mirror of
https://github.com/FFmpeg/FFmpeg.git
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75a37b57a5
* qatar/master: APIchanges: fill in date and commit for request_sample_fmt Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders. Add support for request_sample_format in ffmpeg and ffplay. Add APIchanges entry for request_sample_fmt. Add request_sample_fmt field to AVCodecContext. Add float_interleave() to FmtConvertContext with x86-optimized versions. Remove unused make variable SEEK_REFFILE fate: remove redundant aref and vref references fate: remove do_ffmpeg_nocheck function fate: do not collect -benchmark output mpegaudiodec: remove decode_end() function fate: run aref and vref as regular tests mpegaudio: sanitise compute_antialias_* names mpeg12: add slice-threading checks to slice-threading initializers. h264: copy pixel_shift between slice threading contexts. mdec: enable frame-level multithreading. mdec.c: fix overread. Conflicts: libavcodec/aacdec.c libavcodec/ac3dec.c libavcodec/avcodec.h libavcodec/dca.c libavcodec/h264.c libavcodec/mdec.c libavcodec/mpeg12.c libavcodec/options.c libavcodec/version.h libavcodec/vorbisdec.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
1470 lines
53 KiB
C
1470 lines
53 KiB
C
/*
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* AC-3 Audio Decoder
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* This code was developed as part of Google Summer of Code 2006.
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* E-AC-3 support was added as part of Google Summer of Code 2007.
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*
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* Copyright (c) 2006 Kartikey Mahendra BHATT (bhattkm at gmail dot com)
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* Copyright (c) 2007-2008 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
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* Copyright (c) 2007 Justin Ruggles <justin.ruggles@gmail.com>
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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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#include <stdio.h>
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#include <stddef.h>
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#include <math.h>
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#include <string.h>
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#include "libavutil/crc.h"
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#include "internal.h"
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#include "aac_ac3_parser.h"
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#include "ac3_parser.h"
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#include "ac3dec.h"
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#include "ac3dec_data.h"
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#include "kbdwin.h"
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/**
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* table for ungrouping 3 values in 7 bits.
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* used for exponents and bap=2 mantissas
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*/
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static uint8_t ungroup_3_in_7_bits_tab[128][3];
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/** tables for ungrouping mantissas */
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static int b1_mantissas[32][3];
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static int b2_mantissas[128][3];
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static int b3_mantissas[8];
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static int b4_mantissas[128][2];
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static int b5_mantissas[16];
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/**
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* Quantization table: levels for symmetric. bits for asymmetric.
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* reference: Table 7.18 Mapping of bap to Quantizer
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*/
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static const uint8_t quantization_tab[16] = {
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0, 3, 5, 7, 11, 15,
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5, 6, 7, 8, 9, 10, 11, 12, 14, 16
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};
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/** dynamic range table. converts codes to scale factors. */
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static float dynamic_range_tab[256];
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/** Adjustments in dB gain */
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static const float gain_levels[9] = {
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LEVEL_PLUS_3DB,
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LEVEL_PLUS_1POINT5DB,
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LEVEL_ONE,
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LEVEL_MINUS_1POINT5DB,
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LEVEL_MINUS_3DB,
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LEVEL_MINUS_4POINT5DB,
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LEVEL_MINUS_6DB,
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LEVEL_ZERO,
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LEVEL_MINUS_9DB
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};
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/**
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* Table for center mix levels
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* reference: Section 5.4.2.4 cmixlev
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*/
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static const uint8_t center_levels[4] = { 4, 5, 6, 5 };
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/**
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* Table for surround mix levels
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* reference: Section 5.4.2.5 surmixlev
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*/
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static const uint8_t surround_levels[4] = { 4, 6, 7, 6 };
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/**
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* Table for default stereo downmixing coefficients
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* reference: Section 7.8.2 Downmixing Into Two Channels
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*/
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static const uint8_t ac3_default_coeffs[8][5][2] = {
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{ { 2, 7 }, { 7, 2 }, },
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{ { 4, 4 }, },
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{ { 2, 7 }, { 7, 2 }, },
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{ { 2, 7 }, { 5, 5 }, { 7, 2 }, },
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{ { 2, 7 }, { 7, 2 }, { 6, 6 }, },
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{ { 2, 7 }, { 5, 5 }, { 7, 2 }, { 8, 8 }, },
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{ { 2, 7 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, },
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{ { 2, 7 }, { 5, 5 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, },
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};
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/**
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* Symmetrical Dequantization
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* reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantization
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* Tables 7.19 to 7.23
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*/
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static inline int
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symmetric_dequant(int code, int levels)
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{
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return ((code - (levels >> 1)) << 24) / levels;
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}
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/*
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* Initialize tables at runtime.
