mirrors
/
nothings-stb
mirror of https://github.com/nothings/stb.git
1
0
Fork 0
Code Issues Packages Projects Releases Wiki Activity
nothings-stb/stb_vorbis.c

5398 lines
180 KiB
C
Raw Blame History

// 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>
#if !(defined(__APPLE__) || defined(MACOSX) || defined(macintosh) || defined(Macintosh))
#include <malloc.h>
#if defined(__linux__) || defined(__linux) || defined(__EMSCRIPTEN__)
#include <alloca.h>
#endif
#endif
#else // STB_VORBIS_NO_CRT
#define NULL 0
#define malloc(s) 0
#define free(s) ((void) 0)
#define realloc(s) 0
#endif // STB_VORBIS_NO_CRT
#include <limits.h>
#ifdef __MINGW32__
// eff you mingw:
// "fixed":
// http://sourceforge.net/p/mingw-w64/mailman/message/32882927/
// "no that broke the build, reverted, who cares about C":
// http://sourceforge.net/p/mingw-w64/mailman/message/32890381/
#ifdef __forceinline
#undef __forceinline
#endif
#define __forceinline
#elif !defined(_MSC_VER)
#if __GNUC__
#define __forceinline inline
#else
#define __forceinline
#endif
#endif
#if STB_VORBIS_MAX_CHANNELS > 256
#error "Value of STB_VORBIS_MAX_CHANNELS outside of allowed range"
#endif
#if STB_VORBIS_FAST_HUFFMAN_LENGTH > 24
#error "Value of STB_VORBIS_FAST_HUFFMAN_LENGTH outside of allowed range"
#endif
#if 0
#include <crtdbg.h>
#define CHECK(f) _CrtIsValidHeapPointer(f->channel_buffers[1])
#else
#define CHECK(f) ((void) 0)
#endif
#define MAX_BLOCKSIZE_LOG 13 // from specification
#define MAX_BLOCKSIZE (1 << MAX_BLOCKSIZE_LOG)
typedef unsigned char uint8;
typedef signed char int8;
typedef unsigned short uint16;
typedef signed short int16;
typedef unsigned int uint32;
typedef signed int int32;
#ifndef TRUE
#define TRUE 1
#define FALSE 0
#endif
typedef float codetype;
// @NOTE
//
// Some arrays below are tagged "//varies", which means it's actually
// a variable-sized piece of data, but rather than malloc I assume it's
// small enough it's better to just allocate it all together with the
// main thing
//
// Most of the variables are specified with the smallest size I could pack
// them into. It might give better performance to make them all full-sized
// integers. It should be safe to freely rearrange the structures or change
// the sizes larger--nothing relies on silently truncating etc., nor the
// order of variables.
#define FAST_HUFFMAN_TABLE_SIZE (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH)
#define FAST_HUFFMAN_TABLE_MASK (FAST_HUFFMAN_TABLE_SIZE - 1)
typedef struct
{
int dimensions, entries;
uint8 *codeword_lengths;
float minimum_value;
float delta_value;
uint8 value_bits;
uint8 lookup_type;
uint8 sequence_p;
uint8 sparse;
uint32 lookup_values;
codetype *multiplicands;
uint32 *codewords;
#ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
int16 fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
#else
int32 fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
#endif
uint32 *sorted_codewords;
int *sorted_values;
int sorted_entries;
} Codebook;
typedef struct
{
uint8 order;
uint16 rate;
uint16 bark_map_size;
uint8 amplitude_bits;
uint8 amplitude_offset;
uint8 number_of_books;
uint8 book_list[16]; // varies
} Floor0;
typedef struct
{
uint8 partitions;
uint8 partition_class_list[32]; // varies
uint8 class_dimensions[16]; // varies
uint8 class_subclasses[16]; // varies
uint8 class_masterbooks[16]; // varies
int16 subclass_books[16][8]; // varies
uint16 Xlist[31*8+2]; // varies
uint8 sorted_order[31*8+2];
uint8 neighbors[31*8+2][2];
uint8 floor1_multiplier;
uint8 rangebits;
int values;
} Floor1;
typedef union
{
Floor0 floor0;
Floor1 floor1;
} Floor;
typedef struct
{
uint32 begin, end;
uint32 part_size;
uint8 classifications;
uint8 classbook;
uint8 **classdata;
int16 (*residue_books)[8];
} Residue;
typedef struct
{
uint8 magnitude;
uint8 angle;
uint8 mux;
} MappingChannel;
typedef struct
{
uint16 coupling_steps;
MappingChannel *chan;
uint8 submaps;
uint8 submap_floor[15]; // varies
uint8 submap_residue[15]; // varies
} Mapping;
typedef struct
{
uint8 blockflag;
uint8 mapping;
uint16 windowtype;
uint16 transformtype;
} Mode;
typedef struct
{
uint32 goal_crc; // expected crc if match
int bytes_left; // bytes left in packet
uint32 crc_so_far; // running crc
int bytes_done; // bytes processed in _current_ chunk
uint32 sample_loc; // granule pos encoded in page
} CRCscan;
typedef struct
{
uint32 page_start, page_end;
uint32 last_decoded_sample;
} ProbedPage;
struct stb_vorbis
{
// user-accessible info
unsigned int sample_rate;
int channels;
unsigned int setup_memory_required;
unsigned int temp_memory_required;
unsigned int setup_temp_memory_required;
// input config
#ifndef STB_VORBIS_NO_STDIO
FILE *f;
uint32 f_start;
int close_on_free;
#endif
uint8 *stream;
uint8 *stream_start;
uint8 *stream_end;
uint32 stream_len;
uint8 push_mode;
uint32 first_audio_page_offset;
ProbedPage p_first, p_last;
// memory management
stb_vorbis_alloc alloc;
int setup_offset;
int temp_offset;
// run-time results
int eof;
enum STBVorbisError error;
// user-useful data
// header info
int blocksize[2];
int blocksize_0, blocksize_1;
int codebook_count;
Codebook *codebooks;
int floor_count;
uint16 floor_types[64]; // varies
Floor *floor_config;
int residue_count;
uint16 residue_types[64]; // varies
Residue *residue_config;
int mapping_count;
Mapping *mapping;
int mode_count;
Mode mode_config[64]; // varies
uint32 total_samples;
// decode buffer
float *channel_buffers[STB_VORBIS_MAX_CHANNELS];
float *outputs [STB_VORBIS_MAX_CHANNELS];
float *previous_window[STB_VORBIS_MAX_CHANNELS];
int previous_length;
#ifndef STB_VORBIS_NO_DEFER_FLOOR
int16 *finalY[STB_VORBIS_MAX_CHANNELS];
#else
float *floor_buffers[STB_VORBIS_MAX_CHANNELS];
#endif
uint32 current_loc; // sample location of next frame to decode
int current_loc_valid;
// per-blocksize precomputed data
// twiddle factors
float *A[2],*B[2],*C[2];
float *window[2];
uint16 *bit_reverse[2];
// current page/packet/segment streaming info
uint32 serial; // stream serial number for verification
int last_page;
int segment_count;
uint8 segments[255];
uint8 page_flag;
uint8 bytes_in_seg;
uint8 first_decode;
int next_seg;
int last_seg; // flag that we're on the last segment
int last_seg_which; // what was the segment number of the last seg?
uint32 acc;
int valid_bits;
int packet_bytes;
int end_seg_with_known_loc;
uint32 known_loc_for_packet;
int discard_samples_deferred;
uint32 samples_output;
// push mode scanning
int page_crc_tests; // only in push_mode: number of tests active; -1 if not searching
#ifndef STB_VORBIS_NO_PUSHDATA_API
CRCscan scan[STB_VORBIS_PUSHDATA_CRC_COUNT];
#endif
// sample-access
int channel_buffer_start;
int channel_buffer_end;
};
#if defined(STB_VORBIS_NO_PUSHDATA_API)
#define IS_PUSH_MODE(f) FALSE
#elif defined(STB_VORBIS_NO_PULLDATA_API)
#define IS_PUSH_MODE(f) TRUE
#else
#define IS_PUSH_MODE(f) ((f)->push_mode)
#endif
typedef struct stb_vorbis vorb;
static int error(vorb *f, enum STBVorbisError e)
{
f->error = e;
if (!f->eof && e != VORBIS_need_more_data) {
f->error=e; // breakpoint for debugging
}
return 0;
}
// these functions are used for allocating temporary memory
// while decoding. if you can afford the stack space, use
// alloca(); otherwise, provide a temp buffer and it will
// allocate out of those.
#define array_size_required(count,size) (count*(sizeof(void *)+(size)))
#define temp_alloc(f,size) (f->alloc.alloc_buffer ? setup_temp_malloc(f,size) : alloca(size))
#ifdef dealloca
#define temp_free(f,p) (f->alloc.alloc_buffer ? 0 : dealloca(size))
#else
#define temp_free(f,p) 0
#endif
#define temp_alloc_save(f) ((f)->temp_offset)
#define temp_alloc_restore(f,p) ((f)->temp_offset = (p))
#define temp_block_array(f,count,size) make_block_array(temp_alloc(f,array_size_required(count,size)), count, size)
// given a sufficiently large block of memory, make an array of pointers to subblocks of it
static void *make_block_array(void *mem, int count, int size)
{
int i;
void ** p = (void **) mem;
char *q = (char *) (p + count);
for (i=0; i < count; ++i) {
p[i] = q;
q += size;
}
return p;
}
static void *setup_malloc(vorb *f, int sz)
{
sz = (sz+3) & ~3;
f->setup_memory_required += sz;
if (f->alloc.alloc_buffer) {
void *p = (char *) f->alloc.alloc_buffer + f->setup_offset;
if (f->setup_offset + sz > f->temp_offset) return NULL;
f->setup_offset += sz;
return p;
}
return sz ? malloc(sz) : NULL;
}
static void setup_free(vorb *f, void *p)
{
if (f->alloc.alloc_buffer) return; // do nothing; setup mem is a stack
free(p);
}
static void *setup_temp_malloc(vorb *f, int sz)
{
sz = (sz+3) & ~3;
if (f->alloc.alloc_buffer) {
if (f->temp_offset - sz < f->setup_offset) return NULL;
f->temp_offset -= sz;
return (char *) f->alloc.alloc_buffer + f->temp_offset;
}
return malloc(sz);
}
static void setup_temp_free(vorb *f, void *p, int sz)
{
if (f->alloc.alloc_buffer) {
f->temp_offset += (sz+3)&~3;
return;
}
free(p);
}
#define CRC32_POLY 0x04c11db7 // from spec
static uint32 crc_table[256];
static void crc32_init(void)
{
int i,j;
uint32 s;
for(i=0; i < 256; i++) {
for (s=(uint32) i << 24, j=0; j < 8; ++j)
s = (s << 1) ^ (s >= (1U<<31) ? CRC32_POLY : 0);
crc_table[i] = s;
}
}
static __forceinline uint32 crc32_update(uint32 crc, uint8 byte)
{
return (crc << 8) ^ crc_table[byte ^ (crc >> 24)];
}
// used in setup, and for huffman that doesn't go fast path
static unsigned int bit_reverse(unsigned int n)
{
n = ((n & 0xAAAAAAAA) >> 1) | ((n & 0x55555555) << 1);
n = ((n & 0xCCCCCCCC) >> 2) | ((n & 0x33333333) << 2);
n = ((n & 0xF0F0F0F0) >> 4) | ((n & 0x0F0F0F0F) << 4);
n = ((n & 0xFF00FF00) >> 8) | ((n & 0x00FF00FF) << 8);
return (n >> 16) | (n << 16);
}
static float square(float x)
{
return x*x;
}
// this is a weird definition of log2() for which log2(1) = 1, log2(2) = 2, log2(4) = 3
// as required by the specification. fast(?) implementation from stb.h
// @OPTIMIZE: called multiple times per-packet with "constants"; move to setup
static int ilog(int32 n)
{
static signed char log2_4[16] = { 0,1,2,2,3,3,3,3,4,4,4,4,4,4,4,4 };
// 2 compares if n < 16, 3 compares otherwise (4 if signed or n > 1<<29)
if (n < (1 << 14))
if (n < (1 << 4)) return 0 + log2_4[n ];
else if (n < (1 << 9)) return 5 + log2_4[n >> 5];
else return 10 + log2_4[n >> 10];
else if (n < (1 << 24))
if (n < (1 << 19)) return 15 + log2_4[n >> 15];
else return 20 + log2_4[n >> 20];
else if (n < (1 << 29)) return 25 + log2_4[n >> 25];
else if (n < (1 << 31)) return 30 + log2_4[n >> 30];
else return 0; // signed n returns 0
}
#ifndef M_PI
#define M_PI 3.14159265358979323846264f // from CRC
#endif
// code length assigned to a value with no huffman encoding
#define NO_CODE 255
/////////////////////// LEAF SETUP FUNCTIONS //////////////////////////
//
// these functions are only called at setup, and only a few times
// per file
static float float32_unpack(uint32 x)
{
// from the specification
uint32 mantissa = x & 0x1fffff;
uint32 sign = x & 0x80000000;
uint32 exp = (x & 0x7fe00000) >> 21;
double res = sign ? -(double)mantissa : (double)mantissa;
return (float) ldexp((float)res, exp-788);
}
// zlib & jpeg huffman tables assume that the output symbols
// can either be arbitrarily arranged, or have monotonically
// increasing frequencies--they rely on the lengths being sorted;
// this makes for a very simple generation algorithm.
// vorbis allows a huffman table with non-sorted lengths. This
// requires a more sophisticated construction, since symbols in
// order do not map to huffman codes "in order".
static void add_entry(Codebook *c, uint32 huff_code, int symbol, int count, int len, uint32 *values)
{
if (!c->sparse) {
c->codewords [symbol] = huff_code;
} else {
c->codewords [count] = huff_code;
c->codeword_lengths[count] = len;
values [count] = symbol;
}
}
static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values)
{
int i,k,m=0;
uint32 available[32];
memset(available, 0, sizeof(available));
// find the first entry
for (k=0; k < n; ++k) if (len[k] < NO_CODE) break;
if (k == n) { assert(c->sorted_entries == 0); return TRUE; }
// add to the list
add_entry(c, 0, k, m++, len[k], values);
// add all available leaves
for (i=1; i <= len[k]; ++i)
available[i] = 1U << (32-i);
// note that the above code treats the first case specially,
// but it's really the same as the following code, so they
// could probably be combined (except the initial code is 0,
// and I use 0 in available[] to mean 'empty')
for (i=k+1; i < n; ++i) {
uint32 res;
int z = len[i], y;
if (z == NO_CODE) continue;
// find lowest available leaf (should always be earliest,
// which is what the specification calls for)
// note that this property, and the fact we can never have
// more than one free leaf at a given level, isn't totally
// trivial to prove, but it seems true and the assert never
// fires, so!
while (z > 0 && !available[z]) --z;
if (z == 0) { return FALSE; }
res = available[z];
assert(z >= 0 && z < 32);
available[z] = 0;
add_entry(c, bit_reverse(res), i, m++, len[i], values);
// propogate availability up the tree
if (z != len[i]) {
assert(len[i] >= 0 && len[i] < 32);
for (y=len[i]; y > z; --y) {
assert(available[y] == 0);
available[y] = res + (1 << (32-y));
}
}
}
return TRUE;
}
// accelerated huffman table allows fast O(1) match of all symbols
// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH
static void compute_accelerated_huffman(Codebook *c)
{
int i, len;
for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i)
c->fast_huffman[i] = -1;
len = c->sparse ? c->sorted_entries : c->entries;
#ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
if (len > 32767) len = 32767; // largest possible value we can encode!
#endif
for (i=0; i < len; ++i) {
if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) {
uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i];
// set table entries for all bit combinations in the higher bits
while (z < FAST_HUFFMAN_TABLE_SIZE) {
c->fast_huffman[z] = i;
z += 1 << c->codeword_lengths[i];
}
}
}
}
#ifdef _MSC_VER
#define STBV_CDECL __cdecl
#else
#define STBV_CDECL
#endif
static int STBV_CDECL uint32_compare(const void *p, const void *q)
{
uint32 x = * (uint32 *) p;
uint32 y = * (uint32 *) q;
return x < y ? -1 : x > y;
}
static int include_in_sort(Codebook *c, uint8 len)
{
if (c->sparse) { assert(len != NO_CODE); return TRUE; }
if (len == NO_CODE) return FALSE;
if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE;
return FALSE;
}
// if the fast table above doesn't work, we want to binary
// search them... need to reverse the bits
static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values)
{
int i, len;
// build a list of all the entries
// OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN.
// this is kind of a frivolous optimization--I don't see any performance improvement,
// but it's like 4 extra lines of code, so.
if (!c->sparse) {
int k = 0;
for (i=0; i < c->entries; ++i)
if (include_in_sort(c, lengths[i]))
c->sorted_codewords[k++] = bit_reverse(c->codewords[i]);
assert(k == c->sorted_entries);
} else {
for (i=0; i < c->sorted_entries; ++i)
c->sorted_codewords[i] = bit_reverse(c->codewords[i]);
}
qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare);
c->sorted_codewords[c->sorted_entries] = 0xffffffff;
len = c->sparse ? c->sorted_entries : c->entries;
// now we need to indicate how they correspond; we could either
// #1: sort a different data structure that says who they correspond to
// #2: for each sorted entry, search the original list to find who corresponds
// #3: for each original entry, find the sorted entry
// #1 requires extra storage, #2 is slow, #3 can use binary search!
