diff --git a/internal/c/parts/audio/decode/ogg/download/stb_vorbis.c b/internal/c/parts/audio/decode/ogg/download/stb_vorbis.c index b2d217efa..4d0f0482b 100644 --- a/internal/c/parts/audio/decode/ogg/download/stb_vorbis.c +++ b/internal/c/parts/audio/decode/ogg/download/stb_vorbis.c @@ -1,5370 +1,5445 @@ -// Ogg Vorbis I audio decoder -- version 0.99996 -// -// Written in April 2007 by Sean Barrett, sponsored by RAD Game Tools. -// -// Placed in the public domain April 2007 by the author: no copyright is -// claimed, and you may use it for any purpose you like. -// -// 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. -// -// Get the latest version and other information at: -// http://nothings.org/stb_vorbis/ - - -// Todo: -// -// - seeking (note you can seek yourself using the pushdata API) -// -// Limitations: -// -// - floor 0 not supported (used in old ogg vorbis files) -// - 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) -// -// All of these limitations may be removed in future versions. - - -////////////////////////////////////////////////////////////////////////////// -// -// 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 -#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( - unsigned char *datablock, int datablock_length_in_bytes, - int *datablock_memory_consumed_in_bytes, - int *error, - 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, 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(char *filename, int *channels, short **output); -#endif -extern int stb_vorbis_decode_memory(unsigned char *mem, int len, int *channels, short **output); -// 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(unsigned char *data, int len, - int *error, 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(char *filename, - int *error, 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, 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, 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); -// NOT WORKING YET -// 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), but it -// actually works - -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. 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. 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. - -// 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 - -// STB_VORBIS_CODEBOOK_SHORTS -// The vorbis file format encodes VQ codebook floats as ax+b where a and -// b are floating point per-codebook constants, and x is a 16-bit int. -// Normally, stb_vorbis decodes them to floats rather than leaving them -// as 16-bit ints and computing ax+b while decoding. This is a speed/space -// tradeoff; you can save space by defining this flag. -#ifndef STB_VORBIS_CODEBOOK_SHORTS -#define STB_VORBIS_CODEBOOK_FLOATS -#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 -#endif - -#ifndef STB_VORBIS_NO_CRT -#include -#include -#include -#include -#if !(defined(__APPLE__) || defined(MACOSX) || defined(macintosh) || defined(Macintosh)) -#include -#endif -#else -#define NULL 0 -#endif - -#ifndef _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 - - -#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 - -#ifdef STB_VORBIS_CODEBOOK_FLOATS -typedef float codetype; -#else -typedef uint16 codetype; -#endif - -// @NOTE -// -// Some arrays below are tagged "//varies", which means it's actually -// a variable-sized piece of data, but rather than malloc I assume it's -// small enough it's better to just allocate it all together with the -// main thing -// -// Most of the variables are specified with the smallest size I could pack -// them into. It might give better performance to make them all full-sized -// integers. It should be safe to freely rearrange the structures or change -// the sizes larger--nothing relies on silently truncating etc., nor the -// order of variables. - -#define FAST_HUFFMAN_TABLE_SIZE (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH) -#define FAST_HUFFMAN_TABLE_MASK (FAST_HUFFMAN_TABLE_SIZE - 1) - -typedef struct -{ - int dimensions, entries; - uint8 *codeword_lengths; - float minimum_value; - float delta_value; - uint8 value_bits; - uint8 lookup_type; - uint8 sequence_p; - uint8 sparse; - uint32 lookup_values; - codetype *multiplicands; - uint32 *codewords; - #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT - int16 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; - #else - int32 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; - #endif - uint32 *sorted_codewords; - int *sorted_values; - int sorted_entries; -} Codebook; - -typedef struct -{ - uint8 order; - uint16 rate; - uint16 bark_map_size; - uint8 amplitude_bits; - uint8 amplitude_offset; - uint8 number_of_books; - uint8 book_list[16]; // varies -} Floor0; - -typedef struct -{ - uint8 partitions; - uint8 partition_class_list[32]; // varies - uint8 class_dimensions[16]; // varies - uint8 class_subclasses[16]; // varies - uint8 class_masterbooks[16]; // varies - int16 subclass_books[16][8]; // varies - uint16 Xlist[31*8+2]; // varies - uint8 sorted_order[31*8+2]; - uint8 neighbors[31*8+2][2]; - uint8 floor1_multiplier; - uint8 rangebits; - int values; -} Floor1; - -typedef union -{ - Floor0 floor0; - Floor1 floor1; -} Floor; - -typedef struct -{ - uint32 begin, end; - uint32 part_size; - uint8 classifications; - uint8 classbook; - uint8 **classdata; - int16 (*residue_books)[8]; -} Residue; - -typedef struct -{ - uint8 magnitude; - uint8 angle; - uint8 mux; -} MappingChannel; - -typedef struct -{ - uint16 coupling_steps; - MappingChannel *chan; - uint8 submaps; - uint8 submap_floor[15]; // varies - uint8 submap_residue[15]; // varies -} Mapping; - -typedef struct -{ - uint8 blockflag; - uint8 mapping; - uint16 windowtype; - uint16 transformtype; -} Mode; - -typedef struct -{ - uint32 goal_crc; // expected crc if match - int bytes_left; // bytes left in packet - uint32 crc_so_far; // running crc - int bytes_done; // bytes processed in _current_ chunk - uint32 sample_loc; // granule pos encoded in page -} CRCscan; - -typedef struct -{ - uint32 page_start, page_end; - uint32 after_previous_page_start; - uint32 first_decoded_sample; - 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; -}; - -extern int my_prof(int slot); -//#define stb_prof my_prof - -#ifndef stb_prof -#define stb_prof(x) 0 -#endif - -#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 not 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, size_t 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=i<<24, j=0; j < 8; ++j) - s = (s << 1) ^ (s >= (1<<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 < (1U << 14)) - if (n < (1U << 4)) return 0 + log2_4[n ]; - else if (n < (1U << 9)) return 5 + log2_4[n >> 5]; - else return 10 + log2_4[n >> 10]; - else if (n < (1U << 24)) - if (n < (1U << 19)) return 15 + log2_4[n >> 15]; - else return 20 + log2_4[n >> 20]; - else if (n < (1U << 29)) return 25 + log2_4[n >> 25]; - else if (n < (1U << 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] = 1 << (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) { assert(0); return FALSE; } - res = available[z]; - available[z] = 0; - add_entry(c, bit_reverse(res), i, m++, len[i], values); - // propogate availability up the tree - if (z != len[i]) { - for (y=len[i]; y > z; --y) { - assert(available[y] == 0); - available[y] = res + (1 << (32-y)); - } - } - } - return TRUE; -} - -// accelerated huffman table allows fast O(1) match of all symbols -// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH -static void compute_accelerated_huffman(Codebook *c) -{ - int i, len; - for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i) - c->fast_huffman[i] = -1; - - len = c->sparse ? c->sorted_entries : c->entries; - #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT - if (len > 32767) len = 32767; // largest possible value we can encode! - #endif - for (i=0; i < len; ++i) { - if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) { - uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i]; - // set table entries for all bit combinations in the higher bits - while (z < FAST_HUFFMAN_TABLE_SIZE) { - c->fast_huffman[z] = i; - z += 1 << c->codeword_lengths[i]; - } - } - } -} - -static int 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; - -int 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 += 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,i; - // 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 != ~0 || loc1 != ~0) { - // 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.after_previous_page_start = p.page_start; - p.first_decoded_sample = 0; - 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) - 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; -} - -static int32 get_bits_signed(vorb *f, int n) -{ - uint32 z = get_bits(f, n); - if (z & (1 << (n-1))) - z += ~((1 << n) - 1); - return (int32) 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 += 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); - - assert(c->sorted_codewords || c->codewords); - // 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; -} - -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); -} - -#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 - -#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 -#ifndef STB_VORBIS_CODEBOOK_FLOATS - #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off] * c->delta_value + c->minimum_value) - #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off] * c->delta_value) - #define CODEBOOK_ELEMENT_BASE(c) (c->minimum_value) -#else - #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off]) - #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off]) - #define CODEBOOK_ELEMENT_BASE(c) (0) -#endif - -static int codebook_decode_start(vorb *f, Codebook *c, int len) -{ - 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,len); - 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,len); - 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; - 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; - 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; - 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; -} - -#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK -static int codebook_decode_deinterleave_repeat_2(vorb *f, Codebook *c, float **outputs, 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); - - 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*2 + effective > len * 2) { - effective = len*2 - (p_inter*2 - c_inter); - } - - { - z *= c->dimensions; - stb_prof(11); - if (c->sequence_p) { - // haven't optimized this case because I don't have any examples - for (i=0; i < effective; ++i) { - float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; - outputs[c_inter][p_inter] += val; - if (++c_inter == 2) { c_inter = 0; ++p_inter; } - last = val; - } - } else { - i=0; - if (c_inter == 1) { - float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; - outputs[c_inter][p_inter] += val; - c_inter = 0; ++p_inter; - ++i; - } - { - float *z0 = outputs[0]; - float *z1 = outputs[1]; - for (; i+1 < effective;) { - z0[p_inter] += CODEBOOK_ELEMENT_FAST(c,z+i) + last; - z1[p_inter] += CODEBOOK_ELEMENT_FAST(c,z+i+1) + last; - ++p_inter; - i += 2; - } - } - if (i < effective) { - float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; - outputs[c_inter][p_inter] += val; - if (++c_inter == 2) { c_inter = 0; ++p_inter; } - } - } - } - - total_decode -= effective; - } - *c_inter_p = c_inter; - *p_inter_p = p_inter; - return TRUE; -} -#endif - -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; - 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 - - stb_prof(2); - for (i=0; i < ch; ++i) - if (!do_not_decode[i]) - memset(residue_buffers[i], 0, sizeof(float) * n); - - if (rtype == 2 && ch != 1) { - int len = ch * n; - for (j=0; j < ch; ++j) - if (!do_not_decode[j]) - break; - if (j == ch) - goto done; - - stb_prof(3); - for (pass=0; pass < 8; ++pass) { - int pcount = 0, class_set = 0; - if (ch == 2) { - stb_prof(13); - 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 - } - stb_prof(5); - 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; - stb_prof(20); // accounts for X time - #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_2(f, book, residue_buffers, &c_inter, &p_inter, n, r->part_size)) - goto done; - #endif - stb_prof(7); - } else { - z += r->part_size; - c_inter = z & 1; - p_inter = z >> 1; - } - } - stb_prof(8); - #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; - stb_prof(22); - if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) - goto done; - stb_prof(3); - } 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; - stb_prof(22); - if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) - goto done; - stb_prof(3); - } else { - z += r->part_size; - c_inter = z % ch; - p_inter = z / ch; - } - } - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - ++class_set; - #endif - } - } - } - goto done; - } - stb_prof(9); - - 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: - stb_prof(0); - 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); -} -#else -// 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]; - //acc += x[j] * cos(M_PI / n * (i + 0.5) * (j + 0.5)); - buffer[i] = acc; - } - free(x); -} - -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 k_off = -8; - 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 n3_4 = n - n4, 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_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]; - draw_line(target, lx,ly, hx,hy, n2); - 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]); - } - return TRUE; -} - -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]; - int window_center; - -// WINDOWING - - n = f->blocksize[m->blockflag]; - window_center = n >> 1; - - map = &f->mapping[m->mapping]; - -// FLOORS - n2 = n >> 1; - - stb_prof(1); - 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 - } - } - stb_prof(0); - // 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; - } - -// RESIDUE DECODE - for (i=0; i < map->submaps; ++i) { - float *residue_buffers[STB_VORBIS_MAX_CHANNELS]; - int r,t; - 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]; - t = f->residue_types[r]; - 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); - -// INVERSE COUPLING - stb_prof(14); - 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; - } - } - - // finish decoding the floors -#ifndef STB_VORBIS_NO_DEFER_FLOOR - stb_prof(15); - 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 - stb_prof(16); - for (i=0; i < f->channels; ++i) - inverse_mdct(f->channel_buffers[i], n, f, m->blockflag); - stb_prof(0); - - // 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) { - 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) { - if (current_end < f->current_loc) { - // negative truncation, that's impossible! - *len = 0; - } else { - *len = current_end - f->current_loc; - } - *len += left_start; - 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 - 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 == f->segment_count) - 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; - 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->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; - } 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); - 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; - //assert(total != 0); - } 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; - - 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)); - // 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) { ++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); - - 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; - } - 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; - // 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) { - int z = sparse ? c->sorted_values[j] : j, div=1; - for (k=0; k < c->dimensions; ++k) { - int off = (z / div) % c->lookup_values; - c->multiplicands[j*c->dimensions + k] = - #ifndef STB_VORBIS_CODEBOOK_FLOATS - mults[off]; - #else - mults[off]*c->delta_value + c->minimum_value; - // in this case (and this case only) we could pre-expand c->sequence_p, - // and throw away the decode logic for it; have to ALSO do - // it in the case below, but it can only be done if - // STB_VORBIS_CODEBOOK_FLOATS - // !STB_VORBIS_DIVIDES_IN_CODEBOOK - #endif - div *= c->lookup_values; - } - } - setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); - c->lookup_type = 2; - } - else -#endif - { - c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values); - #ifndef STB_VORBIS_CODEBOOK_FLOATS - memcpy(c->multiplicands, mults, sizeof(c->multiplicands[0]) * c->lookup_values); - #else - for (j=0; j < (int) c->lookup_values; ++j) - c->multiplicands[j] = mults[j] * c->delta_value + c->minimum_value; - setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); - #endif - } - skip:; - - #ifdef STB_VORBIS_CODEBOOK_FLOATS - if (c->lookup_type == 2 && c->sequence_p) { - for (j=1; j < (int) c->lookup_values; ++j) - c->multiplicands[j] = c->multiplicands[j-1]; - c->sequence_p = 0; - } - #endif - } - } - - // 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)); - 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)); - 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); - r->part_size = get_bits(f,24)+1; - r->classifications = get_bits(f,6)+1; - r->classbook = get_bits(f,8); - 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); - 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); - 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)); - 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 (get_bits(f,1)) - m->submaps = get_bits(f,4); - 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); - #ifdef STB_VORBIS_NO_DEFER_FLOOR - f->floor_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); - #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; - 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) { - 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); - for (i=0; i < p->mapping_count; ++i) - setup_free(p, p->mapping[i].chan); - setup_free(p, p->mapping); - for (i=0; i < p->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]); - } - #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, 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 != ~0; - 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 - 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, data, data_len); - } - - f->stream = data; - f->stream_end = 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 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 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 f->stream - data; -} - -stb_vorbis *stb_vorbis_open_pushdata( - unsigned char *data, int data_len, // the memory available for decoding - int *data_used, // only defined if result is not NULL - int *error, stb_vorbis_alloc *alloc) -{ - stb_vorbis *f, p; - vorbis_init(&p, alloc); - p.stream = data; - p.stream_end = 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 = 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 f->stream - f->stream_start; - #ifndef STB_VORBIS_NO_STDIO - return 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 - 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); - } - } -} - -// seek is implemented with 'interpolation search'--this is like -// binary search, but we use the data values to estimate the likely -// location of the data item (plus a bit of a bias so when the -// estimation is wrong we don't waste overly much time) - -#define SAMPLE_unknown 0xffffffff - - -// ogg vorbis, in its insane infinite wisdom, only provides -// information about the sample at the END of the page. -// therefore we COULD have the data we need in the current -// page, and not know it. we could just use the end location -// as our only knowledge for bounds, seek back, and eventually -// the binary search finds it. or we can try to be smart and -// not waste time trying to locate more pages. we try to be -// smart, since this data is already in memory anyway, so -// doing needless I/O would be crazy! -static int vorbis_analyze_page(stb_vorbis *f, ProbedPage *z) -{ - uint8 header[27], lacing[255]; - uint8 packet_type[255]; - int num_packet, packet_start, previous =0; - int i,len; - uint32 samples; - - // record where the page starts - z->page_start = stb_vorbis_get_file_offset(f); - - // parse the header - getn(f, header, 27); - assert(header[0] == 'O' && header[1] == 'g' && header[2] == 'g' && header[3] == 'S'); - 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] << 16); - - if (header[5] & 4) { - // if this is the last page, it's not possible to work - // backwards to figure out the first sample! whoops! fuck. - z->first_decoded_sample = SAMPLE_unknown; - set_file_offset(f, z->page_start); - return 1; - } - - // scan through the frames to determine the sample-count of each one... - // our goal is the sample # of the first fully-decoded sample on the - // page, which is the first decoded sample of the 2nd page - - num_packet=0; - - packet_start = ((header[5] & 1) == 0); - - for (i=0; i < header[26]; ++i) { - if (packet_start) { - uint8 n,b,m; - if (lacing[i] == 0) goto bail; // trying to read from zero-length packet - n = get8(f); - // if bottom bit is non-zero, we've got corruption - if (n & 1) goto bail; - n >>= 1; - b = ilog(f->mode_count-1); - m = n >> b; - n &= (1 << b)-1; - if (n >= f->mode_count) goto bail; - if (num_packet == 0 && f->mode_config[n].blockflag) - previous = (m & 1); - packet_type[num_packet++] = f->mode_config[n].blockflag; - skip(f, lacing[i]-1); - } else - skip(f, lacing[i]); - packet_start = (lacing[i] < 255); - } - - // now that we know the sizes of all the pages, we can start determining - // how much sample data there is. - - samples = 0; - - // for the last packet, we step by its whole length, because the definition - // is that we encoded the end sample loc of the 'last packet completed', - // where 'completed' refers to packets being split, and we are left to guess - // what 'end sample loc' means. we assume it means ignoring the fact that - // the last half of the data is useless without windowing against the next - // packet... (so it's not REALLY complete in that sense) - if (num_packet > 1) - samples += f->blocksize[packet_type[num_packet-1]]; - - for (i=num_packet-2; i >= 1; --i) { - // now, for this packet, how many samples do we have that - // do not overlap the following packet? - if (packet_type[i] == 1) - if (packet_type[i+1] == 1) - samples += f->blocksize_1 >> 1; - else - samples += ((f->blocksize_1 - f->blocksize_0) >> 2) + (f->blocksize_0 >> 1); - else - samples += f->blocksize_0 >> 1; - } - // now, at this point, we've rewound to the very beginning of the - // _second_ packet. if we entirely discard the first packet after - // a seek, this will be exactly the right sample number. HOWEVER! - // we can't as easily compute this number for the LAST page. The - // only way to get the sample offset of the LAST page is to use - // the end loc from the previous page. But what that returns us - // is _exactly_ the place where we get our first non-overlapped - // sample. (I think. Stupid spec for being ambiguous.) So for - // consistency it's better to do that here, too. However, that - // will then require us to NOT discard all of the first frame we - // decode, in some cases, which means an even weirder frame size - // and extra code. what a fucking pain. - - // we're going to discard the first packet if we - // start the seek here, so we don't care about it. (we could actually - // do better; if the first packet is long, and the previous packet - // is short, there's actually data in the first half of the first - // packet that doesn't need discarding... but not worth paying the - // effort of tracking that of that here and in the seeking logic) - // except crap, if we infer it from the _previous_ packet's end - // location, we DO need to use that definition... and we HAVE to - // infer the start loc of the LAST packet from the previous packet's - // end location. fuck you, ogg vorbis. - - z->first_decoded_sample = z->last_decoded_sample - samples; - - // restore file state to where we were - set_file_offset(f, z->page_start); - return 1; - - // restore file state to where we were - bail: - set_file_offset(f, z->page_start); - return 0; -} - -static int vorbis_seek_frame_from_page(stb_vorbis *f, uint32 page_start, uint32 first_sample, uint32 target_sample, int fine) -{ - int left_start, left_end, right_start, right_end, mode,i; - int frame=0; - uint32 frame_start; - int frames_to_skip, data_to_skip; - - // first_sample is the sample # of the first sample that doesn't - // overlap the previous page... note that this requires us to - // _partially_ discard the first packet! bleh. - set_file_offset(f, page_start); - - f->next_seg = -1; // force page resync - - frame_start = first_sample; - // frame start is where the previous packet's last decoded sample - // was, which corresponds to left_end... EXCEPT if the previous - // packet was long and this packet is short? Probably a bug here. - - - // now, we can start decoding frames... we'll only FAKE decode them, - // until we find the frame that contains our sample; then we'll rewind, - // and try again - for (;;) { - int start; - - if (!vorbis_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode)) - return error(f, VORBIS_seek_failed); - - if (frame == 0) - start = left_end; - else - start = left_start; - - // the window starts at left_start; the last valid sample we generate - // before the next frame's window start is right_start-1 - if (target_sample < frame_start + right_start-start) - break; - - flush_packet(f); - if (f->eof) - return error(f, VORBIS_seek_failed); - - frame_start += right_start - start; - - ++frame; - } - - // ok, at this point, the sample we want is contained in frame #'frame' - - // to decode frame #'frame' normally, we have to decode the - // previous frame first... but if it's the FIRST frame of the page - // we can't. if it's the first frame, it means it falls in the part - // of the first frame that doesn't overlap either of the other frames. - // so, if we have to handle that case for the first frame, we might - // as well handle it for all of them, so: - if (target_sample > frame_start + (left_end - left_start)) { - // so what we want to do is go ahead and just immediately decode - // this frame, but then make it so the next get_frame_float() uses - // this already-decoded data? or do we want to go ahead and rewind, - // and leave a flag saying to skip the first N data? let's do that - frames_to_skip = frame; // if this is frame #1, skip 1 frame (#0) - data_to_skip = left_end - left_start; - } else { - // otherwise, we want to skip frames 0, 1, 2, ... frame-2 - // (which means frame-2+1 total frames) then decode frame-1, - // then leave frame pending - frames_to_skip = frame - 1; - assert(frames_to_skip >= 0); - data_to_skip = -1; - } - - set_file_offset(f, page_start); - f->next_seg = - 1; // force page resync - - for (i=0; i < frames_to_skip; ++i) { - maybe_start_packet(f); - flush_packet(f); - } - - if (data_to_skip >= 0) { - int i,j,n = f->blocksize_0 >> 1; - f->discard_samples_deferred = data_to_skip; - for (i=0; i < f->channels; ++i) - for (j=0; j < n; ++j) - f->previous_window[i][j] = 0; - f->previous_length = n; - frame_start += data_to_skip; - } else { - f->previous_length = 0; - vorbis_pump_first_frame(f); - } - - // at this point, the NEXT decoded frame will generate the desired sample - if (fine) { - // so if we're doing sample accurate streaming, we want to go ahead and decode it! - if (target_sample != frame_start) { - int n; - stb_vorbis_get_frame_float(f, &n, NULL); - assert(target_sample > frame_start); - assert(f->channel_buffer_start + (int) (target_sample-frame_start) < f->channel_buffer_end); - f->channel_buffer_start += (target_sample - frame_start); - } - } - - return 0; -} - -static int vorbis_seek_base(stb_vorbis *f, unsigned int sample_number, int fine) -{ - ProbedPage p[2],q; - if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); - - // do we know the location of the last page? - if (f->p_last.page_start == 0) { - uint32 z = stb_vorbis_stream_length_in_samples(f); - if (z == 0) return error(f, VORBIS_cant_find_last_page); - } - - p[0] = f->p_first; - p[1] = f->p_last; - - if (sample_number >= f->p_last.last_decoded_sample) - sample_number = f->p_last.last_decoded_sample-1; - - if (sample_number < f->p_first.last_decoded_sample) { - vorbis_seek_frame_from_page(f, p[0].page_start, 0, sample_number, fine); - return 0; - } else { - int attempts=0; - while (p[0].page_end < p[1].page_start) { - uint32 probe; - uint32 start_offset, end_offset; - uint32 start_sample, end_sample; - - // copy these into local variables so we can tweak them - // if any are unknown - start_offset = p[0].page_end; - end_offset = p[1].after_previous_page_start; // an address known to seek to page p[1] - start_sample = p[0].last_decoded_sample; - end_sample = p[1].last_decoded_sample; - - // currently there is no such tweaking logic needed/possible? - if (start_sample == SAMPLE_unknown || end_sample == SAMPLE_unknown) - return error(f, VORBIS_seek_failed); - - // now we want to lerp between these for the target samples... - - // step 1: we need to bias towards the page start... - if (start_offset + 4000 < end_offset) - end_offset -= 4000; - - // now compute an interpolated search loc - probe = start_offset + (int) floor((float) (end_offset - start_offset) / (end_sample - start_sample) * (sample_number - start_sample)); - - // next we need to bias towards binary search... - // code is a little wonky to allow for full 32-bit unsigned values - if (attempts >= 4) { - uint32 probe2 = start_offset + ((end_offset - start_offset) >> 1); - if (attempts >= 8) - probe = probe2; - else if (probe < probe2) - probe = probe + ((probe2 - probe) >> 1); - else - probe = probe2 + ((probe - probe2) >> 1); - } - ++attempts; - - set_file_offset(f, probe); - if (!vorbis_find_page(f, NULL, NULL)) return error(f, VORBIS_seek_failed); - if (!vorbis_analyze_page(f, &q)) return error(f, VORBIS_seek_failed); - q.after_previous_page_start = probe; - - // it's possible we've just found the last page again - if (q.page_start == p[1].page_start) { - p[1] = q; - continue; - } - - if (sample_number < q.last_decoded_sample) - p[1] = q; - else - p[0] = q; - } - - if (p[0].last_decoded_sample <= sample_number && sample_number < p[1].last_decoded_sample) { - vorbis_seek_frame_from_page(f, p[1].page_start, p[0].last_decoded_sample, sample_number, fine); - return 0; - } - return error(f, VORBIS_seek_failed); - } -} - -int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number) -{ - return vorbis_seek_base(f, sample_number, FALSE); -} - -int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number) -{ - return vorbis_seek_base(f, sample_number, TRUE); -} - -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) { - 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, (int unsigned *)&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, (int unsigned *)&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; - f->p_last.first_decoded_sample = SAMPLE_unknown; - f->p_last.after_previous_page_start = previous_safe; - - 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, stb_vorbis_alloc *alloc, unsigned int length) -{ - stb_vorbis *f, p; - vorbis_init(&p, alloc); - p.f = file; - p.f_start = 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, stb_vorbis_alloc *alloc) -{ - unsigned int len, start; - start = ftell(file); - fseek(file, 0, SEEK_END); - len = 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(char *filename, int *error, 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(unsigned char *data, int len, int *error, stb_vorbis_alloc *alloc) -{ - stb_vorbis *f, p; - if (data == NULL) return NULL; - vorbis_init(&p, alloc); - p.stream = data; - p.stream_end = data + len; - p.stream_start = 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 int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} }; -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], 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(char *filename, int *channels, 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; - 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; - return data_len; -} -#endif // NO_STDIO - -int stb_vorbis_decode_memory(uint8 *mem, int len, int *channels, 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; - 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; - return data_len; -} -#endif - -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+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 - -#endif // STB_VORBIS_HEADER_ONLY +// Ogg Vorbis audio decoder - v1.05 - public domain +// http://nothings.org/stb_vorbis/ +// +// Written by Sean Barrett in 2007, last updated in 2014 +// Sponsored by RAD Game Tools. +// +// Placed in the public domain April 2007 by the author: no copyright +// is claimed, and you may use it for any purpose you like. +// +// 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: +// +// - seeking not supported except manually via PUSHDATA api +// - 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) +// +// 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 +// (If you reported a bug but do not appear in this list, it is because +// someone else reported the bug before you. There were too many of you to +// list them all because I was lax about updating for a long time, sorry.) +// +// Partial history: +// 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 +// 0.99996 - - bracket #include for macintosh compilation +// 0.99995 - - avoid alias-optimization issue in float-to-int conversion +// +// 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 +#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( + unsigned char *datablock, int datablock_length_in_bytes, + int *datablock_memory_consumed_in_bytes, + int *error, + 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, 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, 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, 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, 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, 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); +// NOT WORKING YET +// 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), but it +// actually works + +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. 