Mercurial > libavcodec.hg
view mpegaudiodec.c @ 12530:63edd10ad4bc libavcodec tip
Try to fix crashes introduced by r25218
r25218 made assumptions about the existence of past reference frames that
weren't necessarily true.
| author | darkshikari |
|---|---|
| date | Tue, 28 Sep 2010 09:06:22 +0000 |
| parents | f31ba4ca1397 |
| children |
line wrap: on
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/* * MPEG Audio decoder * Copyright (c) 2001, 2002 Fabrice Bellard * * This file is part of FFmpeg. * * FFmpeg 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 2.1 of the License, or (at your option) any later version. * * FFmpeg 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 * 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 */ /** * @file * MPEG Audio decoder. */ #include "avcodec.h" #include "get_bits.h" #include "dsputil.h" /* * TODO: * - in low precision mode, use more 16 bit multiplies in synth filter * - test lsf / mpeg25 extensively. */ #include "mpegaudio.h" #include "mpegaudiodecheader.h" #include "mathops.h" #if CONFIG_FLOAT # define SHR(a,b) ((a)*(1.0f/(1<<(b)))) # define compute_antialias compute_antialias_float # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5)) # define FIXR(x) ((float)(x)) # define FIXHR(x) ((float)(x)) # define MULH3(x, y, s) ((s)*(y)*(x)) # define MULLx(x, y, s) ((y)*(x)) # define RENAME(a) a ## _float #else # define SHR(a,b) ((a)>>(b)) # define compute_antialias compute_antialias_integer /* WARNING: only correct for posititive numbers */ # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5)) # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5)) # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5)) # define MULH3(x, y, s) MULH((s)*(x), y) # define MULLx(x, y, s) MULL(x,y,s) # define RENAME(a) a #endif /****************/ #define HEADER_SIZE 4 #include "mpegaudiodata.h" #include "mpegaudiodectab.h" #if CONFIG_FLOAT # include "fft.h" #else # include "dct32.c" #endif static void compute_antialias(MPADecodeContext *s, GranuleDef *g); static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window, int *dither_state, OUT_INT *samples, int incr); /* vlc structure for decoding layer 3 huffman tables */ static VLC huff_vlc[16]; static VLC_TYPE huff_vlc_tables[ 0+128+128+128+130+128+154+166+ 142+204+190+170+542+460+662+414 ][2]; static const int huff_vlc_tables_sizes[16] = { 0, 128, 128, 128, 130, 128, 154, 166, 142, 204, 190, 170, 542, 460, 662, 414 }; static VLC huff_quad_vlc[2]; static VLC_TYPE huff_quad_vlc_tables[128+16][2]; static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 }; /* computed from band_size_long */ static uint16_t band_index_long[9][23]; #include "mpegaudio_tablegen.h" /* intensity stereo coef table */ static INTFLOAT is_table[2][16]; static INTFLOAT is_table_lsf[2][2][16]; static int32_t csa_table[8][4]; static float csa_table_float[8][4]; static INTFLOAT mdct_win[8][36]; static int16_t division_tab3[1<<6 ]; static int16_t division_tab5[1<<8 ]; static int16_t division_tab9[1<<11]; static int16_t * const division_tabs[4] = { division_tab3, division_tab5, NULL, division_tab9 }; /* 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_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(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 */ }; DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256]; /** * Convert region offsets to region sizes and truncate * size to big_values. */ static void ff_region_offset2size(GranuleDef *g){ int i, k, j=0; g->region_size[2] = (576 / 2); for(i=0;i<3;i++) { k = FFMIN(g->region_size[i], g->big_values); g->region_size[i] = k - j; j = k; } } static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g){ 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); } static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2){ int l; g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1; /* should not overflow */ l = FFMIN(ra1 + ra2 + 2, 22); g->region_size[1] = band_index_long[s->sample_rate_index][l] >> 1; } static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g){ 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; } } /* 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 av_cold void int_pow_init(void) { int i, a; a = POW_FIX(1.0); for(i=0;i<DEV_ORDER;i++) { a = POW_MULL(a, POW_FIX(4.0 / 3.0) - i * POW_FIX(1.0)) / (i + 1); dev_4_3_coefs[i] = a; } } #if 0 /* unused, remove? */ /* return the mantissa and the binary exponent */ static int int_pow(int i, int *exp_ptr) { int e, er, eq, j; int a, a1; /* renormalize */ a = i; e = POW_FRAC_BITS; while (a < (1 << (POW_FRAC_BITS - 1))) { a = a << 