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*/
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static av_cold void ac3_tables_init(void)
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{
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int i;
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/* generate table for ungrouping 3 values in 7 bits
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reference: Section 7.1.3 Exponent Decoding */
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for(i=0; i<128; i++) {
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ungroup_3_in_7_bits_tab[i][0] = i / 25;
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ungroup_3_in_7_bits_tab[i][1] = (i % 25) / 5;
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ungroup_3_in_7_bits_tab[i][2] = (i % 25) % 5;
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}
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/* generate grouped mantissa tables
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reference: Section 7.3.5 Ungrouping of Mantissas */
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for(i=0; i<32; i++) {
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/* bap=1 mantissas */
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b1_mantissas[i][0] = symmetric_dequant(ff_ac3_ungroup_3_in_5_bits_tab[i][0], 3);
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b1_mantissas[i][1] = symmetric_dequant(ff_ac3_ungroup_3_in_5_bits_tab[i][1], 3);
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b1_mantissas[i][2] = symmetric_dequant(ff_ac3_ungroup_3_in_5_bits_tab[i][2], 3);
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}
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for(i=0; i<128; i++) {
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/* bap=2 mantissas */
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b2_mantissas[i][0] = symmetric_dequant(ungroup_3_in_7_bits_tab[i][0], 5);
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b2_mantissas[i][1] = symmetric_dequant(ungroup_3_in_7_bits_tab[i][1], 5);
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b2_mantissas[i][2] = symmetric_dequant(ungroup_3_in_7_bits_tab[i][2], 5);
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/* bap=4 mantissas */
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b4_mantissas[i][0] = symmetric_dequant(i / 11, 11);
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b4_mantissas[i][1] = symmetric_dequant(i % 11, 11);
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}
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/* generate ungrouped mantissa tables
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reference: Tables 7.21 and 7.23 */
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for(i=0; i<7; i++) {
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/* bap=3 mantissas */
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b3_mantissas[i] = symmetric_dequant(i, 7);
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}
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for(i=0; i<15; i++) {
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/* bap=5 mantissas */
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b5_mantissas[i] = symmetric_dequant(i, 15);
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}
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/* generate dynamic range table
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reference: Section 7.7.1 Dynamic Range Control */
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for(i=0; i<256; i++) {
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int v = (i >> 5) - ((i >> 7) << 3) - 5;
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dynamic_range_tab[i] = powf(2.0f, v) * ((i & 0x1F) | 0x20);
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}
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}
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/**
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* AVCodec initialization
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*/
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static av_cold int ac3_decode_init(AVCodecContext *avctx)
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{
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AC3DecodeContext *s = avctx->priv_data;
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s->avctx = avctx;
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ff_ac3_common_init();
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ac3_tables_init();
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ff_mdct_init(&s->imdct_256, 8, 1, 1.0);
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ff_mdct_init(&s->imdct_512, 9, 1, 1.0);
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ff_kbd_window_init(s->window, 5.0, 256);
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dsputil_init(&s->dsp, avctx);
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ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
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ff_fmt_convert_init(&s->fmt_conv, avctx);
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av_lfg_init(&s->dith_state, 0);
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/* set scale value for float to int16 conversion */
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if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
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s->mul_bias = 1.0f;
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
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} else {
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s->mul_bias = 32767.0f;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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}
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/* allow downmixing to stereo or mono */
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if (avctx->channels > 0 && avctx->request_channels > 0 &&
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avctx->request_channels < avctx->channels &&
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avctx->request_channels <= 2) {
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avctx->channels = avctx->request_channels;
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}
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s->downmixed = 1;
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return 0;
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}
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/**
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* Parse the 'sync info' and 'bit stream info' from the AC-3 bitstream.
