diff --git a/bin/05.ogg b/bin/05.ogg new file mode 100644 index 0000000..2ff42d4 Binary files /dev/null and b/bin/05.ogg differ diff --git a/bin/OpenLara.exe b/bin/OpenLara.exe index de94ff8..84b909d 100644 Binary files a/bin/OpenLara.exe and b/bin/OpenLara.exe differ diff --git a/src/controller.h b/src/controller.h index 98b7386..e37eada 100644 --- a/src/controller.h +++ b/src/controller.h @@ -166,8 +166,9 @@ struct Controller { if (b.chance == 0 || (rand() & 0x7fff) <= b.chance) { uint32 c = level->soundOffsets[b.offset + rand() % ((b.flags & 0xFF) >> 2)]; void *p = &level->soundData[c]; - #ifdef WIN32 - PlaySound((LPSTR)p, NULL, SND_ASYNC | SND_MEMORY); + #ifdef WIN32 + Sound::play(new Stream(p, 1024 * 1024), b.volume / 255.0f, 0.0f, Sound::Flags::PAN); + // PlaySound((LPSTR)p, NULL, SND_ASYNC | SND_MEMORY); #endif } } diff --git a/src/core.h b/src/core.h index 86dd9de..09d4768 100644 --- a/src/core.h +++ b/src/core.h @@ -16,6 +16,7 @@ #include "utils.h" #include "input.h" +#include "sound.h" #ifdef WIN32 #if defined(_MSC_VER) // Visual Studio @@ -119,10 +120,11 @@ namespace Core { GetProcOGL(glBindBuffer); GetProcOGL(glBufferData); #endif + Sound::init(); } void free() { - // + Sound::free(); } void clear(const vec4 &color) { diff --git a/src/debug.h b/src/debug.h index 2d1a6e4..5996322 100644 --- a/src/debug.h +++ b/src/debug.h @@ -525,7 +525,7 @@ namespace Debug { void info(const TR::Level &level, const TR::Entity &entity) { char buf[255]; - sprintf(buf, "DIP = %d, TRI = %d", Core::stats.dips, Core::stats.tris); + sprintf(buf, "DIP = %d, TRI = %d, SND = %d", Core::stats.dips, Core::stats.tris, Sound::channelsCount); Debug::Draw::text(vec2(16, 16), vec4(1.0f), buf); sprintf(buf, "pos = (%d, %d, %d), room = %d", entity.x, entity.y, entity.z, entity.room); Debug::Draw::text(vec2(16, 32), vec4(1.0f), buf); diff --git a/src/game.h b/src/game.h index b55f24a..09636f9 100644 --- a/src/game.h +++ b/src/game.h @@ -14,6 +14,10 @@ namespace Game { Core::init(); Stream stream("LEVEL2_DEMO.PHD"); level = new Level(stream); + + //Sound::play(Sound::openWAD("05_Lara's_Themes.wav"), 1, 1, 0); + Sound::play(new Stream("05.ogg"), 1, 1, 0); + //Sound::play(new Stream("03.mp3"), 1, 1, 0); } void free() { diff --git a/src/libs/minimp3/libc.h b/src/libs/minimp3/libc.h new file mode 100644 index 0000000..d1e2ac2 --- /dev/null +++ b/src/libs/minimp3/libc.h @@ -0,0 +1,90 @@ +// a libc replacement (more or less) for the Microsoft Visual C compiler +// this file is public domain -- do with it whatever you want! +#ifndef __LIBC_H_INCLUDED__ +#define __LIBC_H_INCLUDED__ + +#ifdef _MSC_VER + #define INLINE __forceinline + #define FASTCALL __fastcall + #ifdef NOLIBC + #ifdef MAIN_PROGRAM + int _fltused=0; + #endif + #endif +#else + #define INLINE inline + #define FASTCALL __attribute__((fastcall)) + #include +#endif + +#ifdef _WIN32 + #ifndef WIN32 + #define WIN32 + #endif +#endif +#ifdef WIN32 + #include +#endif + +#if !NEED_MINILIBC + #include + #include + #include +#endif +#include + +#ifndef __int8_t_defined + #define __int8_t_defined + typedef unsigned char uint8_t; + typedef signed char int8_t; + typedef unsigned short uint16_t; + typedef signed short int16_t; + typedef unsigned int uint32_t; + typedef signed int int32_t; + #ifdef _MSC_VER + typedef unsigned __int64 uint64_t; + typedef signed __int64 int64_t; + #else + typedef unsigned long long uint64_t; + typedef signed long long int64_t; + #endif +#endif + +#ifndef NULL + #define NULL 0 +#endif + +#ifndef M_PI + #define M_PI 3.14159265358979 +#endif + +#define libc_malloc malloc +#define libc_calloc calloc +#define libc_realloc realloc +#define libc_free free + +#define libc_memset memset +#define libc_memcpy memcpy +#define libc_memmove memmove + +#if defined(_MSC_VER) && !defined(_DEBUG) +static INLINE double libc_frexp(double x, int *e) { + double res = -9999.999; + unsigned __int64 i = *(unsigned __int64*)(&x); + if (!(i & 0x7F00000000000000UL)) { + *e = 0; + return x; + } + *e = ((i << 1) >> 53) - 1022; + i &= 0x800FFFFFFFFFFFFFUL; + i |= 0x3FF0000000000000UL; + return *(double*)(&i) * 0.5; +} +#else + #define libc_frexp frexp +#endif + +#define libc_exp exp +#define libc_pow pow + +#endif//__LIBC_H_INCLUDED__ diff --git a/src/libs/minimp3/minimp3.cpp b/src/libs/minimp3/minimp3.cpp new file mode 100644 index 0000000..d76e499 --- /dev/null +++ b/src/libs/minimp3/minimp3.cpp @@ -0,0 +1,2666 @@ +/* + * MPEG Audio Layer III decoder + * Copyright (c) 2001, 2002 Fabrice Bellard, + * (c) 2007 Martin J. Fiedler + * + * This file is a stripped-down version of the MPEG Audio decoder from + * the FFmpeg libavcodec library. + * + * FFmpeg and minimp3 are free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg and minimp3 are distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "libc.h" +#include "minimp3.h" + +#define MP3_FRAME_SIZE 1152 +#define MP3_MAX_CODED_FRAME_SIZE 1792 +#define MP3_MAX_CHANNELS 2 +#define SBLIMIT 32 + +#define MP3_STEREO 0 +#define MP3_JSTEREO 1 +#define MP3_DUAL 2 +#define MP3_MONO 3 + +#define SAME_HEADER_MASK \ + (0xffe00000 | (3 << 17) | (0xf << 12) | (3 << 10) | (3 << 19)) + +#define FRAC_BITS 15 +#define WFRAC_BITS 14 + +#define OUT_MAX (32767) +#define OUT_MIN (-32768) +#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15) + +#define MODE_EXT_MS_STEREO 2 +#define MODE_EXT_I_STEREO 1 + +#define FRAC_ONE (1 << FRAC_BITS) +#define FIX(a) ((int)((a) * FRAC_ONE)) +#define FIXR(a) ((int)((a) * FRAC_ONE + 0.5)) +#define FRAC_RND(a) (((a) + (FRAC_ONE/2)) >> FRAC_BITS) +#define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5)) + +#ifndef _MSC_VER + #define MULL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS) + #define MULH(a,b) (((int64_t)(a) * (int64_t)(b)) >> 32) +#else + static INLINE int MULL(int a, int b) { + int res; + __asm { + mov eax, a + imul b + shr eax, 15 + shl edx, 17 + or eax, edx + mov res, eax + } + return res; + } + static INLINE int MULH(int a, int b) { + int res; + __asm { + mov eax, a + imul b + mov res, edx + } + return res; + } +#endif +#define MULS(ra, rb) ((ra) * (rb)) + +#define ISQRT2 FIXR(0.70710678118654752440) + +#define HEADER_SIZE 4 +#define BACKSTEP_SIZE 512 +#define EXTRABYTES 24 + +#define VLC_TYPE int16_t + +//////////////////////////////////////////////////////////////////////////////// + +struct _granule; + +typedef struct _bitstream { + const uint8_t *buffer, *buffer_end; + int index; + int size_in_bits; +} bitstream_t; + +typedef struct _vlc { + int bits; + VLC_TYPE (*table)[2]; ///< code, bits + int table_size, table_allocated; +} vlc_t; + +typedef struct _mp3_context { + uint8_t last_buf[2*BACKSTEP_SIZE + EXTRABYTES]; + int last_buf_size; + int frame_size; + uint32_t free_format_next_header; + int error_protection; + int sample_rate; + int sample_rate_index; + int bit_rate; + bitstream_t gb; + bitstream_t in_gb; + int nb_channels; + int mode; + int mode_ext; + int lsf; + int16_t synth_buf[MP3_MAX_CHANNELS][512 * 2]; + int synth_buf_offset[MP3_MAX_CHANNELS]; + int32_t sb_samples[MP3_MAX_CHANNELS][36][SBLIMIT]; + int32_t mdct_buf[MP3_MAX_CHANNELS][SBLIMIT * 18]; + int dither_state; +} mp3_context_t; + +typedef struct _granule { + uint8_t scfsi; + int part2_3_length; + int big_values; + int global_gain; + int scalefac_compress; + uint8_t block_type; + uint8_t switch_point; + int table_select[3]; + int subblock_gain[3]; + uint8_t scalefac_scale; + uint8_t count1table_select; + int region_size[3]; + int preflag; + int short_start, long_end; + uint8_t scale_factors[40]; + int32_t sb_hybrid[SBLIMIT * 18]; +} granule_t; + +typedef struct _huff_table { + int xsize; + const uint8_t *bits; + const uint16_t *codes; +} huff_table_t; + +static vlc_t huff_vlc[16]; +static vlc_t huff_quad_vlc[2]; +static uint16_t band_index_long[9][23]; +#define TABLE_4_3_SIZE (8191 + 16)*4 +static int8_t *table_4_3_exp; +static uint32_t *table_4_3_value; +static uint32_t exp_table[512]; +static uint32_t expval_table[512][16]; +static int32_t is_table[2][16]; +static int32_t is_table_lsf[2][2][16]; +static int32_t csa_table[8][4]; +static float csa_table_float[8][4]; +static int32_t mdct_win[8][36]; +static int16_t window[512]; + +//////////////////////////////////////////////////////////////////////////////// + +static const uint16_t mp3_bitrate_tab[2][15] = { + {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 }, + {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160} +}; + +static const uint16_t mp3_freq_tab[3] = { 44100, 48000, 32000 }; + +static const int32_t mp3_enwindow[257] = { + 0, -1, -1, -1, -1, -1, -1, -2, + -2, -2, -2, -3, -3, -4, -4, -5, + -5, -6, -7, -7, -8, -9, -10, -11, + -13, -14, -16, -17, -19, -21, -24, -26, + -29, -31, -35, -38, -41, -45, -49, -53, + -58, -63, -68, -73, -79, -85, -91, -97, + -104, -111, -117, -125, -132, -139, -147, -154, + -161, -169, -176, -183, -190, -196, -202, -208, + 213, 218, 222, 225, 227, 228, 228, 227, + 224, 221, 215, 208, 200, 189, 177, 163, + 146, 127, 106, 83, 57, 29, -2, -36, + -72, -111, -153, -197, -244, -294, -347, -401, + -459, -519, -581, -645, -711, -779, -848, -919, + -991, -1064, -1137, -1210, -1283, -1356, -1428, -1498, + -1567, -1634, -1698, -1759, -1817, -1870, -1919, -1962, + -2001, -2032, -2057, -2075, -2085, -2087, -2080, -2063, + 2037, 2000, 1952, 1893, 1822, 1739, 1644, 1535, + 1414, 1280, 1131, 970, 794, 605, 402, 185, + -45, -288, -545, -814, -1095, -1388, -1692, -2006, + -2330, -2663, -3004, -3351, -3705, -4063, -4425, -4788, + -5153, -5517, -5879, -6237, -6589, -6935, -7271, -7597, + -7910, -8209, -8491, -8755, -8998, -9219, -9416, -9585, + -9727, -9838, -9916, -9959, -9966, -9935, -9863, -9750, + -9592, -9389, -9139, -8840, -8492, -8092, -7640, -7134, + 6574, 5959, 5288, 4561, 3776, 2935, 2037, 1082, + 70, -998, -2122, -3300, -4533, -5818, -7154, -8540, + -9975,-11455,-12980,-14548,-16155,-17799,-19478,-21189, +-22929,-24694,-26482,-28289,-30112,-31947,-33791,-35640, +-37489,-39336,-41176,-43006,-44821,-46617,-48390,-50137, +-51853,-53534,-55178,-56778,-58333,-59838,-61289,-62684, +-64019,-65290,-66494,-67629,-68692,-69679,-70590,-71420, +-72169,-72835,-73415,-73908,-74313,-74630,-74856,-74992, + 75038, +}; + +static const uint8_t slen_table[2][16] = { + { 0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4 }, + { 0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3 }, +}; + +static const uint8_t lsf_nsf_table[6][3][4] = { + { { 6, 5, 5, 5 }, { 9, 9, 9, 9 }, { 6, 9, 9, 9 } }, + { { 6, 5, 7, 3 }, { 9, 9, 12, 6 }, { 6, 9, 12, 6 } }, + { { 11, 10, 0, 0 }, { 18, 18, 0, 0 }, { 15, 18, 0, 0 } }, + { { 7, 7, 7, 0 }, { 12, 12, 12, 0 }, { 6, 15, 12, 0 } }, + { { 6, 6, 6, 3 }, { 12, 9, 9, 6 }, { 6, 12, 9, 6 } }, + { { 8, 8, 5, 0 }, { 15, 12, 9, 0 }, { 6, 18, 9, 0 } }, +}; + +static const uint16_t mp3_huffcodes_1[4] = { + 0x0001, 0x0001, 0x0001, 0x0000, +}; + +static const uint8_t mp3_huffbits_1[4] = { + 1, 3, 2, 3, +}; + +static const uint16_t mp3_huffcodes_2[9] = { + 0x0001, 0x0002, 0x0001, 0x0003, 0x0001, 0x0001, 0x0003, 0x0002, + 0x0000, +}; + +static const uint8_t mp3_huffbits_2[9] = { + 1, 3, 6, 3, 3, 5, 5, 5, + 6, +}; + +static const uint16_t mp3_huffcodes_3[9] = { + 0x0003, 0x0002, 0x0001, 0x0001, 0x0001, 0x0001, 0x0003, 0x0002, + 0x0000, +}; + +static const uint8_t mp3_huffbits_3[9] = { + 2, 2, 6, 3, 2, 5, 5, 5, + 6, +}; + +static const uint16_t mp3_huffcodes_5[16] = { + 0x0001, 0x0002, 0x0006, 0x0005, 0x0003, 0x0001, 0x0004, 0x0004, + 0x0007, 0x0005, 0x0007, 0x0001, 0x0006, 0x0001, 0x0001, 0x0000, +}; + +static const uint8_t mp3_huffbits_5[16] = { + 1, 3, 6, 7, 3, 3, 6, 7, + 6, 6, 7, 8, 7, 6, 7, 8, +}; + +static const uint16_t mp3_huffcodes_6[16] = { + 0x0007, 0x0003, 0x0005, 0x0001, 0x0006, 0x0002, 0x0003, 0x0002, + 0x0005, 0x0004, 0x0004, 0x0001, 0x0003, 0x0003, 0x0002, 0x0000, +}; + +static const uint8_t mp3_huffbits_6[16] = { + 3, 3, 5, 7, 3, 2, 4, 5, + 4, 4, 5, 6, 6, 5, 6, 7, +}; + +static const uint16_t mp3_huffcodes_7[36] = { + 0x0001, 0x0002, 0x000a, 0x0013, 0x0010, 0x000a, 0x0003, 0x0003, + 0x0007, 0x000a, 0x0005, 0x0003, 0x000b, 0x0004, 0x000d, 0x0011, + 0x0008, 0x0004, 0x000c, 0x000b, 0x0012, 0x000f, 0x000b, 0x0002, + 0x0007, 0x0006, 0x0009, 0x000e, 0x0003, 0x0001, 0x0006, 0x0004, + 0x0005, 0x0003, 0x0002, 0x0000, +}; + +static const uint8_t mp3_huffbits_7[36] = { + 1, 3, 6, 8, 8, 9, 3, 4, + 6, 7, 7, 8, 6, 5, 7, 8, + 8, 9, 7, 7, 8, 9, 9, 9, + 7, 7, 8, 9, 9, 10, 8, 8, + 9, 10, 10, 10, +}; + +static const uint16_t mp3_huffcodes_8[36] = { + 0x0003, 0x0004, 0x0006, 0x0012, 0x000c, 0x0005, 0x0005, 0x0001, + 0x0002, 0x0010, 0x0009, 0x0003, 0x0007, 0x0003, 0x0005, 0x000e, + 0x0007, 0x0003, 0x0013, 0x0011, 0x000f, 0x000d, 0x000a, 0x0004, + 0x000d, 0x0005, 0x0008, 0x000b, 0x0005, 0x0001, 0x000c, 0x0004, + 0x0004, 0x0001, 0x0001, 0x0000, +}; + +static const uint8_t mp3_huffbits_8[36] = { + 2, 3, 6, 8, 8, 9, 3, 2, + 4, 8, 8, 8, 6, 4, 6, 8, + 8, 9, 8, 8, 8, 9, 9, 10, + 8, 7, 8, 9, 10, 10, 9, 8, + 9, 9, 11, 11, +}; + +static const uint16_t mp3_huffcodes_9[36] = { + 0x0007, 0x0005, 0x0009, 0x000e, 0x000f, 0x0007, 0x0006, 0x0004, + 0x0005, 0x0005, 0x0006, 0x0007, 0x0007, 0x0006, 0x0008, 0x0008, + 0x0008, 0x0005, 0x000f, 0x0006, 0x0009, 0x000a, 0x0005, 0x0001, + 0x000b, 0x0007, 0x0009, 0x0006, 0x0004, 0x0001, 0x000e, 0x0004, + 0x0006, 0x0002, 0x0006, 0x0000, +}; + +static const uint8_t mp3_huffbits_9[36] = { + 3, 3, 5, 6, 8, 9, 3, 3, + 4, 5, 6, 8, 4, 4, 5, 6, + 7, 8, 6, 5, 6, 7, 7, 8, + 7, 6, 7, 7, 8, 9, 8, 7, + 8, 8, 9, 9, +}; + +static const uint16_t mp3_huffcodes_10[64] = { + 0x0001, 0x0002, 0x000a, 0x0017, 0x0023, 0x001e, 0x000c, 0x0011, + 0x0003, 0x0003, 0x0008, 0x000c, 0x0012, 0x0015, 0x000c, 0x0007, + 0x000b, 0x0009, 0x000f, 0x0015, 0x0020, 0x0028, 0x0013, 0x0006, + 0x000e, 0x000d, 0x0016, 0x0022, 0x002e, 0x0017, 0x0012, 0x0007, + 0x0014, 0x0013, 0x0021, 0x002f, 0x001b, 0x0016, 0x0009, 0x0003, + 0x001f, 0x0016, 0x0029, 0x001a, 0x0015, 0x0014, 0x0005, 0x0003, + 0x000e, 0x000d, 0x000a, 0x000b, 0x0010, 0x0006, 0x0005, 0x0001, + 0x0009, 0x0008, 0x0007, 0x0008, 0x0004, 0x0004, 0x0002, 0x0000, +}; + +static const uint8_t mp3_huffbits_10[64] = { + 1, 3, 6, 8, 9, 9, 9, 10, + 3, 4, 6, 7, 8, 9, 8, 8, + 6, 6, 7, 8, 9, 10, 9, 9, + 7, 7, 8, 9, 10, 10, 9, 10, + 8, 8, 9, 10, 10, 10, 10, 10, + 9, 9, 10, 10, 11, 11, 10, 11, + 8, 8, 9, 10, 10, 10, 11, 11, + 9, 8, 9, 10, 10, 11, 11, 11, +}; + +static const uint16_t mp3_huffcodes_11[64] = { + 0x0003, 0x0004, 0x000a, 0x0018, 0x0022, 0x0021, 0x0015, 0x000f, + 0x0005, 0x0003, 0x0004, 0x000a, 0x0020, 0x0011, 0x000b, 0x000a, + 0x000b, 0x0007, 0x000d, 0x0012, 0x001e, 0x001f, 0x0014, 0x0005, + 0x0019, 0x000b, 0x0013, 0x003b, 0x001b, 0x0012, 0x000c, 0x0005, + 0x0023, 0x0021, 0x001f, 0x003a, 0x001e, 0x0010, 0x0007, 0x0005, + 0x001c, 0x001a, 0x0020, 0x0013, 0x0011, 0x000f, 0x0008, 0x000e, + 0x000e, 0x000c, 0x0009, 0x000d, 0x000e, 0x0009, 0x0004, 0x0001, + 0x000b, 0x0004, 0x0006, 0x0006, 0x0006, 0x0003, 0x0002, 0x0000, +}; + +static const uint8_t mp3_huffbits_11[64] = { + 2, 3, 5, 7, 8, 9, 8, 9, + 3, 3, 4, 6, 8, 8, 7, 8, + 5, 5, 6, 7, 8, 9, 8, 8, + 7, 6, 7, 9, 8, 10, 8, 9, + 8, 8, 8, 9, 9, 10, 9, 10, + 8, 8, 9, 10, 10, 11, 10, 11, + 8, 7, 7, 8, 9, 10, 10, 10, + 8, 7, 8, 9, 10, 10, 10, 10, +}; + +static const uint16_t mp3_huffcodes_12[64] = { + 0x0009, 0x0006, 0x0010, 0x0021, 0x0029, 0x0027, 0x0026, 0x001a, + 0x0007, 0x0005, 0x0006, 0x0009, 0x0017, 0x0010, 0x001a, 0x000b, + 0x0011, 0x0007, 0x000b, 0x000e, 0x0015, 0x001e, 0x000a, 0x0007, + 0x0011, 0x000a, 0x000f, 0x000c, 0x0012, 0x001c, 0x000e, 0x0005, + 0x0020, 0x000d, 0x0016, 0x0013, 0x0012, 0x0010, 0x0009, 0x0005, + 0x0028, 0x0011, 0x001f, 0x001d, 0x0011, 0x000d, 0x0004, 0x0002, + 0x001b, 0x000c, 0x000b, 0x000f, 0x000a, 0x0007, 0x0004, 0x0001, + 0x001b, 0x000c, 0x0008, 0x000c, 0x0006, 0x0003, 0x0001, 0x0000, +}; + +static const uint8_t mp3_huffbits_12[64] = { + 4, 3, 5, 7, 8, 9, 9, 9, + 3, 3, 4, 5, 7, 7, 8, 8, + 5, 4, 5, 6, 7, 8, 7, 8, + 6, 5, 6, 6, 7, 8, 8, 8, + 7, 6, 7, 7, 8, 8, 8, 9, + 8, 7, 8, 8, 8, 9, 8, 9, + 8, 7, 7, 8, 8, 9, 9, 10, + 9, 8, 8, 9, 9, 9, 9, 10, +}; + +static const uint16_t mp3_huffcodes_13[256] = { + 0x0001, 0x0005, 0x000e, 0x0015, 0x0022, 0x0033, 0x002e, 0x0047, + 0x002a, 0x0034, 0x0044, 0x0034, 0x0043, 0x002c, 0x002b, 0x0013, + 0x0003, 0x0004, 0x000c, 0x0013, 0x001f, 0x001a, 0x002c, 0x0021, + 0x001f, 0x0018, 0x0020, 0x0018, 0x001f, 0x0023, 0x0016, 0x000e, + 0x000f, 0x000d, 0x0017, 0x0024, 0x003b, 0x0031, 0x004d, 0x0041, + 0x001d, 0x0028, 0x001e, 0x0028, 0x001b, 0x0021, 0x002a, 0x0010, + 0x0016, 0x0014, 0x0025, 0x003d, 0x0038, 0x004f, 0x0049, 0x0040, + 0x002b, 0x004c, 0x0038, 0x0025, 0x001a, 0x001f, 0x0019, 0x000e, + 0x0023, 0x0010, 0x003c, 0x0039, 0x0061, 0x004b, 0x0072, 0x005b, + 0x0036, 0x0049, 0x0037, 0x0029, 0x0030, 0x0035, 0x0017, 0x0018, + 0x003a, 0x001b, 0x0032, 0x0060, 0x004c, 0x0046, 0x005d, 0x0054, + 0x004d, 0x003a, 0x004f, 0x001d, 0x004a, 0x0031, 0x0029, 0x0011, + 0x002f, 0x002d, 0x004e, 0x004a, 0x0073, 0x005e, 0x005a, 0x004f, + 0x0045, 0x0053, 0x0047, 0x0032, 0x003b, 0x0026, 0x0024, 0x000f, + 0x0048, 0x0022, 0x0038, 0x005f, 0x005c, 0x0055, 0x005b, 0x005a, + 0x0056, 0x0049, 0x004d, 0x0041, 0x0033, 0x002c, 0x002b, 0x002a, + 0x002b, 0x0014, 0x001e, 0x002c, 0x0037, 0x004e, 0x0048, 0x0057, + 0x004e, 0x003d, 0x002e, 0x0036, 0x0025, 0x001e, 0x0014, 0x0010, + 0x0035, 0x0019, 0x0029, 0x0025, 0x002c, 0x003b, 0x0036, 0x0051, + 0x0042, 0x004c, 0x0039, 0x0036, 0x0025, 0x0012, 0x0027, 0x000b, + 0x0023, 0x0021, 0x001f, 0x0039, 0x002a, 0x0052, 0x0048, 0x0050, + 0x002f, 0x003a, 0x0037, 0x0015, 0x0016, 0x001a, 0x0026, 0x0016, + 0x0035, 0x0019, 0x0017, 0x0026, 0x0046, 0x003c, 0x0033, 0x0024, + 0x0037, 0x001a, 0x0022, 0x0017, 0x001b, 0x000e, 0x0009, 0x0007, + 0x0022, 0x0020, 0x001c, 0x0027, 0x0031, 0x004b, 0x001e, 0x0034, + 0x0030, 0x0028, 0x0034, 0x001c, 0x0012, 0x0011, 0x0009, 0x0005, + 0x002d, 0x0015, 0x0022, 0x0040, 0x0038, 0x0032, 0x0031, 0x002d, + 0x001f, 0x0013, 0x000c, 0x000f, 0x000a, 0x0007, 0x0006, 0x0003, + 0x0030, 0x0017, 0x0014, 0x0027, 0x0024, 0x0023, 0x0035, 0x0015, + 0x0010, 0x0017, 0x000d, 0x000a, 0x0006, 0x0001, 0x0004, 0x0002, + 0x0010, 0x000f, 0x0011, 0x001b, 0x0019, 0x0014, 0x001d, 0x000b, + 0x0011, 0x000c, 0x0010, 0x0008, 0x0001, 0x0001, 0x0000, 0x0001, +}; + +static const uint8_t mp3_huffbits_13[256] = { + 1, 4, 6, 7, 8, 9, 9, 10, + 9, 10, 11, 11, 12, 12, 13, 13, + 3, 4, 6, 7, 8, 8, 9, 9, + 9, 9, 10, 10, 11, 12, 12, 12, + 6, 6, 7, 8, 9, 9, 10, 10, + 9, 10, 10, 11, 11, 12, 13, 13, + 7, 7, 8, 9, 9, 10, 10, 10, + 10, 11, 11, 11, 11, 12, 13, 13, + 8, 7, 9, 9, 10, 10, 11, 11, + 10, 11, 11, 12, 12, 13, 13, 14, + 9, 8, 9, 10, 10, 10, 11, 11, + 11, 11, 12, 11, 13, 13, 14, 14, + 9, 9, 10, 10, 11, 11, 11, 11, + 11, 12, 12, 12, 13, 13, 14, 14, + 10, 9, 10, 11, 11, 11, 12, 12, + 12, 12, 13, 13, 13, 14, 16, 16, + 9, 8, 9, 10, 10, 11, 11, 12, + 12, 12, 12, 13, 13, 14, 15, 15, + 10, 9, 10, 10, 11, 11, 11, 13, + 12, 13, 13, 14, 14, 14, 16, 15, + 10, 10, 10, 11, 11, 12, 12, 13, + 12, 13, 14, 13, 14, 15, 16, 17, + 11, 10, 10, 11, 12, 12, 12, 12, + 13, 13, 13, 14, 15, 15, 15, 16, + 11, 11, 11, 12, 12, 13, 12, 13, + 14, 14, 15, 15, 15, 16, 16, 16, + 12, 11, 12, 13, 13, 13, 14, 14, + 14, 14, 14, 15, 16, 15, 16, 16, + 13, 12, 12, 13, 13, 13, 15, 14, + 14, 17, 15, 15, 15, 17, 16, 16, + 12, 12, 13, 14, 14, 14, 15, 14, + 15, 15, 16, 16, 19, 18, 19, 16, +}; + +static const uint16_t mp3_huffcodes_15[256] = { + 0x0007, 0x000c, 0x0012, 0x0035, 0x002f, 0x004c, 0x007c, 0x006c, + 0x0059, 0x007b, 0x006c, 0x0077, 0x006b, 0x0051, 0x007a, 0x003f, + 0x000d, 0x0005, 0x0010, 0x001b, 0x002e, 0x0024, 0x003d, 0x0033, + 0x002a, 0x0046, 0x0034, 0x0053, 0x0041, 0x0029, 0x003b, 0x0024, + 0x0013, 0x0011, 0x000f, 0x0018, 0x0029, 0x0022, 0x003b, 0x0030, + 0x0028, 0x0040, 0x0032, 0x004e, 0x003e, 0x0050, 0x0038, 0x0021, + 0x001d, 0x001c, 0x0019, 0x002b, 0x0027, 0x003f, 0x0037, 0x005d, + 0x004c, 0x003b, 0x005d, 0x0048, 0x0036, 0x004b, 0x0032, 0x001d, + 0x0034, 0x0016, 0x002a, 0x0028, 0x0043, 0x0039, 0x005f, 0x004f, + 0x0048, 0x0039, 0x0059, 0x0045, 0x0031, 0x0042, 0x002e, 0x001b, + 0x004d, 0x0025, 0x0023, 0x0042, 0x003a, 0x0034, 0x005b, 0x004a, + 0x003e, 0x0030, 0x004f, 0x003f, 0x005a, 0x003e, 0x0028, 0x0026, + 0x007d, 0x0020, 0x003c, 0x0038, 0x0032, 0x005c, 0x004e, 0x0041, + 0x0037, 0x0057, 0x0047, 0x0033, 0x0049, 0x0033, 0x0046, 0x001e, + 0x006d, 0x0035, 0x0031, 0x005e, 0x0058, 0x004b, 0x0042, 0x007a, + 0x005b, 0x0049, 0x0038, 0x002a, 0x0040, 0x002c, 0x0015, 0x0019, + 0x005a, 0x002b, 0x0029, 0x004d, 0x0049, 0x003f, 0x0038, 0x005c, + 0x004d, 0x0042, 0x002f, 0x0043, 0x0030, 0x0035, 0x0024, 0x0014, + 0x0047, 0x0022, 0x0043, 0x003c, 0x003a, 0x0031, 0x0058, 0x004c, + 0x0043, 0x006a, 0x0047, 0x0036, 0x0026, 0x0027, 0x0017, 0x000f, + 0x006d, 0x0035, 0x0033, 0x002f, 0x005a, 0x0052, 0x003a, 0x0039, + 0x0030, 0x0048, 0x0039, 0x0029, 0x0017, 0x001b, 0x003e, 0x0009, + 0x0056, 0x002a, 0x0028, 0x0025, 0x0046, 0x0040, 0x0034, 0x002b, + 0x0046, 0x0037, 0x002a, 0x0019, 0x001d, 0x0012, 0x000b, 0x000b, + 0x0076, 0x0044, 0x001e, 0x0037, 0x0032, 0x002e, 0x004a, 0x0041, + 0x0031, 0x0027, 0x0018, 0x0010, 0x0016, 0x000d, 0x000e, 0x0007, + 0x005b, 0x002c, 0x0027, 0x0026, 0x0022, 0x003f, 0x0034, 0x002d, + 0x001f, 0x0034, 0x001c, 0x0013, 0x000e, 0x0008, 0x0009, 0x0003, + 0x007b, 0x003c, 0x003a, 0x0035, 0x002f, 0x002b, 0x0020, 0x0016, + 0x0025, 0x0018, 0x0011, 0x000c, 0x000f, 0x000a, 0x0002, 0x0001, + 0x0047, 0x0025, 0x0022, 0x001e, 0x001c, 0x0014, 0x0011, 0x001a, + 0x0015, 0x0010, 0x000a, 0x0006, 0x0008, 0x0006, 0x0002, 0x0000, +}; + +static const uint8_t mp3_huffbits_15[256] = { + 3, 4, 5, 7, 7, 8, 9, 9, + 9, 10, 10, 11, 11, 11, 12, 13, + 4, 3, 5, 6, 7, 7, 8, 8, + 8, 9, 9, 10, 10, 10, 11, 11, + 5, 5, 5, 6, 7, 7, 8, 8, + 8, 9, 9, 10, 10, 11, 11, 11, + 6, 6, 6, 7, 7, 8, 8, 9, + 9, 9, 10, 10, 10, 11, 11, 11, + 7, 6, 7, 7, 8, 8, 9, 9, + 9, 9, 10, 10, 10, 11, 11, 11, + 8, 7, 7, 8, 8, 8, 9, 9, + 9, 9, 10, 10, 11, 11, 11, 12, + 9, 7, 8, 8, 8, 9, 9, 9, + 9, 10, 10, 10, 11, 11, 12, 12, + 9, 8, 8, 9, 9, 9, 9, 10, + 10, 10, 10, 10, 11, 11, 11, 12, + 9, 8, 8, 9, 9, 9, 9, 10, + 10, 10, 10, 11, 11, 12, 12, 12, + 9, 8, 9, 9, 9, 9, 10, 10, + 10, 11, 11, 11, 11, 12, 12, 12, + 10, 9, 9, 9, 10, 10, 10, 10, + 10, 11, 11, 11, 11, 12, 13, 12, + 10, 9, 9, 9, 10, 10, 10, 10, + 11, 11, 11, 11, 12, 12, 12, 13, + 11, 10, 9, 10, 10, 10, 11, 11, + 11, 11, 11, 11, 12, 12, 13, 13, + 11, 10, 10, 10, 10, 11, 11, 11, + 11, 12, 12, 12, 12, 12, 13, 13, + 12, 11, 11, 11, 11, 11, 11, 11, + 12, 12, 12, 12, 13, 13, 12, 13, + 12, 11, 11, 11, 11, 11, 11, 12, + 12, 12, 12, 12, 13, 13, 13, 13, +}; + +static const uint16_t mp3_huffcodes_16[256] = { + 0x0001, 0x0005, 0x000e, 0x002c, 0x004a, 0x003f, 0x006e, 0x005d, + 0x00ac, 0x0095, 0x008a, 0x00f2, 0x00e1, 0x00c3, 0x0178, 0x0011, + 0x0003, 0x0004, 0x000c, 0x0014, 0x0023, 0x003e, 0x0035, 0x002f, + 0x0053, 0x004b, 0x0044, 0x0077, 0x00c9, 0x006b, 0x00cf, 0x0009, + 0x000f, 0x000d, 0x0017, 0x0026, 0x0043, 0x003a, 0x0067, 0x005a, + 0x00a1, 0x0048, 0x007f, 0x0075, 0x006e, 0x00d1, 0x00ce, 0x0010, + 0x002d, 0x0015, 0x0027, 0x0045, 0x0040, 0x0072, 0x0063, 0x0057, + 0x009e, 0x008c, 0x00fc, 0x00d4, 0x00c7, 0x0183, 0x016d, 0x001a, + 0x004b, 0x0024, 0x0044, 0x0041, 0x0073, 0x0065, 0x00b3, 0x00a4, + 0x009b, 0x0108, 0x00f6, 0x00e2, 0x018b, 0x017e, 0x016a, 0x0009, + 0x0042, 0x001e, 0x003b, 0x0038, 0x0066, 0x00b9, 0x00ad, 0x0109, + 0x008e, 0x00fd, 0x00e8, 0x0190, 0x0184, 0x017a, 0x01bd, 0x0010, + 0x006f, 0x0036, 0x0034, 0x0064, 0x00b8, 0x00b2, 0x00a0, 0x0085, + 0x0101, 0x00f4, 0x00e4, 0x00d9, 0x0181, 0x016e, 0x02cb, 0x000a, + 0x0062, 0x0030, 0x005b, 0x0058, 0x00a5, 0x009d, 0x0094, 0x0105, + 0x00f8, 0x0197, 0x018d, 0x0174, 0x017c, 0x0379, 0x0374, 0x0008, + 0x0055, 0x0054, 0x0051, 0x009f, 0x009c, 0x008f, 0x0104, 0x00f9, + 0x01ab, 0x0191, 0x0188, 0x017f, 0x02d7, 0x02c9, 0x02c4, 0x0007, + 0x009a, 0x004c, 0x0049, 0x008d, 0x0083, 0x0100, 0x00f5, 0x01aa, + 0x0196, 0x018a, 0x0180, 0x02df, 0x0167, 0x02c6, 0x0160, 0x000b, + 0x008b, 0x0081, 0x0043, 0x007d, 0x00f7, 0x00e9, 0x00e5, 0x00db, + 0x0189, 0x02e7, 0x02e1, 0x02d0, 0x0375, 0x0372, 0x01b7, 0x0004, + 0x00f3, 0x0078, 0x0076, 0x0073, 0x00e3, 0x00df, 0x018c, 0x02ea, + 0x02e6, 0x02e0, 0x02d1, 0x02c8, 0x02c2, 0x00df, 0x01b4, 0x0006, + 0x00ca, 0x00e0, 0x00de, 0x00da, 0x00d8, 0x0185, 0x0182, 0x017d, + 0x016c, 0x0378, 0x01bb, 0x02c3, 0x01b8, 0x01b5, 0x06c0, 0x0004, + 0x02eb, 0x00d3, 0x00d2, 0x00d0, 0x0172, 0x017b, 0x02de, 0x02d3, + 0x02ca, 0x06c7, 0x0373, 0x036d, 0x036c, 0x0d83, 0x0361, 0x0002, + 0x0179, 0x0171, 0x0066, 0x00bb, 0x02d6, 0x02d2, 0x0166, 0x02c7, + 0x02c5, 0x0362, 0x06c6, 0x0367, 0x0d82, 0x0366, 0x01b2, 0x0000, + 0x000c, 0x000a, 0x0007, 0x000b, 0x000a, 0x0011, 0x000b, 0x0009, + 0x000d, 0x000c, 0x000a, 0x0007, 0x0005, 0x0003, 0x0001, 0x0003, +}; + +static const uint8_t mp3_huffbits_16[256] = { + 1, 4, 6, 8, 9, 9, 10, 10, + 11, 11, 11, 12, 12, 12, 13, 9, + 3, 4, 6, 7, 8, 9, 9, 9, + 10, 10, 10, 11, 12, 11, 12, 8, + 6, 6, 7, 8, 9, 9, 10, 10, + 11, 10, 11, 11, 11, 12, 12, 9, + 8, 7, 8, 9, 9, 10, 10, 10, + 11, 11, 12, 12, 12, 13, 13, 10, + 9, 8, 9, 9, 10, 10, 11, 11, + 11, 12, 12, 12, 13, 13, 13, 9, + 9, 8, 9, 9, 10, 11, 11, 12, + 11, 12, 12, 13, 13, 13, 14, 10, + 10, 9, 9, 10, 11, 11, 11, 11, + 12, 12, 12, 12, 13, 13, 14, 10, + 10, 9, 10, 10, 11, 11, 11, 12, + 12, 13, 13, 13, 13, 15, 15, 10, + 10, 10, 10, 11, 11, 11, 12, 12, + 13, 13, 13, 13, 14, 14, 14, 10, + 11, 10, 10, 11, 11, 12, 12, 13, + 13, 13, 13, 14, 13, 14, 13, 11, + 11, 11, 10, 11, 12, 12, 12, 12, + 13, 14, 14, 14, 15, 15, 14, 10, + 12, 11, 11, 11, 12, 12, 13, 14, + 14, 14, 14, 14, 14, 13, 14, 11, + 12, 12, 12, 12, 12, 13, 13, 13, + 13, 15, 14, 14, 14, 14, 16, 11, + 14, 12, 12, 12, 13, 13, 14, 14, + 14, 16, 15, 15, 15, 17, 15, 11, + 13, 13, 11, 12, 14, 14, 13, 14, + 14, 15, 16, 15, 17, 15, 14, 11, + 9, 8, 8, 9, 9, 10, 10, 10, + 11, 11, 11, 11, 11, 11, 11, 8, +}; + +static const uint16_t mp3_huffcodes_24[256] = { + 0x000f, 0x000d, 0x002e, 0x0050, 0x0092, 0x0106, 0x00f8, 0x01b2, + 0x01aa, 0x029d, 0x028d, 0x0289, 0x026d, 0x0205, 0x0408, 0x0058, + 0x000e, 0x000c, 0x0015, 0x0026, 0x0047, 0x0082, 0x007a, 0x00d8, + 0x00d1, 0x00c6, 0x0147, 0x0159, 0x013f, 0x0129, 0x0117, 0x002a, + 0x002f, 0x0016, 0x0029, 0x004a, 0x0044, 0x0080, 0x0078, 0x00dd, + 0x00cf, 0x00c2, 0x00b6, 0x0154, 0x013b, 0x0127, 0x021d, 0x0012, + 0x0051, 0x0027, 0x004b, 0x0046, 0x0086, 0x007d, 0x0074, 0x00dc, + 0x00cc, 0x00be, 0x00b2, 0x0145, 0x0137, 0x0125, 0x010f, 0x0010, + 0x0093, 0x0048, 0x0045, 0x0087, 0x007f, 0x0076, 0x0070, 0x00d2, + 0x00c8, 0x00bc, 0x0160, 0x0143, 0x0132, 0x011d, 0x021c, 0x000e, + 0x0107, 0x0042, 0x0081, 0x007e, 0x0077, 0x0072, 0x00d6, 0x00ca, + 0x00c0, 0x00b4, 0x0155, 0x013d, 0x012d, 0x0119, 0x0106, 0x000c, + 0x00f9, 0x007b, 0x0079, 0x0075, 0x0071, 0x00d7, 0x00ce, 0x00c3, + 0x00b9, 0x015b, 0x014a, 0x0134, 0x0123, 0x0110, 0x0208, 0x000a, + 0x01b3, 0x0073, 0x006f, 0x006d, 0x00d3, 0x00cb, 0x00c4, 0x00bb, + 0x0161, 0x014c, 0x0139, 0x012a, 0x011b, 0x0213, 0x017d, 0x0011, + 0x01ab, 0x00d4, 0x00d0, 0x00cd, 0x00c9, 0x00c1, 0x00ba, 0x00b1, + 0x00a9, 0x0140, 0x012f, 0x011e, 0x010c, 0x0202, 0x0179, 0x0010, + 0x014f, 0x00c7, 0x00c5, 0x00bf, 0x00bd, 0x00b5, 0x00ae, 0x014d, + 0x0141, 0x0131, 0x0121, 0x0113, 0x0209, 0x017b, 0x0173, 0x000b, + 0x029c, 0x00b8, 0x00b7, 0x00b3, 0x00af, 0x0158, 0x014b, 0x013a, + 0x0130, 0x0122, 0x0115, 0x0212, 0x017f, 0x0175, 0x016e, 0x000a, + 0x028c, 0x015a, 0x00ab, 0x00a8, 0x00a4, 0x013e, 0x0135, 0x012b, + 0x011f, 0x0114, 0x0107, 0x0201, 0x0177, 0x0170, 0x016a, 0x0006, + 0x0288, 0x0142, 0x013c, 0x0138, 0x0133, 0x012e, 0x0124, 0x011c, + 0x010d, 0x0105, 0x0200, 0x0178, 0x0172, 0x016c, 0x0167, 0x0004, + 0x026c, 0x012c, 0x0128, 0x0126, 0x0120, 0x011a, 0x0111, 0x010a, + 0x0203, 0x017c, 0x0176, 0x0171, 0x016d, 0x0169, 0x0165, 0x0002, + 0x0409, 0x0118, 0x0116, 0x0112, 0x010b, 0x0108, 0x0103, 0x017e, + 0x017a, 0x0174, 0x016f, 0x016b, 0x0168, 0x0166, 0x0164, 0x0000, + 0x002b, 0x0014, 0x0013, 0x0011, 0x000f, 0x000d, 0x000b, 0x0009, + 0x0007, 0x0006, 0x0004, 0x0007, 0x0005, 0x0003, 0x0001, 0x0003, +}; + +static const uint8_t mp3_huffbits_24[256] = { + 4, 4, 6, 7, 8, 9, 9, 10, + 10, 11, 11, 11, 11, 11, 12, 9, + 4, 4, 5, 6, 7, 8, 8, 9, + 9, 9, 10, 10, 10, 10, 10, 8, + 6, 5, 6, 7, 7, 8, 8, 9, + 9, 9, 9, 10, 10, 10, 11, 7, + 7, 6, 7, 7, 8, 8, 8, 9, + 9, 9, 9, 10, 10, 10, 10, 7, + 8, 7, 7, 8, 8, 8, 8, 9, + 9, 9, 10, 10, 10, 10, 11, 7, + 9, 7, 8, 8, 8, 8, 9, 9, + 9, 9, 10, 10, 10, 10, 10, 7, + 9, 8, 8, 8, 8, 9, 9, 9, + 9, 10, 10, 10, 10, 10, 11, 7, + 10, 8, 8, 8, 9, 9, 9, 9, + 10, 10, 10, 10, 10, 11, 11, 8, + 10, 9, 9, 9, 9, 9, 9, 9, + 9, 10, 10, 10, 10, 11, 11, 8, + 10, 9, 9, 9, 9, 9, 9, 10, + 10, 10, 10, 10, 11, 11, 11, 8, + 11, 9, 9, 9, 9, 10, 10, 10, + 10, 10, 10, 11, 11, 11, 11, 8, + 11, 10, 9, 9, 9, 10, 10, 10, + 10, 10, 10, 11, 11, 11, 11, 8, + 11, 10, 10, 10, 10, 10, 10, 10, + 10, 10, 11, 11, 11, 11, 11, 8, + 11, 10, 10, 10, 10, 10, 10, 10, + 11, 11, 11, 11, 11, 11, 11, 8, + 12, 10, 10, 10, 10, 10, 10, 11, + 11, 11, 11, 11, 11, 11, 11, 8, + 8, 7, 7, 7, 7, 7, 7, 7, + 7, 7, 7, 8, 8, 8, 8, 4, +}; + +static const huff_table_t mp3_huff_tables[16] = { +{ 1, NULL, NULL }, +{ 2, mp3_huffbits_1, mp3_huffcodes_1 }, +{ 3, mp3_huffbits_2, mp3_huffcodes_2 }, +{ 3, mp3_huffbits_3, mp3_huffcodes_3 }, +{ 4, mp3_huffbits_5, mp3_huffcodes_5 }, +{ 4, mp3_huffbits_6, mp3_huffcodes_6 }, +{ 6, mp3_huffbits_7, mp3_huffcodes_7 }, +{ 6, mp3_huffbits_8, mp3_huffcodes_8 }, +{ 6, mp3_huffbits_9, mp3_huffcodes_9 }, +{ 8, mp3_huffbits_10, mp3_huffcodes_10 }, +{ 8, mp3_huffbits_11, mp3_huffcodes_11 }, +{ 8, mp3_huffbits_12, mp3_huffcodes_12 }, +{ 16, mp3_huffbits_13, mp3_huffcodes_13 }, +{ 16, mp3_huffbits_15, mp3_huffcodes_15 }, +{ 16, mp3_huffbits_16, mp3_huffcodes_16 }, +{ 16, mp3_huffbits_24, mp3_huffcodes_24 }, +}; + +static const uint8_t mp3_huff_data[32][2] = { +{ 0, 0 }, +{ 1, 0 }, +{ 2, 0 }, +{ 3, 0 }, +{ 0, 0 }, +{ 4, 0 }, +{ 5, 0 }, +{ 6, 0 }, +{ 7, 0 }, +{ 8, 0 }, +{ 9, 0 }, +{ 10, 0 }, +{ 11, 0 }, +{ 12, 0 }, +{ 0, 0 }, +{ 13, 0 }, +{ 14, 1 }, +{ 14, 2 }, +{ 14, 3 }, +{ 14, 4 }, +{ 14, 6 }, +{ 14, 8 }, +{ 14, 10 }, +{ 14, 13 }, +{ 15, 4 }, +{ 15, 5 }, +{ 15, 6 }, +{ 15, 7 }, +{ 15, 8 }, +{ 15, 9 }, +{ 15, 11 }, +{ 15, 13 }, +}; + +static const uint8_t mp3_quad_codes[2][16] = { + { 1, 5, 4, 5, 6, 5, 4, 4, 7, 3, 6, 0, 7, 2, 3, 1, }, + { 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0, }, +}; + +static const uint8_t mp3_quad_bits[2][16] = { + { 1, 4, 4, 5, 4, 6, 5, 6, 4, 5, 5, 6, 5, 6, 6, 6, }, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, }, +}; + +static const uint8_t band_size_long[9][22] = { +{ 4, 4, 4, 4, 4, 4, 6, 6, 8, 8, 10, + 12, 16, 20, 24, 28, 34, 42, 50, 54, 76, 158, }, /* 44100 */ +{ 4, 4, 4, 4, 4, 4, 6, 6, 