Use stb_vorbis to load ogg files.

build/android/TemplateApp/jni/Android.mk

@@ -62,21 +62,10 @@ LOCAL_MODULE := box2d
 LOCAL_SRC_FILES := $(LIBDIR)/libbox2d.a
 include $(PREBUILT_STATIC_LIBRARY)
 
-include $(CLEAR_VARS)
-LOCAL_MODULE := ogg
-LOCAL_SRC_FILES := $(LIBDIR)/libogg.so
-include $(PREBUILT_SHARED_LIBRARY)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := vorbis
-LOCAL_SRC_FILES := $(LIBDIR)/libvorbis.so
-include $(PREBUILT_SHARED_LIBRARY)
-
 include $(CLEAR_VARS)
 LOCAL_MODULE := TemplateApp
 LOCAL_LDLIBS := -landroid -lEGL -lGLESv2 -lOpenSLES -lz -llog
 LOCAL_STATIC_LIBRARIES := Polycore PolycodeUI Polycode3DPhysics freetype lua physfs box2d BulletDynamics BulletCollision BulletSoftBody LinearMath
-LOCAL_SHARED_LIBRARIES := ogg vorbis
 LOCAL_CFLAGS += -I$(LOCAL_PATH)/../../../../include -DUSE_EGL -DSTRICT_OPENGLES2 -DNO_FP16
 LOCAL_SRC_FILES := PolycodeTemplate.cpp PolycodeTemplateApp.cpp
 include $(BUILD_SHARED_LIBRARY)

include/stb_vorbis.h

@@ -0,0 +1,5267 @@
+// 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/PolyCoreServices.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

src/core/PolyAAssetFileProvider.cpp

@@ -111,5 +111,5 @@ int AAssetFile::seek(long int offset, int origin) {
 }
 
 long AAssetFile::tell() {
-	return AAsset_getLength64(file);
+	return AAsset_getLength64(file) - AAsset_getRemainingLength64(file);
 }

src/core/PolySound.cpp

@@ -1,582 +1,542 @@
-/*
- Copyright (C) 2011 by Ivan Safrin
- 
- Permission is hereby granted, free of charge, to any person obtaining a copy
- of this software and associated documentation files (the "Software"), to deal
- in the Software without restriction, including without limitation the rights
- to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- copies of the Software, and to permit persons to whom the Software is
- furnished to do so, subject to the following conditions:
- 
- The above copyright notice and this permission notice shall be included in
- all copies or substantial portions of the Software.
- 
- THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- THE SOFTWARE.
-*/
-
-#include "polycode/core/PolySound.h"
-
-#ifndef NO_OGG
-	#define OV_EXCLUDE_STATIC_CALLBACKS
-	#include <vorbis/vorbisfile.h>
-#endif
-
-#undef OV_EXCLUDE_STATIC_CALLBACKS
-#include "polycode/core/PolyString.h"
-#include "polycode/core/PolyLogger.h"
-#include "polycode/core/PolySoundManager.h"
-#include "polycode/core/PolyCore.h"
-#include "polycode/core/PolyCoreServices.h"
-#include <stdlib.h>
-#include <string>
-#include <vector>
-#include <stdint.h>
-#include <limits>
-
-#ifndef MAX_FLOAT
-	#define MAX_FLOAT (std::numeric_limits<double>::infinity())
-#endif
-
-#ifndef INT32_MAX
-	#define INT32_MAX (std::numeric_limits<int32_t>::max())
-#endif
-
-#ifndef INT16_MAX
-	#define INT16_MAX (std::numeric_limits<int16_t>::max())
-#endif
-
-using namespace std;
-using namespace Polycode;
-
-AudioStreamingSource::AudioStreamingSource(unsigned int channels, unsigned int freq) : channels(channels), freq(freq) {
-}
-
-unsigned int AudioStreamingSource::getNumChannels() {
-	return channels;
-}
-
-unsigned int AudioStreamingSource::getFrequency() {
-	return freq;
-}
-
-unsigned int AudioStreamingSource::streamData(int16_t *buffer, unsigned int size) {
-	return 0;
-}
-
-#ifndef NO_OGG
-size_t custom_readfunc(void *ptr, size_t size, size_t nmemb, void *datasource) {
-	Polycode::CoreFile *file = (Polycode::CoreFile*) datasource;
-	return file->read(ptr, size, nmemb);
-}
-
-int custom_seekfunc(void *datasource, ogg_int64_t offset, int whence){
-	Polycode::CoreFile *file = (Polycode::CoreFile*) datasource;
-	return file->seek(offset, whence);
-}
-
-int custom_closefunc(void *datasource) {
-	Polycode::CoreFile *file = (Polycode::CoreFile*) datasource;
-	Services()->getCore()->closeFile(file);
-	return 0;
-}
-
-long custom_tellfunc(void *datasource) {
-	CoreFile *file = (CoreFile*) datasource;
-	return file->tell();
-}
-#endif
-
-Sound::Sound(const String& fileName) :	referenceDistance(1), maxDistance(MAX_FLOAT), pitch(1), volume(1), numSamples(-1), streamingSound(false), playing(false), playbackOffset(0), streamingSource(NULL), frequencyAdjust(1.0) {
-	soundLoaded = false;
-	setIsPositional(false);
-	loadFile(fileName);
-
-    if(soundLoaded) {
-        Services()->getSoundManager()->registerSound(this);
-    }
-}
-
