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2003 lines
65 KiB
2003 lines
65 KiB
/* |
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* MPEG Audio decoder |
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* Copyright (c) 2001, 2002 Fabrice Bellard |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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|
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/** |
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* @file |
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* MPEG Audio decoder |
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*/ |
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#include "libavutil/attributes.h" |
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#include "libavutil/avassert.h" |
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#include "libavutil/channel_layout.h" |
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#include "libavutil/float_dsp.h" |
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#include "libavutil/libm.h" |
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#include "avcodec.h" |
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#include "get_bits.h" |
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#include "internal.h" |
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#include "mathops.h" |
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#include "mpegaudiodsp.h" |
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|
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/* |
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* TODO: |
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* - test lsf / mpeg25 extensively. |
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*/ |
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#include "mpegaudio.h" |
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#include "mpegaudiodecheader.h" |
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#define BACKSTEP_SIZE 512 |
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#define EXTRABYTES 24 |
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#define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES |
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|
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/* layer 3 "granule" */ |
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typedef struct GranuleDef { |
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uint8_t scfsi; |
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int part2_3_length; |
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int big_values; |
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int global_gain; |
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int scalefac_compress; |
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uint8_t block_type; |
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uint8_t switch_point; |
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int table_select[3]; |
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int subblock_gain[3]; |
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uint8_t scalefac_scale; |
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uint8_t count1table_select; |
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int region_size[3]; /* number of huffman codes in each region */ |
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int preflag; |
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int short_start, long_end; /* long/short band indexes */ |
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uint8_t scale_factors[40]; |
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DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */ |
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} GranuleDef; |
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|
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typedef struct MPADecodeContext { |
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MPA_DECODE_HEADER |
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uint8_t last_buf[LAST_BUF_SIZE]; |
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int last_buf_size; |
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/* next header (used in free format parsing) */ |
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uint32_t free_format_next_header; |
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GetBitContext gb; |
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GetBitContext in_gb; |
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DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2]; |
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int synth_buf_offset[MPA_MAX_CHANNELS]; |
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DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT]; |
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INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */ |
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GranuleDef granules[2][2]; /* Used in Layer 3 */ |
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int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3 |
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int dither_state; |
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int err_recognition; |
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AVCodecContext* avctx; |
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MPADSPContext mpadsp; |
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AVFloatDSPContext *fdsp; |
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AVFrame *frame; |
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} MPADecodeContext; |
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#define HEADER_SIZE 4 |
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#include "mpegaudiodata.h" |
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#include "mpegaudiodectab.h" |
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|
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/* vlc structure for decoding layer 3 huffman tables */ |
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static VLC huff_vlc[16]; |
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static VLC_TYPE huff_vlc_tables[ |
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0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 + |
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142 + 204 + 190 + 170 + 542 + 460 + 662 + 414 |
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][2]; |
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static const int huff_vlc_tables_sizes[16] = { |
