X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fmpegaudiodec.c;h=ff1f1113e77c3c288ad83473800adb49c29cece8;hb=f29f3b5d9ff5cc3b70dac295c9589577ce1a41c7;hp=f3fa90af552202b61756a6cc0aee3446b63164f4;hpb=de6d9b6404bfd1c589799142da5a95428f146edd;p=ffmpeg diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c index f3fa90af552..ff1f1113e77 100644 --- a/libavcodec/mpegaudiodec.c +++ b/libavcodec/mpegaudiodec.c @@ -1,288 +1,2841 @@ /* * MPEG Audio decoder - * Copyright (c) 2001 Gerard Lantau. + * Copyright (c) 2001, 2002 Fabrice Bellard. * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. + * This library is 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 of the License, or (at your option) any later version. * - * This program is distributed in the hope that it will be useful, + * This library is 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 General Public License for more details. + * 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 General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#include -#include -#include + +/** + * @file mpegaudiodec.c + * MPEG Audio decoder. + */ + +//#define DEBUG #include "avcodec.h" -#include "mpglib/mpg123.h" +#include "bitstream.h" +#include "dsputil.h" /* - * TODO: - * - add free format - * - do not rely anymore on mpglib (first step: implement dct64 and decoding filter) + * TODO: + * - in low precision mode, use more 16 bit multiplies in synth filter + * - test lsf / mpeg25 extensively. */ +/* define USE_HIGHPRECISION to have a bit exact (but slower) mpeg + audio decoder */ +#ifdef CONFIG_MPEGAUDIO_HP +#define USE_HIGHPRECISION +#endif + +#include "mpegaudio.h" + +#define FRAC_ONE (1 << FRAC_BITS) + +#define MULL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS) +#define MUL64(a,b) ((int64_t)(a) * (int64_t)(b)) +#define FIX(a) ((int)((a) * FRAC_ONE)) +/* WARNING: only correct for posititive numbers */ +#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)) +//#define MULH(a,b) (((int64_t)(a) * (int64_t)(b))>>32) //gcc 3.4 creates an incredibly bloated mess out of this +static always_inline int MULH(int a, int b){ + return ((int64_t)(a) * (int64_t)(b))>>32; +} + +/****************/ + #define HEADER_SIZE 4 #define BACKSTEP_SIZE 512 +struct GranuleDef; + typedef struct MPADecodeContext { - struct mpstr mpstr; - UINT8 inbuf1[2][MAXFRAMESIZE + BACKSTEP_SIZE]; /* input buffer */ + uint8_t inbuf1[2][MPA_MAX_CODED_FRAME_SIZE + BACKSTEP_SIZE]; /* input buffer */ int inbuf_index; - UINT8 *inbuf_ptr, *inbuf; + uint8_t *inbuf_ptr, *inbuf; int frame_size; + int free_format_frame_size; /* frame size in case of free format + (zero if currently unknown) */ + /* next header (used in free format parsing) */ + uint32_t free_format_next_header; int error_protection; int layer; int sample_rate; + int sample_rate_index; /* between 0 and 8 */ int bit_rate; int old_frame_size; GetBitContext gb; + int nb_channels; + int mode; + int mode_ext; + int lsf; + MPA_INT synth_buf[MPA_MAX_CHANNELS][512 * 2] __attribute__((aligned(16))); + int synth_buf_offset[MPA_MAX_CHANNELS]; + int32_t sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT] __attribute__((aligned(16))); + int32_t mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */ +#ifdef DEBUG + int frame_count; +#endif + void (*compute_antialias)(struct MPADecodeContext *s, struct GranuleDef *g); + int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3 + unsigned int dither_state; } MPADecodeContext; -/* XXX: suppress that mess */ -struct mpstr *gmp; -GetBitContext *gmp_gb; -static MPADecodeContext *gmp_s; +/** + * Context for MP3On4 decoder + */ +typedef struct MP3On4DecodeContext { + int frames; ///< number of mp3 frames per block (number of mp3 decoder instances) + int chan_cfg; ///< channel config number + MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance +} MP3On4DecodeContext; + +/* layer 3 "granule" */ +typedef struct GranuleDef { + 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]; /* number of huffman codes in each region */ + int preflag; + int short_start, long_end; /* long/short band indexes */ + uint8_t scale_factors[40]; + int32_t sb_hybrid[SBLIMIT * 18]; /* 576 samples */ +} GranuleDef; -/* XXX: merge constants with encoder */ -static const unsigned short mp_bitrate_tab[2][3][15] = { - { {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 }, - {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384 }, - {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 } }, - { {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256}, - {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160}, - {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160} - } +#define MODE_EXT_MS_STEREO 2 +#define MODE_EXT_I_STEREO 1 + +/* layer 3 huffman tables */ +typedef struct HuffTable { + int xsize; + const uint8_t *bits; + const uint16_t *codes; +} HuffTable; + +#include "mpegaudiodectab.h" + +static void compute_antialias_integer(MPADecodeContext *s, GranuleDef *g); +static void compute_antialias_float(MPADecodeContext *s, GranuleDef *g); + +/* vlc structure for decoding layer 3 huffman tables */ +static VLC huff_vlc[16]; +static uint8_t *huff_code_table[16]; +static VLC huff_quad_vlc[2]; +/* computed from band_size_long */ +static uint16_t band_index_long[9][23]; +/* XXX: free when all decoders are closed */ +#define TABLE_4_3_SIZE (8191 + 16)*4 +static int8_t *table_4_3_exp; +static uint32_t *table_4_3_value; +/* intensity stereo coef table */ +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]; + +/* lower 2 bits: modulo 3, higher bits: shift */ +static uint16_t scale_factor_modshift[64]; +/* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */ +static int32_t scale_factor_mult[15][3]; +/* mult table for layer 2 group quantization */ + +#define SCALE_GEN(v) \ +{ FIXR(1.0 * (v)), FIXR(0.7937005259 * (v)), FIXR(0.6299605249 * (v)) } + +static const int32_t scale_factor_mult2[3][3] = { + SCALE_GEN(4.0 / 3.0), /* 3 steps */ + SCALE_GEN(4.0 / 5.0), /* 5 steps */ + SCALE_GEN(4.0 / 9.0), /* 9 steps */ }; -static unsigned short mp_freq_tab[3] = { 44100, 48000, 32000 }; +void ff_mpa_synth_init(MPA_INT *window); +static MPA_INT window[512] __attribute__((aligned(16))); + +/* layer 1 unscaling */ +/* n = number of bits of the mantissa minus 1 */ +static inline int l1_unscale(int n, int mant, int scale_factor) +{ + int shift, mod; + int64_t val; + + shift = scale_factor_modshift[scale_factor]; + mod = shift & 3; + shift >>= 2; + val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]); + shift += n; + /* NOTE: at this point, 1 <= shift >= 21 + 15 */ + return (int)((val + (1LL << (shift - 1))) >> shift); +} + +static inline int l2_unscale_group(int steps, int mant, int scale_factor) +{ + int shift, mod, val; + + shift = scale_factor_modshift[scale_factor]; + mod = shift & 