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. 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. + +// 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 + +// STB_VORBIS_CODEBOOK_SHORTS +// The vorbis file format encodes VQ codebook floats as ax+b where a and +// b are floating point per-codebook constants, and x is a 16-bit int. +// Normally, stb_vorbis decodes them to floats rather than leaving them +// as 16-bit ints and computing ax+b while decoding. This is a speed/space +// tradeoff; you can save space by defining this flag. +#ifndef STB_VORBIS_CODEBOOK_SHORTS +#define STB_VORBIS_CODEBOOK_FLOATS +#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 +#endif + +#ifndef STB_VORBIS_NO_CRT +#include +#include +#include +#include +#if !(defined(__APPLE__) || defined(MACOSX) || defined(macintosh) || defined(Macintosh)) +#include +#endif +#else +#define NULL 0 +#endif + +#if !defined(_MSC_VER) && !(defined(__MINGW32__) && defined(__forceinline)) + #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 + + +#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 + +#ifdef STB_VORBIS_CODEBOOK_FLOATS +typedef float codetype; +#else +typedef uint16 codetype; +#endif + +// @NOTE +// +// Some arrays below are tagged "//varies", which means it's actually +// a variable-sized piece of data, but rather than malloc I assume it's +// small enough it's better to just allocate it all together with the +// main thing +// +// Most of the variables are specified with the smallest size I could pack +// them into. It might give better performance to make them all full-sized +// integers. It should be safe to freely rearrange the structures or change +// the sizes larger--nothing relies on silently truncating etc., nor the +// order of variables. + +#define FAST_HUFFMAN_TABLE_SIZE (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH) +#define FAST_HUFFMAN_TABLE_MASK (FAST_HUFFMAN_TABLE_SIZE - 1) + +typedef struct +{ + int dimensions, entries; + uint8 *codeword_lengths; + float minimum_value; + float delta_value; + uint8 value_bits; + uint8 lookup_type; + uint8 sequence_p; + uint8 sparse; + uint32 lookup_values; + codetype *multiplicands; + uint32 *codewords; + #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT + int16 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; + #else + int32 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; + #endif + uint32 *sorted_codewords; + int *sorted_values; + int sorted_entries; +} Codebook; + +typedef struct +{ + uint8 order; + uint16 rate; + uint16 bark_map_size; + uint8 amplitude_bits; + uint8 amplitude_offset; + uint8 number_of_books; + uint8 book_list[16]; // varies +} Floor0; + +typedef struct +{ + uint8 partitions; + uint8 partition_class_list[32]; // varies + uint8 class_dimensions[16]; // varies + uint8 class_subclasses[16]; // varies + uint8 class_masterbooks[16]; // varies + int16 subclass_books[16][8]; // varies + uint16 Xlist[31*8+2]; // varies + uint8 sorted_order[31*8+2]; + uint8 neighbors[31*8+2][2]; + uint8 floor1_multiplier; + uint8 rangebits; + int values; +} Floor1; + +typedef union +{ + Floor0 floor0; + Floor1 floor1; +} Floor; + +typedef struct +{ + uint32 begin, end; + uint32 part_size; + uint8 classifications; + uint8 classbook; + uint8 **classdata; + int16 (*residue_books)[8]; +} Residue; + +typedef struct +{ + uint8 magnitude; + uint8 angle; + uint8 mux; +} MappingChannel; + +typedef struct +{ + uint16 coupling_steps; + MappingChannel *chan; + uint8 submaps; + uint8 submap_floor[15]; // varies + uint8 submap_residue[15]; // varies +} Mapping; + +typedef struct +{ + uint8 blockflag; + uint8 mapping; + uint16 windowtype; + uint16 transformtype; +} Mode; + +typedef struct +{ + uint32 goal_crc; // expected crc if match + int bytes_left; // bytes left in packet + uint32 crc_so_far; // running crc + int bytes_done; // bytes processed in _current_ chunk + uint32 sample_loc; // granule pos encoded in page +} CRCscan; + +typedef struct +{ + uint32 page_start, page_end; + uint32 after_previous_page_start; + uint32 first_decoded_sample; + 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; +}; + +extern int my_prof(int slot); +//#define stb_prof my_prof + +#ifndef stb_prof +#define stb_prof(x) ((void) 0) +#endif + +#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 not 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=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] = 1 << (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) { assert(0); return FALSE; } + res = available[z]; + available[z] = 0; + add_entry(c, bit_reverse(res), i, m++, len[i], values); + // propogate availability up the tree + if (z != len[i]) { + for (y=len[i]; y > z; --y) { + assert(available[y] == 0); + available[y] = res + (1 << (32-y)); + } + } + } + return TRUE; +} + +// accelerated huffman table allows fast O(1) match of all symbols +// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH +static void compute_accelerated_huffman(Codebook *c) +{ + int i, len; + for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i) + c->fast_huffman[i] = -1; + + len = c->sparse ? c->sorted_entries : c->entries; + #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT + if (len > 32767) len = 32767; // largest possible value we can encode! + #endif + for (i=0; i < len; ++i) { + if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) { + uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i]; + // set table entries for all bit combinations in the higher bits + while (z < FAST_HUFFMAN_TABLE_SIZE) { + c->fast_huffman[z] = i; + z += 1 << c->codeword_lengths[i]; + } + } + } +} + +#ifdef _MSC_VER +#define STBV_CDECL __cdecl +#else +#define STBV_CDECL +#endif + +static int STBV_CDECL uint32_compare(const void *p, const void *q) +{ + uint32 x = * (uint32 *) p; + uint32 y = * (uint32 *) q; + return x < y ? -1 : x > y; +} + +static int include_in_sort(Codebook *c, uint8 len) +{ + if (c->sparse) { assert(len != NO_CODE); return TRUE; } + if (len == NO_CODE) return FALSE; + if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE; + return FALSE; +} + +// if the fast table above doesn't work, we want to binary +// search them... need to reverse the bits +static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values) +{ + int i, len; + // build a list of all the entries + // OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN. + // this is kind of a frivolous optimization--I don't see any performance improvement, + // but it's like 4 extra lines of code, so. + if (!c->sparse) { + int k = 0; + for (i=0; i < c->entries; ++i) + if (include_in_sort(c, lengths[i])) + c->sorted_codewords[k++] = bit_reverse(c->codewords[i]); + assert(k == c->sorted_entries); + } else { + for (i=0; i < c->sorted_entries; ++i) + c->sorted_codewords[i] = bit_reverse(c->codewords[i]); + } + + qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare); + c->sorted_codewords[c->sorted_entries] = 0xffffffff; + + len = c->sparse ? c->sorted_entries : c->entries; + // now we need to indicate how they correspond; we could either + // #1: sort a different data structure that says who they correspond to + // #2: for each sorted entry, search the original list to find who corresponds + // #3: for each original entry, find the sorted entry + // #1 requires extra storage, #2 is slow, #3 can use binary search! + for (i=0; i < len; ++i) { + int huff_len = c->sparse ? lengths[values[i]] : lengths[i]; + if (include_in_sort(c,huff_len)) { + uint32 code = bit_reverse(c->codewords[i]); + int x=0, n=c->sorted_entries; + while (n > 1) { + // invariant: sc[x] <= code < sc[x+n] + int m = x + (n >> 1); + if (c->sorted_codewords[m] <= code) { + x = m; + n -= (n>>1); + } else { + n >>= 1; + } + } + assert(c->sorted_codewords[x] == code); + if (c->sparse) { + c->sorted_values[x] = values[i]; + c->codeword_lengths[x] = huff_len; + } else { + c->sorted_values[x] = i; + } + } + } +} + +// only run while parsing the header (3 times) +static int vorbis_validate(uint8 *data) +{ + static uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' }; + return memcmp(data, vorbis, 6) == 0; +} + +// called from setup only, once per code book +// (formula implied by specification) +static int lookup1_values(int entries, int dim) +{ + int r = (int) floor(exp((float) log((float) entries) / dim)); + if ((int) floor(pow((float) r+1, dim)) <= entries) // (int) cast for MinGW warning; + ++r; // floor() to avoid _ftol() when non-CRT + 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 += 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.after_previous_page_start = p.page_start; + p.first_decoded_sample = 0; + 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 += 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); + + assert(c->sorted_codewords || c->codewords); + // 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 +#ifndef STB_VORBIS_CODEBOOK_FLOATS + #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off] * c->delta_value + c->minimum_value) + #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off] * c->delta_value) + #define CODEBOOK_ELEMENT_BASE(c) (c->minimum_value) +#else + #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off]) + #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off]) + #define CODEBOOK_ELEMENT_BASE(c) (0) +#endif + +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; +} + +#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK +static int codebook_decode_deinterleave_repeat_2(vorb *f, Codebook *c, float **outputs, 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); + + 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*2 + effective > len * 2) { + effective = len*2 - (p_inter*2 - c_inter); + } + + { + z *= c->dimensions; + stb_prof(11); + if (c->sequence_p) { + // haven't optimized this case because I don't have any examples + 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 == 2) { c_inter = 0; ++p_inter; } + last = val; + } + } else { + i=0; + if (c_inter == 1) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + if (outputs[c_inter]) + outputs[c_inter][p_inter] += val; + c_inter = 0; ++p_inter; + ++i; + } + { + float *z0 = outputs[0]; + float *z1 = outputs[1]; + for (; i+1 < effective;) { + float v0 = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + float v1 = CODEBOOK_ELEMENT_FAST(c,z+i+1) + last; + if (z0) + z0[p_inter] += v0; + if (z1) + z1[p_inter] += v1; + ++p_inter; + i += 2; + } + } + if (i < effective) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + if (outputs[c_inter]) + outputs[c_inter][p_inter] += val; + if (++c_inter == 2) { c_inter = 0; ++p_inter; } + } + } + } + + total_decode -= effective; + } + *c_inter_p = c_inter; + *p_inter_p = p_inter; + return TRUE; +} +#endif + +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; + 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 + + stb_prof(2); + 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; + + stb_prof(3); + for (pass=0; pass < 8; ++pass) { + int pcount = 0, class_set = 0; + if (ch == 2) { + stb_prof(13); + 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 + } + stb_prof(5); + 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; + stb_prof(20); // accounts for X time + #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_2(f, book, residue_buffers, &c_inter, &p_inter, n, r->part_size)) + goto done; + #endif + stb_prof(7); + } else { + z += r->part_size; + c_inter = z & 1; + p_inter = z >> 1; + } + } + stb_prof(8); + #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; + stb_prof(22); + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + stb_prof(3); + } 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; + stb_prof(22); + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + stb_prof(3); + } else { + z += r->part_size; + c_inter = z % ch; + p_inter = z / ch; + } + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + ++class_set; + #endif + } + } + } + goto done; + } + stb_prof(9); + + 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: + stb_prof(0); + 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_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]; + draw_line(target, lx,ly, hx,hy, n2); + 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]); + } + return TRUE; +} + +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; + + stb_prof(1); + 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 + } + } + stb_prof(0); + // 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; + } + +// 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); + +// INVERSE COUPLING + stb_prof(14); + 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; + } + } + + // finish decoding the floors +#ifndef STB_VORBIS_NO_DEFER_FLOOR + stb_prof(15); + 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 + stb_prof(16); + for (i=0; i < f->channels; ++i) + inverse_mdct(f->channel_buffers[i], n, f, m->blockflag); + stb_prof(0); + + // 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) { + 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) { + if (current_end < f->current_loc) { + // negative truncation, that's impossible! + *len = 0; + } else { + *len = current_end - f->current_loc; + } + *len += left_start; + 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 + 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; + 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->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; + } 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); + 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; + + 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)); + // 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) { ++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); + + 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; + } + 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; + // 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) { + int z = sparse ? c->sorted_values[j] : j, div=1; + for (k=0; k < c->dimensions; ++k) { + int off = (z / div) % c->lookup_values; + c->multiplicands[j*c->dimensions + k] = + #ifndef STB_VORBIS_CODEBOOK_FLOATS + mults[off]; + #else + mults[off]*c->delta_value + c->minimum_value; + // in this case (and this case only) we could pre-expand c->sequence_p, + // and throw away the decode logic for it; have to ALSO do + // it in the case below, but it can only be done if + // STB_VORBIS_CODEBOOK_FLOATS + // !STB_VORBIS_DIVIDES_IN_CODEBOOK + #endif + div *= c->lookup_values; + } + } + setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); + c->lookup_type = 2; + } + else +#endif + { + c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values); + #ifndef STB_VORBIS_CODEBOOK_FLOATS + memcpy(c->multiplicands, mults, sizeof(c->multiplicands[0]) * c->lookup_values); + #else + for (j=0; j < (int) c->lookup_values; ++j) + c->multiplicands[j] = mults[j] * c->delta_value + c->minimum_value; + #endif + setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); + } +#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK + skip:; +#endif + + #ifdef STB_VORBIS_CODEBOOK_FLOATS + if (c->lookup_type == 2 && c->sequence_p) { + for (j=1; j < (int) c->lookup_values; ++j) + c->multiplicands[j] = c->multiplicands[j-1]; + c->sequence_p = 0; + } + #endif + } + } + + // 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)); + 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)); + 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); + r->part_size = get_bits(f,24)+1; + r->classifications = get_bits(f,6)+1; + r->classbook = get_bits(f,8); + 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); + 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); + 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)); + 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 (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); + #ifdef STB_VORBIS_NO_DEFER_FLOOR + f->floor_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); + #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; + 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) { + 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); + for (i=0; i < p->mapping_count; ++i) + setup_free(p, p->mapping[i].chan); + setup_free(p, p->mapping); + for (i=0; i < p->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, 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 + 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, data, data_len); + } + + f->stream = data; + f->stream_end = 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 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 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 f->stream - data; +} + +stb_vorbis *stb_vorbis_open_pushdata( + unsigned char *data, int data_len, // the memory available for decoding + int *data_used, // only defined if result is not NULL + int *error, stb_vorbis_alloc *alloc) +{ + stb_vorbis *f, p; + vorbis_init(&p, alloc); + p.stream = data; + p.stream_end = 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 = 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 f->stream - f->stream_start; + #ifndef STB_VORBIS_NO_STDIO + return 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 + 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); + } + } +} + +// seek is implemented with 'interpolation search'--this is like +// binary search, but we use the data values to estimate the likely +// location of the data item (plus a bit of a bias so when the +// estimation is wrong we don't waste overly much time) + +#define SAMPLE_unknown 0xffffffff + + +// ogg vorbis, in its insane infinite wisdom, only provides +// information about the sample at the END of the page. +// therefore we COULD have the data we need in the current +// page, and not know it. we could just use the end location +// as our only knowledge for bounds, seek back, and eventually +// the binary search finds it. or we can try to be smart and +// not waste time trying to locate more pages. we try to be +// smart, since this data is already in memory anyway, so +// doing needless I/O would be crazy! +static int vorbis_analyze_page(stb_vorbis *f, ProbedPage *z) +{ + uint8 header[27], lacing[255]; + uint8 packet_type[255]; + int num_packet, packet_start; + int i,len; + uint32 samples; + + // record where the page starts + z->page_start = stb_vorbis_get_file_offset(f); + + // parse the header + getn(f, header, 27); + assert(header[0] == 'O' && header[1] == 'g' && header[2] == 'g' && header[3] == 'S'); + 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] << 16); + + if (header[5] & 4) { + // if this is the last page, it's not possible to work + // backwards to figure out the first sample! whoops! fuck. + z->first_decoded_sample = SAMPLE_unknown; + set_file_offset(f, z->page_start); + return 1; + } + + // scan through the frames to determine the sample-count of each one... + // our goal is the sample # of the first fully-decoded sample on the + // page, which is the first decoded sample of the 2nd packet + + num_packet=0; + + packet_start = ((header[5] & 1) == 0); + + for (i=0; i < header[26]; ++i) { + if (packet_start) { + uint8 n,b; + if (lacing[i] == 0) goto bail; // trying to read from zero-length packet + n = get8(f); + // if bottom bit is non-zero, we've got corruption + if (n & 1) goto bail; + n >>= 1; + b = ilog(f->mode_count-1); + n &= (1 << b)-1; + if (n >= f->mode_count) goto bail; + packet_type[num_packet++] = f->mode_config[n].blockflag; + skip(f, lacing[i]-1); + } else + skip(f, lacing[i]); + packet_start = (lacing[i] < 255); + } + + // now that we know the sizes of all the pages, we can start determining + // how much sample data there is. + + samples = 0; + + // for the last packet, we step by its whole length, because the definition + // is that we encoded the end sample loc of the 'last packet completed', + // where 'completed' refers to packets being split, and we are left to guess + // what 'end sample loc' means. we assume it means ignoring the fact that + // the last half of the data is useless without windowing against the next + // packet... (so it's not REALLY complete in that sense) + if (num_packet > 1) + samples += f->blocksize[packet_type[num_packet-1]]; + + for (i=num_packet-2; i >= 1; --i) { + // now, for this packet, how many samples do we have that + // do not overlap the following packet? + if (packet_type[i] == 1) + if (packet_type[i+1] == 1) + samples += f->blocksize_1 >> 1; + else + samples += ((f->blocksize_1 - f->blocksize_0) >> 2) + (f->blocksize_0 >> 1); + else + samples += f->blocksize_0 >> 1; + } + // now, at this point, we've rewound to the very beginning of the + // _second_ packet. if we entirely discard the first packet after + // a seek, this will be exactly the right sample number. HOWEVER! + // we can't as easily compute this number for the LAST page. The + // only way to get the sample offset of the LAST page is to use + // the end loc from the previous page. But what that returns us + // is _exactly_ the place where we get our first non-overlapped + // sample. (I think. Stupid spec for being ambiguous.) So for + // consistency it's better to do that here, too. However, that + // will then require us to NOT discard all of the first frame we + // decode, in some cases, which means an even weirder frame size + // and extra code. what a fucking pain. + + // we're going to discard the first packet if we + // start the seek here, so we don't care about it. (we could actually + // do better; if the first packet is long, and the previous packet + // is short, there's actually data in the first half of the first + // packet that doesn't need discarding... but not worth paying the + // effort of tracking that of that here and in the seeking logic) + // except crap, if we infer it from the _previous_ packet's end + // location, we DO need to use that definition... and we HAVE to + // infer the start loc of the LAST packet from the previous packet's + // end location. fuck you, ogg vorbis. + + z->first_decoded_sample = z->last_decoded_sample - samples; + + // restore file state to where we were + set_file_offset(f, z->page_start); + return 1; + + // restore file state to where we were + bail: + set_file_offset(f, z->page_start); + return 0; +} + +static int vorbis_seek_frame_from_page(stb_vorbis *f, uint32 page_start, uint32 first_sample, uint32 target_sample, int fine) +{ + int left_start, left_end, right_start, right_end, mode,i; + int frame=0; + uint32 frame_start; + int frames_to_skip, data_to_skip; + + // first_sample is the sample # of the first sample that doesn't + // overlap the previous page... note that this requires us to + // _partially_ discard the first packet! bleh. + set_file_offset(f, page_start); + + f->next_seg = -1; // force page resync + + frame_start = first_sample; + // frame start is where the previous packet's last decoded sample + // was, which corresponds to left_end... EXCEPT if the previous + // packet was long and this packet is short? Probably a bug here. + + + // now, we can start decoding frames... we'll only FAKE decode them, + // until we find the frame that contains our sample; then we'll rewind, + // and try again + for (;;) { + int start; + + if (!vorbis_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode)) + return error(f, VORBIS_seek_failed); + + if (frame == 0) + start = left_end; + else + start = left_start; + + // the window starts at left_start; the last valid sample we generate + // before the next frame's window start is right_start-1 + if (target_sample < frame_start + right_start-start) + break; + + flush_packet(f); + if (f->eof) + return error(f, VORBIS_seek_failed); + + frame_start += right_start - start; + + ++frame; + } + + // ok, at this point, the sample we want is contained in frame #'frame' + + // to decode frame #'frame' normally, we have to decode the + // previous frame first... but if it's the FIRST frame of the page + // we can't. if it's the first frame, it means it falls in the part + // of the first frame that doesn't overlap either of the other frames. + // so, if we have to handle that case for the first frame, we might + // as well handle it for all of them, so: + if (target_sample > frame_start + (left_end - left_start)) { + // so what we want to do is go ahead and just immediately decode + // this frame, but then make it so the next get_frame_float() uses + // this already-decoded data? or do we want to go ahead and rewind, + // and leave a flag saying to skip the first N data? let's do that + frames_to_skip = frame; // if this is frame #1, skip 1 frame (#0) + data_to_skip = left_end - left_start; + } else { + // otherwise, we want to skip frames 0, 1, 2, ... frame-2 + // (which means frame-2+1 total frames) then decode frame-1, + // then leave frame pending + frames_to_skip = frame - 1; + assert(frames_to_skip >= 0); + data_to_skip = -1; + } + + set_file_offset(f, page_start); + f->next_seg = - 1; // force page resync + + for (i=0; i < frames_to_skip; ++i) { + maybe_start_packet(f); + flush_packet(f); + } + + if (data_to_skip >= 0) { + int i,j,n = f->blocksize_0 >> 1; + f->discard_samples_deferred = data_to_skip; + for (i=0; i < f->channels; ++i) + for (j=0; j < n; ++j) + f->previous_window[i][j] = 0; + f->previous_length = n; + frame_start += data_to_skip; + } else { + f->previous_length = 0; + vorbis_pump_first_frame(f); + } + + // at this point, the NEXT decoded frame will generate the desired sample + if (fine) { + // so if we're doing sample accurate streaming, we want to go ahead and decode it! + if (target_sample != frame_start) { + int n; + stb_vorbis_get_frame_float(f, &n, NULL); + assert(target_sample > frame_start); + assert(f->channel_buffer_start + (int) (target_sample-frame_start) < f->channel_buffer_end); + f->channel_buffer_start += (target_sample - frame_start); + } + } + + return 0; +} + +static int vorbis_seek_base(stb_vorbis *f, unsigned int sample_number, int fine) +{ + ProbedPage p[2],q; + if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + + // do we know the location of the last page? + if (f->p_last.page_start == 0) { + uint32 z = stb_vorbis_stream_length_in_samples(f); + if (z == 0) return error(f, VORBIS_cant_find_last_page); + } + + p[0] = f->p_first; + p[1] = f->p_last; + + if (sample_number >= f->p_last.last_decoded_sample) + sample_number = f->p_last.last_decoded_sample-1; + + if (sample_number < f->p_first.last_decoded_sample) { + vorbis_seek_frame_from_page(f, p[0].page_start, 0, sample_number, fine); + return 0; + } else { + int attempts=0; + while (p[0].page_end < p[1].page_start) { + uint32 probe; + uint32 start_offset, end_offset; + uint32 start_sample, end_sample; + + // copy these into local variables so we can tweak them + // if any are unknown + start_offset = p[0].page_end; + end_offset = p[1].after_previous_page_start; // an address known to seek to page p[1] + start_sample = p[0].last_decoded_sample; + end_sample = p[1].last_decoded_sample; + + // currently there is no such tweaking logic needed/possible? + if (start_sample == SAMPLE_unknown || end_sample == SAMPLE_unknown) + return error(f, VORBIS_seek_failed); + + // now we want to lerp between these for the target samples... + + // step 1: we need to bias towards the page start... + if (start_offset + 4000 < end_offset) + end_offset -= 4000; + + // now compute an interpolated search loc + probe = start_offset + (int) floor((float) (end_offset - start_offset) / (end_sample - start_sample) * (sample_number - start_sample)); + + // next we need to bias towards binary search... + // code is a little wonky to allow for full 32-bit unsigned values + if (attempts >= 4) { + uint32 probe2 = start_offset + ((end_offset - start_offset) >> 1); + if (attempts >= 8) + probe = probe2; + else if (probe < probe2) + probe = probe + ((probe2 - probe) >> 1); + else + probe = probe2 + ((probe - probe2) >> 1); + } + ++attempts; + + set_file_offset(f, probe); + if (!vorbis_find_page(f, NULL, NULL)) return error(f, VORBIS_seek_failed); + if (!vorbis_analyze_page(f, &q)) return error(f, VORBIS_seek_failed); + q.after_previous_page_start = probe; + + // it's possible we've just found the last page again + if (q.page_start == p[1].page_start) { + p[1] = q; + continue; + } + + if (sample_number < q.last_decoded_sample) + p[1] = q; + else + p[0] = q; + } + + if (p[0].last_decoded_sample <= sample_number && sample_number < p[1].last_decoded_sample) { + vorbis_seek_frame_from_page(f, p[1].page_start, p[0].last_decoded_sample, sample_number, fine); + return 0; + } + return error(f, VORBIS_seek_failed); + } +} + +int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number) +{ + return vorbis_seek_base(f, sample_number, FALSE); +} + +int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number) +{ + return vorbis_seek_base(f, sample_number, TRUE); +} + +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; + f->p_last.first_decoded_sample = SAMPLE_unknown; + f->p_last.after_previous_page_start = previous_safe; + + 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, stb_vorbis_alloc *alloc, unsigned int length) +{ + stb_vorbis *f, p; + vorbis_init(&p, alloc); + p.f = file; + p.f_start = 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, stb_vorbis_alloc *alloc) +{ + unsigned int len, start; + start = ftell(file); + fseek(file, 0, SEEK_END); + len = 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, 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, 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 + 1.05 - 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 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 diff --git a/internal/c/parts/audio/decode/ogg/src.c b/internal/c/parts/audio/decode/ogg/src.c index fcabf6863..8e6842661 100644 --- a/internal/c/parts/audio/decode/ogg/src.c +++ b/internal/c/parts/audio/decode/ogg/src.c @@ -12,35 +12,35 @@ #endif snd_sequence_struct *snd_decode_ogg(uint8 *buffer,int32 bytes){ - -int result; -int channels; -short *out; -result=stb_vorbis_decode_memory((unsigned char *)buffer,bytes,&channels,&out); -if (result==-1) return NULL; -//extern int stb_vorbis_decode_memory(unsigned char *mem, int len, int *channels, short **output); -// 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. - -//attach to new sequence -static int32 seq_handle; seq_handle=list_add(snd_sequences); -static snd_sequence_struct *seq; seq=(snd_sequence_struct*)list_get(snd_sequences,seq_handle); -memset(seq,0,sizeof(snd_sequence_struct)); -seq->references=1; - -seq->channels=channels; -seq->sample_rate=44100; -seq->bits_per_sample=16; -seq->endian=0;//native -seq->is_unsigned=0; -seq->data=(uint8*)out; -seq->data_size=result*2*channels; - - -return seq; - + + int result; + int channels; + int samplerate; + short *out; + result=stb_vorbis_decode_memory((unsigned char *)buffer,bytes,&channels,&samplerate,&out); + if (result==-1) return NULL; + //extern int stb_vorbis_decode_memory(unsigned char *mem, int len, int *channels, int *sample_rate, short **output); + // 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. + + //attach to new sequence + static int32 seq_handle; seq_handle=list_add(snd_sequences); + static snd_sequence_struct *seq; seq=(snd_sequence_struct*)list_get(snd_sequences,seq_handle); + memset(seq,0,sizeof(snd_sequence_struct)); + seq->references=1; + + seq->channels=channels; + seq->sample_rate=samplerate; + seq->bits_per_sample=16; + seq->endian=0;//native + seq->is_unsigned=0; + seq->data=(uint8*)out; + seq->data_size=result*2*channels; + + + return seq; } diff --git a/internal/c/parts/audio/decode/ogg/src/stb_vorbis.c b/internal/c/parts/audio/decode/ogg/src/stb_vorbis.c index b2d217efa..48d0c508e 100644 --- a/internal/c/parts/audio/decode/ogg/src/stb_vorbis.c +++ b/internal/c/parts/audio/decode/ogg/src/stb_vorbis.c @@ -1,5370 +1,5447 @@ -// Ogg Vorbis I audio decoder -- version 0.99996 -// -// Written in April 2007 by Sean Barrett, sponsored by RAD Game Tools. -// -// Placed in the public domain April 2007 by the author: no copyright is -// claimed, and you may use it for any purpose you like. -// -// 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. -// -// Get the latest version and other information at: -// http://nothings.org/stb_vorbis/ - - -// Todo: -// -// - seeking (note you can seek yourself using the pushdata API) -// -// Limitations: -// -// - floor 0 not supported (used in old ogg vorbis files) -// - 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) -// -// All of these limitations may be removed in future versions. - - -////////////////////////////////////////////////////////////////////////////// -// -// 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 -#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( - unsigned char *datablock, int datablock_length_in_bytes, - int *datablock_memory_consumed_in_bytes, - int *error, - 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, 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(char *filename, int *channels, short **output); -#endif -extern int stb_vorbis_decode_memory(unsigned char *mem, int len, int *channels, short **output); -// 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(unsigned char *data, int len, - int *error, 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(char *filename, - int *error, 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, 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, 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); -// NOT WORKING YET -// 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), but it -// actually works - -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. 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. 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. - -// 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 - -// STB_VORBIS_CODEBOOK_SHORTS -// The vorbis file format encodes VQ codebook floats as ax+b where a and -// b are floating point per-codebook constants, and x is a 16-bit int. -// Normally, stb_vorbis decodes them to floats rather than leaving them -// as 16-bit ints and computing ax+b while decoding. This is a speed/space -// tradeoff; you can save space by defining this flag. -#ifndef STB_VORBIS_CODEBOOK_SHORTS -#define STB_VORBIS_CODEBOOK_FLOATS -#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 -#endif - -#ifndef STB_VORBIS_NO_CRT -#include -#include -#include -#include -#if !(defined(__APPLE__) || defined(MACOSX) || defined(macintosh) || defined(Macintosh)) -#include -#endif -#else -#define NULL 0 -#endif - -#ifndef _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 - - -#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 - -#ifdef STB_VORBIS_CODEBOOK_FLOATS -typedef float codetype; -#else -typedef uint16 codetype; -#endif - -// @NOTE -// -// Some arrays below are tagged "//varies", which means it's actually -// a variable-sized piece of data, but rather than malloc I assume it's -// small enough it's better to just allocate it all together with the -// main thing -// -// Most of the variables are specified with the smallest size I could pack -// them into. It might give better performance to make them all full-sized -// integers. It should be safe to freely rearrange the structures or change -// the sizes larger--nothing relies on silently truncating etc., nor the -// order of variables. - -#define FAST_HUFFMAN_TABLE_SIZE (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH) -#define FAST_HUFFMAN_TABLE_MASK (FAST_HUFFMAN_TABLE_SIZE - 1) - -typedef struct -{ - int dimensions, entries; - uint8 *codeword_lengths; - float minimum_value; - float delta_value; - uint8 value_bits; - uint8 lookup_type; - uint8 sequence_p; - uint8 sparse; - uint32 lookup_values; - codetype *multiplicands; - uint32 *codewords; - #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT - int16 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; - #else - int32 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; - #endif - uint32 *sorted_codewords; - int *sorted_values; - int sorted_entries; -} Codebook; - -typedef struct -{ - uint8 order; - uint16 rate; - uint16 bark_map_size; - uint8 amplitude_bits; - uint8 amplitude_offset; - uint8 number_of_books; - uint8 book_list[16]; // varies -} Floor0; - -typedef struct -{ - uint8 partitions; - uint8 partition_class_list[32]; // varies - uint8 class_dimensions[16]; // varies - uint8 class_subclasses[16]; // varies - uint8 class_masterbooks[16]; // varies - int16 subclass_books[16][8]; // varies - uint16 Xlist[31*8+2]; // varies - uint8 sorted_order[31*8+2]; - uint8 neighbors[31*8+2][2]; - uint8 floor1_multiplier; - uint8 rangebits; - int values; -} Floor1; - -typedef union -{ - Floor0 floor0; - Floor1 floor1; -} Floor; - -typedef struct -{ - uint32 begin, end; - uint32 part_size; - uint8 classifications; - uint8 classbook; - uint8 **classdata; - int16 (*residue_books)[8]; -} Residue; - -typedef struct -{ - uint8 magnitude; - uint8 angle; - uint8 mux; -} MappingChannel; - -typedef struct -{ - uint16 coupling_steps; - MappingChannel *chan; - uint8 submaps; - uint8 submap_floor[15]; // varies - uint8 submap_residue[15]; // varies -} Mapping; - -typedef struct -{ - uint8 blockflag; - uint8 mapping; - uint16 windowtype; - uint16 transformtype; -} Mode; - -typedef struct -{ - uint32 goal_crc; // expected crc if match - int bytes_left; // bytes left in packet - uint32 crc_so_far; // running crc - int bytes_done; // bytes processed in _current_ chunk - uint32 sample_loc; // granule pos encoded in page -} CRCscan; - -typedef struct -{ - uint32 page_start, page_end; - uint32 after_previous_page_start; - uint32 first_decoded_sample; - 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; -}; - -extern int my_prof(int slot); -//#define stb_prof my_prof - -#ifndef stb_prof -#define stb_prof(x) 0 -#endif - -#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 not 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, size_t 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=i<<24, j=0; j < 8; ++j) - s = (s << 1) ^ (s >= (1<<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 < (1U << 14)) - if (n < (1U << 4)) return 0 + log2_4[n ]; - else if (n < (1U << 9)) return 5 + log2_4[n >> 5]; - else return 10 + log2_4[n >> 10]; - else if (n < (1U << 24)) - if (n < (1U << 19)) return 15 + log2_4[n >> 15]; - else return 20 + log2_4[n >> 20]; - else if (n < (1U << 29)) return 25 + log2_4[n >> 25]; - else if (n < (1U << 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] = 1 << (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) { assert(0); return FALSE; } - res = available[z]; - available[z] = 0; - add_entry(c, bit_reverse(res), i, m++, len[i], values); - // propogate availability up the tree - if (z != len[i]) { - for (y=len[i]; y > z; --y) { - assert(available[y] == 0); - available[y] = res + (1 << (32-y)); - } - } - } - return TRUE; -} - -// accelerated huffman table allows fast O(1) match of all symbols -// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH -static void compute_accelerated_huffman(Codebook *c) -{ - int i, len; - for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i) - c->fast_huffman[i] = -1; - - len = c->sparse ? c->sorted_entries : c->entries; - #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT - if (len > 32767) len = 32767; // largest possible value we can encode! - #endif - for (i=0; i < len; ++i) { - if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) { - uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i]; - // set table entries for all bit combinations in the higher bits - while (z < FAST_HUFFMAN_TABLE_SIZE) { - c->fast_huffman[z] = i; - z += 1 << c->codeword_lengths[i]; - } - } - } -} - -static int 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; - -int 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 += 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,i; - // 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 != ~0 || loc1 != ~0) { - // 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.after_previous_page_start = p.page_start; - p.first_decoded_sample = 0; - 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) - 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; -} - -static int32 get_bits_signed(vorb *f, int n) -{ - uint32 z = get_bits(f, n); - if (z & (1 << (n-1))) - z += ~((1 << n) - 1); - return (int32) 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 += 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); - - assert(c->sorted_codewords || c->codewords); - // 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; -} - -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); -} - -#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 - -#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 -#ifndef STB_VORBIS_CODEBOOK_FLOATS - #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off] * c->delta_value + c->minimum_value) - #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off] * c->delta_value) - #define CODEBOOK_ELEMENT_BASE(c) (c->minimum_value) -#else - #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off]) - #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off]) - #define CODEBOOK_ELEMENT_BASE(c) (0) -#endif - -static int codebook_decode_start(vorb *f, Codebook *c, int len) -{ - 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,len); - 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,len); - 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; - 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; - 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; - 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; -} - -#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK -static int codebook_decode_deinterleave_repeat_2(vorb *f, Codebook *c, float **outputs, 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); - - 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*2 + effective > len * 2) { - effective = len*2 - (p_inter*2 - c_inter); - } - - { - z *= c->dimensions; - stb_prof(11); - if (c->sequence_p) { - // haven't optimized this case because I don't have any examples - for (i=0; i < effective; ++i) { - float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; - outputs[c_inter][p_inter] += val; - if (++c_inter == 2) { c_inter = 0; ++p_inter; } - last = val; - } - } else { - i=0; - if (c_inter == 1) { - float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; - outputs[c_inter][p_inter] += val; - c_inter = 0; ++p_inter; - ++i; - } - { - float *z0 = outputs[0]; - float *z1 = outputs[1]; - for (; i+1 < effective;) { - z0[p_inter] += CODEBOOK_ELEMENT_FAST(c,z+i) + last; - z1[p_inter] += CODEBOOK_ELEMENT_FAST(c,z+i+1) + last; - ++p_inter; - i += 2; - } - } - if (i < effective) { - float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; - outputs[c_inter][p_inter] += val; - if (++c_inter == 2) { c_inter = 0; ++p_inter; } - } - } - } - - total_decode -= effective; - } - *c_inter_p = c_inter; - *p_inter_p = p_inter; - return TRUE; -} -#endif - -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; - 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 - - stb_prof(2); - for (i=0; i < ch; ++i) - if (!do_not_decode[i]) - memset(residue_buffers[i], 0, sizeof(float) * n); - - if (rtype == 2 && ch != 1) { - int len = ch * n; - for (j=0; j < ch; ++j) - if (!do_not_decode[j]) - break; - if (j == ch) - goto done; - - stb_prof(3); - for (pass=0; pass < 8; ++pass) { - int pcount = 0, class_set = 0; - if (ch == 2) { - stb_prof(13); - 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 - } - stb_prof(5); - 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; - stb_prof(20); // accounts for X time - #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_2(f, book, residue_buffers, &c_inter, &p_inter, n, r->part_size)) - goto done; - #endif - stb_prof(7); - } else { - z += r->part_size; - c_inter = z & 1; - p_inter = z >> 1; - } - } - stb_prof(8); - #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; - stb_prof(22); - if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) - goto done; - stb_prof(3); - } 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; - stb_prof(22); - if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) - goto done; - stb_prof(3); - } else { - z += r->part_size; - c_inter = z % ch; - p_inter = z / ch; - } - } - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - ++class_set; - #endif - } - } - } - goto done; - } - stb_prof(9); - - 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: - stb_prof(0); - 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); -} -#else -// 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]; - //acc += x[j] * cos(M_PI / n * (i + 0.5) * (j + 0.5)); - buffer[i] = acc; - } - free(x); -} - -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 k_off = -8; - 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 n3_4 = n - n4, 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_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]; - draw_line(target, lx,ly, hx,hy, n2); - 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]); - } - return TRUE; -} - -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]; - int window_center; - -// WINDOWING - - n = f->blocksize[m->blockflag]; - window_center = n >> 1; - - map = &f->mapping[m->mapping]; - -// FLOORS - n2 = n >> 1; - - stb_prof(1); - 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 - } - } - stb_prof(0); - // 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; - } - -// RESIDUE DECODE - for (i=0; i < map->submaps; ++i) { - float *residue_buffers[STB_VORBIS_MAX_CHANNELS]; - int r,t; - 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]; - t = f->residue_types[r]; - 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); - -// INVERSE COUPLING - stb_prof(14); - 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; - } - } - - // finish decoding the floors -#ifndef STB_VORBIS_NO_DEFER_FLOOR - stb_prof(15); - 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 - stb_prof(16); - for (i=0; i < f->channels; ++i) - inverse_mdct(f->channel_buffers[i], n, f, m->blockflag); - stb_prof(0); - - // 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) { - 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) { - if (current_end < f->current_loc) { - // negative truncation, that's impossible! - *len = 0; - } else { - *len = current_end - f->current_loc; - } - *len += left_start; - 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 - 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 == f->segment_count) - 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; - 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->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; - } 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); - 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; - //assert(total != 0); - } 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; - - 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)); - // 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) { ++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); - - 