1; e--; } a -= (1 << POW_FRAC_BITS); a1 = 0; for(j = DEV_ORDER - 1; j >= 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 av_cold int decode_init(AVCodecContext * avctx) { MPADecodeContext *s = avctx->priv_data; static int init=0; int i, j, k; s->avctx = avctx; s->apply_window_mp3 = apply_window_mp3_c; #if HAVE_MMX && CONFIG_FLOAT ff_mpegaudiodec_init_mmx(s); #endif #if CONFIG_FLOAT ff_dct_init(&s->dct, 5, DCT_II); #endif if (HAVE_ALTIVEC && CONFIG_FLOAT) ff_mpegaudiodec_init_altivec(s); avctx->sample_fmt= OUT_FMT; s->error_recognition= avctx->error_recognition; if (!init && !avctx->parse_only) { int offset; /* 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_C(1) << n) * FRAC_ONE) / ((1 << n) - 1); scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS); scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS); scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS); dprintf(avctx, "%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]); } RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window)); /* huffman decode tables */ offset = 0; for(i=1;i<16;i++) { const HuffTable *h = &mpa_huff_tables[i]; int xsize, x, y; uint8_t tmp_bits [512]; uint16_t tmp_codes[512]; memset(tmp_bits , 0, sizeof(tmp_bits )); memset(tmp_codes, 0, sizeof(tmp_codes)); xsize = h->xsize; j = 0; for(x=0;x<xsize;x++) { for(y=0;y<xsize;y++){ tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ]; tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++]; } } /* XXX: fail test */ huff_vlc[i].table = huff_vlc_tables+offset; huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i]; init_vlc(&huff_vlc[i], 7, 512, tmp_bits, 1, 1, tmp_codes, 2, 2, INIT_VLC_USE_NEW_STATIC); offset += huff_vlc_tables_sizes[i]; } assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables)); offset = 0; for(i=0;i<2;i++) { huff_quad_vlc[i].table = huff_quad_vlc_tables+offset; huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i]; init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16, mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC); offset += huff_quad_vlc_tables_sizes[i]; } assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables)); 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 */ int_pow_init(); mpegaudio_tableinit(); for (i = 0; i < 4; i++) if (ff_mpa_quant_bits[i] < 0) for (j = 0; j < (1<<(-ff_mpa_quant_bits[i]+1)); j++) { int val1, val2, val3, steps; int val = j; steps = ff_mpa_quant_steps[i]; val1 = val % steps; val /= steps; val2 = val % steps; val3 = val / steps; division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8); } for(i=0;i<7;i++) { float f; INTFLOAT 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; } /* invalid values */ 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 = 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(avctx, "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; } /* 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))); } } /* 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]; } } init = 1; } if (avctx->codec_id == CODEC_ID_MP3ADU) s->adu_mode = 1; return 0; } #if CONFIG_FLOAT static inline float round_sample(float *sum) { float sum1=*sum; *sum = 0; return sum1; } /* signed 16x16 -> 32 multiply add accumulate */ #define MACS(rt, ra, rb) rt+=(ra)*(rb) /* signed 16x16 -> 32 multiply */ #define MULS(ra, rb) ((ra)*(rb)) #define MLSS(rt, ra, rb) rt-=(ra)*(rb) #elif FRAC_BITS <= 15 static inline int round_sample(int *sum) { int sum1; sum1 = (*sum) >> OUT_SHIFT; *sum &= (1<<OUT_SHIFT)-1; return av_clip(sum1, OUT_MIN, OUT_MAX); } /* signed 16x16 -> 32 multiply add accumulate */ #define MACS(rt, ra, rb) MAC16(rt, ra, rb) /* signed 16x16 -> 32 multiply */ #define MULS(ra, rb) MUL16(ra, rb) #define MLSS(rt, ra, rb) MLS16(rt, ra, rb) #else static inline int round_sample(int64_t *sum) { int sum1; sum1 = (int)((*sum) >> OUT_SHIFT); *sum &= (1<<OUT_SHIFT)-1; return av_clip(sum1, OUT_MIN, OUT_MAX); } # define MULS(ra, rb) MUL64(ra, rb) # define MACS(rt, ra, rb) MAC64(rt, ra, rb) # define MLSS(rt, ra, rb) MLS64(rt, ra, rb) #endif #define SUM8(op, sum, w, p) \ { \ op(sum, (w)[0 * 64], (p)[0 * 64]); \ op(sum, (w)[1 * 64], (p)[1 * 64]); \ op(sum, (w)[2 * 64], (p)[2 * 64]); \ op(sum, (w)[3 * 64], (p)[3 * 64]); \ op(sum, (w)[4 * 64], (p)[4 * 64]); \ op(sum, (w)[5 * 64], (p)[5 * 64]); \ op(sum, (w)[6 * 64], (p)[6 * 64]); \ op(sum, (w)[7 * 64], (p)[7 * 64]); \ } #define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \ { \ INTFLOAT tmp;\ tmp = p[0 * 64];\ op1(sum1, (w1)[0 * 64], tmp);\ op2(sum2, (w2)[0 * 64], tmp);\ tmp = p[1 * 64];\ op1(sum1, (w1)[1 * 64], tmp);\ op2(sum2, (w2)[1 * 64], tmp);\ tmp = p[2 * 64];\ op1(sum1, (w1)[2 * 64], tmp);\ op2(sum2, (w2)[2 * 64], tmp);\ tmp = p[3 * 64];\ op1(sum1, (w1)[3 * 64], tmp);\ op2(sum2, (w2)[3 * 64], tmp);\ tmp = p[4 * 64];\ op1(sum1, (w1)[4 * 64], tmp);\ op2(sum2, (w2)[4 * 64], tmp);\ tmp = p[5 * 64];\ op1(sum1, (w1)[5 * 64], tmp);\ op2(sum2, (w2)[5 * 64], tmp);\ tmp = p[6 * 64];\ op1(sum1, (w1)[6 * 64], tmp);\ op2(sum2, (w2)[6 * 64], tmp);\ tmp = p[7 * 64];\ op1(sum1, (w1)[7 * 64], tmp);\ op2(sum2, (w2)[7 * 64], tmp);\ } void av_cold RENAME(ff_mpa_synth_init)(MPA_INT *window) { int i, j; /* max = 18760, max sum over all 16 coefs : 44736 */ for(i=0;i<257;i++) { INTFLOAT v; v = ff_mpa_enwindow[i]; #if CONFIG_FLOAT v *= 1.0 / (1LL<<(16 + FRAC_BITS)); #elif 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; } // Needed for avoiding shuffles in ASM implementations for(i=0; i < 8; i++) for(j=0; j < 16; j++) window[512+16*i+j] = window[64*i+32-j]; for(i=0; i < 8; i++) for(j=0; j < 16; j++) window[512+128+16*i+j] = window[64*i+48-j]; } static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window, int *dither_state, OUT_INT *samples, int incr) { register const MPA_INT *w, *w2, *p; int j; OUT_INT *samples2; #if CONFIG_FLOAT float sum, sum2; #elif FRAC_BITS <= 15 int sum, sum2; #else int64_t sum, sum2; #endif /* copy to avoid wrap */ memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf)); samples2 = samples + 31 * incr; w = window; w2 = window + 31; sum = *dither_state; p = synth_buf + 16; SUM8(MACS, sum, w, p); p = synth_buf + 48; SUM8(MLSS, 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, MACS, sum2, MLSS, w, w2, p); p = synth_buf + 48 - j; SUM8P2(sum, MLSS, sum2, MLSS, 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(MLSS, sum, w + 32, p); *samples = round_sample(&sum); *dither_state= sum; } /* 32 sub band synthesis filter. Input: 32 sub band samples, Output: 32 samples. */ /* XXX: optimize by avoiding ring buffer usage */ #if !CONFIG_FLOAT 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, INTFLOAT sb_samples[SBLIMIT]) { register MPA_INT *synth_buf; int offset; #if FRAC_BITS <= 15 int32_t tmp[32]; int j; #endif offset = *synth_buf_offset; synth_buf = synth_buf_ptr + offset; #if FRAC_BITS <= 15 dct32(tmp, sb_samples); for(j=0;j<32;j++) { /* NOTE: can cause a loss in precision if very high amplitude sound */ synth_buf[j] = av_clip_int16(tmp[j]); } #else dct32(synth_buf, sb_samples); #endif apply_window_mp3_c(synth_buf, window, dither_state, samples, incr); offset = (offset - 32) & 511; *synth_buf_offset = offset; } #endif #define C3 FIXHR(0.86602540378443864676/2) /* 0.5 / cos(pi*(2*i+1)/36) */ static const INTFLOAT 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), }; /* 0.5 / cos(pi*(2*i+1)/36) */ static const INTFLOAT 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), }; /* 12 points IMDCT. We compute it "by hand" by factorizing obvious cases. */ static void imdct12(INTFLOAT *out, INTFLOAT *in) { INTFLOAT 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= MULH3(in2, C3, 2); in3= MULH3(in3, C3, 4); t1 = in0 - in4; t2 = MULH3(in1 - in5, icos36h[4], 2); out[ 7]= out[10]= t1 + t2; out[ 1]= out[ 4]= t1 - t2; in0 += SHR(in4, 1); in4 = in0 + in2; in5 += 2*in1; in1 = MULH3(in5 + in3, icos36h[1], 1); out[ 8]= out[ 9]= in4 + in1; out[ 2]= out[ 3]= in4 - in1; in0 -= in2; in5 = MULH3(in5 - in3, icos36h[7], 2); out[ 0]= out[ 5]= in0 - in5; out[ 6]= out[11]= in0 + in5; } /* 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(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in, INTFLOAT *win) { int i, j; INTFLOAT t0, t1, t2, t3, s0, s1, s2, s3; INTFLOAT 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] + SHR(in1[2*6],1); t1 = in1[2*0] - in1[2*6]; tmp1[ 6] = t1 - SHR(t2,1); tmp1[16] = t1 + t2; t0 = MULH3(in1[2*2] + in1[2*4] , C2, 2); t1 = MULH3(in1[2*4] - in1[2*8] , -2*C8, 1); t2 = MULH3(in1[2*2] + in1[2*8] , -C4, 2); tmp1[10] = t3 - t0 - t2; tmp1[ 2] = t3 + t0 + t1; tmp1[14] = t3 + t2 - t1; tmp1[ 4] = MULH3(in1[2*5] + in1[2*7] - in1[2*1], -C3, 2); t2 = MULH3(in1[2*1] + in1[2*5], C1, 2); t3 = MULH3(in1[2*5] - in1[2*7], -2*C7, 1); t0 = MULH3(in1[2*3], C3, 2); t1 = MULH3(in1[2*1] + in1[2*7], -C5, 2); 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 = MULH3(t3 + t2, icos36h[j], 2); s3 = MULLx(t3 - t2, icos36[8 - j], FRAC_BITS); t0 = s0 + s1; t1 = s0 - s1; out[(9 + j)*SBLIMIT] = MULH3(t1, win[9 + j], 1) + buf[9 + j]; out[(8 - j)*SBLIMIT] = MULH3(t1, win[8 - j], 1) + buf[8 - j]; buf[9 + j] = MULH3(t0, win[18 + 9 + j], 1); buf[8 - j] = MULH3(t0, win[18 + 8 - j], 1); t0 = s2 + s3; t1 = s2 - s3; out[(9 + 8 - j)*SBLIMIT] = MULH3(t1, win[9 + 8 - j], 1) + buf[9 + 8 - j]; out[( j)*SBLIMIT] = MULH3(t1, win[ j], 1) + buf[ j]; buf[9 + 8 - j] = MULH3(t0, win[18 + 9 + 8 - j], 1); buf[ + j] = MULH3(t0, win[18 + j], 1); i += 4; } s0 = tmp[16]; s1 = MULH3(tmp[17], icos36h[4], 2); t0 = s0 + s1; t1 = s0 - s1; out[(9 + 4)*SBLIMIT] = MULH3(t1, win[9 + 4], 1) + buf[9 + 4]; out[(8 - 4)*SBLIMIT] = MULH3(t1, win[8 - 4], 1) + buf[8 - 4]; buf[9 + 4] = MULH3(t0, win[18 + 9 + 4], 1); buf[8 - 4] = MULH3(t0, win[18 + 8 - 4], 1); } /* 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;i<bound;i++) { for(ch=0;ch<s->nb_channels;ch++) { allocation[ch][i] = get_bits(&s->gb, 4); } } for(i=bound;i<SBLIMIT;i++) { allocation[0][i] = get_bits(&s->gb, 4); } /* scale factors */ for(i=0;i<bound;i++) { for(ch=0;ch<s->nb_channels;ch++) { if (allocation[ch][i]) scale_factors[ch][i] = get_bits(&s->gb, 6); } } for(i=bound;i<SBLIMIT;i++) { if (allocation[0][i]) { scale_factors[0][i] = get_bits(&s->gb, 6); scale_factors[1][i] = get_bits(&s->gb, 6); } } /* compute samples */ for(j=0;j<12;j++) { for(i=0;i<bound;i++) { for(ch=0;ch<s->nb_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;i<SBLIMIT;i++) { n = allocation[0][i]; if (n) { mant = get_bits(&s->gb, 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; } 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 = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels, s->sample_rate, s->lsf); sblimit = ff_mpa_sblimit_table[table]; alloc_table = ff_mpa_alloc_tables[table]; if (s->mode == MPA_JSTEREO) bound = (s->mode_ext + 1) * 4; else bound = sblimit; dprintf(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit); /* sanity check */ if( bound > sblimit ) bound = sblimit; /* parse bit allocation */ j = 0; for(i=0;i<bound;i++) { bit_alloc_bits = alloc_table[j]; for(ch=0;ch<s->nb_channels;ch++) { bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits); } j += 1 << bit_alloc_bits; } for(i=bound;i<sblimit;i++) { bit_alloc_bits = alloc_table[j]; v = get_bits(&s->gb, bit_alloc_bits); bit_alloc[0][i] = v; bit_alloc[1][i] = v; j += 1 << bit_alloc_bits; } /* scale codes */ for(i=0;i<sblimit;i++) { for(ch=0;ch<s->nb_channels;ch++) { if (bit_alloc[ch][i]) scale_code[ch][i] = get_bits(&s->gb, 2); } } /* scale factors */ for(i=0;i<sblimit;i++) { for(ch=0;ch<s->nb_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; } } } } /* samples */ for(k=0;k<3;k++) { for(l=0;l<12;l+=3) { j = 0; for(i=0;i<bound;i++) { bit_alloc_bits = alloc_table[j]; for(ch=0;ch<s->nb_channels;ch++) { b = bit_alloc[ch][i]; if (b) { scale = scale_factors[ch][i][k]; qindex = alloc_table[j+b]; bits = ff_mpa_quant_bits[qindex]; if (bits < 0) { int v2; /* 3 values at the same time */ v = get_bits(&s->gb, -bits); v2 = division_tabs[qindex][v]; steps = ff_mpa_quant_steps[qindex]; s->sb_samples[ch][k * 12 + l + 0][i] = l2_unscale_group(steps, v2 & 15, scale); s->sb_samples[ch][k * 12 + l + 1][i] = l2_unscale_group(steps, (v2 >> 4) & 15, scale); s->sb_samples[ch][k * 12 + l + 2][i] = l2_unscale_group(steps, v2 >> 8 , 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;i<sblimit;i++) { bit_alloc_bits = alloc_table[j]; b = bit_alloc[0][i]; if (b) { int mant, scale0, scale1; scale0 = scale_factors[0][i][k]; scale1 = scale_factors[1][i][k]; qindex = alloc_table[j+b]; bits = ff_mpa_quant_bits[qindex]; if (bits < 0) { /* 3 values at the same time */ v = get_bits(&s->gb, -bits); steps = ff_mpa_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;i<SBLIMIT;i++) { for(ch=0;ch<s->nb_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; } #define SPLIT(dst,sf,n)\ if(n==3){\ int m= (sf*171)>>9;\ dst= sf - 3*m;\ sf=m;\ }else if(n==4){\ dst= sf&3;\ sf>>=2;\ }else if(n==5){\ int m= (sf*205)>>10;\ dst= sf - 5*m;\ sf=m;\ }else if(n==6){\ int m= (sf*171)>>10;\ dst= sf - 6*m;\ sf=m;\ }else{\ dst=0;\ } static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2, int n3) { SPLIT(slen[3], sf, n3) SPLIT(slen[2], sf, n2) SPLIT(slen[1], 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;i<g->long_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; } } } } /* 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 void switch_buffer(MPADecodeContext *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; assert((get_bits_count(&s->gb) & 7) == 0); 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); } } /* Following is a optimized code for INTFLOAT v = *src if(get_bits1(&s->gb)) v = -v; *dst = v; */ #if CONFIG_FLOAT #define READ_FLIP_SIGN(dst,src)\ v = AV_RN32A(src) ^ (get_bits1(&s->gb)<<31);\ AV_WN32A(dst, v); #else #define READ_FLIP_SIGN(dst,src)\ v= -get_bits1(&s->gb);\ *(dst) = (*(src) ^ v) - v; #endif static int huffman_decode(MPADecodeContext *s, GranuleDef *g, int16_t *exponents, int end_pos2) { int s_index; int i; int last_pos, bits_left; VLC *vlc; int end_pos= FFMIN(end_pos2, s->gb.size_in_bits); /* 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 = mpa_huff_data[k][0]; linbits = mpa_huff_data[k][1]; vlc = &huff_vlc[l]; if(!l){ 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; int v; int pos= get_bits_count(&s->gb); if (pos >= end_pos){ // av_log(NULL, AV_LOG_ERROR, "pos: %d %d %d %d\n", pos, end_pos, end_pos2, s_index); switch_buffer(s, &pos, &end_pos, &end_pos2); // av_log(NULL, AV_LOG_ERROR, "new pos: %d %d\n", pos, end_pos); 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]; dprintf(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n", i, g->region_size[i] - j, x, y, exponent); if(y&16){ x = y >> 5; y = y & 0x0f; if (x < 15){ READ_FLIP_SIGN(g->sb_hybrid+s_index, RENAME(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){ READ_FLIP_SIGN(g->sb_hybrid+s_index+1, RENAME(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){ READ_FLIP_SIGN(g->sb_hybrid+s_index+!!y, RENAME(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); av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos); if(s->error_recognition >= FF_ER_COMPLIANT) s_index=0; break; } // av_log(NULL, AV_LOG_ERROR, "pos2: %d %d %d %d\n", pos, end_pos, end_pos2, s_index); switch_buffer(s, &pos, &end_pos, &end_pos2); // av_log(NULL, AV_LOG_ERROR, "new pos2: %d %d %d\n", pos, end_pos, s_index); if(pos >= end_pos) break; } last_pos= pos; code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1); dprintf(s->avctx, "t=%d code=%d\n", g->count1table_select, code); 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){ static const 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]; READ_FLIP_SIGN(g->sb_hybrid+pos, RENAME(exp_table)+exponents[pos]) } s_index+=4; } /* skip extension bits */ bits_left = end_pos2 - get_bits_count(&s->gb); //av_log(NULL, AV_LOG_ERROR, "left:%d buf:%p\n", bits_left, s->in_gb.buffer); if (bits_left < 0 && s->error_recognition >= FF_ER_COMPLIANT) { av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left); s_index=0; }else if(bits_left > 0 && s->error_recognition >= FF_ER_AGGRESSIVE){ av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left); s_index=0; } memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*(576 - s_index)); skip_bits_long(&s->gb, bits_left); i= get_bits_count(&s->gb); switch_buffer(s, &i, &end_pos, &end_pos2); 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, len; INTFLOAT *ptr, *dst, *ptr1; INTFLOAT 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; memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1)); } } #define ISQRT2 FIXR(0.70710678118654752440) static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1) { int i, j, k, l; int sf_max, sf, len, non_zero_found; INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2; 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;j<len;j++) { if (tab1[j] != 0) { non_zero_found_short[l] = 1; goto found1; } } sf = g1->scale_factors[k + l]; if (sf >= sf_max) goto found1; v1 = is_tab[0][sf]; v2 = is_tab[1][sf]; for(j=0;j<len;j++) { tmp0 = tab0[j]; tab0[j] = MULLx(tmp0, v1, FRAC_BITS); tab1[j] = MULLx(tmp0, v2, FRAC_BITS); } } else { found1: if (s->mode_ext & MODE_EXT_MS_STEREO) { /* lower part of the spectrum : do ms stereo if enabled */ for(j=0;j<len;j++) { tmp0 = tab0[j]; tmp1 = tab1[j]; tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS); tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS); } } } } } non_zero_found = non_zero_found_short[0] | non_zero_found_short[1] | non_zero_found_short[2]; for(i = g1->long_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;j<len;j++) { if (tab1[j] != 0) { non_zero_found = 1; goto found2; } } /* for last band, use previous scale factor */ k = (i == 21) ? 20 : i; sf = g1->scale_factors[k]; if (sf >= sf_max) goto found2; v1 = is_tab[0][sf]; v2 = is_tab[1][sf]; for(j=0;j<len;j++) { tmp0 = tab0[j]; tab0[j] = MULLx(tmp0, v1, FRAC_BITS); tab1[j] = MULLx(tmp0, v2, FRAC_BITS); } } else { found2: if (s->mode_ext & MODE_EXT_MS_STEREO) { /* lower part of the spectrum : do ms stereo if enabled */ for(j=0;j<len;j++) { tmp0 = tab0[j]; tmp1 = tab1[j]; tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS); tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS); } } } } } else if (s->mode_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; } } } #if !CONFIG_FLOAT 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; } } #endif static void compute_imdct(MPADecodeContext *s, GranuleDef *g, INTFLOAT *sb_samples, INTFLOAT *mdct_buf) { INTFLOAT *win, *win1, *out_ptr, *ptr, *buf, *ptr1; INTFLOAT out2[12]; int i, j, mdct_long_end, sblimit; /* find last non zero block */ ptr = g->sb_hybrid + 576; ptr1 = g->sb_hybrid + 2 * 18; while (ptr >= ptr1) { int32_t *p; ptr -= 6; p= (int32_t*)ptr; if(p[0] | p[1] | p[2] | p[3] | p[4] | p[5]) 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;j<mdct_long_end;j++) { /* apply window & overlap with previous buffer */ out_ptr = sb_samples + j; /* select window */ if (g->switch_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<sblimit;j++) { /* select frequency inversion */ win = mdct_win[2] + ((4 * 36) & -(j & 1)); out_ptr = sb_samples + j; for(i=0; i<6; i++){ *out_ptr = buf[i]; out_ptr += SBLIMIT; } imdct12(out2, ptr + 0); for(i=0;i<6;i++) { *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*1]; buf[i + 6*2] = MULH3(out2[i + 6], win[i + 6], 1); out_ptr += SBLIMIT; } imdct12(out2, ptr + 1); for(i=0;i<6;i++) { *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*2]; buf[i + 6*0] = MULH3(out2[i + 6], win[i + 6], 1); out_ptr += SBLIMIT; } imdct12(out2, ptr + 2); for(i=0;i<6;i++) { buf[i + 6*0] = MULH3(out2[i ], win[i ], 1) + buf[i + 6*0]; buf[i + 6*1] = MULH3(out2[i + 6], win[i + 6], 1); buf[i + 6*2] = 0; } ptr += 18; buf += 18; } /* zero bands */ for(j=sblimit;j<SBLIMIT;j++) { /* overlap */ out_ptr = sb_samples + j; for(i=0;i<18;i++) { *out_ptr = buf[i]; buf[i] = 0; out_ptr += SBLIMIT; } buf += 18; } } /* main layer3 decoding function */ static int mp_decode_layer3(MPADecodeContext *s) { int nb_granules, main_data_begin, private_bits; int gr, ch, blocksplit_flag, i, j, k, n, bits_pos; GranuleDef *g; int16_t exponents[576]; //FIXME try INTFLOAT /* read side info */ 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;ch<s->nb_channels;ch++) { s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */ s->granules[ch][1].scfsi = get_bits(&s->gb, 4); } } for(gr=0;gr<nb_granules;gr++) { for(ch=0;ch<s->nb_channels;ch++) { dprintf(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch); g = &s->granules[ch][gr]; g->part2_3_length = get_bits(&s->gb, 12); g->big_values = get_bits(&s->gb, 9); if(g->big_values > 288){ av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n"); return -1; } 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_bits1(&s->gb); if (blocksplit_flag) { g->block_type = get_bits(&s->gb, 2); if (g->block_type == 0){ av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n"); return -1; } g->switch_point = get_bits1(&s->gb); 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); ff_init_short_region(s, g); } else { int region_address1, region_address2; 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(s->avctx, "region1=%d region2=%d\n", region_address1, region_address2); ff_init_long_region(s, g, region_address1, region_address2); } ff_region_offset2size(g); ff_compute_band_indexes(s, g); g->preflag = 0; if (!s->lsf) g->preflag = get_bits1(&s->gb); g->scalefac_scale = get_bits1(&s->gb); g->count1table_select = get_bits1(&s->gb); dprintf(s->avctx, "block_type=%d switch_point=%d\n", g->block_type, g->switch_point); } } if (!s->adu_mode) { const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3); assert((get_bits_count(&s->gb) & 7) == 0); /* now we get bits from the main_data_begin offset */ dprintf(s->avctx, "seekback: %d\n", main_data_begin); //av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size); 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*8); skip_bits_long(&s->gb, 8*(s->last_buf_size - main_data_begin)); } for(gr=0;gr<nb_granules;gr++) { for(ch=0;ch<s->nb_channels;ch++) { g = &s->granules[ch][gr]; if(get_bits_count(&s->gb)<0){ av_log(s->avctx, AV_LOG_DEBUG, "mdb:%d, lastbuf:%d skipping granule %d\n", main_data_begin, s->last_buf_size, gr); skip_bits_long(&s->gb, g->part2_3_length); memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid)); if(get_bits_count(&s->gb) >= s->gb.size_in_bits && s->in_gb.buffer){ skip_bits_long(&s->in_gb, get_bits_count(&s->gb) - s->gb.size_in_bits); s->gb= s->in_gb; s->in_gb.buffer=NULL; } continue; } 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(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2); if (g->block_type == 2) { n = g->switch_point ? 