3; + shift >>= 2; + + val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod]; + /* NOTE: at this point, 0 <= shift <= 21 */ + if (shift > 0) + val = (val + (1 << (shift - 1))) >> shift; + return val; +} + +/* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */ +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); + assert(e>=1); + if (e > 31) + return 0; + m = (m + (1 << (e-1))) >> e; + + return m; +} + +/* all integer n^(4/3) computation code */ +#define DEV_ORDER 13 + +#define POW_FRAC_BITS 24 +#define POW_FRAC_ONE (1 << POW_FRAC_BITS) +#define POW_FIX(a) ((int)((a) * POW_FRAC_ONE)) +#define POW_MULL(a,b) (((int64_t)(a) * (int64_t)(b)) >> POW_FRAC_BITS) + +static int dev_4_3_coefs[DEV_ORDER]; + +#if 0 /* unused */ +static int pow_mult3[3] = { + POW_FIX(1.0), + POW_FIX(1.25992104989487316476), + POW_FIX(1.58740105196819947474), +}; +#endif + +static void int_pow_init(void) +{ + int i, a; + + a = POW_FIX(1.0); + for(i=0;i= 0; j--) + a1 = POW_MULL(a, dev_4_3_coefs[j] + a1); + a = (1 << POW_FRAC_BITS) + a1; + /* exponent compute (exact) */ + e = e * 4; + er = e % 3; + eq = e / 3; + a = POW_MULL(a, pow_mult3[er]); + while (a >= 2 * POW_FRAC_ONE) { + a = a >> 1; + eq++; + } + /* convert to float */ + while (a < POW_FRAC_ONE) { + a = a << 1; + eq--; + } + /* now POW_FRAC_ONE <= a < 2 * POW_FRAC_ONE */ +#if POW_FRAC_BITS > FRAC_BITS + a = (a + (1 << (POW_FRAC_BITS - FRAC_BITS - 1))) >> (POW_FRAC_BITS - FRAC_BITS); + /* correct overflow */ + if (a >= 2 * (1 << FRAC_BITS)) { + a = a >> 1; + eq++; + } +#endif + *exp_ptr = eq; + return a; +} +#endif static int decode_init(AVCodecContext * avctx) { MPADecodeContext *s = avctx->priv_data; - struct mpstr *mp = &s->mpstr; - static int init; + static int init=0; + int i, j, k; + +#if defined(USE_HIGHPRECISION) && defined(CONFIG_AUDIO_NONSHORT) + avctx->sample_fmt= SAMPLE_FMT_S32; +#else + avctx->sample_fmt= SAMPLE_FMT_S16; +#endif + + if(avctx->antialias_algo != FF_AA_FLOAT) + s->compute_antialias= compute_antialias_integer; + else + s->compute_antialias= compute_antialias_float; + + if (!init && !avctx->parse_only) { + /* scale factors table for layer 1/2 */ + for(i=0;i<64;i++) { + int shift, mod; + /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */ + shift = (i / 3); + mod = i % 3; + scale_factor_modshift[i] = mod | (shift << 2); + } + + /* scale factor multiply for layer 1 */ + for(i=0;i<15;i++) { + int n, norm; + n = i + 2; + norm = ((int64_t_C(1) << n) * FRAC_ONE) / ((1 << n) - 1); + scale_factor_mult[i][0] = MULL(FIXR(1.0 * 2.0), norm); + scale_factor_mult[i][1] = MULL(FIXR(0.7937005259 * 2.0), norm); + scale_factor_mult[i][2] = MULL(FIXR(0.6299605249 * 2.0), norm); + dprintf("%d: norm=%x s=%x %x %x\n", + i, norm, + scale_factor_mult[i][0], + scale_factor_mult[i][1], + scale_factor_mult[i][2]); + } - mp->fr.single = -1; - mp->synth_bo = 1; + ff_mpa_synth_init(window); - if(!init) { + /* huffman decode tables */ + huff_code_table[0] = NULL; + for(i=1;i<16;i++) { + const HuffTable *h = &mpa_huff_tables[i]; + int xsize, x, y; + unsigned int n; + uint8_t *code_table; + + xsize = h->xsize; + n = xsize * xsize; + /* XXX: fail test */ + init_vlc(&huff_vlc[i], 8, n, + h->bits, 1, 1, h->codes, 2, 2, 1); + + code_table = av_mallocz(n); + j = 0; + for(x=0;x> 1); + f = 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); + dprintf("is_table_lsf %d %d: %x %x\n", + i, j, is_table_lsf[j][0][i], is_table_lsf[j][1][i]); + } + } + + 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; +// printf("%d %d %d %d\n", FIX(cs), FIX(cs-1), FIX(ca), FIX(cs)-FIX(ca)); +// av_log(NULL, AV_LOG_DEBUG,"%f %f %f %f\n", cs, ca, ca+cs, 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; + } + //merge last stage of imdct into the window coefficients + 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))); +// av_log(NULL, AV_LOG_DEBUG, "%2d %d %f\n", i,j,d / (1<<5)); + } + } + + /* NOTE: we do frequency inversion adter the MDCT by changing + the sign of the right window coefs */ + 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]; + } + } + +#if defined(DEBUG) + for(j=0;j<8;j++) { + printf("win%d=\n", j); + for(i=0;i<36;i++) + printf("%f, ", (double)mdct_win[j][i] / FRAC_ONE); + printf("\n"); + } +#endif init = 1; - make_decode_tables(32767); - init_layer2(); - init_layer3(SBLIMIT); } s->inbuf_index = 0; s->inbuf = &s->inbuf1[s->inbuf_index][BACKSTEP_SIZE]; s->inbuf_ptr = s->inbuf; - +#ifdef DEBUG + s->frame_count = 0; +#endif + if (avctx->codec_id == CODEC_ID_MP3ADU) + s->adu_mode = 1; return 0; } -/* fast header check for resync */ -static int check_header(UINT32 header) -{ - /* header */ - if ((header & 0xffe00000) != 0xffe00000) - return -1; - /* layer check */ - if (((header >> 17) & 3) == 0) - return -1; - /* bit rate : currently no free format supported */ - if (((header >> 12) & 0xf) == 0xf || - ((header >> 12) & 0xf) == 0x0) - return -1; - /* frequency */ - if (((header >> 10) & 3) == 3) - return -1; - return 0; +/* tab[i][j] = 1.0 / (2.0 * cos(pi*(2*k+1) / 2^(6 - j))) */ + +/* cos(i*pi/64) */ + +#define COS0_0 FIXR(0.50060299823519630134) +#define COS0_1 FIXR(0.50547095989754365998) +#define COS0_2 FIXR(0.51544730992262454697) +#define COS0_3 FIXR(0.53104259108978417447) +#define COS0_4 FIXR(0.55310389603444452782) +#define COS0_5 FIXR(0.58293496820613387367) +#define COS0_6 FIXR(0.62250412303566481615) +#define COS0_7 FIXR(0.67480834145500574602) +#define COS0_8 FIXR(0.74453627100229844977) +#define COS0_9 FIXR(0.83934964541552703873) +#define COS0_10 FIXR(0.97256823786196069369) +#define COS0_11 FIXR(1.16943993343288495515) +#define COS0_12 FIXR(1.48416461631416627724) +#define COS0_13 FIXR(2.05778100995341155085) +#define COS0_14 FIXR(3.40760841846871878570) +#define COS0_15 FIXR(10.19000812354805681150) + +#define COS1_0 FIXR(0.50241928618815570551) +#define COS1_1 FIXR(0.52249861493968888062) +#define COS1_2 FIXR(0.56694403481635770368) +#define COS1_3 FIXR(0.64682178335999012954) +#define COS1_4 FIXR(0.78815462345125022473) +#define COS1_5 FIXR(1.06067768599034747134) +#define COS1_6 FIXR(1.72244709823833392782) +#define COS1_7 FIXR(5.10114861868916385802) + +#define COS2_0 FIXR(0.50979557910415916894) +#define COS2_1 FIXR(0.60134488693504528054) +#define COS2_2 FIXR(0.89997622313641570463) +#define COS2_3 FIXR(2.56291544774150617881) + +#define COS3_0 FIXR(0.54119610014619698439) +#define COS3_1 FIXR(1.30656296487637652785) + +#define COS4_0 FIXR(0.70710678118654752439) + +/* butterfly operator */ +#define BF(a, b, c)\ +{\ + tmp0 = tab[a] + tab[b];\ + tmp1 = tab[a] - tab[b];\ + tab[a] = tmp0;\ + tab[b] = MULL(tmp1, c);\ } -/* header decoding. MUST check the header before because no - consistency check is done there */ -static void decode_header(MPADecodeContext *s, UINT32 header) +#define BF1(a, b, c, d)\ +{\ + BF(a, b, COS4_0);\ + BF(c, d, -COS4_0);\ + tab[c] += tab[d];\ +} + +#define BF2(a, b, c, d)\ +{\ + BF(a, b, COS4_0);\ + BF(c, d, -COS4_0);\ + tab[c] += tab[d];\ + tab[a] += tab[c];\ + tab[c] += tab[b];\ + tab[b] += tab[d];\ +} + +#define ADD(a, b) tab[a] += tab[b] + +/* DCT32 without 1/sqrt(2) coef zero scaling. */ +static void dct32(int32_t *out, int32_t *tab) +{ + int tmp0, tmp1; + + /* pass 1 */ + BF(0, 31, COS0_0); + BF(1, 30, COS0_1); + BF(2, 29, COS0_2); + BF(3, 28, COS0_3); + BF(4, 27, COS0_4); + BF(5, 26, COS0_5); + BF(6, 25, COS0_6); + BF(7, 24, COS0_7); + BF(8, 23, COS0_8); + BF(9, 22, COS0_9); + BF(10, 21, COS0_10); + BF(11, 20, COS0_11); + BF(12, 19, COS0_12); + BF(13, 18, COS0_13); + BF(14, 17, COS0_14); + BF(15, 16, COS0_15); + + /* pass 2 */ + BF(0, 15, COS1_0); + BF(1, 14, COS1_1); + BF(2, 13, COS1_2); + BF(3, 12, COS1_3); + BF(4, 11, COS1_4); + BF(5, 10, COS1_5); + BF(6, 9, COS1_6); + BF(7, 8, COS1_7); + + BF(16, 31, -COS1_0); + BF(17, 30, -COS1_1); + BF(18, 29, -COS1_2); + BF(19, 28, -COS1_3); + BF(20, 27, -COS1_4); + BF(21, 26, -COS1_5); + BF(22, 25, -COS1_6); + BF(23, 24, -COS1_7); + + /* pass 3 */ + BF(0, 7, COS2_0); + BF(1, 6, COS2_1); + BF(2, 5, COS2_2); + BF(3, 4, COS2_3); + + BF(8, 15, -COS2_0); + BF(9, 14, -COS2_1); + BF(10, 13, -COS2_2); + BF(11, 12, -COS2_3); + + BF(16, 23, COS2_0); + BF(17, 22, COS2_1); + BF(18, 21, COS2_2); + BF(19, 20, COS2_3); + + BF(24, 31, -COS2_0); + BF(25, 30, -COS2_1); + BF(26, 29, -COS2_2); + BF(27, 28, -COS2_3); + + /* pass 4 */ + BF(0, 3, COS3_0); + BF(1, 2, COS3_1); + + BF(4, 7, -COS3_0); + BF(5, 6, -COS3_1); + + BF(8, 11, COS3_0); + BF(9, 10, COS3_1); + + BF(12, 15, -COS3_0); + BF(13, 14, -COS3_1); + + BF(16, 19, COS3_0); + BF(17, 18, COS3_1); + + BF(20, 23, -COS3_0); + BF(21, 22, -COS3_1); + + BF(24, 27, COS3_0); + BF(25, 26, COS3_1); + + BF(28, 31, -COS3_0); + BF(29, 30, -COS3_1); + + /* pass 5 */ + BF1(0, 1, 2, 3); + BF2(4, 5, 6, 7); + BF1(8, 9, 10, 11); + BF2(12, 13, 14, 15); + BF1(16, 17, 18, 19); + BF2(20, 21, 22, 23); + BF1(24, 25, 26, 27); + BF2(28, 29, 30, 31); + + /* pass 6 */ + + ADD( 8, 12); + ADD(12, 10); + ADD(10, 14); + ADD(14, 9); + ADD( 9, 13); + ADD(13, 11); + ADD(11, 15); + + out[ 0] = tab[0]; + out[16] = tab[1]; + out[ 8] = tab[2]; + out[24] = tab[3]; + out[ 4] = tab[4]; + out[20] = tab[5]; + out[12] = tab[6]; + out[28] = tab[7]; + out[ 2] = tab[8]; + out[18] = tab[9]; + out[10] = tab[10]; + out[26] = tab[11]; + out[ 6] = tab[12]; + out[22] = tab[13]; + out[14] = tab[14]; + out[30] = tab[15]; + + ADD(24, 28); + ADD(28, 26); + ADD(26, 30); + ADD(30, 25); + ADD(25, 29); + ADD(29, 27); + ADD(27, 31); + + out[ 1] = tab[16] + tab[24]; + out[17] = tab[17] + tab[25]; + out[ 9] = tab[18] + tab[26]; + out[25] = tab[19] + tab[27]; + out[ 5] = tab[20] + tab[28]; + out[21] = tab[21] + tab[29]; + out[13] = tab[22] + tab[30]; + out[29] = tab[23] + tab[31]; + out[ 3] = tab[24] + tab[20]; + out[19] = tab[25] + tab[21]; + out[11] = tab[26] + tab[22]; + out[27] = tab[27] + tab[23]; + out[ 7] = tab[28] + tab[18]; + out[23] = tab[29] + tab[19]; + out[15] = tab[30] + tab[17]; + out[31] = tab[31]; +} + +#if FRAC_BITS <= 15 + +static inline int round_sample(int *sum) +{ + int sum1; + sum1 = (*sum) >> OUT_SHIFT; + *sum &= (1< OUT_MAX) + sum1 = OUT_MAX; + return sum1; +} + +#if defined(ARCH_POWERPC_405) + +/* signed 16x16 -> 32 multiply add accumulate */ +#define MACS(rt, ra, rb) \ + asm ("maclhw %0, %2, %3" : "=r" (rt) : "0" (rt), "r" (ra), "r" (rb)); + +/* signed 16x16 -> 32 multiply */ +#define MULS(ra, rb) \ + ({ int __rt; asm ("mullhw %0, %1, %2" : "=r" (__rt) : "r" (ra), "r" (rb)); __rt; }) + +#else + +/* signed 16x16 -> 32 multiply add accumulate */ +#define MACS(rt, ra, rb) rt += (ra) * (rb) + +/* signed 16x16 -> 32 multiply */ +#define MULS(ra, rb) ((ra) * (rb)) + +#endif + +#else + +static inline int round_sample(int64_t *sum) +{ + int sum1; + sum1 = (int)((*sum) >> OUT_SHIFT); + *sum &= (1< OUT_MAX) + sum1 = OUT_MAX; + return sum1; +} + +#define MULS(ra, rb) MUL64(ra, rb) + +#endif + +#define SUM8(sum, op, w, p) \ +{ \ + sum op MULS((w)[0 * 64], p[0 * 64]);\ + sum op MULS((w)[1 * 64], p[1 * 64]);\ + sum op MULS((w)[2 * 64], p[2 * 64]);\ + sum op MULS((w)[3 * 64], p[3 * 64]);\ + sum op MULS((w)[4 * 64], p[4 * 64]);\ + sum op MULS((w)[5 * 64], p[5 * 64]);\ + sum op MULS((w)[6 * 64], p[6 * 64]);\ + sum op MULS((w)[7 * 64], p[7 * 64]);\ +} + +#define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \ +{ \ + int tmp;\ + tmp = p[0 * 64];\ + sum1 op1 MULS((w1)[0 * 64], tmp);\ + sum2 op2 MULS((w2)[0 * 64], tmp);\ + tmp = p[1 * 64];\ + sum1 op1 MULS((w1)[1 * 64], tmp);\ + sum2 op2 MULS((w2)[1 * 64], tmp);\ + tmp = p[2 * 64];\ + sum1 op1 MULS((w1)[2 * 64], tmp);\ + sum2 op2 MULS((w2)[2 * 64], tmp);\ + tmp = p[3 * 64];\ + sum1 op1 MULS((w1)[3 * 64], tmp);\ + sum2 op2 MULS((w2)[3 * 64], tmp);\ + tmp = p[4 * 64];\ + sum1 op1 MULS((w1)[4 * 64], tmp);\ + sum2 op2 MULS((w2)[4 * 64], tmp);\ + tmp = p[5 * 64];\ + sum1 op1 MULS((w1)[5 * 64], tmp);\ + sum2 op2 MULS((w2)[5 * 64], tmp);\ + tmp = p[6 * 64];\ + sum1 op1 MULS((w1)[6 * 64], tmp);\ + sum2 op2 MULS((w2)[6 * 64], tmp);\ + tmp = p[7 * 64];\ + sum1 op1 MULS((w1)[7 * 64], tmp);\ + sum2 op2 MULS((w2)[7 * 64], tmp);\ +} + +void ff_mpa_synth_init(MPA_INT *window) { - struct frame *fr = &s->mpstr.fr; - int sample_rate, frame_size; + int i; + + /* max = 18760, max sum over all 16 coefs : 44736 */ + for(i=0;i<257;i++) { + int v; + v = mpa_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; + } +} + +/* 32 sub band synthesis filter. Input: 32 sub band samples, Output: + 32 samples. */ +/* XXX: optimize by avoiding ring buffer usage */ +void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset, + MPA_INT *window, int *dither_state, + OUT_INT *samples, int incr, + int32_t sb_samples[SBLIMIT]) +{ + int32_t tmp[32]; + register MPA_INT *synth_buf; + register const MPA_INT *w, *w2, *p; + int j, offset, v; + OUT_INT *samples2; +#if FRAC_BITS <= 15 + int sum, sum2; +#else + int64_t sum, sum2; +#endif + + dct32(tmp, sb_samples); + + offset = *synth_buf_offset; + synth_buf = synth_buf_ptr + offset; + + for(j=0;j<32;j++) { + v = tmp[j]; +#if FRAC_BITS <= 15 + /* NOTE: can cause a loss in precision if very high amplitude + sound */ + if (v > 32767) + v = 32767; + else if (v < -32768) + v = -32768; +#endif + synth_buf[j] = v; + } + /* copy to avoid wrap */ + memcpy(synth_buf + 512, synth_buf, 32 * sizeof(MPA_INT)); + + 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; +} + +#define C3 FIXHR(0.86602540378443864676/2) + +/* 0.5 / cos(pi*(2*i+1)/36) */ +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), +}; + +/* 12 points IMDCT. We compute it "by hand" by factorizing obvious + cases. */ +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(2*in3, C3); + + t1 = in0 - in4; + t2 = MULL(in1 - in5, icos36[4]); + + out[ 