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0, 128, 128, 128, 130, 128, 154, 166, |
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142, 204, 190, 170, 542, 460, 662, 414 |
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}; |
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static VLC huff_quad_vlc[2]; |
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static VLC_TYPE huff_quad_vlc_tables[128+16][2]; |
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static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 }; |
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/* computed from band_size_long */ |
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static uint16_t band_index_long[9][23]; |
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#include "mpegaudio_tablegen.h" |
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/* intensity stereo coef table */ |
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static INTFLOAT is_table[2][16]; |
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static INTFLOAT is_table_lsf[2][2][16]; |
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static INTFLOAT csa_table[8][4]; |
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static int16_t division_tab3[1<<6 ]; |
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static int16_t division_tab5[1<<8 ]; |
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static int16_t division_tab9[1<<11]; |
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static int16_t * const division_tabs[4] = { |
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division_tab3, division_tab5, NULL, division_tab9 |
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}; |
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/* lower 2 bits: modulo 3, higher bits: shift */ |
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static uint16_t scale_factor_modshift[64]; |
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/* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */ |
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static int32_t scale_factor_mult[15][3]; |
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/* mult table for layer 2 group quantization */ |
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#define SCALE_GEN(v) \ |
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{ FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) } |
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static const int32_t scale_factor_mult2[3][3] = { |
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SCALE_GEN(4.0 / 3.0), /* 3 steps */ |
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SCALE_GEN(4.0 / 5.0), /* 5 steps */ |
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SCALE_GEN(4.0 / 9.0), /* 9 steps */ |
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}; |
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/** |
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* Convert region offsets to region sizes and truncate |
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* size to big_values. |
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*/ |
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static void region_offset2size(GranuleDef *g) |
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{ |
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int i, k, j = 0; |
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g->region_size[2] = 576 / 2; |
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for (i = 0; i < 3; i++) { |
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k = FFMIN(g->region_size[i], g->big_values); |
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g->region_size[i] = k - j; |
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j = k; |
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} |
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} |
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static void init_short_region(MPADecodeContext *s, GranuleDef *g) |
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{ |
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if (g->block_type == 2) { |
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if (s->sample_rate_index != 8) |
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g->region_size[0] = (36 / 2); |
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else |
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g->region_size[0] = (72 / 2); |
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} else { |
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if (s->sample_rate_index <= 2) |
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g->region_size[0] = (36 / 2); |
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else if (s->sample_rate_index != 8) |
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g->region_size[0] = (54 / 2); |
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else |
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g->region_size[0] = (108 / 2); |
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} |
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g->region_size[1] = (576 / 2); |
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} |
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static void init_long_region(MPADecodeContext *s, GranuleDef *g, |
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int ra1, int ra2) |
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{ |
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int l; |
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g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1; |
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/* should not overflow */ |
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l = FFMIN(ra1 + ra2 + 2, 22); |
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g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1; |
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} |
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static void compute_band_indexes(MPADecodeContext *s, GranuleDef *g) |
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{ |
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if (g->block_type == 2) { |
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if (g->switch_point) { |
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if(s->sample_rate_index == 8) |
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avpriv_request_sample(s->avctx, "switch point in 8khz"); |
