text: // 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>
  #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>
  
     // find definition of alloca if it's not in stdlib.h:
     #ifdef _MSC_VER
        #include <malloc.h>
     #endif
     #if defined(__linux__) || defined(__linux) || defined(__EMSCRIPTEN__)
        #include <alloca.h>
     #endif
  #else // STB_VORBIS_NO_CRT
     #define NULL 0
     #define malloc(s)   0
     #define free(s)     ((void) 0)
     #define realloc(s)  0
  #endif // STB_VORBIS_NO_CRT
  
  #include <limits.h>
  
  #ifdef __MINGW32__
     // eff you mingw:
     //     "fixed":
     //         http://sourceforge.net/p/mingw-w64/mailman/message/32882927/
     //     "no that broke the build, reverted, who cares about C":
     //         http://sourceforge.net/p/mingw-w64/mailman/message/32890381/
     #ifdef __forceinline
     #undef __forceinline
     #endif
     #define __forceinline
  #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)                  (void)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=0., bytes_per_sample=0.;
     int probe = 0;
  
     // find the last page and validate the target sample
     stream_length = stb_vorbis_stream_length_in_samples(f);
     if (stream_length == 0)            return error(f, VORBIS_seek_without_length);
     if (sample_number > stream_length) return error(f, VORBIS_seek_invalid);
  
     // this is the maximum difference between the window-center (which is the
     // actual granule position value), and the right-start (which the spec
     // indicates should be the granule position (give or take one)).
     padding = ((f->blocksize_1 - f->blocksize_0) >> 2);
     if (sample_number < padding)
        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
  