-Sound::Sound(int size, const char *data, int channels, unsigned int freq, SoundFormat format) : referenceDistance(1), maxDistance(MAX_FLOAT), pitch(1), volume(1), numSamples(-1), streamingSound(false), playing(false) , playbackOffset(0), streamingSource(NULL), frequencyAdjust(1.0) {
-	setIsPositional(false);
-	soundLoaded = loadBytes(data, size, channels, freq, format);
-	if(soundLoaded) {
-		Services()->getSoundManager()->registerSound(this);
-	}
-}
-
-Sound::Sound(AudioStreamingSource *streamingSource) : referenceDistance(1), maxDistance(MAX_FLOAT), pitch(1), volume(1),  numSamples(-1), streamingSound(true), streamingSource(streamingSource), playing(false), playbackOffset(0), frequencyAdjust(1.0) {
-
-	soundBuffer = (int16_t*) malloc(sizeof(int16_t) * streamingSource->getNumChannels() * POLY_MIX_BUFFER_SIZE);
-	Services()->getSoundManager()->registerSound(this);
-	numChannels = streamingSource->getNumChannels();
-}
-
-void Sound::updateStream(unsigned int streamCount) {
-	if(streamingSource) {
-		playbackOffset = 0;
-		numSamples = streamCount;
-		streamingSource->streamData(soundBuffer, streamCount);
-	}
-}
-
-void Sound::loadFile(String fileName) {
-
-	if(soundLoaded) {
-		free(soundBuffer);
-	}
-
-	String actualFilename = fileName;
-	CoreFile *test = Services()->getCore()->openFile(fileName, "rb");
-	if(!test) {
-		actualFilename = "default/default.wav";
-	} else {
-		Services()->getCore()->closeFile(test);
-	}
-	
-	String extension;
-	size_t found;
-	found=actualFilename.rfind(".");
-	if (found!=string::npos) {
-		extension = actualFilename.substr(found+1);
-	} else {
-		extension = "";
-	}
-	
-	if(extension == "wav" || extension == "WAV") {
-		soundLoaded = loadWAV(actualFilename);
-	} else if(extension == "ogg" || extension == "OGG") {
-		soundLoaded = loadOGG(actualFilename);
-	}
-	
-	this->fileName = actualFilename;
-}
-
-String Sound::getFileName() {
-	return fileName;
-}
-
-Number Sound::getVolume() {
-	return volume;
-}
-
-Number Sound::getPitch() {
-	return pitch;
-}
-
-Sound::~Sound() {
-	free(soundBuffer);
-	Services()->getSoundManager()->unregisterSound(this);
-}
-
-void Sound::soundCheck(bool result, const String& err) {
-	if(!result)
-		Logger::log(err);
-}
-
-unsigned long Sound::readByte32(const unsigned char data[4]) {
-#if TAU_BIG_ENDIAN
-	return (data[0] << 24) + (data[1] << 16) + (data[2] << 8) + data[3];
-#else
-	return (data[3] << 24) + (data[2] << 16) + (data[1] << 8) + data[0];
-#endif
-}
-
-unsigned short Sound::readByte16(const unsigned char data[2]) {
-#if TAU_BIG_ENDIAN
-	return (data[0] << 8) + data[1];
-#else
-	return (data[1] << 8) + data[0];
-#endif	
-}
-
-void Sound::Play(bool loop, bool restartSound) {
-	if(restartSound) {
-		playbackOffset = 0;
-	}
-	playing = true;
-	looped = loop;
-}
-
-bool Sound::isPlaying() {
-	return playing;
-}
-
-bool Sound::isLooped() {
-	return looped;
-}
-
-
-void Sound::setVolume(Number newVolume) {
-	this->volume = newVolume;
-}
-
-void Sound::setPitch(Number newPitch) {
-	this->pitch = newPitch;
-}
-
-void Sound::setSoundPosition(const Vector3 &position) {
-	this->position = position;
-}
-
-void Sound::setSoundVelocity(const Vector3 &velocity) {
-	this->velocity = velocity;
-}
-
-void Sound::setSoundDirection(const Vector3 &direction) {
-	this->direction = direction;
-}
-
-
-Number Sound::getPlaybackTime() {
-	/*
-	float result = 0.0;
-	alGetSourcef(soundSource, AL_SEC_OFFSET, &result);
-	return result;
-	 */
-		//NOAL_TODO
-	return 0.0;
-}
-
-Number Sound::getPlaybackDuration() {
-	/*
-	ALint sizeInBytes;
-	ALint channels;
-	ALint bits;
-	ALint bufferID;
-	alGetSourcei(soundSource, AL_BUFFER, &bufferID);
-	
-	alGetBufferi(bufferID, AL_SIZE, &sizeInBytes);
-	alGetBufferi(bufferID, AL_CHANNELS, &channels);
-	alGetBufferi(bufferID, AL_BITS, &bits);
-
-	int lengthInSamples = sizeInBytes * 8 / (channels * bits);
-
-	ALint frequency;
-	alGetBufferi(bufferID, AL_FREQUENCY, &frequency);
-	Number durationInSeconds = (float)lengthInSamples / (float)frequency;
-	
-	return durationInSeconds;
-	 */
-		//NOAL_TODO
-	return 0.0;
-}
-		
-int Sound::getOffset() {
-	return playbackOffset;
-}
-
-void Sound::setOffset(unsigned int offset) {
-	playbackOffset = (offset);
-	
-	Number adjustedOffset = ((Number)playbackOffset) * pitch * frequencyAdjust;
-	
-	if((unsigned int)adjustedOffset >= numSamples) {
-		playbackOffset = 0;
-		if(!looped && !streamingSource) {
-			playing = false;
-		}
-	}
-}
-
-void Sound::seekTo(Number time) {
-	/*
-	if(time > getPlaybackDuration())
-		return;