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/* if switched mode, we handle the 36 first samples as |
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long blocks. For 8000Hz, we handle the 72 first |
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exponents as long blocks */ |
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if (s->sample_rate_index <= 2) |
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g->long_end = 8; |
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else |
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g->long_end = 6; |
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g->short_start = 3; |
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} else { |
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g->long_end = 0; |
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g->short_start = 0; |
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} |
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} else { |
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g->short_start = 13; |
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g->long_end = 22; |
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} |
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} |
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/* layer 1 unscaling */ |
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/* n = number of bits of the mantissa minus 1 */ |
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static inline int l1_unscale(int n, int mant, int scale_factor) |
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{ |
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int shift, mod; |
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int64_t val; |
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shift = scale_factor_modshift[scale_factor]; |
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mod = shift & 3; |
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shift >>= 2; |
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val = MUL64((int)(mant + (-1U << n) + 1), scale_factor_mult[n-1][mod]); |
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shift += n; |
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/* NOTE: at this point, 1 <= shift >= 21 + 15 */ |
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return (int)((val + (1LL << (shift - 1))) >> shift); |
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} |
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static inline int l2_unscale_group(int steps, int mant, int scale_factor) |
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{ |
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int shift, mod, val; |
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shift = scale_factor_modshift[scale_factor]; |
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mod = shift & 3; |
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shift >>= 2; |
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val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod]; |
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/* NOTE: at this point, 0 <= shift <= 21 */ |
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if (shift > 0) |
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val = (val + (1 << (shift - 1))) >> shift; |
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return val; |
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} |
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/* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */ |
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static inline int l3_unscale(int value, int exponent) |
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{ |
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unsigned int m; |
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int e; |
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e = table_4_3_exp [4 * value + (exponent & 3)]; |
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m = table_4_3_value[4 * value + (exponent & 3)]; |
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e -= exponent >> 2; |
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#ifdef DEBUG |
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if(e < 1) |
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av_log(NULL, AV_LOG_WARNING, "l3_unscale: e is %d\n", e); |
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#endif |
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if (e > 31) |
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return 0; |
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m = (m + (1 << (e - 1))) >> e; |
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return m; |
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} |
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static av_cold void decode_init_static(void) |
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{ |
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int i, j, k; |
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int offset; |
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|
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/* scale factors table for layer 1/2 */ |
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for (i = 0; i < 64; i++) { |
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int shift, mod; |
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/* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */ |
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shift = i / 3; |
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mod = i % 3; |
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scale_factor_modshift[i] = mod | (shift << 2); |
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} |
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/* scale factor multiply for layer 1 */ |
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for (i = 0; i < 15; i++) { |
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int n, norm; |
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n = i + 2; |
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norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1); |
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scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS); |
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scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS); |
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scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS); |
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ff_dlog(NULL, "%d: norm=%x s=%x %x %x\n", i, norm, |