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; - } - 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; - // 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) { - int z = sparse ? c->sorted_values[j] : j, div=1; - for (k=0; k < c->dimensions; ++k) { - int off = (z / div) % c->lookup_values; - c->multiplicands[j*c->dimensions + k] = - #ifndef STB_VORBIS_CODEBOOK_FLOATS - mults[off]; - #else - mults[off]*c->delta_value + c->minimum_value; - // in this case (and this case only) we could pre-expand c->sequence_p, - // and throw away the decode logic for it; have to ALSO do - // it in the case below, but it can only be done if - // STB_VORBIS_CODEBOOK_FLOATS - // !STB_VORBIS_DIVIDES_IN_CODEBOOK - #endif - div *= c->lookup_values; - } - } - setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); - c->lookup_type = 2; - } - else -#endif - { - c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values); - #ifndef STB_VORBIS_CODEBOOK_FLOATS - memcpy(c->multiplicands, mults, sizeof(c->multiplicands[0]) * c->lookup_values); - #else - for (j=0; j < (int) c->lookup_values; ++j) - c->multiplicands[j] = mults[j] * c->delta_value + c->minimum_value; - setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); - #endif - } - skip:; - - #ifdef STB_VORBIS_CODEBOOK_FLOATS - if (c->lookup_type == 2 && c->sequence_p) { - for (j=1; j < (int) c->lookup_values; ++j) - c->multiplicands[j] = c->multiplicands[j-1]; - c->sequence_p = 0; - } - #endif - } - } - - // 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)); - 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)); - 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); - r->part_size = get_bits(f,24)+1; - r->classifications = get_bits(f,6)+1; - r->classbook = get_bits(f,8); - 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); - 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); - 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)); - 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 (get_bits(f,1)) - m->submaps = get_bits(f,4); - 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); - #ifdef STB_VORBIS_NO_DEFER_FLOOR - f->floor_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); - #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; - 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) { - 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); - for (i=0; i < p->mapping_count; ++i) - setup_free(p, p->mapping[i].chan); - setup_free(p, p->mapping); - for (i=0; i < p->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]); - } - #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, 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 != ~0; - 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 - 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, data, data_len); - } - - f->stream = data; - f->stream_end = 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 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 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 f->stream - data; -} - -stb_vorbis *stb_vorbis_open_pushdata( - unsigned char *data, int data_len, // the memory available for decoding - int *data_used, // only defined if result is not NULL - int *error, stb_vorbis_alloc *alloc) -{ - stb_vorbis *f, p; - vorbis_init(&p, alloc); - p.stream = data; - p.stream_end = 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 = 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 f->stream - f->stream_start; - #ifndef STB_VORBIS_NO_STDIO - return 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 - 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); - } - } -} - -// seek is implemented with 'interpolation search'--this is like -// binary search, but we use the data values to estimate the likely -// location of the data item (plus a bit of a bias so when the -// estimation is wrong we don't waste overly much time) - -#define SAMPLE_unknown 0xffffffff - - -// ogg vorbis, in its insane infinite wisdom, only provides -// information about the sample at the END of the page. -// therefore we COULD have the data we need in the current -// page, and not know it. we could just use the end location -// as our only knowledge for bounds, seek back, and eventually -// the binary search finds it. or we can try to be smart and -// not waste time trying to locate more pages. we try to be -// smart, since this data is already in memory anyway, so -// doing needless I/O would be crazy! -static int vorbis_analyze_page(stb_vorbis *f, ProbedPage *z) -{ - uint8 header[27], lacing[255]; - uint8 packet_type[255]; - int num_packet, packet_start, previous =0; - int i,len; - uint32 samples; - - // record where the page starts - z->page_start = stb_vorbis_get_file_offset(f); - - // parse the header - getn(f, header, 27); - assert(header[0] == 'O' && header[1] == 'g' && header[2] == 'g' && header[3] == 'S'); - 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] << 16); - - if (header[5] & 4) { - // if this is the last page, it's not possible to work - // backwards to figure out the first sample! whoops! fuck. - z->first_decoded_sample = SAMPLE_unknown; - set_file_offset(f, z->page_start); - return 1; - } - - // scan through the frames to determine the sample-count of each one... - // our goal is the sample # of the first fully-decoded sample on the - // page, which is the first decoded sample of the 2nd page - - num_packet=0; - - packet_start = ((header[5] & 1) == 0); - - for (i=0; i < header[26]; ++i) { - if (packet_start) { - uint8 n,b,m; - if (lacing[i] == 0) goto bail; // trying to read from zero-length packet - n = get8(f); - // if bottom bit is non-zero, we've got corruption - if (n & 1) goto bail; - n >>= 1; - b = ilog(f->mode_count-1); - m = n >> b; - n &= (1 << b)-1; - if (n >= f->mode_count) goto bail; - if (num_packet == 0 && f->mode_config[n].blockflag) - previous = (m & 1); - packet_type[num_packet++] = f->mode_config[n].blockflag; - skip(f, lacing[i]-1); - } else - skip(f, lacing[i]); - packet_start = (lacing[i] < 255); - } - - // now that we know the sizes of all the pages, we can start determining - // how much sample data there is. - - samples = 0; - - // for the last packet, we step by its whole length, because the definition - // is that we encoded the end sample loc of the 'last packet completed', - // where 'completed' refers to packets being split, and we are left to guess - // what 'end sample loc' means. we assume it means ignoring the fact that - // the last half of the data is useless without windowing against the next - // packet... (so it's not REALLY complete in that sense) - if (num_packet > 1) - samples += f->blocksize[packet_type[num_packet-1]]; - - for (i=num_packet-2; i >= 1; --i) { - // now, for this packet, how many samples do we have that - // do not overlap the following packet? - if (packet_type[i] == 1) - if (packet_type[i+1] == 1) - samples += f->blocksize_1 >> 1; - else - samples += ((f->blocksize_1 - f->blocksize_0) >> 2) + (f->blocksize_0 >> 1); - else - samples += f->blocksize_0 >> 1; - } - // now, at this point, we've rewound to the very beginning of the - // _second_ packet. if we entirely discard the first packet after - // a seek, this will be exactly the right sample number. HOWEVER! - // we can't as easily compute this number for the LAST page. The - // only way to get the sample offset of the LAST page is to use - // the end loc from the previous page. But what that returns us - // is _exactly_ the place where we get our first non-overlapped - // sample. (I think. Stupid spec for being ambiguous.) So for - // consistency it's better to do that here, too. However, that - // will then require us to NOT discard all of the first frame we - // decode, in some cases, which means an even weirder frame size - // and extra code. what a fucking pain. - - // we're going to discard the first packet if we - // start the seek here, so we don't care about it. (we could actually - // do better; if the first packet is long, and the previous packet - // is short, there's actually data in the first half of the first - // packet that doesn't need discarding... but not worth paying the - // effort of tracking that of that here and in the seeking logic) - // except crap, if we infer it from the _previous_ packet's end - // location, we DO need to use that definition... and we HAVE to - // infer the start loc of the LAST packet from the previous packet's - // end location. fuck you, ogg vorbis. - - z->first_decoded_sample = z->last_decoded_sample - samples; - - // restore file state to where we were - set_file_offset(f, z->page_start); - return 1; - - // restore file state to where we were - bail: - set_file_offset(f, z->page_start); - return 0; -} - -static int vorbis_seek_frame_from_page(stb_vorbis *f, uint32 page_start, uint32 first_sample, uint32 target_sample, int fine) -{ - int left_start, left_end, right_start, right_end, mode,i; - int frame=0; - uint32 frame_start; - int frames_to_skip, data_to_skip; - - // first_sample is the sample # of the first sample that doesn't - // overlap the previous page... note that this requires us to - // _partially_ discard the first packet! bleh. - set_file_offset(f, page_start); - - f->next_seg = -1; // force page resync - - frame_start = first_sample; - // frame start is where the previous packet's last decoded sample - // was, which corresponds to left_end... EXCEPT if the previous - // packet was long and this packet is short? Probably a bug here. - - - // now, we can start decoding frames... we'll only FAKE decode them, - // until we find the frame that contains our sample; then we'll rewind, - // and try again - for (;;) { - int start; - - if (!vorbis_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode)) - return error(f, VORBIS_seek_failed); - - if (frame == 0) - start = left_end; - else - start = left_start; - - // the window starts at left_start; the last valid sample we generate - // before the next frame's window start is right_start-1 - if (target_sample < frame_start + right_start-start) - break; - - flush_packet(f); - if (f->eof) - return error(f, VORBIS_seek_failed); - - frame_start += right_start - start; - - ++frame; - } - - // ok, at this point, the sample we want is contained in frame #'frame' - - // to decode frame #'frame' normally, we have to decode the - // previous frame first... but if it's the FIRST frame of the page - // we can't. if it's the first frame, it means it falls in the part - // of the first frame that doesn't overlap either of the other frames. - // so, if we have to handle that case for the first frame, we might - // as well handle it for all of them, so: - if (target_sample > frame_start + (left_end - left_start)) { - // so what we want to do is go ahead and just immediately decode - // this frame, but then make it so the next get_frame_float() uses - // this already-decoded data? or do we want to go ahead and rewind, - // and leave a flag saying to skip the first N data? let's do that - frames_to_skip = frame; // if this is frame #1, skip 1 frame (#0) - data_to_skip = left_end - left_start; - } else { - // otherwise, we want to skip frames 0, 1, 2, ... frame-2 - // (which means frame-2+1 total frames) then decode frame-1, - // then leave frame pending - frames_to_skip = frame - 1; - assert(frames_to_skip >= 0); - data_to_skip = -1; - } - - set_file_offset(f, page_start); - f->next_seg = - 1; // force page resync - - for (i=0; i < frames_to_skip; ++i) { - maybe_start_packet(f); - flush_packet(f); - } - - if (data_to_skip >= 0) { - int i,j,n = f->blocksize_0 >> 1; - f->discard_samples_deferred = data_to_skip; - for (i=0; i < f->channels; ++i) - for (j=0; j < n; ++j) - f->previous_window[i][j] = 0; - f->previous_length = n; - frame_start += data_to_skip; - } else { - f->previous_length = 0; - vorbis_pump_first_frame(f); - } - - // at this point, the NEXT decoded frame will generate the desired sample - if (fine) { - // so if we're doing sample accurate streaming, we want to go ahead and decode it! - if (target_sample != frame_start) { - int n; - stb_vorbis_get_frame_float(f, &n, NULL); - assert(target_sample > frame_start); - assert(f->channel_buffer_start + (int) (target_sample-frame_start) < f->channel_buffer_end); - f->channel_buffer_start += (target_sample - frame_start); - } - } - - return 0; -} - -static int vorbis_seek_base(stb_vorbis *f, unsigned int sample_number, int fine) -{ - ProbedPage p[2],q; - if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); - - // do we know the location of the last page? - if (f->p_last.page_start == 0) { - uint32 z = stb_vorbis_stream_length_in_samples(f); - if (z == 0) return error(f, VORBIS_cant_find_last_page); - } - - p[0] = f->p_first; - p[1] = f->p_last; - - if (sample_number >= f->p_last.last_decoded_sample) - sample_number = f->p_last.last_decoded_sample-1; - - if (sample_number < f->p_first.last_decoded_sample) { - vorbis_seek_frame_from_page(f, p[0].page_start, 0, sample_number, fine); - return 0; - } else { - int attempts=0; - while (p[0].page_end < p[1].page_start) { - uint32 probe; - uint32 start_offset, end_offset; - uint32 start_sample, end_sample; - - // copy these into local variables so we can tweak them - // if any are unknown - start_offset = p[0].page_end; - end_offset = p[1].after_previous_page_start; // an address known to seek to page p[1] - start_sample = p[0].last_decoded_sample; - end_sample = p[1].last_decoded_sample; - - // currently there is no such tweaking logic needed/possible? - if (start_sample == SAMPLE_unknown || end_sample == SAMPLE_unknown) - return error(f, VORBIS_seek_failed); - - // now we want to lerp between these for the target samples... - - // step 1: we need to bias towards the page start... - if (start_offset + 4000 < end_offset) - end_offset -= 4000; - - // now compute an interpolated search loc - probe = start_offset + (int) floor((float) (end_offset - start_offset) / (end_sample - start_sample) * (sample_number - start_sample)); - - // next we need to bias towards binary search... - // code is a little wonky to allow for full 32-bit unsigned values - if (attempts >= 4) { - uint32 probe2 = start_offset + ((end_offset - start_offset) >> 1); - if (attempts >= 8) - probe = probe2; - else if (probe < probe2) - probe = probe + ((probe2 - probe) >> 1); - else - probe = probe2 + ((probe - probe2) >> 1); - } - ++attempts; - - set_file_offset(f, probe); - if (!vorbis_find_page(f, NULL, NULL)) return error(f, VORBIS_seek_failed); - if (!vorbis_analyze_page(f, &q)) return error(f, VORBIS_seek_failed); - q.after_previous_page_start = probe; - - // it's possible we've just found the last page again - if (q.page_start == p[1].page_start) { - p[1] = q; - continue; - } - - if (sample_number < q.last_decoded_sample) - p[1] = q; - else - p[0] = q; - } - - if (p[0].last_decoded_sample <= sample_number && sample_number < p[1].last_decoded_sample) { - vorbis_seek_frame_from_page(f, p[1].page_start, p[0].last_decoded_sample, sample_number, fine); - return 0; - } - return error(f, VORBIS_seek_failed); - } -} - -int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number) -{ - return vorbis_seek_base(f, sample_number, FALSE); -} - -int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number) -{ - return vorbis_seek_base(f, sample_number, TRUE); -} - -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) { - 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, (int unsigned *)&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, (int unsigned *)&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; - f->p_last.first_decoded_sample = SAMPLE_unknown; - f->p_last.after_previous_page_start = previous_safe; - - 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, stb_vorbis_alloc *alloc, unsigned int length) -{ - stb_vorbis *f, p; - vorbis_init(&p, alloc); - p.f = file; - p.f_start = 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, stb_vorbis_alloc *alloc) -{ - unsigned int len, start; - start = ftell(file); - fseek(file, 0, SEEK_END); - len = 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(char *filename, int *error, 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(unsigned char *data, int len, int *error, stb_vorbis_alloc *alloc) -{ - stb_vorbis *f, p; - if (data == NULL) return