for (i=0; i < len; ++i) {
int huff_len = c->sparse ? lengths[values[i]] : lengths[i];
if (include_in_sort(c,huff_len)) {
uint32 code = bit_reverse(c->codewords[i]);
int x=0, n=c->sorted_entries;
while (n > 1) {
// invariant: sc[x] <= code < sc[x+n]
int m = x + (n >> 1);
if (c->sorted_codewords[m] <= code) {
x = m;
n -= (n>>1);
} else {
n >>= 1;
}
}
assert(c->sorted_codewords[x] == code);
if (c->sparse) {
c->sorted_values[x] = values[i];
c->codeword_lengths[x] = huff_len;
} else {
c->sorted_values[x] = i;
}
}
}
}
// only run while parsing the header (3 times)
static int vorbis_validate(uint8 *data)
{
static uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' };
return memcmp(data, vorbis, 6) == 0;
}
// called from setup only, once per code book
// (formula implied by specification)
static int lookup1_values(int entries, int dim)
{
int r = (int) floor(exp((float) log((float) entries) / dim));
if ((int) floor(pow((float) r+1, dim)) <= entries) // (int) cast for MinGW warning;
++r; // floor() to avoid _ftol() when non-CRT
assert(pow((float) r+1, dim) > entries);
assert((int) floor(pow((float) r, dim)) <= entries); // (int),floor() as above
return r;
}
// called twice per file
static void compute_twiddle_factors(int n, float *A, float *B, float *C)
{
int n4 = n >> 2, n8 = n >> 3;
int k,k2;
for (k=k2=0; k < n4; ++k,k2+=2) {
A[k2 ] = (float) cos(4*k*M_PI/n);
A[k2+1] = (float) -sin(4*k*M_PI/n);
B[k2 ] = (float) cos((k2+1)*M_PI/n/2) * 0.5f;
B[k2+1] = (float) sin((k2+1)*M_PI/n/2) * 0.5f;
}
for (k=k2=0; k < n8; ++k,k2+=2) {
C[k2 ] = (float) cos(2*(k2+1)*M_PI/n);
C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n);
}
}
static void compute_window(int n, float *window)
{
int n2 = n >> 1, i;
for (i=0; i < n2; ++i)
window[i] = (float) sin(0.5 * M_PI * square((float) sin((i - 0 + 0.5) / n2 * 0.5 * M_PI)));
}
static void compute_bitreverse(int n, uint16 *rev)
{
int ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
int i, n8 = n >> 3;
for (i=0; i < n8; ++i)
rev[i] = (bit_reverse(i) >> (32-ld+3)) << 2;
}
static int init_blocksize(vorb *f, int b, int n)
{
int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3;
f->A[b] = (float *) setup_malloc(f, sizeof(float) * n2);
f->B[b] = (float *) setup_malloc(f, sizeof(float) * n2);
f->C[b] = (float *) setup_malloc(f, sizeof(float) * n4);
if (!f->A[b] || !f->B[b] || !f->C[b]) return error(f, VORBIS_outofmem);
compute_twiddle_factors(n, f->A[b], f->B[b], f->C[b]);
f->window[b] = (float *) setup_malloc(f, sizeof(float) * n2);
if (!f->window[b]) return error(f, VORBIS_outofmem);
compute_window(n, f->window[b]);
f->bit_reverse[b] = (uint16 *) setup_malloc(f, sizeof(uint16) * n8);
if (!f->bit_reverse[b]) return error(f, VORBIS_outofmem);
compute_bitreverse(n, f->bit_reverse[b]);
return TRUE;
}
static void neighbors(uint16 *x, int n, int *plow, int *phigh)
{
int low = -1;
int high = 65536;
int i;
for (i=0; i < n; ++i) {
if (x[i] > low && x[i] < x[n]) { *plow = i; low = x[i]; }
if (x[i] < high && x[i] > x[n]) { *phigh = i; high = x[i]; }
}
}
// this has been repurposed so y is now the original index instead of y
typedef struct
{
uint16 x,y;
} Point;
static int STBV_CDECL point_compare(const void *p, const void *q)
{
Point *a = (Point *) p;
Point *b = (Point *) q;
return a->x < b->x ? -1 : a->x > b->x;
}
//
/////////////////////// END LEAF SETUP FUNCTIONS //////////////////////////
#if defined(STB_VORBIS_NO_STDIO)
#define USE_MEMORY(z) TRUE
#else
#define USE_MEMORY(z) ((z)->stream)
#endif
static uint8 get8(vorb *z)
{
if (USE_MEMORY(z)) {
if (z->stream >= z->stream_end) { z->eof = TRUE; return 0; }
return *z->stream++;
}
#ifndef STB_VORBIS_NO_STDIO
{
int c = fgetc(z->f);
if (c == EOF) { z->eof = TRUE; return 0; }
return c;
}
#endif
}
static uint32 get32(vorb *f)
{
uint32 x;
x = get8(f);
x += get8(f) << 8;
x += get8(f) << 16;
x += (uint32) get8(f) << 24;
return x;
}
static int getn(vorb *z, uint8 *data, int n)
{
if (USE_MEMORY(z)) {
if (z->stream+n > z->stream_end) { z->eof = 1; return 0; }
memcpy(data, z->stream, n);
z->stream += n;
return 1;
}
#ifndef STB_VORBIS_NO_STDIO
if (fread(data, n, 1, z->f) == 1)
return 1;
else {
z->eof = 1;
return 0;
}
#endif
}
static void skip(vorb *z, int n)
{
if (USE_MEMORY(z)) {
z->stream += n;
if (z->stream >= z->stream_end) z->eof = 1;
return;
}
#ifndef STB_VORBIS_NO_STDIO
{
long x = ftell(z->f);
fseek(z->f, x+n, SEEK_SET);
}
#endif
}
static int set_file_offset(stb_vorbis *f, unsigned int loc)
{
#ifndef STB_VORBIS_NO_PUSHDATA_API
if (f->push_mode) return 0;
#endif
f->eof = 0;
if (USE_MEMORY(f)) {
if (f->stream_start + loc >= f->stream_end || f->stream_start + loc < f->stream_start) {
f->stream = f->stream_end;
f->eof = 1;
return 0;
} else {
f->stream = f->stream_start + loc;
return 1;
}
}
#ifndef STB_VORBIS_NO_STDIO
if (loc + f->f_start < loc || loc >= 0x80000000) {
loc = 0x7fffffff;
f->eof = 1;
} else {
loc += f->f_start;
}
if (!fseek(f->f, loc, SEEK_SET))
return 1;
f->eof = 1;
fseek(f->f, f->f_start, SEEK_END);
return 0;
#endif
}
static uint8 ogg_page_header[4] = { 0x4f, 0x67, 0x67, 0x53 };
static int capture_pattern(vorb *f)
{
if (0x4f != get8(f)) return FALSE;
if (0x67 != get8(f)) return FALSE;
if (0x67 != get8(f)) return FALSE;
if (0x53 != get8(f)) return FALSE;
return TRUE;
}
#define PAGEFLAG_continued_packet 1
#define PAGEFLAG_first_page 2
#define PAGEFLAG_last_page 4
static int start_page_no_capturepattern(vorb *f)
{
uint32 loc0,loc1,n;
// stream structure version
if (0 != get8(f)) return error(f, VORBIS_invalid_stream_structure_version);
// header flag
f->page_flag = get8(f);
// absolute granule position
loc0 = get32(f);
loc1 = get32(f);
// @TODO: validate loc0,loc1 as valid positions?
// stream serial number -- vorbis doesn't interleave, so discard
get32(f);
//if (f->serial != get32(f)) return error(f, VORBIS_incorrect_stream_serial_number);
// page sequence number
n = get32(f);
f->last_page = n;
// CRC32
get32(f);
// page_segments
f->segment_count = get8(f);
if (!getn(f, f->segments, f->segment_count))
return error(f, VORBIS_unexpected_eof);
// assume we _don't_ know any the sample position of any segments
f->end_seg_with_known_loc = -2;
if (loc0 != ~0U || loc1 != ~0U) {
int i;
// determine which packet is the last one that will complete
for (i=f->segment_count-1; i >= 0; --i)
if (f->segments[i] < 255)
break;
// 'i' is now the index of the _last_ segment of a packet that ends
if (i >= 0) {
f->end_seg_with_known_loc = i;
f->known_loc_for_packet = loc0;
}
}
if (f->first_decode) {
int i,len;
ProbedPage p;
len = 0;
for (i=0; i < f->segment_count; ++i)
len += f->segments[i];
len += 27 + f->segment_count;
p.page_start = f->first_audio_page_offset;
p.page_end = p.page_start + len;
p.last_decoded_sample = loc0;
f->p_first = p;
}
f->next_seg = 0;
return TRUE;
}
static int start_page(vorb *f)
{
if (!capture_pattern(f)) return error(f, VORBIS_missing_capture_pattern);
return start_page_no_capturepattern(f);
}
static int start_packet(vorb *f)
{
while (f->next_seg == -1) {
if (!start_page(f)) return FALSE;
if (f->page_flag & PAGEFLAG_continued_packet)
return error(f, VORBIS_continued_packet_flag_invalid);
}
f->last_seg = FALSE;
f->valid_bits = 0;
f->packet_bytes = 0;
f->bytes_in_seg = 0;
// f->next_seg is now valid
return TRUE;
}
static int maybe_start_packet(vorb *f)
{
if (f->next_seg == -1) {
int x = get8(f);
if (f->eof) return FALSE; // EOF at page boundary is not an error!
if (0x4f != x ) return error(f, VORBIS_missing_capture_pattern);
if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
if (0x53 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
if (!start_page_no_capturepattern(f)) return FALSE;
if (f->page_flag & PAGEFLAG_continued_packet) {
// set up enough state that we can read this packet if we want,
// e.g. during recovery
f->last_seg = FALSE;
f->bytes_in_seg = 0;
return error(f, VORBIS_continued_packet_flag_invalid);
}
}
return start_packet(f);
}
static int next_segment(vorb *f)
{
int len;
if (f->last_seg) return 0;
if (f->next_seg == -1) {
f->last_seg_which = f->segment_count-1; // in case start_page fails
if (!start_page(f)) { f->last_seg = 1; return 0; }
if (!(f->page_flag & PAGEFLAG_continued_packet)) return error(f, VORBIS_continued_packet_flag_invalid);
}
len = f->segments[f->next_seg++];
if (len < 255) {
f->last_seg = TRUE;
f->last_seg_which = f->next_seg-1;
}
if (f->next_seg >= f->segment_count)
f->next_seg = -1;
assert(f->bytes_in_seg == 0);
f->bytes_in_seg = len;
return len;
}
#define EOP (-1)
#define INVALID_BITS (-1)
static int get8_packet_raw(vorb *f)
{
if (!f->bytes_in_seg) { // CLANG!
if (f->last_seg) return EOP;
else if (!next_segment(f)) return EOP;
}
assert(f->bytes_in_seg > 0);
--f->bytes_in_seg;
++f->packet_bytes;
return get8(f);
}
static int get8_packet(vorb *f)
{
int x = get8_packet_raw(f);
f->valid_bits = 0;
return x;
}
static void flush_packet(vorb *f)
{
while (get8_packet_raw(f) != EOP);
}
// @OPTIMIZE: this is the secondary bit decoder, so it's probably not as important
// as the huffman decoder?
static uint32 get_bits(vorb *f, int n)
{
uint32 z;
if (f->valid_bits < 0) return 0;
if (f->valid_bits < n) {
if (n > 24) {
// the accumulator technique below would not work correctly in this case
z = get_bits(f, 24);
z += get_bits(f, n-24) << 24;
return z;
}
if (f->valid_bits == 0) f->acc = 0;
while (f->valid_bits < n) {
int z = get8_packet_raw(f);
if (z == EOP) {
f->valid_bits = INVALID_BITS;
return 0;
}
f->acc += z << f->valid_bits;
f->valid_bits += 8;
}
}
if (f->valid_bits < 0) return 0;
z = f->acc & ((1 << n)-1);
f->acc >>= n;
f->valid_bits -= n;
return z;
}
// @OPTIMIZE: primary accumulator for huffman
// expand the buffer to as many bits as possible without reading off end of packet
// it might be nice to allow f->valid_bits and f->acc to be stored in registers,
// e.g. cache them locally and decode locally
static __forceinline void prep_huffman(vorb *f)
{
if (f->valid_bits <= 24) {
if (f->valid_bits == 0) f->acc = 0;
do {
int z;
if (f->last_seg && !f->bytes_in_seg) return;
z = get8_packet_raw(f);
if (z == EOP) return;
f->acc += (unsigned) z << f->valid_bits;
f->valid_bits += 8;
} while (f->valid_bits <= 24);
}
}
enum
{
VORBIS_packet_id = 1,
VORBIS_packet_comment = 3,
VORBIS_packet_setup = 5
};
static int codebook_decode_scalar_raw(vorb *f, Codebook *c)
{
int i;
prep_huffman(f);
if (c->codewords == NULL && c->sorted_codewords == NULL)
return -1;
// cases to use binary search: sorted_codewords && !c->codewords
// sorted_codewords && c->entries > 8
if (c->entries > 8 ? c->sorted_codewords!=NULL : !c->codewords) {
// binary search
uint32 code = bit_reverse(f->acc);
int x=0, n=c->sorted_entries, len;
while (n > 1) {
// invariant: sc[x] <= code < sc[x+n]
int m = x + (n >> 1);
if (c->sorted_codewords[m] <= code) {
x = m;
n -= (n>>1);
} else {
n >>= 1;
}
}
// x is now the sorted index
if (!c->sparse) x = c->sorted_values[x];
// x is now sorted index if sparse, or symbol otherwise
len = c->codeword_lengths[x];
if (f->valid_bits >= len) {
f->acc >>= len;
f->valid_bits -= len;
return x;
}
f->valid_bits = 0;
return -1;
}
// if small, linear search
assert(!c->sparse);
for (i=0; i < c->entries; ++i) {
if (c->codeword_lengths[i] == NO_CODE) continue;
if (c->codewords[i] == (f->acc & ((1 << c->codeword_lengths[i])-1))) {
if (f->valid_bits >= c->codeword_lengths[i]) {
f->acc >>= c->codeword_lengths[i];
f->valid_bits -= c->codeword_lengths[i];
return i;
}
f->valid_bits = 0;
return -1;
}
}
error(f, VORBIS_invalid_stream);
f->valid_bits = 0;
return -1;
}
#ifndef STB_VORBIS_NO_INLINE_DECODE
#define DECODE_RAW(var, f,c) \
if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH) \
prep_huffman(f); \
var = f->acc & FAST_HUFFMAN_TABLE_MASK; \
var = c->fast_huffman[var]; \
if (var >= 0) { \
int n = c->codeword_lengths[var]; \
f->acc >>= n; \
f->valid_bits -= n; \
if (f->valid_bits < 0) { f->valid_bits = 0; var = -1; } \
} else { \
var = codebook_decode_scalar_raw(f,c); \
}
#else
static int codebook_decode_scalar(vorb *f, Codebook *c)
{
int i;
if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH)
prep_huffman(f);
// fast huffman table lookup
i = f->acc & FAST_HUFFMAN_TABLE_MASK;
i = c->fast_huffman[i];
if (i >= 0) {
f->acc >>= c->codeword_lengths[i];
f->valid_bits -= c->codeword_lengths[i];
if (f->valid_bits < 0) { f->valid_bits = 0; return -1; }
return i;
}
return codebook_decode_scalar_raw(f,c);
}
#define DECODE_RAW(var,f,c) var = codebook_decode_scalar(f,c);
#endif
#define DECODE(var,f,c) \
DECODE_RAW(var,f,c) \
if (c->sparse) var = c->sorted_values[var];
#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
#define DECODE_VQ(var,f,c) DECODE_RAW(var,f,c)
#else
#define DECODE_VQ(var,f,c) DECODE(var,f,c)
#endif
// CODEBOOK_ELEMENT_FAST is an optimization for the CODEBOOK_FLOATS case
// where we avoid one addition
#define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off])
#define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off])
#define CODEBOOK_ELEMENT_BASE(c) (0)
static int codebook_decode_start(vorb *f, Codebook *c)
{
int z = -1;
// type 0 is only legal in a scalar context
if (c->lookup_type == 0)
error(f, VORBIS_invalid_stream);
else {
DECODE_VQ(z,f,c);
if (c->sparse) assert(z < c->sorted_entries);
if (z < 0) { // check for EOP
if (!f->bytes_in_seg)
if (f->last_seg)
return z;
error(f, VORBIS_invalid_stream);
}
}
return z;
}
static int codebook_decode(vorb *f, Codebook *c, float *output, int len)
{
int i,z = codebook_decode_start(f,c);
if (z < 0) return FALSE;
if (len > c->dimensions) len = c->dimensions;
#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
if (c->lookup_type == 1) {
float last = CODEBOOK_ELEMENT_BASE(c);
int div = 1;
for (i=0; i < len; ++i) {
int off = (z / div) % c->lookup_values;
float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
output[i] += val;
if (c->sequence_p) last = val + c->minimum_value;
div *= c->lookup_values;
}
return TRUE;
}
#endif
z *= c->dimensions;
if (c->sequence_p) {
float last = CODEBOOK_ELEMENT_BASE(c);
for (i=0; i < len; ++i) {
float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
output[i] += val;
last = val + c->minimum_value;
}
} else {
float last = CODEBOOK_ELEMENT_BASE(c);
for (i=0; i < len; ++i) {
output[i] += CODEBOOK_ELEMENT_FAST(c,z+i) + last;
}
}
return TRUE;
}
static int codebook_decode_step(vorb *f, Codebook *c, float *output, int len, int step)
{
int i,z = codebook_decode_start(f,c);
float last = CODEBOOK_ELEMENT_BASE(c);
if (z < 0) return FALSE;
if (len > c->dimensions) len = c->dimensions;
#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
if (c->lookup_type == 1) {
int div = 1;
for (i=0; i < len; ++i) {
int off = (z / div) % c->lookup_values;
float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
output[i*step] += val;
if (c->sequence_p) last = val;
div *= c->lookup_values;
}
return TRUE;
}
#endif
z *= c->dimensions;
for (i=0; i < len; ++i) {
float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
output[i*step] += val;
if (c->sequence_p) last = val;
}
return TRUE;
}
static int codebook_decode_deinterleave_repeat(vorb *f, Codebook *c, float **outputs, int ch, int *c_inter_p, int *p_inter_p, int len, int total_decode)
{
int c_inter = *c_inter_p;
int p_inter = *p_inter_p;
int i,z, effective = c->dimensions;
// type 0 is only legal in a scalar context
if (c->lookup_type == 0) return error(f, VORBIS_invalid_stream);
while (total_decode > 0) {
float last = CODEBOOK_ELEMENT_BASE(c);
DECODE_VQ(z,f,c);
#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
assert(!c->sparse || z < c->sorted_entries);
#endif
if (z < 0) {
if (!f->bytes_in_seg)
if (f->last_seg) return FALSE;
return error(f, VORBIS_invalid_stream);
}
// if this will take us off the end of the buffers, stop short!