6, 8, 10, + 12, 16, 18, 22, 28, 34, 40, 46, 54, 54, 192, }, /* 48000 */ +{ 4, 4, 4, 4, 4, 4, 6, 6, 8, 10, 12, + 16, 20, 24, 30, 38, 46, 56, 68, 84, 102, 26, }, /* 32000 */ +{ 6, 6, 6, 6, 6, 6, 8, 10, 12, 14, 16, + 20, 24, 28, 32, 38, 46, 52, 60, 68, 58, 54, }, /* 22050 */ +{ 6, 6, 6, 6, 6, 6, 8, 10, 12, 14, 16, + 18, 22, 26, 32, 38, 46, 52, 64, 70, 76, 36, }, /* 24000 */ +{ 6, 6, 6, 6, 6, 6, 8, 10, 12, 14, 16, + 20, 24, 28, 32, 38, 46, 52, 60, 68, 58, 54, }, /* 16000 */ +{ 6, 6, 6, 6, 6, 6, 8, 10, 12, 14, 16, + 20, 24, 28, 32, 38, 46, 52, 60, 68, 58, 54, }, /* 11025 */ +{ 6, 6, 6, 6, 6, 6, 8, 10, 12, 14, 16, + 20, 24, 28, 32, 38, 46, 52, 60, 68, 58, 54, }, /* 12000 */ +{ 12, 12, 12, 12, 12, 12, 16, 20, 24, 28, 32, + 40, 48, 56, 64, 76, 90, 2, 2, 2, 2, 2, }, /* 8000 */ +}; + +static const uint8_t band_size_short[9][13] = { +{ 4, 4, 4, 4, 6, 8, 10, 12, 14, 18, 22, 30, 56, }, /* 44100 */ +{ 4, 4, 4, 4, 6, 6, 10, 12, 14, 16, 20, 26, 66, }, /* 48000 */ +{ 4, 4, 4, 4, 6, 8, 12, 16, 20, 26, 34, 42, 12, }, /* 32000 */ +{ 4, 4, 4, 6, 6, 8, 10, 14, 18, 26, 32, 42, 18, }, /* 22050 */ +{ 4, 4, 4, 6, 8, 10, 12, 14, 18, 24, 32, 44, 12, }, /* 24000 */ +{ 4, 4, 4, 6, 8, 10, 12, 14, 18, 24, 30, 40, 18, }, /* 16000 */ +{ 4, 4, 4, 6, 8, 10, 12, 14, 18, 24, 30, 40, 18, }, /* 11025 */ +{ 4, 4, 4, 6, 8, 10, 12, 14, 18, 24, 30, 40, 18, }, /* 12000 */ +{ 8, 8, 8, 12, 16, 20, 24, 28, 36, 2, 2, 2, 26, }, /* 8000 */ +}; + +static const uint8_t mp3_pretab[2][22] = { + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 2, 2, 3, 3, 3, 2, 0 }, +}; + +static const float ci_table[8] = { + -0.6f, -0.535f, -0.33f, -0.185f, -0.095f, -0.041f, -0.0142f, -0.0037f, +}; + +#define C1 FIXHR(0.98480775301220805936/2) +#define C2 FIXHR(0.93969262078590838405/2) +#define C3 FIXHR(0.86602540378443864676/2) +#define C4 FIXHR(0.76604444311897803520/2) +#define C5 FIXHR(0.64278760968653932632/2) +#define C6 FIXHR(0.5/2) +#define C7 FIXHR(0.34202014332566873304/2) +#define C8 FIXHR(0.17364817766693034885/2) + +static const int icos36[9] = { + FIXR(0.50190991877167369479), + FIXR(0.51763809020504152469), //0 + FIXR(0.55168895948124587824), + FIXR(0.61038729438072803416), + FIXR(0.70710678118654752439), //1 + FIXR(0.87172339781054900991), + FIXR(1.18310079157624925896), + FIXR(1.93185165257813657349), //2 + FIXR(5.73685662283492756461), +}; + +static const int icos36h[9] = { + FIXHR(0.50190991877167369479/2), + FIXHR(0.51763809020504152469/2), //0 + FIXHR(0.55168895948124587824/2), + FIXHR(0.61038729438072803416/2), + FIXHR(0.70710678118654752439/2), //1 + FIXHR(0.87172339781054900991/2), + FIXHR(1.18310079157624925896/4), + FIXHR(1.93185165257813657349/4), //2 +// FIXHR(5.73685662283492756461), +}; + +//////////////////////////////////////////////////////////////////////////////// + +static INLINE int unaligned32_be(const uint8_t *p) +{ + return (((p[0]<<8) | p[1])<<16) | (p[2]<<8) | (p[3]); +} + +#define MIN_CACHE_BITS 25 + +#define NEG_SSR32(a,s) ((( int32_t)(a))>>(32-(s))) +#define NEG_USR32(a,s) (((uint32_t)(a))>>(32-(s))) + +#define OPEN_READER(name, gb) \ + int name##_index= (gb)->index;\ + int name##_cache= 0;\ + +#define CLOSE_READER(name, gb)\ + (gb)->index= name##_index;\ + +#define UPDATE_CACHE(name, gb)\ + name##_cache= unaligned32_be(&((gb)->buffer[name##_index>>3])) << (name##_index&0x07); \ + +#define SKIP_CACHE(name, gb, num)\ + name##_cache <<= (num); + +#define SKIP_COUNTER(name, gb, num)\ + name##_index += (num);\ + +#define SKIP_BITS(name, gb, num)\ + {\ + SKIP_CACHE(name, gb, num)\ + SKIP_COUNTER(name, gb, num)\ + }\ + +#define LAST_SKIP_BITS(name, gb, num) SKIP_COUNTER(name, gb, num) +#define LAST_SKIP_CACHE(name, gb, num) ; + +#define SHOW_UBITS(name, gb, num)\ + NEG_USR32(name##_cache, num) + +#define SHOW_SBITS(name, gb, num)\ + NEG_SSR32(name##_cache, num) + +#define GET_CACHE(name, gb)\ + ((uint32_t)name##_cache) + +static INLINE int get_bits_count(bitstream_t *s){ + return s->index; +} + +static INLINE void skip_bits_long(bitstream_t *s, int n){ + s->index += n; +} +#define skip_bits skip_bits_long + +static void init_get_bits(bitstream_t *s, const uint8_t *buffer, int bit_size) { + int buffer_size= (bit_size+7)>>3; + if(buffer_size < 0 || bit_size < 0) { + buffer_size = bit_size = 0; + buffer = NULL; + } + s->buffer= buffer; + s->size_in_bits= bit_size; + s->buffer_end= buffer + buffer_size; + s->index=0; +} + +static INLINE unsigned int get_bits(bitstream_t *s, int n){ + register int tmp; + OPEN_READER(re, s) + UPDATE_CACHE(re, s) + tmp= SHOW_UBITS(re, s, n); + LAST_SKIP_BITS(re, s, n) + CLOSE_READER(re, s) + return tmp; +} + +static INLINE int get_bitsz(bitstream_t *s, int n) +{ + if (n == 0) + return 0; + else + return get_bits(s, n); +} + +static INLINE unsigned int get_bits1(bitstream_t *s){ + int index= s->index; + uint8_t result= s->buffer[ index>>3 ]; + result<<= (index&0x07); + result>>= 8 - 1; + index++; + s->index= index; + return result; +} + +static INLINE void align_get_bits(bitstream_t *s) +{ + int n= (-get_bits_count(s)) & 7; + if(n) skip_bits(s, n); +} + +#define GET_DATA(v, table, i, wrap, size) \ +{\ + const uint8_t *ptr = (const uint8_t *)table + i * wrap;\ + switch(size) {\ + case 1:\ + v = *(const uint8_t *)ptr;\ + break;\ + case 2:\ + v = *(const uint16_t *)ptr;\ + break;\ + default:\ + v = *(const uint32_t *)ptr;\ + break;\ + }\ +} + +static INLINE int alloc_table(vlc_t *vlc, int size) { + int index; + index = vlc->table_size; + vlc->table_size += size; + if (vlc->table_size > vlc->table_allocated) { + vlc->table_allocated += (1 << vlc->bits); + vlc->table = (VLC_TYPE(*)[2])libc_realloc(vlc->table, sizeof(VLC_TYPE) * 2 * vlc->table_allocated); + if (!vlc->table) + return -1; + } + return index; +} + +static int build_table( + vlc_t *vlc, int table_nb_bits, + int nb_codes, + const void *bits, int bits_wrap, int bits_size, + const void *codes, int codes_wrap, int codes_size, + uint32_t code_prefix, int n_prefix +) { + int i, j, k, n, table_size, table_index, nb, n1, index, code_prefix2; + uint32_t code; + VLC_TYPE (*table)[2]; + + table_size = 1 << table_nb_bits; + table_index = alloc_table(vlc, table_size); + if (table_index < 0) + return -1; + table = &vlc->table[table_index]; + + for(i=0;i> n; + if (n > 0 && code_prefix2 == code_prefix) { + if (n <= table_nb_bits) { + j = (code << (table_nb_bits - n)) & (table_size - 1); + nb = 1 << (table_nb_bits - n); + for(k=0;k> n) & ((1 << table_nb_bits) - 1); + n1 = -table[j][1]; //bits + if (n > n1) + n1 = n; + table[j][1] = -n1; //bits + } + } + } + for(i=0;i table_nb_bits) { + n = table_nb_bits; + table[i][1] = -n; //bits + } + index = build_table(vlc, n, nb_codes, + bits, bits_wrap, bits_size, + codes, codes_wrap, codes_size, + (code_prefix << table_nb_bits) | i, + n_prefix + table_nb_bits); + if (index < 0) + return -1; + table = &vlc->table[table_index]; + table[i][0] = index; //code + } + } + return table_index; +} + +static INLINE int init_vlc( + vlc_t *vlc, int nb_bits, int nb_codes, + const void *bits, int bits_wrap, int bits_size, + const void *codes, int codes_wrap, int codes_size +) { + vlc->bits = nb_bits; + if (build_table(vlc, nb_bits, nb_codes, + bits, bits_wrap, bits_size, + codes, codes_wrap, codes_size, + 0, 0) < 0) { + libc_free(vlc->table); + return -1; + } + return 0; +} + +#define GET_VLC(code, name, gb, table, bits, max_depth)\ +{\ + int n, index, nb_bits;\ +\ + index= SHOW_UBITS(name, gb, bits);\ + code = table[index][0];\ + n = table[index][1];\ +\ + if(max_depth > 1 && n < 0){\ + LAST_SKIP_BITS(name, gb, bits)\ + UPDATE_CACHE(name, gb)\ +\ + nb_bits = -n;\ +\ + index= SHOW_UBITS(name, gb, nb_bits) + code;\ + code = table[index][0];\ + n = table[index][1];\ + if(max_depth > 2 && n < 0){\ + LAST_SKIP_BITS(name, gb, nb_bits)\ + UPDATE_CACHE(name, gb)\ +\ + nb_bits = -n;\ +\ + index= SHOW_UBITS(name, gb, nb_bits) + code;\ + code = table[index][0];\ + n = table[index][1];\ + }\ + }\ + SKIP_BITS(name, gb, n)\ +} + +static INLINE int get_vlc2(bitstream_t *s, VLC_TYPE (*table)[2], int bits, int max_depth) { + int code; + + OPEN_READER(re, s) + UPDATE_CACHE(re, s) + + GET_VLC(code, re, s, table, bits, max_depth) + + CLOSE_READER(re, s) + return code; +} + +static void switch_buffer(mp3_context_t *s, int *pos, int *end_pos, int *end_pos2) { + if(s->in_gb.buffer && *pos >= s->gb.size_in_bits){ + s->gb= s->in_gb; + s->in_gb.buffer=NULL; + skip_bits_long(&s->gb, *pos - *end_pos); + *end_pos2= + *end_pos= *end_pos2 + get_bits_count(&s->gb) - *pos; + *pos= get_bits_count(&s->gb); + } +} + +//////////////////////////////////////////////////////////////////////////////// + +static INLINE int mp3_check_header(uint32_t header){ + /* header */ + if ((header & 0xffe00000) != 0xffe00000) + return -1; + /* layer check */ + if ((header & (3<<17)) != (1 << 17)) + return -1; + /* bit rate */ + if ((header & (0xf<<12)) == 0xf<<12) + return -1; + /* frequency */ + if ((header & (3<<10)) == 3<<10) + return -1; + return 0; +} + + +static void lsf_sf_expand( + int *slen, int sf, int n1, int n2, int n3 +) { + if (n3) { + slen[3] = sf % n3; + sf /= n3; + } else { + slen[3] = 0; + } + if (n2) { + slen[2] = sf % n2; + sf /= n2; + } else { + slen[2] = 0; + } + slen[1] = sf % n1; + sf /= n1; + slen[0] = sf; +} + +static INLINE int l3_unscale(int value, int exponent) +{ + unsigned int m; + int e; + + e = table_4_3_exp [4*value + (exponent&3)]; + m = table_4_3_value[4*value + (exponent&3)]; + e -= (exponent >> 2); + if (e > 31) + return 0; + m = (m + (1 << (e-1))) >> e; + + return m; +} + +static INLINE int round_sample(int *sum) { + int sum1; + sum1 = (*sum) >> OUT_SHIFT; + *sum &= (1< OUT_MAX) + sum1 = OUT_MAX; + return sum1; +} + +static void exponents_from_scale_factors( + mp3_context_t *s, granule_t *g, int16_t *exponents +) { + const uint8_t *bstab, *pretab; + int len, i, j, k, l, v0, shift, gain, gains[3]; + int16_t *exp_ptr; + + exp_ptr = exponents; + gain = g->global_gain - 210; + shift = g->scalefac_scale + 1; + + bstab = band_size_long[s->sample_rate_index]; + pretab = mp3_pretab[g->preflag]; + for(i=0;ilong_end;i++) { + v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400; + len = bstab[i]; + for(j=len;j>0;j--) + *exp_ptr++ = v0; + } + + if (g->short_start < 13) { + bstab = band_size_short[s->sample_rate_index]; + gains[0] = gain - (g->subblock_gain[0] << 3); + gains[1] = gain - (g->subblock_gain[1] << 3); + gains[2] = gain - (g->subblock_gain[2] << 3); + k = g->long_end; + for(i=g->short_start;i<13;i++) { + len = bstab[i]; + for(l=0;l<3;l++) { + v0 = gains[l] - (g->scale_factors[k++] << shift) + 400; + for(j=len;j>0;j--) + *exp_ptr++ = v0; + } + } + } +} + +static void reorder_block(mp3_context_t *s, granule_t *g) +{ + int i, j, len; + int32_t *ptr, *dst, *ptr1; + int32_t tmp[576]; + + if (g->block_type != 2) + return; + + if (g->switch_point) { + if (s->sample_rate_index != 8) { + ptr = g->sb_hybrid + 36; + } else { + ptr = g->sb_hybrid + 48; + } + } else { + ptr = g->sb_hybrid; + } + + for(i=g->short_start;i<13;i++) { + len = band_size_short[s->sample_rate_index][i]; + ptr1 = ptr; + dst = tmp; + for(j=len;j>0;j--) { + *dst++ = ptr[0*len]; + *dst++ = ptr[1*len]; + *dst++ = ptr[2*len]; + ptr++; + } + ptr+=2*len; + libc_memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1)); + } +} + +static void compute_antialias(mp3_context_t *s, granule_t *g) { + int32_t *ptr, *csa; + int n, i; + + /* we antialias only "long" bands */ + if (g->block_type == 2) { + if (!g->switch_point) + return; + /* XXX: check this for 8000Hz case */ + n = 1; + } else { + n = SBLIMIT - 1; + } + + ptr = g->sb_hybrid + 18; + for(i = n;i > 0;i--) { + int tmp0, tmp1, tmp2; + csa = &csa_table[0][0]; +#define INT_AA(j) \ + tmp0 = ptr[-1-j];\ + tmp1 = ptr[ j];\ + tmp2= MULH(tmp0 + tmp1, csa[0+4*j]);\ + ptr[-1-j] = 4*(tmp2 - MULH(tmp1, csa[2+4*j]));\ + ptr[ j] = 4*(tmp2 + MULH(tmp0, csa[3+4*j])); + + INT_AA(0) + INT_AA(1) + INT_AA(2) + INT_AA(3) + INT_AA(4) + INT_AA(5) + INT_AA(6) + INT_AA(7) + + ptr += 18; + } +} + +static void compute_stereo( + mp3_context_t *s, granule_t *g0, granule_t *g1 +) { + int i, j, k, l; + int32_t v1, v2; + int sf_max, tmp0, tmp1, sf, len, non_zero_found; + int32_t (*is_tab)[16]; + int32_t *tab0, *tab1; + int non_zero_found_short[3]; + + if (s->mode_ext & MODE_EXT_I_STEREO) { + if (!s->lsf) { + is_tab = is_table; + sf_max = 7; + } else { + is_tab = is_table_lsf[g1->scalefac_compress & 1]; + sf_max = 16; + } + + tab0 = g0->sb_hybrid + 576; + tab1 = g1->sb_hybrid + 576; + + non_zero_found_short[0] = 0; + non_zero_found_short[1] = 0; + non_zero_found_short[2] = 0; + k = (13 - g1->short_start) * 3 + g1->long_end - 3; + for(i = 12;i >= g1->short_start;i--) { + /* for last band, use previous scale factor */ + if (i != 11) + k -= 3; + len = band_size_short[s->sample_rate_index][i]; + for(l=2;l>=0;l--) { + tab0 -= len; + tab1 -= len; + if (!non_zero_found_short[l]) { + /* test if non zero band. if so, stop doing i-stereo */ + for(j=0;jscale_factors[k + l]; + if (sf >= sf_max) + goto found1; + + v1 = is_tab[0][sf]; + v2 = is_tab[1][sf]; + for(j=0;jmode_ext & MODE_EXT_MS_STEREO) { + /* lower part of the spectrum : do ms stereo + if enabled */ + for(j=0;jlong_end - 1;i >= 0;i--) { + len = band_size_long[s->sample_rate_index][i]; + tab0 -= len; + tab1 -= len; + /* test if non zero band. if so, stop doing i-stereo */ + if (!non_zero_found) { + for(j=0;jscale_factors[k]; + if (sf >= sf_max) + goto found2; + v1 = is_tab[0][sf]; + v2 = is_tab[1][sf]; + for(j=0;jmode_ext & MODE_EXT_MS_STEREO) { + /* lower part of the spectrum : do ms stereo + if enabled */ + for(j=0;jmode_ext & MODE_EXT_MS_STEREO) { + /* ms stereo ONLY */ + /* NOTE: the 1/sqrt(2) normalization factor is included in the + global gain */ + tab0 = g0->sb_hybrid; + tab1 = g1->sb_hybrid; + for(i=0;i<576;i++) { + tmp0 = tab0[i]; + tmp1 = tab1[i]; + tab0[i] = tmp0 + tmp1; + tab1[i] = tmp0 - tmp1; + } + } +} + +static int huffman_decode( + mp3_context_t *s, granule_t *g, int16_t *exponents, int end_pos2 +) { + int s_index; + int i; + int last_pos, bits_left; + vlc_t *vlc; + int end_pos= s->gb.size_in_bits; + if (end_pos2 < end_pos) end_pos = end_pos2; + + /* low frequencies (called big values) */ + s_index = 0; + for(i=0;i<3;i++) { + int j, k, l, linbits; + j = g->region_size[i]; + if (j == 0) + continue; + /* select vlc table */ + k = g->table_select[i]; + l = mp3_huff_data[k][0]; + linbits = mp3_huff_data[k][1]; + vlc = &huff_vlc[l]; + + if(!l){ + libc_memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*2*j); + s_index += 2*j; + continue; + } + + /* read huffcode and compute each couple */ + for(;j>0;j--) { + int exponent, x, y, v; + int pos= get_bits_count(&s->gb); + + if (pos >= end_pos){ + switch_buffer(s, &pos, &end_pos, &end_pos2); + if(pos >= end_pos) + break; + } + y = get_vlc2(&s->gb, vlc->table, 7, 3); + + if(!y){ + g->sb_hybrid[s_index ] = + g->sb_hybrid[s_index+1] = 0; + s_index += 2; + continue; + } + + exponent= exponents[s_index]; + + if(y&16){ + x = y >> 5; + y = y & 0x0f; + if (x < 15){ + v = expval_table[ exponent ][ x ]; + }else{ + x += get_bitsz(&s->gb, linbits); + v = l3_unscale(x, exponent); + } + if (get_bits1(&s->gb)) + v = -v; + g->sb_hybrid[s_index] = v; + if (y < 15){ + v = expval_table[ exponent ][ y ]; + }else{ + y += get_bitsz(&s->gb, linbits); + v = l3_unscale(y, exponent); + } + if (get_bits1(&s->gb)) + v = -v; + g->sb_hybrid[s_index+1] = v; + }else{ + x = y >> 5; + y = y & 0x0f; + x += y; + if (x < 15){ + v = expval_table[ exponent ][ x ]; + }else{ + x += get_bitsz(&s->gb, linbits); + v = l3_unscale(x, exponent); + } + if (get_bits1(&s->gb)) + v = -v; + g->sb_hybrid[s_index+!!y] = v; + g->sb_hybrid[s_index+ !y] = 0; + } + s_index+=2; + } + } + + /* high frequencies */ + vlc = &huff_quad_vlc[g->count1table_select]; + last_pos=0; + while (s_index <= 572) { + int pos, code; + pos = get_bits_count(&s->gb); + if (pos >= end_pos) { + if (pos > end_pos2 && last_pos){ + /* some encoders generate an incorrect size for this + part. We must go back into the data */ + s_index -= 4; + skip_bits_long(&s->gb, last_pos - pos); + break; + } + switch_buffer(s, &pos, &end_pos, &end_pos2); + if(pos >= end_pos) + break; + } + last_pos= pos; + + code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1); + g->sb_hybrid[s_index+0]= + g->sb_hybrid[s_index+1]= + g->sb_hybrid[s_index+2]= + g->sb_hybrid[s_index+3]= 0; + while(code){ + const static int idxtab[16]={3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0}; + int v; + int pos= s_index+idxtab[code]; + code ^= 8>>idxtab[code]; + v = exp_table[ exponents[pos] ]; + if(get_bits1(&s->gb)) + v = -v; + g->sb_hybrid[pos] = v; + } + s_index+=4; + } + libc_memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*(576 - s_index)); + + /* skip extension bits */ + bits_left = end_pos2 - get_bits_count(&s->gb); + if (bits_left < 0) { + return -1; + } + skip_bits_long(&s->gb, bits_left); + + i= get_bits_count(&s->gb); + switch_buffer(s, &i, &end_pos, &end_pos2); + + return 0; +} + +//////////////////////////////////////////////////////////////////////////////// + +static void imdct12(int *out, int *in) +{ + int in0, in1, in2, in3, in4, in5, t1, t2; + + in0= in[0*3]; + in1= in[1*3] + in[0*3]; + in2= in[2*3] + in[1*3]; + in3= in[3*3] + in[2*3]; + in4= in[4*3] + in[3*3]; + in5= in[5*3] + in[4*3]; + in5 += in3; + in3 += in1; + + in2= MULH(2*in2, C3); + in3= MULH(4*in3, C3); + + t1 = in0 - in4; + t2 = MULH(2*(in1 - in5), icos36h[4]); + + out[ 7]= + out[10]= t1 + t2; + out[ 1]= + out[ 4]= t1 - t2; + + in0 += in4>>1; + in4 = in0 + in2; + in5 += 2*in1; + in1 = MULH(in5 + in3, icos36h[1]); + out[ 8]= + out[ 9]= in4 + in1; + out[ 2]= + out[ 3]= in4 - in1; + + in0 -= in2; + in5 = MULH(2*(in5 - in3), icos36h[7]); + out[ 0]= + out[ 5]= in0 - in5; + out[ 6]= + out[11]= in0 + in5; +} + +static void imdct36(int *out, int *buf, int *in, int *win) +{ + int i, j, t0, t1, t2, t3, s0, s1, s2, s3; + int tmp[18], *tmp1, *in1; + + for(i=17;i>=1;i--) + in[i] += in[i-1]; + for(i=17;i>=3;i-=2) + in[i] += in[i-2]; + + for(j=0;j<2;j++) { + tmp1 = tmp + j; + in1 = in + j; + t2 = in1[2*4] + in1[2*8] - in1[2*2]; + + t3 = in1[2*0] + (in1[2*6]>>1); + t1 = in1[2*0] - in1[2*6]; + tmp1[ 6] = t1 - (t2>>1); + tmp1[16] = t1 + t2; + + t0 = MULH(2*(in1[2*2] + in1[2*4]), C2); + t1 = MULH( in1[2*4] - in1[2*8] , -2*C8); + t2 = MULH(2*(in1[2*2] + in1[2*8]), -C4); + + tmp1[10] = t3 - t0 - t2; + tmp1[ 2] = t3 + t0 + t1; + tmp1[14] = t3 + t2 - t1; + + tmp1[ 4] = MULH(2*(in1[2*5] + in1[2*7] - in1[2*1]), -C3); + t2 = MULH(2*(in1[2*1] + in1[2*5]), C1); + t3 = MULH( in1[2*5] - in1[2*7] , -2*C7); + t0 = MULH(2*in1[2*3], C3); + + t1 = MULH(2*(in1[2*1] + in1[2*7]), -C5); + + tmp1[ 0] = t2 + t3 + t0; + tmp1[12] = t2 + t1 - t0; + tmp1[ 8] = t3 - t1 - t0; + } + + i = 0; + for(j=0;j<4;j++) { + t0 = tmp[i]; + t1 = tmp[i + 2]; + s0 = t1 + t0; + s2 = t1 - t0; + + t2 = tmp[i + 1]; + t3 = tmp[i + 3]; + s1 = MULH(2*(t3 + t2), icos36h[j]); + s3 = MULL(t3 - t2, icos36[8 - j]); + + t0 = s0 + s1; + t1 = s0 - s1; + out[(9 + j)*SBLIMIT] = MULH(t1, win[9 + j]) + buf[9 + j]; + out[(8 - j)*SBLIMIT] = MULH(t1, win[8 - j]) + buf[8 - j]; + buf[9 + j] = MULH(t0, win[18 + 9 + j]); + buf[8 - j] = MULH(t0, win[18 + 8 - j]); + + t0 = s2 + s3; + t1 = s2 - s3; + out[(9 + 8 - j)*SBLIMIT] = MULH(t1, win[9 + 8 - j]) + buf[9 + 8 - j]; + out[( j)*SBLIMIT] = MULH(t1, win[ j]) + buf[ j]; + buf[9 + 8 - j] = MULH(t0, win[18 + 9 + 8 - j]); + buf[ + j] = MULH(t0, win[18 + j]); + i += 4; + } + + s0 = tmp[16]; + s1 = MULH(2*tmp[17], icos36h[4]); + t0 = s0 + s1; + t1 = s0 - s1; + out[(9 + 4)*SBLIMIT] = MULH(t1, win[9 + 4]) + buf[9 + 4]; + out[(8 - 4)*SBLIMIT] = MULH(t1, win[8 - 4]) + buf[8 - 4]; + buf[9 + 4] = MULH(t0, win[18 + 9 + 4]); + buf[8 - 4] = MULH(t0, win[18 + 8 - 4]); +} + +static void compute_imdct( + mp3_context_t *s, granule_t *g, int32_t *sb_samples, int32_t *mdct_buf +) { + int32_t *ptr, *win, *win1, *buf, *out_ptr, *ptr1; + int32_t out2[12]; + int i, j, mdct_long_end, v, sblimit; + + /* find last non zero block */ + ptr = g->sb_hybrid + 576; + ptr1 = g->sb_hybrid + 2 * 18; + while (ptr >= ptr1) { + ptr -= 6; + v = ptr[0] | ptr[1] | ptr[2] | ptr[3] | ptr[4] | ptr[5]; + if (v != 0) + break; + } + sblimit = ((ptr - g->sb_hybrid) / 18) + 1; + + if (g->block_type == 2) { + /* XXX: check for 8000 Hz */ + if (g->switch_point) + mdct_long_end = 2; + else + mdct_long_end = 0; + } else { + mdct_long_end = sblimit; + } + + buf = mdct_buf; + ptr = g->sb_hybrid; + for(j=0;jswitch_point && j < 2) + win1 = mdct_win[0]; + else + win1 = mdct_win[g->block_type]; + /* select frequency inversion */ + win = win1 + ((4 * 36) & -(j & 1)); + imdct36(out_ptr, buf, ptr, win); + out_ptr += 18*SBLIMIT; + ptr += 18; + buf += 18; + } + for(j=mdct_long_end;j 32767) + v = 32767; + else if (v < -32768) + v = -32768; + synth_buf[j] = v; + } + /* copy to avoid wrap */ + libc_memcpy(synth_buf + 512, synth_buf, 32 * sizeof(int16_t)); + + samples2 = samples + 31 * incr; + w = window; + w2 = window + 31; + + sum = *dither_state; + p = synth_buf + 16; + SUM8(sum, +=, w, p); + p = synth_buf + 48; + SUM8(sum, -=, w + 32, p); + *samples = round_sample(&sum); + samples += incr; + w++; + + /* we calculate two samples at the same time to avoid one memory + access per two sample */ + for(j=1;j<16;j++) { + sum2 = 0; + p = synth_buf + 16 + j; + SUM8P2(sum, +=, sum2, -=, w, w2, p); + p = synth_buf + 48 - j; + SUM8P2(sum, -=, sum2, -=, w + 32, w2 + 32, p); + + *samples = round_sample(&sum); + samples += incr; + sum += sum2; + *samples2 = round_sample(&sum); + samples2 -= incr; + w++; + w2--; + } + + p = synth_buf + 32; + SUM8(sum, -=, w + 32, p); + *samples = round_sample(&sum); + *dither_state= sum; + + offset = (offset - 32) & 511; + *synth_buf_offset = offset; +} + +//////////////////////////////////////////////////////////////////////////////// + +static int decode_header(mp3_context_t *s, uint32_t header) { + int sample_rate, frame_size, mpeg25, padding; + int sample_rate_index, bitrate_index; + if (header & (1<<20)) { + s->lsf = (header & (1<<19)) ? 0 : 1; + mpeg25 = 0; + } else { + s->lsf = 1; + mpeg25 = 1; + } + + sample_rate_index = (header >> 10) & 3; + sample_rate = mp3_freq_tab[sample_rate_index] >> (s->lsf + mpeg25); + sample_rate_index += 3 * (s->lsf + mpeg25); + s->sample_rate_index = sample_rate_index; + s->error_protection = ((header >> 16) & 1) ^ 1; + s->sample_rate = sample_rate; + + bitrate_index = (header >> 12) & 0xf; + padding = (header >> 9) & 1; + s->mode = (header >> 6) & 3; + s->mode_ext = (header >> 4) & 3; + s->nb_channels = (s->mode == MP3_MONO) ? 1 : 2; + + if (bitrate_index != 0) { + frame_size = mp3_bitrate_tab[s->lsf][bitrate_index]; + s->bit_rate = frame_size * 1000; + s->frame_size = (frame_size * 144000) / (sample_rate << s->lsf) + padding; + } else { + /* if no frame size computed, signal it */ + return 1; + } + return 0; +} + +static int mp_decode_layer3(mp3_context_t *s) { + int nb_granules, main_data_begin, private_bits; + int gr, ch, blocksplit_flag, i, j, k, n, bits_pos; + granule_t *g; + static granule_t granules[2][2]; + static int16_t exponents[576]; + const uint8_t *ptr; + + if (s->lsf) { + main_data_begin = get_bits(&s->gb, 8); + private_bits = get_bits(&s->gb, s->nb_channels); + nb_granules = 1; + } else { + main_data_begin = get_bits(&s->gb, 9); + if (s->nb_channels == 2) + private_bits = get_bits(&s->gb, 3); + else + private_bits = get_bits(&s->gb, 5); + nb_granules = 2; + for(ch=0;chnb_channels;ch++) { + granules[ch][0].scfsi = 0; /* all scale factors are transmitted */ + granules[ch][1].scfsi = get_bits(&s->gb, 4); + } + } + + for(gr=0;grnb_channels;ch++) { + g = &granules[ch][gr]; + g->part2_3_length = get_bits(&s->gb, 12); + g->big_values = get_bits(&s->gb, 9); + g->global_gain = get_bits(&s->gb, 8); + /* if MS stereo only is selected, we precompute the + 1/sqrt(2) renormalization factor */ + if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) == + MODE_EXT_MS_STEREO) + g->global_gain -= 2; + if (s->lsf) + g->scalefac_compress = get_bits(&s->gb, 9); + else + g->scalefac_compress = get_bits(&s->gb, 4); + blocksplit_flag = get_bits(&s->gb, 1); + if (blocksplit_flag) { + g->block_type = get_bits(&s->gb, 2); + if (g->block_type == 0) + return -1; + g->switch_point = get_bits(&s->gb, 1); + for(i=0;i<2;i++) + g->table_select[i] = get_bits(&s->gb, 5); + for(i=0;i<3;i++) + g->subblock_gain[i] = get_bits(&s->gb, 3); + /* compute huffman coded region sizes */ + if (g->block_type == 2) + g->region_size[0] = (36 / 2); + else { + if (s->sample_rate_index <= 2) + g->region_size[0] = (36 / 2); + else if (s->sample_rate_index != 8) + g->region_size[0] = (54 / 2); + else + g->region_size[0] = (108 / 2); + } + g->region_size[1] = (576 / 2); + } else { + int region_address1, region_address2, l; + g->block_type = 0; + g->switch_point = 0; + for(i=0;i<3;i++) + g->table_select[i] = get_bits(&s->gb, 5); + /* compute huffman coded region sizes */ + region_address1 = get_bits(&s->gb, 4); + region_address2 = get_bits(&s->gb, 3); + g->region_size[0] = + band_index_long[s->sample_rate_index][region_address1 + 1] >> 1; + l = region_address1 + region_address2 + 2; + /* should not overflow */ + if (l > 22) + l = 22; + g->region_size[1] = + band_index_long[s->sample_rate_index][l] >> 1; + } + /* convert region offsets to region sizes and truncate + size to big_values */ + g->region_size[2] = (576 / 2); + j = 0; + for(i=0;i<3;i++) { + k = g->region_size[i]; + if (g->big_values < k) k = g->big_values; + g->region_size[i] = k - j; + j = k; + } + + /* compute band indexes */ + if (g->block_type == 2) { + if (g->switch_point) { + /* if switched mode, we handle the 36 first samples as + long blocks. For 8000Hz, we handle the 48 first + exponents as long blocks (XXX: check this!) */ + if (s->sample_rate_index <= 2) + g->long_end = 8; + else if (s->sample_rate_index != 8) + g->long_end = 6; + else + g->long_end = 4; /* 8000 Hz */ + + g->short_start = 2 + (s->sample_rate_index != 8); + } else { + g->long_end = 0; + g->short_start = 0; + } + } else { + g->short_start = 13; + g->long_end = 22; + } + + g->preflag = 0; + if (!s->lsf) + g->preflag = get_bits(&s->gb, 1); + g->scalefac_scale = get_bits(&s->gb, 1); + g->count1table_select = get_bits(&s->gb, 1); + } + } + + ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3); + /* now we get bits from the main_data_begin offset */ + if(main_data_begin > s->last_buf_size){ + s->last_buf_size= main_data_begin; + } + + memcpy(s->last_buf + s->last_buf_size, ptr, EXTRABYTES); + s->in_gb= s->gb; + init_get_bits(&s->gb, s->last_buf + s->last_buf_size - main_data_begin, main_data_begin*8); + + for(gr=0;grnb_channels;ch++) { + g = &granules[ch][gr]; + + bits_pos = get_bits_count(&s->gb); + + if (!s->lsf) { + uint8_t *sc; + int slen, slen1, slen2; + + /* MPEG1 scale factors */ + slen1 = slen_table[0][g->scalefac_compress]; + slen2 = slen_table[1][g->scalefac_compress]; + if (g->block_type == 2) { + n = g->switch_point ? 17 : 18; + j = 0; + if(slen1){ + for(i=0;iscale_factors[j++] = get_bits(&s->gb, slen1); + }else{ + libc_memset((void*) &g->scale_factors[j], 0, n); + j += n; +// for(i=0;iscale_factors[j++] = 0; + } + if(slen2){ + for(i=0;i<18;i++) + g->scale_factors[j++] = get_bits(&s->gb, slen2); + for(i=0;i<3;i++) + g->scale_factors[j++] = 0; + }else{ + for(i=0;i<21;i++) + g->scale_factors[j++] = 0; + } + } else { + sc = granules[ch][0].scale_factors; + j = 0; + for(k=0;k<4;k++) { + n = (k == 0 ? 6 : 5); + if ((g->scfsi & (0x8 >> k)) == 0) { + slen = (k < 2) ? slen1 : slen2; + if(slen){ + for(i=0;iscale_factors[j++] = get_bits(&s->gb, slen); + }else{ + libc_memset((void*) &g->scale_factors[j], 0, n); + j += n; +// for(i=0;iscale_factors[j++] = 0; + } + } else { + /* simply copy from last granule */ + for(i=0;iscale_factors[j] = sc[j]; + j++; + } + } + } + g->scale_factors[j++] = 0; + } + } else { + int tindex, tindex2, slen[4], sl, sf; + + /* LSF scale factors */ + if (g->block_type == 2) { + tindex = g->switch_point ? 