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scale_factor_mult[i][0], |
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scale_factor_mult[i][1], |
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scale_factor_mult[i][2]); |
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} |
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RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window)); |
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/* huffman decode tables */ |
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offset = 0; |
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for (i = 1; i < 16; i++) { |
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const HuffTable *h = &mpa_huff_tables[i]; |
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int xsize, x, y; |
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uint8_t tmp_bits [512] = { 0 }; |
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uint16_t tmp_codes[512] = { 0 }; |
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xsize = h->xsize; |
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j = 0; |
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for (x = 0; x < xsize; x++) { |
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for (y = 0; y < xsize; y++) { |
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tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ]; |
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tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++]; |
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} |
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} |
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/* XXX: fail test */ |
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huff_vlc[i].table = huff_vlc_tables+offset; |
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huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i]; |
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init_vlc(&huff_vlc[i], 7, 512, |
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tmp_bits, 1, 1, tmp_codes, 2, 2, |
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INIT_VLC_USE_NEW_STATIC); |
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offset += huff_vlc_tables_sizes[i]; |
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} |
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av_assert0(offset == FF_ARRAY_ELEMS(huff_vlc_tables)); |
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offset = 0; |
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for (i = 0; i < 2; i++) { |
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huff_quad_vlc[i].table = huff_quad_vlc_tables+offset; |
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huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i]; |
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init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16, |
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mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1, |
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INIT_VLC_USE_NEW_STATIC); |
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offset += huff_quad_vlc_tables_sizes[i]; |
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} |
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av_assert0(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables)); |
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for (i = 0; i < 9; i++) { |
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k = 0; |
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for (j = 0; j < 22; j++) { |
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band_index_long[i][j] = k; |
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k += band_size_long[i][j]; |
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} |
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band_index_long[i][22] = k; |
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} |
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/* compute n ^ (4/3) and store it in mantissa/exp format */ |
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mpegaudio_tableinit(); |
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for (i = 0; i < 4; i++) { |
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if (ff_mpa_quant_bits[i] < 0) { |
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for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) { |
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int val1, val2, val3, steps; |
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int val = j; |
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steps = ff_mpa_quant_steps[i]; |
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val1 = val % steps; |
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val /= steps; |
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val2 = val % steps; |
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val3 = val / steps; |
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division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8); |
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} |
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} |
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} |
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for (i = 0; i < 7; i++) { |
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float f; |
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INTFLOAT v; |
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if (i != 6) { |
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f = tan((double)i * M_PI / 12.0); |
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v = FIXR(f / (1.0 + f)); |
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} else { |
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v = FIXR(1.0); |
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} |
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is_table[0][ i] = v; |
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is_table[1][6 - i] = v; |
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} |
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/* invalid values */ |
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for (i = 7; i < 16; i++) |
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is_table[0][i] = is_table[1][i] = 0.0; |
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for (i = 0; i < 16; i++) { |
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double f; |
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int e, k; |
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for (j = 0; j < 2; j++) { |