// we check by computing the length of the virtual interleaved
// buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter),
// and the length we'll be using (effective)
if (c_inter + p_inter*ch + effective > len * ch) {
effective = len*ch - (p_inter*ch - c_inter);
}
#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
if (c->lookup_type == 1) {
int div = 1;
for (i=0; i < effective; ++i) {
int off = (z / div) % c->lookup_values;
float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
if (outputs[c_inter])
outputs[c_inter][p_inter] += val;
if (++c_inter == ch) { c_inter = 0; ++p_inter; }
if (c->sequence_p) last = val;
div *= c->lookup_values;
}
} else
#endif
{
z *= c->dimensions;
if (c->sequence_p) {
for (i=0; i < effective; ++i) {
float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
if (outputs[c_inter])
outputs[c_inter][p_inter] += val;
if (++c_inter == ch) { c_inter = 0; ++p_inter; }
last = val;
}
} else {
for (i=0; i < effective; ++i) {
float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
if (outputs[c_inter])
outputs[c_inter][p_inter] += val;
if (++c_inter == ch) { c_inter = 0; ++p_inter; }
}
}
}
total_decode -= effective;
}
*c_inter_p = c_inter;
*p_inter_p = p_inter;
return TRUE;
}
static int predict_point(int x, int x0, int x1, int y0, int y1)
{
int dy = y1 - y0;
int adx = x1 - x0;
// @OPTIMIZE: force int division to round in the right direction... is this necessary on x86?
int err = abs(dy) * (x - x0);
int off = err / adx;
return dy < 0 ? y0 - off : y0 + off;
}
// the following table is block-copied from the specification
static float inverse_db_table[256] =
{
1.0649863e-07f, 1.1341951e-07f, 1.2079015e-07f, 1.2863978e-07f,
1.3699951e-07f, 1.4590251e-07f, 1.5538408e-07f, 1.6548181e-07f,
1.7623575e-07f, 1.8768855e-07f, 1.9988561e-07f, 2.1287530e-07f,
2.2670913e-07f, 2.4144197e-07f, 2.5713223e-07f, 2.7384213e-07f,
2.9163793e-07f, 3.1059021e-07f, 3.3077411e-07f, 3.5226968e-07f,
3.7516214e-07f, 3.9954229e-07f, 4.2550680e-07f, 4.5315863e-07f,
4.8260743e-07f, 5.1396998e-07f, 5.4737065e-07f, 5.8294187e-07f,
6.2082472e-07f, 6.6116941e-07f, 7.0413592e-07f, 7.4989464e-07f,
7.9862701e-07f, 8.5052630e-07f, 9.0579828e-07f, 9.6466216e-07f,
1.0273513e-06f, 1.0941144e-06f, 1.1652161e-06f, 1.2409384e-06f,
1.3215816e-06f, 1.4074654e-06f, 1.4989305e-06f, 1.5963394e-06f,
1.7000785e-06f, 1.8105592e-06f, 1.9282195e-06f, 2.0535261e-06f,
2.1869758e-06f, 2.3290978e-06f, 2.4804557e-06f, 2.6416497e-06f,
2.8133190e-06f, 2.9961443e-06f, 3.1908506e-06f, 3.3982101e-06f,
3.6190449e-06f, 3.8542308e-06f, 4.1047004e-06f, 4.3714470e-06f,
4.6555282e-06f, 4.9580707e-06f, 5.2802740e-06f, 5.6234160e-06f,
5.9888572e-06f, 6.3780469e-06f, 6.7925283e-06f, 7.2339451e-06f,
7.7040476e-06f, 8.2047000e-06f, 8.7378876e-06f, 9.3057248e-06f,
9.9104632e-06f, 1.0554501e-05f, 1.1240392e-05f, 1.1970856e-05f,
1.2748789e-05f, 1.3577278e-05f, 1.4459606e-05f, 1.5399272e-05f,
1.6400004e-05f, 1.7465768e-05f, 1.8600792e-05f, 1.9809576e-05f,
2.1096914e-05f, 2.2467911e-05f, 2.3928002e-05f, 2.5482978e-05f,
2.7139006e-05f, 2.8902651e-05f, 3.0780908e-05f, 3.2781225e-05f,
3.4911534e-05f, 3.7180282e-05f, 3.9596466e-05f, 4.2169667e-05f,
4.4910090e-05f, 4.7828601e-05f, 5.0936773e-05f, 5.4246931e-05f,
5.7772202e-05f, 6.1526565e-05f, 6.5524908e-05f, 6.9783085e-05f,
7.4317983e-05f, 7.9147585e-05f, 8.4291040e-05f, 8.9768747e-05f,
9.5602426e-05f, 0.00010181521f, 0.00010843174f, 0.00011547824f,
0.00012298267f, 0.00013097477f, 0.00013948625f, 0.00014855085f,
0.00015820453f, 0.00016848555f, 0.00017943469f, 0.00019109536f,
0.00020351382f, 0.00021673929f, 0.00023082423f, 0.00024582449f,
0.00026179955f, 0.00027881276f, 0.00029693158f, 0.00031622787f,
0.00033677814f, 0.00035866388f, 0.00038197188f, 0.00040679456f,
0.00043323036f, 0.00046138411f, 0.00049136745f, 0.00052329927f,
0.00055730621f, 0.00059352311f, 0.00063209358f, 0.00067317058f,
0.00071691700f, 0.00076350630f, 0.00081312324f, 0.00086596457f,
0.00092223983f, 0.00098217216f, 0.0010459992f, 0.0011139742f,
0.0011863665f, 0.0012634633f, 0.0013455702f, 0.0014330129f,
0.0015261382f, 0.0016253153f, 0.0017309374f, 0.0018434235f,
0.0019632195f, 0.0020908006f, 0.0022266726f, 0.0023713743f,
0.0025254795f, 0.0026895994f, 0.0028643847f, 0.0030505286f,
0.0032487691f, 0.0034598925f, 0.0036847358f, 0.0039241906f,
0.0041792066f, 0.0044507950f, 0.0047400328f, 0.0050480668f,
0.0053761186f, 0.0057254891f, 0.0060975636f, 0.0064938176f,
0.0069158225f, 0.0073652516f, 0.0078438871f, 0.0083536271f,
0.0088964928f, 0.009474637f, 0.010090352f, 0.010746080f,
0.011444421f, 0.012188144f, 0.012980198f, 0.013823725f,
0.014722068f, 0.015678791f, 0.016697687f, 0.017782797f,
0.018938423f, 0.020169149f, 0.021479854f, 0.022875735f,
0.024362330f, 0.025945531f, 0.027631618f, 0.029427276f,
0.031339626f, 0.033376252f, 0.035545228f, 0.037855157f,
0.040315199f, 0.042935108f, 0.045725273f, 0.048696758f,
0.051861348f, 0.055231591f, 0.058820850f, 0.062643361f,
0.066714279f, 0.071049749f, 0.075666962f, 0.080584227f,
0.085821044f, 0.091398179f, 0.097337747f, 0.10366330f,
0.11039993f, 0.11757434f, 0.12521498f, 0.13335215f,
0.14201813f, 0.15124727f, 0.16107617f, 0.17154380f,
0.18269168f, 0.19456402f, 0.20720788f, 0.22067342f,
0.23501402f, 0.25028656f, 0.26655159f, 0.28387361f,
0.30232132f, 0.32196786f, 0.34289114f, 0.36517414f,
0.38890521f, 0.41417847f, 0.44109412f, 0.46975890f,
0.50028648f, 0.53279791f, 0.56742212f, 0.60429640f,
0.64356699f, 0.68538959f, 0.72993007f, 0.77736504f,
0.82788260f, 0.88168307f, 0.9389798f, 1.0f
};
// @OPTIMIZE: if you want to replace this bresenham line-drawing routine,
// note that you must produce bit-identical output to decode correctly;
// this specific sequence of operations is specified in the spec (it's
// drawing integer-quantized frequency-space lines that the encoder
// expects to be exactly the same)
// ... also, isn't the whole point of Bresenham's algorithm to NOT
// have to divide in the setup? sigh.
#ifndef STB_VORBIS_NO_DEFER_FLOOR
#define LINE_OP(a,b) a *= b
#else
#define LINE_OP(a,b) a = b
#endif
#ifdef STB_VORBIS_DIVIDE_TABLE
#define DIVTAB_NUMER 32
#define DIVTAB_DENOM 64
int8 integer_divide_table[DIVTAB_NUMER][DIVTAB_DENOM]; // 2KB
#endif
static __forceinline void draw_line(float *output, int x0, int y0, int x1, int y1, int n)
{
int dy = y1 - y0;
int adx = x1 - x0;
int ady = abs(dy);
int base;
int x=x0,y=y0;
int err = 0;
int sy;
#ifdef STB_VORBIS_DIVIDE_TABLE
if (adx < DIVTAB_DENOM && ady < DIVTAB_NUMER) {
if (dy < 0) {
base = -integer_divide_table[ady][adx];
sy = base-1;
} else {
base = integer_divide_table[ady][adx];
sy = base+1;
}
} else {
base = dy / adx;
if (dy < 0)
sy = base - 1;
else
sy = base+1;
}
#else
base = dy / adx;
if (dy < 0)
sy = base - 1;
else
sy = base+1;
#endif
ady -= abs(base) * adx;
if (x1 > n) x1 = n;
if (x < x1) {
LINE_OP(output[x], inverse_db_table[y]);
for (++x; x < x1; ++x) {
err += ady;
if (err >= adx) {
err -= adx;
y += sy;
} else
y += base;
LINE_OP(output[x], inverse_db_table[y]);
}
}
}
static int residue_decode(vorb *f, Codebook *book, float *target, int offset, int n, int rtype)
{
int k;
if (rtype == 0) {
int step = n / book->dimensions;
for (k=0; k < step; ++k)
if (!codebook_decode_step(f, book, target+offset+k, n-offset-k, step))
return FALSE;
} else {
for (k=0; k < n; ) {
if (!codebook_decode(f, book, target+offset, n-k))
return FALSE;
k += book->dimensions;
offset += book->dimensions;
}
}
return TRUE;
}
static void decode_residue(vorb *f, float *residue_buffers[], int ch, int n, int rn, uint8 *do_not_decode)
{
int i,j,pass;
Residue *r = f->residue_config + rn;
int rtype = f->residue_types[rn];
int c = r->classbook;
int classwords = f->codebooks[c].dimensions;
int n_read = r->end - r->begin;
int part_read = n_read / r->part_size;
int temp_alloc_point = temp_alloc_save(f);
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
uint8 ***part_classdata = (uint8 ***) temp_block_array(f,f->channels, part_read * sizeof(**part_classdata));
#else
int **classifications = (int **) temp_block_array(f,f->channels, part_read * sizeof(**classifications));
#endif
CHECK(f);
for (i=0; i < ch; ++i)
if (!do_not_decode[i])
memset(residue_buffers[i], 0, sizeof(float) * n);
if (rtype == 2 && ch != 1) {
for (j=0; j < ch; ++j)
if (!do_not_decode[j])
break;
if (j == ch)
goto done;
for (pass=0; pass < 8; ++pass) {
int pcount = 0, class_set = 0;
if (ch == 2) {
while (pcount < part_read) {
int z = r->begin + pcount*r->part_size;
int c_inter = (z & 1), p_inter = z>>1;
if (pass == 0) {
Codebook *c = f->codebooks+r->classbook;
int q;
DECODE(q,f,c);
if (q == EOP) goto done;
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
part_classdata[0][class_set] = r->classdata[q];
#else
for (i=classwords-1; i >= 0; --i) {
classifications[0][i+pcount] = q % r->classifications;
q /= r->classifications;
}
#endif
}
for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
int z = r->begin + pcount*r->part_size;
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
int c = part_classdata[0][class_set][i];
#else
int c = classifications[0][pcount];
#endif
int b = r->residue_books[c][pass];
if (b >= 0) {
Codebook *book = f->codebooks + b;
#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
goto done;
#else
// saves 1%
if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
goto done;
#endif
} else {
z += r->part_size;
c_inter = z & 1;
p_inter = z >> 1;
}
}
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
++class_set;
#endif
}
} else if (ch == 1) {
while (pcount < part_read) {
int z = r->begin + pcount*r->part_size;
int c_inter = 0, p_inter = z;
if (pass == 0) {
Codebook *c = f->codebooks+r->classbook;
int q;
DECODE(q,f,c);
if (q == EOP) goto done;
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
part_classdata[0][class_set] = r->classdata[q];
#else
for (i=classwords-1; i >= 0; --i) {
classifications[0][i+pcount] = q % r->classifications;
q /= r->classifications;
}
#endif
}
for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
int z = r->begin + pcount*r->part_size;
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
int c = part_classdata[0][class_set][i];
#else
int c = classifications[0][pcount];
#endif
int b = r->residue_books[c][pass];
if (b >= 0) {
Codebook *book = f->codebooks + b;
if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
goto done;
} else {
z += r->part_size;
c_inter = 0;
p_inter = z;
}
}
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
++class_set;
#endif
}
} else {
while (pcount < part_read) {
int z = r->begin + pcount*r->part_size;
int c_inter = z % ch, p_inter = z/ch;
if (pass == 0) {
Codebook *c = f->codebooks+r->classbook;
int q;
DECODE(q,f,c);
if (q == EOP) goto done;
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
part_classdata[0][class_set] = r->classdata[q];
#else
for (i=classwords-1; i >= 0; --i) {
classifications[0][i+pcount] = q % r->classifications;
q /= r->classifications;
}
#endif
}
for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
int z = r->begin + pcount*r->part_size;
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
int c = part_classdata[0][class_set][i];
#else
int c = classifications[0][pcount];
#endif
int b = r->residue_books[c][pass];
if (b >= 0) {
Codebook *book = f->codebooks + b;
if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
goto done;
} else {
z += r->part_size;
c_inter = z % ch;
p_inter = z / ch;
}
}
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
++class_set;
#endif
}
}
}
goto done;
}
CHECK(f);
for (pass=0; pass < 8; ++pass) {
int pcount = 0, class_set=0;
while (pcount < part_read) {
if (pass == 0) {
for (j=0; j < ch; ++j) {
if (!do_not_decode[j]) {
Codebook *c = f->codebooks+r->classbook;
int temp;
DECODE(temp,f,c);
if (temp == EOP) goto done;
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
part_classdata[j][class_set] = r->classdata[temp];
#else
for (i=classwords-1; i >= 0; --i) {
classifications[j][i+pcount] = temp % r->classifications;
temp /= r->classifications;
}
#endif
}
}
}
for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
for (j=0; j < ch; ++j) {
if (!do_not_decode[j]) {
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
int c = part_classdata[j][class_set][i];
#else
int c = classifications[j][pcount];
#endif
int b = r->residue_books[c][pass];
if (b >= 0) {
float *target = residue_buffers[j];
int offset = r->begin + pcount * r->part_size;
int n = r->part_size;
Codebook *book = f->codebooks + b;
if (!residue_decode(f, book, target, offset, n, rtype))
goto done;
}
}
}
}
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
++class_set;
#endif
}
}
done:
CHECK(f);
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
temp_free(f,part_classdata);
#else
temp_free(f,classifications);
#endif
temp_alloc_restore(f,temp_alloc_point);
}
#if 0
// slow way for debugging
void inverse_mdct_slow(float *buffer, int n)
{
int i,j;
int n2 = n >> 1;
float *x = (float *) malloc(sizeof(*x) * n2);
memcpy(x, buffer, sizeof(*x) * n2);
for (i=0; i < n; ++i) {
float acc = 0;
for (j=0; j < n2; ++j)
// formula from paper:
//acc += n/4.0f * x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1));
// formula from wikipedia
//acc += 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5));
// these are equivalent, except the formula from the paper inverts the multiplier!
// however, what actually works is NO MULTIPLIER!?!
//acc += 64 * 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5));
acc += x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1));
buffer[i] = acc;
}
free(x);
}
#elif 0
// same as above, but just barely able to run in real time on modern machines
void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype)
{
float mcos[16384];
int i,j;
int n2 = n >> 1, nmask = (n << 2) -1;
float *x = (float *) malloc(sizeof(*x) * n2);
memcpy(x, buffer, sizeof(*x) * n2);
for (i=0; i < 4*n; ++i)
mcos[i] = (float) cos(M_PI / 2 * i / n);
for (i=0; i < n; ++i) {
float acc = 0;
for (j=0; j < n2; ++j)
acc += x[j] * mcos[(2 * i + 1 + n2)*(2*j+1) & nmask];
buffer[i] = acc;
}
free(x);
}
#elif 0
// transform to use a slow dct-iv; this is STILL basically trivial,
// but only requires half as many ops
void dct_iv_slow(float *buffer, int n)
{
float mcos[16384];
float x[2048];
int i,j;
int n2 = n >> 1, nmask = (n << 3) - 1;
memcpy(x, buffer, sizeof(*x) * n);
for (i=0; i < 8*n; ++i)
mcos[i] = (float) cos(M_PI / 4 * i / n);
for (i=0; i < n; ++i) {
float acc = 0;
for (j=0; j < n; ++j)
acc += x[j] * mcos[((2 * i + 1)*(2*j+1)) & nmask];
buffer[i] = acc;
}
}
void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype)
{
int i, n4 = n >> 2, n2 = n >> 1, n3_4 = n - n4;
float temp[4096];
memcpy(temp, buffer, n2 * sizeof(float));
dct_iv_slow(temp, n2); // returns -c'-d, a-b'
for (i=0; i < n4 ; ++i) buffer[i] = temp[i+n4]; // a-b'
for ( ; i < n3_4; ++i) buffer[i] = -temp[n3_4 - i - 1]; // b-a', c+d'
for ( ; i < n ; ++i) buffer[i] = -temp[i - n3_4]; // c'+d
}
#endif
#ifndef LIBVORBIS_MDCT
#define LIBVORBIS_MDCT 0
#endif
#if LIBVORBIS_MDCT
// directly call the vorbis MDCT using an interface documented
// by Jeff Roberts... useful for performance comparison
typedef struct
{
int n;
int log2n;
float *trig;
int *bitrev;
float scale;
} mdct_lookup;
extern void mdct_init(mdct_lookup *lookup, int n);
extern void mdct_clear(mdct_lookup *l);
extern void mdct_backward(mdct_lookup *init, float *in, float *out);
mdct_lookup M1,M2;
void inverse_mdct(float *buffer, int n, vorb *f, int blocktype)
{
mdct_lookup *M;
if (M1.n == n) M = &M1;
else if (M2.n == n) M = &M2;
else if (M1.n == 0) { mdct_init(&M1, n); M = &M1; }
else {
if (M2.n) __asm int 3;
mdct_init(&M2, n);
M = &M2;
}
mdct_backward(M, buffer, buffer);
}
#endif
// the following were split out into separate functions while optimizing;
// they could be pushed back up but eh. __forceinline showed no change;
// they're probably already being inlined.