2 : 1; + } else { + tindex = 0; + } + sf = g->scalefac_compress; + if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) { + /* intensity stereo case */ + sf >>= 1; + if (sf < 180) { + lsf_sf_expand(slen, sf, 6, 6, 0); + tindex2 = 3; + } else if (sf < 244) { + lsf_sf_expand(slen, sf - 180, 4, 4, 0); + tindex2 = 4; + } else { + lsf_sf_expand(slen, sf - 244, 3, 0, 0); + tindex2 = 5; + } + } else { + /* normal case */ + if (sf < 400) { + lsf_sf_expand(slen, sf, 5, 4, 4); + tindex2 = 0; + } else if (sf < 500) { + lsf_sf_expand(slen, sf - 400, 5, 4, 0); + tindex2 = 1; + } else { + lsf_sf_expand(slen, sf - 500, 3, 0, 0); + tindex2 = 2; + g->preflag = 1; + } + } + + j = 0; + for(k=0;k<4;k++) { + n = lsf_nsf_table[tindex2][tindex][k]; + sl = slen[k]; + if(sl){ + for(i=0;iscale_factors[j++] = get_bits(&s->gb, sl); + }else{ + libc_memset((void*) &g->scale_factors[j], 0, n); + j += n; +// for(i=0;iscale_factors[j++] = 0; + } + } + /* XXX: should compute exact size */ + libc_memset((void*) &g->scale_factors[j], 0, 40 - j); +// for(;j<40;j++) +// g->scale_factors[j] = 0; + } + + exponents_from_scale_factors(s, g, exponents); + + /* read Huffman coded residue */ + if (huffman_decode(s, g, exponents, + bits_pos + g->part2_3_length) < 0) + return -1; + } /* ch */ + + if (s->nb_channels == 2) + compute_stereo(s, &granules[0][gr], &granules[1][gr]); + + for(ch=0;chnb_channels;ch++) { + g = &granules[ch][gr]; + reorder_block(s, g); + compute_antialias(s, g); + compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]); + } + } /* gr */ + return nb_granules * 18; +} + +static int mp3_decode_main( + mp3_context_t *s, + int16_t *samples, const uint8_t *buf, int buf_size +) { + int i, nb_frames, ch; + int16_t *samples_ptr; + + init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE)*8); + + if (s->error_protection) + get_bits(&s->gb, 16); + + nb_frames = mp_decode_layer3(s); + + s->last_buf_size=0; + if(s->in_gb.buffer){ + align_get_bits(&s->gb); + i= (s->gb.size_in_bits - get_bits_count(&s->gb))>>3; + if(i >= 0 && i <= BACKSTEP_SIZE){ + libc_memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i); + s->last_buf_size=i; + } + s->gb= s->in_gb; + } + + align_get_bits(&s->gb); + i= (s->gb.size_in_bits - get_bits_count(&s->gb))>>3; + + if(i<0 || i > BACKSTEP_SIZE || nb_frames<0){ + i = buf_size - HEADER_SIZE; + if (BACKSTEP_SIZE < i) i = BACKSTEP_SIZE; + } + libc_memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i); + s->last_buf_size += i; + + /* apply the synthesis filter */ + for(ch=0;chnb_channels;ch++) { + samples_ptr = samples + ch; + for(i=0;isynth_buf[ch], &(s->synth_buf_offset[ch]), + window, &s->dither_state, + samples_ptr, s->nb_channels, + s->sb_samples[ch][i] + ); + samples_ptr += 32 * s->nb_channels; + } + } + return nb_frames * 32 * sizeof(uint16_t) * s->nb_channels; +} + +//////////////////////////////////////////////////////////////////////////////// + +int mp3_decode_init() { + static int init=0; + int i, j, k; + + if (!init) { + /* synth init */ + for(i=0;i<257;i++) { + int v; + v = mp3_enwindow[i]; + #if WFRAC_BITS < 16 + v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS); + #endif + window[i] = v; + if ((i & 63) != 0) + v = -v; + if (i != 0) + window[512 - i] = v; + } + + /* huffman decode tables */ + for(i=1;i<16;i++) { + const huff_table_t *h = &mp3_huff_tables[i]; + int xsize, x, y; + unsigned int n; + uint8_t tmp_bits [512]; + uint16_t tmp_codes[512]; + + libc_memset(tmp_bits , 0, sizeof(tmp_bits )); + libc_memset(tmp_codes, 0, sizeof(tmp_codes)); + + xsize = h->xsize; + n = xsize * xsize; + + j = 0; + for(x=0;xbits [j ]; + tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++]; + } + } + + init_vlc(&huff_vlc[i], 7, 512, + tmp_bits, 1, 1, tmp_codes, 2, 2); + } + for(i=0;i<2;i++) { + init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16, + mp3_quad_bits[i], 1, 1, mp3_quad_codes[i], 1, 1); + } + + for(i=0;i<9;i++) { + k = 0; + for(j=0;j<22;j++) { + band_index_long[i][j] = k; + k += band_size_long[i][j]; + } + band_index_long[i][22] = k; + } + + /* compute n ^ (4/3) and store it in mantissa/exp format */ + table_4_3_exp = (int8_t*)libc_malloc(TABLE_4_3_SIZE * sizeof(table_4_3_exp[0])); + if(!table_4_3_exp) + return -1; + table_4_3_value = (uint32_t*)libc_malloc(TABLE_4_3_SIZE * sizeof(table_4_3_value[0])); + if(!table_4_3_value) + return -1; + + for(i=1;i>4); + double f= libc_pow(i&15, 4.0 / 3.0) * libc_pow(2, (exponent-400)*0.25 + FRAC_BITS + 5); + expval_table[exponent][i&15]= f; + if((i&15)==1) + exp_table[exponent]= f; + } + + for(i=0;i<7;i++) { + float f; + int v; + if (i != 6) { + f = tan((double)i * M_PI / 12.0); + v = FIXR(f / (1.0 + f)); + } else { + v = FIXR(1.0); + } + is_table[0][i] = v; + is_table[1][6 - i] = v; + } + for(i=7;i<16;i++) + is_table[0][i] = is_table[1][i] = 0.0; + + for(i=0;i<16;i++) { + double f; + int e, k; + + for(j=0;j<2;j++) { + e = -(j + 1) * ((i + 1) >> 1); + f = libc_pow(2.0, e / 4.0); + k = i & 1; + is_table_lsf[j][k ^ 1][i] = FIXR(f); + is_table_lsf[j][k][i] = FIXR(1.0); + } + } + + for(i=0;i<8;i++) { + float ci, cs, ca; + ci = ci_table[i]; + cs = 1.0 / sqrt(1.0 + ci * ci); + ca = cs * ci; + csa_table[i][0] = FIXHR(cs/4); + csa_table[i][1] = FIXHR(ca/4); + csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4); + csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4); + csa_table_float[i][0] = cs; + csa_table_float[i][1] = ca; + csa_table_float[i][2] = ca + cs; + csa_table_float[i][3] = ca - cs; + } + + /* compute mdct windows */ + for(i=0;i<36;i++) { + for(j=0; j<4; j++){ + double d; + + if(j==2 && i%3 != 1) + continue; + + d= sin(M_PI * (i + 0.5) / 36.0); + if(j==1){ + if (i>=30) d= 0; + else if(i>=24) d= sin(M_PI * (i - 18 + 0.5) / 12.0); + else if(i>=18) d= 1; + }else if(j==3){ + if (i< 6) d= 0; + else if(i< 12) d= sin(M_PI * (i - 6 + 0.5) / 12.0); + else if(i< 18) d= 1; + } + d*= 0.5 / cos(M_PI*(2*i + 19)/72); + if(j==2) + mdct_win[j][i/3] = FIXHR((d / (1<<5))); + else + mdct_win[j][i ] = FIXHR((d / (1<<5))); + } + } + for(j=0;j<4;j++) { + for(i=0;i<36;i+=2) { + mdct_win[j + 4][i] = mdct_win[j][i]; + mdct_win[j + 4][i + 1] = -mdct_win[j][i + 1]; + } + } + init = 1; + } + return 0; +} + +void mp3_decode_free() { + libc_free(table_4_3_exp); + libc_free(table_4_3_value); + + for (int i = 0; i < 16; i++) libc_free(huff_vlc[i].table); + for (int i = 0; i < 2; i++) libc_free(huff_quad_vlc[i].table); +} + +static int mp3_decode_frame( + mp3_context_t *s, + int16_t *out_samples, int *data_size, + uint8_t *buf, int buf_size +) { + uint32_t header; + int out_size; + int extra_bytes = 0; + +retry: + if(buf_size < HEADER_SIZE) + return -1; + + header = (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3]; + if(mp3_check_header(header) < 0){ + buf++; + buf_size--; + extra_bytes++; + goto retry; + } + + if (decode_header(s, header) == 1) { + s->frame_size = -1; + return -1; + } + + if(s->frame_size<=0 || s->frame_size > buf_size){ + return -1; // incomplete frame + } + if(s->frame_size < buf_size) { + buf_size = s->frame_size; + } + + out_size = mp3_decode_main(s, out_samples, buf, buf_size); + if(out_size>=0) + *data_size = out_size; + // else: Error while decoding MPEG audio frame. + s->frame_size += extra_bytes; + return buf_size; +} + +//////////////////////////////////////////////////////////////////////////////// + +mp3_decoder_t mp3_create(void) { + void *dec = libc_calloc(sizeof(mp3_context_t), 1); + return (mp3_decoder_t) dec; +} + +void mp3_done(mp3_decoder_t dec) { + if (dec) { + mp3_context_t *ctx = (mp3_context_t*)dec; + libc_free(dec); + } +} + +int mp3_decode(mp3_decoder_t dec, void *buf, int bytes, signed short *out, mp3_info_t *info) { + int res, size = -1; + mp3_context_t *s = (mp3_context_t*) dec; + if (!s) return 0; + res = mp3_decode_frame(s, (int16_t*) out, &size, (uint8_t*)buf, bytes); + if (res < 0) return 0; + if (info) { + info->sample_rate = s->sample_rate; + info->channels = s->nb_channels; + info->audio_bytes = size; + } + return s->frame_size; +} diff --git a/src/libs/minimp3/minimp3.h b/src/libs/minimp3/minimp3.h new file mode 100644 index 0000000..0acdbd7 --- /dev/null +++ b/src/libs/minimp3/minimp3.h @@ -0,0 +1,18 @@ +#ifndef __MINIMP3_H_INCLUDED__ +#define __MINIMP3_H_INCLUDED__ + +typedef struct _mp3_info { + int sample_rate; + int channels; + int audio_bytes; // generated amount of audio per frame +} mp3_info_t; + +typedef void* mp3_decoder_t; + +mp3_decoder_t mp3_create(void); +extern int mp3_decode(mp3_decoder_t dec, void *buf, int bytes, signed short *out, mp3_info_t *info); +void mp3_done(mp3_decoder_t dec); +int mp3_decode_init(); +void mp3_decode_free(); + +#endif//__MINIMP3_H_INCLUDED__ diff --git a/src/libs/stb_vorbis/stb_vorbis.c b/src/libs/stb_vorbis/stb_vorbis.c new file mode 100644 index 0000000..021f5c7 --- /dev/null +++ b/src/libs/stb_vorbis/stb_vorbis.c @@ -0,0 +1,5422 @@ +// Ogg Vorbis audio decoder - v1.09 - public domain +// http://nothings.org/stb_vorbis/ +// +// Original version written by Sean Barrett in 2007. +// +// Originally sponsored by RAD Game Tools. Seeking sponsored +// by Phillip Bennefall, Marc Andersen, Aaron Baker, Elias Software, +// Aras Pranckevicius, and Sean Barrett. +// +// LICENSE +// +// This software is dual-licensed to the public domain and under the following +// license: you are granted a perpetual, irrevocable license to copy, modify, +// publish, and distribute this file as you see fit. +// +// No warranty for any purpose is expressed or implied by the author (nor +// by RAD Game Tools). Report bugs and send enhancements to the author. +// +// Limitations: +// +// - floor 0 not supported (used in old ogg vorbis files pre-2004) +// - lossless sample-truncation at beginning ignored +// - cannot concatenate multiple vorbis streams +// - sample positions are 32-bit, limiting seekable 192Khz +// files to around 6 hours (Ogg supports 64-bit) +// +// Feature contributors: +// Dougall Johnson (sample-exact seeking) +// +// Bugfix/warning contributors: +// Terje Mathisen Niklas Frykholm Andy Hill +// Casey Muratori John Bolton Gargaj +// Laurent Gomila Marc LeBlanc Ronny Chevalier +// Bernhard Wodo Evan Balster alxprd@github +// Tom Beaumont Ingo Leitgeb Nicolas Guillemot +// Phillip Bennefall Rohit Thiago Goulart +// manxorist@github saga musix +// +// Partial history: +// 1.09 - 2016/04/04 - back out 'truncation of last frame' fix from previous version +// 1.08 - 2016/04/02 - warnings; setup memory leaks; truncation of last frame +// 1.07 - 2015/01/16 - fixes for crashes on invalid files; warning fixes; const +// 1.06 - 2015/08/31 - full, correct support for seeking API (Dougall Johnson) +// some crash fixes when out of memory or with corrupt files +// fix some inappropriately signed shifts +// 1.05 - 2015/04/19 - don't define __forceinline if it's redundant +// 1.04 - 2014/08/27 - fix missing const-correct case in API +// 1.03 - 2014/08/07 - warning fixes +// 1.02 - 2014/07/09 - declare qsort comparison as explicitly _cdecl in Windows +// 1.01 - 2014/06/18 - fix stb_vorbis_get_samples_float (interleaved was correct) +// 1.0 - 2014/05/26 - fix memory leaks; fix warnings; fix bugs in >2-channel; +// (API change) report sample rate for decode-full-file funcs +// +// See end of file for full version history. + + +////////////////////////////////////////////////////////////////////////////// +// +// HEADER BEGINS HERE +// + +#ifndef STB_VORBIS_INCLUDE_STB_VORBIS_H +#define STB_VORBIS_INCLUDE_STB_VORBIS_H + +#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO) +#define STB_VORBIS_NO_STDIO 1 +#endif + +#ifndef STB_VORBIS_NO_STDIO +#include +#endif + +#ifdef __cplusplus +extern "C" { +#endif + +/////////// THREAD SAFETY + +// Individual stb_vorbis* handles are not thread-safe; you cannot decode from +// them from multiple threads at the same time. However, you can have multiple +// stb_vorbis* handles and decode from them independently in multiple thrads. + + +/////////// MEMORY ALLOCATION + +// normally stb_vorbis uses malloc() to allocate memory at startup, +// and alloca() to allocate temporary memory during a frame on the +// stack. (Memory consumption will depend on the amount of setup +// data in the file and how you set the compile flags for speed +// vs. size. In my test files the maximal-size usage is ~150KB.) +// +// You can modify the wrapper functions in the source (setup_malloc, +// setup_temp_malloc, temp_malloc) to change this behavior, or you +// can use a simpler allocation model: you pass in a buffer from +// which stb_vorbis will allocate _all_ its memory (including the +// temp memory). "open" may fail with a VORBIS_outofmem if you +// do not pass in enough data; there is no way to determine how +// much you do need except to succeed (at which point you can +// query get_info to find the exact amount required. yes I know +// this is lame). +// +// If you pass in a non-NULL buffer of the type below, allocation +// will occur from it as described above. Otherwise just pass NULL +// to use malloc()/alloca() + +typedef struct +{ + char *alloc_buffer; + int alloc_buffer_length_in_bytes; +} stb_vorbis_alloc; + + +/////////// FUNCTIONS USEABLE WITH ALL INPUT MODES + +typedef struct stb_vorbis stb_vorbis; + +typedef struct +{ + unsigned int sample_rate; + int channels; + + unsigned int setup_memory_required; + unsigned int setup_temp_memory_required; + unsigned int temp_memory_required; + + int max_frame_size; +} stb_vorbis_info; + +// get general information about the file +extern stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f); + +// get the last error detected (clears it, too) +extern int stb_vorbis_get_error(stb_vorbis *f); + +// close an ogg vorbis file and free all memory in use +extern void stb_vorbis_close(stb_vorbis *f); + +// this function returns the offset (in samples) from the beginning of the +// file that will be returned by the next decode, if it is known, or -1 +// otherwise. after a flush_pushdata() call, this may take a while before +// it becomes valid again. +// NOT WORKING YET after a seek with PULLDATA API +extern int stb_vorbis_get_sample_offset(stb_vorbis *f); + +// returns the current seek point within the file, or offset from the beginning +// of the memory buffer. In pushdata mode it returns 0. +extern unsigned int stb_vorbis_get_file_offset(stb_vorbis *f); + +/////////// PUSHDATA API + +#ifndef STB_VORBIS_NO_PUSHDATA_API + +// this API allows you to get blocks of data from any source and hand +// them to stb_vorbis. you have to buffer them; stb_vorbis will tell +// you how much it used, and you have to give it the rest next time; +// and stb_vorbis may not have enough data to work with and you will +// need to give it the same data again PLUS more. Note that the Vorbis +// specification does not bound the size of an individual frame. + +extern stb_vorbis *stb_vorbis_open_pushdata( + const unsigned char * datablock, int datablock_length_in_bytes, + int *datablock_memory_consumed_in_bytes, + int *error, + const stb_vorbis_alloc *alloc_buffer); +// create a vorbis decoder by passing in the initial data block containing +// the ogg&vorbis headers (you don't need to do parse them, just provide +// the first N bytes of the file--you're told if it's not enough, see below) +// on success, returns an stb_vorbis *, does not set error, returns the amount of +// data parsed/consumed on this call in *datablock_memory_consumed_in_bytes; +// on failure, returns NULL on error and sets *error, does not change *datablock_memory_consumed +// if returns NULL and *error is VORBIS_need_more_data, then the input block was +// incomplete and you need to pass in a larger block from the start of the file + +extern int stb_vorbis_decode_frame_pushdata( + stb_vorbis *f, + const unsigned char *datablock, int datablock_length_in_bytes, + int *channels, // place to write number of float * buffers + float ***output, // place to write float ** array of float * buffers + int *samples // place to write number of output samples + ); +// decode a frame of audio sample data if possible from the passed-in data block +// +// return value: number of bytes we used from datablock +// +// possible cases: +// 0 bytes used, 0 samples output (need more data) +// N bytes used, 0 samples output (resynching the stream, keep going) +// N bytes used, M samples output (one frame of data) +// note that after opening a file, you will ALWAYS get one N-bytes,0-sample +// frame, because Vorbis always "discards" the first frame. +// +// Note that on resynch, stb_vorbis will rarely consume all of the buffer, +// instead only datablock_length_in_bytes-3 or less. This is because it wants +// to avoid missing parts of a page header if they cross a datablock boundary, +// without writing state-machiney code to record a partial detection. +// +// The number of channels returned are stored in *channels (which can be +// NULL--it is always the same as the number of channels reported by +// get_info). *output will contain an array of float* buffers, one per +// channel. In other words, (*output)[0][0] contains the first sample from +// the first channel, and (*output)[1][0] contains the first sample from +// the second channel. + +extern void stb_vorbis_flush_pushdata(stb_vorbis *f); +// inform stb_vorbis that your next datablock will not be contiguous with +// previous ones (e.g. you've seeked in the data); future attempts to decode +// frames will cause stb_vorbis to resynchronize (as noted above), and +// once it sees a valid Ogg page (typically 4-8KB, as large as 64KB), it +// will begin decoding the _next_ frame. +// +// if you want to seek using pushdata, you need to seek in your file, then +// call stb_vorbis_flush_pushdata(), then start calling decoding, then once +// decoding is returning you data, call stb_vorbis_get_sample_offset, and +// if you don't like the result, seek your file again and repeat. +#endif + + +////////// PULLING INPUT API + +#ifndef STB_VORBIS_NO_PULLDATA_API +// This API assumes stb_vorbis is allowed to pull data from a source-- +// either a block of memory containing the _entire_ vorbis stream, or a +// FILE * that you or it create, or possibly some other reading mechanism +// if you go modify the source to replace the FILE * case with some kind +// of callback to your code. (But if you don't support seeking, you may +// just want to go ahead and use pushdata.) + +#if !defined(STB_VORBIS_NO_STDIO) && !defined(STB_VORBIS_NO_INTEGER_CONVERSION) +extern int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output); +#endif +#if !defined(STB_VORBIS_NO_INTEGER_CONVERSION) +extern int stb_vorbis_decode_memory(const unsigned char *mem, int len, int *channels, int *sample_rate, short **output); +#endif +// decode an entire file and output the data interleaved into a malloc()ed +// buffer stored in *output. The return value is the number of samples +// decoded, or -1 if the file could not be opened or was not an ogg vorbis file. +// When you're done with it, just free() the pointer returned in *output. + +extern stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, + int *error, const stb_vorbis_alloc *alloc_buffer); +// create an ogg vorbis decoder from an ogg vorbis stream in memory (note +// this must be the entire stream!). on failure, returns NULL and sets *error + +#ifndef STB_VORBIS_NO_STDIO +extern stb_vorbis * stb_vorbis_open_filename(const char *filename, + int *error, const stb_vorbis_alloc *alloc_buffer); +// create an ogg vorbis decoder from a filename via fopen(). on failure, +// returns NULL and sets *error (possibly to VORBIS_file_open_failure). + +extern stb_vorbis * stb_vorbis_open_file(FILE *f, int close_handle_on_close, + int *error, const stb_vorbis_alloc *alloc_buffer); +// create an ogg vorbis decoder from an open FILE *, looking for a stream at +// the _current_ seek point (ftell). on failure, returns NULL and sets *error. +// note that stb_vorbis must "own" this stream; if you seek it in between +// calls to stb_vorbis, it will become confused. Morever, if you attempt to +// perform stb_vorbis_seek_*() operations on this file, it will assume it +// owns the _entire_ rest of the file after the start point. Use the next +// function, stb_vorbis_open_file_section(), to limit it. + +extern stb_vorbis * stb_vorbis_open_file_section(FILE *f, int close_handle_on_close, + int *error, const stb_vorbis_alloc *alloc_buffer, unsigned int len); +// create an ogg vorbis decoder from an open FILE *, looking for a stream at +// the _current_ seek point (ftell); the stream will be of length 'len' bytes. +// on failure, returns NULL and sets *error. note that stb_vorbis must "own" +// this stream; if you seek it in between calls to stb_vorbis, it will become +// confused. +#endif + +extern int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number); +extern int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number); +// these functions seek in the Vorbis file to (approximately) 'sample_number'. +// after calling seek_frame(), the next call to get_frame_*() will include +// the specified sample. after calling stb_vorbis_seek(), the next call to +// stb_vorbis_get_samples_* will start with the specified sample. If you +// do not need to seek to EXACTLY the target sample when using get_samples_*, +// you can also use seek_frame(). + +extern void stb_vorbis_seek_start(stb_vorbis *f); +// this function is equivalent to stb_vorbis_seek(f,0) + +extern unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f); +extern float stb_vorbis_stream_length_in_seconds(stb_vorbis *f); +// these functions return the total length of the vorbis stream + +extern int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output); +// decode the next frame and return the number of samples. the number of +// channels returned are stored in *channels (which can be NULL--it is always +// the same as the number of channels reported by get_info). *output will +// contain an array of float* buffers, one per channel. These outputs will +// be overwritten on the next call to stb_vorbis_get_frame_*. +// +// You generally should not intermix calls to stb_vorbis_get_frame_*() +// and stb_vorbis_get_samples_*(), since the latter calls the former. + +#ifndef STB_VORBIS_NO_INTEGER_CONVERSION +extern int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts); +extern int stb_vorbis_get_frame_short (stb_vorbis *f, int num_c, short **buffer, int num_samples); +#endif +// decode the next frame and return the number of *samples* per channel. +// Note that for interleaved data, you pass in the number of shorts (the +// size of your array), but the return value is the number of samples per +// channel, not the total number of samples. +// +// The data is coerced to the number of channels you request according to the +// channel coercion rules (see below). You must pass in the size of your +// buffer(s) so that stb_vorbis will not overwrite the end of the buffer. +// The maximum buffer size needed can be gotten from get_info(); however, +// the Vorbis I specification implies an absolute maximum of 4096 samples +// per channel. + +// Channel coercion rules: +// Let M be the number of channels requested, and N the number of channels present, +// and Cn be the nth channel; let stereo L be the sum of all L and center channels, +// and stereo R be the sum of all R and center channels (channel assignment from the +// vorbis spec). +// M N output +// 1 k sum(Ck) for all k +// 2 * stereo L, stereo R +// k l k > l, the first l channels, then 0s +// k l k <= l, the first k channels +// Note that this is not _good_ surround etc. mixing at all! It's just so +// you get something useful. + +extern int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats); +extern int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples); +// gets num_samples samples, not necessarily on a frame boundary--this requires +// buffering so you have to supply the buffers. DOES NOT APPLY THE COERCION RULES. +// Returns the number of samples stored per channel; it may be less than requested +// at the end of the file. If there are no more samples in the file, returns 0. + +#ifndef STB_VORBIS_NO_INTEGER_CONVERSION +extern int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts); +extern int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int num_samples); +#endif +// gets num_samples samples, not necessarily on a frame boundary--this requires +// buffering so you have to supply the buffers. Applies the coercion rules above +// to produce 'channels' channels. Returns the number of samples stored per channel; +// it may be less than requested at the end of the file. If there are no more +// samples in the file, returns 0. + +#endif + +//////// ERROR CODES + +enum STBVorbisError +{ + VORBIS__no_error, + + VORBIS_need_more_data=1, // not a real error + + VORBIS_invalid_api_mixing, // can't mix API modes + VORBIS_outofmem, // not enough memory + VORBIS_feature_not_supported, // uses floor 0 + VORBIS_too_many_channels, // STB_VORBIS_MAX_CHANNELS is too small + VORBIS_file_open_failure, // fopen() failed + VORBIS_seek_without_length, // can't seek in unknown-length file + + VORBIS_unexpected_eof=10, // file is truncated? + VORBIS_seek_invalid, // seek past EOF + + // decoding errors (corrupt/invalid stream) -- you probably + // don't care about the exact details of these + + // vorbis errors: + VORBIS_invalid_setup=20, + VORBIS_invalid_stream, + + // ogg errors: + VORBIS_missing_capture_pattern=30, + VORBIS_invalid_stream_structure_version, + VORBIS_continued_packet_flag_invalid, + VORBIS_incorrect_stream_serial_number, + VORBIS_invalid_first_page, + VORBIS_bad_packet_type, + VORBIS_cant_find_last_page, + VORBIS_seek_failed +}; + + +#ifdef __cplusplus +} +#endif + +#endif // STB_VORBIS_INCLUDE_STB_VORBIS_H +// +// HEADER ENDS HERE +// +////////////////////////////////////////////////////////////////////////////// + +#ifndef STB_VORBIS_HEADER_ONLY + +// global configuration settings (e.g. set these in the project/makefile), +// or just set them in this file at the top (although ideally the first few +// should be visible when the header file is compiled too, although it's not +// crucial) + +// STB_VORBIS_NO_PUSHDATA_API +// does not compile the code for the various stb_vorbis_*_pushdata() +// functions +// #define STB_VORBIS_NO_PUSHDATA_API + +// STB_VORBIS_NO_PULLDATA_API +// does not compile the code for the non-pushdata APIs +// #define STB_VORBIS_NO_PULLDATA_API + +// STB_VORBIS_NO_STDIO +// does not compile the code for the APIs that use FILE *s internally +// or externally (implied by STB_VORBIS_NO_PULLDATA_API) +// #define STB_VORBIS_NO_STDIO + +// STB_VORBIS_NO_INTEGER_CONVERSION +// does not compile the code for converting audio sample data from +// float to integer (implied by STB_VORBIS_NO_PULLDATA_API) +// #define STB_VORBIS_NO_INTEGER_CONVERSION + +// STB_VORBIS_NO_FAST_SCALED_FLOAT +// does not use a fast float-to-int trick to accelerate float-to-int on +// most platforms which requires endianness be defined correctly. +//#define STB_VORBIS_NO_FAST_SCALED_FLOAT + + +// STB_VORBIS_MAX_CHANNELS [number] +// globally define this to the maximum number of channels you need. +// The spec does not put a restriction on channels except that +// the count is stored in a byte, so 255 is the hard limit. +// Reducing this saves about 16 bytes per value, so using 16 saves +// (255-16)*16 or around 4KB. Plus anything other memory usage +// I forgot to account for. Can probably go as low as 8 (7.1 audio), +// 6 (5.1 audio), or 2 (stereo only). +#ifndef STB_VORBIS_MAX_CHANNELS +#define STB_VORBIS_MAX_CHANNELS 16 // enough for anyone? +#endif + +// STB_VORBIS_PUSHDATA_CRC_COUNT [number] +// after a flush_pushdata(), stb_vorbis begins scanning for the +// next valid page, without backtracking. when it finds something +// that looks like a page, it streams through it and verifies its +// CRC32. Should that validation fail, it keeps scanning. But it's +// possible that _while_ streaming through to check the CRC32 of +// one candidate page, it sees another candidate page. This #define +// determines how many "overlapping" candidate pages it can search +// at once. Note that "real" pages are typically ~4KB to ~8KB, whereas +// garbage pages could be as big as 64KB, but probably average ~16KB. +// So don't hose ourselves by scanning an apparent 64KB page and +// missing a ton of real ones in the interim; so minimum of 2 +#ifndef STB_VORBIS_PUSHDATA_CRC_COUNT +#define STB_VORBIS_PUSHDATA_CRC_COUNT 4 +#endif + +// STB_VORBIS_FAST_HUFFMAN_LENGTH [number] +// sets the log size of the huffman-acceleration table. Maximum +// supported value is 24. with larger numbers, more decodings are O(1), +// but the table size is larger so worse cache missing, so you'll have +// to probe (and try multiple ogg vorbis files) to find the sweet spot. +#ifndef STB_VORBIS_FAST_HUFFMAN_LENGTH +#define STB_VORBIS_FAST_HUFFMAN_LENGTH 10 +#endif + +// STB_VORBIS_FAST_BINARY_LENGTH [number] +// sets the log size of the binary-search acceleration table. this +// is used in similar fashion to the fast-huffman size to set initial +// parameters for the binary search + +// STB_VORBIS_FAST_HUFFMAN_INT +// The fast huffman tables are much more efficient if they can be +// stored as 16-bit results instead of 32-bit results. This restricts +// the codebooks to having only 65535 possible outcomes, though. +// (At least, accelerated by the huffman table.) +#ifndef STB_VORBIS_FAST_HUFFMAN_INT +#define STB_VORBIS_FAST_HUFFMAN_SHORT +#endif + +// STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH +// If the 'fast huffman' search doesn't succeed, then stb_vorbis falls +// back on binary searching for the correct one. This requires storing +// extra tables with the huffman codes in sorted order. Defining this +// symbol trades off space for speed by forcing a linear search in the +// non-fast case, except for "sparse" codebooks. +// #define STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH + +// STB_VORBIS_DIVIDES_IN_RESIDUE +// stb_vorbis precomputes the result of the scalar residue decoding +// that would otherwise require a divide per chunk. you can trade off +// space for time by defining this symbol. +// #define STB_VORBIS_DIVIDES_IN_RESIDUE + +// STB_VORBIS_DIVIDES_IN_CODEBOOK +// vorbis VQ codebooks can be encoded two ways: with every case explicitly +// stored, or with all elements being chosen from a small range of values, +// and all values possible in all elements. By default, stb_vorbis expands +// this latter kind out to look like the former kind for ease of decoding, +// because otherwise an integer divide-per-vector-element is required to +// unpack the index. If you define STB_VORBIS_DIVIDES_IN_CODEBOOK, you can +// trade off storage for speed. +//#define STB_VORBIS_DIVIDES_IN_CODEBOOK + +#ifdef STB_VORBIS_CODEBOOK_SHORTS +#error "STB_VORBIS_CODEBOOK_SHORTS is no longer supported as it produced incorrect results for some input formats" +#endif + +// STB_VORBIS_DIVIDE_TABLE +// this replaces small integer divides in the floor decode loop with +// table lookups. made less than 1% difference, so disabled by default. + +// STB_VORBIS_NO_INLINE_DECODE +// disables the inlining of the scalar codebook fast-huffman decode. +// might save a little codespace; useful for debugging +// #define STB_VORBIS_NO_INLINE_DECODE + +// STB_VORBIS_NO_DEFER_FLOOR +// Normally we only decode the floor without synthesizing the actual +// full curve. We can instead synthesize the curve immediately. This +// requires more memory and is very likely slower, so I don't think +// you'd ever want to do it except for debugging. +// #define STB_VORBIS_NO_DEFER_FLOOR + + + + +////////////////////////////////////////////////////////////////////////////// + +#ifdef STB_VORBIS_NO_PULLDATA_API + #define STB_VORBIS_NO_INTEGER_CONVERSION + #define STB_VORBIS_NO_STDIO +#endif + +#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO) + #define STB_VORBIS_NO_STDIO 1 +#endif + +#ifndef STB_VORBIS_NO_INTEGER_CONVERSION +#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT + + // only need endianness for fast-float-to-int, which we don't + // use for pushdata + + #ifndef STB_VORBIS_BIG_ENDIAN + #define STB_VORBIS_ENDIAN 0 + #else + #define STB_VORBIS_ENDIAN 1 + #endif + +#endif +#endif + + +#ifndef STB_VORBIS_NO_STDIO +#include +#endif + +#ifndef STB_VORBIS_NO_CRT + #include + #include + #include + #include + + // find definition of alloca if it's not in stdlib.h: + #ifdef _MSC_VER + #include + #endif + #if defined(__linux__) || defined(__linux) || defined(__EMSCRIPTEN__) + #include + #endif +#else // STB_VORBIS_NO_CRT + #define NULL 0 + #define malloc(s) 0 + #define free(s) ((void) 0) + #define realloc(s) 0 +#endif // STB_VORBIS_NO_CRT + +#include + +#ifdef __MINGW32__ + // eff you mingw: + // "fixed": + // http://sourceforge.net/p/mingw-w64/mailman/message/32882927/ + // "no that broke the build, reverted, who cares about C": + // http://sourceforge.net/p/mingw-w64/mailman/message/32890381/ + #ifdef __forceinline + #undef __forceinline + #endif + #define __forceinline +#elif !defined(_MSC_VER) + #if __GNUC__ + #define __forceinline inline + #else + #define __forceinline + #endif +#endif + +#if STB_VORBIS_MAX_CHANNELS > 256 +#error "Value of STB_VORBIS_MAX_CHANNELS outside of allowed range" +#endif + +#if STB_VORBIS_FAST_HUFFMAN_LENGTH > 24 +#error "Value of STB_VORBIS_FAST_HUFFMAN_LENGTH outside of allowed range" +#endif + + +#if 0 +#include +#define CHECK(f) _CrtIsValidHeapPointer(f->channel_buffers[1]) +#else +#define CHECK(f) ((void) 0) +#endif + +#define MAX_BLOCKSIZE_LOG 13 // from specification +#define MAX_BLOCKSIZE (1 << MAX_BLOCKSIZE_LOG) + + +typedef unsigned char uint8; +typedef signed char int8; +typedef unsigned short uint16; +typedef signed short int16; +typedef unsigned int uint32; +typedef signed int int32; + +#ifndef TRUE +#define TRUE 1 +#define FALSE 0 +#endif + +typedef float codetype; + +// @NOTE +// +// Some arrays below are tagged "//varies", which means it's actually +// a variable-sized piece of data, but rather than malloc I assume it's +// small enough it's better to just allocate it all together with the +// main thing +// +// Most of the variables are specified with the smallest size I could pack +// them into. It might give better performance to make them all full-sized +// integers. It should be safe to freely rearrange the structures or change +// the sizes larger--nothing relies on silently truncating etc., nor the +// order of variables. + +#define FAST_HUFFMAN_TABLE_SIZE (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH) +#define FAST_HUFFMAN_TABLE_MASK (FAST_HUFFMAN_TABLE_SIZE - 1) + +typedef struct +{ + int dimensions, entries; + uint8 *codeword_lengths; + float minimum_value; + float delta_value; + uint8 value_bits; + uint8 lookup_type; + uint8 sequence_p; + uint8 sparse; + uint32 lookup_values; + codetype *multiplicands; + uint32 *codewords; + #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT + int16 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; + #else + int32 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; + #endif + uint32 *sorted_codewords; + int *sorted_values; + int sorted_entries; +} Codebook; + +typedef struct +{ + uint8 order; + uint16 rate; + uint16 bark_map_size; + uint8 amplitude_bits; + uint8 amplitude_offset; + uint8 number_of_books; + uint8 book_list[16]; // varies +} Floor0; + +typedef struct +{ + uint8 partitions; + uint8 partition_class_list[32]; // varies + uint8 class_dimensions[16]; // varies + uint8 class_subclasses[16]; // varies + uint8 class_masterbooks[16]; // varies + int16 subclass_books[16][8]; // varies + uint16 Xlist[31*8+2]; // varies + uint8 sorted_order[31*8+2]; + uint8 neighbors[31*8+2][2]; + uint8 floor1_multiplier; + uint8 rangebits; + int values; +} Floor1; + +typedef union +{ + Floor0 floor0; + Floor1 floor1; +} Floor; + +typedef struct +{ + uint32 begin, end; + uint32 part_size; + uint8 classifications; + uint8 classbook; + uint8 **classdata; + int16 (*residue_books)[8]; +} Residue; + +typedef struct +{ + uint8 magnitude; + uint8 angle; + uint8 mux; +} MappingChannel; + +typedef struct +{ + uint16 coupling_steps; + MappingChannel *chan; + uint8 submaps; + uint8 submap_floor[15]; // varies + uint8 submap_residue[15]; // varies +} Mapping; + +typedef struct +{ + uint8 blockflag; + uint8 mapping; + uint16 windowtype; + uint16 transformtype; +} Mode; + +typedef struct +{ + uint32 goal_crc; // expected crc if match + int bytes_left; // bytes left in packet + uint32 crc_so_far; // running crc + int bytes_done; // bytes processed in _current_ chunk + uint32 sample_loc; // granule pos encoded in page +} CRCscan; + +typedef struct +{ + uint32 page_start, page_end; + uint32 last_decoded_sample; +} ProbedPage; + +struct stb_vorbis +{ + // user-accessible info + unsigned int sample_rate; + int channels; + + unsigned int setup_memory_required; + unsigned int temp_memory_required; + unsigned int setup_temp_memory_required; + + // input config +#ifndef STB_VORBIS_NO_STDIO + FILE *f; + uint32 f_start; + int close_on_free; +#endif + + uint8 *stream; + uint8 *stream_start; + uint8 *stream_end; + + uint32 stream_len; + + uint8 push_mode; + + uint32 first_audio_page_offset; + + ProbedPage p_first, p_last; + + // memory management + stb_vorbis_alloc alloc; + int setup_offset; + int temp_offset; + + // run-time results + int eof; + enum STBVorbisError error; + + // user-useful data + + // header info + int blocksize[2]; + int blocksize_0, blocksize_1; + int codebook_count; + Codebook *codebooks; + int floor_count; + uint16 floor_types[64]; // varies + Floor *floor_config; + int residue_count; + uint16 residue_types[64]; // varies + Residue *residue_config; + int mapping_count; + Mapping *mapping; + int mode_count; + Mode mode_config[64]; // varies + + uint32 total_samples; + + // decode buffer + float *channel_buffers[STB_VORBIS_MAX_CHANNELS]; + float *outputs [STB_VORBIS_MAX_CHANNELS]; + + float *previous_window[STB_VORBIS_MAX_CHANNELS]; + int previous_length; + + #ifndef STB_VORBIS_NO_DEFER_FLOOR + int16 *finalY[STB_VORBIS_MAX_CHANNELS]; + #else + float *floor_buffers[STB_VORBIS_MAX_CHANNELS]; + #endif + + uint32 current_loc; // sample location of next frame to decode + int current_loc_valid; + + // per-blocksize precomputed data + + // twiddle factors + float *A[2],*B[2],*C[2]; + float *window[2]; + uint16 *bit_reverse[2]; + + // current page/packet/segment streaming info + uint32 serial; // stream serial number for verification + int last_page; + int segment_count; + uint8 segments[255]; + uint8 page_flag; + uint8 bytes_in_seg; + uint8 first_decode; + int next_seg; + int last_seg; // flag that we're on the last segment + int last_seg_which; // what was the segment number of the last seg? + uint32 acc; + int valid_bits; + int packet_bytes; + int end_seg_with_known_loc; + uint32 known_loc_for_packet; + int discard_samples_deferred; + uint32 samples_output; + + // push mode scanning + int page_crc_tests; // only in push_mode: number of tests active; -1 if not searching +#ifndef STB_VORBIS_NO_PUSHDATA_API + CRCscan scan[STB_VORBIS_PUSHDATA_CRC_COUNT]; +#endif + + // sample-access + int channel_buffer_start; + int channel_buffer_end; +}; + +#if defined(STB_VORBIS_NO_PUSHDATA_API) + #define IS_PUSH_MODE(f) FALSE +#elif defined(STB_VORBIS_NO_PULLDATA_API) + #define IS_PUSH_MODE(f) TRUE +#else + #define IS_PUSH_MODE(f) ((f)->push_mode) +#endif + +typedef struct stb_vorbis vorb; + +static int error(vorb *f, enum STBVorbisError e) +{ + f->error = e; + if (!f->eof && e != VORBIS_need_more_data) { + f->error=e; // breakpoint for debugging + } + return 0; +} + + +// these functions are used for allocating temporary memory +// while decoding. if you can afford the stack space, use +// alloca(); otherwise, provide a temp buffer and it will +// allocate out of those. + +#define array_size_required(count,size) (count*(sizeof(void *)+(size))) + +#define temp_alloc(f,size) setup_temp_malloc(f,size)//(f->alloc.alloc_buffer ? setup_temp_malloc(f,size) : alloca(size)) +#ifdef dealloca +#define temp_free(f,p) (f->alloc.alloc_buffer ? 0 : dealloca(size)) +#else +#define temp_free(f,p) 0 +#endif +#define temp_alloc_save(f) ((f)->temp_offset) +#define temp_alloc_restore(f,p) ((f)->temp_offset = (p)) + +#define temp_block_array(f,count,size) make_block_array(temp_alloc(f,array_size_required(count,size)), count, size) + +// given a sufficiently large block of memory, make an array of pointers to subblocks of it +static void *make_block_array(void *mem, int count, int size) +{ + int i; + void ** p = (void **) mem; + char *q = (char *) (p + count); + for (i=0; i < count; ++i) { + p[i] = q; + q += size; + } + return p; +} + +static void *setup_malloc(vorb *f, int sz) +{ + sz = (sz+3) & ~3; + f->setup_memory_required += sz; + if (f->alloc.alloc_buffer) { + void *p = (char *) f->alloc.alloc_buffer + f->setup_offset; + if (f->setup_offset + sz > f->temp_offset) return NULL; + f->setup_offset += sz; + return p; + } + return sz ? malloc(sz) : NULL; +} + +static void setup_free(vorb *f, void *p) +{ + if (f->alloc.alloc_buffer) return; // do nothing; setup mem is a stack + free(p); +} + +static void *setup_temp_malloc(vorb *f, int sz) +{ + sz = (sz+3) & ~3; + if (f->alloc.alloc_buffer) { + if (f->temp_offset - sz < f->setup_offset) return NULL; + f->temp_offset -= sz; + return (char *) f->alloc.alloc_buffer + f->temp_offset; + } + return malloc(sz); +} + +static void setup_temp_free(vorb *f, void *p, int sz) +{ + if (f->alloc.alloc_buffer) { + f->temp_offset += (sz+3)&~3; + return; + } + free(p); +} + +#define CRC32_POLY 0x04c11db7 // from spec + +static uint32 crc_table[256]; +static void crc32_init(void) +{ + int i,j; + uint32 s; + for(i=0; i < 256; i++) { + for (s=(uint32) i << 24, j=0; j < 8; ++j) + s = (s << 1) ^ (s >= (1U<<31) ? CRC32_POLY : 0); + crc_table[i] = s; + } +} + +static __forceinline uint32 crc32_update(uint32 crc, uint8 byte) +{ + return (crc << 8) ^ crc_table[byte ^ (crc >> 24)]; +} + + +// used in setup, and for huffman that doesn't go fast path +static unsigned int bit_reverse(unsigned int n) +{ + n = ((n & 0xAAAAAAAA) >> 1) | ((n & 0x55555555) << 1); + n = ((n & 0xCCCCCCCC) >> 2) | ((n & 0x33333333) << 2); + n = ((n & 0xF0F0F0F0) >> 4) | ((n & 0x0F0F0F0F) << 4); + n = ((n & 0xFF00FF00) >> 8) | ((n & 0x00FF00FF) << 8); + return (n >> 16) | (n << 16); +} + +static float square(float x) +{ + return x*x; +} + +// this is a weird definition of log2() for which log2(1) = 1, log2(2) = 2, log2(4) = 3 +// as required by the specification. fast(?) implementation from stb.h +// @OPTIMIZE: called multiple times per-packet with "constants"; move to setup +static int ilog(int32 n) +{ + static signed char log2_4[16] = { 0,1,2,2,3,3,3,3,4,4,4,4,4,4,4,4 }; + + // 2 compares if n < 16, 3 compares otherwise (4 if signed or n > 1<<29) + if (n < (1 << 14)) + if (n < (1 << 4)) return 0 + log2_4[n ]; + else if (n < (1 << 9)) return 5 + log2_4[n >> 5]; + else return 10 + log2_4[n >> 10]; + else if (n < (1 << 24)) + if (n < (1 << 19)) return 15 + log2_4[n >> 15]; + else return 20 + log2_4[n >> 20]; + else if (n < (1 << 29)) return 25 + log2_4[n >> 25]; + else if (n < (1 << 31)) return 30 + log2_4[n >> 30]; + else return 0; // signed n returns 0 +} + +#ifndef M_PI + #define M_PI 3.14159265358979323846264f // from CRC +#endif + +// code length assigned to a value with no huffman encoding +#define NO_CODE 255 + +/////////////////////// LEAF SETUP FUNCTIONS ////////////////////////// +// +// these functions are only called at setup, and only a few times +// per file +/* +static float float32_unpack(uint32 x) +{ + // from the specification + uint32 mantissa = x & 0x1fffff; + uint32 sign = x & 0x80000000; + uint32 exp = (x & 0x7fe00000) >> 21; + double res = sign ? -(double)mantissa : (double)mantissa; + return (float) ldexp((float)res, exp-788); +} +*/ +float ldexpi(int m, int e) { + return m * pow(2, e); +} + +static float float32_unpack(uint32 x) +{ + // from the specification + uint32 s = x & 0x80000000; + int32 m = x & 0x1fffff; + int32 e = (x & 0x7fe00000) >> 21; + return ldexpi(s ? -m : m, e - 788); +} + +// zlib & jpeg huffman tables assume that the output symbols +// can either be arbitrarily arranged, or have monotonically +// increasing frequencies--they rely on the lengths being sorted; +// this makes for a very simple generation algorithm. +// vorbis allows a huffman table with non-sorted lengths. This +// requires a more sophisticated construction, since symbols in +// order do not map to huffman codes "in order". +static void add_entry(Codebook *c, uint32 huff_code, int symbol, int count, int len, uint32 *values) +{ + if (!c->sparse) { + c->codewords [symbol] = huff_code; + } else { + c->codewords [count] = huff_code; + c->codeword_lengths[count] = len; + values [count] = symbol; + } +} + +static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values) +{ + int i,k,m=0; + uint32 available[32]; + + memset(available, 0, sizeof(available)); + // find the first entry + for (k=0; k < n; ++k) if (len[k] < NO_CODE) break; + if (k == n) { assert(c->sorted_entries == 0); return TRUE; } + // add to the list + add_entry(c, 0, k, m++, len[k], values); + // add all available leaves + for (i=1; i <= len[k]; ++i) + available[i] = 1U << (32-i); + // note that the above code treats the first case specially, + // but it's really the same as the following code, so they + // could probably be combined (except the initial code is 0, + // and I use 0 in available[] to mean 'empty') + for (i=k+1; i < n; ++i) { + uint32 res; + int z = len[i], y; + if (z == NO_CODE) continue; + // find lowest available leaf (should always be earliest, + // which is what the specification calls for) + // note that this property, and the fact we can never have + // more than one free leaf at a given level, isn't totally + // trivial to prove, but it seems true and the assert never + // fires, so! + while (z > 0 && !available[z]) --z; + if (z == 0) { return FALSE; } + res = available[z]; + assert(z >= 0 && z < 32); + available[z] = 0; + add_entry(c, bit_reverse(res), i, m++, len[i], values); + // propogate availability up the tree + if (z != len[i]) { + assert(len[i] >= 0 && len[i] < 32); + for (y=len[i]; y > z; --y) { + assert(available[y] == 0); + available[y] = res + (1 << (32-y)); + } + } + } + return TRUE; +} + +// accelerated huffman table allows fast O(1) match of all symbols +// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH +static void compute_accelerated_huffman(Codebook *c) +{ + int i, len; + for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i) + c->fast_huffman[i] = -1; + + len = c->sparse ? c->sorted_entries : c->entries; + #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT + if (len > 32767) len = 32767; // largest possible value we can encode! + #endif + for (i=0; i < len; ++i) { + if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) { + uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i]; + // set table entries for all bit combinations in the higher bits + while (z < FAST_HUFFMAN_TABLE_SIZE) { + c->fast_huffman[z] = i; + z += 1 << c->codeword_lengths[i]; + } + } + } +} + +#ifdef _MSC_VER +#define STBV_CDECL __cdecl +#else +#define STBV_CDECL +#endif + +static int STBV_CDECL uint32_compare(const void *p, const void *q) +{ + uint32 x = * (uint32 *) p; + uint32 y = * (uint32 *) q; + return x < y ? -1 : x > y; +} + +static int include_in_sort(Codebook *c, uint8 len) +{ + if (c->sparse) { assert(len != NO_CODE); return TRUE; } + if (len == NO_CODE) return FALSE; + if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE; + return FALSE; +} + +// if the fast table above doesn't work, we want to binary +// search them... need to reverse the bits +static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values) +{ + int i, len; + // build a list of all the entries + // OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN. + // this is kind of a frivolous optimization--I don't see any performance improvement, + // but it's like 4 extra lines of code, so. + if (!c->sparse) { + int k = 0; + for (i=0; i < c->entries; ++i) + if (include_in_sort(c, lengths[i])) + c->sorted_codewords[k++] = bit_reverse(c->codewords[i]); + assert(k == c->sorted_entries); + } else { + for (i=0; i < c->sorted_entries; ++i) + c->sorted_codewords[i] = bit_reverse(c->codewords[i]); + } + + qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare); + c->sorted_codewords[c->sorted_entries] = 0xffffffff; + + len = c->sparse ? c->sorted_entries : c->entries; + // now we need to indicate how they correspond; we could either + // #1: sort a different data structure that says who they correspond to + // #2: for each sorted entry, search the original list to find who corresponds + // #3: for each original entry, find the sorted entry + // #1 requires extra storage, #2 is slow, #3 can use binary search! + for (i=0; i < len; ++i) { + int huff_len = c->sparse ? lengths[values[i]] : lengths[i]; + if (include_in_sort(c,huff_len)) { + uint32 code = bit_reverse(c->codewords[i]); + int x=0, n=c->sorted_entries; + while (n > 1) { + // invariant: sc[x] <= code < sc[x+n] + int m = x + (n >> 1); + if (c->sorted_codewords[m] <= code) { + x = m; + n -= (n>>1); + } else { + n >>= 1; + } + } + assert(c->sorted_codewords[x] == code); + if (c->sparse) { + c->sorted_values[x] = values[i]; + c->codeword_lengths[x] = huff_len; + } else { + c->sorted_values[x] = i; + } + } + } +} + +// only run while parsing the header (3 times) +static int vorbis_validate(uint8 *data) +{ + static uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' }; + return memcmp(data, vorbis, 6) == 0; +} + +// called from setup only, once per code book +// (formula implied by specification) +/* +static int lookup1_values(int entries, int dim) +{ + int r = (int) floor(exp((float) log((float) entries) / dim)); + if ((int) floor(pow((float) r+1, dim)) <= entries) // (int) cast for MinGW warning; + ++r; // floor() to avoid _ftol() when non-CRT + assert(pow((float) r+1, dim) > entries); + assert((int) floor(pow((float) r, dim)) <= entries); // (int),floor() as above + return r; +} +*/ +static int lookup1_values(int a, int b) +{ + int r = 0, p = 0; + do { + r++; + p = pow(r, b); + } while (p <= a); + return r - 1; +} + +// called twice per file +static void compute_twiddle_factors(int n, float *A, float *B, float *C) +{ + int n4 = n >> 2, n8 = n >> 3; + int k,k2; + + for (k=k2=0; k < n4; ++k,k2+=2) { + A[k2 ] = (float) cos(4*k*M_PI/n); + A[k2+1] = (float) -sin(4*k*M_PI/n); + B[k2 ] = (float) cos((k2+1)*M_PI/n/2) * 0.5f; + B[k2+1] = (float) sin((k2+1)*M_PI/n/2) * 0.5f; + } + for (k=k2=0; k < n8; ++k,k2+=2) { + C[k2 ] = (float) cos(2*(k2+1)*M_PI/n); + C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n); + } +} + +static void compute_window(int n, float *window) +{ + int n2 = n >> 1, i; + for (i=0; i < n2; ++i) + window[i] = (float) sin(0.5 * M_PI * square((float) sin((i - 0 + 0.5) / n2 * 0.5 * M_PI))); +} + +static void compute_bitreverse(int n, uint16 *rev) +{ + int ld = ilog(n) - 1; // ilog is off-by-one from normal definitions + int i, n8 = n >> 3; + for (i=0; i < n8; ++i) + rev[i] = (bit_reverse(i) >> (32-ld+3)) << 2; +} + +static int init_blocksize(vorb *f, int b, int n) +{ + int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3; + f->A[b] = (float *) setup_malloc(f, sizeof(float) * n2); + f->B[b] = (float *) setup_malloc(f, sizeof(float) * n2); + f->C[b] = (float *) setup_malloc(f, sizeof(float) * n4); + if (!f->A[b] || !f->B[b] || !f->C[b]) return error(f, VORBIS_outofmem); + compute_twiddle_factors(n, f->A[b], f->B[b], f->C[b]); + f->window[b] = (float *) setup_malloc(f, sizeof(float) * n2); + if (!f->window[b]) return error(f, VORBIS_outofmem); + compute_window(n, f->window[b]); + f->bit_reverse[b] = (uint16 *) setup_malloc(f, sizeof(uint16) * n8); + if (!f->bit_reverse[b]) return error(f, VORBIS_outofmem); + compute_bitreverse(n, f->bit_reverse[b]); + return TRUE; +} + +static void neighbors(uint16 *x, int n, int *plow, int *phigh) +{ + int low = -1; + int high = 65536; + int i; + for (i=0; i < n; ++i) { + if (x[i] > low && x[i] < x[n]) { *plow = i; low = x[i]; } + if (x[i] < high && x[i] > x[n]) { *phigh = i; high = x[i]; } + } +} + +// this has been repurposed so y is now the original index instead of y +typedef struct +{ + uint16 x,y; +} Point; + +static int STBV_CDECL point_compare(const void *p, const void *q) +{ + Point *a = (Point *) p; + Point *b = (Point *) q; + return a->x < b->x ? -1 : a->x > b->x; +} + +// +/////////////////////// END LEAF SETUP FUNCTIONS ////////////////////////// + + +#if defined(STB_VORBIS_NO_STDIO) + #define USE_MEMORY(z) TRUE +#else + #define USE_MEMORY(z) ((z)->stream) +#endif + +static uint8 get8(vorb *z) +{ + if (USE_MEMORY(z)) { + if (z->stream >= z->stream_end) { z->eof = TRUE; return 0; } + return *z->stream++; + } + + #ifndef STB_VORBIS_NO_STDIO + { + int c = fgetc(z->f); + if (c == EOF) { z->eof = TRUE; return 0; } + return c; + } + #endif +} + +static uint32 get32(vorb *f) +{ + uint32 x; + x = get8(f); + x += get8(f) << 8; + x += get8(f) << 16; + x += (uint32) get8(f) << 24; + return x; +} + +static int getn(vorb *z, uint8 *data, int n) +{ + if (USE_MEMORY(z)) { + if (z->stream+n > z->stream_end) { z->eof = 1; return 0; } + memcpy(data, z->stream, n); + z->stream += n; + return 1; + } + + #ifndef STB_VORBIS_NO_STDIO + if (fread(data, n, 1, z->f) == 1) + return 1; + else { + z->eof = 1; + return 0; + } + #endif +} + +static void skip(vorb *z, int n) +{ + if (USE_MEMORY(z)) { + z->stream += n; + if (z->stream >= z->stream_end) z->eof = 1; + return; + } + #ifndef STB_VORBIS_NO_STDIO + { + long x = ftell(z->f); + fseek(z->f, x+n, SEEK_SET); + } + #endif +} + +static int set_file_offset(stb_vorbis *f, unsigned int loc) +{ + #ifndef STB_VORBIS_NO_PUSHDATA_API + if (f->push_mode) return 0; + #endif + f->eof = 0; + if (USE_MEMORY(f)) { + if (f->stream_start + loc >= f->stream_end || f->stream_start + loc < f->stream_start) { + f->stream = f->stream_end; + f->eof = 1; + return 0; + } else { + f->stream = f->stream_start + loc; + return 1; + } + } + #ifndef STB_VORBIS_NO_STDIO + if (loc + f->f_start < loc || loc >= 0x80000000) { + loc = 0x7fffffff; + f->eof = 1; + } else { + loc += f->f_start; + } + if (!fseek(f->f, loc, SEEK_SET)) + return 1; + f->eof = 1; + fseek(f->f, f->f_start, SEEK_END); + return 0; + #endif +} + + +static uint8 ogg_page_header[4] = { 0x4f, 0x67, 0x67, 0x53 }; + +static int capture_pattern(vorb *f) +{ + if (0x4f != get8(f)) return FALSE; + if (0x67 != get8(f)) return FALSE; + if (0x67 != get8(f)) return FALSE; + if (0x53 != get8(f)) return FALSE; + return TRUE; +} + +#define PAGEFLAG_continued_packet 1 +#define PAGEFLAG_first_page 2 +#define PAGEFLAG_last_page 4 + +static int start_page_no_capturepattern(vorb *f) +{ + uint32 loc0,loc1,n; + // stream structure version + if (0 != get8(f)) return error(f, VORBIS_invalid_stream_structure_version); + // header flag + f->page_flag = get8(f); + // absolute granule position + loc0 = get32(f); + loc1 = get32(f); + // @TODO: validate loc0,loc1 as valid positions? + // stream serial number -- vorbis doesn't interleave, so discard + get32(f); + //if (f->serial != get32(f)) return error(f, VORBIS_incorrect_stream_serial_number); + // page sequence number + n = get32(f); + f->last_page = n; + // CRC32 + get32(f); + // page_segments + f->segment_count = get8(f); + if (!getn(f, f->segments, f->segment_count)) + return error(f, VORBIS_unexpected_eof); + // assume we _don't_ know any the sample position of any segments + f->end_seg_with_known_loc = -2; + if (loc0 != ~0U || loc1 != ~0U) { + int i; + // determine which packet is the last one that will complete + for (i=f->segment_count-1; i >= 0; --i) + if (f->segments[i] < 255) + break; + // 'i' is now the index of the _last_ segment of a packet that ends + if (i >= 0) { + f->end_seg_with_known_loc = i; + f->known_loc_for_packet = loc0; + } + } + if (f->first_decode) { + int i,len; + ProbedPage p; + len = 0; + for (i=0; i < f->segment_count; ++i) + len += f->segments[i]; + len += 27 + f->segment_count; + p.page_start = f->first_audio_page_offset; + p.page_end = p.page_start + len; + p.last_decoded_sample = loc0; + f->p_first = p; + } + f->next_seg = 0; + return TRUE; +} + +static int start_page(vorb *f) +{ + if (!capture_pattern(f)) return error(f, VORBIS_missing_capture_pattern); + return start_page_no_capturepattern(f); +} + +static int start_packet(vorb *f) +{ + while (f->next_seg == -1) { + if (!start_page(f)) return FALSE; + if (f->page_flag & PAGEFLAG_continued_packet) + return error(f, VORBIS_continued_packet_flag_invalid); + } + f->last_seg = FALSE; + f->valid_bits = 0; + f->packet_bytes = 0; + f->bytes_in_seg = 0; + // f->next_seg is now valid + return TRUE; +} + +static int maybe_start_packet(vorb *f) +{ + if (f->next_seg == -1) { + int x = get8(f); + if (f->eof) return FALSE; // EOF at page boundary is not an error! + if (0x4f != x ) return error(f, VORBIS_missing_capture_pattern); + if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern); + if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern); + if (0x53 != get8(f)) return error(f, VORBIS_missing_capture_pattern); + if (!start_page_no_capturepattern(f)) return FALSE; + if (f->page_flag & PAGEFLAG_continued_packet) { + // set up enough state that we can read this packet if we want, + // e.g. during recovery + f->last_seg = FALSE; + f->bytes_in_seg = 0; + return error(f, VORBIS_continued_packet_flag_invalid); + } + } + return start_packet(f); +} + +static int next_segment(vorb *f) +{ + int len; + if (f->last_seg) return 0; + if (f->next_seg == -1) { + f->last_seg_which = f->segment_count-1; // in case start_page fails + if (!start_page(f)) { f->last_seg = 1; return 0; } + if (!(f->page_flag & PAGEFLAG_continued_packet)) return error(f, VORBIS_continued_packet_flag_invalid); + } + len = f->segments[f->next_seg++]; + if (len < 255) { + f->last_seg = TRUE; + f->last_seg_which = f->next_seg-1; + } + if (f->next_seg >= f->segment_count) + f->next_seg = -1; + assert(f->bytes_in_seg == 0); + f->bytes_in_seg = len; + return len; +} + +#define EOP (-1) +#define INVALID_BITS (-1) + +static int get8_packet_raw(vorb *f) +{ + if (!f->bytes_in_seg) { // CLANG! + if (f->last_seg) return EOP; + else if (!next_segment(f)) return EOP; + } + assert(f->bytes_in_seg > 0); + --f->bytes_in_seg; + ++f->packet_bytes; + return get8(f); +} + +static int get8_packet(vorb *f) +{ + int x = get8_packet_raw(f); + f->valid_bits = 0; + return x; +} + +static void flush_packet(vorb *f) +{ + while (get8_packet_raw(f) != EOP); +} + +// @OPTIMIZE: this is the secondary bit decoder, so it's probably not as important +// as the huffman decoder? +static uint32 get_bits(vorb *f, int n) +{ + uint32 z; + + if (f->valid_bits < 0) return 0; + if (f->valid_bits < n) { + if (n > 24) { + // the accumulator technique below would not work correctly in this case + z = get_bits(f, 24); + z += get_bits(f, n-24) << 24; + return z; + } + if (f->valid_bits == 0) f->acc = 0; + while (f->valid_bits < n) { + int z = get8_packet_raw(f); + if (z == EOP) { + f->valid_bits = INVALID_BITS; + return 0; + } + f->acc += z << f->valid_bits; + f->valid_bits += 8; + } + } + if (f->valid_bits < 0) return 0; + z = f->acc & ((1 << n)-1); + f->acc >>= n; + f->valid_bits -= n; + return z; +} + +// @OPTIMIZE: primary accumulator for huffman +// expand the buffer to as many bits as possible without reading off end of packet +// it might be nice to allow f->valid_bits and f->acc to be stored in registers, +// e.g. cache them locally and decode locally +static __forceinline void prep_huffman(vorb *f) +{ + if (f->valid_bits <= 24) { + if (f->valid_bits == 0) f->acc = 0; + do { + int z; + if (f->last_seg && !f->bytes_in_seg) return; + z = get8_packet_raw(f); + if (z == EOP) return; + f->acc += (unsigned) z << f->valid_bits; + f->valid_bits += 8; + } while (f->valid_bits <= 24); + } +} + +enum +{ + VORBIS_packet_id = 1, + VORBIS_packet_comment = 3, + VORBIS_packet_setup = 5 +}; + +static int codebook_decode_scalar_raw(vorb *f, Codebook *c) +{ + int i; + prep_huffman(f); + + if (c->codewords == NULL && c->sorted_codewords == NULL) + return -1; + + // cases to use binary search: sorted_codewords && !c->codewords + // sorted_codewords && c->entries > 8 + if (c->entries > 8 ? c->sorted_codewords!