/* * 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 "libavutil/avassert.h" #include "libavutil/channel_layout.h" #include "libavutil/float_dsp.h" #include "libavutil/libm.h" #include "avcodec.h" #include "get_bits.h" #include "internal.h" #include "mathops.h" #include "mpegaudiodsp.h" /* * TODO: * - test lsf / mpeg25 extensively. */ #include "mpegaudio.h" #include "mpegaudiodecheader.h" #define BACKSTEP_SIZE 512 #define EXTRABYTES 24 #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES /* 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]; DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */ } GranuleDef; typedef struct MPADecodeContext { MPA_DECODE_HEADER uint8_t last_buf[LAST_BUF_SIZE]; int last_buf_size; /* next header (used in free format parsing) */ uint32_t free_format_next_header; GetBitContext gb; GetBitContext in_gb; DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2]; int synth_buf_offset[MPA_MAX_CHANNELS]; DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT]; INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */ GranuleDef granules[2][2]; /* Used in Layer 3 */ int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3 int dither_state; int err_recognition; AVCodecContext* avctx; MPADSPContext mpadsp; AVFloatDSPContext fdsp; AVFrame *frame; } MPADecodeContext; #if CONFIG_FLOAT # define SHR(a,b) ((a)*(1.0f/(1<<(b)))) # 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 # define OUT_FMT AV_SAMPLE_FMT_FLT # define OUT_FMT_P AV_SAMPLE_FMT_FLTP #else # define SHR(a,b) ((a)>>(b)) /* WARNING: only correct for positive 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 ## _fixed # define OUT_FMT AV_SAMPLE_FMT_S16 # define OUT_FMT_P AV_SAMPLE_FMT_S16P #endif /****************/ #define HEADER_SIZE 4 #include "mpegaudiodata.h" #include "mpegaudiodectab.h" /* 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 INTFLOAT csa_table[8][4]; 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 */ }; /** * 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) { if (s->sample_rate_index != 8) g->region_size[0] = (36 / 2); else g->region_size[0] = (72 / 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(s->sample_rate_index == 8) av_log_ask_for_sample(s->avctx, "switch point in 8khz\n"); /* if switched mode, we handle the 36 first samples as long blocks. For 8000Hz, we handle the 72 first exponents as long blocks */ if (s->sample_rate_index <= 2) g->long_end = 8; else g->long_end = 6; g->short_start = 3; } 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; #ifdef DEBUG if(e < 1) av_log(NULL, AV_LOG_WARNING, "l3_unscale: e is %d\n", e); #endif if (e > 31) return 0; m = (m + (1 << (e - 1))) >> e; return m; } static av_cold void decode_init_static(void) { int i, j, k; 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); av_dlog(NULL, "%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] = { 0 }; uint16_t tmp_codes[512] = { 0 }; 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]; } av_assert0(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]; } av_assert0(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 */ 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 = exp2(e / 4.0); k = i & 1; is_table_lsf[j][k ^ 1][i] = FIXR(f); is_table_lsf[j][k ][i] = FIXR(1.0); av_dlog(NULL, "is_table_lsf %d %d: %f %f\n", i, j, (float) is_table_lsf[j][0][i], (float) 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; #if !CONFIG_FLOAT 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); #else csa_table[i][0] = cs; csa_table[i][1] = ca; csa_table[i][2] = ca + cs; csa_table[i][3] = ca - cs; #endif } } static av_cold int decode_init(AVCodecContext * avctx) { static int initialized_tables = 0; MPADecodeContext *s = avctx->priv_data; if (!initialized_tables) { decode_init_static(); initialized_tables = 1; } s->avctx = avctx; avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); ff_mpadsp_init(&s->mpadsp); if (avctx->request_sample_fmt == OUT_FMT && avctx->codec_id != AV_CODEC_ID_MP3ON4) avctx->sample_fmt = OUT_FMT; else avctx->sample_fmt = OUT_FMT_P; s->err_recognition = avctx->err_recognition; if (avctx->codec_id == AV_CODEC_ID_MP3ADU) s->adu_mode = 1; return 0; } #define C3 FIXHR(0.86602540378443864676/2) #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36) #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36) #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36) /* 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, C4, 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, C5, 1); out[ 8] = out[ 9] = in4 + in1; out[ 2] = out[ 3] = in4 - in1; in0 -= in2; in5 = MULH3(in5 - in3, C6, 2); out[ 0] = out[ 5] = in0 - in5; out[ 6] = out[11] = in0 + in5; } /* 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; av_dlog(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) { return n ? get_bits(s, n) : 0; } 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; av_assert2((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){ switch_buffer(s, &pos, &end_pos, &end_pos2); if (pos >= end_pos) break; } y = get_vlc2(&s->gb, vlc->table, 7, 3); if (!y) { g->sb_hybrid[s_index ] = g->sb_hybrid[s_index+1] = 0; s_index += 2; continue; } exponent= exponents[s_index]; av_dlog(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->err_recognition & (AV_EF_BITSTREAM|AV_EF_COMPLIANT)) s_index=0; break; } switch_buffer(s, &pos, &end_pos, &end_pos2); if (pos >= end_pos) break; } last_pos = pos; code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1); av_dlog(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); if (bits_left < 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_COMPLIANT))) { av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left); s_index=0; } else if (bits_left > 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_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 + 72; } 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 */ #if CONFIG_FLOAT s->fdsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576); #else 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; } #endif } } #if CONFIG_FLOAT #if HAVE_MIPSFPU # include "mips/compute_antialias_float.h" #endif /* HAVE_MIPSFPU */ #else #if HAVE_MIPSDSPR1 # include "mips/compute_antialias_fixed.h" #endif /* HAVE_MIPSDSPR1 */ #endif /* CONFIG_FLOAT */ #ifndef compute_antialias #if CONFIG_FLOAT #define AA(j) do { \ float tmp0 = ptr[-1-j]; \ float tmp1 = ptr[ j]; \ ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \ ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \ } while (0) #else #define AA(j) do { \ int tmp0 = ptr[-1-j]; \ int tmp1 = ptr[ j]; \ int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \ ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \ ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \ } while (0) #endif static void compute_antialias(MPADecodeContext *s, GranuleDef *g) { INTFLOAT *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--) { AA(0); AA(1); AA(2); AA(3); AA(4); AA(5); AA(6); AA(7); ptr += 18; } } #endif /* compute_antialias */ static void compute_imdct(MPADecodeContext *s, GranuleDef *g, INTFLOAT *sb_samples, INTFLOAT *mdct_buf) { INTFLOAT *win, *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; } s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid, mdct_long_end, g->switch_point, g->block_type); buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3); ptr = g->sb_hybrid + 18 * mdct_long_end; for (j = mdct_long_end; j < sblimit; j++) { /* select frequency inversion */ win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))]; out_ptr = sb_samples + j; for (i = 0; i < 6; i++) { *out_ptr = buf[4*i]; out_ptr += SBLIMIT; } imdct12(out2, ptr + 0); for (i = 0; i < 6; i++) { *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)]; buf[4*(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[4*(i + 6*2)]; buf[4*(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[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)]; buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1); buf[4*(i + 6*2)] = 0; } ptr += 18; buf += (j&3) != 3 ? 1 : (4*18-3); } /* zero bands */ for (j = sblimit; j < SBLIMIT; j++) { /* overlap */ out_ptr = sb_samples + j; for (i = 0; i < 18; i++) { *out_ptr = buf[4*i]; buf[4*i] = 0; out_ptr += SBLIMIT; } buf += (j&3) != 3 ? 1 : (4*18-3); } } /* main layer3 decoding function */ static int mp_decode_layer3(MPADecodeContext *s) { int nb_granules, main_data_begin; 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); skip_bits(&s->gb, s->nb_channels); nb_granules = 1; } else { main_data_begin = get_bits(&s->gb, 9); if (s->nb_channels == 2) skip_bits(&s->gb, 3); else skip_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++) { av_dlog(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 AVERROR_INVALIDDATA; } 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 AVERROR_INVALIDDATA; } 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); av_dlog(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); av_dlog(s->avctx, "block_type=%d switch_point=%d\n", g->block_type, g->switch_point); } } if (!s->adu_mode) { int skip; const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3); int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0, EXTRABYTES); av_assert1((get_bits_count(&s->gb) & 7) == 0); /* now we get bits from the main_data_begin offset */ av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size); memcpy(s->last_buf + s->last_buf_size, ptr, extrasize); s->in_gb = s->gb; init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8); #if !UNCHECKED_BITSTREAM_READER s->gb.size_in_bits_plus8 += FFMAX(extrasize, LAST_BUF_SIZE - s->last_buf_size) * 8; #endif s->last_buf_size <<= 3; for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) { for (ch = 0; ch < s->nb_channels; ch++) { g = &s->granules[ch][gr]; s->last_buf_size += g->part2_3_length; memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid)); compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]); } } skip = s->last_buf_size - 8 * main_data_begin; if (skip >= s->gb.size_in_bits && s->in_gb.buffer) { skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits); s->gb = s->in_gb; s->in_gb.buffer = NULL; } else { skip_bits_long(&s->gb, skip); } } else { gr = 0; } for (; gr < nb_granules; gr++) { for (ch = 0; ch < s->nb_channels; ch++) { g = &s->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]; av_dlog(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->mode == MPA_JSTEREO) 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, ret; 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); 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); av_assert1((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); } av_assert1(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; } if(nb_frames < 0) return nb_frames; /* get output buffer */ if (!samples) { av_assert0(s->frame != NULL); s->frame->nb_samples = s->avctx->frame_size; if ((ret = ff_get_buffer(s->avctx, s->frame, 0)) < 0) return ret; samples = (OUT_INT **)s->frame->extended_data; } /* apply the synthesis filter */ for (ch = 0; ch < s->nb_channels; ch++) { int sample_stride; if (s->avctx->sample_fmt == OUT_FMT_P) { samples_ptr = samples[ch]; sample_stride = 1; } else { samples_ptr = samples[0] + ch; sample_stride = s->nb_channels; } for (i = 0; i < nb_frames; i++) { RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch], &(s->synth_buf_offset[ch]), RENAME(ff_mpa_synth_window), &s->dither_state, samples_ptr, sample_stride, s->sb_samples[ch][i]); samples_ptr += 32 * sample_stride; } } return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels; } static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; MPADecodeContext *s = avctx->priv_data; uint32_t header; int ret; while(buf_size && !*buf){ buf++; buf_size--; } if (buf_size < HEADER_SIZE) return AVERROR_INVALIDDATA; header = AV_RB32(buf); if (header>>8 == AV_RB32("TAG")>>8) { av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n"); return buf_size; } if (ff_mpa_check_header(header) < 0) { av_log(avctx, AV_LOG_ERROR, "Header missing\n"); return AVERROR_INVALIDDATA; } if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) { /* free format: prepare to compute frame size */ s->frame_size = -1; return AVERROR_INVALIDDATA; } /* update codec info */ avctx->channels = s->nb_channels; avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO; if (!avctx->bit_rate) avctx->bit_rate = s->bit_rate; if (s->frame_size <= 0 || s->frame_size > buf_size) { av_log(avctx, AV_LOG_ERROR, "incomplete frame\n"); return AVERROR_INVALIDDATA; } else if (s->frame_size < buf_size) { av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n"); buf_size= s->frame_size; } s->frame = data; ret = mp_decode_frame(s, NULL, buf, buf_size); if (ret >= 0) { s->frame->nb_samples = avctx->frame_size; *got_frame_ptr = 1; avctx->sample_rate = s->sample_rate; //FIXME maybe move the other codec info stuff from above here too } else { av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n"); /* Only return an error if the bad frame makes up the whole packet or * the error is related to buffer management. * If there is more data in the packet, just consume the bad frame * instead of returning an error, which would discard the whole * packet. */ *got_frame_ptr = 0; if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA) return ret; } s->frame_size = 0; return buf_size; } static void mp_flush(MPADecodeContext *ctx) { memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf)); ctx->last_buf_size = 0; } static void flush(AVCodecContext *avctx) { mp_flush(avctx->priv_data); } #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER static int decode_frame_adu(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; MPADecodeContext *s = avctx->priv_data; uint32_t header; int len, ret; int av_unused out_size; len = buf_size; // Discard too short frames if (buf_size < HEADER_SIZE) { av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); return AVERROR_INVALIDDATA; } 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 av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n"); return AVERROR_INVALIDDATA; } avpriv_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; s->frame_size = len; s->frame = data; ret = mp_decode_frame(s, NULL, buf, buf_size); if (ret < 0) { av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n"); return ret; } *got_frame_ptr = 1; 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; ///< channel 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) */ /* number of mp3 decoder instances */ static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 }; /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */ static const uint8_t chan_offset[8][5] = { { 0 }, { 0 }, // C { 0 }, // FLR { 2, 0 }, // C FLR { 2, 0, 3 }, // C FLR BS { 2, 0, 3 }, // C FLR BLRS { 2, 0, 4, 3 }, // C FLR BLRS LFE { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE }; /* mp3on4 channel layouts */ static const int16_t chan_layout[8] = { 0, AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_SURROUND, AV_CH_LAYOUT_4POINT0, AV_CH_LAYOUT_5POINT0, AV_CH_LAYOUT_5POINT1, AV_CH_LAYOUT_7POINT1 }; static av_cold int decode_close_mp3on4(AVCodecContext * avctx) { MP3On4DecodeContext *s = avctx->priv_data; int i; for (i = 0; i < s->frames; i++) av_free(s->mp3decctx[i]); return 0; } 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 AVERROR_INVALIDDATA; } avpriv_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size * 8, 1); if (!cfg.chan_config || cfg.chan_config > 7) { av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n"); return AVERROR_INVALIDDATA; } s->frames = mp3Frames[cfg.chan_config]; s->coff = chan_offset[cfg.chan_config]; avctx->channels = ff_mpeg4audio_channels[cfg.chan_config]; avctx->channel_layout = chan_layout[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)); if (!s->mp3decctx[0]) goto alloc_fail; // 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)); if (!s->mp3decctx[i]) goto alloc_fail; s->mp3decctx[i]->adu_mode = 1; s->mp3decctx[i]->avctx = avctx; s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp; } return 0; alloc_fail: decode_close_mp3on4(avctx); return AVERROR(ENOMEM); } static void flush_mp3on4(AVCodecContext *avctx) { int i; MP3On4DecodeContext *s = avctx->priv_data; for (i = 0; i < s->frames; i++) mp_flush(s->mp3decctx[i]); } static int decode_frame_mp3on4(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { AVFrame *frame = data; 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; OUT_INT *outptr[2]; int fr, ch, ret; /* get output buffer */ frame->nb_samples = MPA_FRAME_SIZE; if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; out_samples = (OUT_INT **)frame->extended_data; // Discard too short frames if (buf_size < HEADER_SIZE) return AVERROR_INVALIDDATA; avctx->bit_rate = 0; ch = 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]; av_assert1(m); if (fsize < HEADER_SIZE) { av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n"); return AVERROR_INVALIDDATA; } header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header if (ff_mpa_check_header(header) < 0) // Bad header, discard block break; avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header); if (ch + m->nb_channels > avctx->channels || s->coff[fr] + m->nb_channels > avctx->channels) { av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec " "channel count\n"); return AVERROR_INVALIDDATA; } ch += m->nb_channels; outptr[0] = out_samples[s->coff[fr]]; if (m->nb_channels > 1) outptr[1] = out_samples[s->coff[fr] + 1]; if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0) return ret; out_size += ret; buf += fsize; len -= fsize; avctx->bit_rate += m->bit_rate; } /* update codec info */ avctx->sample_rate = s->mp3decctx[0]->sample_rate; frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT)); *got_frame_ptr = 1; return buf_size; } #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */ #if !CONFIG_FLOAT #if CONFIG_MP1_DECODER AVCodec ff_mp1_decoder = { .name = "mp1", .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_MP1, .priv_data_size = sizeof(MPADecodeContext), .init = decode_init, .decode = decode_frame, .capabilities = CODEC_CAP_DR1, .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"), .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, }; #endif #if CONFIG_MP2_DECODER AVCodec ff_mp2_decoder = { .name = "mp2", .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_MP2, .priv_data_size = sizeof(MPADecodeContext), .init = decode_init, .decode = decode_frame, .capabilities = CODEC_CAP_DR1, .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, }; #endif #if CONFIG_MP3_DECODER AVCodec ff_mp3_decoder = { .name = "mp3", .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_MP3, .priv_data_size = sizeof(MPADecodeContext), .init = decode_init, .decode = decode_frame, .capabilities = CODEC_CAP_DR1, .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"), .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, }; #endif #if CONFIG_MP3ADU_DECODER AVCodec ff_mp3adu_decoder = { .name = "mp3adu", .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_MP3ADU, .priv_data_size = sizeof(MPADecodeContext), .init = decode_init, .decode = decode_frame_adu, .capabilities = CODEC_CAP_DR1, .flush = flush, .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"), .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, }; #endif #if CONFIG_MP3ON4_DECODER AVCodec ff_mp3on4_decoder = { .name = "mp3on4", .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_MP3ON4, .priv_data_size = sizeof(MP3On4DecodeContext), .init = decode_init_mp3on4, .close = decode_close_mp3on4, .decode = decode_frame_mp3on4, .capabilities = CODEC_CAP_DR1, .flush = flush_mp3on4, .long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"), .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE }, }; #endif #endif