-	alSourcef(soundSource, AL_SEC_OFFSET, time);
-	checkALError("Seek");
-	 */
-			//NOAL_TODO
-}
-
-int Sound::getSampleLength() {
-	return numSamples;
-}
-
-void Sound::setPositionalProperties(Number referenceDistance, Number maxDistance) { 
-	setReferenceDistance(referenceDistance);
-	setMaxDistance(maxDistance);
-}
-
-void Sound::setReferenceDistance(Number referenceDistance) {
-	this->referenceDistance = referenceDistance;
-}
-
-void Sound::setMaxDistance(Number maxDistance) {
-	this->maxDistance = maxDistance;
-}
-		
-Number Sound::getReferenceDistance() {
-	return referenceDistance;
-}
-
-Number Sound::getMaxDistance() {
-	return maxDistance;
-}
-
-void Sound::setIsPositional(bool isPositional) {
-	this->isPositional = isPositional;
-}
-
-void Sound::Stop() {
-	playing = false;
-}
-
-
-Number Sound::getSampleAsNumber(unsigned int offset, unsigned int channel, const Vector3 &position, const Quaternion &orientation) {
-	Number adjustedOffset = ((Number)offset) * pitch * frequencyAdjust;
-	Number ret;
-	if(isPositional) {
-		ret = (((Number)(soundBuffer[((((unsigned int )adjustedOffset)%numSamples)*numChannels)])/((Number)INT16_MAX))) * volume;
-		ret = modulateSampleForListener(ret, channel, position, orientation);
-	} else {
-		ret = (((Number)(soundBuffer[((((unsigned int )adjustedOffset)%numSamples)*numChannels)+(channel % numChannels)])/((Number)INT16_MAX))) * volume;
-	}
-	return ret;
-}
-
-Number Sound::modulateSampleForListener(Number sample, unsigned int channel, const Vector3 &position, const Quaternion &orientation) {
-	
-	// setup different channel configurations here
-	// if(STEREO) {
-	Vector3 earDirection;
-	if(channel) {
-		earDirection = Vector3(-1.0, 0.0, 0.0);
-	} else {
-		earDirection = Vector3(1.0, 0.0, 0.0);
-	}
-	earDirection = orientation.applyTo(earDirection);
-	
-	Vector3 dir = position - this->position;
-	dir.Normalize();
-	Number muliplier = earDirection.dot(dir);
-	if(muliplier < 0.0) {
-		muliplier = 0.0;
-	}
-	
-	Number ret = sample * (0.1 + (muliplier * 0.9)); // bleed 0.1 into the other ear
-	Number distance = position.distance(this->position);
-	Number attenuate = 0.5 * pow(referenceDistance/distance, 2.0);
-	
-	attenuate = MIN(attenuate, 1.0);
-	attenuate = MAX(attenuate, 0.0);
-	ret *= attenuate;
-	return ret;
-}
-
-bool Sound::loadBytes(const char *data, int size, int channels, unsigned int freq, SoundFormat format) {
-	
-	if(format == SoundFormatUnsupported) {
-		Logger::log("[%s] Error: sound format unsupported!\n", fileName.c_str());
-		return false;
-	}
-	
-	soundBuffer = (int16_t*) malloc(sizeof(int16_t) * channels * size);
-	
-	int16_t *soundBufferPtr = soundBuffer;
-	
-	unsigned int dataOffset = 0;
-	
-	switch(format) {
-		case SoundFormat8:
-			numSamples = size / channels;
-			break;
-		case SoundFormat16:
-			numSamples = size / channels / 2;
-			break;
-		case SoundFormat32:
-			numSamples = size / channels / 4;
-			break;
-		default:
-		break;
-	}
-	
-	for(int i=0; i < numSamples; i++){
-		for(int c=0; c < channels; c++) {
-			switch(format) {
-				case SoundFormat8:
-					*soundBufferPtr = ((int8_t*)data)[dataOffset];
-				break;
-				case SoundFormat16:
-					*soundBufferPtr = ((int16_t*)data)[dataOffset];
-				break;
-				case SoundFormat32:
-					*soundBufferPtr = ((int32_t*)data)[dataOffset];
-				break;
-				default:
-				break;
-			}
-			soundBufferPtr++;
-			dataOffset++;
-		}
-	}
-	
-	numChannels = channels;
-	frequency = freq;
-	
-	// adjust for different frequency
-	frequencyAdjust = (Number)freq/(Number)POLY_AUDIO_FREQ;
-	
-	return true;
-}
-
-unsigned int Sound::getFrequency() {
-	return frequency;
-}
-
-
-bool Sound::loadOGG(const String& fileName) {
-#ifndef NO_OGG
-
-	vector<char> data;
-	int bitStream;
-	long bytes;
-	char array[BUFFER_SIZE];
-	
-	CoreFile *f = Services()->getCore()->openFile(fileName.c_str(), "rb");
-	if(!f) {
-		Logger::log("Error loading OGG file!\n");
-		return false;
-	}
-	vorbis_info *pInfo;
-	OggVorbis_File oggFile; 
-	
-	ov_callbacks callbacks;
-	callbacks.read_func = custom_readfunc;
-	callbacks.seek_func = custom_seekfunc;
-	callbacks.close_func = custom_closefunc;
-	callbacks.tell_func = custom_tellfunc;
-	
-	ov_open_callbacks( (void*)f, &oggFile, NULL, 0, callbacks);
-	pInfo = ov_info(&oggFile, -1);
-
-	do {
-		bytes = ov_read(&oggFile, array, BUFFER_SIZE, 0, 2, 1, &bitStream);
-		data.insert(data.end(), array, array + bytes);
-	} while (bytes > 0);
-	