=NULL : !c->codewords) { + // binary search + uint32 code = bit_reverse(f->acc); + int x=0, n=c->sorted_entries, len; + + while (n > 1) { + // invariant: sc[x] <= code < sc[x+n] + int m = x + (n >> 1); + if (c->sorted_codewords[m] <= code) { + x = m; + n -= (n>>1); + } else { + n >>= 1; + } + } + // x is now the sorted index + if (!c->sparse) x = c->sorted_values[x]; + // x is now sorted index if sparse, or symbol otherwise + len = c->codeword_lengths[x]; + if (f->valid_bits >= len) { + f->acc >>= len; + f->valid_bits -= len; + return x; + } + + f->valid_bits = 0; + return -1; + } + + // if small, linear search + assert(!c->sparse); + for (i=0; i < c->entries; ++i) { + if (c->codeword_lengths[i] == NO_CODE) continue; + if (c->codewords[i] == (f->acc & ((1 << c->codeword_lengths[i])-1))) { + if (f->valid_bits >= c->codeword_lengths[i]) { + f->acc >>= c->codeword_lengths[i]; + f->valid_bits -= c->codeword_lengths[i]; + return i; + } + f->valid_bits = 0; + return -1; + } + } + + error(f, VORBIS_invalid_stream); + f->valid_bits = 0; + return -1; +} + +#ifndef STB_VORBIS_NO_INLINE_DECODE + +#define DECODE_RAW(var, f,c) \ + if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH) \ + prep_huffman(f); \ + var = f->acc & FAST_HUFFMAN_TABLE_MASK; \ + var = c->fast_huffman[var]; \ + if (var >= 0) { \ + int n = c->codeword_lengths[var]; \ + f->acc >>= n; \ + f->valid_bits -= n; \ + if (f->valid_bits < 0) { f->valid_bits = 0; var = -1; } \ + } else { \ + var = codebook_decode_scalar_raw(f,c); \ + } + +#else + +static int codebook_decode_scalar(vorb *f, Codebook *c) +{ + int i; + if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH) + prep_huffman(f); + // fast huffman table lookup + i = f->acc & FAST_HUFFMAN_TABLE_MASK; + i = c->fast_huffman[i]; + if (i >= 0) { + f->acc >>= c->codeword_lengths[i]; + f->valid_bits -= c->codeword_lengths[i]; + if (f->valid_bits < 0) { f->valid_bits = 0; return -1; } + return i; + } + return codebook_decode_scalar_raw(f,c); +} + +#define DECODE_RAW(var,f,c) var = codebook_decode_scalar(f,c); + +#endif + +#define DECODE(var,f,c) \ + DECODE_RAW(var,f,c) \ + if (c->sparse) var = c->sorted_values[var]; + +#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK + #define DECODE_VQ(var,f,c) DECODE_RAW(var,f,c) +#else + #define DECODE_VQ(var,f,c) DECODE(var,f,c) +#endif + + + + + + +// CODEBOOK_ELEMENT_FAST is an optimization for the CODEBOOK_FLOATS case +// where we avoid one addition +#define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off]) +#define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off]) +#define CODEBOOK_ELEMENT_BASE(c) (0) + +static int codebook_decode_start(vorb *f, Codebook *c) +{ + int z = -1; + + // type 0 is only legal in a scalar context + if (c->lookup_type == 0) + error(f, VORBIS_invalid_stream); + else { + DECODE_VQ(z,f,c); + if (c->sparse) assert(z < c->sorted_entries); + if (z < 0) { // check for EOP + if (!f->bytes_in_seg) + if (f->last_seg) + return z; + error(f, VORBIS_invalid_stream); + } + } + return z; +} + +static int codebook_decode(vorb *f, Codebook *c, float *output, int len) +{ + int i,z = codebook_decode_start(f,c); + if (z < 0) return FALSE; + if (len > c->dimensions) len = c->dimensions; + +#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (c->lookup_type == 1) { + float last = CODEBOOK_ELEMENT_BASE(c); + int div = 1; + for (i=0; i < len; ++i) { + int off = (z / div) % c->lookup_values; + float val = CODEBOOK_ELEMENT_FAST(c,off) + last; + output[i] += val; + if (c->sequence_p) last = val + c->minimum_value; + div *= c->lookup_values; + } + return TRUE; + } +#endif + + z *= c->dimensions; + if (c->sequence_p) { + float last = CODEBOOK_ELEMENT_BASE(c); + for (i=0; i < len; ++i) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + output[i] += val; + last = val + c->minimum_value; + } + } else { + float last = CODEBOOK_ELEMENT_BASE(c); + for (i=0; i < len; ++i) { + output[i] += CODEBOOK_ELEMENT_FAST(c,z+i) + last; + } + } + + return TRUE; +} + +static int codebook_decode_step(vorb *f, Codebook *c, float *output, int len, int step) +{ + int i,z = codebook_decode_start(f,c); + float last = CODEBOOK_ELEMENT_BASE(c); + if (z < 0) return FALSE; + if (len > c->dimensions) len = c->dimensions; + +#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (c->lookup_type == 1) { + int div = 1; + for (i=0; i < len; ++i) { + int off = (z / div) % c->lookup_values; + float val = CODEBOOK_ELEMENT_FAST(c,off) + last; + output[i*step] += val; + if (c->sequence_p) last = val; + div *= c->lookup_values; + } + return TRUE; + } +#endif + + z *= c->dimensions; + for (i=0; i < len; ++i) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + output[i*step] += val; + if (c->sequence_p) last = val; + } + + return TRUE; +} + +static int codebook_decode_deinterleave_repeat(vorb *f, Codebook *c, float **outputs, int ch, int *c_inter_p, int *p_inter_p, int len, int total_decode) +{ + int c_inter = *c_inter_p; + int p_inter = *p_inter_p; + int i,z, effective = c->dimensions; + + // type 0 is only legal in a scalar context + if (c->lookup_type == 0) return error(f, VORBIS_invalid_stream); + + while (total_decode > 0) { + float last = CODEBOOK_ELEMENT_BASE(c); + DECODE_VQ(z,f,c); + #ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK + assert(!c->sparse || z < c->sorted_entries); + #endif + if (z < 0) { + if (!f->bytes_in_seg) + if (f->last_seg) return FALSE; + return error(f, VORBIS_invalid_stream); + } + + // if this will take us off the end of the buffers, stop short! + // we check by computing the length of the virtual interleaved + // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter), + // and the length we'll be using (effective) + if (c_inter + p_inter*ch + effective > len * ch) { + effective = len*ch - (p_inter*ch - c_inter); + } + + #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (c->lookup_type == 1) { + int div = 1; + for (i=0; i < effective; ++i) { + int off = (z / div) % c->lookup_values; + float val = CODEBOOK_ELEMENT_FAST(c,off) + last; + if (outputs[c_inter]) + outputs[c_inter][p_inter] += val; + if (++c_inter == ch) { c_inter = 0; ++p_inter; } + if (c->sequence_p) last = val; + div *= c->lookup_values; + } + } else + #endif + { + z *= c->dimensions; + if (c->sequence_p) { + for (i=0; i < effective; ++i) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + if (outputs[c_inter]) + outputs[c_inter][p_inter] += val; + if (++c_inter == ch) { c_inter = 0; ++p_inter; } + last = val; + } + } else { + for (i=0; i < effective; ++i) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + if (outputs[c_inter]) + outputs[c_inter][p_inter] += val; + if (++c_inter == ch) { c_inter = 0; ++p_inter; } + } + } + } + + total_decode -= effective; + } + *c_inter_p = c_inter; + *p_inter_p = p_inter; + return TRUE; +} + +static int predict_point(int x, int x0, int x1, int y0, int y1) +{ + int dy = y1 - y0; + int adx = x1 - x0; + // @OPTIMIZE: force int division to round in the right direction... is this necessary on x86? + int err = abs(dy) * (x - x0); + int off = err / adx; + return dy < 0 ? y0 - off : y0 + off; +} + +// the following table is block-copied from the specification +static float inverse_db_table[256] = +{ + 1.0649863e-07f, 1.1341951e-07f, 1.2079015e-07f, 1.2863978e-07f, + 1.3699951e-07f, 1.4590251e-07f, 1.5538408e-07f, 1.6548181e-07f, + 1.7623575e-07f, 1.8768855e-07f, 1.9988561e-07f, 2.1287530e-07f, + 2.2670913e-07f, 2.4144197e-07f, 2.5713223e-07f, 2.7384213e-07f, + 2.9163793e-07f, 3.1059021e-07f, 3.3077411e-07f, 3.5226968e-07f, + 3.7516214e-07f, 3.9954229e-07f, 4.2550680e-07f, 4.5315863e-07f, + 4.8260743e-07f, 5.1396998e-07f, 5.4737065e-07f, 5.8294187e-07f, + 6.2082472e-07f, 6.6116941e-07f, 7.0413592e-07f, 7.4989464e-07f, + 7.9862701e-07f, 8.5052630e-07f, 9.0579828e-07f, 9.6466216e-07f, + 1.0273513e-06f, 1.0941144e-06f, 1.1652161e-06f, 1.2409384e-06f, + 1.3215816e-06f, 1.4074654e-06f, 1.4989305e-06f, 1.5963394e-06f, + 1.7000785e-06f, 1.8105592e-06f, 1.9282195e-06f, 2.0535261e-06f, + 2.1869758e-06f, 2.3290978e-06f, 2.4804557e-06f, 2.6416497e-06f, + 2.8133190e-06f, 2.9961443e-06f, 3.1908506e-06f, 3.3982101e-06f, + 3.6190449e-06f, 3.8542308e-06f, 4.1047004e-06f, 4.3714470e-06f, + 4.6555282e-06f, 4.9580707e-06f, 5.2802740e-06f, 5.6234160e-06f, + 5.9888572e-06f, 6.3780469e-06f, 6.7925283e-06f, 7.2339451e-06f, + 7.7040476e-06f, 8.2047000e-06f, 8.7378876e-06f, 9.3057248e-06f, + 9.9104632e-06f, 1.0554501e-05f, 1.1240392e-05f, 1.1970856e-05f, + 1.2748789e-05f, 1.3577278e-05f, 1.4459606e-05f, 1.5399272e-05f, + 1.6400004e-05f, 1.7465768e-05f, 1.8600792e-05f, 1.9809576e-05f, + 2.1096914e-05f, 2.2467911e-05f, 2.3928002e-05f, 2.5482978e-05f, + 2.7139006e-05f, 2.8902651e-05f, 3.0780908e-05f, 3.2781225e-05f, + 3.4911534e-05f, 3.7180282e-05f, 3.9596466e-05f, 4.2169667e-05f, + 4.4910090e-05f, 4.7828601e-05f, 5.0936773e-05f, 5.4246931e-05f, + 5.7772202e-05f, 6.1526565e-05f, 6.5524908e-05f, 6.9783085e-05f, + 7.4317983e-05f, 7.9147585e-05f, 8.4291040e-05f, 8.9768747e-05f, + 9.5602426e-05f, 0.00010181521f, 0.00010843174f, 0.00011547824f, + 0.00012298267f, 0.00013097477f, 0.00013948625f, 0.00014855085f, + 0.00015820453f, 0.00016848555f, 0.00017943469f, 0.00019109536f, + 0.00020351382f, 0.00021673929f, 0.00023082423f, 0.00024582449f, + 0.00026179955f, 0.00027881276f, 0.00029693158f, 0.00031622787f, + 0.00033677814f, 0.00035866388f, 0.00038197188f, 0.00040679456f, + 0.00043323036f, 0.00046138411f, 0.00049136745f, 0.00052329927f, + 0.00055730621f, 0.00059352311f, 0.00063209358f, 0.00067317058f, + 0.00071691700f, 0.00076350630f, 0.00081312324f, 0.00086596457f, + 0.00092223983f, 0.00098217216f, 0.0010459992f, 0.0011139742f, + 0.0011863665f, 0.0012634633f, 0.0013455702f, 0.0014330129f, + 0.0015261382f, 0.0016253153f, 0.0017309374f, 0.0018434235f, + 0.0019632195f, 0.0020908006f, 0.0022266726f, 0.0023713743f, + 0.0025254795f, 0.0026895994f, 0.0028643847f, 0.0030505286f, + 0.0032487691f, 0.0034598925f, 0.0036847358f, 0.0039241906f, + 0.0041792066f, 0.0044507950f, 0.0047400328f, 0.0050480668f, + 0.0053761186f, 0.0057254891f, 0.0060975636f, 0.0064938176f, + 0.0069158225f, 0.0073652516f, 0.0078438871f, 0.0083536271f, + 0.0088964928f, 0.009474637f, 0.010090352f, 0.010746080f, + 0.011444421f, 0.012188144f, 0.012980198f, 0.013823725f, + 0.014722068f, 0.015678791f, 0.016697687f, 0.017782797f, + 0.018938423f, 0.020169149f, 0.021479854f, 0.022875735f, + 0.024362330f, 0.025945531f, 0.027631618f, 0.029427276f, + 0.031339626f, 0.033376252f, 0.035545228f, 0.037855157f, + 0.040315199f, 0.042935108f, 0.045725273f, 0.048696758f, + 0.051861348f, 0.055231591f, 0.058820850f, 0.062643361f, + 0.066714279f, 0.071049749f, 0.075666962f, 0.080584227f, + 0.085821044f, 0.091398179f, 0.097337747f, 0.10366330f, + 0.11039993f, 0.11757434f, 0.12521498f, 0.13335215f, + 0.14201813f, 0.15124727f, 0.16107617f, 0.17154380f, + 0.18269168f, 0.19456402f, 0.20720788f, 0.22067342f, + 0.23501402f, 0.25028656f, 0.26655159f, 0.28387361f, + 0.30232132f, 0.32196786f, 0.34289114f, 0.36517414f, + 0.38890521f, 0.41417847f, 0.44109412f, 0.46975890f, + 0.50028648f, 0.53279791f, 0.56742212f, 0.60429640f, + 0.64356699f, 0.68538959f, 0.72993007f, 0.77736504f, + 0.82788260f, 0.88168307f, 0.9389798f, 1.0f +}; + + +// @OPTIMIZE: if you want to replace this bresenham line-drawing routine, +// note that you must produce bit-identical output to decode correctly; +// this specific sequence of operations is specified in the spec (it's +// drawing integer-quantized frequency-space lines that the encoder +// expects to be exactly the same) +// ... also, isn't the whole point of Bresenham's algorithm to NOT +// have to divide in the setup? sigh. +#ifndef STB_VORBIS_NO_DEFER_FLOOR +#define LINE_OP(a,b) a *= b +#else +#define LINE_OP(a,b) a = b +#endif + +#ifdef STB_VORBIS_DIVIDE_TABLE +#define DIVTAB_NUMER 32 +#define DIVTAB_DENOM 64 +int8 integer_divide_table[DIVTAB_NUMER][DIVTAB_DENOM]; // 2KB +#endif + +static __forceinline void draw_line(float *output, int x0, int y0, int x1, int y1, int n) +{ + int dy = y1 - y0; + int adx = x1 - x0; + int ady = abs(dy); + int base; + int x=x0,y=y0; + int err = 0; + int sy; + +#ifdef STB_VORBIS_DIVIDE_TABLE + if (adx < DIVTAB_DENOM && ady < DIVTAB_NUMER) { + if (dy < 0) { + base = -integer_divide_table[ady][adx]; + sy = base-1; + } else { + base = integer_divide_table[ady][adx]; + sy = base+1; + } + } else { + base = dy / adx; + if (dy < 0) + sy = base - 1; + else + sy = base+1; + } +#else + base = dy / adx; + if (dy < 0) + sy = base - 1; + else + sy = base+1; +#endif + ady -= abs(base) * adx; + if (x1 > n) x1 = n; + if (x < x1) { + LINE_OP(output[x], inverse_db_table[y]); + for (++x; x < x1; ++x) { + err += ady; + if (err >= adx) { + err -= adx; + y += sy; + } else + y += base; + LINE_OP(output[x], inverse_db_table[y]); + } + } +} + +static int residue_decode(vorb *f, Codebook *book, float *target, int offset, int n, int rtype) +{ + int k; + if (rtype == 0) { + int step = n / book->dimensions; + for (k=0; k < step; ++k) + if (!codebook_decode_step(f, book, target+offset+k, n-offset-k, step)) + return FALSE; + } else { + for (k=0; k < n; ) { + if (!codebook_decode(f, book, target+offset, n-k)) + return FALSE; + k += book->dimensions; + offset += book->dimensions; + } + } + return TRUE; +} + +static void decode_residue(vorb *f, float *residue_buffers[], int ch, int n, int rn, uint8 *do_not_decode) +{ + int i,j,pass; + Residue *r = f->residue_config + rn; + int rtype = f->residue_types[rn]; + int c = r->classbook; + int classwords = f->codebooks[c].dimensions; + int n_read = r->end - r->begin; + int part_read = n_read / r->part_size; + int temp_alloc_point = temp_alloc_save(f); + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + uint8 ***part_classdata = (uint8 ***) temp_block_array(f,f->channels, part_read * sizeof(**part_classdata)); + #else + int **classifications = (int **) temp_block_array(f,f->channels, part_read * sizeof(**classifications)); + #endif + + CHECK(f); + + for (i=0; i < ch; ++i) + if (!do_not_decode[i]) + memset(residue_buffers[i], 0, sizeof(float) * n); + + if (rtype == 2 && ch != 1) { + for (j=0; j < ch; ++j) + if (!do_not_decode[j]) + break; + if (j == ch) + goto done; + + for (pass=0; pass < 8; ++pass) { + int pcount = 0, class_set = 0; + if (ch == 2) { + while (pcount < part_read) { + int z = r->begin + pcount*r->part_size; + int c_inter = (z & 1), p_inter = z>>1; + if (pass == 0) { + Codebook *c = f->codebooks+r->classbook; + int q; + DECODE(q,f,c); + if (q == EOP) goto done; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + part_classdata[0][class_set] = r->classdata[q]; + #else + for (i=classwords-1; i >= 0; --i) { + classifications[0][i+pcount] = q % r->classifications; + q /= r->classifications; + } + #endif + } + for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { + int z = r->begin + pcount*r->part_size; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + int c = part_classdata[0][class_set][i]; + #else + int c = classifications[0][pcount]; + #endif + int b = r->residue_books[c][pass]; + if (b >= 0) { + Codebook *book = f->codebooks + b; + #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + #else + // saves 1% + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + #endif + } else { + z += r->part_size; + c_inter = z & 1; + p_inter = z >> 1; + } + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + ++class_set; + #endif + } + } else if (ch == 1) { + while (pcount < part_read) { + int z = r->begin + pcount*r->part_size; + int c_inter = 0, p_inter = z; + if (pass == 0) { + Codebook *c = f->codebooks+r->classbook; + int q; + DECODE(q,f,c); + if (q == EOP) goto done; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + part_classdata[0][class_set] = r->classdata[q]; + #else + for (i=classwords-1; i >= 0; --i) { + classifications[0][i+pcount] = q % r->classifications; + q /= r->classifications; + } + #endif + } + for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { + int z = r->begin + pcount*r->part_size; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + int c = part_classdata[0][class_set][i]; + #else + int c = classifications[0][pcount]; + #endif + int b = r->residue_books[c][pass]; + if (b >= 0) { + Codebook *book = f->codebooks + b; + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + } else { + z += r->part_size; + c_inter = 0; + p_inter = z; + } + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + ++class_set; + #endif + } + } else { + while (pcount < part_read) { + int z = r->begin + pcount*r->part_size; + int c_inter = z % ch, p_inter = z/ch; + if (pass == 0) { + Codebook *c = f->codebooks+r->classbook; + int q; + DECODE(q,f,c); + if (q == EOP) goto done; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + part_classdata[0][class_set] = r->classdata[q]; + #else + for (i=classwords-1; i >= 0; --i) { + classifications[0][i+pcount] = q % r->classifications; + q /= r->classifications; + } + #endif + } + for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { + int z = r->begin + pcount*r->part_size; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + int c = part_classdata[0][class_set][i]; + #else + int c = classifications[0][pcount]; + #endif + int b = r->residue_books[c][pass]; + if (b >= 0) { + Codebook *book = f->codebooks + b; + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + } else { + z += r->part_size; + c_inter = z % ch; + p_inter = z / ch; + } + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + ++class_set; + #endif + } + } + } + goto done; + } + CHECK(f); + + for (pass=0; pass < 8; ++pass) { + int pcount = 0, class_set=0; + while (pcount < part_read) { + if (pass == 0) { + for (j=0; j < ch; ++j) { + if (!do_not_decode[j]) { + Codebook *c = f->codebooks+r->classbook; + int temp; + DECODE(temp,f,c); + if (temp == EOP) goto done; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + part_classdata[j][class_set] = r->classdata[temp]; + #else + for (i=classwords-1; i >= 0; --i) { + classifications[j][i+pcount] = temp % r->classifications; + temp /= r->classifications; + } + #endif + } + } + } + for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { + for (j=0; j < ch; ++j) { + if (!do_not_decode[j]) { + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + int c = part_classdata[j][class_set][i]; + #else + int c = classifications[j][pcount]; + #endif + int b = r->residue_books[c][pass]; + if (b >= 0) { + float *target = residue_buffers[j]; + int offset = r->begin + pcount * r->part_size; + int n = r->part_size; + Codebook *book = f->codebooks + b; + if (!residue_decode(f, book, target, offset, n, rtype)) + goto done; + } + } + } + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + ++class_set; + #endif + } + } + done: + CHECK(f); + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + temp_free(f,part_classdata); + #else + temp_free(f,classifications); + #endif + temp_alloc_restore(f,temp_alloc_point); +} + + +#if 0 +// slow way for debugging +void inverse_mdct_slow(float *buffer, int n) +{ + int i,j; + int n2 = n >> 1; + float *x = (float *) malloc(sizeof(*x) * n2); + memcpy(x, buffer, sizeof(*x) * n2); + for (i=0; i < n; ++i) { + float acc = 0; + for (j=0; j < n2; ++j) + // formula from paper: + //acc += n/4.0f * x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1)); + // formula from wikipedia + //acc += 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5)); + // these are equivalent, except the formula from the paper inverts the multiplier! + // however, what actually works is NO MULTIPLIER!?! + //acc += 64 * 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5)); + acc += x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1)); + buffer[i] = acc; + } + free(x); +} +#elif 0 +// same as above, but just barely able to run in real time on modern machines +void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype) +{ + float mcos[16384]; + int i,j; + int n2 = n >> 1, nmask = (n << 2) -1; + float *x = (float *) malloc(sizeof(*x) * n2); + memcpy(x, buffer, sizeof(*x) * n2); + for (i=0; i < 4*n; ++i) + mcos[i] = (float) cos(M_PI / 2 * i / n); + + for (i=0; i < n; ++i) { + float acc = 0; + for (j=0; j < n2; ++j) + acc += x[j] * mcos[(2 * i + 1 + n2)*(2*j+1) & nmask]; + buffer[i] = acc; + } + free(x); +} +#elif 0 +// transform to use a slow dct-iv; this is STILL basically trivial, +// but only requires half as many ops +void dct_iv_slow(float *buffer, int n) +{ + float mcos[16384]; + float x[2048]; + int i,j; + int n2 = n >> 1, nmask = (n << 3) - 1; + memcpy(x, buffer, sizeof(*x) * n); + for (i=0; i < 8*n; ++i) + mcos[i] = (float) cos(M_PI / 4 * i / n); + for (i=0; i < n; ++i) { + float acc = 0; + for (j=0; j < n; ++j) + acc += x[j] * mcos[((2 * i + 1)*(2*j+1)) & nmask]; + buffer[i] = acc; + } +} + +void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype) +{ + int i, n4 = n >> 2, n2 = n >> 1, n3_4 = n - n4; + float temp[4096]; + + memcpy(temp, buffer, n2 * sizeof(float)); + dct_iv_slow(temp, n2); // returns -c'-d, a-b' + + for (i=0; i < n4 ; ++i) buffer[i] = temp[i+n4]; // a-b' + for ( ; i < n3_4; ++i) buffer[i] = -temp[n3_4 - i - 1]; // b-a', c+d' + for ( ; i < n ; ++i) buffer[i] = -temp[i - n3_4]; // c'+d +} +#endif + +#ifndef LIBVORBIS_MDCT +#define LIBVORBIS_MDCT 0 +#endif + +#if LIBVORBIS_MDCT +// directly call the vorbis MDCT using an interface documented +// by Jeff Roberts... useful for performance comparison +typedef struct +{ + int n; + int log2n; + + float *trig; + int *bitrev; + + float scale; +} mdct_lookup; + +extern void mdct_init(mdct_lookup *lookup, int n); +extern void mdct_clear(mdct_lookup *l); +extern void mdct_backward(mdct_lookup *init, float *in, float *out); + +mdct_lookup M1,M2; + +void inverse_mdct(float *buffer, int n, vorb *f, int blocktype) +{ + mdct_lookup *M; + if (M1.n == n) M = &M1; + else if (M2.n == n) M = &M2; + else if (M1.n == 0) { mdct_init(&M1, n); M = &M1; } + else { + if (M2.n) __asm int 3; + mdct_init(&M2, n); + M = &M2; + } + + mdct_backward(M, buffer, buffer); +} +#endif + + +// the following were split out into separate functions while optimizing; +// they could be pushed back up but eh. __forceinline showed no change; +// they're probably already being inlined. +static void imdct_step3_iter0_loop(int n, float *e, int i_off, int k_off, float *A) +{ + float *ee0 = e + i_off; + float *ee2 = ee0 + k_off; + int i; + + assert((n & 3) == 0); + for (i=(n>>2); i > 0; --i) { + float k00_20, k01_21; + k00_20 = ee0[ 0] - ee2[ 0]; + k01_21 = ee0[-1] - ee2[-1]; + ee0[ 0] += ee2[ 0];//ee0[ 0] = ee0[ 0] + ee2[ 0]; + ee0[-1] += ee2[-1];//ee0[-1] = ee0[-1] + ee2[-1]; + ee2[ 0] = k00_20 * A[0] - k01_21 * A[1]; + ee2[-1] = k01_21 * A[0] + k00_20 * A[1]; + A += 8; + + k00_20 = ee0[-2] - ee2[-2]; + k01_21 = ee0[-3] - ee2[-3]; + ee0[-2] += ee2[-2];//ee0[-2] = ee0[-2] + ee2[-2]; + ee0[-3] += ee2[-3];//ee0[-3] = ee0[-3] + ee2[-3]; + ee2[-2] = k00_20 * A[0] - k01_21 * A[1]; + ee2[-3] = k01_21 * A[0] + k00_20 * A[1]; + A += 8; + + k00_20 = ee0[-4] - ee2[-4]; + k01_21 = ee0[-5] - ee2[-5]; + ee0[-4] += ee2[-4];//ee0[-4] = ee0[-4] + ee2[-4]; + ee0[-5] += ee2[-5];//ee0[-5] = ee0[-5] + ee2[-5]; + ee2[-4] = k00_20 * A[0] - k01_21 * A[1]; + ee2[-5] = k01_21 * A[0] + k00_20 * A[1]; + A += 8; + + k00_20 = ee0[-6] - ee2[-6]; + k01_21 = ee0[-7] - ee2[-7]; + ee0[-6] += ee2[-6];//ee0[-6] = ee0[-6] + ee2[-6]; + ee0[-7] += ee2[-7];//ee0[-7] = ee0[-7] + ee2[-7]; + ee2[-6] = k00_20 * A[0] - k01_21 * A[1]; + ee2[-7] = k01_21 * A[0] + k00_20 * A[1]; + A += 8; + ee0 -= 8; + ee2 -= 8; + } +} + +static void imdct_step3_inner_r_loop(int lim, float *e, int d0, int k_off, float *A, int k1) +{ + int i; + float k00_20, k01_21; + + float *e0 = e + d0; + float *e2 = e0 + k_off; + + for (i=lim >> 2; i > 0; --i) { + k00_20 = e0[-0] - e2[-0]; + k01_21 = e0[-1] - e2[-1]; + e0[-0] += e2[-0];//e0[-0] = e0[-0] + e2[-0]; + e0[-1] += e2[-1];//e0[-1] = e0[-1] + e2[-1]; + e2[-0] = (k00_20)*A[0] - (k01_21) * A[1]; + e2[-1] = (k01_21)*A[0] + (k00_20) * A[1]; + + A += k1; + + k00_20 = e0[-2] - e2[-2]; + k01_21 = e0[-3] - e2[-3]; + e0[-2] += e2[-2];//e0[-2] = e0[-2] + e2[-2]; + e0[-3] += e2[-3];//e0[-3] = e0[-3] + e2[-3]; + e2[-2] = (k00_20)*A[0] - (k01_21) * A[1]; + e2[-3] = (k01_21)*A[0] + (k00_20) * A[1]; + + A += k1; + + k00_20 = e0[-4] - e2[-4]; + k01_21 = e0[-5] - e2[-5]; + e0[-4] += e2[-4];//e0[-4] = e0[-4] + e2[-4]; + e0[-5] += e2[-5];//e0[-5] = e0[-5] + e2[-5]; + e2[-4] = (k00_20)*A[0] - (k01_21) * A[1]; + e2[-5] = (k01_21)*A[0] + (k00_20) * A[1]; + + A += k1; + + k00_20 = e0[-6] - e2[-6]; + k01_21 = e0[-7] - e2[-7]; + e0[-6] += e2[-6];//e0[-6] = e0[-6] + e2[-6]; + e0[-7] += e2[-7];//e0[-7] = e0[-7] + e2[-7]; + e2[-6] = (k00_20)*A[0] - (k01_21) * A[1]; + e2[-7] = (k01_21)*A[0] + (k00_20) * A[1]; + + e0 -= 8; + e2 -= 8; + + A += k1; + } +} + +static void imdct_step3_inner_s_loop(int n, float *e, int i_off, int k_off, float *A, int a_off, int k0) +{ + int i; + float A0 = A[0]; + float A1 = A[0+1]; + float A2 = A[0+a_off]; + float A3 = A[0+a_off+1]; + float A4 = A[0+a_off*2+0]; + float A5 = A[0+a_off*2+1]; + float A6 = A[0+a_off*3+0]; + float A7 = A[0+a_off*3+1]; + + float k00,k11; + + float *ee0 = e +i_off; + float *ee2 = ee0+k_off; + + for (i=n; i > 0; --i) { + k00 = ee0[ 0] - ee2[ 0]; + k11 = ee0[-1] - ee2[-1]; + ee0[ 0] = ee0[ 0] + ee2[ 0]; + ee0[-1] = ee0[-1] + ee2[-1]; + ee2[ 0] = (k00) * A0 - (k11) * A1; + ee2[-1] = (k11) * A0 + (k00) * A1; + + k00 = ee0[-2] - ee2[-2]; + k11 = ee0[-3] - ee2[-3]; + ee0[-2] = ee0[-2] + ee2[-2]; + ee0[-3] = ee0[-3] + ee2[-3]; + ee2[-2] = (k00) * A2 - (k11) * A3; + ee2[-3] = (k11) * A2 + (k00) * A3; + + k00 = ee0[-4] - ee2[-4]; + k11 = ee0[-5] - ee2[-5]; + ee0[-4] = ee0[-4] + ee2[-4]; + ee0[-5] = ee0[-5] + ee2[-5]; + ee2[-4] = (k00) * A4 - (k11) * A5; + ee2[-5] = (k11) * A4 + (k00) * A5; + + k00 = ee0[-6] - ee2[-6]; + k11 = ee0[-7] - ee2[-7]; + ee0[-6] = ee0[-6] + ee2[-6]; + ee0[-7] = ee0[-7] + ee2[-7]; + ee2[-6] = (k00) * A6 - (k11) * A7; + ee2[-7] = (k11) * A6 + (k00) * A7; + + ee0 -= k0; + ee2 -= k0; + } +} + +static __forceinline void iter_54(float *z) +{ + float k00,k11,k22,k33; + float y0,y1,y2,y3; + + k00 = z[ 0] - z[-4]; + y0 = z[ 0] + z[-4]; + y2 = z[-2] + z[-6]; + k22 = z[-2] - z[-6]; + + z[-0] = y0 + y2; // z0 + z4 + z2 + z6 + z[-2] = y0 - y2; // z0 + z4 - z2 - z6 + + // done with y0,y2 + + k33 = z[-3] - z[-7]; + + z[-4] = k00 + k33; // z0 - z4 + z3 - z7 + z[-6] = k00 - k33; // z0 - z4 - z3 + z7 + + // done with k33 + + k11 = z[-1] - z[-5]; + y1 = z[-1] + z[-5]; + y3 = z[-3] + z[-7]; + + z[-1] = y1 + y3; // z1 + z5 + z3 + z7 + z[-3] = y1 - y3; // z1 + z5 - z3 - z7 + z[-5] = k11 - k22; // z1 - z5 + z2 - z6 + z[-7] = k11 + k22; // z1 - z5 - z2 + z6 +} + +static void imdct_step3_inner_s_loop_ld654(int n, float *e, int i_off, float *A, int base_n) +{ + int a_off = base_n >> 3; + float A2 = A[0+a_off]; + float *z = e + i_off; + float *base = z - 16 * n; + + while (z > base) { + float k00,k11; + + k00 = z[-0] - z[-8]; + k11 = z[-1] - z[-9]; + z[-0] = z[-0] + z[-8]; + z[-1] = z[-1] + z[-9]; + z[-8] = k00; + z[-9] = k11 ; + + k00 = z[ -2] - z[-10]; + k11 = z[ -3] - z[-11]; + z[ -2] = z[ -2] + z[-10]; + z[ -3] = z[ -3] + z[-11]; + z[-10] = (k00+k11) * A2; + z[-11] = (k11-k00) * A2; + + k00 = z[-12] - z[ -4]; // reverse to avoid a unary negation + k11 = z[ -5] - z[-13]; + z[ -4] = z[ -4] + z[-12]; + z[ -5] = z[ -5] + z[-13]; + z[-12] = k11; + z[-13] = k00; + + k00 = z[-14] - z[ -6]; // reverse to avoid a unary negation + k11 = z[ -7] - z[-15]; + z[ -6] = z[ -6] + z[-14]; + z[ -7] = z[ -7] + z[-15]; + z[-14] = (k00+k11) * A2; + z[-15] = (k00-k11) * A2; + + iter_54(z); + iter_54(z-8); + z -= 16; + } +} + +static void inverse_mdct(float *buffer, int n, vorb *f, int blocktype) +{ + int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l; + int ld; + // @OPTIMIZE: reduce register pressure by using fewer variables? + int save_point = temp_alloc_save(f); + float *buf2 = (float *) temp_alloc(f, n2 * sizeof(*buf2)); + float *u=NULL,*v=NULL; + // twiddle factors + float *A = f->A[blocktype]; + + // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio" + // See notes about bugs in that paper in less-optimal implementation 'inverse_mdct_old' after this function. + + // kernel from paper + + + // merged: + // copy and reflect spectral data + // step 0 + + // note that it turns out that the items added together during + // this step are, in fact, being added to themselves (as reflected + // by step 0). inexplicable inefficiency! this became obvious + // once I combined the passes. + + // so there's a missing 'times 2' here (for adding X to itself). + // this propogates through linearly to the end, where the numbers + // are 1/2 too small, and need to be compensated for. + + { + float *d,*e, *AA, *e_stop; + d = &buf2[n2-2]; + AA = A; + e = &buffer[0]; + e_stop = &buffer[n2]; + while (e != e_stop) { + d[1] = (e[0] * AA[0] - e[2]*AA[1]); + d[0] = (e[0] * AA[1] + e[2]*AA[0]); + d -= 2; + AA += 2; + e += 4; + } + + e = &buffer[n2-3]; + while (d >= buf2) { + d[1] = (-e[2] * AA[0] - -e[0]*AA[1]); + d[0] = (-e[2] * AA[1] + -e[0]*AA[0]); + d -= 2; + AA += 2; + e -= 4; + } + } + + // now we use symbolic names for these, so that we can + // possibly swap their meaning as we change which operations + // are in place + + u = buffer; + v = buf2; + + // step 2 (paper output is w, now u) + // this could be in place, but the data ends up in the wrong + // place... _somebody_'s got to swap it, so this is nominated + { + float *AA = &A[n2-8]; + float *d0,*d1, *e0, *e1; + + e0 = &v[n4]; + e1 = &v[0]; + + d0 = &u[n4]; + d1 = &u[0]; + + while (AA >= A) { + float v40_20, v41_21; + + v41_21 = e0[1] - e1[1]; + v40_20 = e0[0] - e1[0]; + d0[1] = e0[1] + e1[1]; + d0[0] = e0[0] + e1[0]; + d1[1] = v41_21*AA[4] - v40_20*AA[5]; + d1[0] = v40_20*AA[4] + v41_21*AA[5]; + + v41_21 = e0[3] - e1[3]; + v40_20 = e0[2] - e1[2]; + d0[3] = e0[3] + e1[3]; + d0[2] = e0[2] + e1[2]; + d1[3] = v41_21*AA[0] - v40_20*AA[1]; + d1[2] = v40_20*AA[0] + v41_21*AA[1]; + + AA -= 8; + + d0 += 4; + d1 += 4; + e0 += 4; + e1 += 4; + } + } + + // step 3 + ld = ilog(n) - 1; // ilog is off-by-one from normal definitions + + // optimized step 3: + + // the original step3 loop can be nested r inside s or s inside r; + // it's written originally as s inside r, but this is dumb when r + // iterates many times, and s few. So I have two copies of it and + // switch between them halfway. + + // this is iteration 0 of step 3 + imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*0, -(n >> 3), A); + imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*1, -(n >> 3), A); + + // this is iteration 1 of step 3 + imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*0, -(n >> 4), A, 16); + imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*1, -(n >> 4), A, 16); + imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*2, -(n >> 4), A, 16); + imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*3, -(n >> 4), A, 16); + + l=2; + for (; l < (ld-3)>>1; ++l) { + int k0 = n >> (l+2), k0_2 = k0>>1; + int lim = 1 << (l+1); + int i; + for (i=0; i < lim; ++i) + imdct_step3_inner_r_loop(n >> (l+4), u, n2-1 - k0*i, -k0_2, A, 1 << (l+3)); + } + + for (; l < ld-6; ++l) { + int k0 = n >> (l+2), k1 = 1 << (l+3), k0_2 = k0>>1; + int rlim = n >> (l+6), r; + int lim = 1 << (l+1); + int i_off; + float *A0 = A; + i_off = n2-1; + for (r=rlim; r > 0; --r) { + imdct_step3_inner_s_loop(lim, u, i_off, -k0_2, A0, k1, k0); + A0 += k1*4; + i_off -= 8; + } + } + + // iterations with count: + // ld-6,-5,-4 all interleaved together + // the big win comes from getting rid of needless flops + // due to the constants on pass 5 & 4 being all 1 and 0; + // combining them to be simultaneous to improve cache made little difference + imdct_step3_inner_s_loop_ld654(n >> 5, u, n2-1, A, n); + + // output is u + + // step 4, 5, and 6 + // cannot be in-place because of step 5 + { + uint16 *bitrev = f->bit_reverse[blocktype]; + // weirdly, I'd have thought reading sequentially and writing + // erratically would have been better than vice-versa, but in + // fact that's not what my testing showed. (That is, with + // j = bitreverse(i), do you read i and write j, or read j and write i.) + + float *d0 = &v[n4-4]; + float *d1 = &v[n2-4]; + while (d0 >= v) { + int k4; + + k4 = bitrev[0]; + d1[3] = u[k4+0]; + d1[2] = u[k4+1]; + d0[3] = u[k4+2]; + d0[2] = u[k4+3]; + + k4 = bitrev[1]; + d1[1] = u[k4+0]; + d1[0] = u[k4+1]; + d0[1] = u[k4+2]; + d0[0] = u[k4+3]; + + d0 -= 4; + d1 -= 4; + bitrev += 2; + } + } + // (paper output is u, now v) + + + // data must be in buf2 + assert(v == buf2); + + // step 7 (paper output is v, now v) + // this is now in place + { + float *C = f->C[blocktype]; + float *d, *e; + + d = v; + e = v + n2 - 4; + + while (d < e) { + float a02,a11,b0,b1,b2,b3; + + a02 = d[0] - e[2]; + a11 = d[1] + e[3]; + + b0 = C[1]*a02 + C[0]*a11; + b1 = C[1]*a11 - C[0]*a02; + + b2 = d[0] + e[ 2]; + b3 = d[1] - e[ 3]; + + d[0] = b2 + b0; + d[1] = b3 + b1; + e[2] = b2 - b0; + e[3] = b1 - b3; + + a02 = d[2] - e[0]; + a11 = d[3] + e[1]; + + b0 = C[3]*a02 + C[2]*a11; + b1 = C[3]*a11 - C[2]*a02; + + b2 = d[2] + e[ 0]; + b3 = d[3] - e[ 1]; + + d[2] = b2 + b0; + d[3] = b3 + b1; + e[0] = b2 - b0; + e[1] = b1 - b3; + + C += 4; + d += 4; + e -= 4; + } + } + + // data must be in buf2 + + + // step 8+decode (paper output is X, now buffer) + // this generates pairs of data a la 8 and pushes them directly through + // the decode kernel (pushing rather than pulling) to avoid having + // to make another pass later + + // this cannot POSSIBLY be in place, so we refer to the buffers directly + + { + float *d0,*d1,*d2,*d3; + + float *B = f->B[blocktype] + n2 - 8; + float *e = buf2 + n2 - 8; + d0 = &buffer[0]; + d1 = &buffer[n2-4]; + d2 = &buffer[n2]; + d3 = &buffer[n-4]; + while (e >= v) { + float p0,p1,p2,p3; + + p3 = e[6]*B[7] - e[7]*B[6]; + p2 = -e[6]*B[6] - e[7]*B[7]; + + d0[0] = p3; + d1[3] = - p3; + d2[0] = p2; + d3[3] = p2; + + p1 = e[4]*B[5] - e[5]*B[4]; + p0 = -e[4]*B[4] - e[5]*B[5]; + + d0[1] = p1; + d1[2] = - p1; + d2[1] = p0; + d3[2] = p0; + + p3 = e[2]*B[3] - e[3]*B[2]; + p2 = -e[2]*B[2] - e[3]*B[3]; + + d0[2] = p3; + d1[1] = - p3; + d2[2] = p2; + d3[1] = p2; + + p1 = e[0]*B[1] - e[1]*B[0]; + p0 = -e[0]*B[0] - e[1]*B[1]; + + d0[3] = p1; + d1[0] = - p1; + d2[3] = p0; + d3[0] = p0; + + B -= 8; + e -= 8; + d0 += 4; + d2 += 4; + d1 -= 4; + d3 -= 4; + } + } + + temp_free(f,buf2); + temp_alloc_restore(f,save_point); +} + +#if 0 +// this is the original version of the above code, if you want to optimize it from scratch +void inverse_mdct_naive(float *buffer, int n) +{ + float s; + float A[1 << 12], B[1 << 12], C[1 << 11]; + int i,k,k2,k4, n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l; + int n3_4 = n - n4, ld; + // how can they claim this only uses N words?! + // oh, because they're only used sparsely, whoops + float u[1 << 13], X[1 << 13], v[1 << 13], w[1 << 13]; + // set up twiddle factors + + for (k=k2=0; k < n4; ++k,k2+=2) { + A[k2 ] = (float) cos(4*k*M_PI/n); + A[k2+1] = (float) -sin(4*k*M_PI/n); + B[k2 ] = (float) cos((k2+1)*M_PI/n/2); + B[k2+1] = (float) sin((k2+1)*M_PI/n/2); + } + for (k=k2=0; k < n8; ++k,k2+=2) { + C[k2 ] = (float) cos(2*(k2+1)*M_PI/n); + C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n); + } + + // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio" + // Note there are bugs in that pseudocode, presumably due to them attempting + // to rename the arrays nicely rather than representing the way their actual + // implementation bounces buffers back and forth. As a result, even in the + // "some formulars corrected" version, a direct implementation fails. These + // are noted below as "paper bug". + + // copy and reflect spectral data + for (k=0; k < n2; ++k) u[k] = buffer[k]; + for ( ; k < n ; ++k) u[k] = -buffer[n - k - 1]; + // kernel from paper + // step 1 + for (k=k2=k4=0; k < n4; k+=1, k2+=2, k4+=4) { + v[n-k4-1] = (u[k4] - u[n-k4-1]) * A[k2] - (u[k4+2] - u[n-k4-3])*A[k2+1]; + v[n-k4-3] = (u[k4] - u[n-k4-1]) * A[k2+1] + (u[k4+2] - u[n-k4-3])*A[k2]; + } + // step 2 + for (k=k4=0; k < n8; k+=1, k4+=4) { + w[n2+3+k4] = v[n2+3+k4] + v[k4+3]; + w[n2+1+k4] = v[n2+1+k4] + v[k4+1]; + w[k4+3] = (v[n2+3+k4] - v[k4+3])*A[n2-4-k4] - (v[n2+1+k4]-v[k4+1])*A[n2-3-k4]; + w[k4+1] = (v[n2+1+k4] - v[k4+1])*A[n2-4-k4] + (v[n2+3+k4]-v[k4+3])*A[n2-3-k4]; + } + // step 3 + ld = ilog(n) - 1; // ilog is off-by-one from normal definitions + for (l=0; l < ld-3; ++l) { + int k0 = n >> (l+2), k1 = 1 << (l+3); + int rlim = n >> (l+4), r4, r; + int s2lim = 1 << (l+2), s2; + for (r=r4=0; r < rlim; r4+=4,++r) { + for (s2=0; s2 < s2lim; s2+=2) { + u[n-1-k0*s2-r4] = w[n-1-k0*s2-r4] + w[n-1-k0*(s2+1)-r4]; + u[n-3-k0*s2-r4] = w[n-3-k0*s2-r4] + w[n-3-k0*(s2+1)-r4]; + u[n-1-k0*(s2+1)-r4] = (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1] + - (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1+1]; + u[n-3-k0*(s2+1)-r4] = (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1] + + (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1+1]; + } + } + if (l+1 < ld-3) { + // paper bug: ping-ponging of u&w here is omitted + memcpy(w, u, sizeof(u)); + } + } + + // step 4 + for (i=0; i < n8; ++i) { + int j = bit_reverse(i) >> (32-ld+3); + assert(j < n8); + if (i == j) { + // paper bug: original code probably swapped in place; if copying, + // need to directly copy in this case + int i8 = i << 3; + v[i8+1] = u[i8+1]; + v[i8+3] = u[i8+3]; + v[i8+5] = u[i8+5]; + v[i8+7] = u[i8+7]; + } else if (i < j) { + int i8 = i << 3, j8 = j << 3; + v[j8+1] = u[i8+1], v[i8+1] = u[j8 + 1]; + v[j8+3] = u[i8+3], v[i8+3] = u[j8 + 3]; + v[j8+5] = u[i8+5], v[i8+5] = u[j8 + 5]; + v[j8+7] = u[i8+7], v[i8+7] = u[j8 + 7]; + } + } + // step 5 + for (k=0; k < n2; ++k) { + w[k] = v[k*2+1]; + } + // step 6 + for (k=k2=k4=0; k < n8; ++k, k2 += 2, k4 += 4) { + u[n-1-k2] = w[k4]; + u[n-2-k2] = w[k4+1]; + u[n3_4 - 1 - k2] = w[k4+2]; + u[n3_4 - 2 - k2] = w[k4+3]; + } + // step 7 + for (k=k2=0; k < n8; ++k, k2 += 2) { + v[n2 + k2 ] = ( u[n2 + k2] + u[n-2-k2] + C[k2+1]*(u[n2+k2]-u[n-2-k2]) + C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2; + v[n-2 - k2] = ( u[n2 + k2] + u[n-2-k2] - C[k2+1]*(u[n2+k2]-u[n-2-k2]) - C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2; + v[n2+1+ k2] = ( u[n2+1+k2] - u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2; + v[n-1 - k2] = (-u[n2+1+k2] + u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2; + } + // step 8 + for (k=k2=0; k < n4; ++k,k2 += 2) { + X[k] = v[k2+n2]*B[k2 ] + v[k2+1+n2]*B[k2+1]; + X[n2-1-k] = v[k2+n2]*B[k2+1] - v[k2+1+n2]*B[k2 ]; + } + + // decode kernel to output + // determined the following value experimentally + // (by first figuring out what made inverse_mdct_slow work); then matching that here + // (probably vorbis encoder premultiplies by n or n/2, to save it on the decoder?) + s = 0.5; // theoretically would be n4 + + // [[[ note! the s value of 0.5 is compensated for by the B[] in the current code, + // so it needs to use the "old" B values to behave correctly, or else + // set s to 1.0 ]]] + for (i=0; i < n4 ; ++i) buffer[i] = s * X[i+n4]; + for ( ; i < n3_4; ++i) buffer[i] = -s * X[n3_4 - i - 1]; + for ( ; i < n ; ++i) buffer[i] = -s * X[i - n3_4]; +} +#endif + +static float *get_window(vorb *f, int len) +{ + len <<= 1; + if (len == f->blocksize_0) return f->window[0]; + if (len == f->blocksize_1) return f->window[1]; + assert(0); + return NULL; +} + +#ifndef STB_VORBIS_NO_DEFER_FLOOR +typedef int16 YTYPE; +#else +typedef int YTYPE; +#endif +static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *finalY, uint8 *step2_flag) +{ + int n2 = n >> 1; + int s = map->chan[i].mux, floor; + floor = map->submap_floor[s]; + if (f->floor_types[floor] == 0) { + return error(f, VORBIS_invalid_stream); + } else { + Floor1 *g = &f->floor_config[floor].floor1; + int j,q; + int lx = 0, ly = finalY[0] * g->floor1_multiplier; + for (q=1; q < g->values; ++q) { + j = g->sorted_order[q]; + #ifndef STB_VORBIS_NO_DEFER_FLOOR + if (finalY[j] >= 0) + #else + if (step2_flag[j]) + #endif + { + int hy = finalY[j] * g->floor1_multiplier; + int hx = g->Xlist[j]; + if (lx != hx) + draw_line(target, lx,ly, hx,hy, n2); + CHECK(f); + lx = hx, ly = hy; + } + } + if (lx < n2) { + // optimization of: draw_line(target, lx,ly, n,ly, n2); + for (j=lx; j < n2; ++j) + LINE_OP(target[j], inverse_db_table[ly]); + CHECK(f); + } + } + return TRUE; +} + +// The meaning of "left" and "right" +// +// For a given frame: +// we compute samples from 0..n +// window_center is n/2 +// we'll window and mix the samples from left_start to left_end with data from the previous frame +// all of the samples from left_end to right_start can be output without mixing; however, +// this interval is 0-length except when transitioning between short and long frames +// all of the samples from right_start to right_end need to be mixed with the next frame, +// which we don't have, so those get saved in a buffer +// frame N's right_end-right_start, the number of samples to mix with the next frame, +// has to be the same as frame N+1's left_end-left_start (which they are by +// construction) + +static int vorbis_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode) +{ + Mode *m; + int i, n, prev, next, window_center; + f->channel_buffer_start = f->channel_buffer_end = 0; + + retry: + if (f->eof) return FALSE; + if (!maybe_start_packet(f)) + return FALSE; + // check packet type + if (get_bits(f,1) != 0) { + if (IS_PUSH_MODE(f)) + return error(f,VORBIS_bad_packet_type); + while (EOP != get8_packet(f)); + goto retry; + } + + if (f->alloc.alloc_buffer) + assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); + + i = get_bits(f, ilog(f->mode_count-1)); + if (i == EOP) return FALSE; + if (i >= f->mode_count) return FALSE; + *mode = i; + m = f->mode_config + i; + if (m->blockflag) { + n = f->blocksize_1; + prev = get_bits(f,1); + next = get_bits(f,1); + } else { + prev = next = 0; + n = f->blocksize_0; + } + +// WINDOWING + + window_center = n >> 1; + if (m->blockflag && !prev) { + *p_left_start = (n - f->blocksize_0) >> 2; + *p_left_end = (n + f->blocksize_0) >> 2; + } else { + *p_left_start = 0; + *p_left_end = window_center; + } + if (m->blockflag && !next) { + *p_right_start = (n*3 - f->blocksize_0) >> 2; + *p_right_end = (n*3 + f->blocksize_0) >> 2; + } else { + *p_right_start = window_center; + *p_right_end = n; + } + + return TRUE; +} + +static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start, int left_end, int right_start, int right_end, int *p_left) +{ + Mapping *map; + int i,j,k,n,n2; + int zero_channel[256]; + int really_zero_channel[256]; + +// WINDOWING + + n = f->blocksize[m->blockflag]; + map = &f->mapping[m->mapping]; + +// FLOORS + n2 = n >> 1; + + CHECK(f); + + for (i=0; i < f->channels; ++i) { + int s = map->chan[i].mux, floor; + zero_channel[i] = FALSE; + floor = map->submap_floor[s]; + if (f->floor_types[floor] == 0) { + return error(f, VORBIS_invalid_stream); + } else { + Floor1 *g = &f->floor_config[floor].floor1; + if (get_bits(f, 1)) { + short *finalY; + uint8 step2_flag[256]; + static int range_list[4] = { 256, 128, 86, 64 }; + int range = range_list[g->floor1_multiplier-1]; + int offset = 2; + finalY = f->finalY[i]; + finalY[0] = get_bits(f, ilog(range)-1); + finalY[1] = get_bits(f, ilog(range)-1); + for (j=0; j < g->partitions; ++j) { + int pclass = g->partition_class_list[j]; + int cdim = g->class_dimensions[pclass]; + int cbits = g->class_subclasses[pclass]; + int csub = (1 << cbits)-1; + int cval = 0; + if (cbits) { + Codebook *c = f->codebooks + g->class_masterbooks[pclass]; + DECODE(cval,f,c); + } + for (k=0; k < cdim; ++k) { + int book = g->subclass_books[pclass][cval & csub]; + cval = cval >> cbits; + if (book >= 0) { + int temp; + Codebook *c = f->codebooks + book; + DECODE(temp,f,c); + finalY[offset++] = temp; + } else + finalY[offset++] = 0; + } + } + if (f->valid_bits == INVALID_BITS) goto error; // behavior according to spec + step2_flag[0] = step2_flag[1] = 1; + for (j=2; j < g->values; ++j) { + int low, high, pred, highroom, lowroom, room, val; + low = g->neighbors[j][0]; + high = g->neighbors[j][1]; + //neighbors(g->Xlist, j, &low, &high); + pred = predict_point(g->Xlist[j], g->Xlist[low], g->Xlist[high], finalY[low], finalY[high]); + val = finalY[j]; + highroom = range - pred; + lowroom = pred; + if (highroom < lowroom) + room = highroom * 2; + else + room = lowroom * 2; + if (val) { + step2_flag[low] = step2_flag[high] = 1; + step2_flag[j] = 1; + if (val >= room) + if (highroom > lowroom) + finalY[j] = val - lowroom + pred; + else + finalY[j] = pred - val + highroom - 1; + else + if (val & 1) + finalY[j] = pred - ((val+1)>>1); + else + finalY[j] = pred + (val>>1); + } else { + step2_flag[j] = 0; + finalY[j] = pred; + } + } + +#ifdef STB_VORBIS_NO_DEFER_FLOOR + do_floor(f, map, i, n, f->floor_buffers[i], finalY, step2_flag); +#else + // defer final floor computation until _after_ residue + for (j=0; j < g->values; ++j) { + if (!step2_flag[j]) + finalY[j] = -1; + } +#endif + } else { + error: + zero_channel[i] = TRUE; + } + // So we just defer everything else to later + + // at this point we've decoded the floor into buffer + } + } + CHECK(f); + // at this point we've decoded all floors + + if (f->alloc.alloc_buffer) + assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); + + // re-enable coupled channels if necessary + memcpy(really_zero_channel, zero_channel, sizeof(really_zero_channel[0]) * f->channels); + for (i=0; i < map->coupling_steps; ++i) + if (!zero_channel[map->chan[i].magnitude] || !zero_channel[map->chan[i].angle]) { + zero_channel[map->chan[i].magnitude] = zero_channel[map->chan[i].angle] = FALSE; + } + + CHECK(f); +// RESIDUE DECODE + for (i=0; i < map->submaps; ++i) { + float *residue_buffers[STB_VORBIS_MAX_CHANNELS]; + int r; + uint8 do_not_decode[256]; + int ch = 0; + for (j=0; j < f->channels; ++j) { + if (map->chan[j].mux == i) { + if (zero_channel[j]) { + do_not_decode[ch] = TRUE; + residue_buffers[ch] = NULL; + } else { + do_not_decode[ch] = FALSE; + residue_buffers[ch] = f->channel_buffers[j]; + } + ++ch; + } + } + r = map->submap_residue[i]; + decode_residue(f, residue_buffers, ch, n2, r, do_not_decode); + } + + if (f->alloc.alloc_buffer) + assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); + CHECK(f); + +// INVERSE COUPLING + for (i = map->coupling_steps-1; i >= 0; --i) { + int n2 = n >> 1; + float *m = f->channel_buffers[map->chan[i].magnitude]; + float *a = f->channel_buffers[map->chan[i].angle ]; + for (j=0; j < n2; ++j) { + float a2,m2; + if (m[j] > 0) + if (a[j] > 0) + m2 = m[j], a2 = m[j] - a[j]; + else + a2 = m[j], m2 = m[j] + a[j]; + else + if (a[j] > 0) + m2 = m[j], a2 = m[j] + a[j]; + else + a2 = m[j], m2 = m[j] - a[j]; + m[j] = m2; + a[j] = a2; + } + } + CHECK(f); + + // finish decoding the floors +#ifndef STB_VORBIS_NO_DEFER_FLOOR + for (i=0; i < f->channels; ++i) { + if (really_zero_channel[i]) { + memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2); + } else { + do_floor(f, map, i, n, f->channel_buffers[i], f->finalY[i], NULL); + } + } +#else + for (i=0; i < f->channels; ++i) { + if (really_zero_channel[i]) { + memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2); + } else { + for (j=0; j < n2; ++j) + f->channel_buffers[i][j] *= f->floor_buffers[i][j]; + } + } +#endif + +// INVERSE MDCT + CHECK(f); + for (i=0; i < f->channels; ++i) + inverse_mdct(f->channel_buffers[i], n, f, m->blockflag); + CHECK(f); + + // this shouldn't be necessary, unless we exited on an error + // and want to flush to get to the next packet + flush_packet(f); + + if (f->first_decode) { + // assume we start so first non-discarded sample is sample 0 + // this isn't to spec, but spec would require us to read ahead + // and decode the size of all current frames--could be done, + // but presumably it's not a commonly used feature + f->current_loc = -n2; // start of first frame is positioned for discard + // we might have to discard samples "from" the next frame too, + // if we're lapping a large block then a small at the start? + f->discard_samples_deferred = n - right_end; + f->current_loc_valid = TRUE; + f->first_decode = FALSE; + } else if (f->discard_samples_deferred) { + if (f->discard_samples_deferred >= right_start - left_start) { + f->discard_samples_deferred -= (right_start - left_start); + left_start = right_start; + *p_left = left_start; + } else { + left_start += f->discard_samples_deferred; + *p_left = left_start; + f->discard_samples_deferred = 0; + } + } else if (f->previous_length == 0 && f->current_loc_valid) { + // we're recovering from a seek... that means we're going to discard + // the samples from this packet even though we know our position from + // the last page header, so we need to update the position based on + // the discarded samples here + // but wait, the code below is going to add this in itself even + // on a discard, so we don't need to do it here... + } + + // check if we have ogg information about the sample # for this packet + if (f->last_seg_which == f->end_seg_with_known_loc) { + // if we have a valid current loc, and this is final: + if (f->current_loc_valid && (f->page_flag & PAGEFLAG_last_page)) { + uint32 current_end = f->known_loc_for_packet - (n-right_end); + // then let's infer the size of the (probably) short final frame + if (current_end < f->current_loc + (right_end-left_start)) { + if (current_end < f->current_loc) { + // negative truncation, that's impossible! + *len = 0; + } else { + *len = current_end - f->current_loc; + } + *len += left_start; + if (*len > right_end) *len = right_end; // this should never happen + f->current_loc += *len; + return TRUE; + } + } + // otherwise, just set our sample loc + // guess that the ogg granule pos refers to the _middle_ of the + // last frame? + // set f->current_loc to the position of left_start + f->current_loc = f->known_loc_for_packet - (n2-left_start); + f->current_loc_valid = TRUE; + } + if (f->current_loc_valid) + f->current_loc += (right_start - left_start); + + if (f->alloc.alloc_buffer) + assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); + *len = right_end; // ignore samples after the window goes to 0 + CHECK(f); + + return TRUE; +} + +static int vorbis_decode_packet(vorb *f, int *len, int *p_left, int *p_right) +{ + int mode, left_end, right_end; + if (!vorbis_decode_initial(f, p_left, &left_end, p_right, &right_end, &mode)) return 0; + return vorbis_decode_packet_rest(f, len, f->mode_config + mode, *p_left, left_end, *p_right, right_end, p_left); +} + +static int vorbis_finish_frame(stb_vorbis *f, int len, int left, int right) +{ + int prev,i,j; + // we use right&left (the start of the right- and left-window sin()-regions) + // to determine how much to return, rather than inferring from the rules + // (same result, clearer code); 'left' indicates where our sin() window + // starts, therefore where the previous window's right edge starts, and + // therefore where to start mixing from the previous buffer. 'right' + // indicates where our sin() ending-window starts, therefore that's where + // we start saving, and where our returned-data ends. + + // mixin from previous window + if (f->previous_length) { + int i,j, n = f->previous_length; + float *w = get_window(f, n); + for (i=0; i < f->channels; ++i) { + for (j=0; j < n; ++j) + f->channel_buffers[i][left+j] = + f->channel_buffers[i][left+j]*w[ j] + + f->previous_window[i][ j]*w[n-1-j]; + } + } + + prev = f->previous_length; + + // last half of this data becomes previous window + f->previous_length = len - right; + + // @OPTIMIZE: could avoid this copy by double-buffering the + // output (flipping previous_window with channel_buffers), but + // then previous_window would have to be 2x as large, and + // channel_buffers couldn't be temp mem (although they're NOT + // currently temp mem, they could be (unless we want to level + // performance by spreading out the computation)) + for (i=0; i < f->channels; ++i) + for (j=0; right+j < len; ++j) + f->previous_window[i][j] = f->channel_buffers[i][right+j]; + + if (!prev) + // there was no previous packet, so this data isn't valid... + // this isn't entirely true, only the would-have-overlapped data + // isn't valid, but this seems to be what the spec requires + return 0; + + // truncate a short frame + if (len < right) right = len; + + f->samples_output += right-left; + + return right - left; +} + +static void vorbis_pump_first_frame(stb_vorbis *f) +{ + int len, right, left; + if (vorbis_decode_packet(f, &len, &left, &right)) + vorbis_finish_frame(f, len, left, right); +} + +#ifndef STB_VORBIS_NO_PUSHDATA_API +static int is_whole_packet_present(stb_vorbis *f, int end_page) +{ + // make sure that we have the packet available before continuing... + // this requires a full ogg parse, but we know we can fetch from f->stream + + // instead of coding this out explicitly, we could save the current read state, + // read the next packet with get8() until end-of-packet, check f->eof, then + // reset the state? but that would be slower, esp. since we'd have over 256 bytes + // of state to restore (primarily the page segment table) + + int s = f->next_seg, first = TRUE; + uint8 *p = f->stream; + + if (s != -1) { // if we're not starting the packet with a 'continue on next page' flag + for (; s < f->segment_count; ++s) { + p += f->segments[s]; + if (f->segments[s] < 255) // stop at first short segment + break; + } + // either this continues, or it ends it... + if (end_page) + if (s < f->segment_count-1) return error(f, VORBIS_invalid_stream); + if (s == f->segment_count) + s = -1; // set 'crosses page' flag + if (p > f->stream_end) return error(f, VORBIS_need_more_data); + first = FALSE; + } + for (; s == -1;) { + uint8 *q; + int n; + + // check that we have the page header ready + if (p + 26 >= f->stream_end) return error(f, VORBIS_need_more_data); + // validate the page + if (memcmp(p, ogg_page_header, 4)) return error(f, VORBIS_invalid_stream); + if (p[4] != 0) return error(f, VORBIS_invalid_stream); + if (first) { // the first segment must NOT have 'continued_packet', later ones MUST + if (f->previous_length) + if ((p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream); + // if no previous length, we're resynching, so we can come in on a continued-packet, + // which we'll just drop + } else { + if (!(p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream); + } + n = p[26]; // segment counts + q = p+27; // q points to segment table + p = q + n; // advance past header + // make sure we've read the segment table + if (p > f->stream_end) return error(f, VORBIS_need_more_data); + for (s=0; s < n; ++s) { + p += q[s]; + if (q[s] < 255) + break; + } + if (end_page) + if (s < n-1) return error(f, VORBIS_invalid_stream); + if (s == n) + s = -1; // set 'crosses page' flag + if (p > f->stream_end) return error(f, VORBIS_need_more_data); + first = FALSE; + } + return TRUE; +} +#endif // !STB_VORBIS_NO_PUSHDATA_API + +static int start_decoder(vorb *f) +{ + uint8 header[6], x,y; + int len,i,j,k, max_submaps = 0; + int longest_floorlist=0; + + // first page, first packet + + if (!start_page(f)) return FALSE; + // validate page flag + if (!(f->page_flag & PAGEFLAG_first_page)) return error(f, VORBIS_invalid_first_page); + if (f->page_flag & PAGEFLAG_last_page) return error(f, VORBIS_invalid_first_page); + if (f->page_flag & PAGEFLAG_continued_packet) return error(f, VORBIS_invalid_first_page); + // check for expected packet length + if (f->segment_count != 1) return error(f, VORBIS_invalid_first_page); + if (f->segments[0] != 30) return error(f, VORBIS_invalid_first_page); + // read packet + // check packet header + if (get8(f) != VORBIS_packet_id) return error(f, VORBIS_invalid_first_page); + if (!getn(f, header, 6)) return error(f, VORBIS_unexpected_eof); + if (!vorbis_validate(header)) return error(f, VORBIS_invalid_first_page); + // vorbis_version + if (get32(f) != 0) return error(f, VORBIS_invalid_first_page); + f->channels = get8(f); if (!f->channels) return error(f, VORBIS_invalid_first_page); + if (f->channels > STB_VORBIS_MAX_CHANNELS) return error(f, VORBIS_too_many_channels); + f->sample_rate = get32(f); if (!f->sample_rate) return error(f, VORBIS_invalid_first_page); + get32(f); // bitrate_maximum + get32(f); // bitrate_nominal + get32(f); // bitrate_minimum + x = get8(f); + { + int log0,log1; + log0 = x & 15; + log1 = x >> 4; + f->blocksize_0 = 1 << log0; + f->blocksize_1 = 1 << log1; + if (log0 < 6 || log0 > 13) return error(f, VORBIS_invalid_setup); + if (log1 < 6 || log1 > 13) return error(f, VORBIS_invalid_setup); + if (log0 > log1) return error(f, VORBIS_invalid_setup); + } + + // framing_flag + x = get8(f); + if (!(x & 1)) return error(f, VORBIS_invalid_first_page); + + // second packet! + if (!start_page(f)) return FALSE; + + if (!start_packet(f)) return FALSE; + do { + len = next_segment(f); + skip(f, len); + f->bytes_in_seg = 0; + } while (len); + + // third packet! + if (!start_packet(f)) return FALSE; + + #ifndef STB_VORBIS_NO_PUSHDATA_API + if (IS_PUSH_MODE(f)) { + if (!is_whole_packet_present(f, TRUE)) { + // convert error in ogg header to write type + if (f->error == VORBIS_invalid_stream) + f->error = VORBIS_invalid_setup; + return FALSE; + } + } + #endif + + crc32_init(); // always init it, to avoid multithread race conditions + + if (get8_packet(f) != VORBIS_packet_setup) return error(f, VORBIS_invalid_setup); + for (i=0; i < 6; ++i) header[i] = get8_packet(f); + if (!vorbis_validate(header)) return error(f, VORBIS_invalid_setup); + + // codebooks + + f->codebook_count = get_bits(f,8) + 1; + f->codebooks = (Codebook *) setup_malloc(f, sizeof(*f->codebooks) * f->codebook_count); + if (f->codebooks == NULL) return error(f, VORBIS_outofmem); + memset(f->codebooks, 0, sizeof(*f->codebooks) * f->codebook_count); + for (i=0; i < f->codebook_count; ++i) { + uint32 *values; + int ordered, sorted_count; + int total=0; + uint8 *lengths; + Codebook *c = f->codebooks+i; + CHECK(f); + x = get_bits(f, 8); if (x != 0x42) return error(f, VORBIS_invalid_setup); + x = get_bits(f, 8); if (x != 0x43) return error(f, VORBIS_invalid_setup); + x = get_bits(f, 8); if (x != 0x56) return error(f, VORBIS_invalid_setup); + x = get_bits(f, 8); + c->dimensions = (get_bits(f, 8)<<8) + x; + x = get_bits(f, 8); + y = get_bits(f, 8); + c->entries = (get_bits(f, 8)<<16) + (y<<8) + x; + ordered = get_bits(f,1); + c->sparse = ordered ? 0 : get_bits(f,1); + + if (c->dimensions == 0 && c->entries != 0) return error(f, VORBIS_invalid_setup); + + if (c->sparse) + lengths = (uint8 *) setup_temp_malloc(f, c->entries); + else + lengths = c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries); + + if (!lengths) return error(f, VORBIS_outofmem); + + if (ordered) { + int current_entry = 0; + int current_length = get_bits(f,5) + 1; + while (current_entry < c->entries) { + int limit = c->entries - current_entry; + int n = get_bits(f, ilog(limit)); + if (current_entry + n > (int) c->entries) { return error(f, VORBIS_invalid_setup); } + memset(lengths + current_entry, current_length, n); + current_entry += n; + ++current_length; + } + } else { + for (j=0; j < c->entries; ++j) { + int present = c->sparse ? get_bits(f,1) : 1; + if (present) { + lengths[j] = get_bits(f, 5) + 1; + ++total; + if (lengths[j] == 32) + return error(f, VORBIS_invalid_setup); + } else { + lengths[j] = NO_CODE; + } + } + } + + if (c->sparse && total >= c->entries >> 2) { + // convert sparse items to non-sparse! + if (c->entries > (int) f->setup_temp_memory_required) + f->setup_temp_memory_required = c->entries; + + c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries); + if (c->codeword_lengths == NULL) return error(f, VORBIS_outofmem); + memcpy(c->codeword_lengths, lengths, c->entries); + setup_temp_free(f, lengths, c->entries); // note this is only safe if there have been no intervening temp mallocs! + lengths = c->codeword_lengths; + c->sparse = 0; + } + + // compute the size of the sorted tables + if (c->sparse) { + sorted_count = total; + } else { + sorted_count = 0; + #ifndef STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH + for (j=0; j < c->entries; ++j) + if (lengths[j] > STB_VORBIS_FAST_HUFFMAN_LENGTH && lengths[j] != NO_CODE) + ++sorted_count; + #endif + } + + c->sorted_entries = sorted_count; + values = NULL; + + CHECK(f); + if (!c->sparse) { + c->codewords = (uint32 *) setup_malloc(f, sizeof(c->codewords[0]) * c->entries); + if (!c->codewords) return error(f, VORBIS_outofmem); + } else { + unsigned int size; + if (c->sorted_entries) { + c->codeword_lengths = (uint8 *) setup_malloc(f, c->sorted_entries); + if (!c->codeword_lengths) return error(f, VORBIS_outofmem); + c->codewords = (uint32 *) setup_temp_malloc(f, sizeof(*c->codewords) * c->sorted_entries); + if (!c->codewords) return error(f, VORBIS_outofmem); + values = (uint32 *) setup_temp_malloc(f, sizeof(*values) * c->sorted_entries); + if (!values) return error(f, VORBIS_outofmem); + } + size = c->entries + (sizeof(*c->codewords) + sizeof(*values)) * c->sorted_entries; + if (size > f->setup_temp_memory_required) + f->setup_temp_memory_required = size; + } + + if (!compute_codewords(c, lengths, c->entries, values)) { + if (c->sparse) setup_temp_free(f, values, 0); + return error(f, VORBIS_invalid_setup); + } + + if (c->sorted_entries) { + // allocate an extra slot for sentinels + c->sorted_codewords = (uint32 *) setup_malloc(f, sizeof(*c->sorted_codewords) * (c->sorted_entries+1)); + if (c->sorted_codewords == NULL) return error(f, VORBIS_outofmem); + // allocate an extra slot at the front so that c->sorted_values[-1] is defined + // so that we can catch that case without an extra if + c->sorted_values = ( int *) setup_malloc(f, sizeof(*c->sorted_values ) * (c->sorted_entries+1)); + if (c->sorted_values == NULL) return error(f, VORBIS_outofmem); + ++c->sorted_values; + c->sorted_values[-1] = -1; + compute_sorted_huffman(c, lengths, values); + } + + if (c->sparse) { + setup_temp_free(f, values, sizeof(*values)*c->sorted_entries); + setup_temp_free(f, c->codewords, sizeof(*c->codewords)*c->sorted_entries); + setup_temp_free(f, lengths, c->entries); + c->codewords = NULL; + } + + compute_accelerated_huffman(c); + + CHECK(f); + c->lookup_type = get_bits(f, 4); + if (c->lookup_type > 2) return error(f, VORBIS_invalid_setup); + if (c->lookup_type > 0) { + uint16 *mults; + c->minimum_value = float32_unpack(get_bits(f, 32)); + c->delta_value = float32_unpack(get_bits(f, 32)); + c->value_bits = get_bits(f, 4)+1; + c->sequence_p = get_bits(f,1); + if (c->lookup_type == 1) { + c->lookup_values = lookup1_values(c->entries, c->dimensions); + } else { + c->lookup_values = c->entries * c->dimensions; + } + if (c->lookup_values == 0) return error(f, VORBIS_invalid_setup); + mults = (uint16 *) setup_temp_malloc(f, sizeof(mults[0]) * c->lookup_values); + if (mults == NULL) return error(f, VORBIS_outofmem); + for (j=0; j < (int) c->lookup_values; ++j) { + int q = get_bits(f, c->value_bits); + if (q == EOP) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_invalid_setup); } + mults[j] = q; + } + +#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (c->lookup_type == 1) { + int len, sparse = c->sparse; + float last=0; + // pre-expand the lookup1-style multiplicands, to avoid a divide in the inner loop + if (sparse) { + if (c->sorted_entries == 0) goto skip; + c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->sorted_entries * c->dimensions); + } else + c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->entries * c->dimensions); + if (c->multiplicands == NULL) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); } + len = sparse ? c->sorted_entries : c->entries; + for (j=0; j < len; ++j) { + unsigned int z = sparse ? c->sorted_values[j] : j; + unsigned int div=1; + for (k=0; k < c->dimensions; ++k) { + int off = (z / div) % c->lookup_values; + float val = mults[off]; + val = mults[off]*c->delta_value + c->minimum_value + last; + c->multiplicands[j*c->dimensions + k] = val; + if (c->sequence_p) + last = val; + if (k+1 < c->dimensions) { + if (div > UINT_MAX / (unsigned int) c->lookup_values) { + setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); + return error(f, VORBIS_invalid_setup); + } + div *= c->lookup_values; + } + } + } + c->lookup_type = 2; + } + else +#endif + { + float last=0; + CHECK(f); + c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values); + if (c->multiplicands == NULL) { setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); } + for (j=0; j < (int) c->lookup_values; ++j) { + float val = mults[j] * c->delta_value + c->minimum_value + last; + c->multiplicands[j] = val; + if (c->sequence_p) + last = val; + } + } +#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK + skip:; +#endif + setup_temp_free(f, mults, sizeof(mults[0])*c->lookup_values); + + CHECK(f); + } + CHECK(f); + } + + // time domain transfers (notused) + + x = get_bits(f, 6) + 1; + for (i=0; i < x; ++i) { + uint32 z = get_bits(f, 16); + if (z != 0) return error(f, VORBIS_invalid_setup); + } + + // Floors + f->floor_count = get_bits(f, 6)+1; + f->floor_config = (Floor *) setup_malloc(f, f->floor_count * sizeof(*f->floor_config)); + if (f->floor_config == NULL) return error(f, VORBIS_outofmem); + for (i=0; i < f->floor_count; ++i) { + f->floor_types[i] = get_bits(f, 16); + if (f->floor_types[i] > 1) return error(f, VORBIS_invalid_setup); + if (f->floor_types[i] == 0) { + Floor0 *g = &f->floor_config[i].floor0; + g->order = get_bits(f,8); + g->rate = get_bits(f,16); + g->bark_map_size = get_bits(f,16); + g->amplitude_bits = get_bits(f,6); + g->amplitude_offset = get_bits(f,8); + g->number_of_books = get_bits(f,4) + 1; + for (j=0; j < g->number_of_books; ++j) + g->book_list[j] = get_bits(f,8); + return error(f, VORBIS_feature_not_supported); + } else { + Point p[31*8+2]; + Floor1 *g = &f->floor_config[i].floor1; + int max_class = -1; + g->partitions = get_bits(f, 5); + for (j=0; j < g->partitions; ++j) { + g->partition_class_list[j] = get_bits(f, 4); + if (g->partition_class_list[j] > max_class) + max_class = g->partition_class_list[j]; + } + for (j=0; j <= max_class; ++j) { + g->class_dimensions[j] = get_bits(f, 3)+1; + g->class_subclasses[j] = get_bits(f, 2); + if (g->class_subclasses[j]) { + g->class_masterbooks[j] = get_bits(f, 8); + if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup); + } + for (k=0; k < 1 << g->class_subclasses[j]; ++k) { + g->subclass_books[j][k] = get_bits(f,8)-1; + if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup); + } + } + g->floor1_multiplier = get_bits(f,2)+1; + g->rangebits = get_bits(f,4); + g->Xlist[0] = 0; + g->Xlist[1] = 1 << g->rangebits; + g->values = 2; + for (j=0; j < g->partitions; ++j) { + int c = g->partition_class_list[j]; + for (k=0; k < g->class_dimensions[c]; ++k) { + g->Xlist[g->values] = get_bits(f, g->rangebits); + ++g->values; + } + } + // precompute the sorting + for (j=0; j < g->values; ++j) { + p[j].x = g->Xlist[j]; + p[j].y = j; + } + qsort(p, g->values, sizeof(p[0]), point_compare); + for (j=0; j < g->values; ++j) + g->sorted_order[j] = (uint8) p[j].y; + // precompute the neighbors + for (j=2; j < g->values; ++j) { + int low,hi; + neighbors(g->Xlist, j, &low,&hi); + g->neighbors[j][0] = low; + g->neighbors[j][1] = hi; + } + + if (g->values > longest_floorlist) + longest_floorlist = g->values; + } + } + + // Residue + f->residue_count = get_bits(f, 6)+1; + f->residue_config = (Residue *) setup_malloc(f, f->residue_count * sizeof(f->residue_config[0])); + if (f->residue_config == NULL) return error(f, VORBIS_outofmem); + memset(f->residue_config, 0, f->residue_count * sizeof(f->residue_config[0])); + for (i=0; i < f->residue_count; ++i) { + uint8 residue_cascade[64]; + Residue *r = f->residue_config+i; + f->residue_types[i] = get_bits(f, 16); + if (f->residue_types[i] > 2) return error(f, VORBIS_invalid_setup); + r->begin = get_bits(f, 24); + r->end = get_bits(f, 24); + if (r->end < r->begin) return error(f, VORBIS_invalid_setup); + r->part_size = get_bits(f,24)+1; + r->classifications = get_bits(f,6)+1; + r->classbook = get_bits(f,8); + if (r->classbook >= f->codebook_count) return error(f, VORBIS_invalid_setup); + for (j=0; j < r->classifications; ++j) { + uint8 high_bits=0; + uint8 low_bits=get_bits(f,3); + if (get_bits(f,1)) + high_bits = get_bits(f,5); + residue_cascade[j] = high_bits*8 + low_bits; + } + r->residue_books = (short (*)[8]) setup_malloc(f, sizeof(r->residue_books[0]) * r->classifications); + if (r->residue_books == NULL) return error(f, VORBIS_outofmem); + for (j=0; j < r->classifications; ++j) { + for (k=0; k < 8; ++k) { + if (residue_cascade[j] & (1 << k)) { + r->residue_books[j][k] = get_bits(f, 8); + if (r->residue_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup); + } else { + r->residue_books[j][k] = -1; + } + } + } + // precompute the classifications[] array to avoid inner-loop mod/divide + // call it 'classdata' since we already have r->classifications + r->classdata = (uint8 **) setup_malloc(f, sizeof(*r->classdata) * f->codebooks[r->classbook].entries); + if (!r->classdata) return error(f, VORBIS_outofmem); + memset(r->classdata, 0, sizeof(*r->classdata) * f->codebooks[r->classbook].entries); + for (j=0; j < f->codebooks[r->classbook].entries; ++j) { + int classwords = f->codebooks[r->classbook].dimensions; + int temp = j; + r->classdata[j] = (uint8 *) setup_malloc(f, sizeof(r->classdata[j][0]) * classwords); + if (r->classdata[j] == NULL) return error(f, VORBIS_outofmem); + for (k=classwords-1; k >= 0; --k) { + r->classdata[j][k] = temp % r->classifications; + temp /= r->classifications; + } + } + } + + f->mapping_count = get_bits(f,6)+1; + f->mapping = (Mapping *) setup_malloc(f, f->mapping_count * sizeof(*f->mapping)); + if (f->mapping == NULL) return error(f, VORBIS_outofmem); + memset(f->mapping, 0, f->mapping_count * sizeof(*f->mapping)); + for (i=0; i < f->mapping_count; ++i) { + Mapping *m = f->mapping + i; + int mapping_type = get_bits(f,16); + if (mapping_type != 0) return error(f, VORBIS_invalid_setup); + m->chan = (MappingChannel *) setup_malloc(f, f->channels * sizeof(*m->chan)); + if (m->chan == NULL) return error(f, VORBIS_outofmem); + if (get_bits(f,1)) + m->submaps = get_bits(f,4)+1; + else + m->submaps = 1; + if (m->submaps > max_submaps) + max_submaps = m->submaps; + if (get_bits(f,1)) { + m->coupling_steps = get_bits(f,8)+1; + for (k=0; k < m->coupling_steps; ++k) { + m->chan[k].magnitude = get_bits(f, ilog(f->channels-1)); + m->chan[k].angle = get_bits(f, ilog(f->channels-1)); + if (m->chan[k].magnitude >= f->channels) return error(f, VORBIS_invalid_setup); + if (m->chan[k].angle >= f->channels) return error(f, VORBIS_invalid_setup); + if (m->chan[k].magnitude == m->chan[k].angle) return error(f, VORBIS_invalid_setup); + } + } else + m->coupling_steps = 0; + + // reserved field + if (get_bits(f,2)) return error(f, VORBIS_invalid_setup); + if (m->submaps > 1) { + for (j=0; j < f->channels; ++j) { + m->chan[j].mux = get_bits(f, 4); + if (m->chan[j].mux >= m->submaps) return error(f, VORBIS_invalid_setup); + } + } else + // @SPECIFICATION: this case is missing from the spec + for (j=0; j < f->channels; ++j) + m->chan[j].mux = 0; + + for (j=0; j < m->submaps; ++j) { + get_bits(f,8); // discard + m->submap_floor[j] = get_bits(f,8); + m->submap_residue[j] = get_bits(f,8); + if (m->submap_floor[j] >= f->floor_count) return error(f, VORBIS_invalid_setup); + if (m->submap_residue[j] >= f->residue_count) return error(f, VORBIS_invalid_setup); + } + } + + // Modes + f->mode_count = get_bits(f, 6)+1; + for (i=0; i < f->mode_count; ++i) { + Mode *m = f->mode_config+i; + m->blockflag = get_bits(f,1); + m->windowtype = get_bits(f,16); + m->transformtype = get_bits(f,16); + m->mapping = get_bits(f,8); + if (m->windowtype != 0) return error(f, VORBIS_invalid_setup); + if (m->transformtype != 0) return error(f, VORBIS_invalid_setup); + if (m->mapping >= f->mapping_count) return error(f, VORBIS_invalid_setup); + } + + flush_packet(f); + + f->previous_length = 0; + + for (i=0; i < f->channels; ++i) { + f->channel_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1); + f->previous_window[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); + f->finalY[i] = (int16 *) setup_malloc(f, sizeof(int16) * longest_floorlist); + if (f->channel_buffers[i] == NULL || f->previous_window[i] == NULL || f->finalY[i] == NULL) return error(f, VORBIS_outofmem); + #ifdef STB_VORBIS_NO_DEFER_FLOOR + f->floor_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); + if (f->floor_buffers[i] == NULL) return error(f, VORBIS_outofmem); + #endif + } + + if (!init_blocksize(f, 0, f->blocksize_0)) return FALSE; + if (!init_blocksize(f, 1, f->blocksize_1)) return FALSE; + f->blocksize[0] = f->blocksize_0; + f->blocksize[1] = f->blocksize_1; + +#ifdef STB_VORBIS_DIVIDE_TABLE + if (integer_divide_table[1][1]==0) + for (i=0; i < DIVTAB_NUMER; ++i) + for (j=1; j < DIVTAB_DENOM; ++j) + integer_divide_table[i][j] = i / j; +#endif + + // compute how much temporary memory is needed + + // 1. + { + uint32 imdct_mem = (f->blocksize_1 * sizeof(float) >> 1); + uint32 classify_mem; + int i,max_part_read=0; + for (i=0; i < f->residue_count; ++i) { + Residue *r = f->residue_config + i; + int n_read = r->end - r->begin; + int part_read = n_read / r->part_size; + if (part_read > max_part_read) + max_part_read = part_read; + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(uint8 *)); + #else + classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(int *)); + #endif + + f->temp_memory_required = classify_mem; + if (imdct_mem > f->temp_memory_required) + f->temp_memory_required = imdct_mem; + } + + f->first_decode = TRUE; + + if (f->alloc.alloc_buffer) { + assert(f->temp_offset == f->alloc.alloc_buffer_length_in_bytes); + // check if there's enough temp memory so we don't error later + if (f->setup_offset + sizeof(*f) + f->temp_memory_required > (unsigned) f->temp_offset) + return error(f, VORBIS_outofmem); + } + + f->first_audio_page_offset = stb_vorbis_get_file_offset(f); + + return TRUE; +} + +static void vorbis_deinit(stb_vorbis *p) +{ + int i,j; + if (p->residue_config) { + for (i=0; i < p->residue_count; ++i) { + Residue *r = p->residue_config+i; + if (r->classdata) { + for (j=0; j < p->codebooks[r->classbook].entries; ++j) + setup_free(p, r->classdata[j]); + setup_free(p, r->classdata); + } + setup_free(p, r->residue_books); + } + } + + if (p->codebooks) { + CHECK(p); + for (i=0; i < p->codebook_count; ++i) { + Codebook *c = p->codebooks + i; + setup_free(p, c->codeword_lengths); + setup_free(p, c->multiplicands); + setup_free(p, c->codewords); + setup_free(p, c->sorted_codewords); + // c->sorted_values[-1] is the first entry in the array + setup_free(p, c->sorted_values ? c->sorted_values-1 : NULL); + } + setup_free(p, p->codebooks); + } + setup_free(p, p->floor_config); + setup_free(p, p->residue_config); + if (p->mapping) { + for (i=0; i < p->mapping_count; ++i) + setup_free(p, p->mapping[i].chan); + setup_free(p, p->mapping); + } + CHECK(p); + for (i=0; i < p->channels && i < STB_VORBIS_MAX_CHANNELS; ++i) { + setup_free(p, p->channel_buffers[i]); + setup_free(p, p->previous_window[i]); + #ifdef STB_VORBIS_NO_DEFER_FLOOR + setup_free(p, p->floor_buffers[i]); + #endif + setup_free(p, p->finalY[i]); + } + for (i=0; i < 2; ++i) { + setup_free(p, p->A[i]); + setup_free(p, p->B[i]); + setup_free(p, p->C[i]); + setup_free(p, p->window[i]); + setup_free(p, p->bit_reverse[i]); + } + #ifndef STB_VORBIS_NO_STDIO + if (p->close_on_free) fclose(p->f); + #endif +} + +void stb_vorbis_close(stb_vorbis *p) +{ + if (p == NULL) return; + vorbis_deinit(p); + setup_free(p,p); +} + +static void vorbis_init(stb_vorbis *p, const stb_vorbis_alloc *z) +{ + memset(p, 0, sizeof(*p)); // NULL out all malloc'd pointers to start + if (z) { + p->alloc = *z; + p->alloc.alloc_buffer_length_in_bytes = (p->alloc.alloc_buffer_length_in_bytes+3) & ~3; + p->temp_offset = p->alloc.alloc_buffer_length_in_bytes; + } + p->eof = 0; + p->error = VORBIS__no_error; + p->stream = NULL; + p->codebooks = NULL; + p->page_crc_tests = -1; + #ifndef STB_VORBIS_NO_STDIO + p->close_on_free = FALSE; + p->f = NULL; + #endif +} + +int stb_vorbis_get_sample_offset(stb_vorbis *f) +{ + if (f->current_loc_valid) + return f->current_loc; + else + return -1; +} + +stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f) +{ + stb_vorbis_info d; + d.channels = f->channels; + d.sample_rate = f->sample_rate; + d.setup_memory_required = f->setup_memory_required; + d.setup_temp_memory_required = f->setup_temp_memory_required; + d.temp_memory_required = f->temp_memory_required; + d.max_frame_size = f->blocksize_1 >> 1; + return d; +} + +int stb_vorbis_get_error(stb_vorbis *f) +{ + int e = f->error; + f->error = VORBIS__no_error; + return e; +} + +static stb_vorbis * vorbis_alloc(stb_vorbis *f) +{ + stb_vorbis *p = (stb_vorbis *) setup_malloc(f, sizeof(*p)); + return p; +} + +#ifndef STB_VORBIS_NO_PUSHDATA_API + +void stb_vorbis_flush_pushdata(stb_vorbis *f) +{ + f->previous_length = 0; + f->page_crc_tests = 0; + f->discard_samples_deferred = 0; + f->current_loc_valid = FALSE; + f->first_decode = FALSE; + f->samples_output = 0; + f->channel_buffer_start = 0; + f->channel_buffer_end = 0; +} + +static int vorbis_search_for_page_pushdata(vorb *f, uint8 *data, int data_len) +{ + int i,n; + for (i=0; i < f->page_crc_tests; ++i) + f->scan[i].bytes_done = 0; + + // if we have room for more scans, search for them first, because + // they may cause us to stop early if their header is incomplete + if (f->page_crc_tests < STB_VORBIS_PUSHDATA_CRC_COUNT) { + if (data_len < 4) return 0; + data_len -= 3; // need to look for 4-byte sequence, so don't miss + // one that straddles a boundary + for (i=0; i < data_len; ++i) { + if (data[i] == 0x4f) { + if (0==memcmp(data+i, ogg_page_header, 4)) { + int j,len; + uint32 crc; + // make sure we have the whole page header + if (i+26 >= data_len || i+27+data[i+26] >= data_len) { + // only read up to this page start, so hopefully we'll + // have the whole page header start next time + data_len = i; + break; + } + // ok, we have it all; compute the length of the page + len = 27 + data[i+26]; + for (j=0; j < data[i+26]; ++j) + len += data[i+27+j]; + // scan everything up to the embedded crc (which we must 0) + crc = 0; + for (j=0; j < 22; ++j) + crc = crc32_update(crc, data[i+j]); + // now process 4 0-bytes + for ( ; j < 26; ++j) + crc = crc32_update(crc, 0); + // len is the total number of bytes we need to scan + n = f->page_crc_tests++; + f->scan[n].bytes_left = len-j; + f->scan[n].crc_so_far = crc; + f->scan[n].goal_crc = data[i+22] + (data[i+23] << 8) + (data[i+24]<<16) + (data[i+25]<<24); + // if the last frame on a page is continued to the next, then + // we can't recover the sample_loc immediately + if (data[i+27+data[i+26]-1] == 255) + f->scan[n].sample_loc = ~0; + else + f->scan[n].sample_loc = data[i+6] + (data[i+7] << 8) + (data[i+ 8]<<16) + (data[i+ 9]<<24); + f->scan[n].bytes_done = i+j; + if (f->page_crc_tests == STB_VORBIS_PUSHDATA_CRC_COUNT) + break; + // keep going if we still have room for more + } + } + } + } + + for (i=0; i < f->page_crc_tests;) { + uint32 crc; + int j; + int n = f->scan[i].bytes_done; + int m = f->scan[i].bytes_left; + if (m > data_len - n) m = data_len - n; + // m is the bytes to scan in the current chunk + crc = f->scan[i].crc_so_far; + for (j=0; j < m; ++j) + crc = crc32_update(crc, data[n+j]); + f->scan[i].bytes_left -= m; + f->scan[i].crc_so_far = crc; + if (f->scan[i].bytes_left == 0) { + // does it match? + if (f->scan[i].crc_so_far == f->scan[i].goal_crc) { + // Houston, we have page + data_len = n+m; // consumption amount is wherever that scan ended + f->page_crc_tests = -1; // drop out of page scan mode + f->previous_length = 0; // decode-but-don't-output one frame + f->next_seg = -1; // start a new page + f->current_loc = f->scan[i].sample_loc; // set the current sample location + // to the amount we'd have decoded had we decoded this page + f->current_loc_valid = f->current_loc != ~0U; + return data_len; + } + // delete entry + f->scan[i] = f->scan[--f->page_crc_tests]; + } else { + ++i; + } + } + + return data_len; +} + +// return value: number of bytes we used +int stb_vorbis_decode_frame_pushdata( + stb_vorbis *f, // the file we're decoding + const uint8 *data, int data_len, // the memory available for decoding + int *channels, // place to write number of float * buffers + float ***output, // place to write float ** array of float * buffers + int *samples // place to write number of output samples + ) +{ + int i; + int len,right,left; + + if (!IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + + if (f->page_crc_tests >= 0) { + *samples = 0; + return vorbis_search_for_page_pushdata(f, (uint8 *) data, data_len); + } + + f->stream = (uint8 *) data; + f->stream_end = (uint8 *) data + data_len; + f->error = VORBIS__no_error; + + // check that we have the entire packet in memory + if (!is_whole_packet_present(f, FALSE)) { + *samples = 0; + return 0; + } + + if (!vorbis_decode_packet(f, &len, &left, &right)) { + // save the actual error we encountered + enum STBVorbisError error = f->error; + if (error == VORBIS_bad_packet_type) { + // flush and resynch + f->error = VORBIS__no_error; + while (get8_packet(f) != EOP) + if (f->eof) break; + *samples = 0; + return (int) (f->stream - data); + } + if (error == VORBIS_continued_packet_flag_invalid) { + if (f->previous_length == 0) { + // we may be resynching, in which case it's ok to hit one + // of these; just discard the packet + f->error = VORBIS__no_error; + while (get8_packet(f) != EOP) + if (f->eof) break; + *samples = 0; + return (int) (f->stream - data); + } + } + // if we get an error while parsing, what to do? + // well, it DEFINITELY won't work to continue from where we are! + stb_vorbis_flush_pushdata(f); + // restore the error that actually made us bail + f->error = error; + *samples = 0; + return 1; + } + + // success! + len = vorbis_finish_frame(f, len, left, right); + for (i=0; i < f->channels; ++i) + f->outputs[i] = f->channel_buffers[i] + left; + + if (channels) *channels = f->channels; + *samples = len; + *output = f->outputs; + return (int) (f->stream - data); +} + +stb_vorbis *stb_vorbis_open_pushdata( + const unsigned char *data, int data_len, // the memory available for decoding + int *data_used, // only defined if result is not NULL + int *error, const stb_vorbis_alloc *alloc) +{ + stb_vorbis *f, p; + vorbis_init(&p, alloc); + p.stream = (uint8 *) data; + p.stream_end = (uint8 *) data + data_len; + p.push_mode = TRUE; + if (!start_decoder(&p)) { + if (p.eof) + *error = VORBIS_need_more_data; + else + *error = p.error; + return NULL; + } + f = vorbis_alloc(&p); + if (f) { + *f = p; + *data_used = (int) (f->stream - data); + *error = 0; + return f; + } else { + vorbis_deinit(&p); + return NULL; + } +} +#endif // STB_VORBIS_NO_PUSHDATA_API + +unsigned int stb_vorbis_get_file_offset(stb_vorbis *f) +{ + #ifndef STB_VORBIS_NO_PUSHDATA_API + if (f->push_mode) return 0; + #endif + if (USE_MEMORY(f)) return (unsigned int) (f->stream - f->stream_start); + #ifndef STB_VORBIS_NO_STDIO + return (unsigned int) (ftell(f->f) - f->f_start); + #endif +} + +#ifndef STB_VORBIS_NO_PULLDATA_API +// +// DATA-PULLING API +// + +static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last) +{ + for(;;) { + int n; + if (f->eof) return 0; + n = get8(f); + if (n == 0x4f) { // page header candidate + unsigned int retry_loc = stb_vorbis_get_file_offset(f); + int i; + // check if we're off the end of a file_section stream + if (retry_loc - 25 > f->stream_len) + return 0; + // check the rest of the header + for (i=1; i < 4; ++i) + if (get8(f) != ogg_page_header[i]) + break; + if (f->eof) return 0; + if (i == 4) { + uint8 header[27]; + uint32 i, crc, goal, len; + for (i=0; i < 4; ++i) + header[i] = ogg_page_header[i]; + for (; i < 27; ++i) + header[i] = get8(f); + if (f->eof) return 0; + if (header[4] != 0) goto invalid; + goal = header[22] + (header[23] << 8) + (header[24]<<16) + (header[25]<<24); + for (i=22; i < 26; ++i) + header[i] = 0; + crc = 0; + for (i=0; i < 27; ++i) + crc = crc32_update(crc, header[i]); + len = 0; + for (i=0; i < header[26]; ++i) { + int s = get8(f); + crc = crc32_update(crc, s); + len += s; + } + if (len && f->eof) return 0; + for (i=0; i < len; ++i) + crc = crc32_update(crc, get8(f)); + // finished parsing probable page + if (crc == goal) { + // we could now check that it's either got the last + // page flag set, OR it's followed by the capture + // pattern, but I guess TECHNICALLY you could have + // a file with garbage between each ogg page and recover + // from it automatically? So even though that paranoia + // might decrease the chance of an invalid decode by + // another 2^32, not worth it since it would hose those + // invalid-but-useful files? + if (end) + *end = stb_vorbis_get_file_offset(f); + if (last) { + if (header[5] & 0x04) + *last = 1; + else + *last = 0; + } + set_file_offset(f, retry_loc-1); + return 1; + } + } + invalid: + // not a valid page, so rewind and look for next one + set_file_offset(f, retry_loc); + } + } +} + + +#define SAMPLE_unknown 0xffffffff + +// seeking is implemented with a binary search, which narrows down the range to +// 64K, before using a linear search (because finding the synchronization +// pattern can be expensive, and the chance we'd find the end page again is +// relatively high for small ranges) +// +// two initial interpolation-style probes are used at the start of the search +// to try to bound either side of the binary search sensibly, while still +// working in O(log n) time if they fail. + +static int get_seek_page_info(stb_vorbis *f, ProbedPage *z) +{ + uint8 header[27], lacing[255]; + int i,len; + + // record where the page starts + z->page_start = stb_vorbis_get_file_offset(f); + + // parse the header + getn(f, header, 27); + if (header[0] != 'O' || header[1] != 'g' || header[2] != 'g' || header[3] != 'S') + return 0; + getn(f, lacing, header[26]); + + // determine the length of the payload + len = 0; + for (i=0; i < header[26]; ++i) + len += lacing[i]; + + // this implies where the page ends + z->page_end = z->page_start + 27 + header[26] + len; + + // read the last-decoded sample out of the data + z->last_decoded_sample = header[6] + (header[7] << 8) + (header[8] << 16) + (header[9] << 24); + + // restore file state to where we were + set_file_offset(f, z->page_start); + return 1; +} + +// rarely used function to seek back to the preceeding page while finding the +// start of a packet +static int go_to_page_before(stb_vorbis *f, unsigned int limit_offset) +{ + unsigned int previous_safe, end; + + // now we want to seek back 64K from the limit + if (limit_offset >= 65536 && limit_offset-65536 >= f->first_audio_page_offset) + previous_safe = limit_offset - 65536; + else + previous_safe = f->first_audio_page_offset; + + set_file_offset(f, previous_safe); + + while (vorbis_find_page(f, &end, NULL)) { + if (end >= limit_offset && stb_vorbis_get_file_offset(f) < limit_offset) + return 1; + set_file_offset(f, end); + } + + return 0; +} + +// implements the search logic for finding a page and starting decoding. if +// the function succeeds, current_loc_valid will be true and current_loc will +// be less than or equal to the provided sample number (the closer the +// better). +static int seek_to_sample_coarse(stb_vorbis *f, uint32 sample_number) +{ + ProbedPage left, right, mid; + int i, start_seg_with_known_loc, end_pos, page_start; + uint32 delta, stream_length, padding; + double offset, bytes_per_sample; + int probe = 0; + + // find the last page and validate the target sample + stream_length = stb_vorbis_stream_length_in_samples(f); + if (stream_length == 0) return error(f, VORBIS_seek_without_length); + if (sample_number > stream_length) return error(f, VORBIS_seek_invalid); + + // this is the maximum difference between the window-center (which is the + // actual granule position value), and the right-start (which the spec + // indicates should be the granule position (give or take one)). + padding = ((f->blocksize_1 - f->blocksize_0) >> 2); + if (sample_number < padding) + sample_number = 0; + else + sample_number -= padding; + + left = f->p_first; + while (left.last_decoded_sample == ~0U) { + // (untested) the first page does not have a 'last_decoded_sample' + set_file_offset(f, left.page_end); + if (!get_seek_page_info(f, &left)) goto error; + } + + right = f->p_last; + assert(right.last_decoded_sample != ~0U); + + // starting from the start is handled differently + if (sample_number <= left.last_decoded_sample) { + stb_vorbis_seek_start(f); + return 1; + } + + while (left.page_end != right.page_start) { + assert(left.page_end < right.page_start); + // search range in bytes + delta = right.page_start - left.page_end; + if (delta <= 65536) { + // there's only 64K left to search - handle it linearly + set_file_offset(f, left.page_end); + } else { + if (probe < 2) { + if (probe == 0) { + // first probe (interpolate) + double data_bytes = right.page_end - left.page_start; + bytes_per_sample = data_bytes / right.last_decoded_sample; + offset = left.page_start + bytes_per_sample * (sample_number - left.last_decoded_sample); + } else { + // second probe (try to bound the other side) + double error = ((double) sample_number - mid.last_decoded_sample) * bytes_per_sample; + if (error >= 0 && error < 8000) error = 8000; + if (error < 0 && error > -8000) error = -8000; + offset += error * 2; + } + + // ensure the offset is valid + if (offset < left.page_end) + offset = left.page_end; + if (offset > right.page_start - 65536) + offset = right.page_start - 65536; + + set_file_offset(f, (unsigned int) offset); + } else { + // binary search for large ranges (offset by 32K to ensure + // we don't hit the right page) + set_file_offset(f, left.page_end + (delta / 2) - 32768); + } + + if (!vorbis_find_page(f, NULL, NULL)) goto error; + } + + for (;;) { + if (!get_seek_page_info(f, &mid)) goto error; + if (mid.last_decoded_sample != ~0U) break; + // (untested) no frames end on this page + set_file_offset(f, mid.page_end); + assert(mid.page_start < right.page_start); + } + + // if we've just found the last page again then we're in a tricky file, + // and we're close enough. + if (mid.page_start == right.page_start) + break; + + if (sample_number < mid.last_decoded_sample) + right = mid; + else + left = mid; + + ++probe; + } + + // seek back to start of the last packet + page_start = left.page_start; + set_file_offset(f, page_start); + if (!start_page(f)) return error(f, VORBIS_seek_failed); + end_pos = f->end_seg_with_known_loc; + assert(end_pos >= 0); + + for (;;) { + for (i = end_pos; i > 0; --i) + if (f->segments[i-1] != 255) + break; + + start_seg_with_known_loc = i; + + if (start_seg_with_known_loc > 0 || !(f->page_flag & PAGEFLAG_continued_packet)) + break; + + // (untested) the final packet begins on an earlier page + if (!go_to_page_before(f, page_start)) + goto error; + + page_start = stb_vorbis_get_file_offset(f); + if (!start_page(f)) goto error; + end_pos = f->segment_count - 1; + } + + // prepare to start decoding + f->current_loc_valid = FALSE; + f->last_seg = FALSE; + f->valid_bits = 0; + f->packet_bytes = 0; + f->bytes_in_seg = 0; + f->previous_length = 0; + f->next_seg = start_seg_with_known_loc; + + for (i = 0; i < start_seg_with_known_loc; i++) + skip(f, f->segments[i]); + + // start decoding (optimizable - this frame is generally discarded) + vorbis_pump_first_frame(f); + return 1; + +error: + // try to restore the file to a valid state + stb_vorbis_seek_start(f); + return error(f, VORBIS_seek_failed); +} + +// the same as vorbis_decode_initial, but without advancing +static int peek_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode) +{ + int bits_read, bytes_read; + + if (!vorbis_decode_initial(f, p_left_start, p_left_end, p_right_start, p_right_end, mode)) + return 0; + + // either 1 or 2 bytes were read, figure out which so we can rewind + bits_read = 1 + ilog(f->mode_count-1); + if (f->mode_config[*mode].blockflag) + bits_read += 2; + bytes_read = (bits_read + 7) / 8; + + f->bytes_in_seg += bytes_read; + f->packet_bytes -= bytes_read; + skip(f, -bytes_read); + if (f->next_seg == -1) + f->next_seg = f->segment_count - 1; + else + f->next_seg--; + f->valid_bits = 0; + + return 1; +} + +int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number) +{ + uint32 max_frame_samples; + + if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + + // fast page-level search + if (!seek_to_sample_coarse(f, sample_number)) + return 0; + + assert(f->current_loc_valid); + assert(f->current_loc <= sample_number); + + // linear search for the relevant packet + max_frame_samples = (f->blocksize_1*3 - f->blocksize_0) >> 2; + while (f->current_loc < sample_number) { + int left_start, left_end, right_start, right_end, mode, frame_samples; + if (!peek_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode)) + return error(f, VORBIS_seek_failed); + // calculate the number of samples returned by the next frame + frame_samples = right_start - left_start; + if (f->current_loc + frame_samples > sample_number) { + return 1; // the next frame will contain the sample + } else if (f->current_loc + frame_samples + max_frame_samples > sample_number) { + // there's a chance the frame after this could contain the sample + vorbis_pump_first_frame(f); + } else { + // this frame is too early to be relevant + f->current_loc += frame_samples; + f->previous_length = 0; + maybe_start_packet(f); + flush_packet(f); + } + } + // the next frame will start with the sample + assert(f->current_loc == sample_number); + return 1; +} + +int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number) +{ + if (!stb_vorbis_seek_frame(f, sample_number)) + return 0; + + if (sample_number != f->current_loc) { + int n; + uint32 frame_start = f->current_loc; + stb_vorbis_get_frame_float(f, &n, NULL); + assert(sample_number > frame_start); + assert(f->channel_buffer_start + (int) (sample_number-frame_start) <= f->channel_buffer_end); + f->channel_buffer_start += (sample_number - frame_start); + } + + return 1; +} + +void stb_vorbis_seek_start(stb_vorbis *f) +{ + if (IS_PUSH_MODE(f)) { error(f, VORBIS_invalid_api_mixing); return; } + set_file_offset(f, f->first_audio_page_offset); + f->previous_length = 0; + f->first_decode = TRUE; + f->next_seg = -1; + vorbis_pump_first_frame(f); +} + +unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f) +{ + unsigned int restore_offset, previous_safe; + unsigned int end, last_page_loc; + + if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + if (!f->total_samples) { + unsigned int last; + uint32 lo,hi; + char header[6]; + + // first, store the current decode position so we can restore it + restore_offset = stb_vorbis_get_file_offset(f); + + // now we want to seek back 64K from the end (the last page must + // be at most a little less than 64K, but let's allow a little slop) + if (f->stream_len >= 65536 && f->stream_len-65536 >= f->first_audio_page_offset) + previous_safe = f->stream_len - 65536; + else + previous_safe = f->first_audio_page_offset; + + set_file_offset(f, previous_safe); + // previous_safe is now our candidate 'earliest known place that seeking + // to will lead to the final page' + + if (!vorbis_find_page(f, &end, &last)) { + // if we can't find a page, we're hosed! + f->error = VORBIS_cant_find_last_page; + f->total_samples = 0xffffffff; + goto done; + } + + // check if there are more pages + last_page_loc = stb_vorbis_get_file_offset(f); + + // stop when the last_page flag is set, not when we reach eof; + // this allows us to stop short of a 'file_section' end without + // explicitly checking the length of the section + while (!last) { + set_file_offset(f, end); + if (!vorbis_find_page(f, &end, &last)) { + // the last page we found didn't have the 'last page' flag + // set. whoops! + break; + } + previous_safe = last_page_loc+1; + last_page_loc = stb_vorbis_get_file_offset(f); + } + + set_file_offset(f, last_page_loc); + + // parse the header + getn(f, (unsigned char *)header, 6); + // extract the absolute granule position + lo = get32(f); + hi = get32(f); + if (lo == 0xffffffff && hi == 0xffffffff) { + f->error = VORBIS_cant_find_last_page; + f->total_samples = SAMPLE_unknown; + goto done; + } + if (hi) + lo = 0xfffffffe; // saturate + f->total_samples = lo; + + f->p_last.page_start = last_page_loc; + f->p_last.page_end = end; + f->p_last.last_decoded_sample = lo; + + done: + set_file_offset(f, restore_offset); + } + return f->total_samples == SAMPLE_unknown ? 