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* GetBitContext within AC3DecodeContext must point to
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* the start of the synchronized AC-3 bitstream.
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*/
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static int ac3_parse_header(AC3DecodeContext *s)
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{
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GetBitContext *gbc = &s->gbc;
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int i;
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/* read the rest of the bsi. read twice for dual mono mode. */
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i = !(s->channel_mode);
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do {
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skip_bits(gbc, 5); // skip dialog normalization
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if (get_bits1(gbc))
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skip_bits(gbc, 8); //skip compression
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if (get_bits1(gbc))
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skip_bits(gbc, 8); //skip language code
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if (get_bits1(gbc))
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skip_bits(gbc, 7); //skip audio production information
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} while (i--);
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skip_bits(gbc, 2); //skip copyright bit and original bitstream bit
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/* skip the timecodes (or extra bitstream information for Alternate Syntax)
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TODO: read & use the xbsi1 downmix levels */
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if (get_bits1(gbc))
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skip_bits(gbc, 14); //skip timecode1 / xbsi1
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if (get_bits1(gbc))
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skip_bits(gbc, 14); //skip timecode2 / xbsi2
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/* skip additional bitstream info */
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if (get_bits1(gbc)) {
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i = get_bits(gbc, 6);
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do {
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skip_bits(gbc, 8);
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} while(i--);
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}
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return 0;
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}
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/**
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* Common function to parse AC-3 or E-AC-3 frame header
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*/
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static int parse_frame_header(AC3DecodeContext *s)
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{
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AC3HeaderInfo hdr;
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int err;
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err = ff_ac3_parse_header(&s->gbc, &hdr);
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if(err)
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return err;
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/* get decoding parameters from header info */
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s->bit_alloc_params.sr_code = hdr.sr_code;
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s->bitstream_mode = hdr.bitstream_mode;
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s->channel_mode = hdr.channel_mode;
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s->channel_layout = hdr.channel_layout;
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s->lfe_on = hdr.lfe_on;
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s->bit_alloc_params.sr_shift = hdr.sr_shift;
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s->sample_rate = hdr.sample_rate;
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s->bit_rate = hdr.bit_rate;
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s->channels = hdr.channels;
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s->fbw_channels = s->channels - s->lfe_on;
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s->lfe_ch = s->fbw_channels + 1;
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s->frame_size = hdr.frame_size;
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s->center_mix_level = hdr.center_mix_level;
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s->surround_mix_level = hdr.surround_mix_level;
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s->num_blocks = hdr.num_blocks;
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s->frame_type = hdr.frame_type;
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s->substreamid = hdr.substreamid;
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if(s->lfe_on) {
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s->start_freq[s->lfe_ch] = 0;
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s->end_freq[s->lfe_ch] = 7;
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s->num_exp_groups[s->lfe_ch] = 2;
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s->channel_in_cpl[s->lfe_ch] = 0;
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}
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if (hdr.bitstream_id <= 10) {
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s->eac3 = 0;
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s->snr_offset_strategy = 2;
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s->block_switch_syntax = 1;
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s->dither_flag_syntax = 1;
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s->bit_allocation_syntax = 1;
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s->fast_gain_syntax = 0;
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s->first_cpl_leak = 0;
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s->dba_syntax = 1;
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s->skip_syntax = 1;
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memset(s->channel_uses_aht, 0, sizeof(s->channel_uses_aht));
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return ac3_parse_header(s);
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} else if (CONFIG_EAC3_DECODER) {
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s->eac3 = 1;
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return ff_eac3_parse_header(s);
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} else {
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av_log(s->avctx, AV_LOG_ERROR, "E-AC-3 support not compiled in\n");
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return -1;
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}
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}
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/**
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* Set stereo downmixing coefficients based on frame header info.