7]= + out[10]= t1 + t2; + out[ 1]= + out[ 4]= t1 - t2; + + in0 += in4>>1; + in4 = in0 + in2; + in1 += in5>>1; + in5 = MULL(in1 + in3, icos36[1]); + out[ 8]= + out[ 9]= in4 + in5; + out[ 2]= + out[ 3]= in4 - in5; + + in0 -= in2; + in1 = MULL(in1 - in3, icos36[7]); + out[ 0]= + out[ 5]= in0 - in1; + out[ 6]= + out[11]= in0 + in1; +} + +/* cos(pi*i/18) */ +#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) + + +/* using Lee like decomposition followed by hand coded 9 points DCT */ +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; +#if 0 +//more accurate but slower + int64_t t0, t1, t2, t3; + t2 = in1[2*4] + in1[2*8] - in1[2*2]; + + t3 = (in1[2*0] + (int64_t)(in1[2*6]>>1))<<32; + t1 = in1[2*0] - in1[2*6]; + tmp1[ 6] = t1 - (t2>>1); + tmp1[16] = t1 + t2; + + t0 = MUL64(2*(in1[2*2] + in1[2*4]), C2); + t1 = MUL64( in1[2*4] - in1[2*8] , -2*C8); + t2 = MUL64(2*(in1[2*2] + in1[2*8]), -C4); + + tmp1[10] = (t3 - t0 - t2) >> 32; + tmp1[ 2] = (t3 + t0 + t1) >> 32; + tmp1[14] = (t3 + t2 - t1) >> 32; + + tmp1[ 4] = MULH(2*(in1[2*5] + in1[2*7] - in1[2*1]), -C3); + t2 = MUL64(2*(in1[2*1] + in1[2*5]), C1); + t3 = MUL64( in1[2*5] - in1[2*7] , -2*C7); + t0 = MUL64(2*in1[2*3], C3); + + t1 = MUL64(2*(in1[2*1] + in1[2*7]), -C5); + + tmp1[ 0] = (t2 + t3 + t0) >> 32; + tmp1[12] = (t2 + t1 - t0) >> 32; + tmp1[ 8] = (t3 - t1 - t0) >> 32; +#else + 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; +#endif + } + + 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 = MULL(t3 + t2, icos36[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 = MULL(tmp[17], icos36[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]); +} + +/* header decoding. MUST check the header before because no + consistency check is done there. Return 1 if free format found and + that the frame size must be computed externally */ +static int decode_header(MPADecodeContext *s, uint32_t header) +{ + int sample_rate, frame_size, mpeg25, padding; + int sample_rate_index, bitrate_index; if (header & (1<<20)) { - fr->lsf = (header & (1<<19)) ? 0 : 1; - fr->mpeg25 = 0; + s->lsf = (header & (1<<19)) ? 0 : 1; + mpeg25 = 0; } else { - fr->lsf = 1; - fr->mpeg25 = 1; + s->lsf = 1; + mpeg25 = 1; } - + s->layer = 4 - ((header >> 17) & 3); /* extract frequency */ - fr->sampling_frequency = ((header >> 10) & 3); - sample_rate = mp_freq_tab[fr->sampling_frequency] >> (fr->lsf + fr->mpeg25); - fr->sampling_frequency += 3 * (fr->lsf + fr->mpeg25); - - s->error_protection = ((header>>16) & 1) ^ 1; - - fr->bitrate_index = ((header>>12)&0xf); - fr->padding = ((header>>9)&0x1); - fr->extension = ((header>>8)&0x1); - fr->mode = ((header>>6)&0x3); - fr->mode_ext = ((header>>4)&0x3); - fr->copyright = ((header>>3)&0x1); - fr->original = ((header>>2)&0x1); - fr->emphasis = header & 0x3; - - fr->stereo = (fr->mode == MPG_MD_MONO) ? 1 : 2; - - - frame_size = mp_bitrate_tab[fr->lsf][s->layer - 1][fr->bitrate_index]; - s->bit_rate = frame_size * 1000; - switch(s->layer) { - case 1: - frame_size = (frame_size * 12000) / sample_rate; - frame_size = ((frame_size + fr->padding) << 2); - break; - case 2: - frame_size = (frame_size * 144000) / sample_rate; - frame_size += fr->padding; - break; - case 3: - frame_size = (frame_size * 144000) / (sample_rate << fr->lsf); - frame_size += fr->padding; - break; - } - s->frame_size = frame_size; + sample_rate_index = (header >> 10) & 3; + sample_rate = mpa_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; -#if 0 - printf("layer%d, %d Hz, %d kbits/s, %s\n", - s->layer, s->sample_rate, s->bit_rate, fr->stereo ? "stereo" : "mono"); + bitrate_index = (header >> 12) & 0xf; + padding = (header >> 9) & 1; + //extension = (header >> 8) & 1; + s->mode = (header >> 6) & 3; + s->mode_ext = (header >> 4) & 3; + //copyright = (header >> 3) & 1; + //original = (header >> 2) & 1; + //emphasis = header & 3; + + if (s->mode == MPA_MONO) + s->nb_channels = 1; + else + s->nb_channels = 2; + + if (bitrate_index != 0) { + frame_size = mpa_bitrate_tab[s->lsf][s->layer - 1][bitrate_index]; + s->bit_rate = frame_size * 1000; + switch(s->layer) { + case 1: + frame_size = (frame_size * 12000) / sample_rate; + frame_size = (frame_size + padding) * 4; + break; + case 2: + frame_size = (frame_size * 144000) / sample_rate; + frame_size += padding; + break; + default: + case 3: + frame_size = (frame_size * 144000) / (sample_rate << s->lsf); + frame_size += padding; + break; + } + s->frame_size = frame_size; + } else { + /* if no frame size computed, signal it */ + if (!s->free_format_frame_size) + return 1; + /* free format: compute bitrate and real frame size from the + frame size we extracted by reading the bitstream */ + s->frame_size = s->free_format_frame_size; + switch(s->layer) { + case 1: + s->frame_size += padding * 4; + s->bit_rate = (s->frame_size * sample_rate) / 48000; + break; + case 2: + s->frame_size += padding; + s->bit_rate = (s->frame_size * sample_rate) / 144000; + break; + default: + case 3: + s->frame_size += padding; + s->bit_rate = (s->frame_size * (sample_rate << s->lsf)) / 144000; + break; + } + } + +#if defined(DEBUG) + printf("layer%d, %d Hz, %d kbits/s, ", + s->layer, s->sample_rate, s->bit_rate); + if (s->nb_channels == 2) { + if (s->layer == 3) { + if (s->mode_ext & MODE_EXT_MS_STEREO) + printf("ms-"); + if (s->mode_ext & MODE_EXT_I_STEREO) + printf("i-"); + } + printf("stereo"); + } else { + printf("mono"); + } + printf("\n"); #endif + return 0; } -static int mp_decode_frame(MPADecodeContext *s, - short *samples) +/* useful helper to get mpeg audio stream infos. Return -1 if error in + header, otherwise the coded frame size in bytes */ +int mpa_decode_header(AVCodecContext *avctx, uint32_t head) { - int nb_bytes; - - init_get_bits(&s->gb, s->inbuf + HEADER_SIZE, s->inbuf_ptr - s->inbuf - HEADER_SIZE); - - /* skip error protection field */ - if (s->error_protection) - get_bits(&s->gb, 16); + MPADecodeContext s1, *s = &s1; + memset( s, 0, sizeof(MPADecodeContext) ); - /* XXX: horrible: global! */ - gmp = &s->mpstr; - gmp_s = s; - gmp_gb = &s->gb; + if (ff_mpa_check_header(head) != 0) + return -1; + + if (decode_header(s, head) != 0) { + return -1; + } - nb_bytes = 0; switch(s->layer) { case 1: - do_layer1(&s->mpstr.fr,(unsigned char *)samples, &nb_bytes); + avctx->frame_size = 384; break; case 2: - do_layer2(&s->mpstr.fr,(unsigned char *)samples, &nb_bytes); - break; - case 3: - do_layer3(&s->mpstr.fr,(unsigned char *)samples, &nb_bytes); - s->inbuf_index ^= 1; - s->inbuf = &s->inbuf1[s->inbuf_index][BACKSTEP_SIZE]; - s->old_frame_size = s->frame_size; + avctx->frame_size = 1152; break; default: + case 3: + if (s->lsf) + avctx->frame_size = 576; + else + avctx->frame_size = 1152; break; } - return nb_bytes; + + avctx->sample_rate = s->sample_rate; + avctx->channels = s->nb_channels; + avctx->bit_rate = s->bit_rate; + avctx->sub_id = s->layer; + return s->frame_size; +} + +/* return the number of decoded frames */ +static int mp_decode_layer1(MPADecodeContext *s) +{ + int bound, i, v, n, ch, j, mant; + uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT]; + uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT]; + + if (s->mode == MPA_JSTEREO) + bound = (s->mode_ext + 1) * 4; + else + bound = SBLIMIT; + + /* allocation bits */ + for(i=0;inb_channels;ch++) { + allocation[ch][i] = get_bits(&s->gb, 4); + } + } + for(i=bound;igb, 4); + } + + /* scale factors */ + for(i=0;inb_channels;ch++) { + if (allocation[ch][i]) + scale_factors[ch][i] = get_bits(&s->gb, 6); + } + } + for(i=bound;igb, 6); + scale_factors[1][i] = get_bits(&s->gb, 6); + } + } + + /* compute samples */ + for(j=0;j<12;j++) { + for(i=0;inb_channels;ch++) { + n = allocation[ch][i]; + if (n) { + mant = get_bits(&s->gb, n + 1); + v = l1_unscale(n, mant, scale_factors[ch][i]); + } else { + v = 0; + } + s->sb_samples[ch][j][i] = v; + } + } + for(i=bound;igb, n + 1); + v = l1_unscale(n, mant, scale_factors[0][i]); + s->sb_samples[0][j][i] = v; + v = l1_unscale(n, mant, scale_factors[1][i]); + s->sb_samples[1][j][i] = v; + } else { + s->sb_samples[0][j][i] = 0; + s->sb_samples[1][j][i] = 0; + } + } + } + return 12; +} + +/* bitrate is in kb/s */ +int l2_select_table(int bitrate, int nb_channels, int freq, int lsf) +{ + int ch_bitrate, table; + + ch_bitrate = bitrate / nb_channels; + if (!lsf) { + if ((freq == 48000 && ch_bitrate >= 56) || + (ch_bitrate >= 56 && ch_bitrate <= 80)) + table = 0; + else if (freq != 48000 && ch_bitrate >= 96) + table = 1; + else if (freq != 32000 && ch_bitrate <= 48) + table = 2; + else + table = 3; + } else { + table = 4; + } + return table; +} + +static int mp_decode_layer2(MPADecodeContext *s) +{ + int sblimit; /* number of used subbands */ + const unsigned char *alloc_table; + int table, bit_alloc_bits, i, j, ch, bound, v; + unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT]; + unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; + unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf; + int scale, qindex, bits, steps, k, l, m, b; + + /* select decoding table */ + table = l2_select_table(s->bit_rate / 1000, s->nb_channels, + s->sample_rate, s->lsf); + sblimit = sblimit_table[table]; + alloc_table = alloc_tables[table]; + + if (s->mode == MPA_JSTEREO) + bound = (s->mode_ext + 1) * 4; + else + bound = sblimit; + + dprintf("bound=%d sblimit=%d\n", bound, sblimit); + + /* sanity check */ + if( bound > sblimit ) bound = sblimit; + + /* parse bit allocation */ + j = 0; + for(i=0;inb_channels;ch++) { + bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits); + } + j += 1 << bit_alloc_bits; + } + for(i=bound;igb, bit_alloc_bits); + bit_alloc[0][i] = v; + bit_alloc[1][i] = v; + j += 1 << bit_alloc_bits; + } + +#ifdef DEBUG + { + for(ch=0;chnb_channels;ch++) { + for(i=0;inb_channels;ch++) { + if (bit_alloc[ch][i]) + scale_code[ch][i] = get_bits(&s->gb, 2); + } + } + + /* scale factors */ + for(i=0;inb_channels;ch++) { + if (bit_alloc[ch][i]) { + sf = scale_factors[ch][i]; + switch(scale_code[ch][i]) { + default: + case 0: + sf[0] = get_bits(&s->gb, 6); + sf[1] = get_bits(&s->gb, 6); + sf[2] = get_bits(&s->gb, 6); + break; + case 2: + sf[0] = get_bits(&s->gb, 6); + sf[1] = sf[0]; + sf[2] = sf[0]; + break; + case 1: + sf[0] = get_bits(&s->gb, 6); + sf[2] = get_bits(&s->gb, 6); + sf[1] = sf[0]; + break; + case 3: + sf[0] = get_bits(&s->gb, 6); + sf[2] = get_bits(&s->gb, 6); + sf[1] = sf[2]; + break; + } + } + } + } + +#ifdef DEBUG + for(ch=0;chnb_channels;ch++) { + for(i=0;inb_channels;ch++) { + b = bit_alloc[ch][i]; + if (b) { + scale = scale_factors[ch][i][k]; + qindex = alloc_table[j+b]; + bits = quant_bits[qindex]; + if (bits < 0) { + /* 3 values at the same time */ + v = get_bits(&s->gb, -bits); + steps = quant_steps[qindex]; + s->sb_samples[ch][k * 12 + l + 0][i] = + l2_unscale_group(steps, v % steps, scale); + v = v / steps; + s->sb_samples[ch][k * 12 + l + 1][i] = + l2_unscale_group(steps, v % steps, scale); + v = v / steps; + s->sb_samples[ch][k * 12 + l + 2][i] = + l2_unscale_group(steps, v, scale); + } else { + for(m=0;m<3;m++) { + v = get_bits(&s->gb, bits); + v = l1_unscale(bits - 1, v, scale); + s->sb_samples[ch][k * 12 + l + m][i] = v; + } + } + } else { + s->sb_samples[ch][k * 12 + l + 0][i] = 0; + s->sb_samples[ch][k * 12 + l + 1][i] = 0; + s->sb_samples[ch][k * 12 + l + 2][i] = 0; + } + } + /* next subband in alloc table */ + j += 1 << bit_alloc_bits; + } + /* XXX: find a way to avoid this duplication of code */ + for(i=bound;igb, -bits); + steps = quant_steps[qindex]; + mant = v % steps; + v = v / steps; + s->sb_samples[0][k * 12 + l + 0][i] = + l2_unscale_group(steps, mant, scale0); + s->sb_samples[1][k * 12 + l + 0][i] = + l2_unscale_group(steps, mant, scale1); + mant = v % steps; + v = v / steps; + s->sb_samples[0][k * 12 + l + 1][i] = + l2_unscale_group(steps, mant, scale0); + s->sb_samples[1][k * 12 + l + 1][i] = + l2_unscale_group(steps, mant, scale1); + s->sb_samples[0][k * 12 + l + 2][i] = + l2_unscale_group(steps, v, scale0); + s->sb_samples[1][k * 12 + l + 2][i] = + l2_unscale_group(steps, v, scale1); + } else { + for(m=0;m<3;m++) { + mant = get_bits(&s->gb, bits); + s->sb_samples[0][k * 12 + l + m][i] = + l1_unscale(bits - 1, mant, scale0); + s->sb_samples[1][k * 12 + l + m][i] = + l1_unscale(bits - 1, mant, scale1); + } + } + } else { + s->sb_samples[0][k * 12 + l + 0][i] = 0; + s->sb_samples[0][k * 12 + l + 1][i] = 0; + s->sb_samples[0][k * 12 + l + 2][i] = 0; + s->sb_samples[1][k * 12 + l + 0][i] = 0; + s->sb_samples[1][k * 12 + l + 1][i] = 0; + s->sb_samples[1][k * 12 + l + 2][i] = 0; + } + /* next subband in alloc table */ + j += 1 << bit_alloc_bits; + } + /* fill remaining samples to zero */ + for(i=sblimit;inb_channels;ch++) { + s->sb_samples[ch][k * 12 + l + 0][i] = 0; + s->sb_samples[ch][k * 12 + l + 1][i] = 0; + s->sb_samples[ch][k * 12 + l + 2][i] = 0; + } + } + } + } + return 3 * 12; } /* - * seek back in the stream for backstep bytes (at most 511 bytes, and - * at most in last frame). Note that this is slightly incorrect (data - * can span more than one block!) + * Seek back in the stream for backstep bytes (at most 511 bytes) */ -int set_pointer(long backstep) +static void seek_to_maindata(MPADecodeContext *s, unsigned int backstep) { - UINT8 *ptr; + uint8_t *ptr; /* compute current position in stream */ - ptr = gmp_gb->buf_ptr - (gmp_gb->bit_cnt >> 3); + ptr = (uint8_t *)(s->gb.buffer + (get_bits_count(&s->gb)>>3)); + /* copy old data before current one */ ptr -= backstep; - memcpy(ptr, gmp_s->inbuf1[gmp_s->inbuf_index ^ 1] + - BACKSTEP_SIZE + gmp_s->old_frame_size - backstep, backstep); + memcpy(ptr, s->inbuf1[s->inbuf_index ^ 1] + + BACKSTEP_SIZE + s->old_frame_size - backstep, backstep); /* init get bits again */ - init_get_bits(gmp_gb, ptr, gmp_s->frame_size + backstep); + init_get_bits(&s->gb, ptr, (s->frame_size + backstep)*8); + + /* prepare next buffer */ + s->inbuf_index ^= 1; + s->inbuf = &s->inbuf1[s->inbuf_index][BACKSTEP_SIZE]; + s->old_frame_size = s->frame_size; +} + +static inline 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 void exponents_from_scale_factors(MPADecodeContext *s, + GranuleDef *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 = mpa_pretab[g->preflag]; + for(i=0;ilong_end;i++) { + v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift); + 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); + for(j=len;j>0;j--) + *exp_ptr++ = v0; + } + } + } +} + +/* handle n = 0 too */ +static inline int get_bitsz(GetBitContext *s, int n) +{ + if (n == 0) + return 0; + else + return get_bits(s, n); +} + +static int huffman_decode(MPADecodeContext *s, GranuleDef *g, + int16_t *exponents, int end_pos) +{ + int s_index; + int linbits, code, x, y, l, v, i, j, k, pos; + GetBitContext last_gb; + VLC *vlc; + uint8_t *code_table; + + /* low frequencies (called big values) */ + s_index = 0; + for(i=0;i<3;i++) { + j = g->region_size[i]; + if (j == 0) + continue; + /* select vlc table */ + k = g->table_select[i]; + l = mpa_huff_data[k][0]; + linbits = mpa_huff_data[k][1]; + vlc = &huff_vlc[l]; + code_table = huff_code_table[l]; + + /* read huffcode and compute each couple */ + for(;j>0;j--) { + if (get_bits_count(&s->gb) >= end_pos) + break; + if (code_table) { + code = get_vlc2(&s->gb, vlc->table, 8, 2); + if (code < 0) + return -1; + y = code_table[code]; + x = y >> 4; + y = y & 0x0f; + } else { + x = 0; + y = 0; + } + dprintf("region=%d n=%d x=%d y=%d exp=%d\n", + i, g->region_size[i] - j, x, y, exponents[s_index]); + if (x) { + if (x == 15) + x += get_bitsz(&s->gb, linbits); + v = l3_unscale(x, exponents[s_index]); + if (get_bits1(&s->gb)) + v = -v; + } else { + v = 0; + } + g->sb_hybrid[s_index++] = v; + if (y) { + if (y == 15) + y += get_bitsz(&s->gb, linbits); + v = l3_unscale(y, exponents[s_index]); + if (get_bits1(&s->gb)) + v = -v; + } else { + v = 0; + } + g->sb_hybrid[s_index++] = v; + } + } + + /* high frequencies */ + vlc = &huff_quad_vlc[g->count1table_select]; + last_gb.buffer = NULL; + while (s_index <= 572) { + pos = get_bits_count(&s->gb); + if (pos >= end_pos) { + if (pos > end_pos && last_gb.buffer != NULL) { + /* some encoders generate an incorrect size for this + part. We must go back into the data */ + s_index -= 4; + s->gb = last_gb; + } + break; + } + last_gb= s->gb; + + code = get_vlc2(&s->gb, vlc->table, vlc->bits, 2); + dprintf("t=%d code=%d\n", g->count1table_select, code); + if (code < 0) + return -1; + for(i=0;i<4;i++) { + if (code & (8 >> i)) { + /* non zero value. Could use a hand coded function for + 'one' value */ + v = l3_unscale(1, exponents[s_index]); + if(get_bits1(&s->gb)) + v = -v; + } else { + v = 0; + } + g->sb_hybrid[s_index++] = v; + } + } + while (s_index < 576) + g->sb_hybrid[s_index++] = 0; return 0; } +/* Reorder short blocks from bitstream order to interleaved order. It + would be faster to do it in parsing, but the code would be far more + complicated */ +static void reorder_block(MPADecodeContext *s, GranuleDef *g) +{ + int i, j, k, 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; + for(k=0;k<3;k++) { + dst = tmp + k; + for(j=len;j>0;j--) { + *dst = *ptr++; + dst += 3; + } + } + memcpy(ptr1, tmp, len * 3 * sizeof(int32_t)); + } +} + +#define ISQRT2 FIXR(0.70710678118654752440) + +static void compute_stereo(MPADecodeContext *s, + GranuleDef *g0, GranuleDef *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]; + + /* intensity stereo */ + 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 void compute_antialias_integer(MPADecodeContext *s, + GranuleDef *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_antialias_float(MPADecodeContext *s, + GranuleDef *g) +{ + int32_t *ptr; + 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--) { + float tmp0, tmp1; + float *csa = &csa_table_float[0][0]; +#define FLOAT_AA(j)\ + tmp0= ptr[-1-j];\ + tmp1= ptr[ j];\ + ptr[-1-j] = lrintf(tmp0 * csa[0+4*j] - tmp1 * csa[1+4*j]);\ + ptr[ j] = lrintf(tmp0 * csa[1+4*j] + tmp1 * csa[0+4*j]); + + FLOAT_AA(0) + FLOAT_AA(1) + FLOAT_AA(2) + FLOAT_AA(3) + FLOAT_AA(4) + FLOAT_AA(5) + FLOAT_AA(6) + FLOAT_AA(7) + + ptr += 18; + } +} + +static void compute_imdct(MPADecodeContext *s, + GranuleDef *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;jlsf) { + main_data_begin = get_bits(&s->gb, 8); + if (s->nb_channels == 2) + private_bits = get_bits(&s->gb, 2); + else + private_bits = get_bits(&s->gb, 1); + 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++) { + dprintf("gr=%d ch=%d: side_info\n", gr, 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); + dprintf("region1=%d region2=%d\n", + region_address1, region_address2); + 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 (k > g->big_values) + 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 */ + + if (s->sample_rate_index != 8) + g->short_start = 3; + else + g->short_start = 2; + } 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); + dprintf("block_type=%d switch_point=%d\n", + g->block_type, g->switch_point); + } + } + + if (!s->adu_mode) { + /* now we get bits from the main_data_begin offset */ + dprintf("seekback: %d\n", main_data_begin); + seek_to_maindata(s, main_data_begin); + } + + 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]; + dprintf("slen1=%d slen2=%d\n", slen1, slen2); + if (g->block_type == 2) { + n = g->switch_point ? 17 : 18; + j = 0; + for(i=0;iscale_factors[j++] = get_bitsz(&s->gb, slen1); + for(i=0;i<18;i++) + g->scale_factors[j++] = get_bitsz(&s->gb, slen2); + for(i=0;i<3;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; + for(i=0;iscale_factors[j++] = get_bitsz(&s->gb, slen); + } else { + /* simply copy from last granule */ + for(i=0;iscale_factors[j] = sc[j]; + j++; + } + } + } + g->scale_factors[j++] = 0; + } +#if defined(DEBUG) + { + printf("scfsi=%x gr=%d ch=%d scale_factors:\n", + g->scfsi, gr, ch); + for(i=0;iscale_factors[i]); + printf("\n"); + } +#endif + } 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]; + for(i=0;iscale_factors[j++] = get_bitsz(&s->gb, sl); + } + /* XXX: should compute exact size */ + for(;j<40;j++) + g->scale_factors[j] = 0; +#if defined(DEBUG) + { + printf("gr=%d ch=%d scale_factors:\n", + gr, ch); + for(i=0;i<40;i++) + printf(" %d", g->scale_factors[i]); + printf("\n"); + } +#endif + } + + 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; +#if defined(DEBUG) + sample_dump(0, g->sb_hybrid, 576); +#endif + + /* skip extension bits */ + bits_left = g->part2_3_length - (get_bits_count(&s->gb) - bits_pos); + if (bits_left < 