static void imdct_step3_iter0_loop(int n, float *e, int i_off, int k_off, float *A)
{
float *ee0 = e + i_off;
float *ee2 = ee0 + k_off;
int i;
assert((n & 3) == 0);
for (i=(n>>2); i > 0; --i) {
float k00_20, k01_21;
k00_20 = ee0[ 0] - ee2[ 0];
k01_21 = ee0[-1] - ee2[-1];
ee0[ 0] += ee2[ 0];//ee0[ 0] = ee0[ 0] + ee2[ 0];
ee0[-1] += ee2[-1];//ee0[-1] = ee0[-1] + ee2[-1];
ee2[ 0] = k00_20 * A[0] - k01_21 * A[1];
ee2[-1] = k01_21 * A[0] + k00_20 * A[1];
A += 8;
k00_20 = ee0[-2] - ee2[-2];
k01_21 = ee0[-3] - ee2[-3];
ee0[-2] += ee2[-2];//ee0[-2] = ee0[-2] + ee2[-2];
ee0[-3] += ee2[-3];//ee0[-3] = ee0[-3] + ee2[-3];
ee2[-2] = k00_20 * A[0] - k01_21 * A[1];
ee2[-3] = k01_21 * A[0] + k00_20 * A[1];
A += 8;
k00_20 = ee0[-4] - ee2[-4];
k01_21 = ee0[-5] - ee2[-5];
ee0[-4] += ee2[-4];//ee0[-4] = ee0[-4] + ee2[-4];
ee0[-5] += ee2[-5];//ee0[-5] = ee0[-5] + ee2[-5];
ee2[-4] = k00_20 * A[0] - k01_21 * A[1];
ee2[-5] = k01_21 * A[0] + k00_20 * A[1];
A += 8;
k00_20 = ee0[-6] - ee2[-6];
k01_21 = ee0[-7] - ee2[-7];
ee0[-6] += ee2[-6];//ee0[-6] = ee0[-6] + ee2[-6];
ee0[-7] += ee2[-7];//ee0[-7] = ee0[-7] + ee2[-7];
ee2[-6] = k00_20 * A[0] - k01_21 * A[1];
ee2[-7] = k01_21 * A[0] + k00_20 * A[1];
A += 8;
ee0 -= 8;
ee2 -= 8;
}
}
static void imdct_step3_inner_r_loop(int lim, float *e, int d0, int k_off, float *A, int k1)
{
int i;
float k00_20, k01_21;
float *e0 = e + d0;
float *e2 = e0 + k_off;
for (i=lim >> 2; i > 0; --i) {
k00_20 = e0[-0] - e2[-0];
k01_21 = e0[-1] - e2[-1];
e0[-0] += e2[-0];//e0[-0] = e0[-0] + e2[-0];
e0[-1] += e2[-1];//e0[-1] = e0[-1] + e2[-1];
e2[-0] = (k00_20)*A[0] - (k01_21) * A[1];
e2[-1] = (k01_21)*A[0] + (k00_20) * A[1];
A += k1;
k00_20 = e0[-2] - e2[-2];
k01_21 = e0[-3] - e2[-3];
e0[-2] += e2[-2];//e0[-2] = e0[-2] + e2[-2];
e0[-3] += e2[-3];//e0[-3] = e0[-3] + e2[-3];
e2[-2] = (k00_20)*A[0] - (k01_21) * A[1];
e2[-3] = (k01_21)*A[0] + (k00_20) * A[1];
A += k1;
k00_20 = e0[-4] - e2[-4];
k01_21 = e0[-5] - e2[-5];
e0[-4] += e2[-4];//e0[-4] = e0[-4] + e2[-4];
e0[-5] += e2[-5];//e0[-5] = e0[-5] + e2[-5];
e2[-4] = (k00_20)*A[0] - (k01_21) * A[1];
e2[-5] = (k01_21)*A[0] + (k00_20) * A[1];
A += k1;
k00_20 = e0[-6] - e2[-6];
k01_21 = e0[-7] - e2[-7];
e0[-6] += e2[-6];//e0[-6] = e0[-6] + e2[-6];
e0[-7] += e2[-7];//e0[-7] = e0[-7] + e2[-7];
e2[-6] = (k00_20)*A[0] - (k01_21) * A[1];
e2[-7] = (k01_21)*A[0] + (k00_20) * A[1];
e0 -= 8;
e2 -= 8;
A += k1;
}
}
static void imdct_step3_inner_s_loop(int n, float *e, int i_off, int k_off, float *A, int a_off, int k0)
{
int i;
float A0 = A[0];
float A1 = A[0+1];
float A2 = A[0+a_off];
float A3 = A[0+a_off+1];
float A4 = A[0+a_off*2+0];
float A5 = A[0+a_off*2+1];
float A6 = A[0+a_off*3+0];
float A7 = A[0+a_off*3+1];
float k00,k11;
float *ee0 = e +i_off;
float *ee2 = ee0+k_off;
for (i=n; i > 0; --i) {
k00 = ee0[ 0] - ee2[ 0];
k11 = ee0[-1] - ee2[-1];
ee0[ 0] = ee0[ 0] + ee2[ 0];
ee0[-1] = ee0[-1] + ee2[-1];
ee2[ 0] = (k00) * A0 - (k11) * A1;
ee2[-1] = (k11) * A0 + (k00) * A1;
k00 = ee0[-2] - ee2[-2];
k11 = ee0[-3] - ee2[-3];
ee0[-2] = ee0[-2] + ee2[-2];
ee0[-3] = ee0[-3] + ee2[-3];
ee2[-2] = (k00) * A2 - (k11) * A3;
ee2[-3] = (k11) * A2 + (k00) * A3;
k00 = ee0[-4] - ee2[-4];
k11 = ee0[-5] - ee2[-5];
ee0[-4] = ee0[-4] + ee2[-4];
ee0[-5] = ee0[-5] + ee2[-5];
ee2[-4] = (k00) * A4 - (k11) * A5;
ee2[-5] = (k11) * A4 + (k00) * A5;
k00 = ee0[-6] - ee2[-6];
k11 = ee0[-7] - ee2[-7];
ee0[-6] = ee0[-6] + ee2[-6];
ee0[-7] = ee0[-7] + ee2[-7];
ee2[-6] = (k00) * A6 - (k11) * A7;
ee2[-7] = (k11) * A6 + (k00) * A7;
ee0 -= k0;
ee2 -= k0;
}
}
static __forceinline void iter_54(float *z)
{
float k00,k11,k22,k33;
float y0,y1,y2,y3;
k00 = z[ 0] - z[-4];
y0 = z[ 0] + z[-4];
y2 = z[-2] + z[-6];
k22 = z[-2] - z[-6];
z[-0] = y0 + y2; // z0 + z4 + z2 + z6
z[-2] = y0 - y2; // z0 + z4 - z2 - z6
// done with y0,y2
k33 = z[-3] - z[-7];
z[-4] = k00 + k33; // z0 - z4 + z3 - z7
z[-6] = k00 - k33; // z0 - z4 - z3 + z7
// done with k33
k11 = z[-1] - z[-5];
y1 = z[-1] + z[-5];
y3 = z[-3] + z[-7];
z[-1] = y1 + y3; // z1 + z5 + z3 + z7
z[-3] = y1 - y3; // z1 + z5 - z3 - z7
z[-5] = k11 - k22; // z1 - z5 + z2 - z6
z[-7] = k11 + k22; // z1 - z5 - z2 + z6
}
static void imdct_step3_inner_s_loop_ld654(int n, float *e, int i_off, float *A, int base_n)
{
int a_off = base_n >> 3;
float A2 = A[0+a_off];
float *z = e + i_off;
float *base = z - 16 * n;
while (z > base) {
float k00,k11;
k00 = z[-0] - z[-8];
k11 = z[-1] - z[-9];
z[-0] = z[-0] + z[-8];
z[-1] = z[-1] + z[-9];
z[-8] = k00;
z[-9] = k11 ;
k00 = z[ -2] - z[-10];
k11 = z[ -3] - z[-11];
z[ -2] = z[ -2] + z[-10];
z[ -3] = z[ -3] + z[-11];
z[-10] = (k00+k11) * A2;
z[-11] = (k11-k00) * A2;
k00 = z[-12] - z[ -4]; // reverse to avoid a unary negation
k11 = z[ -5] - z[-13];
z[ -4] = z[ -4] + z[-12];
z[ -5] = z[ -5] + z[-13];
z[-12] = k11;
z[-13] = k00;
k00 = z[-14] - z[ -6]; // reverse to avoid a unary negation
k11 = z[ -7] - z[-15];
z[ -6] = z[ -6] + z[-14];
z[ -7] = z[ -7] + z[-15];
z[-14] = (k00+k11) * A2;
z[-15] = (k00-k11) * A2;
iter_54(z);
iter_54(z-8);
z -= 16;
}
}
static void inverse_mdct(float *buffer, int n, vorb *f, int blocktype)
{
int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l;
int ld;
// @OPTIMIZE: reduce register pressure by using fewer variables?
int save_point = temp_alloc_save(f);
float *buf2 = (float *) temp_alloc(f, n2 * sizeof(*buf2));
float *u=NULL,*v=NULL;
// twiddle factors
float *A = f->A[blocktype];
// IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio"
// See notes about bugs in that paper in less-optimal implementation 'inverse_mdct_old' after this function.
// kernel from paper
// merged:
// copy and reflect spectral data
// step 0
// note that it turns out that the items added together during
// this step are, in fact, being added to themselves (as reflected
// by step 0). inexplicable inefficiency! this became obvious
// once I combined the passes.
// so there's a missing 'times 2' here (for adding X to itself).
// this propogates through linearly to the end, where the numbers
// are 1/2 too small, and need to be compensated for.
{
float *d,*e, *AA, *e_stop;
d = &buf2[n2-2];
AA = A;
e = &buffer[0];
e_stop = &buffer[n2];
while (e != e_stop) {
d[1] = (e[0] * AA[0] - e[2]*AA[1]);
d[0] = (e[0] * AA[1] + e[2]*AA[0]);
d -= 2;
AA += 2;
e += 4;
}
e = &buffer[n2-3];
while (d >= buf2) {
d[1] = (-e[2] * AA[0] - -e[0]*AA[1]);
d[0] = (-e[2] * AA[1] + -e[0]*AA[0]);
d -= 2;
AA += 2;
e -= 4;
}
}
// now we use symbolic names for these, so that we can
// possibly swap their meaning as we change which operations
// are in place
u = buffer;
v = buf2;
// step 2 (paper output is w, now u)
// this could be in place, but the data ends up in the wrong
// place... _somebody_'s got to swap it, so this is nominated
{
float *AA = &A[n2-8];
float *d0,*d1, *e0, *e1;
e0 = &v[n4];
e1 = &v[0];
d0 = &u[n4];
d1 = &u[0];
while (AA >= A) {
float v40_20, v41_21;
v41_21 = e0[1] - e1[1];
v40_20 = e0[0] - e1[0];
d0[1] = e0[1] + e1[1];
d0[0] = e0[0] + e1[0];
d1[1] = v41_21*AA[4] - v40_20*AA[5];
d1[0] = v40_20*AA[4] + v41_21*AA[5];
v41_21 = e0[3] - e1[3];
v40_20 = e0[2] - e1[2];
d0[3] = e0[3] + e1[3];
d0[2] = e0[2] + e1[2];
d1[3] = v41_21*AA[0] - v40_20*AA[1];
d1[2] = v40_20*AA[0] + v41_21*AA[1];
AA -= 8;
d0 += 4;
d1 += 4;
e0 += 4;
e1 += 4;
}
}
// step 3
ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
// optimized step 3:
// the original step3 loop can be nested r inside s or s inside r;
// it's written originally as s inside r, but this is dumb when r
// iterates many times, and s few. So I have two copies of it and
// switch between them halfway.
// this is iteration 0 of step 3
imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*0, -(n >> 3), A);
imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*1, -(n >> 3), A);
// this is iteration 1 of step 3
imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*0, -(n >> 4), A, 16);
imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*1, -(n >> 4), A, 16);
imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*2, -(n >> 4), A, 16);
imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*3, -(n >> 4), A, 16);
l=2;
for (; l < (ld-3)>>1; ++l) {
int k0 = n >> (l+2), k0_2 = k0>>1;
int lim = 1 << (l+1);
int i;
for (i=0; i < lim; ++i)
imdct_step3_inner_r_loop(n >> (l+4), u, n2-1 - k0*i, -k0_2, A, 1 << (l+3));
}
for (; l < ld-6; ++l) {
int k0 = n >> (l+2), k1 = 1 << (l+3), k0_2 = k0>>1;
int rlim = n >> (l+6), r;
int lim = 1 << (l+1);
int i_off;
float *A0 = A;
i_off = n2-1;
for (r=rlim; r > 0; --r) {
imdct_step3_inner_s_loop(lim, u, i_off, -k0_2, A0, k1, k0);
A0 += k1*4;
i_off -= 8;
}
}
// iterations with count:
// ld-6,-5,-4 all interleaved together
// the big win comes from getting rid of needless flops
// due to the constants on pass 5 & 4 being all 1 and 0;
// combining them to be simultaneous to improve cache made little difference
imdct_step3_inner_s_loop_ld654(n >> 5, u, n2-1, A, n);
// output is u
// step 4, 5, and 6
// cannot be in-place because of step 5
{
uint16 *bitrev = f->bit_reverse[blocktype];
// weirdly, I'd have thought reading sequentially and writing
// erratically would have been better than vice-versa, but in
// fact that's not what my testing showed. (That is, with
// j = bitreverse(i), do you read i and write j, or read j and write i.)
float *d0 = &v[n4-4];
float *d1 = &v[n2-4];
while (d0 >= v) {
int k4;
k4 = bitrev[0];
d1[3] = u[k4+0];
d1[2] = u[k4+1];
d0[3] = u[k4+2];
d0[2] = u[k4+3];
k4 = bitrev[1];
d1[1] = u[k4+0];
d1[0] = u[k4+1];
d0[1] = u[k4+2];
d0[0] = u[k4+3];
d0 -= 4;
d1 -= 4;
bitrev += 2;
}
}
// (paper output is u, now v)
// data must be in buf2
assert(v == buf2);
// step 7 (paper output is v, now v)
// this is now in place
{
float *C = f->C[blocktype];
float *d, *e;
d = v;
e = v + n2 - 4;
while (d < e) {
float a02,a11,b0,b1,b2,b3;
a02 = d[0] - e[2];
a11 = d[1] + e[3];
b0 = C[1]*a02 + C[0]*a11;
b1 = C[1]*a11 - C[0]*a02;
b2 = d[0] + e[ 2];
b3 = d[1] - e[ 3];
d[0] = b2 + b0;
d[1] = b3 + b1;
e[2] = b2 - b0;
e[3] = b1 - b3;
a02 = d[2] - e[0];
a11 = d[3] + e[1];
b0 = C[3]*a02 + C[2]*a11;
b1 = C[3]*a11 - C[2]*a02;
b2 = d[2] + e[ 0];
b3 = d[3] - e[ 1];
d[2] = b2 + b0;
d[3] = b3 + b1;
e[0] = b2 - b0;
e[1] = b1 - b3;
C += 4;
d += 4;
e -= 4;
}
}
// data must be in buf2
// step 8+decode (paper output is X, now buffer)
// this generates pairs of data a la 8 and pushes them directly through
// the decode kernel (pushing rather than pulling) to avoid having
// to make another pass later
// this cannot POSSIBLY be in place, so we refer to the buffers directly
{
float *d0,*d1,*d2,*d3;
float *B = f->B[blocktype] + n2 - 8;
float *e = buf2 + n2 - 8;
d0 = &buffer[0];
d1 = &buffer[n2-4];
d2 = &buffer[n2];
d3 = &buffer[n-4];
while (e >= v) {
float p0,p1,p2,p3;
p3 = e[6]*B[7] - e[7]*B[6];
p2 = -e[6]*B[6] - e[7]*B[7];
d0[0] = p3;
d1[3] = - p3;
d2[0] = p2;
d3[3] = p2;
p1 = e[4]*B[5] - e[5]*B[4];
p0 = -e[4]*B[4] - e[5]*B[5];
d0[1] = p1;
d1[2] = - p1;
d2[1] = p0;
d3[2] = p0;
p3 = e[2]*B[3] - e[3]*B[2];
p2 = -e[2]*B[2] - e[3]*B[3];
d0[2] = p3;
d1[1] = - p3;
d2[2] = p2;
d3[1] = p2;
p1 = e[0]*B[1] - e[1]*B[0];
p0 = -e[0]*B[0] - e[1]*B[1];
d0[3] = p1;
d1[0] = - p1;
d2[3] = p0;
d3[0] = p0;
B -= 8;
e -= 8;
d0 += 4;
d2 += 4;
d1 -= 4;
d3 -= 4;
}
}
temp_free(f,buf2);
temp_alloc_restore(f,save_point);
}
#if 0
// this is the original version of the above code, if you want to optimize it from scratch
void inverse_mdct_naive(float *buffer, int n)
{
float s;
float A[1 << 12], B[1 << 12], C[1 << 11];
int i,k,k2,k4, n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l;
int n3_4 = n - n4, ld;
// how can they claim this only uses N words?!
// oh, because they're only used sparsely, whoops
float u[1 << 13], X[1 << 13], v[1 << 13], w[1 << 13];
// set up twiddle factors
for (k=k2=0; k < n4; ++k,k2+=2) {
A[k2 ] = (float) cos(4*k*M_PI/n);
A[k2+1] = (float) -sin(4*k*M_PI/n);
B[k2 ] = (float) cos((k2+1)*M_PI/n/2);
B[k2+1] = (float) sin((k2+1)*M_PI/n/2);
}
for (k=k2=0; k < n8; ++k,k2+=2) {
C[k2 ] = (float) cos(2*(k2+1)*M_PI/n);
C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n);
}
// IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio"
// Note there are bugs in that pseudocode, presumably due to them attempting
// to rename the arrays nicely rather than representing the way their actual
// implementation bounces buffers back and forth. As a result, even in the
// "some formulars corrected" version, a direct implementation fails. These
// are noted below as "paper bug".
// copy and reflect spectral data
for (k=0; k < n2; ++k) u[k] = buffer[k];
for ( ; k < n ; ++k) u[k] = -buffer[n - k - 1];
// kernel from paper
// step 1
for (k=k2=k4=0; k < n4; k+=1, k2+=2, k4+=4) {
v[n-k4-1] = (u[k4] - u[n-k4-1]) * A[k2] - (u[k4+2] - u[n-k4-3])*A[k2+1];
v[n-k4-3] = (u[k4] - u[n-k4-1]) * A[k2+1] + (u[k4+2] - u[n-k4-3])*A[k2];
}
// step 2
for (k=k4=0; k < n8; k+=1, k4+=4) {
w[n2+3+k4] = v[n2+3+k4] + v[k4+3];
w[n2+1+k4] = v[n2+1+k4] + v[k4+1];
w[k4+3] = (v[n2+3+k4] - v[k4+3])*A[n2-4-k4] - (v[n2+1+k4]-v[k4+1])*A[n2-3-k4];
w[k4+1] = (v[n2+1+k4] - v[k4+1])*A[n2-4-k4] + (v[n2+3+k4]-v[k4+3])*A[n2-3-k4];
}
// step 3
ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
for (l=0; l < ld-3; ++l) {
int k0 = n >> (l+2), k1 = 1 << (l+3);
int rlim = n >> (l+4), r4, r;
int s2lim = 1 << (l+2), s2;
for (r=r4=0; r < rlim; r4+=4,++r) {
for (s2=0; s2 < s2lim; s2+=2) {
u[n-1-k0*s2-r4] = w[n-1-k0*s2-r4] + w[n-1-k0*(s2+1)-r4];
u[n-3-k0*s2-r4] = w[n-3-k0*s2-r4] + w[n-3-k0*(s2+1)-r4];
u[n-1-k0*(s2+1)-r4] = (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1]
- (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1+1];
u[n-3-k0*(s2+1)-r4] = (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1]
+ (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1+1];
}
}
if (l+1 < ld-3) {
// paper bug: ping-ponging of u&w here is omitted
memcpy(w, u, sizeof(u));
}
}
// step 4
for (i=0; i < n8; ++i) {
int j = bit_reverse(i) >> (32-ld+3);
assert(j < n8);
if (i == j) {
// paper bug: original code probably swapped in place; if copying,
// need to directly copy in this case
int i8 = i << 3;
v[i8+1] = u[i8+1];
v[i8+3] = u[i8+3];
v[i8+5] = u[i8+5];
v[i8+7] = u[i8+7];
} else if (i < j) {
int i8 = i << 3, j8 = j << 3;
v[j8+1] = u[i8+1], v[i8+1] = u[j8 + 1];
v[j8+3] = u[i8+3], v[i8+3] = u[j8 + 3];
v[j8+5] = u[i8+5], v[i8+5] = u[j8 + 5];
v[j8+7] = u[i8+7], v[i8+7] = u[j8 + 7];
}
}
// step 5
for (k=0; k < n2; ++k) {
w[k] = v[k*2+1];
}
// step 6
for (k=k2=k4=0; k < n8; ++k, k2 += 2, k4 += 4) {
u[n-1-k2] = w[k4];
u[n-2-k2] = w[k4+1];
u[n3_4 - 1 - k2] = w[k4+2];
u[n3_4 - 2 - k2] = w[k4+3];
}
// step 7
for (k=k2=0; k < n8; ++k, k2 += 2) {
v[n2 + k2 ] = ( u[n2 + k2] + u[n-2-k2] + C[k2+1]*(u[n2+k2]-u[n-2-k2]) + C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2;
v[n-2 - k2] = ( u[n2 + k2] + u[n-2-k2] - C[k2+1]*(u[n2+k2]-u[n-2-k2]) - C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2;
v[n2+1+ k2] = ( u[n2+1+k2] - u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2;
v[n-1 - k2] = (-u[n2+1+k2] + u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2;
}
// step 8
for (k=k2=0; k < n4; ++k,k2 += 2) {
X[k] = v[k2+n2]*B[k2 ] + v[k2+1+n2]*B[k2+1];
X[n2-1-k] = v[k2+n2]*B[k2+1] - v[k2+1+n2]*B[k2 ];
}
// decode kernel to output
// determined the following value experimentally
// (by first figuring out what made inverse_mdct_slow work); then matching that here
// (probably vorbis encoder premultiplies by n or n/2, to save it on the decoder?)
