diff options
author | Diego Biurrun <diego@biurrun.de> | 2013-11-19 21:47:39 +0100 |
---|---|---|
committer | Diego Biurrun <diego@biurrun.de> | 2013-11-23 21:36:49 +0100 |
commit | 0eeeb9647e9c92c9edfd0b18c7cb5da7ac666f85 (patch) | |
tree | f3faf163a9e384dff056d53a272b1e912e988d27 /libavcodec/mpegaudiodec.c | |
parent | 48b24bd2d208ce0f124029ac4c5ac5cb1fca4175 (diff) | |
download | ffmpeg-0eeeb9647e9c92c9edfd0b18c7cb5da7ac666f85.tar.gz |
mpegaudiodec: Consistently handle fixed/float templating
Diffstat (limited to 'libavcodec/mpegaudiodec.c')
-rw-r--r-- | libavcodec/mpegaudiodec.c | 2055 |
1 files changed, 0 insertions, 2055 deletions
diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c deleted file mode 100644 index 0c702b054f..0000000000 --- a/libavcodec/mpegaudiodec.c +++ /dev/null @@ -1,2055 +0,0 @@ -/* - * MPEG Audio decoder - * Copyright (c) 2001, 2002 Fabrice Bellard - * - * This file is part of Libav. - * - * Libav 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. - * - * Libav 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 Libav; 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/attributes.h" -#include "libavutil/avassert.h" -#include "libavutil/channel_layout.h" -#include "libavutil/float_dsp.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 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 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 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 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 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; - assert(e >= 1); - 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]; - } - 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 */ - - 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); - 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; - 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){ - 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) - 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_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_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 -#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; - } -} - -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); - 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); - init_long_region(s, g, region_address1, region_address2); - } - region_offset2size(g); - 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, - FFMAX(0, LAST_BUF_SIZE - s->last_buf_size)); - assert((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 += extrasize * 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); - - if (nb_frames < 0) - return nb_frames; - - 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; - } - - /* 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) { - av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n"); - 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; - - if (buf_size < HEADER_SIZE) - return AVERROR_INVALIDDATA; - - header = AV_RB32(buf); - 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) { - 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; - - 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 av_cold 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) { - av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); - 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]; - assert(m != NULL); - - 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", - .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"), - .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, - .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", - .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), - .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, - .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", - .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"), - .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, - .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", - .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"), - .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, - .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", - .long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"), - .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, - .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, - AV_SAMPLE_FMT_NONE }, -}; -#endif -#endif |