-	bool retVal = loadBytes(data.data(), data.size(), pInfo->channels, pInfo->rate, SoundFormat16);
-	ov_clear(&oggFile);
-	return retVal;
-#else
-	return false;
-#endif
-}
-
-bool Sound::loadWAV(const String& fileName) {
-	
-	long bytes;
-	vector <char> data;
-	
-	// Local resources
-	CoreFile *f = NULL;
-	char *array = NULL;
-	
-	// Open for binary reading
-	f = Services()->getCore()->openFile(fileName.c_str(), "rb");
-	if (!f) {
-		Logger::log("LoadWav: Could not load wav from " + fileName);
-		return false;
-	}
-	
-	// buffers
-	char magic[5];
-	magic[4] = '\0';
-	unsigned char data32[4];
-	unsigned char data16[2];
-	
-	// check magic
-	soundCheck(f->read(magic,4,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
-	soundCheck(String(magic) == "RIFF", "LoadWav: Wrong wav file format. This file is not a .wav file (no RIFF magic): "+ fileName );
-	
-	// skip 4 bytes (file size)
-	f->seek(4,SEEK_CUR);
-	
-	// check file format
-	soundCheck(f->read(magic,4,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
-	soundCheck(String(magic) == "WAVE", "LoadWav: Wrong wav file format. This file is not a .wav file (no WAVE format): "+ fileName );
-	
-	// check 'fmt ' sub chunk (1)
-	soundCheck(f->read(magic,4,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
-	soundCheck(String(magic) == "fmt ", "LoadWav: Wrong wav file format. This file is not a .wav file (no 'fmt ' subchunk): "+ fileName );
-	
-	// read (1)'s size
-	soundCheck(f->read(data32,4,1)	 == 1, "LoadWav: Cannot read wav file "+ fileName );
-	unsigned long subChunk1Size = readByte32(data32);
-	soundCheck(subChunk1Size >= 16, "Wrong wav file format. This file is not a .wav file ('fmt ' chunk too small, truncated file?): "+ fileName );
-	
-	// check PCM audio format
-	soundCheck(f->read(data16,2,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
-	unsigned short audioFormat = readByte16(data16);
-	soundCheck(audioFormat == 1, "LoadWav: Wrong wav file format. This file is not a .wav file (audio format is not PCM): "+ fileName );
-	
-	// read number of channels
-	soundCheck(f->read(data16,2,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
-	unsigned short channels = readByte16(data16);
-	
-	// read frequency (sample rate)
-	soundCheck(f->read(data32,4,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
-	unsigned long frequency = readByte32(data32);
-	
-	// skip 6 bytes (Byte rate (4), Block align (2))
-	f->seek(6,SEEK_CUR);
-	
-	// read bits per sample
-	soundCheck(f->read(data16,2,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
-	unsigned short bps = readByte16(data16);
-	
-	SoundFormat format = SoundFormatUnsupported;
-	
-	switch(bps) {
-		case 8:
-			format = SoundFormat8;
-		break;
-		case 16:
-			format = SoundFormat16;
-		break;
-		case 32:
-			format = SoundFormat32;
-		break;
-			
-	}
-	
-	// check 'data' sub chunk (2)
-	soundCheck(f->read(magic,4,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
-	soundCheck(String(magic) == "data", "LoadWav: Wrong wav file format. This file is not a .wav file (no data subchunk): "+ fileName );
-	
-	soundCheck(f->read(data32,4,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
-	unsigned long subChunk2Size = readByte32(data32);
-	
-	array = new char[BUFFER_SIZE];
-	
-	while (data.size() != subChunk2Size) {
-		// Read up to a buffer's worth of decoded sound data
-		bytes = f->read(array, 1, BUFFER_SIZE);
-		
-		if (bytes <= 0)
-			break;
-		
-		if (data.size() + bytes > subChunk2Size)
-			bytes = subChunk2Size - data.size();
-		
-		// Append to end of buffer
-		data.insert(data.end(), array, array + bytes);
-	};
-	
-	delete []array;
-	array = NULL;
-	
-	Services()->getCore()->closeFile(f);
-	f = NULL;
-
-	return loadBytes(&data[0], data.size(), channels, frequency, format);
-
-}
- 
-
-//NOAL_TODO
+/*
+ Copyright (C) 2011 by Ivan Safrin
+ 
+ Permission is hereby granted, free of charge, to any person obtaining a copy
+ of this software and associated documentation files (the "Software"), to deal
+ in the Software without restriction, including without limitation the rights
+ to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ copies of the Software, and to permit persons to whom the Software is
+ furnished to do so, subject to the following conditions:
+ 
+ The above copyright notice and this permission notice shall be included in
+ all copies or substantial portions of the Software.
+ 
+ THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ THE SOFTWARE.