s = 0.5; // theoretically would be n4
// [[[ note! the s value of 0.5 is compensated for by the B[] in the current code,
// so it needs to use the "old" B values to behave correctly, or else
// set s to 1.0 ]]]
for (i=0; i < n4 ; ++i) buffer[i] = s * X[i+n4];
for ( ; i < n3_4; ++i) buffer[i] = -s * X[n3_4 - i - 1];
for ( ; i < n ; ++i) buffer[i] = -s * X[i - n3_4];
}
#endif
static float *get_window(vorb *f, int len)
{
len <<= 1;
if (len == f->blocksize_0) return f->window[0];
if (len == f->blocksize_1) return f->window[1];
assert(0);
return NULL;
}
#ifndef STB_VORBIS_NO_DEFER_FLOOR
typedef int16 YTYPE;
#else
typedef int YTYPE;
#endif
static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *finalY, uint8 *step2_flag)
{
int n2 = n >> 1;
int s = map->chan[i].mux, floor;
floor = map->submap_floor[s];
if (f->floor_types[floor] == 0) {
return error(f, VORBIS_invalid_stream);
} else {
Floor1 *g = &f->floor_config[floor].floor1;
int j,q;
int lx = 0, ly = finalY[0] * g->floor1_multiplier;
for (q=1; q < g->values; ++q) {
j = g->sorted_order[q];
#ifndef STB_VORBIS_NO_DEFER_FLOOR
if (finalY[j] >= 0)
#else
if (step2_flag[j])
#endif
{
int hy = finalY[j] * g->floor1_multiplier;
int hx = g->Xlist[j];
if (lx != hx)
draw_line(target, lx,ly, hx,hy, n2);
CHECK(f);
lx = hx, ly = hy;
}
}
if (lx < n2) {
// optimization of: draw_line(target, lx,ly, n,ly, n2);
for (j=lx; j < n2; ++j)
LINE_OP(target[j], inverse_db_table[ly]);
CHECK(f);
}
}
return TRUE;
}
// The meaning of "left" and "right"
//
// For a given frame:
// we compute samples from 0..n
// window_center is n/2
// we'll window and mix the samples from left_start to left_end with data from the previous frame
// all of the samples from left_end to right_start can be output without mixing; however,
// this interval is 0-length except when transitioning between short and long frames
// all of the samples from right_start to right_end need to be mixed with the next frame,
// which we don't have, so those get saved in a buffer
// frame N's right_end-right_start, the number of samples to mix with the next frame,
// has to be the same as frame N+1's left_end-left_start (which they are by
// construction)
static int vorbis_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode)
{
Mode *m;
int i, n, prev, next, window_center;
f->channel_buffer_start = f->channel_buffer_end = 0;
retry:
if (f->eof) return FALSE;
if (!maybe_start_packet(f))
return FALSE;
// check packet type
if (get_bits(f,1) != 0) {
if (IS_PUSH_MODE(f))
return error(f,VORBIS_bad_packet_type);
while (EOP != get8_packet(f));
goto retry;
}
if (f->alloc.alloc_buffer)
assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
i = get_bits(f, ilog(f->mode_count-1));
if (i == EOP) return FALSE;
if (i >= f->mode_count) return FALSE;
*mode = i;
m = f->mode_config + i;
if (m->blockflag) {
n = f->blocksize_1;
prev = get_bits(f,1);
next = get_bits(f,1);
} else {
prev = next = 0;
n = f->blocksize_0;
}
// WINDOWING
window_center = n >> 1;
if (m->blockflag && !prev) {
*p_left_start = (n - f->blocksize_0) >> 2;
*p_left_end = (n + f->blocksize_0) >> 2;
} else {
*p_left_start = 0;
*p_left_end = window_center;
}
if (m->blockflag && !next) {
*p_right_start = (n*3 - f->blocksize_0) >> 2;
*p_right_end = (n*3 + f->blocksize_0) >> 2;
} else {
*p_right_start = window_center;
*p_right_end = n;
}
return TRUE;
}
static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start, int left_end, int right_start, int right_end, int *p_left)
{
Mapping *map;
int i,j,k,n,n2;
int zero_channel[256];
int really_zero_channel[256];
// WINDOWING
n = f->blocksize[m->blockflag];
map = &f->mapping[m->mapping];
// FLOORS
n2 = n >> 1;
CHECK(f);
for (i=0; i < f->channels; ++i) {
int s = map->chan[i].mux, floor;
zero_channel[i] = FALSE;
floor = map->submap_floor[s];
if (f->floor_types[floor] == 0) {
return error(f, VORBIS_invalid_stream);
} else {
Floor1 *g = &f->floor_config[floor].floor1;
if (get_bits(f, 1)) {
short *finalY;
uint8 step2_flag[256];
static int range_list[4] = { 256, 128, 86, 64 };
int range = range_list[g->floor1_multiplier-1];
int offset = 2;
finalY = f->finalY[i];
finalY[0] = get_bits(f, ilog(range)-1);
finalY[1] = get_bits(f, ilog(range)-1);
for (j=0; j < g->partitions; ++j) {
int pclass = g->partition_class_list[j];
int cdim = g->class_dimensions[pclass];
int cbits = g->class_subclasses[pclass];
int csub = (1 << cbits)-1;
int cval = 0;
if (cbits) {
Codebook *c = f->codebooks + g->class_masterbooks[pclass];
DECODE(cval,f,c);
}
for (k=0; k < cdim; ++k) {
int book = g->subclass_books[pclass][cval & csub];
cval = cval >> cbits;
if (book >= 0) {
int temp;
Codebook *c = f->codebooks + book;
DECODE(temp,f,c);
finalY[offset++] = temp;
} else
finalY[offset++] = 0;
}
}
if (f->valid_bits == INVALID_BITS) goto error; // behavior according to spec
step2_flag[0] = step2_flag[1] = 1;
for (j=2; j < g->values; ++j) {
int low, high, pred, highroom, lowroom, room, val;
low = g->neighbors[j][0];
high = g->neighbors[j][1];
//neighbors(g->Xlist, j, &low, &high);
pred = predict_point(g->Xlist[j], g->Xlist[low], g->Xlist[high], finalY[low], finalY[high]);
val = finalY[j];
highroom = range - pred;
lowroom = pred;
if (highroom < lowroom)
room = highroom * 2;
else
room = lowroom * 2;
if (val) {
step2_flag[low] = step2_flag[high] = 1;
step2_flag[j] = 1;
if (val >= room)
if (highroom > lowroom)
finalY[j] = val - lowroom + pred;
else
finalY[j] = pred - val + highroom - 1;
else
if (val & 1)
finalY[j] = pred - ((val+1)>>1);
else
finalY[j] = pred + (val>>1);
} else {
step2_flag[j] = 0;
finalY[j] = pred;
}
}
#ifdef STB_VORBIS_NO_DEFER_FLOOR
do_floor(f, map, i, n, f->floor_buffers[i], finalY, step2_flag);
#else
// defer final floor computation until _after_ residue
for (j=0; j < g->values; ++j) {
if (!step2_flag[j])
finalY[j] = -1;
}
#endif
} else {
error:
zero_channel[i] = TRUE;
}
// So we just defer everything else to later
// at this point we've decoded the floor into buffer
}
}
CHECK(f);
// at this point we've decoded all floors
if (f->alloc.alloc_buffer)
assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
// re-enable coupled channels if necessary
memcpy(really_zero_channel, zero_channel, sizeof(really_zero_channel[0]) * f->channels);
for (i=0; i < map->coupling_steps; ++i)
if (!zero_channel[map->chan[i].magnitude] || !zero_channel[map->chan[i].angle]) {
zero_channel[map->chan[i].magnitude] = zero_channel[map->chan[i].angle] = FALSE;
}
CHECK(f);
// RESIDUE DECODE
for (i=0; i < map->submaps; ++i) {
float *residue_buffers[STB_VORBIS_MAX_CHANNELS];
int r;
uint8 do_not_decode[256];
int ch = 0;
for (j=0; j < f->channels; ++j) {
if (map->chan[j].mux == i) {
if (zero_channel[j]) {
do_not_decode[ch] = TRUE;
residue_buffers[ch] = NULL;
} else {
do_not_decode[ch] = FALSE;
residue_buffers[ch] = f->channel_buffers[j];
}
++ch;
}
}
r = map->submap_residue[i];
decode_residue(f, residue_buffers, ch, n2, r, do_not_decode);
}
if (f->alloc.alloc_buffer)
assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
CHECK(f);
// INVERSE COUPLING
for (i = map->coupling_steps-1; i >= 0; --i) {
int n2 = n >> 1;
float *m = f->channel_buffers[map->chan[i].magnitude];
float *a = f->channel_buffers[map->chan[i].angle ];
for (j=0; j < n2; ++j) {
float a2,m2;
if (m[j] > 0)
if (a[j] > 0)
m2 = m[j], a2 = m[j] - a[j];
else
a2 = m[j], m2 = m[j] + a[j];
else
if (a[j] > 0)
m2 = m[j], a2 = m[j] + a[j];
else
a2 = m[j], m2 = m[j] - a[j];
m[j] = m2;
a[j] = a2;
}
}
CHECK(f);
// finish decoding the floors
#ifndef STB_VORBIS_NO_DEFER_FLOOR
for (i=0; i < f->channels; ++i) {
if (really_zero_channel[i]) {
memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2);
} else {
do_floor(f, map, i, n, f->channel_buffers[i], f->finalY[i], NULL);
}
}
#else
for (i=0; i < f->channels; ++i) {
if (really_zero_channel[i]) {
memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2);
} else {
for (j=0; j < n2; ++j)
f->channel_buffers[i][j] *= f->floor_buffers[i][j];
}
}
#endif
// INVERSE MDCT
CHECK(f);
for (i=0; i < f->channels; ++i)
inverse_mdct(f->channel_buffers[i], n, f, m->blockflag);
CHECK(f);
// this shouldn't be necessary, unless we exited on an error
// and want to flush to get to the next packet
flush_packet(f);
if (f->first_decode) {
// assume we start so first non-discarded sample is sample 0
// this isn't to spec, but spec would require us to read ahead
// and decode the size of all current frames--could be done,
// but presumably it's not a commonly used feature
f->current_loc = -n2; // start of first frame is positioned for discard
// we might have to discard samples "from" the next frame too,
// if we're lapping a large block then a small at the start?
f->discard_samples_deferred = n - right_end;
f->current_loc_valid = TRUE;
f->first_decode = FALSE;
} else if (f->discard_samples_deferred) {
if (f->discard_samples_deferred >= right_start - left_start) {
f->discard_samples_deferred -= (right_start - left_start);
left_start = right_start;
*p_left = left_start;
} else {
left_start += f->discard_samples_deferred;
*p_left = left_start;
f->discard_samples_deferred = 0;
}
} else if (f->previous_length == 0 && f->current_loc_valid) {
// we're recovering from a seek... that means we're going to discard
// the samples from this packet even though we know our position from
// the last page header, so we need to update the position based on
// the discarded samples here
// but wait, the code below is going to add this in itself even
// on a discard, so we don't need to do it here...
}
// check if we have ogg information about the sample # for this packet
if (f->last_seg_which == f->end_seg_with_known_loc) {
// if we have a valid current loc, and this is final:
if (f->current_loc_valid && (f->page_flag & PAGEFLAG_last_page)) {
uint32 current_end = f->known_loc_for_packet - (n-right_end);
// then let's infer the size of the (probably) short final frame
if (current_end < f->current_loc + (right_end-left_start)) {
if (current_end < f->current_loc) {
// negative truncation, that's impossible!
*len = 0;
} else {
*len = current_end - f->current_loc;
}
*len += left_start;
if (*len > right_end) *len = right_end; // this should never happen
f->current_loc += *len;
return TRUE;
}
}
// otherwise, just set our sample loc
// guess that the ogg granule pos refers to the _middle_ of the
// last frame?
// set f->current_loc to the position of left_start
f->current_loc = f->known_loc_for_packet - (n2-left_start);
f->current_loc_valid = TRUE;
}
if (f->current_loc_valid)
f->current_loc += (right_start - left_start);
if (f->alloc.alloc_buffer)
assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
*len = right_end; // ignore samples after the window goes to 0
CHECK(f);
return TRUE;
}
static int vorbis_decode_packet(vorb *f, int *len, int *p_left, int *p_right)
{
int mode, left_end, right_end;
if (!vorbis_decode_initial(f, p_left, &left_end, p_right, &right_end, &mode)) return 0;
return vorbis_decode_packet_rest(f, len, f->mode_config + mode, *p_left, left_end, *p_right, right_end, p_left);
}
static int vorbis_finish_frame(stb_vorbis *f, int len, int left, int right)
{
int prev,i,j;
// we use right&left (the start of the right- and left-window sin()-regions)
// to determine how much to return, rather than inferring from the rules
// (same result, clearer code); 'left' indicates where our sin() window
// starts, therefore where the previous window's right edge starts, and
// therefore where to start mixing from the previous buffer. 'right'
// indicates where our sin() ending-window starts, therefore that's where
// we start saving, and where our returned-data ends.
// mixin from previous window
if (f->previous_length) {
int i,j, n = f->previous_length;
float *w = get_window(f, n);
for (i=0; i < f->channels; ++i) {
for (j=0; j < n; ++j)
f->channel_buffers[i][left+j] =
f->channel_buffers[i][left+j]*w[ j] +
f->previous_window[i][ j]*w[n-1-j];
}
}
prev = f->previous_length;
// last half of this data becomes previous window
f->previous_length = len - right;
// @OPTIMIZE: could avoid this copy by double-buffering the
// output (flipping previous_window with channel_buffers), but
// then previous_window would have to be 2x as large, and
// channel_buffers couldn't be temp mem (although they're NOT
// currently temp mem, they could be (unless we want to level
// performance by spreading out the computation))
for (i=0; i < f->channels; ++i)
for (j=0; right+j < len; ++j)
f->previous_window[i][j] = f->channel_buffers[i][right+j];
if (!prev)
// there was no previous packet, so this data isn't valid...
// this isn't entirely true, only the would-have-overlapped data
// isn't valid, but this seems to be what the spec requires
return 0;
// truncate a short frame
if (len < right) right = len;
f->samples_output += right-left;
return right - left;
}
static void vorbis_pump_first_frame(stb_vorbis *f)
{
int len, right, left;
if (vorbis_decode_packet(f, &len, &left, &right))
vorbis_finish_frame(f, len, left, right);
}
#ifndef STB_VORBIS_NO_PUSHDATA_API
static int is_whole_packet_present(stb_vorbis *f, int end_page)
{
// make sure that we have the packet available before continuing...
// this requires a full ogg parse, but we know we can fetch from f->stream
// instead of coding this out explicitly, we could save the current read state,
// read the next packet with get8() until end-of-packet, check f->eof, then
// reset the state? but that would be slower, esp. since we'd have over 256 bytes
// of state to restore (primarily the page segment table)
int s = f->next_seg, first = TRUE;
uint8 *p = f->stream;
if (s != -1) { // if we're not starting the packet with a 'continue on next page' flag
for (; s < f->segment_count; ++s) {
p += f->segments[s];
if (f->segments[s] < 255) // stop at first short segment
break;
}
// either this continues, or it ends it...
if (end_page)
if (s < f->segment_count-1) return error(f, VORBIS_invalid_stream);
if (s == f->segment_count)
s = -1; // set 'crosses page' flag
if (p > f->stream_end) return error(f, VORBIS_need_more_data);
first = FALSE;
}
for (; s == -1;) {
uint8 *q;
int n;
// check that we have the page header ready
if (p + 26 >= f->stream_end) return error(f, VORBIS_need_more_data);
// validate the page
if (memcmp(p, ogg_page_header, 4)) return error(f, VORBIS_invalid_stream);
if (p[4] != 0) return error(f, VORBIS_invalid_stream);
if (first) { // the first segment must NOT have 'continued_packet', later ones MUST
if (f->previous_length)
if ((p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream);
// if no previous length, we're resynching, so we can come in on a continued-packet,
// which we'll just drop
} else {
if (!(p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream);
}
n = p[26]; // segment counts
q = p+27; // q points to segment table
p = q + n; // advance past header
// make sure we've read the segment table
if (p > f->stream_end) return error(f, VORBIS_need_more_data);
for (s=0; s < n; ++s) {
p += q[s];
if (q[s] < 255)
break;
}
if (end_page)
if (s < n-1) return error(f, VORBIS_invalid_stream);
if (s == n)
s = -1; // set 'crosses page' flag
if (p > f->stream_end) return error(f, VORBIS_need_more_data);
first = FALSE;
}
return TRUE;
}
#endif // !STB_VORBIS_NO_PUSHDATA_API
static int start_decoder(vorb *f)
{
uint8 header[6], x,y;
int len,i,j,k, max_submaps = 0;
int longest_floorlist=0;
// first page, first packet
if (!start_page(f)) return FALSE;
// validate page flag
if (!(f->page_flag & PAGEFLAG_first_page)) return error(f, VORBIS_invalid_first_page);
if (f->page_flag & PAGEFLAG_last_page) return error(f, VORBIS_invalid_first_page);
if (f->page_flag & PAGEFLAG_continued_packet) return error(f, VORBIS_invalid_first_page);
// check for expected packet length
if (f->segment_count != 1) return error(f, VORBIS_invalid_first_page);
if (f->segments[0] != 30) return error(f, VORBIS_invalid_first_page);
// read packet
// check packet header
if (get8(f) != VORBIS_packet_id) return error(f, VORBIS_invalid_first_page);
if (!getn(f, header, 6)) return error(f, VORBIS_unexpected_eof);
if (!vorbis_validate(header)) return error(f, VORBIS_invalid_first_page);
// vorbis_version
if (get32(f) != 0) return error(f, VORBIS_invalid_first_page);
f->channels = get8(f); if (!f->channels) return error(f, VORBIS_invalid_first_page);
if (f->channels > STB_VORBIS_MAX_CHANNELS) return error(f, VORBIS_too_many_channels);
f->sample_rate = get32(f); if (!f->sample_rate) return error(f, VORBIS_invalid_first_page);
get32(f); // bitrate_maximum
get32(f); // bitrate_nominal
get32(f); // bitrate_minimum
x = get8(f);
{
int log0,log1;
log0 = x & 15;
log1 = x >> 4;
f->blocksize_0 = 1 << log0;
f->blocksize_1 = 1 << log1;
if (log0 < 6 || log0 > 13) return error(f, VORBIS_invalid_setup);
if (log1 < 6 || log1 > 13) return error(f, VORBIS_invalid_setup);
if (log0 > log1) return error(f, VORBIS_invalid_setup);
}
// framing_flag
x = get8(f);
if (!(x & 1)) return error(f, VORBIS_invalid_first_page);
// second packet!
if (!start_page(f)) return FALSE;
if (!start_packet(f)) return FALSE;
do {
len = next_segment(f);
skip(f, len);
f->bytes_in_seg = 0;
} while (len);
// third packet!