NULL; - vorbis_init(&p, alloc); - p.stream = data; - p.stream_end = data + len; - p.stream_start = 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 int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} }; -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], 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(char *filename, int *channels, 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; - 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; - return data_len; -} -#endif // NO_STDIO - -int stb_vorbis_decode_memory(uint8 *mem, int len, int *channels, 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; - 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; - return data_len; -} -#endif - -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+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 - -#endif // STB_VORBIS_HEADER_ONLY +// Ogg Vorbis audio decoder - v1.05 - public domain +// http://nothings.org/stb_vorbis/ +// +// Written by Sean Barrett in 2007, last updated in 2014 +// Sponsored by RAD Game Tools. +// +// Placed in the public domain April 2007 by the author: no copyright +// is claimed, and you may use it for any purpose you like. +// +// 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: +// +// - seeking not supported except manually via PUSHDATA api +// - 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) +// +// 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 +// (If you reported a bug but do not appear in this list, it is because +// someone else reported the bug before you. There were too many of you to +// list them all because I was lax about updating for a long time, sorry.) +// +// Partial history: +// 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 +// 0.99996 - - bracket #include for macintosh compilation +// 0.99995 - - avoid alias-optimization issue in float-to-int conversion +// +// 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 +#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( + unsigned char *datablock, int datablock_length_in_bytes, + int *datablock_memory_consumed_in_bytes, + int *error, + 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, 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, 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, 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, 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, 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); +// NOT WORKING YET +// 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), but it +// actually works + +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. 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. 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. + +// 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 + +// STB_VORBIS_CODEBOOK_SHORTS +// The vorbis file format encodes VQ codebook floats as ax+b where a and +// b are floating point per-codebook constants, and x is a 16-bit int. +// Normally, stb_vorbis decodes them to floats rather than leaving them +// as 16-bit ints and computing ax+b while decoding. This is a speed/space +// tradeoff; you can save space by defining this flag. +#ifndef STB_VORBIS_CODEBOOK_SHORTS +#define STB_VORBIS_CODEBOOK_FLOATS +#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 +#endif + +#ifndef STB_VORBIS_NO_CRT +#include +#include +#include +#include +#if !(defined(__APPLE__) || defined(MACOSX) || defined(macintosh) || defined(Macintosh)) +#include +#endif +#else +#define NULL 0 +#endif + +#if 0 +#if !defined(_MSC_VER) && !(defined(__MINGW32__) && defined(__forceinline)) + #if __GNUC__ + #define __forceinline inline + #else + #define __forceinline + #endif +#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 + + +#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 + +#ifdef STB_VORBIS_CODEBOOK_FLOATS +typedef float codetype; +#else +typedef uint16 codetype; +#endif + +// @NOTE +// +// Some arrays below are tagged "//varies", which means it's actually +// a variable-sized piece of data, but rather than malloc I assume it's +// small enough it's better to just allocate it all together with the +// main thing +// +// Most of the variables are specified with the smallest size I could pack +// them into. It might give better performance to make them all full-sized +// integers. It should be safe to freely rearrange the structures or change +// the sizes larger--nothing relies on silently truncating etc., nor the +// order of variables. + +#define FAST_HUFFMAN_TABLE_SIZE (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH) +#define FAST_HUFFMAN_TABLE_MASK (FAST_HUFFMAN_TABLE_SIZE - 1) + +typedef struct +{ + int dimensions, entries; + uint8 *codeword_lengths; + float minimum_value; + float delta_value; + uint8 value_bits; + uint8 lookup_type; + uint8 sequence_p; + uint8 sparse; + uint32 lookup_values; + codetype *multiplicands; + uint32 *codewords; + #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT + int16 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; + #else + int32 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; + #endif + uint32 *sorted_codewords; + int *sorted_values; + int sorted_entries; +} Codebook; + +typedef struct +{ + uint8 order; + uint16 rate; + uint16 bark_map_size; + uint8 amplitude_bits; + uint8 amplitude_offset; + uint8 number_of_books; + uint8 book_list[16]; // varies +} Floor0; + +typedef struct +{ + uint8 partitions; + uint8 partition_class_list[32]; // varies + uint8 class_dimensions[16]; // varies + uint8 class_subclasses[16]; // varies + uint8 class_masterbooks[16]; // varies + int16 subclass_books[16][8]; // varies + uint16 Xlist[31*8+2]; // varies + uint8 sorted_order[31*8+2]; + uint8 neighbors[31*8+2][2]; + uint8 floor1_multiplier; + uint8 rangebits; + int values; +} Floor1; + +typedef union +{ + Floor0 floor0; + Floor1 floor1; +} Floor; + +typedef struct +{ + uint32 begin, end; + uint32 part_size; + uint8 classifications; + uint8 classbook; + uint8 **classdata; + int16 (*residue_books)[8]; +} Residue; + +typedef struct +{ + uint8 magnitude; + uint8 angle; + uint8 mux; +} MappingChannel; + +typedef struct +{ + uint16 coupling_steps; + MappingChannel *chan; + uint8 submaps; + uint8 submap_floor[15]; // varies + uint8 submap_residue[15]; // varies +} Mapping; + +typedef struct +{ + uint8 blockflag; + uint8 mapping; + uint16 windowtype; + uint16 transformtype; +} Mode; + +typedef struct +{ + uint32 goal_crc; // expected crc if match + int bytes_left; // bytes left in packet + uint32 crc_so_far; // running crc + int bytes_done; // bytes processed in _current_ chunk + uint32 sample_loc; // granule pos encoded in page +} CRCscan; + +typedef struct +{ + uint32 page_start, page_end; + uint32 after_previous_page_start; + uint32 first_decoded_sample; + 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; +}; + +extern int my_prof(int slot); +//#define stb_prof my_prof + +#ifndef stb_prof +#define stb_prof(x) ((void) 0) +#endif + +#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 not 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=i<<24, j=0; j < 8; ++j) + s = (s << 1) ^ (s >= (1U<<31) ? CRC32_POLY : 0); + crc_table[i] = s; + } +} + +static inline 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] = 1 << (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) { assert(0); return FALSE; } + res = available[z]; + available[z] = 0; + add_entry(c, bit_reverse(res), i, m++, len[i], values); + // propogate availability up the tree + if (z != len[i]) { + for (y=len[i]; y > z; --y) { + assert(available[y] == 0); + available[y] = res + (1 << (32-y)); + } + } + } + return TRUE; +} + +// accelerated huffman table allows fast O(1) match of all symbols +// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH +static void compute_accelerated_huffman(Codebook *c) +{ + int i, len; + for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i) + c->fast_huffman[i] = -1; + + len = c->sparse ? c->sorted_entries : c->entries; + #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT + if (len > 32767) len = 32767; // largest possible value we can encode! + #endif + for (i=0; i < len; ++i) { + if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) { + uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i]; + // set table entries for all bit combinations in the higher bits + while (z < FAST_HUFFMAN_TABLE_SIZE) { + c->fast_huffman[z] = i; + z += 1 << c->codeword_lengths[i]; + } + } + } +} + +#ifdef _MSC_VER +#define STBV_CDECL __cdecl +#else +#define STBV_CDECL +#endif + +static int STBV_CDECL uint32_compare(const void *p, const void *q) +{ + uint32 x = * (uint32 *) p; + uint32 y = * (uint32 *) q; + return x < y ? -1 : x > y; +} + +static int include_in_sort(Codebook *c, uint8 len) +{ + if (c->sparse) { assert(len != NO_CODE); return TRUE; } + if (len == NO_CODE) return FALSE; + if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE; + return FALSE; +} + +// if the fast table above doesn't work, we want to binary +// search them... need to reverse the bits +static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values) +{ + int i, len; + // build a list of all the entries + // OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN. + // this is kind of a frivolous optimization--I don't see any performance improvement, + // but it's like 4 extra lines of code, so. + if (!c->sparse) { + int k = 0; + for (i=0; i < c->entries; ++i) + if (include_in_sort(c, lengths[i])) + c->sorted_codewords[k++] = bit_reverse(c->codewords[i]); + assert(k == c->sorted_entries); + } else { + for (i=0; i < c->sorted_entries; ++i) + c->sorted_codewords[i] = bit_reverse(c->codewords[i]); + } + + qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare); + c->sorted_codewords[c->sorted_entries] = 0xffffffff; + + len = c->sparse ? c->sorted_entries : c->entries; + // now we need to indicate how they correspond; we could either + // #1: sort a different data structure that says who they correspond to + // #2: for each sorted entry, search the original list to find who corresponds + // #3: for each original entry, find the sorted entry + // #1 requires extra storage, #2 is slow, #3 can use binary search! + for (i=0; i < len; ++i) { + int huff_len = c->sparse ? lengths[values[i]] : lengths[i]; + if (include_in_sort(c,huff_len)) { + uint32 code = bit_reverse(c->codewords[i]); + int x=0, n=c->sorted_entries; + while (n > 1) { + // invariant: sc[x] <= code < sc[x+n] + int m = x + (n >> 1); + if (c->sorted_codewords[m] <= code) { + x = m; + n -= (n>>1); + } else { + n >>= 1; + } + } + assert(c->sorted_codewords[x] == code); + if (c->sparse) { + c->sorted_values[x] = values[i]; + c->codeword_lengths[x] = huff_len; + } else { + c->sorted_values[x] = i; + } + } + } +} + +// only run while parsing the header (3 times) +static int vorbis_validate(uint8 *data) +{ + static uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' }; + return memcmp(data, vorbis, 6) == 0; +} + +// called from setup only, once per code book +// (formula implied by specification) +static int lookup1_values(int entries, int dim) +{ + int r = (int) floor(exp((float) log((float) entries) / dim)); + if ((int) floor(pow((float) r+1, dim)) <= entries) // (int) cast for MinGW warning; + ++r; // floor() to avoid _ftol() when non-CRT + 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 += 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.after_previous_page_start = p.page_start; + p.first_decoded_sample = 0; + 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 inline 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 += 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); + + assert(c->sorted_codewords || c->codewords); + // 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 +#ifndef STB_VORBIS_CODEBOOK_FLOATS + #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off] * c->delta_value + c->minimum_value) + #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off] * c->delta_value) + #define CODEBOOK_ELEMENT_BASE(c) (c->minimum_value) +#else + #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off]) + #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off]) + #define CODEBOOK_ELEMENT_BASE(c) (0) +#endif + +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; +} + +#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK +static int codebook_decode_deinterleave_repeat_2(vorb *f, Codebook *c, float **outputs, 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); + + 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*2 + effective > len * 2) { + effective = len*2 - (p_inter*2 - c_inter); + } + + { + z *= c->dimensions; + stb_prof(11); + if (c->sequence_p) { + // haven't optimized this case because I don't have any examples + 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 == 2) { c_inter = 0; ++p_inter; } + last = val; + } + } else { + i=0; + if (c_inter == 1) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + if (outputs[c_inter]) + outputs[c_inter][p_inter] += val; + c_inter = 0; ++p_inter; + ++i; + } + { + float *z0 = outputs[0]; + float *z1 = outputs[1]; + for (; i+1 < effective;) { + float v0 = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + float v1 = CODEBOOK_ELEMENT_FAST(c,z+i+1) + last; + if (z0) + z0[p_inter] += v0; + if (z1) + z1[p_inter] += v1; + ++p_inter; + i += 2; + } + } + if (i < effective) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + if (outputs[c_inter]) + outputs[c_inter][p_inter] += val; + if (++c_inter == 2) { c_inter = 0; ++p_inter; } + } + } + } + + total_decode -= effective; + } + *c_inter_p = c_inter; + *p_inter_p = p_inter; + return TRUE; +} +#endif + +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 inline 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; + 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 + + stb_prof(2); + 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; + + stb_prof(3); + for (pass=0; pass < 8; ++pass) { + int pcount = 0, class_set = 0; + if (ch == 2) { + stb_prof(13); + 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 + } + stb_prof(5); + 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; + stb_prof(20); // accounts for X time + #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_2(f, book, residue_buffers, &c_inter, &p_inter, n, r->part_size)) + goto done; + #endif + stb_prof(7); + } else { + z += r->part_size; + c_inter = z & 1; + p_inter = z >> 1; + } + } + stb_prof(8); + #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; + stb_prof(22); + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + stb_prof(3); + } 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; + stb_prof(22); + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + stb_prof(3); + } else { + z += r->part_size; + c_inter = z % ch; + p_inter = z / ch; + } + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + ++class_set; + #endif + } + } + } + goto done; + } + stb_prof(9); + + 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: + stb_prof(0); + 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 inline 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_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]; + draw_line(target, lx,ly, hx,hy, n2); + 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]); + } + return TRUE; +} + +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; + + stb_prof(1); + 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 + } + } + stb_prof(0); + // 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; + } + +// 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); + +// INVERSE COUPLING + stb_prof(14); + 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; + } + } + + // finish decoding the floors +#ifndef STB_VORBIS_NO_DEFER_FLOOR + stb_prof(15); + 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 + stb_prof(16); + for (i=0; i < f->channels; ++i) + inverse_mdct(f->channel_buffers[i], n, f, m->blockflag); + stb_prof(0); + + // 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) { + 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) { + if (current_end < f->current_loc) { + // negative truncation, that's impossible! + *len = 0; + } else { + *len = current_end - f->current_loc; + } + *len += left_start; + 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 + 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; + 