0 : f->total_samples; +} + +float stb_vorbis_stream_length_in_seconds(stb_vorbis *f) +{ + return stb_vorbis_stream_length_in_samples(f) / (float) f->sample_rate; +} + + + +int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output) +{ + int len, right,left,i; + if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + + if (!vorbis_decode_packet(f, &len, &left, &right)) { + f->channel_buffer_start = f->channel_buffer_end = 0; + return 0; + } + + len = vorbis_finish_frame(f, len, left, right); + for (i=0; i < f->channels; ++i) + f->outputs[i] = f->channel_buffers[i] + left; + + f->channel_buffer_start = left; + f->channel_buffer_end = left+len; + + if (channels) *channels = f->channels; + if (output) *output = f->outputs; + return len; +} + +#ifndef STB_VORBIS_NO_STDIO + +stb_vorbis * stb_vorbis_open_file_section(FILE *file, int close_on_free, int *error, const stb_vorbis_alloc *alloc, unsigned int length) +{ + stb_vorbis *f, p; + vorbis_init(&p, alloc); + p.f = file; + p.f_start = (uint32) ftell(file); + p.stream_len = length; + p.close_on_free = close_on_free; + if (start_decoder(&p)) { + f = vorbis_alloc(&p); + if (f) { + *f = p; + vorbis_pump_first_frame(f); + return f; + } + } + if (error) *error = p.error; + vorbis_deinit(&p); + return NULL; +} + +stb_vorbis * stb_vorbis_open_file(FILE *file, int close_on_free, int *error, const stb_vorbis_alloc *alloc) +{ + unsigned int len, start; + start = (unsigned int) ftell(file); + fseek(file, 0, SEEK_END); + len = (unsigned int) (ftell(file) - start); + fseek(file, start, SEEK_SET); + return stb_vorbis_open_file_section(file, close_on_free, error, alloc, len); +} + +stb_vorbis * stb_vorbis_open_filename(const char *filename, int *error, const stb_vorbis_alloc *alloc) +{ + FILE *f = fopen(filename, "rb"); + if (f) + return stb_vorbis_open_file(f, TRUE, error, alloc); + if (error) *error = VORBIS_file_open_failure; + return NULL; +} +#endif // STB_VORBIS_NO_STDIO + +stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, int *error, const stb_vorbis_alloc *alloc) +{ + stb_vorbis *f, p; + if (data == NULL) return NULL; + vorbis_init(&p, alloc); + p.stream = (uint8 *) data; + p.stream_end = (uint8 *) data + len; + p.stream_start = (uint8 *) p.stream; + p.stream_len = len; + p.push_mode = FALSE; + if (start_decoder(&p)) { + f = vorbis_alloc(&p); + if (f) { + *f = p; + vorbis_pump_first_frame(f); + return f; + } + } + if (error) *error = p.error; + vorbis_deinit(&p); + return NULL; +} + +#ifndef STB_VORBIS_NO_INTEGER_CONVERSION +#define PLAYBACK_MONO 1 +#define PLAYBACK_LEFT 2 +#define PLAYBACK_RIGHT 4 + +#define L (PLAYBACK_LEFT | PLAYBACK_MONO) +#define C (PLAYBACK_LEFT | PLAYBACK_RIGHT | PLAYBACK_MONO) +#define R (PLAYBACK_RIGHT | PLAYBACK_MONO) + +static int8 channel_position[7][6] = +{ + { 0 }, + { C }, + { L, R }, + { L, C, R }, + { L, R, L, R }, + { L, C, R, L, R }, + { L, C, R, L, R, C }, +}; + + +#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT + typedef union { + float f; + int i; + } float_conv; + typedef char stb_vorbis_float_size_test[sizeof(float)==4 && sizeof(int) == 4]; + #define FASTDEF(x) float_conv x + // add (1<<23) to convert to int, then divide by 2^SHIFT, then add 0.5/2^SHIFT to round + #define MAGIC(SHIFT) (1.5f * (1 << (23-SHIFT)) + 0.5f/(1 << SHIFT)) + #define ADDEND(SHIFT) (((150-SHIFT) << 23) + (1 << 22)) + #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) (temp.f = (x) + MAGIC(s), temp.i - ADDEND(s)) + #define check_endianness() +#else + #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) ((int) ((x) * (1 << (s)))) + #define check_endianness() + #define FASTDEF(x) +#endif + +static void copy_samples(short *dest, float *src, int len) +{ + int i; + check_endianness(); + for (i=0; i < len; ++i) { + FASTDEF(temp); + int v = FAST_SCALED_FLOAT_TO_INT(temp, src[i],15); + if ((unsigned int) (v + 32768) > 65535) + v = v < 0 ? -32768 : 32767; + dest[i] = v; + } +} + +static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len) +{ + #define BUFFER_SIZE 32 + float buffer[BUFFER_SIZE]; + int i,j,o,n = BUFFER_SIZE; + check_endianness(); + for (o = 0; o < len; o += BUFFER_SIZE) { + memset(buffer, 0, sizeof(buffer)); + if (o + n > len) n = len - o; + for (j=0; j < num_c; ++j) { + if (channel_position[num_c][j] & mask) { + for (i=0; i < n; ++i) + buffer[i] += data[j][d_offset+o+i]; + } + } + for (i=0; i < n; ++i) { + FASTDEF(temp); + int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15); + if ((unsigned int) (v + 32768) > 65535) + v = v < 0 ? -32768 : 32767; + output[o+i] = v; + } + } +} + +static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len) +{ + #define BUFFER_SIZE 32 + float buffer[BUFFER_SIZE]; + int i,j,o,n = BUFFER_SIZE >> 1; + // o is the offset in the source data + check_endianness(); + for (o = 0; o < len; o += BUFFER_SIZE >> 1) { + // o2 is the offset in the output data + int o2 = o << 1; + memset(buffer, 0, sizeof(buffer)); + if (o + n > len) n = len - o; + for (j=0; j < num_c; ++j) { + int m = channel_position[num_c][j] & (PLAYBACK_LEFT | PLAYBACK_RIGHT); + if (m == (PLAYBACK_LEFT | PLAYBACK_RIGHT)) { + for (i=0; i < n; ++i) { + buffer[i*2+0] += data[j][d_offset+o+i]; + buffer[i*2+1] += data[j][d_offset+o+i]; + } + } else if (m == PLAYBACK_LEFT) { + for (i=0; i < n; ++i) { + buffer[i*2+0] += data[j][d_offset+o+i]; + } + } else if (m == PLAYBACK_RIGHT) { + for (i=0; i < n; ++i) { + buffer[i*2+1] += data[j][d_offset+o+i]; + } + } + } + for (i=0; i < (n<<1); ++i) { + FASTDEF(temp); + int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15); + if ((unsigned int) (v + 32768) > 65535) + v = v < 0 ? -32768 : 32767; + output[o2+i] = v; + } + } +} + +static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples) +{ + int i; + if (buf_c != data_c && buf_c <= 2 && data_c <= 6) { + static int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} }; + for (i=0; i < buf_c; ++i) + compute_samples(channel_selector[buf_c][i], buffer[i]+b_offset, data_c, data, d_offset, samples); + } else { + int limit = buf_c < data_c ? buf_c : data_c; + for (i=0; i < limit; ++i) + copy_samples(buffer[i]+b_offset, data[i]+d_offset, samples); + for ( ; i < buf_c; ++i) + memset(buffer[i]+b_offset, 0, sizeof(short) * samples); + } +} + +int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples) +{ + float **output; + int len = stb_vorbis_get_frame_float(f, NULL, &output); + if (len > num_samples) len = num_samples; + if (len) + convert_samples_short(num_c, buffer, 0, f->channels, output, 0, len); + return len; +} + +static void convert_channels_short_interleaved(int buf_c, short *buffer, int data_c, float **data, int d_offset, int len) +{ + int i; + check_endianness(); + if (buf_c != data_c && buf_c <= 2 && data_c <= 6) { + assert(buf_c == 2); + for (i=0; i < buf_c; ++i) + compute_stereo_samples(buffer, data_c, data, d_offset, len); + } else { + int limit = buf_c < data_c ? buf_c : data_c; + int j; + for (j=0; j < len; ++j) { + for (i=0; i < limit; ++i) { + FASTDEF(temp); + float f = data[i][d_offset+j]; + int v = FAST_SCALED_FLOAT_TO_INT(temp, f,15);//data[i][d_offset+j],15); + if ((unsigned int) (v + 32768) > 65535) + v = v < 0 ? -32768 : 32767; + *buffer++ = v; + } + for ( ; i < buf_c; ++i) + *buffer++ = 0; + } + } +} + +int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts) +{ + float **output; + int len; + if (num_c == 1) return stb_vorbis_get_frame_short(f,num_c,&buffer, num_shorts); + len = stb_vorbis_get_frame_float(f, NULL, &output); + if (len) { + if (len*num_c > num_shorts) len = num_shorts / num_c; + convert_channels_short_interleaved(num_c, buffer, f->channels, output, 0, len); + } + return len; +} + +int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts) +{ + float **outputs; + int len = num_shorts / channels; + int n=0; + int z = f->channels; + if (z > channels) z = channels; + while (n < len) { + int k = f->channel_buffer_end - f->channel_buffer_start; + if (n+k >= len) k = len - n; + if (k) + convert_channels_short_interleaved(channels, buffer, f->channels, f->channel_buffers, f->channel_buffer_start, k); + buffer += k*channels; + n += k; + f->channel_buffer_start += k; + if (n == len) break; + if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; + } + return n; +} + +int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int len) +{ + float **outputs; + int n=0; + int z = f->channels; + if (z > channels) z = channels; + while (n < len) { + int k = f->channel_buffer_end - f->channel_buffer_start; + if (n+k >= len) k = len - n; + if (k) + convert_samples_short(channels, buffer, n, f->channels, f->channel_buffers, f->channel_buffer_start, k); + n += k; + f->channel_buffer_start += k; + if (n == len) break; + if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; + } + return n; +} + +#ifndef STB_VORBIS_NO_STDIO +int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output) +{ + int data_len, offset, total, limit, error; + short *data; + stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL); + if (v == NULL) return -1; + limit = v->channels * 4096; + *channels = v->channels; + if (sample_rate) + *sample_rate = v->sample_rate; + offset = data_len = 0; + total = limit; + data = (short *) malloc(total * sizeof(*data)); + if (data == NULL) { + stb_vorbis_close(v); + return -2; + } + for (;;) { + int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset); + if (n == 0) break; + data_len += n; + offset += n * v->channels; + if (offset + limit > total) { + short *data2; + total *= 2; + data2 = (short *) realloc(data, total * sizeof(*data)); + if (data2 == NULL) { + free(data); + stb_vorbis_close(v); + return -2; + } + data = data2; + } + } + *output = data; + stb_vorbis_close(v); + return data_len; +} +#endif // NO_STDIO + +int stb_vorbis_decode_memory(const uint8 *mem, int len, int *channels, int *sample_rate, short **output) +{ + int data_len, offset, total, limit, error; + short *data; + stb_vorbis *v = stb_vorbis_open_memory(mem, len, &error, NULL); + if (v == NULL) return -1; + limit = v->channels * 4096; + *channels = v->channels; + if (sample_rate) + *sample_rate = v->sample_rate; + offset = data_len = 0; + total = limit; + data = (short *) malloc(total * sizeof(*data)); + if (data == NULL) { + stb_vorbis_close(v); + return -2; + } + for (;;) { + int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset); + if (n == 0) break; + data_len += n; + offset += n * v->channels; + if (offset + limit > total) { + short *data2; + total *= 2; + data2 = (short *) realloc(data, total * sizeof(*data)); + if (data2 == NULL) { + free(data); + stb_vorbis_close(v); + return -2; + } + data = data2; + } + } + *output = data; + stb_vorbis_close(v); + return data_len; +} +#endif // STB_VORBIS_NO_INTEGER_CONVERSION + +int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats) +{ + float **outputs; + int len = num_floats / channels; + int n=0; + int z = f->channels; + if (z > channels) z = channels; + while (n < len) { + int i,j; + int k = f->channel_buffer_end - f->channel_buffer_start; + if (n+k >= len) k = len - n; + for (j=0; j < k; ++j) { + for (i=0; i < z; ++i) + *buffer++ = f->channel_buffers[i][f->channel_buffer_start+j]; + for ( ; i < channels; ++i) + *buffer++ = 0; + } + n += k; + f->channel_buffer_start += k; + if (n == len) + break; + if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) + break; + } + return n; +} + +int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples) +{ + float **outputs; + int n=0; + int z = f->channels; + if (z > channels) z = channels; + while (n < num_samples) { + int i; + int k = f->channel_buffer_end - f->channel_buffer_start; + if (n+k >= num_samples) k = num_samples - n; + if (k) { + for (i=0; i < z; ++i) + memcpy(buffer[i]+n, f->channel_buffers[i]+f->channel_buffer_start, sizeof(float)*k); + for ( ; i < channels; ++i) + memset(buffer[i]+n, 0, sizeof(float) * k); + } + n += k; + f->channel_buffer_start += k; + if (n == num_samples) + break; + if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) + break; + } + return n; +} +#endif // STB_VORBIS_NO_PULLDATA_API + +/* Version history + 1.09 - 2016/04/04 - back out 'avoid discarding last frame' fix from previous version + 1.08 - 2016/04/02 - fixed multiple warnings; fix setup memory leaks; + avoid discarding last frame of audio data + 1.07 - 2015/01/16 - fixed some warnings, fix mingw, const-correct API + some more crash fixes when out of memory or with corrupt files + 1.06 - 2015/08/31 - full, correct support for seeking API (Dougall Johnson) + some crash fixes when out of memory or with corrupt files + 1.05 - 2015/04/19 - don't define __forceinline if it's redundant + 1.04 - 2014/08/27 - fix missing const-correct case in API + 1.03 - 2014/08/07 - Warning fixes + 1.02 - 2014/07/09 - Declare qsort compare function _cdecl on windows + 1.01 - 2014/06/18 - fix stb_vorbis_get_samples_float + 1.0 - 2014/05/26 - fix memory leaks; fix warnings; fix bugs in multichannel + (API change) report sample rate for decode-full-file funcs + 0.99996 - bracket #include 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/src/sound.h b/src/sound.h new file mode 100644 index 0000000..8bdbbcf --- /dev/null +++ b/src/sound.h @@ -0,0 +1,356 @@ +#ifndef H_SOUND +#define H_SOUND + +#include "utils.h" +#include "libs/minimp3/minimp3.h" +#define STB_VORBIS_HEADER_ONLY +#include "libs/stb_vorbis/stb_vorbis.c" + +#define SND_CHANNELS_MAX 32 +#define BUFFER_SIZE_MP3 8192 + +namespace Sound { + + struct Frame { + short L, R; + }; + + struct Decoder { + Stream *stream; + int channels, offset; + + Decoder(Stream *stream, int channels) : stream(stream), channels(channels), offset(stream->pos) {} + virtual ~Decoder() { delete stream; } + virtual int decode(Frame *frames, int count) { return 0; } + }; + + struct PCM : Decoder { + int freq, size, bits; + + PCM(Stream *stream, int channels, int freq, int size, int bits) : Decoder(stream, channels), freq(freq), size(size), bits(bits) {} + + virtual int decode(Frame *frames, int count) { + if (stream->pos - offset >= size) return 0; + if (bits == 16) { + int16 value; + if (channels == 2) { + frames[0].L = stream->read(value); + frames[0].R = stream->read(value); + } else + frames[0].L = frames[0].R = stream->read(value); + } else if (bits == 8) { + uint8 value; + if (channels == 2) { + frames[0].L = stream->read(value) * 257 - 32768; + frames[0].R = stream->read(value) * 257 - 32768; + } else + frames[0].L = frames[0].R = stream->read(value) * 257 - 32768; + } else { + ASSERT(false); + return 0; + } + + int k = 44100 / freq; + for (int i = 1; i < k; i++) frames[i] = frames[0]; + return k; + } + }; + + struct ADPCM : Decoder { // https://wiki.multimedia.cx/?title=Microsoft_ADPCM + int size, block; + + ADPCM(Stream *stream, int channels, int size, int block) : Decoder(stream, channels), size(size), block(block) {} + + struct Channel { + int16 c1, c2; + int16 delta; + int16 sample1; + int16 sample2; + + int predicate(uint8 nibble) { + static const int table[] = { 230, 230, 230, 230, 307, 409, 512, 614, 768, 614, 512, 409, 307, 230, 230, 230 }; + + int8 ns = nibble; + if (ns & 8) ns -= 16; + + int sample = (sample1 * c1 + sample2 * c2) / 256 + ns * delta; + sample = clamp(sample, -32768, 32767); + sample2 = sample1; + sample1 = sample; + delta = max(table[nibble] * delta / 256, 16); + return sample; + } + } channel[2]; + + virtual int decode(Frame *frames, int count) { + static const int coeff1[] = { 256, 512, 0, 192, 240, 460, 392 }; + static const int coeff2[] = { 0, -256, 0, 64, 0, -208, -232 }; + + int seek = stream->pos - offset; + if (seek >= size) return 0; + + if (seek % block == 0) { + for (int i = 0; i < channels; i++) { + char index; + stream->read(index); + channel[i].c1 = coeff1[index]; + channel[i].c2 = coeff2[index]; + } + for (int i = 0; i < channels; i++) stream->read(channel[i].delta); + for (int i = 0; i < channels; i++) stream->read(channel[i].sample1); + for (int i = 0; i < channels; i++) stream->read(channel[i].sample2); + + if (channels == 1) { + frames[0].L = frames[0].R = channel[0].sample2; + frames[1].L = frames[1].R = channel[0].sample1; + } else { + frames[0].L = channel[0].sample2; + frames[0].R = channel[1].sample2; + frames[1].L = channel[0].sample1; + frames[1].R = channel[1].sample1; + } + return 2; + } else { + uint8 value; + stream->read(value); + uint8 n1 = value >> 4, n2 = value & 0xF; + + if (channels == 1) { + frames[0].L = frames[0].R = channel[0].predicate(n1); + frames[1].L = frames[1].R = channel[0].predicate(n2); + return 2; + } else { + frames[0].L = channel[0].predicate(n1); + frames[0].R = channel[1].predicate(n2); + return 1; + } + } + } + }; + + struct MP3 : Decoder { + mp3_decoder_t mp3; + char *buffer; + int size, pos; + + MP3(Stream *stream, int channels) : Decoder(stream, channels), size(stream->size), pos(0) { + mp3 = mp3_create(); + buffer = new char[size]; // TODO: file streaming + stream->raw(buffer, size); + } + + virtual ~MP3() { + delete[] buffer; + mp3_done(mp3); + } + + virtual int decode(Frame *frames, int count) { + mp3_info_t info; + int i = 0; + char *ptr = (char*)frames; + while (ptr < (char*)&frames[count]) { + int res = mp3_decode(mp3, buffer + pos, size - pos, (short*)ptr, &info); + if (res) { + pos += res; + ptr += info.audio_bytes; + i += info.audio_bytes; + } else + break; + } + return i; + } + }; + + struct OGG : Decoder { + stb_vorbis *ogg; + stb_vorbis_alloc alloc; + + char *buffer; + int size, pos; + + OGG(Stream *stream, int channels) : Decoder(stream, channels), size(stream->size), pos(0) { + buffer = new char[size]; // TODO: file streaming + stream->raw(buffer, size); + int error; + alloc.alloc_buffer_length_in_bytes = 256 * 1024; + alloc.alloc_buffer = new char[alloc.alloc_buffer_length_in_bytes]; + ogg = stb_vorbis_open_memory((unsigned char*)buffer, size, &error, &alloc); + stb_vorbis_info info = stb_vorbis_get_info(ogg); + channels = info.channels; + } + + virtual ~OGG() { + stb_vorbis_close(ogg); + delete[] alloc.alloc_buffer; + delete[] buffer; + } + + virtual int decode(Frame *frames, int count) { + int i = 0; + while (i < count) { + int res = stb_vorbis_get_samples_short_interleaved(ogg, channels, (short*)frames + i, (count - i) * 2); + if (!res) break; + i += res; + } + return i; + } + }; + + struct Listener { + mat4 matrix; + vec3 velocity; + } listener; + + enum Flags { + LOOP = 1, + PAN = 2, + REVERB_NEAR = 4, + REVERB_MIDDLE = 8, + REVERB_FAR = 16, + }; + + struct Sample { + Decoder *decoder; + float volume; + float pitch; + int flags; + bool isPlaying; + + Sample(Stream *stream, float volume, float pitch, int flags) : decoder(NULL), volume(volume), pitch(pitch), flags(flags), isPlaying(true) { + uint32 fourcc; + stream->read(fourcc); + if (fourcc == FOURCC("RIFF")) { // wav + + struct { + uint16 format; + uint16 channels; + uint32 samplesPerSec; + uint32 bytesPerSec; + uint16 block; + uint16 sampleBits; + } waveFmt; + + stream->seek(8); + while (stream->pos < stream->size) { + uint32 type, size; + stream->read(type); + stream->read(size); + if (type == FOURCC("fmt ")) { + stream->read(waveFmt); + stream->seek(size - sizeof(waveFmt)); + } else if (type == FOURCC("data")) { + if (waveFmt.format == 1) decoder = new PCM(stream, waveFmt.channels, waveFmt.samplesPerSec, size, waveFmt.sampleBits); + if (waveFmt.format == 2) decoder = new ADPCM(stream, waveFmt.channels, size, waveFmt.block); + break; + } else + stream->seek(size); + } + } else if (fourcc == FOURCC("OggS")) { // ogg + stream->seek(-4); + decoder = new OGG(stream, 2); + } else if (fourcc == FOURCC("ID3\3")) { // mp3 + decoder = new MP3(stream, 2); + } + + ASSERT(decoder != NULL); + } + + ~Sample() { + delete decoder; + } + + bool render(Frame *frames, int count) { + int i = 0; + while (i < count) { + int res = decoder->decode(&frames[i], count - i); + if (res == 0) { + if (i == 0) isPlaying = false; + break; + } + i += res; + } + return true; + } + + } *channels[SND_CHANNELS_MAX]; + int channelsCount; + + void init() { + channelsCount = 0; + mp3_decode_init(); + } + + void free() { + for (int i = 0; i < channelsCount; i++) + delete channels[i]; + mp3_decode_free(); + } + + void fill(Frame *frames, int count) { + struct FrameHI { + int L, R; + }; + + FrameHI *result = new FrameHI[count]; + memset(result, 0, sizeof(FrameHI) * count); + + Frame *buffer = new Frame[count]; + + for (int i = 0; i < channelsCount; i++) { + + memset(buffer, 0, sizeof(Frame) * count); + channels[i]->render(buffer, count); + + for (int j = 0; j < count; j++) { + result[j].L += buffer[j].L; + result[j].R += buffer[j].R; + } + } + + for (int i = 0; i < count; i++) { + frames[i].L = clamp(result[i].L, -32768, 32767); + frames[i].R = clamp(result[i].R, -32768, 32767); + } + + delete[] buffer; + delete[] result; + + for (int i = 0; i < channelsCount; i++) + if (!channels[i]->isPlaying) { + delete channels[i]; + channels[i] = channels[--channelsCount]; + i--; + } + } + + Stream *openWAD(const char *name) { + Stream *stream = new Stream("cdaudio.wad"); + if (stream->size) { + struct Item { + char name[260]; + int size; + int offset; + } entity; + + for (int i = 0; i < 130; i++) { + stream->read(entity); + if (strcmp(name, entity.name) == 0) { + stream->setPos(entity.offset); + return stream; + } + } + } + delete stream; + return NULL; + } + + void play(Stream *stream, float volume, float pitch, int flags) { + if (!stream) return; + if (channelsCount < SND_CHANNELS_MAX) + channels[channelsCount++] = new Sample(stream, volume, pitch, flags); + else + LOG("! no free channels\n"); + } +} + +#endif \ No newline at end of file diff --git a/src/utils.h b/src/utils.h index 6e840a6..ac33f32 100644 --- a/src/utils.h +++ b/src/utils.h @@ -36,6 +36,8 @@ typedef unsigned char uint8; typedef unsigned short uint16; typedef unsigned int uint32; +#define FOURCC(str) (*((uint32*)str)) + struct ubyte4 { uint8 x, y, z, w; }; @@ -473,9 +475,12 @@ struct mat4 { struct Stream { FILE *f; - int size, pos; + const char *data; + int size, pos; - Stream(const char *name) : pos(0) { + Stream(const void *data, int size) : f(NULL), data((char*)data), size(size), pos(0) {} + + Stream(const char *name) : data(NULL), size(-1), pos(0) { f = fopen(name, "rb"); if (!f) LOG("error loading file\n"); fseek(f, 0, SEEK_END); @@ -484,32 +489,36 @@ struct Stream { } ~Stream() { - fclose(f); + if (f) fclose(f); } void setPos(int pos) { this->pos = pos; - fseek(f, pos, SEEK_SET); + if (f) fseek(f, pos, SEEK_SET); } void seek(int offset) { - fseek(f, offset, SEEK_CUR); + if (!offset) return; + if (f) fseek(f, offset, SEEK_CUR); pos += offset; } - int raw(void *data, int size) { - pos += size; - return fread(data, 1, size, f); + void raw(void *data, int count) { + if (f) + fread(data, 1, count, f); + else + memcpy(data, this->data + pos, count); + pos += count; } template - T& read(T &x) { + inline T& read(T &x) { raw(&x, sizeof(x)); return x; } template - T* read(T *&a, int count) { + inline T* read(T *&a, int count) { if (count) { a = new T[count]; raw(a, count * sizeof(T)); diff --git a/src/web/build.bat b/src/web/build.bat index a526f12..39b9900 100644 --- a/src/web/build.bat +++ b/src/web/build.bat @@ -1,3 +1,3 @@ -set SRC=main.cpp -call em++ %SRC% -O2 --llvm-opts 2 --closure 1 -std=c++11 -o OpenLara.js --preload-file ./LEVEL2_DEMO.PHD -I..\ +set SRC=main.cpp ../libs/minimp3/minimp3.cpp ../libs/stb_vorbis/stb_vorbis.c +call em++ %SRC% -O2 -s ASSERTIONS=1 -Wno-deprecated-register --llvm-opts 2 --closure 2 -std=c++11 -o OpenLara.js --preload-file ./LEVEL2_DEMO.PHD -I..\ gzip.exe -9 -f OpenLara.data OpenLara.js OpenLara.js.mem \ No newline at end of file diff --git a/src/win/OpenLara.vcxproj b/src/win/OpenLara.vcxproj index 22d45ab..6f9e84c 100644 --- a/src/win/OpenLara.vcxproj +++ b/src/win/OpenLara.vcxproj @@ -56,8 +56,10 @@ Level3 Disabled - NOMINMAX;_CRT_SECURE_NO_WARNINGS;WIN32;_DEBUG;_CONSOLE;_LIB;%(PreprocessorDefinitions) + STB_VORBIS_NO_STDIO;NOMINMAX;_CRT_SECURE_NO_WARNINGS;WIN32;_DEBUG;_CONSOLE;_LIB;%(PreprocessorDefinitions) Strict + + Console @@ -73,26 +75,30 @@ Full true true - NOMINMAX;_CRT_SECURE_NO_WARNINGS;WIN32;NDEBUG;_CONSOLE;_LIB;%(PreprocessorDefinitions) + STB_VORBIS_NO_STDIO;NOMINMAX;_CRT_SECURE_NO_WARNINGS;WIN32;NDEBUG;_CONSOLE;_LIB;%(PreprocessorDefinitions) false false MultiThreaded Strict false Size + /d2noftol3 %(AdditionalOptions) + true Console - false + true true true - opengl32.lib;winmm.lib;wcrt.lib;kernel32.lib;user32.lib;gdi32.lib;winspool.lib;comdlg32.lib;advapi32.lib;shell32.lib;ole32.lib;oleaut32.lib;uuid.lib;odbc32.lib;odbccp32.lib;%(AdditionalDependencies) + wcrt.lib;opengl32.lib;winmm.lib;kernel32.lib;user32.lib;gdi32.lib;winspool.lib;comdlg32.lib;advapi32.lib;shell32.lib;ole32.lib;oleaut32.lib;uuid.lib;odbc32.lib;odbccp32.lib;%(AdditionalDependencies) true false false + + @@ -105,8 +111,11 @@ + + + diff --git a/src/win/main.cpp b/src/win/main.cpp index 94a8214..75e54dc 100644 --- a/src/win/main.cpp +++ b/src/win/main.cpp @@ -1,7 +1,13 @@ #ifdef _DEBUG #include "crtdbg.h" #endif +/* // VS2015 ? +#include +void __cdecl operator delete(void *ptr, unsigned int size) { + // +} +*/ #include "game.h" DWORD getTime() { @@ -16,6 +22,7 @@ DWORD getTime() { #endif } +// common input functions InputKey keyToInputKey(int code) { int codes[] = { VK_LEFT, VK_RIGHT, VK_UP, VK_DOWN, VK_SPACE, VK_RETURN, VK_ESCAPE, VK_SHIFT, VK_CONTROL, VK_MENU, @@ -35,7 +42,8 @@ InputKey mouseToInputKey(int msg) { (msg >= WM_RBUTTONDOWN && msg <= WM_RBUTTONDBLCLK) ? ikMouseR : ikMouseM; } -#define JOY_DEAD_ZONE_STICK 0.3f +// joystick +#define JOY_DEAD_ZONE_STICK 0.3f #define JOY_DEAD_ZONE_TRIGGER 0.01f bool joyReady; @@ -99,11 +107,75 @@ void joyUpdate() { joyFree(); } +// sound +#define SND_SIZE 4608*2 + +bool sndReady; +char *sndData; +CRITICAL_SECTION sndCS; +HWAVEOUT waveOut; +WAVEFORMATEX waveFmt = { WAVE_FORMAT_PCM, 2, 44100, 44100 * 4, 4, 16, sizeof(waveFmt) }; +WAVEHDR waveBuf[2]; + +void soundFree() { + if (!sndReady) return; + sndReady = false; + EnterCriticalSection(&sndCS); + waveOutUnprepareHeader(waveOut, &waveBuf[0], sizeof(WAVEHDR)); + waveOutUnprepareHeader(waveOut, &waveBuf[1], sizeof(WAVEHDR)); + waveOutReset(waveOut); + waveOutClose(waveOut); + delete[] sndData; + LeaveCriticalSection(&sndCS); + DeleteCriticalSection(&sndCS); +} + +void CALLBACK sndFill(HWAVEOUT waveOut, UINT uMsg, DWORD_PTR dwInstance, LPWAVEHDR waveBuf, DWORD dwParam2) { + if (!sndReady) return; + if (uMsg == MM_WOM_CLOSE) { + soundFree(); + return; + } + + EnterCriticalSection(&sndCS); + waveOutUnprepareHeader(waveOut, waveBuf, sizeof(WAVEHDR)); + Sound::fill((Sound::Frame*)waveBuf->lpData, SND_SIZE / 4); + waveOutPrepareHeader(waveOut, waveBuf, sizeof(WAVEHDR)); + waveOutWrite(waveOut, waveBuf, sizeof(WAVEHDR)); + LeaveCriticalSection(&sndCS); +} + +void soundInit(HWND hwnd) { + InitializeCriticalSection(&sndCS); + if (waveOutOpen(&waveOut, WAVE_MAPPER, &waveFmt, (INT_PTR)sndFill, 0, CALLBACK_FUNCTION) == MMSYSERR_NOERROR) { + sndReady = true; + sndData = new char[SND_SIZE * 2]; + memset(&waveBuf, 0, sizeof(waveBuf)); + for (int i = 0; i < 2; i++) { + waveBuf[i].dwBufferLength = SND_SIZE; + waveBuf[i].lpData = sndData + SND_SIZE * i; + sndFill(waveOut, 0, 0, &waveBuf[i], 0); + } + } else { + sndReady = false; + sndData = NULL; + } +} + + static LRESULT CALLBACK WndProc(HWND hWnd, UINT msg, WPARAM wParam, LPARAM lParam) { switch (msg) { + // window case WM_ACTIVATE : Input::reset(); break; + case WM_SIZE: + Core::width = LOWORD(lParam); + Core::height = HIWORD(lParam); + break; + case WM_DESTROY: + PostQuitMessage(0); + break; // keyboard case WM_KEYDOWN : case WM_KEYUP : @@ -134,19 +206,13 @@ static LRESULT CALLBACK WndProc(HWND hWnd, UINT msg, WPARAM wParam, LPARAM lPara case WM_MOUSEMOVE : Input::setPos(ikMouseL, vec2((float)(short)LOWORD(lParam), (float)(short)HIWORD(lParam))); break; - // gamepad + // joystick case WM_DEVICECHANGE : joyInit(); return 1; // touch - // ... - case WM_SIZE : - Core::width = LOWORD(lParam); - Core::height = HIWORD(lParam); - break; - case WM_DESTROY : - PostQuitMessage(0); - break; + // TODO + // sound default : return DefWindowProc(hWnd, msg, wParam, lParam); } @@ -187,11 +253,12 @@ int main() { HWND hWnd = CreateWindow("static", "OpenLara", WS_OVERLAPPEDWINDOW, 0, 0, r.right - r.left, r.bottom - r.top, 0, 0, 0, 0); - joyInit(); - - HDC hDC = GetDC(hWnd); + HDC hDC = GetDC(hWnd); HGLRC hRC = initGL(hDC); - Game::init(); + + joyInit(); + soundInit(hWnd); + Game::init(); SetWindowLong(hWnd, GWL_WNDPROC, (LONG)&WndProc); ShowWindow(hWnd, SW_SHOWDEFAULT); @@ -211,11 +278,13 @@ int main() { continue; float delta = (time - lastTime) * 0.001f; + EnterCriticalSection(&sndCS); while (delta > EPS) { Core::deltaTime = min(delta, 1.0f / 30.0f); Game::update(); delta -= Core::deltaTime; } + LeaveCriticalSection(&sndCS); lastTime = time; Core::stats.dips = 0; @@ -232,7 +301,9 @@ int main() { } } while (msg.message != WM_QUIT); + soundFree(); Game::free(); + freeGL(hRC); ReleaseDC(hWnd, hDC);