if (!start_packet(f)) return FALSE;
#ifndef STB_VORBIS_NO_PUSHDATA_API
if (IS_PUSH_MODE(f)) {
if (!is_whole_packet_present(f, TRUE)) {
// convert error in ogg header to write type
if (f->error == VORBIS_invalid_stream)
f->error = VORBIS_invalid_setup;
return FALSE;
}
}
#endif
crc32_init(); // always init it, to avoid multithread race conditions
if (get8_packet(f) != VORBIS_packet_setup) return error(f, VORBIS_invalid_setup);
for (i=0; i < 6; ++i) header[i] = get8_packet(f);
if (!vorbis_validate(header)) return error(f, VORBIS_invalid_setup);
// codebooks
f->codebook_count = get_bits(f,8) + 1;
f->codebooks = (Codebook *) setup_malloc(f, sizeof(*f->codebooks) * f->codebook_count);
if (f->codebooks == NULL) return error(f, VORBIS_outofmem);
memset(f->codebooks, 0, sizeof(*f->codebooks) * f->codebook_count);
for (i=0; i < f->codebook_count; ++i) {
uint32 *values;
int ordered, sorted_count;
int total=0;
uint8 *lengths;
Codebook *c = f->codebooks+i;
CHECK(f);
x = get_bits(f, 8); if (x != 0x42) return error(f, VORBIS_invalid_setup);
x = get_bits(f, 8); if (x != 0x43) return error(f, VORBIS_invalid_setup);
x = get_bits(f, 8); if (x != 0x56) return error(f, VORBIS_invalid_setup);
x = get_bits(f, 8);
c->dimensions = (get_bits(f, 8)<<8) + x;
x = get_bits(f, 8);
y = get_bits(f, 8);
c->entries = (get_bits(f, 8)<<16) + (y<<8) + x;
ordered = get_bits(f,1);
c->sparse = ordered ? 0 : get_bits(f,1);
if (c->dimensions == 0 && c->entries != 0) return error(f, VORBIS_invalid_setup);
if (c->sparse)
lengths = (uint8 *) setup_temp_malloc(f, c->entries);
else
lengths = c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries);
if (!lengths) return error(f, VORBIS_outofmem);
if (ordered) {
int current_entry = 0;
int current_length = get_bits(f,5) + 1;
while (current_entry < c->entries) {
int limit = c->entries - current_entry;
int n = get_bits(f, ilog(limit));
if (current_entry + n > (int) c->entries) { return error(f, VORBIS_invalid_setup); }
memset(lengths + current_entry, current_length, n);
current_entry += n;
++current_length;
}
} else {
for (j=0; j < c->entries; ++j) {
int present = c->sparse ? get_bits(f,1) : 1;
if (present) {
lengths[j] = get_bits(f, 5) + 1;
++total;
if (lengths[j] == 32)
return error(f, VORBIS_invalid_setup);
} else {
lengths[j] = NO_CODE;
}
}
}
if (c->sparse && total >= c->entries >> 2) {
// convert sparse items to non-sparse!
if (c->entries > (int) f->setup_temp_memory_required)
f->setup_temp_memory_required = c->entries;
c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries);
if (c->codeword_lengths == NULL) return error(f, VORBIS_outofmem);
memcpy(c->codeword_lengths, lengths, c->entries);
setup_temp_free(f, lengths, c->entries); // note this is only safe if there have been no intervening temp mallocs!
lengths = c->codeword_lengths;
c->sparse = 0;
}
// compute the size of the sorted tables
if (c->sparse) {
sorted_count = total;
} else {
sorted_count = 0;
#ifndef STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
for (j=0; j < c->entries; ++j)
if (lengths[j] > STB_VORBIS_FAST_HUFFMAN_LENGTH && lengths[j] != NO_CODE)
++sorted_count;
#endif
}
c->sorted_entries = sorted_count;
values = NULL;
CHECK(f);
if (!c->sparse) {
c->codewords = (uint32 *) setup_malloc(f, sizeof(c->codewords[0]) * c->entries);
if (!c->codewords) return error(f, VORBIS_outofmem);
} else {
unsigned int size;
if (c->sorted_entries) {
c->codeword_lengths = (uint8 *) setup_malloc(f, c->sorted_entries);
if (!c->codeword_lengths) return error(f, VORBIS_outofmem);
c->codewords = (uint32 *) setup_temp_malloc(f, sizeof(*c->codewords) * c->sorted_entries);
if (!c->codewords) return error(f, VORBIS_outofmem);
values = (uint32 *) setup_temp_malloc(f, sizeof(*values) * c->sorted_entries);
if (!values) return error(f, VORBIS_outofmem);
}
size = c->entries + (sizeof(*c->codewords) + sizeof(*values)) * c->sorted_entries;
if (size > f->setup_temp_memory_required)
f->setup_temp_memory_required = size;
}
if (!compute_codewords(c, lengths, c->entries, values)) {
if (c->sparse) setup_temp_free(f, values, 0);
return error(f, VORBIS_invalid_setup);
}
if (c->sorted_entries) {
// allocate an extra slot for sentinels
c->sorted_codewords = (uint32 *) setup_malloc(f, sizeof(*c->sorted_codewords) * (c->sorted_entries+1));
if (c->sorted_codewords == NULL) return error(f, VORBIS_outofmem);
// allocate an extra slot at the front so that c->sorted_values[-1] is defined
// so that we can catch that case without an extra if
c->sorted_values = ( int *) setup_malloc(f, sizeof(*c->sorted_values ) * (c->sorted_entries+1));
if (c->sorted_values == NULL) return error(f, VORBIS_outofmem);
++c->sorted_values;
c->sorted_values[-1] = -1;
compute_sorted_huffman(c, lengths, values);
}
if (c->sparse) {
setup_temp_free(f, values, sizeof(*values)*c->sorted_entries);
setup_temp_free(f, c->codewords, sizeof(*c->codewords)*c->sorted_entries);
setup_temp_free(f, lengths, c->entries);
c->codewords = NULL;
}
compute_accelerated_huffman(c);
CHECK(f);
c->lookup_type = get_bits(f, 4);
if (c->lookup_type > 2) return error(f, VORBIS_invalid_setup);
if (c->lookup_type > 0) {
uint16 *mults;
c->minimum_value = float32_unpack(get_bits(f, 32));
c->delta_value = float32_unpack(get_bits(f, 32));
c->value_bits = get_bits(f, 4)+1;
c->sequence_p = get_bits(f,1);
if (c->lookup_type == 1) {
c->lookup_values = lookup1_values(c->entries, c->dimensions);
} else {
c->lookup_values = c->entries * c->dimensions;
}
if (c->lookup_values == 0) return error(f, VORBIS_invalid_setup);
mults = (uint16 *) setup_temp_malloc(f, sizeof(mults[0]) * c->lookup_values);
if (mults == NULL) return error(f, VORBIS_outofmem);
for (j=0; j < (int) c->lookup_values; ++j) {
int q = get_bits(f, c->value_bits);
if (q == EOP) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_invalid_setup); }
mults[j] = q;
}
#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
if (c->lookup_type == 1) {
int len, sparse = c->sparse;
float last=0;
// pre-expand the lookup1-style multiplicands, to avoid a divide in the inner loop
if (sparse) {
if (c->sorted_entries == 0) goto skip;
c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->sorted_entries * c->dimensions);
} else
c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->entries * c->dimensions);
if (c->multiplicands == NULL) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); }
len = sparse ? c->sorted_entries : c->entries;
for (j=0; j < len; ++j) {
unsigned int z = sparse ? c->sorted_values[j] : j;
unsigned int div=1;
for (k=0; k < c->dimensions; ++k) {
int off = (z / div) % c->lookup_values;
float val = mults[off];
val = mults[off]*c->delta_value + c->minimum_value + last;
c->multiplicands[j*c->dimensions + k] = val;
if (c->sequence_p)
last = val;
if (k+1 < c->dimensions) {
if (div > UINT_MAX / (unsigned int) c->lookup_values) {
setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values);
return error(f, VORBIS_invalid_setup);
}
div *= c->lookup_values;
}
}
}
c->lookup_type = 2;
}
else
#endif
{
float last=0;
CHECK(f);
c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values);
if (c->multiplicands == NULL) { setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); }
for (j=0; j < (int) c->lookup_values; ++j) {
float val = mults[j] * c->delta_value + c->minimum_value + last;
c->multiplicands[j] = val;
if (c->sequence_p)
last = val;
}
}
#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
skip:;
#endif
setup_temp_free(f, mults, sizeof(mults[0])*c->lookup_values);
CHECK(f);
}
CHECK(f);
}
// time domain transfers (notused)
x = get_bits(f, 6) + 1;
for (i=0; i < x; ++i) {
uint32 z = get_bits(f, 16);
if (z != 0) return error(f, VORBIS_invalid_setup);
}
// Floors
f->floor_count = get_bits(f, 6)+1;
f->floor_config = (Floor *) setup_malloc(f, f->floor_count * sizeof(*f->floor_config));
if (f->floor_config == NULL) return error(f, VORBIS_outofmem);
for (i=0; i < f->floor_count; ++i) {
f->floor_types[i] = get_bits(f, 16);
if (f->floor_types[i] > 1) return error(f, VORBIS_invalid_setup);
if (f->floor_types[i] == 0) {
Floor0 *g = &f->floor_config[i].floor0;
g->order = get_bits(f,8);
g->rate = get_bits(f,16);
g->bark_map_size = get_bits(f,16);
g->amplitude_bits = get_bits(f,6);
g->amplitude_offset = get_bits(f,8);
g->number_of_books = get_bits(f,4) + 1;
for (j=0; j < g->number_of_books; ++j)
g->book_list[j] = get_bits(f,8);
return error(f, VORBIS_feature_not_supported);
} else {
Point p[31*8+2];
Floor1 *g = &f->floor_config[i].floor1;
int max_class = -1;
g->partitions = get_bits(f, 5);
for (j=0; j < g->partitions; ++j) {
g->partition_class_list[j] = get_bits(f, 4);
if (g->partition_class_list[j] > max_class)
max_class = g->partition_class_list[j];
}
for (j=0; j <= max_class; ++j) {
g->class_dimensions[j] = get_bits(f, 3)+1;
g->class_subclasses[j] = get_bits(f, 2);
if (g->class_subclasses[j]) {
g->class_masterbooks[j] = get_bits(f, 8);
if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
}
for (k=0; k < 1 << g->class_subclasses[j]; ++k) {
g->subclass_books[j][k] = get_bits(f,8)-1;
if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
}
}
g->floor1_multiplier = get_bits(f,2)+1;
g->rangebits = get_bits(f,4);
g->Xlist[0] = 0;
g->Xlist[1] = 1 << g->rangebits;
g->values = 2;
for (j=0; j < g->partitions; ++j) {
int c = g->partition_class_list[j];
for (k=0; k < g->class_dimensions[c]; ++k) {
g->Xlist[g->values] = get_bits(f, g->rangebits);
++g->values;
}
}
// precompute the sorting
for (j=0; j < g->values; ++j) {
p[j].x = g->Xlist[j];
p[j].y = j;
}
qsort(p, g->values, sizeof(p[0]), point_compare);
for (j=0; j < g->values; ++j)
g->sorted_order[j] = (uint8) p[j].y;
// precompute the neighbors
for (j=2; j < g->values; ++j) {
int low,hi;
neighbors(g->Xlist, j, &low,&hi);
g->neighbors[j][0] = low;
g->neighbors[j][1] = hi;
}
if (g->values > longest_floorlist)
longest_floorlist = g->values;
}
}
// Residue
f->residue_count = get_bits(f, 6)+1;
f->residue_config = (Residue *) setup_malloc(f, f->residue_count * sizeof(f->residue_config[0]));
if (f->residue_config == NULL) return error(f, VORBIS_outofmem);
memset(f->residue_config, 0, f->residue_count * sizeof(f->residue_config[0]));
for (i=0; i < f->residue_count; ++i) {
uint8 residue_cascade[64];
Residue *r = f->residue_config+i;
f->residue_types[i] = get_bits(f, 16);
if (f->residue_types[i] > 2) return error(f, VORBIS_invalid_setup);
r->begin = get_bits(f, 24);
r->end = get_bits(f, 24);
if (r->end < r->begin) return error(f, VORBIS_invalid_setup);
r->part_size = get_bits(f,24)+1;
r->classifications = get_bits(f,6)+1;
r->classbook = get_bits(f,8);
if (r->classbook >= f->codebook_count) return error(f, VORBIS_invalid_setup);
for (j=0; j < r->classifications; ++j) {
uint8 high_bits=0;
uint8 low_bits=get_bits(f,3);
if (get_bits(f,1))
high_bits = get_bits(f,5);
residue_cascade[j] = high_bits*8 + low_bits;
}
r->residue_books = (short (*)[8]) setup_malloc(f, sizeof(r->residue_books[0]) * r->classifications);
if (r->residue_books == NULL) return error(f, VORBIS_outofmem);
for (j=0; j < r->classifications; ++j) {
for (k=0; k < 8; ++k) {
if (residue_cascade[j] & (1 << k)) {
r->residue_books[j][k] = get_bits(f, 8);
if (r->residue_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
} else {
r->residue_books[j][k] = -1;
}
}
}
// precompute the classifications[] array to avoid inner-loop mod/divide
// call it 'classdata' since we already have r->classifications
r->classdata = (uint8 **) setup_malloc(f, sizeof(*r->classdata) * f->codebooks[r->classbook].entries);
if (!r->classdata) return error(f, VORBIS_outofmem);
memset(r->classdata, 0, sizeof(*r->classdata) * f->codebooks[r->classbook].entries);
for (j=0; j < f->codebooks[r->classbook].entries; ++j) {
int classwords = f->codebooks[r->classbook].dimensions;
int temp = j;
r->classdata[j] = (uint8 *) setup_malloc(f, sizeof(r->classdata[j][0]) * classwords);
if (r->classdata[j] == NULL) return error(f, VORBIS_outofmem);
for (k=classwords-1; k >= 0; --k) {
r->classdata[j][k] = temp % r->classifications;
temp /= r->classifications;
}
}
}
f->mapping_count = get_bits(f,6)+1;
f->mapping = (Mapping *) setup_malloc(f, f->mapping_count * sizeof(*f->mapping));
if (f->mapping == NULL) return error(f, VORBIS_outofmem);
memset(f->mapping, 0, f->mapping_count * sizeof(*f->mapping));
for (i=0; i < f->mapping_count; ++i) {
Mapping *m = f->mapping + i;
int mapping_type = get_bits(f,16);
if (mapping_type != 0) return error(f, VORBIS_invalid_setup);
m->chan = (MappingChannel *) setup_malloc(f, f->channels * sizeof(*m->chan));
if (m->chan == NULL) return error(f, VORBIS_outofmem);
if (get_bits(f,1))
m->submaps = get_bits(f,4)+1;
else
m->submaps = 1;
if (m->submaps > max_submaps)
max_submaps = m->submaps;
if (get_bits(f,1)) {
m->coupling_steps = get_bits(f,8)+1;
for (k=0; k < m->coupling_steps; ++k) {
m->chan[k].magnitude = get_bits(f, ilog(f->channels-1));
m->chan[k].angle = get_bits(f, ilog(f->channels-1));
if (m->chan[k].magnitude >= f->channels) return error(f, VORBIS_invalid_setup);
if (m->chan[k].angle >= f->channels) return error(f, VORBIS_invalid_setup);
if (m->chan[k].magnitude == m->chan[k].angle) return error(f, VORBIS_invalid_setup);
}
} else
m->coupling_steps = 0;
// reserved field
if (get_bits(f,2)) return error(f, VORBIS_invalid_setup);
if (m->submaps > 1) {
for (j=0; j < f->channels; ++j) {
m->chan[j].mux = get_bits(f, 4);
if (m->chan[j].mux >= m->submaps) return error(f, VORBIS_invalid_setup);
}
} else
// @SPECIFICATION: this case is missing from the spec
for (j=0; j < f->channels; ++j)
m->chan[j].mux = 0;
for (j=0; j < m->submaps; ++j) {
get_bits(f,8); // discard
m->submap_floor[j] = get_bits(f,8);
m->submap_residue[j] = get_bits(f,8);
if (m->submap_floor[j] >= f->floor_count) return error(f, VORBIS_invalid_setup);
if (m->submap_residue[j] >= f->residue_count) return error(f, VORBIS_invalid_setup);
}
}
// Modes
f->mode_count = get_bits(f, 6)+1;
for (i=0; i < f->mode_count; ++i) {
Mode *m = f->mode_config+i;
m->blockflag = get_bits(f,1);
m->windowtype = get_bits(f,16);
m->transformtype = get_bits(f,16);
m->mapping = get_bits(f,8);
if (m->windowtype != 0) return error(f, VORBIS_invalid_setup);
if (m->transformtype != 0) return error(f, VORBIS_invalid_setup);
if (m->mapping >= f->mapping_count) return error(f, VORBIS_invalid_setup);
}
flush_packet(f);
f->previous_length = 0;
for (i=0; i < f->channels; ++i) {
f->channel_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1);
f->previous_window[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2);
f->finalY[i] = (int16 *) setup_malloc(f, sizeof(int16) * longest_floorlist);
if (f->channel_buffers[i] == NULL || f->previous_window[i] == NULL || f->finalY[i] == NULL) return error(f, VORBIS_outofmem);
#ifdef STB_VORBIS_NO_DEFER_FLOOR
f->floor_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2);
if (f->floor_buffers[i] == NULL) return error(f, VORBIS_outofmem);
#endif
}
if (!init_blocksize(f, 0, f->blocksize_0)) return FALSE;
if (!init_blocksize(f, 1, f->blocksize_1)) return FALSE;
f->blocksize[0] = f->blocksize_0;
f->blocksize[1] = f->blocksize_1;
#ifdef STB_VORBIS_DIVIDE_TABLE
if (integer_divide_table[1][1]==0)
for (i=0; i < DIVTAB_NUMER; ++i)
for (j=1; j < DIVTAB_DENOM; ++j)
integer_divide_table[i][j] = i / j;
#endif
// compute how much temporary memory is needed
// 1.