0) { + dprintf("bits_left=%d\n", bits_left); + return -1; + } + while (bits_left >= 16) { + skip_bits(&s->gb, 16); + bits_left -= 16; + } + if (bits_left > 0) + skip_bits(&s->gb, bits_left); + } /* 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); +#if defined(DEBUG) + sample_dump(0, g->sb_hybrid, 576); +#endif + s->compute_antialias(s, g); +#if defined(DEBUG) + sample_dump(1, g->sb_hybrid, 576); +#endif + compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]); +#if defined(DEBUG) + sample_dump(2, &s->sb_samples[ch][18 * gr][0], 576); +#endif + } + } /* gr */ + return nb_granules * 18; +} + +static int mp_decode_frame(MPADecodeContext *s, + OUT_INT *samples) +{ + int i, nb_frames, ch; + OUT_INT *samples_ptr; + + init_get_bits(&s->gb, s->inbuf + HEADER_SIZE, + (s->inbuf_ptr - s->inbuf - HEADER_SIZE)*8); + + /* skip error protection field */ + if (s->error_protection) + get_bits(&s->gb, 16); + + dprintf("frame %d:\n", s->frame_count); + switch(s->layer) { + case 1: + nb_frames = mp_decode_layer1(s); + break; + case 2: + nb_frames = mp_decode_layer2(s); + break; + case 3: + default: + nb_frames = mp_decode_layer3(s); + break; + } +#if defined(DEBUG) + for(i=0;inb_channels;ch++) { + int j; + printf("%d-%d:", i, ch); + for(j=0;jsb_samples[ch][i][j] / FRAC_ONE); + printf("\n"); + } + } +#endif + /* 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; + } + } +#ifdef DEBUG + s->frame_count++; +#endif + return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels; +} + static int decode_frame(AVCodecContext * avctx, - void *data, int *data_size, - UINT8 * buf, int buf_size) + void *data, int *data_size, + uint8_t * buf, int buf_size) { MPADecodeContext *s = avctx->priv_data; - UINT32 header; - UINT8 *buf_ptr; + uint32_t header; + uint8_t *buf_ptr; int len, out_size; - short *out_samples = data; + OUT_INT *out_samples = data; - *data_size = 0; buf_ptr = buf; while (buf_size > 0) { - len = s->inbuf_ptr - s->inbuf; - if (s->frame_size == 0) { - /* no header seen : find one. We need at least 7 bytes to parse it */ - len = HEADER_SIZE - len; - if (len > buf_size) - len = buf_size; - memcpy(s->inbuf_ptr, buf_ptr, len); - buf_ptr += len; - s->inbuf_ptr += len; - buf_size -= len; - if ((s->inbuf_ptr - s->inbuf) == HEADER_SIZE) { - header = (s->inbuf[0] << 24) | (s->inbuf[1] << 16) | - (s->inbuf[2] << 8) | s->inbuf[3]; - if (check_header(header) < 0) { - /* no sync found : move by one byte (inefficient, but simple!) */ - memcpy(s->inbuf, s->inbuf + 1, HEADER_SIZE - 1); - s->inbuf_ptr--; - } else { - decode_header(s, header); - /* update codec info */ - avctx->sample_rate = s->sample_rate; - avctx->channels = s->mpstr.fr.stereo ? 2 : 1; - avctx->bit_rate = s->bit_rate; - } - } - } else if (len < s->frame_size) { - len = s->frame_size - len; - if (len > buf_size) - len = buf_size; - - memcpy(s->inbuf_ptr, buf_ptr, len); - buf_ptr += len; - s->inbuf_ptr += len; - buf_size -= len; - } else { - out_size = mp_decode_frame(s, out_samples); - s->inbuf_ptr = s->inbuf; - s->frame_size = 0; - *data_size = out_size; - break; - } + len = s->inbuf_ptr - s->inbuf; + if (s->frame_size == 0) { + /* special case for next header for first frame in free + format case (XXX: find a simpler method) */ + if (s->free_format_next_header != 0) { + s->inbuf[0] = s->free_format_next_header >> 24; + s->inbuf[1] = s->free_format_next_header >> 16; + s->inbuf[2] = s->free_format_next_header >> 8; + s->inbuf[3] = s->free_format_next_header; + s->inbuf_ptr = s->inbuf + 4; + s->free_format_next_header = 0; + goto got_header; + } + /* no header seen : find one. We need at least HEADER_SIZE + bytes to parse it */ + len = HEADER_SIZE - len; + if (len > buf_size) + len = buf_size; + if (len > 0) { + memcpy(s->inbuf_ptr, buf_ptr, len); + buf_ptr += len; + buf_size -= len; + s->inbuf_ptr += len; + } + if ((s->inbuf_ptr - s->inbuf) >= HEADER_SIZE) { + got_header: + header = (s->inbuf[0] << 24) | (s->inbuf[1] << 16) | + (s->inbuf[2] << 8) | s->inbuf[3]; + + if (ff_mpa_check_header(header) < 0) { + /* no sync found : move by one byte (inefficient, but simple!) */ + memmove(s->inbuf, s->inbuf + 1, s->inbuf_ptr - s->inbuf - 1); + s->inbuf_ptr--; + dprintf("skip %x\n", header); + /* reset free format frame size to give a chance + to get a new bitrate */ + s->free_format_frame_size = 0; + } else { + if (decode_header(s, header) == 1) { + /* free format: prepare to compute frame size */ + s->frame_size = -1; + } + /* update codec info */ + avctx->sample_rate = s->sample_rate; + avctx->channels = s->nb_channels; + avctx->bit_rate = s->bit_rate; + avctx->sub_id = s->layer; + switch(s->layer) { + case 1: + avctx->frame_size = 384; + break; + case 2: + avctx->frame_size = 1152; + break; + case 3: + if (s->lsf) + avctx->frame_size = 576; + else + avctx->frame_size = 1152; + break; + } + } + } + } else if (s->frame_size == -1) { + /* free format : find next sync to compute frame size */ + len = MPA_MAX_CODED_FRAME_SIZE - len; + if (len > buf_size) + len = buf_size; + if (len == 0) { + /* frame too long: resync */ + s->frame_size = 0; + memmove(s->inbuf, s->inbuf + 1, s->inbuf_ptr - s->inbuf - 1); + s->inbuf_ptr--; + } else { + uint8_t *p, *pend; + uint32_t header1; + int padding; + + memcpy(s->inbuf_ptr, buf_ptr, len); + /* check for header */ + p = s->inbuf_ptr - 3; + pend = s->inbuf_ptr + len - 4; + while (p <= pend) { + header = (p[0] << 24) | (p[1] << 16) | + (p[2] << 8) | p[3]; + header1 = (s->inbuf[0] << 24) | (s->inbuf[1] << 16) | + (s->inbuf[2] << 8) | s->inbuf[3]; + /* check with high probability that we have a + valid header */ + if ((header & SAME_HEADER_MASK) == + (header1 & SAME_HEADER_MASK)) { + /* header found: update pointers */ + len = (p + 4) - s->inbuf_ptr; + buf_ptr += len; + buf_size -= len; + s->inbuf_ptr = p; + /* compute frame size */ + s->free_format_next_header = header; + s->free_format_frame_size = s->inbuf_ptr - s->inbuf; + padding = (header1 >> 9) & 1; + if (s->layer == 1) + s->free_format_frame_size -= padding * 4; + else + s->free_format_frame_size -= padding; + dprintf("free frame size=%d padding=%d\n", + s->free_format_frame_size, padding); + decode_header(s, header1); + goto next_data; + } + p++; + } + /* not found: simply increase pointers */ + buf_ptr += len; + s->inbuf_ptr += len; + buf_size -= len; + } + } else if (len < s->frame_size) { + if (s->frame_size > MPA_MAX_CODED_FRAME_SIZE) + s->frame_size = MPA_MAX_CODED_FRAME_SIZE; + len = s->frame_size - len; + if (len > buf_size) + len = buf_size; + memcpy(s->inbuf_ptr, buf_ptr, len); + buf_ptr += len; + s->inbuf_ptr += len; + buf_size -= len; + } + next_data: + if (s->frame_size > 0 && + (s->inbuf_ptr - s->inbuf) >= s->frame_size) { + if (avctx->parse_only) { + /* simply return the frame data */ + *(uint8_t **)data = s->inbuf; + out_size = s->inbuf_ptr - s->inbuf; + } else { + out_size = mp_decode_frame(s, out_samples); + } + s->inbuf_ptr = s->inbuf; + s->frame_size = 0; + if(out_size>=0) + *data_size = out_size; + else + av_log(avctx, AV_LOG_DEBUG, "Error while decoding mpeg audio frame\n"); //FIXME return -1 / but also return the number of bytes consumed + break; + } } return buf_ptr - buf; } -AVCodec mp3_decoder = + +static int decode_frame_adu(AVCodecContext * avctx, + void *data, int *data_size, + uint8_t * buf, int buf_size) { - "mpegaudio", + MPADecodeContext *s = avctx->priv_data; + uint32_t header; + int len, out_size; + OUT_INT *out_samples = data; + + len = buf_size; + + // Discard too short frames + if (buf_size < HEADER_SIZE) { + *data_size = 0; + return buf_size; + } + + + if (len > MPA_MAX_CODED_FRAME_SIZE) + len = MPA_MAX_CODED_FRAME_SIZE; + + memcpy(s->inbuf, buf, len); + s->inbuf_ptr = s->inbuf + len; + + // Get header and restore sync word + header = (s->inbuf[0] << 24) | (s->inbuf[1] << 16) | + (s->inbuf[2] << 8) | s->inbuf[3] | 0xffe00000; + + if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame + *data_size = 0; + return buf_size; + } + + decode_header(s, header); + /* update codec info */ + avctx->sample_rate = s->sample_rate; + avctx->channels = s->nb_channels; + avctx->bit_rate = s->bit_rate; + avctx->sub_id = s->layer; + + avctx->frame_size=s->frame_size = len; + + if (avctx->parse_only) { + /* simply return the frame data */ + *(uint8_t **)data = s->inbuf; + out_size = s->inbuf_ptr - s->inbuf; + } else { + out_size = mp_decode_frame(s, out_samples); + } + + *data_size = out_size; + return buf_size; +} + + +/* Next 3 arrays are indexed by channel config number (passed via codecdata) */ +static int mp3Frames[16] = {0,1,1,2,3,3,4,5,2}; /* number of mp3 decoder instances */ +static int mp3Channels[16] = {0,1,2,3,4,5,6,8,4}; /* total output channels */ +/* offsets into output buffer, assume output order is FL FR BL BR C LFE */ +static int chan_offset[9][5] = { + {0}, + {0}, // C + {0}, // FLR + {2,0}, // C FLR + {2,0,3}, // C FLR BS + {4,0,2}, // C FLR BLRS + {4,0,2,5}, // C FLR BLRS LFE + {4,0,2,6,5}, // C FLR BLRS BLR LFE + {0,2} // FLR BLRS +}; + + +static int decode_init_mp3on4(AVCodecContext * avctx) +{ + MP3On4DecodeContext *s = avctx->priv_data; + int i; + + if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) { + av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n"); + return -1; + } + + s->chan_cfg = (((unsigned char *)avctx->extradata)[1] >> 3) & 0x0f; + s->frames = mp3Frames[s->chan_cfg]; + if(!s->frames) { + av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n"); + return -1; + } + avctx->channels = mp3Channels[s->chan_cfg]; + + /* Init the first mp3 decoder in standard way, so that all tables get builded + * We replace avctx->priv_data with the context of the first decoder so that + * decode_init() does not have to be changed. + * Other decoders will be inited here copying data from the first context + */ + // Allocate zeroed memory for the first decoder context + s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext)); + // Put decoder context in place to make init_decode() happy + avctx->priv_data = s->mp3decctx[0]; + decode_init(avctx); + // Restore mp3on4 context pointer + avctx->priv_data = s; + s->mp3decctx[0]->adu_mode = 1; // Set adu mode + + /* Create a separate codec/context for each frame (first is already ok). + * Each frame is 1 or 2 channels - up to 5 frames allowed + */ + for (i = 1; i < s->frames; i++) { + s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext)); + s->mp3decctx[i]->compute_antialias = s->mp3decctx[0]->compute_antialias; + s->mp3decctx[i]->inbuf = &s->mp3decctx[i]->inbuf1[0][BACKSTEP_SIZE]; + s->mp3decctx[i]->inbuf_ptr = s->mp3decctx[i]->inbuf; + s->mp3decctx[i]->adu_mode = 1; + } + + return 0; +} + + +static int decode_close_mp3on4(AVCodecContext * avctx) +{ + MP3On4DecodeContext *s = avctx->priv_data; + int i; + + for (i = 0; i < s->frames; i++) + if (s->mp3decctx[i]) + av_free(s->mp3decctx[i]); + + return 0; +} + + +static int decode_frame_mp3on4(AVCodecContext * avctx, + void *data, int *data_size, + uint8_t * buf, int buf_size) +{ + MP3On4DecodeContext *s = avctx->priv_data; + MPADecodeContext *m; + int len, out_size = 0; + uint32_t header; + OUT_INT *out_samples = data; + OUT_INT decoded_buf[MPA_FRAME_SIZE * MPA_MAX_CHANNELS]; + OUT_INT *outptr, *bp; + int fsize; + unsigned char *start2 = buf, *start; + int fr, i, j, n; + int off = avctx->channels; + int *coff = chan_offset[s->chan_cfg]; + + len = buf_size; + + // Discard too short frames + if (buf_size < HEADER_SIZE) { + *data_size = 0; + return buf_size; + } + + // If only one decoder interleave is not needed + outptr = s->frames == 1 ? out_samples : decoded_buf; + + for (fr = 0; fr < s->frames; fr++) { + start = start2; + fsize = (start[0] << 4) | (start[1] >> 4); + start2 += fsize; + if (fsize > len) + fsize = len; + len -= fsize; + if (fsize > MPA_MAX_CODED_FRAME_SIZE) + fsize = MPA_MAX_CODED_FRAME_SIZE; + m = s->mp3decctx[fr]; + assert (m != NULL); + /* copy original to new */ + m->inbuf_ptr = m->inbuf + fsize; + memcpy(m->inbuf, start, fsize); + + // Get header + header = (m->inbuf[0] << 24) | (m->inbuf[1] << 16) | + (m->inbuf[2] << 8) | m->inbuf[3] | 0xfff00000; + + if (ff_mpa_check_header(header) < 0) { // Bad header, discard block + *data_size = 0; + return buf_size; + } + + decode_header(m, header); + mp_decode_frame(m, decoded_buf); + + n = MPA_FRAME_SIZE * m->nb_channels; + out_size += n * sizeof(OUT_INT); + if(s->frames > 1) { + /* interleave output data */ + bp = out_samples + coff[fr]; + if(m->nb_channels == 1) { + for(j = 0; j < n; j++) { + *bp = decoded_buf[j]; + bp += off; + } + } else { + for(j = 0; j < n; j++) { + bp[0] = decoded_buf[j++]; + bp[1] = decoded_buf[j]; + bp += off; + } + } + } + } + + /* update codec info */ + avctx->sample_rate = s->mp3decctx[0]->sample_rate; + avctx->frame_size= buf_size; + avctx->bit_rate = 0; + for (i = 0; i < s->frames; i++) + avctx->bit_rate += s->mp3decctx[i]->bit_rate; + + *data_size = out_size; + return buf_size; +} + + +AVCodec mp2_decoder = +{ + "mp2", CODEC_TYPE_AUDIO, CODEC_ID_MP2, sizeof(MPADecodeContext), @@ -290,4 +2843,44 @@ AVCodec mp3_decoder = NULL, NULL, decode_frame, + CODEC_CAP_PARSE_ONLY, +}; + +AVCodec mp3_decoder = +{ + "mp3", + CODEC_TYPE_AUDIO, + CODEC_ID_MP3, + sizeof(MPADecodeContext), + decode_init, + NULL, + NULL, + decode_frame, + CODEC_CAP_PARSE_ONLY, +}; + +AVCodec mp3adu_decoder = +{ + "mp3adu", + CODEC_TYPE_AUDIO, + CODEC_ID_MP3ADU, + sizeof(MPADecodeContext), + decode_init, + NULL, + NULL, + decode_frame_adu, + CODEC_CAP_PARSE_ONLY, +}; + +AVCodec mp3on4_decoder = +{ + "mp3on4", + CODEC_TYPE_AUDIO, + CODEC_ID_MP3ON4, + sizeof(MP3On4DecodeContext), + decode_init_mp3on4, + NULL, + decode_close_mp3on4, + decode_frame_mp3on4, + 0 };