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e = -(j + 1) * ((i + 1) >> 1); |
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f = exp2(e / 4.0); |
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k = i & 1; |
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is_table_lsf[j][k ^ 1][i] = FIXR(f); |
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is_table_lsf[j][k ][i] = FIXR(1.0); |
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ff_dlog(NULL, "is_table_lsf %d %d: %f %f\n", |
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i, j, (float) is_table_lsf[j][0][i], |
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(float) is_table_lsf[j][1][i]); |
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} |
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} |
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for (i = 0; i < 8; i++) { |
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double ci, cs, ca; |
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ci = ci_table[i]; |
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cs = 1.0 / sqrt(1.0 + ci * ci); |
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ca = cs * ci; |
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#if !USE_FLOATS |
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csa_table[i][0] = FIXHR(cs/4); |
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csa_table[i][1] = FIXHR(ca/4); |
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csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4); |
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csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4); |
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#else |
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csa_table[i][0] = cs; |
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csa_table[i][1] = ca; |
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csa_table[i][2] = ca + cs; |
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csa_table[i][3] = ca - cs; |
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#endif |
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} |
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} |
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#if USE_FLOATS |
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static av_cold int decode_close(AVCodecContext * avctx) |
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{ |
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MPADecodeContext *s = avctx->priv_data; |
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av_freep(&s->fdsp); |
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return 0; |
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} |
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#endif |
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static av_cold int decode_init(AVCodecContext * avctx) |
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{ |
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static int initialized_tables = 0; |
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MPADecodeContext *s = avctx->priv_data; |
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|
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if (!initialized_tables) { |
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decode_init_static(); |
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initialized_tables = 1; |
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} |
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s->avctx = avctx; |
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#if USE_FLOATS |
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s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT); |
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if (!s->fdsp) |
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return AVERROR(ENOMEM); |
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#endif |
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ff_mpadsp_init(&s->mpadsp); |
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if (avctx->request_sample_fmt == OUT_FMT && |
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avctx->codec_id != AV_CODEC_ID_MP3ON4) |
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avctx->sample_fmt = OUT_FMT; |
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else |
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avctx->sample_fmt = OUT_FMT_P; |
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s->err_recognition = avctx->err_recognition; |
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if (avctx->codec_id == AV_CODEC_ID_MP3ADU) |
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s->adu_mode = 1; |
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return 0; |
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} |
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#define C3 FIXHR(0.86602540378443864676/2) |
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#define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36) |
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#define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36) |
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#define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36) |
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|
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/* 12 points IMDCT. We compute it "by hand" by factorizing obvious |
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cases. */ |
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static void imdct12(INTFLOAT *out, INTFLOAT *in) |
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{ |
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INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2; |
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|
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in0 = in[0*3]; |
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in1 = in[1*3] + in[0*3]; |
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in2 = in[2*3] + in[1*3]; |
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in3 = in[3*3] + in[2*3]; |
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in4 = in[4*3] + in[3*3]; |
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in5 = in[5*3] + in[4*3]; |
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in5 += in3; |
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in3 += in1; |
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|
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in2 = MULH3(in2, C3, 2); |
