_vorbis *f) {
	int len, right, left;
	if (vorbis_decode_packet(f, &len, &left, &right))
		vorbis_finish_frame(f, len, left, right);
}

#ifndef STB_VORBIS_NO_PUSHDATA_API
static int is_whole_packet_present(stb_vorbis *f, int end_page) {
	// 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 (end_page)
			if (s < f->segment_count - 1)             return error(f, VORBIS_invalid_stream);
		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 (end_page)
			if (s < n - 1)                            return error(f, VORBIS_invalid_stream);
		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

	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)                        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;
	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, TRUE)) {
			// 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_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) {
				c->lookup_values = lookup1_values(c->entries, c->dimensions);
			} 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];
						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 {
			Point 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] = 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].y = j;
			}
			qsort(p, g->values, sizeof(p[0]), point_compare);
			for (j = 0; j < g->values; ++j)
				g->sorted_order[j] = (uint8) p[j].y;
			// precompute the neighbors
			for (j = 2; j < g->values; ++j) {
				int low, hi;
				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;
			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);
#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;
			int n_read = r->end - 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

		f->temp_memory_required = classify_mem;
		if (imdct_mem > f->temp_memory_required)
			f->temp_memory_required = imdct_mem;
	}

	f->first_decode = TRUE;

	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);
	}

	f->first_audio_page_offset = stb_vorbis_get_file_offset(f);

	return TRUE;
}

static void vorbis_deinit(stb_vorbis *p) {
	int i, j;
	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 = (p->alloc.alloc_buffer_length_in_bytes + 3) & ~3;
		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;
}

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, FALSE)) {
		*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;
		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) + (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 preceeding 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;
	double offset, bytes_per_sample;
	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)
		sample_number = 0;
	else
		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 (sample_number <= left.last_decoded_sample) {
		stb_vorbis_seek_start(f);
		return 1;
	}

	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 * (sample_number - left.last_decoded_sample);
				} else {
					// second probe (try to bound the other side)
					double error = ((double) sample_number - 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 (mid.page_start == right.page_start)
			break;

		if (sample_number < 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)
	vorbis_pump_first_frame(f);
	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 will start with the sample
	assert(f->current_loc == sample_number);
	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;
}

void stb_vorbis_seek_start(stb_vorbis *f) {
	if (IS_PUSH_MODE(f)) { error(f, VORBIS_invalid_api_mixing); return; }
	set_file_offset(f, f->first_audio_page_offset);
	f->previous_length = 0;
	f->first_decode = TRUE;
	f->next_seg = -1;
	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;
			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 = fopen(filename, "rb");
	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 == NULL) 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);
			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 BUFFER_SIZE  32
	float buffer[BUFFER_SIZE];
	int i, j, o, n = BUFFER_SIZE;
	check_endianness();
	for (o = 0; o < len; o += 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;
		}
	}
}

static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len) {
#define BUFFER_SIZE  32
	float buffer[BUFFER_SIZE];
	int i, j, o, n = BUFFER_SIZE >> 1;
	// o is the offset in the source data
	check_endianness();
	for (o = 0; o < len; o += 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;
		}
	}
}

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;
	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;
	int z = f->channels;
	if (z > channels) z = channels;
	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;
	int z = f->channels;
	if (z > channels) z = channels;
	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
1.09    - 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