{
uint32 imdct_mem = (f->blocksize_1 * sizeof(float) >> 1);
uint32 classify_mem;
int i,max_part_read=0;
for (i=0; i < f->residue_count; ++i) {
Residue *r = f->residue_config + i;
int n_read = r->end - r->begin;
int part_read = n_read / r->part_size;
if (part_read > max_part_read)
max_part_read = part_read;
}
#ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(uint8 *));
#else
classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(int *));
#endif
f->temp_memory_required = classify_mem;
if (imdct_mem > f->temp_memory_required)
f->temp_memory_required = imdct_mem;
}
f->first_decode = TRUE;
if (f->alloc.alloc_buffer) {
assert(f->temp_offset == f->alloc.alloc_buffer_length_in_bytes);
// check if there's enough temp memory so we don't error later
if (f->setup_offset + sizeof(*f) + f->temp_memory_required > (unsigned) f->temp_offset)
return error(f, VORBIS_outofmem);
}
f->first_audio_page_offset = stb_vorbis_get_file_offset(f);
return TRUE;
}
static void vorbis_deinit(stb_vorbis *p)
{
int i,j;
if (p->residue_config) {
for (i=0; i < p->residue_count; ++i) {
Residue *r = p->residue_config+i;
if (r->classdata) {
for (j=0; j < p->codebooks[r->classbook].entries; ++j)
setup_free(p, r->classdata[j]);
setup_free(p, r->classdata);
}
setup_free(p, r->residue_books);
}
}
if (p->codebooks) {
CHECK(p);
for (i=0; i < p->codebook_count; ++i) {
Codebook *c = p->codebooks + i;
setup_free(p, c->codeword_lengths);
setup_free(p, c->multiplicands);
setup_free(p, c->codewords);
setup_free(p, c->sorted_codewords);
// c->sorted_values[-1] is the first entry in the array
setup_free(p, c->sorted_values ? c->sorted_values-1 : NULL);
}
setup_free(p, p->codebooks);
}
setup_free(p, p->floor_config);
setup_free(p, p->residue_config);
if (p->mapping) {
for (i=0; i < p->mapping_count; ++i)
setup_free(p, p->mapping[i].chan);
setup_free(p, p->mapping);
}
CHECK(p);
for (i=0; i < p->channels && i < STB_VORBIS_MAX_CHANNELS; ++i) {
setup_free(p, p->channel_buffers[i]);
setup_free(p, p->previous_window[i]);
#ifdef STB_VORBIS_NO_DEFER_FLOOR
setup_free(p, p->floor_buffers[i]);
#endif
setup_free(p, p->finalY[i]);
}
for (i=0; i < 2; ++i) {
setup_free(p, p->A[i]);
setup_free(p, p->B[i]);
setup_free(p, p->C[i]);
setup_free(p, p->window[i]);
setup_free(p, p->bit_reverse[i]);
}
#ifndef STB_VORBIS_NO_STDIO
if (p->close_on_free) fclose(p->f);
#endif
}
void stb_vorbis_close(stb_vorbis *p)
{
if (p == NULL) return;
vorbis_deinit(p);
setup_free(p,p);
}
static void vorbis_init(stb_vorbis *p, const stb_vorbis_alloc *z)
{
memset(p, 0, sizeof(*p)); // NULL out all malloc'd pointers to start
if (z) {
p->alloc = *z;
p->alloc.alloc_buffer_length_in_bytes = (p->alloc.alloc_buffer_length_in_bytes+3) & ~3;
p->temp_offset = p->alloc.alloc_buffer_length_in_bytes;
}
p->eof = 0;
p->error = VORBIS__no_error;
p->stream = NULL;
p->codebooks = NULL;
p->page_crc_tests = -1;
#ifndef STB_VORBIS_NO_STDIO
p->close_on_free = FALSE;
p->f = NULL;
#endif
}
int stb_vorbis_get_sample_offset(stb_vorbis *f)
{
if (f->current_loc_valid)
return f->current_loc;
else
return -1;
}
stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f)
{
stb_vorbis_info d;
d.channels = f->channels;
d.sample_rate = f->sample_rate;
d.setup_memory_required = f->setup_memory_required;
d.setup_temp_memory_required = f->setup_temp_memory_required;
d.temp_memory_required = f->temp_memory_required;
d.max_frame_size = f->blocksize_1 >> 1;
return d;
}
int stb_vorbis_get_error(stb_vorbis *f)
{
int e = f->error;
f->error = VORBIS__no_error;
return e;
}
static stb_vorbis * vorbis_alloc(stb_vorbis *f)
{
stb_vorbis *p = (stb_vorbis *) setup_malloc(f, sizeof(*p));
return p;
}
#ifndef STB_VORBIS_NO_PUSHDATA_API
void stb_vorbis_flush_pushdata(stb_vorbis *f)
{
f->previous_length = 0;
f->page_crc_tests = 0;
f->discard_samples_deferred = 0;
f->current_loc_valid = FALSE;
f->first_decode = FALSE;
f->samples_output = 0;
f->channel_buffer_start = 0;
f->channel_buffer_end = 0;
}
static int vorbis_search_for_page_pushdata(vorb *f, uint8 *data, int data_len)
{
int i,n;
for (i=0; i < f->page_crc_tests; ++i)
f->scan[i].bytes_done = 0;
// if we have room for more scans, search for them first, because
// they may cause us to stop early if their header is incomplete
if (f->page_crc_tests < STB_VORBIS_PUSHDATA_CRC_COUNT) {
if (data_len < 4) return 0;
data_len -= 3; // need to look for 4-byte sequence, so don't miss
// one that straddles a boundary
for (i=0; i < data_len; ++i) {
if (data[i] == 0x4f) {
if (0==memcmp(data+i, ogg_page_header, 4)) {
int j,len;
uint32 crc;
// make sure we have the whole page header
if (i+26 >= data_len || i+27+data[i+26] >= data_len) {
// only read up to this page start, so hopefully we'll
// have the whole page header start next time
data_len = i;
break;
}
// ok, we have it all; compute the length of the page
len = 27 + data[i+26];
for (j=0; j < data[i+26]; ++j)
len += data[i+27+j];
// scan everything up to the embedded crc (which we must 0)
crc = 0;
for (j=0; j < 22; ++j)
crc = crc32_update(crc, data[i+j]);
// now process 4 0-bytes
for ( ; j < 26; ++j)
crc = crc32_update(crc, 0);
// len is the total number of bytes we need to scan
n = f->page_crc_tests++;
f->scan[n].bytes_left = len-j;
f->scan[n].crc_so_far = crc;
f->scan[n].goal_crc = data[i+22] + (data[i+23] << 8) + (data[i+24]<<16) + (data[i+25]<<24);
// if the last frame on a page is continued to the next, then
// we can't recover the sample_loc immediately
if (data[i+27+data[i+26]-1] == 255)
f->scan[n].sample_loc = ~0;
else
f->scan[n].sample_loc = data[i+6] + (data[i+7] << 8) + (data[i+ 8]<<16) + (data[i+ 9]<<24);
f->scan[n].bytes_done = i+j;
if (f->page_crc_tests == STB_VORBIS_PUSHDATA_CRC_COUNT)
break;
// keep going if we still have room for more
}
}
}
}
for (i=0; i < f->page_crc_tests;) {
uint32 crc;
int j;
int n = f->scan[i].bytes_done;
int m = f->scan[i].bytes_left;
if (m > data_len - n) m = data_len - n;
// m is the bytes to scan in the current chunk
crc = f->scan[i].crc_so_far;
for (j=0; j < m; ++j)
crc = crc32_update(crc, data[n+j]);
f->scan[i].bytes_left -= m;
f->scan[i].crc_so_far = crc;
if (f->scan[i].bytes_left == 0) {
// does it match?
if (f->scan[i].crc_so_far == f->scan[i].goal_crc) {
// Houston, we have page
data_len = n+m; // consumption amount is wherever that scan ended
f->page_crc_tests = -1; // drop out of page scan mode
f->previous_length = 0; // decode-but-don't-output one frame
f->next_seg = -1; // start a new page
f->current_loc = f->scan[i].sample_loc; // set the current sample location
// to the amount we'd have decoded had we decoded this page
f->current_loc_valid = f->current_loc != ~0U;
return data_len;
}
// delete entry
f->scan[i] = f->scan[--f->page_crc_tests];
} else {
++i;
}
}
return data_len;
}
// return value: number of bytes we used
int stb_vorbis_decode_frame_pushdata(
stb_vorbis *f, // the file we're decoding
const uint8 *data, int data_len, // the memory available for decoding
int *channels, // place to write number of float * buffers
float ***output, // place to write float ** array of float * buffers
int *samples // place to write number of output samples
)
{
int i;
int len,right,left;
if (!IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
if (f->page_crc_tests >= 0) {
*samples = 0;
return vorbis_search_for_page_pushdata(f, (uint8 *) data, data_len);
}
f->stream = (uint8 *) data;
f->stream_end = (uint8 *) data + data_len;
f->error = VORBIS__no_error;
// check that we have the entire packet in memory
if (!is_whole_packet_present(f, FALSE)) {
*samples = 0;
return 0;
}
if (!vorbis_decode_packet(f, &len, &left, &right)) {
// save the actual error we encountered
enum STBVorbisError error = f->error;
if (error == VORBIS_bad_packet_type) {
// flush and resynch
f->error = VORBIS__no_error;
while (get8_packet(f) != EOP)
if (f->eof) break;
*samples = 0;
return (int) (f->stream - data);
}
if (error == VORBIS_continued_packet_flag_invalid) {
if (f->previous_length == 0) {
// we may be resynching, in which case it's ok to hit one
// of these; just discard the packet
f->error = VORBIS__no_error;
while (get8_packet(f) != EOP)
if (f->eof) break;
*samples = 0;
return (int) (f->stream - data);
}
}
// if we get an error while parsing, what to do?
// well, it DEFINITELY won't work to continue from where we are!
stb_vorbis_flush_pushdata(f);
// restore the error that actually made us bail
f->error = error;
*samples = 0;
return 1;
}
// success!
len = vorbis_finish_frame(f, len, left, right);
for (i=0; i < f->channels; ++i)
f->outputs[i] = f->channel_buffers[i] + left;
if (channels) *channels = f->channels;
*samples = len;
*output = f->outputs;
return (int) (f->stream - data);
}
stb_vorbis *stb_vorbis_open_pushdata(
const unsigned char *data, int data_len, // the memory available for decoding
int *data_used, // only defined if result is not NULL
int *error, const stb_vorbis_alloc *alloc)
{
stb_vorbis *f, p;
vorbis_init(&p, alloc);
p.stream = (uint8 *) data;
p.stream_end = (uint8 *) data + data_len;
p.push_mode = TRUE;
if (!start_decoder(&p)) {
if (p.eof)
*error = VORBIS_need_more_data;
else
*error = p.error;
return NULL;
}
f = vorbis_alloc(&p);
if (f) {
*f = p;
*data_used = (int) (f->stream - data);
*error = 0;
return f;
} else {
vorbis_deinit(&p);
return NULL;
}
}
#endif // STB_VORBIS_NO_PUSHDATA_API
unsigned int stb_vorbis_get_file_offset(stb_vorbis *f)
{
#ifndef STB_VORBIS_NO_PUSHDATA_API
if (f->push_mode) return 0;
#endif
if (USE_MEMORY(f)) return (unsigned int) (f->stream - f->stream_start);
#ifndef STB_VORBIS_NO_STDIO
return (unsigned int) (ftell(f->f) - f->f_start);
#endif
}
#ifndef STB_VORBIS_NO_PULLDATA_API
//
// DATA-PULLING API
//
static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last)
{
for(;;) {
int n;
if (f->eof) return 0;
n = get8(f);
if (n == 0x4f) { // page header candidate
unsigned int retry_loc = stb_vorbis_get_file_offset(f);
int i;
// check if we're off the end of a file_section stream
if (retry_loc - 25 > f->stream_len)
return 0;
// check the rest of the header
for (i=1; i < 4; ++i)
if (get8(f) != ogg_page_header[i])
break;
if (f->eof) return 0;
if (i == 4) {
uint8 header[27];
uint32 i, crc, goal, len;
for (i=0; i < 4; ++i)
header[i] = ogg_page_header[i];
for (; i < 27; ++i)
header[i] = get8(f);
if (f->eof) return 0;
if (header[4] != 0) goto invalid;
goal = header[22] + (header[23] << 8) + (header[24]<<16) + (header[25]<<24);
for (i=22; i < 26; ++i)
header[i] = 0;
crc = 0;
for (i=0; i < 27; ++i)
crc = crc32_update(crc, header[i]);
len = 0;
for (i=0; i < header[26]; ++i) {
int s = get8(f);
crc = crc32_update(crc, s);
len += s;
}
if (len && f->eof) return 0;
for (i=0; i < len; ++i)
crc = crc32_update(crc, get8(f));
// finished parsing probable page
if (crc == goal) {
// we could now check that it's either got the last
// page flag set, OR it's followed by the capture
// pattern, but I guess TECHNICALLY you could have
// a file with garbage between each ogg page and recover
// from it automatically? So even though that paranoia
// might decrease the chance of an invalid decode by
// another 2^32, not worth it since it would hose those
// invalid-but-useful files?
if (end)
*end = stb_vorbis_get_file_offset(f);
if (last) {
if (header[5] & 0x04)
*last = 1;
else
*last = 0;
}
set_file_offset(f, retry_loc-1);
return 1;
}
}
invalid:
// not a valid page, so rewind and look for next one
set_file_offset(f, retry_loc);
}
}
}
#define SAMPLE_unknown 0xffffffff
// seeking is implemented with a binary search, which narrows down the range to
// 64K, before using a linear search (because finding the synchronization
// pattern can be expensive, and the chance we'd find the end page again is
// relatively high for small ranges)
//
// two initial interpolation-style probes are used at the start of the search
// to try to bound either side of the binary search sensibly, while still
// working in O(log n) time if they fail.
static int get_seek_page_info(stb_vorbis *f, ProbedPage *z)
{
uint8 header[27], lacing[255];
int i,len;
// record where the page starts
z->page_start = stb_vorbis_get_file_offset(f);
// parse the header
getn(f, header, 27);
if (header[0] != 'O' || header[1] != 'g' || header[2] != 'g' || header[3] != 'S')
return 0;
getn(f, lacing, header[26]);
// determine the length of the payload
len = 0;
for (i=0; i < header[26]; ++i)
len += lacing[i];
// this implies where the page ends
z->page_end = z->page_start + 27 + header[26] + len;
// read the last-decoded sample out of the data
z->last_decoded_sample = header[6] + (header[7] << 8) + (header[8] << 16) + (header[9] << 24);
// restore file state to where we were
set_file_offset(f, z->page_start);
return 1;
}
// rarely used function to seek back to the preceeding page while finding the
// start of a packet
static int go_to_page_before(stb_vorbis *f, unsigned int limit_offset)
{
unsigned int previous_safe, end;
// now we want to seek back 64K from the limit
if (limit_offset >= 65536 && limit_offset-65536 >= f->first_audio_page_offset)
previous_safe = limit_offset - 65536;
else
previous_safe = f->first_audio_page_offset;
set_file_offset(f, previous_safe);
while (vorbis_find_page(f, &end, NULL)) {
if (end >= limit_offset && stb_vorbis_get_file_offset(f) < limit_offset)
return 1;
set_file_offset(f, end);
}
return 0;
}
// implements the search logic for finding a page and starting decoding. if
// the function succeeds, current_loc_valid will be true and current_loc will
// be less than or equal to the provided sample number (the closer the
// better).
static int seek_to_sample_coarse(stb_vorbis *f, uint32 sample_number)
{
ProbedPage left, right, mid;
int i, start_seg_with_known_loc, end_pos, page_start;
uint32 delta, stream_length, padding;
double offset, bytes_per_sample;
int probe = 0;
// find the last page and validate the target sample
stream_length = stb_vorbis_stream_length_in_samples(f);
if (stream_length == 0) return error(f, VORBIS_seek_without_length);
if (sample_number > stream_length) return error(f, VORBIS_seek_invalid);
// this is the maximum difference between the window-center (which is the
// actual granule position value), and the right-start (which the spec
// indicates should be the granule position (give or take one)).
padding = ((f->blocksize_1 - f->blocksize_0) >> 2);
if (sample_number < padding)
sample_number = 0;
else
sample_number -= padding;
left = f->p_first;
while (left.last_decoded_sample == ~0U) {
// (untested) the first page does not have a 'last_decoded_sample'
set_file_offset(f, left.page_end);
if (!get_seek_page_info(f, &left)) goto error;
}
right = f->p_last;
assert(right.last_decoded_sample != ~0U);
// starting from the start is handled differently
if (sample_number <= left.last_decoded_sample) {
stb_vorbis_seek_start(f);
return 1;
}
while (left.page_end != right.page_start) {
assert(left.page_end < right.page_start);
// search range in bytes
delta = right.page_start - left.page_end;
if (delta <= 65536) {
// there's only 64K left to search - handle it linearly
set_file_offset(f, left.page_end);
} else {
if (probe < 2) {
if (probe == 0) {
// first probe (interpolate)
double data_bytes = right.page_end - left.page_start;
bytes_per_sample = data_bytes / right.last_decoded_sample;
offset = left.page_start + bytes_per_sample * (sample_number - left.last_decoded_sample);
} else {
// second probe (try to bound the other side)
double error = ((double) sample_number - mid.last_decoded_sample) * bytes_per_sample;
if (error >= 0 && error < 8000) error = 8000;
if (error < 0 && error > -8000) error = -8000;
offset += error * 2;
}
// ensure the offset is valid
if (offset < left.page_end)
offset = left.page_end;
if (offset > right.page_start - 65536)
offset = right.page_start - 65536;
set_file_offset(f, (unsigned int) offset);
} else {
// binary search for large ranges (offset by 32K to ensure
// we don't hit the right page)
set_file_offset(f, left.page_end + (delta / 2) - 32768);
}
if (!vorbis_find_page(f, NULL, NULL)) goto error;
}
for (;;) {
if (!get_seek_page_info(f, &mid)) goto error;
if (mid.last_decoded_sample != ~0U) break;
// (untested) no frames end on this page
set_file_offset(f, mid.page_end);
assert(mid.page_start < right.page_start);
}
// if we've just found the last page again then we're in a tricky file,
// and we're close enough.
if (mid.page_start == right.page_start)
break;
if (sample_number < mid.last_decoded_sample)
right = mid;
else
left = mid;
++probe;
}
// seek back to start of the last packet
page_start = left.page_start;
set_file_offset(f, page_start);
if (!start_page(f)) return error(f, VORBIS_seek_failed);
end_pos = f->end_seg_with_known_loc;
assert(end_pos >= 0);
for (;;) {
for (i = end_pos; i > 0; --i)
if (f->segments[i-1] != 255)
break;
start_seg_with_known_loc = i;
if (start_seg_with_known_loc > 0 || !(f->page_flag & PAGEFLAG_continued_packet))
break;
// (untested) the final packet begins on an earlier page
if (!go_to_page_before(f, page_start))
goto error;
page_start = stb_vorbis_get_file_offset(f);
if (!start_page(f)) goto error;
end_pos = f->segment_count - 1;
}
// prepare to start decoding
f->current_loc_valid = FALSE;
f->last_seg = FALSE;
f->valid_bits = 0;
f->packet_bytes = 0;
f->bytes_in_seg = 0;
f->previous_length = 0;
f->next_seg = start_seg_with_known_loc;
for (i = 0; i < start_seg_with_known_loc; i++)
skip(f, f->segments[i]);
// start decoding (optimizable - this frame is generally discarded)
vorbis_pump_first_frame(f);
return 1;
error:
// try to restore the file to a valid state
stb_vorbis_seek_start(f);
return error(f, VORBIS_seek_failed);
}
// the same as vorbis_decode_initial, but without advancing
static int peek_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode)
{
int bits_read, bytes_read;
if (!vorbis_decode_initial(f, p_left_start, p_left_end, p_right_start, p_right_end, mode))
return 0;
// either 1 or 2 bytes were read, figure out which so we can rewind
bits_read = 1 + ilog(f->mode_count-1);
if (f->mode_config[*mode].blockflag)
bits_read += 2;
bytes_read = (bits_read + 7) / 8;
f->bytes_in_seg += bytes_read;
f->packet_bytes -= bytes_read;
skip(f, -bytes_read);
if (f->next_seg == -1)
f->next_seg = f->segment_count - 1;
else
f->next_seg--;
f->valid_bits = 0;
return 1;
}
int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number)
{
uint32 max_frame_samples;
if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
// fast page-level search
if (!seek_to_sample_coarse(f, sample_number))
return 0;
assert(f->current_loc_valid);
assert(f->current_loc <= sample_number);
// linear search for the relevant packet
max_frame_samples = (f->blocksize_1*3 - f->blocksize_0) >> 2;
while (f->current_loc < sample_number) {
int left_start, left_end, right_start, right_end, mode, frame_samples;
if (!peek_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode))
return error(f, VORBIS_seek_failed);
// calculate the number of samples returned by the next frame
frame_samples = right_start - left_start;
if (f->current_loc + frame_samples > sample_number) {
return 1; // the next frame will contain the sample
} else if (f->current_loc + frame_samples + max_frame_samples > sample_number) {
// there's a chance the frame after this could contain the sample
vorbis_pump_first_frame(f);
} else {
// this frame is too early to be relevant
f->current_loc += frame_samples;
f->previous_length = 0;
maybe_start_packet(f);
flush_packet(f);
}
}
// the next frame will start with the sample
assert(f->current_loc == sample_number);
return 1;
}
int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number)
{
if (!stb_vorbis_seek_frame(f, sample_number))
return 0;
if (sample_number != f->current_loc) {
int n;
uint32 frame_start = f->current_loc;
stb_vorbis_get_frame_float(f, &n, NULL);
assert(sample_number > frame_start);
assert(f->channel_buffer_start + (int) (sample_number-frame_start) <= f->channel_buffer_end);
f->channel_buffer_start += (sample_number - frame_start);
}
return 1;
}
void stb_vorbis_seek_start(stb_vorbis *f)
{
if (IS_PUSH_MODE(f)) { error(f, VORBIS_invalid_api_mixing); return; }
set_file_offset(f, f->first_audio_page_offset);
f->previous_length = 0;
f->first_decode = TRUE;
f->next_seg = -1;
vorbis_pump_first_frame(f);
}
unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f)
{
unsigned int restore_offset, previous_safe;
unsigned int end, last_page_loc;
if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
if (!f->total_samples) {
unsigned int last;
uint32 lo,hi;
char header[6];
// first, store the current decode position so we can restore it
restore_offset = stb_vorbis_get_file_offset(f);
// now we want to seek back 64K from the end (the last page must
// be at most a little less than 64K, but let's allow a little slop)
if (f->stream_len >= 65536 && f->stream_len-65536 >= f->first_audio_page_offset)
previous_safe = f->stream_len - 65536;
else
previous_safe = f->first_audio_page_offset;
set_file_offset(f, previous_safe);
// previous_safe is now our candidate 'earliest known place that seeking
// to will lead to the final page'
if (!vorbis_find_page(f, &end, &last)) {
// if we can't find a page, we're hosed!