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in3 = MULH3(in3, C3, 4); |
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|
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t1 = in0 - in4; |
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t2 = MULH3(in1 - in5, C4, 2); |
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out[ 7] = |
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out[10] = t1 + t2; |
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out[ 1] = |
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out[ 4] = t1 - t2; |
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|
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in0 += SHR(in4, 1); |
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in4 = in0 + in2; |
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in5 += 2*in1; |
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in1 = MULH3(in5 + in3, C5, 1); |
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out[ 8] = |
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out[ 9] = in4 + in1; |
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out[ 2] = |
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out[ 3] = in4 - in1; |
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in0 -= in2; |
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in5 = MULH3(in5 - in3, C6, 2); |
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out[ 0] = |
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out[ 5] = in0 - in5; |
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out[ 6] = |
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out[11] = in0 + in5; |
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} |
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|
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/* return the number of decoded frames */ |
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static int mp_decode_layer1(MPADecodeContext *s) |
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{ |
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int bound, i, v, n, ch, j, mant; |
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uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT]; |
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uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT]; |
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|
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if (s->mode == MPA_JSTEREO) |
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bound = (s->mode_ext + 1) * 4; |
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else |
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bound = SBLIMIT; |
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|
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/* allocation bits */ |
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for (i = 0; i < bound; i++) { |
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for (ch = 0; ch < s->nb_channels; ch++) { |
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allocation[ch][i] = get_bits(&s->gb, 4); |
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} |
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} |
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for (i = bound; i < SBLIMIT; i++) |
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allocation[0][i] = get_bits(&s->gb, 4); |
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|
|
/* 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; |
|
} |
|
} |
|
} |
|
} |
|
|
|
/* handle n = 0 too */ |
|
static inline int get_bitsz(GetBitContext *s, int n) |
|
{ |
|
return n ? get_bits(s, n) : 0; |
|
} |
|
|
|
|
|
static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, |
|
int *end_pos2) |
|
{ |
|
if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) { |
|
s->gb = s->in_gb; |
|
s->in_gb.buffer = NULL; |
|
av_assert2((get_bits_count(&s->gb) & 7) == 0); |
|
skip_bits_long(&s->gb, *pos - *end_pos); |
|
*end_pos2 = |
|
*end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos; |
|
*pos = get_bits_count(&s->gb); |
|
} |
|
} |
|
|
|
/* Following is a optimized code for |
|
INTFLOAT v = *src |
|
if(get_bits1(&s->gb)) |
|
v = -v; |
|
*dst = v; |
|
*/ |
|
#if 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); |
|
|
|
/* 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 x=%d y=%d exp=%d\n", |
|
i, g->region_size[i] - j, x, y, exponent); |
|
if (y & 16) { |
|
x = y >> 5; |
|
y = y & 0x0f; |
|
if (x < 15) { |
|
READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x) |
|
} else { |
|
x += get_bitsz(&s->gb, linbits); |
|
v = l3_unscale(x, exponent); |
|
if (get_bits1(&s->gb)) |
|
v = -v; |
|
g->sb_hybrid[s_index] = v; |
|
} |
|
if (y < 15) { |
|
READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y) |
|
} else { |
|
y += get_bitsz(&s->gb, linbits); |
|
v = l3_unscale(y, exponent); |
|
if (get_bits1(&s->gb)) |
|
v = -v; |
|
g->sb_hybrid[s_index+1] = v; |
|
} |
|
} else { |
|
x = y >> 5; |
|
y = y & 0x0f; |
|
x += y; |
|
if (x < 15) { |
|
READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x) |
|
} else { |
|
x += get_bitsz(&s->gb, linbits); |
|
v = l3_unscale(x, exponent); |
|
if (get_bits1(&s->gb)) |
|
v = -v; |
|
g->sb_hybrid[s_index+!!y] = v; |
|
} |
|
g->sb_hybrid[s_index + !y] = 0; |
|
} |
|
s_index += 2; |
|
} |
|
} |
|
|
|
/* high frequencies */ |
|
vlc = &huff_quad_vlc[g->count1table_select]; |
|
last_pos = 0; |
|
while (s_index <= 572) { |
|
int pos, code; |
|
pos = get_bits_count(&s->gb); |
|
if (pos >= end_pos) { |
|
if (pos > end_pos2 && last_pos) { |
|
/* some encoders generate an incorrect size for this |
|
part. We must go back into the data */ |
|
s_index -= 4; |
|
skip_bits_long(&s->gb, last_pos - pos); |
|
av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos); |
|
if(s->err_recognition & (AV_EF_BITSTREAM|AV_EF_COMPLIANT)) |
|
s_index=0; |
|
break; |
|
} |
|
switch_buffer(s, &pos, &end_pos, &end_pos2); |
|
if (pos >= end_pos) |
|
break; |
|
} |
|
last_pos = pos; |
|
|
|
code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1); |
|
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, 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 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 { \ |
|
int tmp0 = ptr[-1-j]; \ |
|
int tmp1 = ptr[ j]; \ |
|
int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \ |
|
ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \ |
|
ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \ |
|
} while (0) |
|
#endif |
|
|
|
static void compute_antialias(MPADecodeContext *s, GranuleDef *g) |
|
{ |
|
INTFLOAT *ptr; |