+*/
+
+#include "polycode/core/PolySound.h"
+
+#define STB_VORBIS_HEADER_ONLY
+#include "stb_vorbis.h"
+
+#include "polycode/core/PolyString.h"
+#include "polycode/core/PolyLogger.h"
+#include "polycode/core/PolySoundManager.h"
+#include "polycode/core/PolyCore.h"
+#include "polycode/core/PolyCoreServices.h"
+#include <stdlib.h>
+#include <string>
+#include <vector>
+#include <stdint.h>
+#include <limits>
+
+#ifndef MAX_FLOAT
+	#define MAX_FLOAT (std::numeric_limits<double>::infinity())
+#endif
+
+#ifndef INT32_MAX
+	#define INT32_MAX (std::numeric_limits<int32_t>::max())
+#endif
+
+#ifndef INT16_MAX
+	#define INT16_MAX (std::numeric_limits<int16_t>::max())
+#endif
+
+using namespace std;
+using namespace Polycode;
+
+AudioStreamingSource::AudioStreamingSource(unsigned int channels, unsigned int freq) : channels(channels), freq(freq) {
+}
+
+unsigned int AudioStreamingSource::getNumChannels() {
+	return channels;
+}
+
+unsigned int AudioStreamingSource::getFrequency() {
+	return freq;
+}
+
+unsigned int AudioStreamingSource::streamData(int16_t *buffer, unsigned int size) {
+	return 0;
+}
+
+Sound::Sound(const String& fileName) :	referenceDistance(1), maxDistance(MAX_FLOAT), pitch(1), volume(1), numSamples(-1), streamingSound(false), playing(false), playbackOffset(0), streamingSource(NULL), frequencyAdjust(1.0) {
+	soundLoaded = false;
+	setIsPositional(false);
+	loadFile(fileName);
+
+    if(soundLoaded) {
+        Services()->getSoundManager()->registerSound(this);
+    }
+}
+
+Sound::Sound(int size, const char *data, int channels, unsigned int freq, SoundFormat format) : referenceDistance(1), maxDistance(MAX_FLOAT), pitch(1), volume(1), numSamples(-1), streamingSound(false), playing(false) , playbackOffset(0), streamingSource(NULL), frequencyAdjust(1.0) {
+	setIsPositional(false);
+	soundLoaded = loadBytes(data, size, channels, freq, format);
+	if(soundLoaded) {
+		Services()->getSoundManager()->registerSound(this);
+	}
+}
+
+Sound::Sound(AudioStreamingSource *streamingSource) : referenceDistance(1), maxDistance(MAX_FLOAT), pitch(1), volume(1),  numSamples(-1), streamingSound(true), streamingSource(streamingSource), playing(false), playbackOffset(0), frequencyAdjust(1.0) {
+
+	soundBuffer = (int16_t*) malloc(sizeof(int16_t) * streamingSource->getNumChannels() * POLY_MIX_BUFFER_SIZE);
+	Services()->getSoundManager()->registerSound(this);
+	numChannels = streamingSource->getNumChannels();
+}
+
+void Sound::updateStream(unsigned int streamCount) {
+	if(streamingSource) {
+		playbackOffset = 0;
+		numSamples = streamCount;
+		streamingSource->streamData(soundBuffer, streamCount);
+	}
+}
+
+void Sound::loadFile(String fileName) {
+
+	if(soundLoaded) {
+		free(soundBuffer);
+	}
+
+	String actualFilename = fileName;
+	CoreFile *test = Services()->getCore()->openFile(fileName, "rb");
+	if(!test) {
+		actualFilename = "default/default.wav";
+	} else {
+		Services()->getCore()->closeFile(test);
+	}
+	
+	String extension;
+	size_t found;
+	found=actualFilename.rfind(".");
+	if (found!=string::npos) {
+		extension = actualFilename.substr(found+1);
+	} else {
+		extension = "";
+	}
+	
+	if(extension == "wav" || extension == "WAV") {
+		soundLoaded = loadWAV(actualFilename);
+	} else if(extension == "ogg" || extension == "OGG") {
+		soundLoaded = loadOGG(actualFilename);
+	}
+	
+	this->fileName = actualFilename;
+}
+
+String Sound::getFileName() {
+	return fileName;
+}
+
+Number Sound::getVolume() {
+	return volume;
+}
+
+Number Sound::getPitch() {
+	return pitch;
+}
+
+Sound::~Sound() {
+	free(soundBuffer);
+	Services()->getSoundManager()->unregisterSound(this);
+}
+
+void Sound::soundCheck(bool result, const String& err) {
+	if(!result)
+		Logger::log(err);
+}
+
+unsigned long Sound::readByte32(const unsigned char data[4]) {
+#if TAU_BIG_ENDIAN
+	return (data[0] << 24) + (data[1] << 16) + (data[2] << 8) + data[3];
+#else
+	return (data[3] << 24) + (data[2] << 16) + (data[1] << 8) + data[0];
+#endif
+}
+
+unsigned short Sound::readByte16(const unsigned char data[2]) {
+#if TAU_BIG_ENDIAN
+	return (data[0] << 8) + data[1];
+#else
+	return (data[1] << 8) + data[0];
+#endif	
+}
+
+void Sound::Play(bool loop, bool restartSound) {
+	if(restartSound) {
+		playbackOffset = 0;
+	}
+	playing = true;
+	looped = loop;
+}
+
+bool Sound::isPlaying() {
+	return playing;
+}
+
+bool Sound::isLooped() {
+	return looped;
+}
+
+
+void Sound::setVolume(Number newVolume) {
+	this->volume = newVolume;
+}
+
+void Sound::setPitch(Number newPitch) {
+	this->pitch = newPitch;
+}
+
+void Sound::setSoundPosition(const Vector3 &position) {
+	this->position = position;
+}
+
+void Sound::setSoundVelocity(const Vector3 &velocity) {