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* reference: Section 7.8.2 Downmixing Into Two Channels
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*/
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static void set_downmix_coeffs(AC3DecodeContext *s)
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{
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int i;
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float cmix = gain_levels[center_levels[s->center_mix_level]];
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float smix = gain_levels[surround_levels[s->surround_mix_level]];
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float norm0, norm1;
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for(i=0; i<s->fbw_channels; i++) {
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s->downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[s->channel_mode][i][0]];
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s->downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[s->channel_mode][i][1]];
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}
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if(s->channel_mode > 1 && s->channel_mode & 1) {
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s->downmix_coeffs[1][0] = s->downmix_coeffs[1][1] = cmix;
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}
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if(s->channel_mode == AC3_CHMODE_2F1R || s->channel_mode == AC3_CHMODE_3F1R) {
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int nf = s->channel_mode - 2;
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s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf][1] = smix * LEVEL_MINUS_3DB;
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}
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if(s->channel_mode == AC3_CHMODE_2F2R || s->channel_mode == AC3_CHMODE_3F2R) {
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int nf = s->channel_mode - 4;
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s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf+1][1] = smix;
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}
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/* renormalize */
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norm0 = norm1 = 0.0;
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for(i=0; i<s->fbw_channels; i++) {
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norm0 += s->downmix_coeffs[i][0];
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norm1 += s->downmix_coeffs[i][1];
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}
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norm0 = 1.0f / norm0;
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norm1 = 1.0f / norm1;
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for(i=0; i<s->fbw_channels; i++) {
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s->downmix_coeffs[i][0] *= norm0;
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s->downmix_coeffs[i][1] *= norm1;
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}
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if(s->output_mode == AC3_CHMODE_MONO) {
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for(i=0; i<s->fbw_channels; i++)
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s->downmix_coeffs[i][0] = (s->downmix_coeffs[i][0] + s->downmix_coeffs[i][1]) * LEVEL_MINUS_3DB;
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}
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}
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/**
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* Decode the grouped exponents according to exponent strategy.
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* reference: Section 7.1.3 Exponent Decoding
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*/
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static int decode_exponents(GetBitContext *gbc, int exp_strategy, int ngrps,
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uint8_t absexp, int8_t *dexps)
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{
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int i, j, grp, group_size;
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int dexp[256];
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int expacc, prevexp;
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/* unpack groups */
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group_size = exp_strategy + (exp_strategy == EXP_D45);
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for(grp=0,i=0; grp<ngrps; grp++) {
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expacc = get_bits(gbc, 7);
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dexp[i++] = ungroup_3_in_7_bits_tab[expacc][0];
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dexp[i++] = ungroup_3_in_7_bits_tab[expacc][1];
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dexp[i++] = ungroup_3_in_7_bits_tab[expacc][2];
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}
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/* convert to absolute exps and expand groups */
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prevexp = absexp;
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for(i=0,j=0; i<ngrps*3; i++) {
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prevexp += dexp[i] - 2;
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if (prevexp > 24U)
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return -1;
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switch (group_size) {
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case 4: dexps[j++] = prevexp;
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dexps[j++] = prevexp;
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case 2: dexps[j++] = prevexp;
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case 1: dexps[j++] = prevexp;
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}
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}
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return 0;
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}
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/**
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* Generate transform coefficients for each coupled channel in the coupling
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* range using the coupling coefficients and coupling coordinates.