|
int n, i; |
|
|
|
/* we antialias only "long" bands */ |
|
if (g->block_type == 2) { |
|
if (!g->switch_point) |
|
return; |
|
/* XXX: check this for 8000Hz case */ |
|
n = 1; |
|
} else { |
|
n = SBLIMIT - 1; |
|
} |
|
|
|
ptr = g->sb_hybrid + 18; |
|
for (i = n; i > 0; i--) { |
|
AA(0); |
|
AA(1); |
|
AA(2); |
|
AA(3); |
|
AA(4); |
|
AA(5); |
|
AA(6); |
|
AA(7); |
|
|
|
ptr += 18; |
|
} |
|
} |
|
#endif /* compute_antialias */ |
|
|
|
static void compute_imdct(MPADecodeContext *s, GranuleDef *g, |
|
INTFLOAT *sb_samples, INTFLOAT *mdct_buf) |
|
{ |
|
INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1; |
|
INTFLOAT out2[12]; |
|
int i, j, mdct_long_end, sblimit; |
|
|
|
/* find last non zero block */ |
|
ptr = g->sb_hybrid + 576; |
|
ptr1 = g->sb_hybrid + 2 * 18; |
|
while (ptr >= ptr1) { |
|
int32_t *p; |
|
ptr -= 6; |
|
p = (int32_t*)ptr; |
|
if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5]) |
|
break; |
|
} |
|
sblimit = ((ptr - g->sb_hybrid) / 18) + 1; |
|
|
|
if (g->block_type == 2) { |
|
/* XXX: check for 8000 Hz */ |
|
if (g->switch_point) |
|
mdct_long_end = 2; |
|
else |
|
mdct_long_end = 0; |
|
} else { |
|
mdct_long_end = sblimit; |
|
} |
|
|
|
s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid, |
|
mdct_long_end, g->switch_point, |
|
g->block_type); |
|
|
|
buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3); |
|
ptr = g->sb_hybrid + 18 * mdct_long_end; |
|
|
|
for (j = mdct_long_end; j < sblimit; j++) { |
|
/* select frequency inversion */ |
|
win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))]; |
|
out_ptr = sb_samples + j; |
|
|
|
for (i = 0; i < 6; i++) { |
|
*out_ptr = buf[4*i]; |
|
out_ptr += SBLIMIT; |
|
} |
|
imdct12(out2, ptr + 0); |
|
for (i = 0; i < 6; i++) { |
|
*out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)]; |
|
buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1); |
|
out_ptr += SBLIMIT; |
|
} |
|
imdct12(out2, ptr + 1); |
|
for (i = 0; i < 6; i++) { |
|
*out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)]; |
|
buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1); |
|
out_ptr += SBLIMIT; |
|
} |
|
imdct12(out2, ptr + 2); |
|
for (i = 0; i < 6; i++) { |
|
buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)]; |
|
buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1); |
|
buf[4*(i + 6*2)] = 0; |
|
} |
|
ptr += 18; |
|
buf += (j&3) != 3 ? 1 : (4*18-3); |
|
} |
|
/* zero bands */ |
|
for (j = sblimit; j < SBLIMIT; j++) { |
|
/* overlap */ |
|
out_ptr = sb_samples + j; |
|
for (i = 0; i < 18; i++) { |
|
*out_ptr = buf[4*i]; |
|
buf[4*i] = 0; |
|
out_ptr += SBLIMIT; |
|
} |
|
buf += (j&3) != 3 ? 1 : (4*18-3); |
|
} |
|
} |
|
|
|
/* main layer3 decoding function */ |
|
static int mp_decode_layer3(MPADecodeContext *s) |
|
{ |
|
int nb_granules, main_data_begin; |
|
int gr, ch, blocksplit_flag, i, j, k, n, bits_pos; |
|
GranuleDef *g; |
|
int16_t exponents[576]; //FIXME try INTFLOAT |
|
|
|
/* read side info */ |
|
if (s->lsf) { |
|
main_data_begin = get_bits(&s->gb, 8); |
|
skip_bits(&s->gb, s->nb_channels); |
|
nb_granules = 1; |
|
} else { |
|
main_data_begin = get_bits(&s->gb, 9); |
|
if (s->nb_channels == 2) |
|
skip_bits(&s->gb, 3); |
|
else |
|
skip_bits(&s->gb, 5); |
|
nb_granules = 2; |
|
for (ch = 0; ch < s->nb_channels; ch++) { |
|
s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */ |
|
s->granules[ch][1].scfsi = get_bits(&s->gb, 4); |
|
} |
|
} |
|
|
|
for (gr = 0; gr < nb_granules; gr++) { |
|
for (ch = 0; ch < s->nb_channels; ch++) { |
|
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); |
|
int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0, EXTRABYTES); |
|
av_assert1((get_bits_count(&s->gb) & 7) == 0); |
|
/* now we get bits from the main_data_begin offset */ |
|
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, extrasize); |
|
s->in_gb = s->gb; |
|
init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8); |
|
#if !UNCHECKED_BITSTREAM_READER |
|
s->gb.size_in_bits_plus8 += FFMAX(extrasize, LAST_BUF_SIZE - s->last_buf_size) * 8; |
|
#endif |
|
s->last_buf_size <<= 3; |
|
for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) { |
|
for (ch = 0; ch < s->nb_channels; ch++) { |
|
g = &s->granules[ch][gr]; |
|
s->last_buf_size += g->part2_3_length; |
|
memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid)); |
|
compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]); |
|
} |
|
} |
|
skip = s->last_buf_size - 8 * main_data_begin; |
|
if (skip >= s->gb.size_in_bits && s->in_gb.buffer) { |
|
skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits); |
|
s->gb = s->in_gb; |
|
s->in_gb.buffer = NULL; |
|
} else { |
|
skip_bits_long(&s->gb, skip); |
|
} |
|
} else { |
|
gr = 0; |
|
} |
|
|
|
for (; gr < nb_granules; gr++) { |
|
for (ch = 0; ch < s->nb_channels; ch++) { |
|
g = &s->granules[ch][gr]; |
|
bits_pos = get_bits_count(&s->gb); |
|
|
|
if (!s->lsf) { |
|
uint8_t *sc; |
|
int slen, slen1, slen2; |
|
|
|
/* MPEG1 scale factors */ |
|
slen1 = slen_table[0][g->scalefac_compress]; |
|
slen2 = slen_table[1][g->scalefac_compress]; |
|
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; |
|
if (i >= 0 && i <= BACKSTEP_SIZE) { |
|
memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i); |
|
s->last_buf_size=i; |
|
} else |
|
av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i); |
|
s->gb = s->in_gb; |
|
s->in_gb.buffer = NULL; |
|
} |
|
|
|
align_get_bits(&s->gb); |
|
av_assert1((get_bits_count(&s->gb) & 7) == 0); |
|
i = get_bits_left(&s->gb) >> 3; |
|
|
|
if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) { |
|
if (i < 0) |
|
av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i); |
|
i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE); |
|
} |
|
av_assert1(i <= buf_size - HEADER_SIZE && i >= 0); |
|
memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i); |
|
s->last_buf_size += i; |
|
} |
|
|
|
if(nb_frames < 0) |
|
return nb_frames; |
|
|
|
/* get output buffer */ |
|
if (!samples) { |
|
av_assert0(s->frame); |
|
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; |
|
} |
|
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) { |
|
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; |
|
|
|
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; |
|
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; |
|
|
|
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 |
|
|
|
if (ff_mpa_check_header(header) < 0) { |
|
av_log(avctx, AV_LOG_ERROR, "Bad header, discard block\n"); |
|
return AVERROR_INVALIDDATA; |
|
} |
|
|
|
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) { |
|
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 */
|
|
|