+	this->velocity = velocity;
+}
+
+void Sound::setSoundDirection(const Vector3 &direction) {
+	this->direction = direction;
+}
+
+
+Number Sound::getPlaybackTime() {
+	/*
+	float result = 0.0;
+	alGetSourcef(soundSource, AL_SEC_OFFSET, &result);
+	return result;
+	 */
+		//NOAL_TODO
+	return 0.0;
+}
+
+Number Sound::getPlaybackDuration() {
+	/*
+	ALint sizeInBytes;
+	ALint channels;
+	ALint bits;
+	ALint bufferID;
+	alGetSourcei(soundSource, AL_BUFFER, &bufferID);
+	
+	alGetBufferi(bufferID, AL_SIZE, &sizeInBytes);
+	alGetBufferi(bufferID, AL_CHANNELS, &channels);
+	alGetBufferi(bufferID, AL_BITS, &bits);
+
+	int lengthInSamples = sizeInBytes * 8 / (channels * bits);
+
+	ALint frequency;
+	alGetBufferi(bufferID, AL_FREQUENCY, &frequency);
+	Number durationInSeconds = (float)lengthInSamples / (float)frequency;
+	
+	return durationInSeconds;
+	 */
+		//NOAL_TODO
+	return 0.0;
+}
+		
+int Sound::getOffset() {
+	return playbackOffset;
+}
+
+void Sound::setOffset(unsigned int offset) {
+	playbackOffset = (offset);
+	
+	Number adjustedOffset = ((Number)playbackOffset) * pitch * frequencyAdjust;
+	
+	if((unsigned int)adjustedOffset >= numSamples) {
+		playbackOffset = 0;
+		if(!looped && !streamingSource) {
+			playing = false;
+		}
+	}
+}
+
+void Sound::seekTo(Number time) {
+	/*
+	if(time > getPlaybackDuration())
+		return;
+	alSourcef(soundSource, AL_SEC_OFFSET, time);
+	checkALError("Seek");
+	 */
+			//NOAL_TODO
+}
+
+int Sound::getSampleLength() {
+	return numSamples;
+}
+
+void Sound::setPositionalProperties(Number referenceDistance, Number maxDistance) { 
+	setReferenceDistance(referenceDistance);
+	setMaxDistance(maxDistance);
+}
+
+void Sound::setReferenceDistance(Number referenceDistance) {
+	this->referenceDistance = referenceDistance;
+}
+
+void Sound::setMaxDistance(Number maxDistance) {
+	this->maxDistance = maxDistance;
+}
+		
+Number Sound::getReferenceDistance() {
+	return referenceDistance;
+}
+
+Number Sound::getMaxDistance() {
+	return maxDistance;
+}
+
+void Sound::setIsPositional(bool isPositional) {
+	this->isPositional = isPositional;
+}
+
+void Sound::Stop() {
+	playing = false;
+}
+
+
+Number Sound::getSampleAsNumber(unsigned int offset, unsigned int channel, const Vector3 &position, const Quaternion &orientation) {
+	Number adjustedOffset = ((Number)offset) * pitch * frequencyAdjust;
+	Number ret;
+	if(isPositional) {
+		ret = (((Number)(soundBuffer[((((unsigned int )adjustedOffset)%numSamples)*numChannels)])/((Number)INT16_MAX))) * volume;
+		ret = modulateSampleForListener(ret, channel, position, orientation);
+	} else {
+		ret = (((Number)(soundBuffer[((((unsigned int )adjustedOffset)%numSamples)*numChannels)+(channel % numChannels)])/((Number)INT16_MAX))) * volume;
+	}
+	return ret;
+}
+
+Number Sound::modulateSampleForListener(Number sample, unsigned int channel, const Vector3 &position, const Quaternion &orientation) {
+	
+	// setup different channel configurations here
+	// if(STEREO) {
+	Vector3 earDirection;
+	if(channel) {
+		earDirection = Vector3(-1.0, 0.0, 0.0);
+	} else {
+		earDirection = Vector3(1.0, 0.0, 0.0);
+	}
+	earDirection = orientation.applyTo(earDirection);
+	
+	Vector3 dir = position - this->position;
+	dir.Normalize();
+	Number muliplier = earDirection.dot(dir);
+	if(muliplier < 0.0) {
+		muliplier = 0.0;
+	}
+	
+	Number ret = sample * (0.1 + (muliplier * 0.9)); // bleed 0.1 into the other ear
+	Number distance = position.distance(this->position);
+	Number attenuate = 0.5 * pow(referenceDistance/distance, 2.0);
+	
+	attenuate = MIN(attenuate, 1.0);
+	attenuate = MAX(attenuate, 0.0);
+	ret *= attenuate;
+	return ret;
+}
+
+bool Sound::loadBytes(const char *data, int size, int channels, unsigned int freq, SoundFormat format) {
+	
+	if(format == SoundFormatUnsupported) {
+		Logger::log("[%s] Error: sound format unsupported!\n", fileName.c_str());
+		return false;
+	}
+	
+	soundBuffer = (int16_t*) malloc(sizeof(int16_t) * channels * size);
+	
+	int16_t *soundBufferPtr = soundBuffer;
+	
+	unsigned int dataOffset = 0;
+	
+	switch(format) {
+		case SoundFormat8:
+			numSamples = size / channels;
+			break;
+		case SoundFormat16:
+			numSamples = size / channels / 2;
+			break;
+		case SoundFormat32:
+			numSamples = size / channels / 4;
+			break;
+		default:
+		break;
+	}
+	
+	for(int i=0; i < numSamples; i++){
+		for(int c=0; c < channels; c++) {
+			switch(format) {
+				case SoundFormat8:
+					*soundBufferPtr = ((int8_t*)data)[dataOffset];
+				break;
+				case SoundFormat16:
+					*soundBufferPtr = ((int16_t*)data)[dataOffset];
+				break;
+				case SoundFormat32:
+					*soundBufferPtr = ((int32_t*)data)[dataOffset];