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* reference: Section 7.4.3 Coupling Coordinate Format
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*/
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static void calc_transform_coeffs_cpl(AC3DecodeContext *s)
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{
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int bin, band, ch;
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bin = s->start_freq[CPL_CH];
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for (band = 0; band < s->num_cpl_bands; band++) {
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int band_start = bin;
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int band_end = bin + s->cpl_band_sizes[band];
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for (ch = 1; ch <= s->fbw_channels; ch++) {
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if (s->channel_in_cpl[ch]) {
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int cpl_coord = s->cpl_coords[ch][band] << 5;
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for (bin = band_start; bin < band_end; bin++) {
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s->fixed_coeffs[ch][bin] = MULH(s->fixed_coeffs[CPL_CH][bin] << 4, cpl_coord);
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}
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if (ch == 2 && s->phase_flags[band]) {
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for (bin = band_start; bin < band_end; bin++)
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s->fixed_coeffs[2][bin] = -s->fixed_coeffs[2][bin];
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}
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}
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}
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bin = band_end;
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}
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}
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/**
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* Grouped mantissas for 3-level 5-level and 11-level quantization
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*/
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typedef struct {
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int b1_mant[2];
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int b2_mant[2];
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int b4_mant;
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int b1;
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|
int b2;
|
|
int b4;
|
|
} mant_groups;
|
|
|
|
/**
|
|
* Decode the transform coefficients for a particular channel
|
|
* reference: Section 7.3 Quantization and Decoding of Mantissas
|
|
*/
|
|
static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, mant_groups *m)
|
|
{
|
|
int start_freq = s->start_freq[ch_index];
|
|
int end_freq = s->end_freq[ch_index];
|
|
uint8_t *baps = s->bap[ch_index];
|
|
int8_t *exps = s->dexps[ch_index];
|
|
int *coeffs = s->fixed_coeffs[ch_index];
|
|
int dither = (ch_index == CPL_CH) || s->dither_flag[ch_index];
|
|
GetBitContext *gbc = &s->gbc;
|
|
int freq;
|
|
|
|
for(freq = start_freq; freq < end_freq; freq++){
|
|
int bap = baps[freq];
|
|
int mantissa;
|
|
switch(bap){
|
|
case 0:
|
|
if (dither)
|
|
mantissa = (av_lfg_get(&s->dith_state) & 0x7FFFFF) - 0x400000;
|
|
else
|
|
mantissa = 0;
|
|
break;
|
|
case 1:
|
|
if(m->b1){
|
|
m->b1--;
|
|
mantissa = m->b1_mant[m->b1];
|
|
}
|
|
else{
|
|
int bits = get_bits(gbc, 5);
|
|
mantissa = b1_mantissas[bits][0];
|
|
m->b1_mant[1] = b1_mantissas[bits][1];
|
|
m->b1_mant[0] = b1_mantissas[bits][2];
|
|
m->b1 = 2;
|
|
}
|
|
break;
|
|
case 2:
|
|
if(m->b2){
|
|
m->b2--;
|
|
mantissa = m->b2_mant[m->b2];
|
|
}
|
|
else{
|
|
int bits = get_bits(gbc, 7);
|
|
mantissa = b2_mantissas[bits][0];
|
|
m->b2_mant[1] = b2_mantissas[bits][1];
|
|
m->b2_mant[0] = b2_mantissas[bits][2];
|
|
m->b2 = 2;
|
|
}
|
|
break;
|
|
case 3:
|
|
mantissa = b3_mantissas[get_bits(gbc, 3)];
|
|
break;
|
|
case 4:
|
|
if(m->b4){
|
|
m->b4 = 0;
|
|
mantissa = m->b4_mant;
|
|
}
|
|
else{
|
|
int bits = get_bits(gbc, 7);
|
|
mantissa = b4_mantissas[bits][0];
|
|
m->b4_mant = b4_mantissas[bits][1];
|
|
m->b4 = 1;
|
|
}
|
|
break;
|
|
case 5:
|
|
mantissa = b5_mantissas[get_bits(gbc, 4)];
|
|
break;
|
|
default: /* 6 to 15 */
|
|
mantissa = get_bits(gbc, quantization_tab[bap]);
|
|
/* Shift mantissa and sign-extend it. */
|
|
mantissa = (mantissa << (32-quantization_tab[bap]))>>8;
|
|
break;
|
|
}
|
|
coeffs[freq] = mantissa >> exps[freq];
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Remove random dithering from coupling range coefficients with zero-bit
|
|
* mantissas for coupled channels which do not use dithering.
