diff options
Diffstat (limited to '')
| -rw-r--r-- | media/ffvpx/libavcodec/mpegaudiodec_template.c | 2000 |
1 files changed, 2000 insertions, 0 deletions
diff --git a/media/ffvpx/libavcodec/mpegaudiodec_template.c b/media/ffvpx/libavcodec/mpegaudiodec_template.c new file mode 100644 index 0000000000..9cce88e263 --- /dev/null +++ b/media/ffvpx/libavcodec/mpegaudiodec_template.c @@ -0,0 +1,2000 @@ +/* + * 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/attributes.h" +#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; + int extrasize; + /* 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; + +#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(s->sample_rate_index == 8) + avpriv_request_sample(s->avctx, "switch point in 8khz"); + /* 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((int)(mant + (-1U << 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 > (SUINT)31) + return 0; + m = (m + ((1U << 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); + ff_dlog(NULL, "%d: norm=%x s=%"PRIx32" %"PRIx32" %"PRIx32"\n", i, + (unsigned)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); + ff_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++) { + double ci, cs, ca; + ci = ci_table[i]; + cs = 1.0 / sqrt(1.0 + ci * ci); + ca = cs * ci; +#if !USE_FLOATS + 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 + } +} + +#if USE_FLOATS +static av_cold int decode_close(AVCodecContext * avctx) +{ + MPADecodeContext *s = avctx->priv_data; + av_freep(&s->fdsp); + + return 0; +} +#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; + +#if USE_FLOATS + s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT); + if (!s->fdsp) + return AVERROR(ENOMEM); +#endif + + 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, SUINTFLOAT *in) +{ + SUINTFLOAT 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; + + ff_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; + } + } + } +} + +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->extrasize * 8) { + s->gb = s->in_gb; + s->in_gb.buffer = NULL; + s->extrasize = 0; + 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 an optimized code for + INTFLOAT v = *src + if(get_bits1(&s->gb)) + v = -v; + *dst = v; +*/ +#if USE_FLOATS +#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 - s->extrasize * 8); + + /* 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]; + + ff_dlog(s->avctx, "region=%d n=%d y=%d exp=%d\n", + i, g->region_size[i] - j, 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); + ff_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, v1, v2; + SUINTFLOAT tmp0, tmp1; + 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 USE_FLOATS + 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 USE_FLOATS +#if HAVE_MIPSFPU +# include "mips/compute_antialias_float.h" +#endif /* HAVE_MIPSFPU */ +#else +#if HAVE_MIPSDSP +# include "mips/compute_antialias_fixed.h" +#endif /* HAVE_MIPSDSP */ +#endif /* USE_FLOATS */ + +#ifndef compute_antialias +#if USE_FLOATS +#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 { \ + SUINT tmp0 = ptr[-1-j]; \ + SUINT tmp1 = ptr[ j]; \ + SUINT 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++) { + ff_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); + ff_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); + ff_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); + s->extrasize = av_clip((get_bits_left(&s->gb) >> 3) - s->extrasize, 0, + FFMAX(0, LAST_BUF_SIZE - s->last_buf_size)); + av_assert1((get_bits_count(&s->gb) & 7) == 0); + /* now we get bits from the main_data_begin offset */ + ff_dlog(s->avctx, "seekback:%d, lastbuf:%d\n", + main_data_begin, s->last_buf_size); + + memcpy(s->last_buf + s->last_buf_size, ptr, s->extrasize); + s->in_gb = s->gb; + init_get_bits(&s->gb, s->last_buf, (s->last_buf_size + s->extrasize) * 8); + 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->extrasize * 8 && s->in_gb.buffer) { + skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits + s->extrasize * 8); + s->gb = s->in_gb; + s->in_gb.buffer = NULL; + s->extrasize = 0; + } else { + skip_bits_long(&s->gb, skip); + } + } else { + gr = 0; + s->extrasize = 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; + + /* MPEG-1 scale factors */ + slen1 = slen_table[0][g->scalefac_compress]; + slen2 = slen_table[1][g->scalefac_compress]; + ff_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) - s->extrasize; + 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; + s->extrasize = 0; + } + + align_get_bits(&s->gb); + av_assert1((get_bits_count(&s->gb) & 7) == 0); + i = (get_bits_left(&s->gb) >> 3) - s->extrasize; + 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); + 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; + + int skipped = 0; + while(buf_size && !*buf){ + buf++; + buf_size--; + skipped++; + } + + 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 + skipped; + } + ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header); + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "Header missing\n"); + return AVERROR_INVALIDDATA; + } else if (ret == 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) { + 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 + skipped; +} + +static void mp_flush(MPADecodeContext *ctx) +{ + memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf)); + memset(ctx->mdct_buf, 0, sizeof(ctx->mdct_buf)); + ctx->last_buf_size = 0; + ctx->dither_state = 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; + + ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header); + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n"); + return ret; + } + /* update codec info */ + avctx->sample_rate = s->sample_rate; + 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; + + 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; + + if (s->mp3decctx[0]) + av_freep(&s->mp3decctx[0]->fdsp); + + for (i = 0; i < s->frames; i++) + av_freep(&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) { + 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; + s->mp3decctx[i]->fdsp = s->mp3decctx[0]->fdsp; + } + + 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 + + ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header); + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "Bad header, discard block\n"); + return AVERROR_INVALIDDATA; + } + + 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) { + av_log(avctx, AV_LOG_ERROR, "failed to decode channel %d\n", ch); + memset(outptr[0], 0, MPA_FRAME_SIZE*sizeof(OUT_INT)); + if (m->nb_channels > 1) + memset(outptr[1], 0, MPA_FRAME_SIZE*sizeof(OUT_INT)); + ret = m->nb_channels * MPA_FRAME_SIZE*sizeof(OUT_INT); + } + + out_size += ret; + buf += fsize; + len -= fsize; + + avctx->bit_rate += m->bit_rate; + } + if (ch != avctx->channels) { + av_log(avctx, AV_LOG_ERROR, "failed to decode all channels\n"); + return AVERROR_INVALIDDATA; + } + + /* 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 */ |