+				break;
+				default:
+				break;
+			}
+			soundBufferPtr++;
+			dataOffset++;
+		}
+	}
+	
+	numChannels = channels;
+	frequency = freq;
+	
+	// adjust for different frequency
+	frequencyAdjust = (Number)freq/(Number)POLY_AUDIO_FREQ;
+	
+	return true;
+}
+
+unsigned int Sound::getFrequency() {
+	return frequency;
+}
+
+
+bool Sound::loadOGG(const String& fileName) {
+	CoreFile *f = Services()->getCore()->openFile(fileName.c_str(), "rb");
+	if (!f) {
+		Logger::log("Error loading OGG file!\n");
+		return false;
+	}
+	Services()->getCore()->closeFile(f);
+
+	short *decoded;
+	int channels, len, sample_rate;
+	len = stb_vorbis_decode_filename(fileName.c_str(), &channels, &sample_rate, &decoded);
+	if (len <= 0) {
+		return false;
+	}
+
+	numChannels = channels;
+	numSamples = len;
+	soundBuffer = decoded;
+	frequency = sample_rate;
+	frequencyAdjust = sample_rate / POLY_AUDIO_FREQ;
+
+	return true;
+}
+
+bool Sound::loadWAV(const String& fileName) {
+	
+	long bytes;
+	vector <char> data;
+	
+	// Local resources
+	CoreFile *f = NULL;
+	char *array = NULL;
+	
+	// Open for binary reading
+	f = Services()->getCore()->openFile(fileName.c_str(), "rb");
+	if (!f) {
+		Logger::log("LoadWav: Could not load wav from " + fileName);
+		return false;
+	}
+	
+	// buffers
+	char magic[5];
+	magic[4] = '\0';
+	unsigned char data32[4];
+	unsigned char data16[2];
+	
+	// check magic
+	soundCheck(f->read(magic,4,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
+	soundCheck(String(magic) == "RIFF", "LoadWav: Wrong wav file format. This file is not a .wav file (no RIFF magic): "+ fileName );
+	
+	// skip 4 bytes (file size)
+	f->seek(4,SEEK_CUR);
+	
+	// check file format
+	soundCheck(f->read(magic,4,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
+	soundCheck(String(magic) == "WAVE", "LoadWav: Wrong wav file format. This file is not a .wav file (no WAVE format): "+ fileName );
+	
+	// check 'fmt ' sub chunk (1)
+	soundCheck(f->read(magic,4,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
+	soundCheck(String(magic) == "fmt ", "LoadWav: Wrong wav file format. This file is not a .wav file (no 'fmt ' subchunk): "+ fileName );
+	
+	// read (1)'s size
+	soundCheck(f->read(data32,4,1)	 == 1, "LoadWav: Cannot read wav file "+ fileName );
+	unsigned long subChunk1Size = readByte32(data32);
+	soundCheck(subChunk1Size >= 16, "Wrong wav file format. This file is not a .wav file ('fmt ' chunk too small, truncated file?): "+ fileName );
+	
+	// check PCM audio format
+	soundCheck(f->read(data16,2,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
+	unsigned short audioFormat = readByte16(data16);
+	soundCheck(audioFormat == 1, "LoadWav: Wrong wav file format. This file is not a .wav file (audio format is not PCM): "+ fileName );
+	
+	// read number of channels
+	soundCheck(f->read(data16,2,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
+	unsigned short channels = readByte16(data16);
+	
+	// read frequency (sample rate)
+	soundCheck(f->read(data32,4,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
+	unsigned long frequency = readByte32(data32);
+	
+	// skip 6 bytes (Byte rate (4), Block align (2))
+	f->seek(6,SEEK_CUR);
+	
+	// read bits per sample
+	soundCheck(f->read(data16,2,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
+	unsigned short bps = readByte16(data16);
+	
+	SoundFormat format = SoundFormatUnsupported;
+	
+	switch(bps) {
+		case 8:
+			format = SoundFormat8;
+		break;
+		case 16:
+			format = SoundFormat16;
+		break;
+		case 32:
+			format = SoundFormat32;
+		break;
+			
+	}
+	
+	// check 'data' sub chunk (2)
+	soundCheck(f->read(magic,4,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
+	soundCheck(String(magic) == "data", "LoadWav: Wrong wav file format. This file is not a .wav file (no data subchunk): "+ fileName );
+	
+	soundCheck(f->read(data32,4,1) == 1, "LoadWav: Cannot read wav file "+ fileName );
+	unsigned long subChunk2Size = readByte32(data32);
+	
+	array = new char[BUFFER_SIZE];
+	
+	while (data.size() != subChunk2Size) {
+		// Read up to a buffer's worth of decoded sound data
+		bytes = f->read(array, 1, BUFFER_SIZE);
+		
+		if (bytes <= 0)
+			break;
+		
+		if (data.size() + bytes > subChunk2Size)
+			bytes = subChunk2Size - data.size();
+		
+		// Append to end of buffer
+		data.insert(data.end(), array, array + bytes);
+	};
+	
+	delete []array;
+	array = NULL;
+	
+	Services()->getCore()->closeFile(f);
+	f = NULL;
+
+	return loadBytes(&data[0], data.size(), channels, frequency, format);
+
+}
+ 
+
+//NOAL_TODO

src/core/stb_vorbis.cpp

@@ -0,0 +1,5269 @@
+// 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.