|
|
* reference: Section 7.3.4 Dither for Zero Bit Mantissas (bap=0)
|
|
*/
|
|
static void remove_dithering(AC3DecodeContext *s) {
|
|
int ch, i;
|
|
|
|
for(ch=1; ch<=s->fbw_channels; ch++) {
|
|
if(!s->dither_flag[ch] && s->channel_in_cpl[ch]) {
|
|
for(i = s->start_freq[CPL_CH]; i<s->end_freq[CPL_CH]; i++) {
|
|
if(!s->bap[CPL_CH][i])
|
|
s->fixed_coeffs[ch][i] = 0;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static void decode_transform_coeffs_ch(AC3DecodeContext *s, int blk, int ch,
|
|
mant_groups *m)
|
|
{
|
|
if (!s->channel_uses_aht[ch]) {
|
|
ac3_decode_transform_coeffs_ch(s, ch, m);
|
|
} else {
|
|
/* if AHT is used, mantissas for all blocks are encoded in the first
|
|
block of the frame. */
|
|
int bin;
|
|
if (!blk && CONFIG_EAC3_DECODER)
|
|
ff_eac3_decode_transform_coeffs_aht_ch(s, ch);
|
|
for (bin = s->start_freq[ch]; bin < s->end_freq[ch]; bin++) {
|
|
s->fixed_coeffs[ch][bin] = s->pre_mantissa[ch][bin][blk] >> s->dexps[ch][bin];
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Decode the transform coefficients.
|
|
*/
|
|
static void decode_transform_coeffs(AC3DecodeContext *s, int blk)
|
|
{
|
|
int ch, end;
|
|
int got_cplchan = 0;
|
|
mant_groups m;
|
|
|
|
m.b1 = m.b2 = m.b4 = 0;
|
|
|
|
for (ch = 1; ch <= s->channels; ch++) {
|
|
/* transform coefficients for full-bandwidth channel */
|
|
decode_transform_coeffs_ch(s, blk, ch, &m);
|
|
/* tranform coefficients for coupling channel come right after the
|
|
coefficients for the first coupled channel*/
|
|
if (s->channel_in_cpl[ch]) {
|
|
if (!got_cplchan) {
|
|
decode_transform_coeffs_ch(s, blk, CPL_CH, &m);
|
|
calc_transform_coeffs_cpl(s);
|
|
got_cplchan = 1;
|
|
}
|
|
end = s->end_freq[CPL_CH];
|
|
} else {
|
|
end = s->end_freq[ch];
|
|
}
|
|
do
|
|
s->fixed_coeffs[ch][end] = 0;
|
|
while(++end < 256);
|
|
}
|
|
|
|
/* zero the dithered coefficients for appropriate channels */
|
|
remove_dithering(s);
|
|
}
|
|
|
|
/**
|
|
* Stereo rematrixing.
|
|
* reference: Section 7.5.4 Rematrixing : Decoding Technique
|
|
*/
|
|
static void do_rematrixing(AC3DecodeContext *s)
|
|
{
|
|
int bnd, i;
|
|
int end, bndend;
|
|
|
|
end = FFMIN(s->end_freq[1], s->end_freq[2]);
|
|
|
|
for(bnd=0; bnd<s->num_rematrixing_bands; bnd++) {
|
|
if(s->rematrixing_flags[bnd]) {
|
|
bndend = FFMIN(end, ff_ac3_rematrix_band_tab[bnd+1]);
|
|
for(i=ff_ac3_rematrix_band_tab[bnd]; i<bndend; i++) {
|
|
int tmp0 = s->fixed_coeffs[1][i];
|
|
s->fixed_coeffs[1][i] += s->fixed_coeffs[2][i];
|
|
s->fixed_coeffs[2][i] = tmp0 - s->fixed_coeffs[2][i];
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Inverse MDCT Transform.