f->error = VORBIS_cant_find_last_page;
f->total_samples = 0xffffffff;
goto done;
}
// check if there are more pages
last_page_loc = stb_vorbis_get_file_offset(f);
// stop when the last_page flag is set, not when we reach eof;
// this allows us to stop short of a 'file_section' end without
// explicitly checking the length of the section
while (!last) {
set_file_offset(f, end);
if (!vorbis_find_page(f, &end, &last)) {
// the last page we found didn't have the 'last page' flag
// set. whoops!
break;
}
previous_safe = last_page_loc+1;
last_page_loc = stb_vorbis_get_file_offset(f);
}
set_file_offset(f, last_page_loc);
// parse the header
getn(f, (unsigned char *)header, 6);
// extract the absolute granule position
lo = get32(f);
hi = get32(f);
if (lo == 0xffffffff && hi == 0xffffffff) {
f->error = VORBIS_cant_find_last_page;
f->total_samples = SAMPLE_unknown;
goto done;
}
if (hi)
lo = 0xfffffffe; // saturate
f->total_samples = lo;
f->p_last.page_start = last_page_loc;
f->p_last.page_end = end;
f->p_last.last_decoded_sample = lo;
done:
set_file_offset(f, restore_offset);
}
return f->total_samples == SAMPLE_unknown ? 0 : f->total_samples;
}
float stb_vorbis_stream_length_in_seconds(stb_vorbis *f)
{
return stb_vorbis_stream_length_in_samples(f) / (float) f->sample_rate;
}
int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output)
{
int len, right,left,i;
if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
if (!vorbis_decode_packet(f, &len, &left, &right)) {
f->channel_buffer_start = f->channel_buffer_end = 0;
return 0;
}
len = vorbis_finish_frame(f, len, left, right);
for (i=0; i < f->channels; ++i)
f->outputs[i] = f->channel_buffers[i] + left;
f->channel_buffer_start = left;
f->channel_buffer_end = left+len;
if (channels) *channels = f->channels;
if (output) *output = f->outputs;
return len;
}
#ifndef STB_VORBIS_NO_STDIO
stb_vorbis * stb_vorbis_open_file_section(FILE *file, int close_on_free, int *error, const stb_vorbis_alloc *alloc, unsigned int length)
{
stb_vorbis *f, p;
vorbis_init(&p, alloc);
p.f = file;
p.f_start = (uint32) ftell(file);
p.stream_len = length;
p.close_on_free = close_on_free;
if (start_decoder(&p)) {
f = vorbis_alloc(&p);
if (f) {
*f = p;
vorbis_pump_first_frame(f);
return f;
}
}
if (error) *error = p.error;
vorbis_deinit(&p);
return NULL;
}
stb_vorbis * stb_vorbis_open_file(FILE *file, int close_on_free, int *error, const stb_vorbis_alloc *alloc)
{
unsigned int len, start;
start = (unsigned int) ftell(file);
fseek(file, 0, SEEK_END);
len = (unsigned int) (ftell(file) - start);
fseek(file, start, SEEK_SET);
return stb_vorbis_open_file_section(file, close_on_free, error, alloc, len);
}
stb_vorbis * stb_vorbis_open_filename(const char *filename, int *error, const stb_vorbis_alloc *alloc)
{
FILE *f = fopen(filename, "rb");
if (f)
return stb_vorbis_open_file(f, TRUE, error, alloc);
if (error) *error = VORBIS_file_open_failure;
return NULL;
}
#endif // STB_VORBIS_NO_STDIO
stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, int *error, const stb_vorbis_alloc *alloc)
{
stb_vorbis *f, p;
if (data == NULL) return NULL;
vorbis_init(&p, alloc);
p.stream = (uint8 *) data;
p.stream_end = (uint8 *) data + len;
p.stream_start = (uint8 *) p.stream;
p.stream_len = len;
p.push_mode = FALSE;
if (start_decoder(&p)) {
f = vorbis_alloc(&p);
if (f) {
*f = p;
vorbis_pump_first_frame(f);
return f;
}
}
if (error) *error = p.error;
vorbis_deinit(&p);
return NULL;
}
#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
#define PLAYBACK_MONO 1
#define PLAYBACK_LEFT 2
#define PLAYBACK_RIGHT 4
#define L (PLAYBACK_LEFT | PLAYBACK_MONO)
#define C (PLAYBACK_LEFT | PLAYBACK_RIGHT | PLAYBACK_MONO)
#define R (PLAYBACK_RIGHT | PLAYBACK_MONO)
static int8 channel_position[7][6] =
{
{ 0 },
{ C },
{ L, R },
{ L, C, R },
{ L, R, L, R },
{ L, C, R, L, R },
{ L, C, R, L, R, C },
};
#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT
typedef union {
float f;
int i;
} float_conv;
typedef char stb_vorbis_float_size_test[sizeof(float)==4 && sizeof(int) == 4];
#define FASTDEF(x) float_conv x
// add (1<<23) to convert to int, then divide by 2^SHIFT, then add 0.5/2^SHIFT to round
#define MAGIC(SHIFT) (1.5f * (1 << (23-SHIFT)) + 0.5f/(1 << SHIFT))
#define ADDEND(SHIFT) (((150-SHIFT) << 23) + (1 << 22))
#define FAST_SCALED_FLOAT_TO_INT(temp,x,s) (temp.f = (x) + MAGIC(s), temp.i - ADDEND(s))
#define check_endianness()
#else
#define FAST_SCALED_FLOAT_TO_INT(temp,x,s) ((int) ((x) * (1 << (s))))
#define check_endianness()
#define FASTDEF(x)
#endif
static void copy_samples(short *dest, float *src, int len)
{
int i;
check_endianness();
for (i=0; i < len; ++i) {
FASTDEF(temp);
int v = FAST_SCALED_FLOAT_TO_INT(temp, src[i],15);
if ((unsigned int) (v + 32768) > 65535)
v = v < 0 ? -32768 : 32767;
dest[i] = v;
}
}
static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len)
{
#define BUFFER_SIZE 32
float buffer[BUFFER_SIZE];
int i,j,o,n = BUFFER_SIZE;
check_endianness();
for (o = 0; o < len; o += BUFFER_SIZE) {
memset(buffer, 0, sizeof(buffer));
if (o + n > len) n = len - o;
for (j=0; j < num_c; ++j) {
if (channel_position[num_c][j] & mask) {
for (i=0; i < n; ++i)
buffer[i] += data[j][d_offset+o+i];
}
}
for (i=0; i < n; ++i) {
FASTDEF(temp);
int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15);
if ((unsigned int) (v + 32768) > 65535)
v = v < 0 ? -32768 : 32767;
output[o+i] = v;
}
}
}
static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len)
{
#define BUFFER_SIZE 32
float buffer[BUFFER_SIZE];
int i,j,o,n = BUFFER_SIZE >> 1;
// o is the offset in the source data
check_endianness();
for (o = 0; o < len; o += BUFFER_SIZE >> 1) {
// o2 is the offset in the output data
int o2 = o << 1;
memset(buffer, 0, sizeof(buffer));
if (o + n > len) n = len - o;
for (j=0; j < num_c; ++j) {
int m = channel_position[num_c][j] & (PLAYBACK_LEFT | PLAYBACK_RIGHT);
if (m == (PLAYBACK_LEFT | PLAYBACK_RIGHT)) {
for (i=0; i < n; ++i) {
buffer[i*2+0] += data[j][d_offset+o+i];
buffer[i*2+1] += data[j][d_offset+o+i];
}
} else if (m == PLAYBACK_LEFT) {
for (i=0; i < n; ++i) {
buffer[i*2+0] += data[j][d_offset+o+i];
}
} else if (m == PLAYBACK_RIGHT) {
for (i=0; i < n; ++i) {
buffer[i*2+1] += data[j][d_offset+o+i];
}
}
}
for (i=0; i < (n<<1); ++i) {
FASTDEF(temp);
int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15);
if ((unsigned int) (v + 32768) > 65535)
v = v < 0 ? -32768 : 32767;
output[o2+i] = v;
}
}
}
static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples)
{
int i;
if (buf_c != data_c && buf_c <= 2 && data_c <= 6) {
static int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} };
for (i=0; i < buf_c; ++i)
compute_samples(channel_selector[buf_c][i], buffer[i]+b_offset, data_c, data, d_offset, samples);
} else {
int limit = buf_c < data_c ? buf_c : data_c;
for (i=0; i < limit; ++i)
copy_samples(buffer[i]+b_offset, data[i]+d_offset, samples);
for ( ; i < buf_c; ++i)
memset(buffer[i]+b_offset, 0, sizeof(short) * samples);
}
}
int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples)
{
float **output;
int len = stb_vorbis_get_frame_float(f, NULL, &output);
if (len > num_samples) len = num_samples;
if (len)
convert_samples_short(num_c, buffer, 0, f->channels, output, 0, len);
return len;
}
static void convert_channels_short_interleaved(int buf_c, short *buffer, int data_c, float **data, int d_offset, int len)
{
int i;
check_endianness();
if (buf_c != data_c && buf_c <= 2 && data_c <= 6) {
assert(buf_c == 2);
for (i=0; i < buf_c; ++i)
compute_stereo_samples(buffer, data_c, data, d_offset, len);
} else {
int limit = buf_c < data_c ? buf_c : data_c;
int j;
for (j=0; j < len; ++j) {
for (i=0; i < limit; ++i) {
FASTDEF(temp);
float f = data[i][d_offset+j];
int v = FAST_SCALED_FLOAT_TO_INT(temp, f,15);//data[i][d_offset+j],15);
if ((unsigned int) (v + 32768) > 65535)
v = v < 0 ? -32768 : 32767;
*buffer++ = v;
}
for ( ; i < buf_c; ++i)
*buffer++ = 0;
}
}
}
int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts)
{
float **output;
int len;
if (num_c == 1) return stb_vorbis_get_frame_short(f,num_c,&buffer, num_shorts);
len = stb_vorbis_get_frame_float(f, NULL, &output);
if (len) {
if (len*num_c > num_shorts) len = num_shorts / num_c;
convert_channels_short_interleaved(num_c, buffer, f->channels, output, 0, len);
}
return len;
}
int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts)
{
float **outputs;
int len = num_shorts / channels;
int n=0;
int z = f->channels;
if (z > channels) z = channels;
while (n < len) {
int k = f->channel_buffer_end - f->channel_buffer_start;
if (n+k >= len) k = len - n;
if (k)
convert_channels_short_interleaved(channels, buffer, f->channels, f->channel_buffers, f->channel_buffer_start, k);
buffer += k*channels;
n += k;
f->channel_buffer_start += k;
if (n == len) break;
if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
}
return n;
}
int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int len)
{
float **outputs;
int n=0;
int z = f->channels;
if (z > channels) z = channels;
while (n < len) {
int k = f->channel_buffer_end - f->channel_buffer_start;
if (n+k >= len) k = len - n;
if (k)
convert_samples_short(channels, buffer, n, f->channels, f->channel_buffers, f->channel_buffer_start, k);
n += k;
f->channel_buffer_start += k;
if (n == len) break;
if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
}
return n;
}
#ifndef STB_VORBIS_NO_STDIO
int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output)
{
int data_len, offset, total, limit, error;
short *data;
stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL);
if (v == NULL) return -1;
limit = v->channels * 4096;
*channels = v->channels;
if (sample_rate)
*sample_rate = v->sample_rate;
offset = data_len = 0;
total = limit;
data = (short *) malloc(total * sizeof(*data));
if (data == NULL) {
stb_vorbis_close(v);
return -2;
}
for (;;) {
int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset);
if (n == 0) break;
data_len += n;
offset += n * v->channels;
if (offset + limit > total) {
short *data2;
total *= 2;
data2 = (short *) realloc(data, total * sizeof(*data));
if (data2 == NULL) {
free(data);
stb_vorbis_close(v);
return -2;
}
data = data2;
}
}
*output = data;
stb_vorbis_close(v);
return data_len;
}
#endif // NO_STDIO
int stb_vorbis_decode_memory(const uint8 *mem, int len, int *channels, int *sample_rate, short **output)
{
int data_len, offset, total, limit, error;
short *data;
stb_vorbis *v = stb_vorbis_open_memory(mem, len, &error, NULL);
if (v == NULL) return -1;
limit = v->channels * 4096;
*channels = v->channels;
if (sample_rate)
*sample_rate = v->sample_rate;
offset = data_len = 0;
total = limit;
data = (short *) malloc(total * sizeof(*data));
if (data == NULL) {
stb_vorbis_close(v);
return -2;
}
for (;;) {
int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset);
if (n == 0) break;
data_len += n;
offset += n * v->channels;
if (offset + limit > total) {
short *data2;
total *= 2;
data2 = (short *) realloc(data, total * sizeof(*data));
if (data2 == NULL) {
free(data);
stb_vorbis_close(v);
return -2;
}
data = data2;
}
}
*output = data;
stb_vorbis_close(v);
return data_len;
}
#endif // STB_VORBIS_NO_INTEGER_CONVERSION
int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats)
{
float **outputs;
int len = num_floats / channels;
int n=0;
int z = f->channels;
if (z > channels) z = channels;
while (n < len) {
int i,j;
int k = f->channel_buffer_end - f->channel_buffer_start;
if (n+k >= len) k = len - n;
for (j=0; j < k; ++j) {
for (i=0; i < z; ++i)
*buffer++ = f->channel_buffers[i][f->channel_buffer_start+j];
for ( ; i < channels; ++i)
*buffer++ = 0;
}
n += k;
f->channel_buffer_start += k;
if (n == len)
break;
if (!stb_vorbis_get_frame_float(f, NULL, &outputs))
break;
}
return n;
}
int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples)
{
float **outputs;
int n=0;
int z = f->channels;
if (z > channels) z = channels;
while (n < num_samples) {
int i;
int k = f->channel_buffer_end - f->channel_buffer_start;
if (n+k >= num_samples) k = num_samples - n;
if (k) {
for (i=0; i < z; ++i)
memcpy(buffer[i]+n, f->channel_buffers[i]+f->channel_buffer_start, sizeof(float)*k);
for ( ; i < channels; ++i)
memset(buffer[i]+n, 0, sizeof(float) * k);
}
n += k;
f->channel_buffer_start += k;
if (n == num_samples)
break;
if (!stb_vorbis_get_frame_float(f, NULL, &outputs))
break;
}
return n;
}
#endif // STB_VORBIS_NO_PULLDATA_API
/* Version history
1.09 - 2016/04/04 - back out 'avoid discarding last frame' fix from previous version
1.08 - 2016/04/02 - fixed multiple warnings; fix setup memory leaks;
avoid discarding last frame of audio data
1.07 - 2015/01/16 - fixed some warnings, fix mingw, const-correct API
some more crash fixes when out of memory or with corrupt files
1.06 - 2015/08/31 - full, correct support for seeking API (Dougall Johnson)
some crash fixes when out of memory or with corrupt files
1.05 - 2015/04/19 - don't define __forceinline if it's redundant
1.04 - 2014/08/27 - fix missing const-correct case in API
1.03 - 2014/08/07 - Warning fixes
1.02 - 2014/07/09 - Declare qsort compare function _cdecl on windows
1.01 - 2014/06/18 - fix stb_vorbis_get_samples_float
1.0 - 2014/05/26 - fix memory leaks; fix warnings; fix bugs in multichannel
(API change) report sample rate for decode-full-file funcs
0.99996 - bracket #include <malloc.h> for macintosh compilation by Laurent Gomila
0.99995 - use union instead of pointer-cast for fast-float-to-int to avoid alias-optimization problem
0.99994 - change fast-float-to-int to work in single-precision FPU mode, remove endian-dependence
0.99993 - remove assert that fired on legal files with empty tables
0.99992 - rewind-to-start
0.99991 - bugfix to stb_vorbis_get_samples_short by Bernhard Wodo
0.9999 - (should have been 0.99990) fix no-CRT support, compiling as C++
0.9998 - add a full-decode function with a memory source
0.9997 - fix a bug in the read-from-FILE case in 0.9996 addition
0.9996 - query length of vorbis stream in samples/seconds
0.9995 - bugfix to another optimization that only happened in certain files
0.9994 - bugfix to one of the optimizations that caused significant (but inaudible?) errors
0.9993 - performance improvements; runs in 99% to 104% of time of reference implementation
0.9992 - performance improvement of IMDCT; now performs close to reference implementation
0.9991 - performance improvement of IMDCT
0.999 - (should have been 0.9990) performance improvement of IMDCT
0.998 - no-CRT support from Casey Muratori
0.997 - bugfixes for bugs found by Terje Mathisen
0.996 - bugfix: fast-huffman decode initialized incorrectly for sparse codebooks; fixing gives 10% speedup - found by Terje Mathisen
0.995 - bugfix: fix to 'effective' overrun detection - found by Terje Mathisen
0.994 - bugfix: garbage decode on final VQ symbol of a non-multiple - found by Terje Mathisen
0.993 - bugfix: pushdata API required 1 extra byte for empty page (failed to consume final page if empty) - found by Terje Mathisen
0.992 - fixes for MinGW warning
0.991 - turn fast-float-conversion on by default
0.990 - fix push-mode seek recovery if you seek into the headers
0.98b - fix to bad release of 0.98
0.98 - fix push-mode seek recovery; robustify float-to-int and support non-fast mode
0.97 - builds under c++ (typecasting, don't use 'class' keyword)
0.96 - somehow MY 0.95 was right, but the web one was wrong, so here's my 0.95 rereleased as 0.96, fixes a typo in the clamping code
0.95 - clamping code for 16-bit functions
0.94 - not publically released
0.93 - fixed all-zero-floor case (was decoding garbage)
0.92 - fixed a memory leak
0.91 - conditional compiles to omit parts of the API and the infrastructure to support them: STB_VORBIS_NO_PULLDATA_API, STB_VORBIS_NO_PUSHDATA_API, STB_VORBIS_NO_STDIO, STB_VORBIS_NO_INTEGER_CONVERSION
0.90 - first public release
*/
#endif // STB_VORBIS_HEADER_ONLY
Reference in New Issue View Git Blame Copy Permalink