17 : 18; j = 0; if(slen1){ for(i=0;i<n;i++) g->scale_factors[j++] = get_bits(&s->gb, slen1); }else{ for(i=0;i<n;i++) g->scale_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 = s->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;i<n;i++) g->scale_factors[j++] = get_bits(&s->gb, slen); }else{ for(i=0;i<n;i++) g->scale_factors[j++] = 0; } } else { /* simply copy from last granule */ for(i=0;i<n;i++) { g->scale_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;i<n;i++) g->scale_factors[j++] = get_bits(&s->gb, sl); }else{ for(i=0;i<n;i++) g->scale_factors[j++] = 0; } } /* XXX: should compute exact size */ for(;j<40;j++) g->scale_factors[j] = 0; } exponents_from_scale_factors(s, g, exponents); /* read Huffman coded residue */ huffman_decode(s, g, exponents, bits_pos + g->part2_3_length); } /* ch */ if (s->nb_channels == 2) compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]); for(ch=0;ch<s->nb_channels;ch++) { g = &s->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 */ if(get_bits_count(&s->gb)<0) skip_bits_long(&s->gb, -get_bits_count(&s->gb)); return nb_granules * 18; } static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples, const uint8_t *buf, int buf_size) { int i, nb_frames, ch; OUT_INT *samples_ptr; init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE)*8); /* skip error protection field */ if (s->error_protection) skip_bits(&s->gb, 16); dprintf(s->avctx, "frame %d:\n", s->frame_count); switch(s->layer) { case 1: s->avctx->frame_size = 384; nb_frames = mp_decode_layer1(s); break; case 2: s->avctx->frame_size = 1152; nb_frames = mp_decode_layer2(s); break; case 3: s->avctx->frame_size = s->lsf ? 576 : 1152; default: nb_frames = mp_decode_layer3(s); s->last_buf_size=0; if(s->in_gb.buffer){ align_get_bits(&s->gb); i= get_bits_left(&s->gb)>>3; if(i >= 0 && i <= BACKSTEP_SIZE){ memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i); s->last_buf_size=i; }else av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i); s->gb= s->in_gb; s->in_gb.buffer= NULL; } align_get_bits(&s->gb); assert((get_bits_count(&s->gb) & 7) == 0); i= get_bits_left(&s->gb)>>3; if(i<0 || i > BACKSTEP_SIZE || nb_frames<0){ if(i<0) av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i); i= FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE); } assert(i <= buf_size - HEADER_SIZE && i>= 0); memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i); s->last_buf_size += i; break; } /* apply the synthesis filter */ for(ch=0;ch<s->nb_channels;ch++) { samples_ptr = samples + ch; for(i=0;i<nb_frames;i++) { RENAME(ff_mpa_synth_filter)( #if CONFIG_FLOAT s, #endif s->synth_buf[ch], &(s->synth_buf_offset[ch]), RENAME(ff_mpa_synth_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(OUT_INT) * s->nb_channels; } static int decode_frame(AVCodecContext * avctx, void *data, int *data_size, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; MPADecodeContext *s = avctx->priv_data; uint32_t header; int out_size; OUT_INT *out_samples = data; if(buf_size < HEADER_SIZE) return -1; header = AV_RB32(buf); if(ff_mpa_check_header(header) < 0){ av_log(avctx, AV_LOG_ERROR, "Header missing\n"); return -1; } if (ff_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) { /* free format: prepare to compute frame size */ s->frame_size = -1; return -1; } /* update codec info */ avctx->channels = s->nb_channels; if (!avctx->bit_rate) avctx->bit_rate = s->bit_rate; avctx->sub_id = s->layer; if(*data_size < 1152*avctx->channels*sizeof(OUT_INT)) return -1; *data_size = 0; if(s->frame_size<=0 || s->frame_size > buf_size){ av_log(avctx, AV_LOG_ERROR, "incomplete frame\n"); return -1; }else if(s->frame_size < buf_size){ av_log(avctx, AV_LOG_ERROR, "incorrect frame size\n"); buf_size= s->frame_size; } out_size = mp_decode_frame(s, out_samples, buf, buf_size); if(out_size>=0){ *data_size = out_size; avctx->sample_rate = s->sample_rate; //FIXME maybe move the other codec info stuff from above here too }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 s->frame_size = 0; return buf_size; } static void flush(AVCodecContext *avctx){ MPADecodeContext *s = avctx->priv_data; memset(s->synth_buf, 0, sizeof(s->synth_buf)); s->last_buf_size= 0; } #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER static int decode_frame_adu(AVCodecContext * avctx, void *data, int *data_size, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; 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; // Get header and restore sync word header = AV_RB32(buf) | 0xffe00000; if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame *data_size = 0; return buf_size; } ff_mpegaudio_decode_header((MPADecodeHeader *)s, header); /* update codec info */ avctx->sample_rate = s->sample_rate; avctx->channels = s->nb_channels; if (!avctx->bit_rate) avctx->bit_rate = s->bit_rate; avctx->sub_id = s->layer; s->frame_size = len; if (avctx->parse_only) { out_size = buf_size; } else { out_size = mp_decode_frame(s, out_samples, buf, buf_size); } *data_size = out_size; return buf_size; } #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */ #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER /** * Context for MP3On4 decoder */ typedef struct MP3On4DecodeContext { int frames; ///< number of mp3 frames per block (number of mp3 decoder instances) int syncword; ///< syncword patch const uint8_t *coff; ///< channels offsets in output buffer MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance } MP3On4DecodeContext; #include "mpeg4audio.h" /* Next 3 arrays are indexed by channel config number (passed via codecdata) */ static const uint8_t mp3Frames[8] = {0,1,1,2,3,3,4,5}; /* number of mp3 decoder instances */ /* offsets into output buffer, assume output order is FL FR BL BR C LFE */ static const uint8_t chan_offset[8][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 }; static int decode_init_mp3on4(AVCodecContext * avctx) { MP3On4DecodeContext *s = avctx->priv_data; MPEG4AudioConfig cfg; 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; } ff_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size); if (!cfg.chan_config || cfg.chan_config > 7) { av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n"); return -1; } s->frames = mp3Frames[cfg.chan_config]; s->coff = chan_offset[cfg.chan_config]; avctx->channels = ff_mpeg4audio_channels[cfg.chan_config]; if (cfg.sample_rate < 16000) s->syncword = 0xffe00000; else s->syncword = 0xfff00000; /* 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 initialized 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]->adu_mode = 1; s->mp3decctx[i]->avctx = avctx; } return 0; } static av_cold 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, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; MP3On4DecodeContext *s = avctx->priv_data; MPADecodeContext *m; int fsize, len = buf_size, 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 fr, j, n; if(*data_size < MPA_FRAME_SIZE * MPA_MAX_CHANNELS * s->frames * sizeof(OUT_INT)) return -1; *data_size = 0; // Discard too short frames if (buf_size < HEADER_SIZE) return -1; // If only one decoder interleave is not needed outptr = s->frames == 1 ? out_samples : decoded_buf; avctx->bit_rate = 0; for (fr = 0; fr < s->frames; fr++) { fsize = AV_RB16(buf) >> 4; fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE); m = s->mp3decctx[fr]; assert (m != NULL); header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header if (ff_mpa_check_header(header) < 0) // Bad header, discard block break; ff_mpegaudio_decode_header((MPADecodeHeader *)m, header); out_size += mp_decode_frame(m, outptr, buf, fsize); buf += fsize; len -= fsize; if(s->frames > 1) { n = m->avctx->frame_size*m->nb_channels; /* interleave output data */ bp = out_samples + s->coff[fr]; if(m->nb_channels == 1) { for(j = 0; j < n; j++) { *bp = decoded_buf[j]; bp += avctx->channels; } } else { for(j = 0; j < n; j++) { bp[0] = decoded_buf[j++]; bp[1] = decoded_buf[j]; bp += avctx->channels; } } } avctx->bit_rate += m->bit_rate; } /* update codec info */ avctx->sample_rate = s->mp3decctx[0]->sample_rate; *data_size = out_size; return buf_size; } #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */ #if !CONFIG_FLOAT #if CONFIG_MP1_DECODER AVCodec mp1_decoder = { "mp1", AVMEDIA_TYPE_AUDIO, CODEC_ID_MP1, sizeof(MPADecodeContext), decode_init, NULL, NULL, decode_frame, CODEC_CAP_PARSE_ONLY, .flush= flush, .long_name= NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"), }; #endif #if CONFIG_MP2_DECODER AVCodec mp2_decoder = { "mp2", AVMEDIA_TYPE_AUDIO, CODEC_ID_MP2, sizeof(MPADecodeContext), decode_init, NULL, NULL, decode_frame, CODEC_CAP_PARSE_ONLY, .flush= flush, .long_name= NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), }; #endif #if CONFIG_MP3_DECODER AVCodec mp3_decoder = { "mp3", AVMEDIA_TYPE_AUDIO, CODEC_ID_MP3, sizeof(MPADecodeContext), decode_init, NULL, NULL, decode_frame, CODEC_CAP_PARSE_ONLY, .flush= flush, .long_name= NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"), }; #endif #if CONFIG_MP3ADU_DECODER AVCodec mp3adu_decoder = { "mp3adu", AVMEDIA_TYPE_AUDIO, CODEC_ID_MP3ADU, sizeof(MPADecodeContext), decode_init, NULL, NULL, decode_frame_adu, CODEC_CAP_PARSE_ONLY, .flush= flush, .long_name= NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"), }; #endif #if CONFIG_MP3ON4_DECODER AVCodec mp3on4_decoder = { "mp3on4", AVMEDIA_TYPE_AUDIO, CODEC_ID_MP3ON4, sizeof(MP3On4DecodeContext), decode_init_mp3on4, NULL, decode_close_mp3on4, decode_frame_mp3on4, .flush= flush, .long_name= NULL_IF_CONFIG_SMALL("MP3onMP4"), }; #endif #endif