|
|
* Convert frequency domain coefficients to time-domain audio samples.
|
|
* reference: Section 7.9.4 Transformation Equations
|
|
*/
|
|
static inline void do_imdct(AC3DecodeContext *s, int channels)
|
|
{
|
|
int ch;
|
|
|
|
for (ch=1; ch<=channels; ch++) {
|
|
if (s->block_switch[ch]) {
|
|
int i;
|
|
float *x = s->tmp_output+128;
|
|
for(i=0; i<128; i++)
|
|
x[i] = s->transform_coeffs[ch][2*i];
|
|
s->imdct_256.imdct_half(&s->imdct_256, s->tmp_output, x);
|
|
s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 128);
|
|
for(i=0; i<128; i++)
|
|
x[i] = s->transform_coeffs[ch][2*i+1];
|
|
s->imdct_256.imdct_half(&s->imdct_256, s->delay[ch-1], x);
|
|
} else {
|
|
s->imdct_512.imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
|
|
s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 128);
|
|
memcpy(s->delay[ch-1], s->tmp_output+128, 128*sizeof(float));
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Downmix the output to mono or stereo.
|
|
*/
|
|
void ff_ac3_downmix_c(float (*samples)[256], float (*matrix)[2], int out_ch, int in_ch, int len)
|
|
{
|
|
int i, j;
|
|
float v0, v1;
|
|
if(out_ch == 2) {
|
|
for(i=0; i<len; i++) {
|
|
v0 = v1 = 0.0f;
|
|
for(j=0; j<in_ch; j++) {
|
|
v0 += samples[j][i] * matrix[j][0];
|
|
v1 += samples[j][i] * matrix[j][1];
|
|
}
|
|
samples[0][i] = v0;
|
|
samples[1][i] = v1;
|
|
}
|
|
} else if(out_ch == 1) {
|
|
for(i=0; i<len; i++) {
|
|
v0 = 0.0f;
|
|
for(j=0; j<in_ch; j++)
|
|
v0 += samples[j][i] * matrix[j][0];
|
|
samples[0][i] = v0;
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Upmix delay samples from stereo to original channel layout.
|
|
*/
|
|
static void ac3_upmix_delay(AC3DecodeContext *s)
|
|
{
|
|
int channel_data_size = sizeof(s->delay[0]);
|
|
switch(s->channel_mode) {
|
|
case AC3_CHMODE_DUALMONO:
|
|
case AC3_CHMODE_STEREO:
|
|
/* upmix mono to stereo */
|
|
memcpy(s->delay[1], s->delay[0], channel_data_size);
|
|
break;
|
|
case AC3_CHMODE_2F2R:
|
|
memset(s->delay[3], 0, channel_data_size);
|
|
case AC3_CHMODE_2F1R:
|
|
memset(s->delay[2], 0, channel_data_size);
|
|
break;
|
|
case AC3_CHMODE_3F2R:
|
|
memset(s->delay[4], 0, channel_data_size);
|
|
case AC3_CHMODE_3F1R:
|
|
memset(s->delay[3], 0, channel_data_size);
|
|
case AC3_CHMODE_3F:
|
|
memcpy(s->delay[2], s->delay[1], channel_data_size);
|
|
memset(s->delay[1], 0, channel_data_size);
|
|
break;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Decode band structure for coupling, spectral extension, or enhanced coupling.
|
|
* The band structure defines how many subbands are in each band. For each
|
|
* subband in the range, 1 means it is combined with the previous band, and 0
|
|
* means that it starts a new band.