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->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; + } 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); + 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; + + 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)); + // 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) { ++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); + + 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; + } + 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; + // 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) { + int z = sparse ? c->sorted_values[j] : j, div=1; + for (k=0; k < c->dimensions; ++k) { + int off = (z / div) % c->lookup_values; + c->multiplicands[j*c->dimensions + k] = + #ifndef STB_VORBIS_CODEBOOK_FLOATS + mults[off]; + #else + mults[off]*c->delta_value + c->minimum_value; + // in this case (and this case only) we could pre-expand c->sequence_p, + // and throw away the decode logic for it; have to ALSO do + // it in the case below, but it can only be done if + // STB_VORBIS_CODEBOOK_FLOATS + // !STB_VORBIS_DIVIDES_IN_CODEBOOK + #endif + div *= c->lookup_values; + } + } + setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); + c->lookup_type = 2; + } + else +#endif + { + c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values); + #ifndef STB_VORBIS_CODEBOOK_FLOATS + memcpy(c->multiplicands, mults, sizeof(c->multiplicands[0]) * c->lookup_values); + #else + for (j=0; j < (int) c->lookup_values; ++j) + c->multiplicands[j] = mults[j] * c->delta_value + c->minimum_value; + #endif + setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); + } +#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK + skip:; +#endif + + #ifdef STB_VORBIS_CODEBOOK_FLOATS + if (c->lookup_type == 2 && c->sequence_p) { + for (j=1; j < (int) c->lookup_values; ++j) + c->multiplicands[j] = c->multiplicands[j-1]; + c->sequence_p = 0; + } + #endif + } + } + + // 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)); + 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)); + 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); + r->part_size = get_bits(f,24)+1; + r->classifications = get_bits(f,6)+1; + r->classbook = get_bits(f,8); + 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); + 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); + 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)); + 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 (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); + #ifdef STB_VORBIS_NO_DEFER_FLOOR + f->floor_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); + #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; + 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) { + 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); + for (i=0; i < p->mapping_count; ++i) + setup_free(p, p->mapping[i].chan); + setup_free(p, p->mapping); + for (i=0; i < p->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, 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 + 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, data, data_len); + } + + f->stream = data; + f->stream_end = 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 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 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 f->stream - data; +} + +stb_vorbis *stb_vorbis_open_pushdata( + unsigned char *data, int data_len, // the memory available for decoding + int *data_used, // only defined if result is not NULL + int *error, stb_vorbis_alloc *alloc) +{ + stb_vorbis *f, p; + vorbis_init(&p, alloc); + p.stream = data; + p.stream_end = 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 = 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 f->stream - f->stream_start; + #ifndef STB_VORBIS_NO_STDIO + return 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 + 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); + } + } +} + +// seek is implemented with 'interpolation search'--this is like +// binary search, but we use the data values to estimate the likely +// location of the data item (plus a bit of a bias so when the +// estimation is wrong we don't waste overly much time) + +#define SAMPLE_unknown 0xffffffff + + +// ogg vorbis, in its insane infinite wisdom, only provides +// information about the sample at the END of the page. +// therefore we COULD have the data we need in the current +// page, and not know it. we could just use the end location +// as our only knowledge for bounds, seek back, and eventually +// the binary search finds it. or we can try to be smart and +// not waste time trying to locate more pages. we try to be +// smart, since this data is already in memory anyway, so +// doing needless I/O would be crazy! +static int vorbis_analyze_page(stb_vorbis *f, ProbedPage *z) +{ + uint8 header[27], lacing[255]; + uint8 packet_type[255]; + int num_packet, packet_start; + int i,len; + uint32 samples; + + // record where the page starts + z->page_start = stb_vorbis_get_file_offset(f); + + // parse the header + getn(f, header, 27); + assert(header[0] == 'O' && header[1] == 'g' && header[2] == 'g' && header[3] == 'S'); + 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] << 16); + + if (header[5] & 4) { + // if this is the last page, it's not possible to work + // backwards to figure out the first sample! whoops! fuck. + z->first_decoded_sample = SAMPLE_unknown; + set_file_offset(f, z->page_start); + return 1; + } + + // scan through the frames to determine the sample-count of each one... + // our goal is the sample # of the first fully-decoded sample on the + // page, which is the first decoded sample of the 2nd packet + + num_packet=0; + + packet_start = ((header[5] & 1) == 0); + + for (i=0; i < header[26]; ++i) { + if (packet_start) { + uint8 n,b; + if (lacing[i] == 0) goto bail; // trying to read from zero-length packet + n = get8(f); + // if bottom bit is non-zero, we've got corruption + if (n & 1) goto bail; + n >>= 1; + b = ilog(f->mode_count-1); + n &= (1 << b)-1; + if (n >= f->mode_count) goto bail; + packet_type[num_packet++] = f->mode_config[n].blockflag; + skip(f, lacing[i]-1); + } else + skip(f, lacing[i]); + packet_start = (lacing[i] < 255); + } + + // now that we know the sizes of all the pages, we can start determining + // how much sample data there is. + + samples = 0; + + // for the last packet, we step by its whole length, because the definition + // is that we encoded the end sample loc of the 'last packet completed', + // where 'completed' refers to packets being split, and we are left to guess + // what 'end sample loc' means. we assume it means ignoring the fact that + // the last half of the data is useless without windowing against the next + // packet... (so it's not REALLY complete in that sense) + if (num_packet > 1) + samples += f->blocksize[packet_type[num_packet-1]]; + + for (i=num_packet-2; i >= 1; --i) { + // now, for this packet, how many samples do we have that + // do not overlap the following packet? + if (packet_type[i] == 1) + if (packet_type[i+1] == 1) + samples += f->blocksize_1 >> 1; + else + samples += ((f->blocksize_1 - f->blocksize_0) >> 2) + (f->blocksize_0 >> 1); + else + samples += f->blocksize_0 >> 1; + } + // now, at this point, we've rewound to the very beginning of the + // _second_ packet. if we entirely discard the first packet after + // a seek, this will be exactly the right sample number. HOWEVER! + // we can't as easily compute this number for the LAST page. The + // only way to get the sample offset of the LAST page is to use + // the end loc from the previous page. But what that returns us + // is _exactly_ the place where we get our first non-overlapped + // sample. (I think. Stupid spec for being ambiguous.) So for + // consistency it's better to do that here, too. However, that + // will then require us to NOT discard all of the first frame we + // decode, in some cases, which means an even weirder frame size + // and extra code. what a fucking pain. + + // we're going to discard the first packet if we + // start the seek here, so we don't care about it. (we could actually + // do better; if the first packet is long, and the previous packet + // is short, there's actually data in the first half of the first + // packet that doesn't need discarding... but not worth paying the + // effort of tracking that of that here and in the seeking logic) + // except crap, if we infer it from the _previous_ packet's end + // location, we DO need to use that definition... and we HAVE to + // infer the start loc of the LAST packet from the previous packet's + // end location. fuck you, ogg vorbis. + + z->first_decoded_sample = z->last_decoded_sample - samples; + + // restore file state to where we were + set_file_offset(f, z->page_start); + return 1; + + // restore file state to where we were + bail: + set_file_offset(f, z->page_start); + return 0; +} + +static int vorbis_seek_frame_from_page(stb_vorbis *f, uint32 page_start, uint32 first_sample, uint32 target_sample, int fine) +{ + int left_start, left_end, right_start, right_end, mode,i; + int frame=0; + uint32 frame_start; + int frames_to_skip, data_to_skip; + + // first_sample is the sample # of the first sample that doesn't + // overlap the previous page... note that this requires us to + // _partially_ discard the first packet! bleh. + set_file_offset(f, page_start); + + f->next_seg = -1; // force page resync + + frame_start = first_sample; + // frame start is where the previous packet's last decoded sample + // was, which corresponds to left_end... EXCEPT if the previous + // packet was long and this packet is short? Probably a bug here. + + + // now, we can start decoding frames... we'll only FAKE decode them, + // until we find the frame that contains our sample; then we'll rewind, + // and try again + for (;;) { + int start; + + if (!vorbis_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode)) + return error(f, VORBIS_seek_failed); + + if (frame == 0) + start = left_end; + else + start = left_start; + + // the window starts at left_start; the last valid sample we generate + // before the next frame's window start is right_start-1 + if (target_sample < frame_start + right_start-start) + break; + + flush_packet(f); + if (f->eof) + return error(f, VORBIS_seek_failed); + + frame_start += right_start - start; + + ++frame; + } + + // ok, at this point, the sample we want is contained in frame #'frame' + + // to decode frame #'frame' normally, we have to decode the + // previous frame first... but if it's the FIRST frame of the page + // we can't. if it's the first frame, it means it falls in the part + // of the first frame that doesn't overlap either of the other frames. + // so, if we have to handle that case for the first frame, we might + // as well handle it for all of them, so: + if (target_sample > frame_start + (left_end - left_start)) { + // so what we want to do is go ahead and just immediately decode + // this frame, but then make it so the next get_frame_float() uses + // this already-decoded data? or do we want to go ahead and rewind, + // and leave a flag saying to skip the first N data? let's do that + frames_to_skip = frame; // if this is frame #1, skip 1 frame (#0) + data_to_skip = left_end - left_start; + } else { + // otherwise, we want to skip frames 0, 1, 2, ... frame-2 + // (which means frame-2+1 total frames) then decode frame-1, + // then leave frame pending + frames_to_skip = frame - 1; + assert(frames_to_skip >= 0); + data_to_skip = -1; + } + + set_file_offset(f, page_start); + f->next_seg = - 1; // force page resync + + for (i=0; i < frames_to_skip; ++i) { + maybe_start_packet(f); + flush_packet(f); + } + + if (data_to_skip >= 0) { + int i,j,n = f->blocksize_0 >> 1; + f->discard_samples_deferred = data_to_skip; + for (i=0; i < f->channels; ++i) + for (j=0; j < n; ++j) + f->previous_window[i][j] = 0; + f->previous_length = n; + frame_start += data_to_skip; + } else { + f->previous_length = 0; + vorbis_pump_first_frame(f); + } + + // at this point, the NEXT decoded frame will generate the desired sample + if (fine) { + // so if we're doing sample accurate streaming, we want to go ahead and decode it! + if (target_sample != frame_start) { + int n; + stb_vorbis_get_frame_float(f, &n, NULL); + assert(target_sample > frame_start); + assert(f->channel_buffer_start + (int) (target_sample-frame_start) < f->channel_buffer_end); + f->channel_buffer_start += (target_sample - frame_start); + } + } + + return 0; +} + +static int vorbis_seek_base(stb_vorbis *f, unsigned int sample_number, int fine) +{ + ProbedPage p[2],q; + if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + + // do we know the location of the last page? + if (f->p_last.page_start == 0) { + uint32 z = stb_vorbis_stream_length_in_samples(f); + if (z == 0) return error(f, VORBIS_cant_find_last_page); + } + + p[0] = f->p_first; + p[1] = f->p_last; + + if (sample_number >= f->p_last.last_decoded_sample) + sample_number = f->p_last.last_decoded_sample-1; + + if (sample_number < f->p_first.last_decoded_sample) { + vorbis_seek_frame_from_page(f, p[0].page_start, 0, sample_number, fine); + return 0; + } else { + int attempts=0; + while (p[0].page_end < p[1].page_start) { + uint32 probe; + uint32 start_offset, end_offset; + uint32 start_sample, end_sample; + + // copy these into local variables so we can tweak them + // if any are unknown + start_offset = p[0].page_end; + end_offset = p[1].after_previous_page_start; // an address known to seek to page p[1] + start_sample = p[0].last_decoded_sample; + end_sample = p[1].last_decoded_sample; + + // currently there is no such tweaking logic needed/possible? + if (start_sample == SAMPLE_unknown || end_sample == SAMPLE_unknown) + return error(f, VORBIS_seek_failed); + + // now we want to lerp between these for the target samples... + + // step 1: we need to bias towards the page start... + if (start_offset + 4000 < end_offset) + end_offset -= 4000; + + // now compute an interpolated search loc + probe = start_offset + (int) floor((float) (end_offset - start_offset) / (end_sample - start_sample) * (sample_number - start_sample)); + + // next we need to bias towards binary search... + // code is a little wonky to allow for full 32-bit unsigned values + if (attempts >= 4) { + uint32 probe2 = start_offset + ((end_offset - start_offset) >> 1); + if (attempts >= 8) + probe = probe2; + else if (probe < probe2) + probe = probe + ((probe2 - probe) >> 1); + else + probe = probe2 + ((probe - probe2) >> 1); + } + ++attempts; + + set_file_offset(f, probe); + if (!vorbis_find_page(f, NULL, NULL)) return error(f, VORBIS_seek_failed); + if (!vorbis_analyze_page(f, &q)) return error(f, VORBIS_seek_failed); + q.after_previous_page_start = probe; + + // it's possible we've just found the last page again + if (q.page_start == p[1].page_start) { + p[1] = q; + continue; + } + + if (sample_number < q.last_decoded_sample) + p[1] = q; + else + p[0] = q; + } + + if (p[0].last_decoded_sample <= sample_number && sample_number < p[1].last_decoded_sample) { + vorbis_seek_frame_from_page(f, p[1].page_start, p[0].last_decoded_sample, sample_number, fine); + return 0; + } + return error(f, VORBIS_seek_failed); + } +} + +int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number) +{ + return vorbis_seek_base(f, sample_number, FALSE); +} + +int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number) +{ + return vorbis_seek_base(f, sample_number, TRUE); +} + +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; + f->p_last.first_decoded_sample = SAMPLE_unknown; + f->p_last.after_previous_page_start = previous_safe; + + 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, stb_vorbis_alloc *alloc, unsigned int length) +{ + stb_vorbis *f, p; + vorbis_init(&p, alloc); + p.f = file; + p.f_start = 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, stb_vorbis_alloc *alloc) +{ + unsigned int len, start; + start = ftell(file); + fseek(file, 0, SEEK_END); + len = 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, 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, 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 + 1.05 - 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 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