+#define STB_VORBIS_HEADER_ONLY
+#include "stb_vorbis.h"
+#undef STB_VORBIS_HEADER_ONLY
+//////////////////////////////////////////////////////////////////////////////
+//
+//  HEADER BEGINS HERE
+//
+
+#ifndef STB_VORBIS_INCLUDE_STB_VORBIS_H
+#define STB_VORBIS_INCLUDE_STB_VORBIS_H
+
+#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
+#define STB_VORBIS_NO_STDIO 1
+#endif
+
+#ifndef STB_VORBIS_NO_STDIO
+#include <stdio.h>
+#endif
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+	///////////   THREAD SAFETY
+
+	// Individual stb_vorbis* handles are not thread-safe; you cannot decode from
+	// them from multiple threads at the same time. However, you can have multiple
+	// stb_vorbis* handles and decode from them independently in multiple thrads.
+
+
+	///////////   MEMORY ALLOCATION
+
+	// normally stb_vorbis uses malloc() to allocate memory at startup,
+	// and alloca() to allocate temporary memory during a frame on the
+	// stack. (Memory consumption will depend on the amount of setup
+	// data in the file and how you set the compile flags for speed
+	// vs. size. In my test files the maximal-size usage is ~150KB.)
+	//
+	// You can modify the wrapper functions in the source (setup_malloc,
+	// setup_temp_malloc, temp_malloc) to change this behavior, or you
+	// can use a simpler allocation model: you pass in a buffer from
+	// which stb_vorbis will allocate _all_ its memory (including the
+	// temp memory). "open" may fail with a VORBIS_outofmem if you
+	// do not pass in enough data; there is no way to determine how
+	// much you do need except to succeed (at which point you can
+	// query get_info to find the exact amount required. yes I know
+	// this is lame).
+	//
+	// If you pass in a non-NULL buffer of the type below, allocation
+	// will occur from it as described above. Otherwise just pass NULL
+	// to use malloc()/alloca()
+
+	typedef struct {
+		char *alloc_buffer;
+		int   alloc_buffer_length_in_bytes;
+	} stb_vorbis_alloc;
+
+
+	///////////   FUNCTIONS USEABLE WITH ALL INPUT MODES
+
+	typedef struct stb_vorbis stb_vorbis;
+
+	typedef struct {
+		unsigned int sample_rate;
+		int channels;
+
+		unsigned int setup_memory_required;
+		unsigned int setup_temp_memory_required;
+		unsigned int temp_memory_required;
+
+		int max_frame_size;
+	} stb_vorbis_info;
+
+	// 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(FILE *f, int close_handle_on_close,
+		int *error, const stb_vorbis_alloc *alloc_buffer);
+	// create an ogg vorbis decoder from an open FILE *, looking for a stream at
+	// the _current_ seek point (ftell). on failure, returns NULL and sets *error.
+	// note that stb_vorbis must "own" this stream; if you seek it in between
+	// calls to stb_vorbis, it will become confused. 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(FILE *f, int close_handle_on_close,
+		int *error, const stb_vorbis_alloc *alloc_buffer, unsigned int len);
+	// create an ogg vorbis decoder from an open FILE *, looking for a stream at
+	// the _current_ seek point (ftell); the stream will be of length 'len' bytes.
+	// on failure, returns NULL and sets *error. note that stb_vorbis must "own"
+	// this stream; if you seek it in between calls to stb_vorbis, it will become
+	// confused.
+#endif
+
+	extern int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number);
+	extern int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number);
+	// these functions seek in the Vorbis file to (approximately) 'sample_number'.
+	// after calling seek_frame(), the next call to get_frame_*() will include
+	// the specified sample. after calling stb_vorbis_seek(), the next call to
+	// stb_vorbis_get_samples_* will start with the specified sample. If you
+	// do not need to seek to EXACTLY the target sample when using get_samples_*,
+	// you can also use seek_frame().
+
+	extern 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
+	Polycode::CoreFile *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* ptr = new int[1];
+		z->f->read(ptr, sizeof(uint8), 1);
+		int c = ptr[0];
+		//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 (z->f->read(data, n, 1) == 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 = z->f->tell();
+		z->f->seek(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 (!f->f->seek(loc, SEEK_SET))
+		return 1;
+	f->eof = 1;
+	f->f->seek(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) p->f->provider->closeFile(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) (f->f->tell() - 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(Polycode::CoreFile *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) file->tell();
+	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(Polycode::CoreFile *file, int close_on_free, int *error, const stb_vorbis_alloc *alloc) {
+	unsigned int len, start;
+	start = (unsigned int) file->tell();
+	file->seek(0, SEEK_END);
+	len = (unsigned int) (file->tell() - start);
+	file->seek(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) {
+	Polycode::CoreFile *f = Polycode::Services()->getCore()->openFile(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