|
|
*
|
|
* @param[in] gbc bit reader context
|
|
* @param[in] blk block number
|
|
* @param[in] eac3 flag to indicate E-AC-3
|
|
* @param[in] ecpl flag to indicate enhanced coupling
|
|
* @param[in] start_subband subband number for start of range
|
|
* @param[in] end_subband subband number for end of range
|
|
* @param[in] default_band_struct default band structure table
|
|
* @param[out] num_bands number of bands (optionally NULL)
|
|
* @param[out] band_sizes array containing the number of bins in each band (optionally NULL)
|
|
*/
|
|
static void decode_band_structure(GetBitContext *gbc, int blk, int eac3,
|
|
int ecpl, int start_subband, int end_subband,
|
|
const uint8_t *default_band_struct,
|
|
int *num_bands, uint8_t *band_sizes)
|
|
{
|
|
int subbnd, bnd, n_subbands, n_bands=0;
|
|
uint8_t bnd_sz[22];
|
|
uint8_t coded_band_struct[22];
|
|
const uint8_t *band_struct;
|
|
|
|
n_subbands = end_subband - start_subband;
|
|
|
|
/* decode band structure from bitstream or use default */
|
|
if (!eac3 || get_bits1(gbc)) {
|
|
for (subbnd = 0; subbnd < n_subbands - 1; subbnd++) {
|
|
coded_band_struct[subbnd] = get_bits1(gbc);
|
|
}
|
|
band_struct = coded_band_struct;
|
|
} else if (!blk) {
|
|
band_struct = &default_band_struct[start_subband+1];
|
|
} else {
|
|
/* no change in band structure */
|
|
return;
|
|
}
|
|
|
|
/* calculate number of bands and band sizes based on band structure.
|
|
note that the first 4 subbands in enhanced coupling span only 6 bins
|
|
instead of 12. */
|
|
if (num_bands || band_sizes ) {
|
|
n_bands = n_subbands;
|
|
bnd_sz[0] = ecpl ? 6 : 12;
|
|
for (bnd = 0, subbnd = 1; subbnd < n_subbands; subbnd++) {
|
|
int subbnd_size = (ecpl && subbnd < 4) ? 6 : 12;
|
|
if (band_struct[subbnd-1]) {
|
|
n_bands--;
|
|
bnd_sz[bnd] += subbnd_size;
|
|
} else {
|
|
bnd_sz[++bnd] = subbnd_size;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* set optional output params */
|
|
if (num_bands)
|
|
*num_bands = n_bands;
|
|
if (band_sizes)
|
|
memcpy(band_sizes, bnd_sz, n_bands);
|
|
}
|
|
|
|
/**
|
|
* Decode a single audio block from the AC-3 bitstream.
|
|
*/
|
|
static int decode_audio_block(AC3DecodeContext *s, int blk)
|
|
{
|
|
int fbw_channels = s->fbw_channels;
|
|
int channel_mode = s->channel_mode;
|
|
int i, bnd, seg, ch;
|
|
int different_transforms;
|
|
int downmix_output;
|
|
int cpl_in_use;
|
|
GetBitContext *gbc = &s->gbc;
|
|
uint8_t bit_alloc_stages[AC3_MAX_CHANNELS];
|
|
|
|
memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS);
|
|
|
|
/* block switch flags */
|
|
different_transforms = 0;
|
|
if (s->block_switch_syntax) {
|
|
for (ch = 1; ch <= fbw_channels; ch++) {
|
|
s->block_switch[ch] = get_bits1(gbc);
|
|
if(ch > 1 && s->block_switch[ch] != s->block_switch[1])
|
|
different_transforms = 1;
|
|
}
|
|
}
|
|
|
|
/* dithering flags */
|
|
if (s->dither_flag_syntax) {
|
|
for (ch = 1; ch <= fbw_channels; ch++) {
|
|
s->dither_flag[ch] = get_bits1(gbc);
|
|
}
|
|
}
|
|
|
|
/* dynamic range */
|
|
i = !(s->channel_mode);
|
|
do {
|
|
if(get_bits1(gbc)) {
|
|
s->dynamic_range[i] = ((dynamic_range_tab[get_bits(gbc, 8)]-1.0) *
|
|
s->avctx->drc_scale)+1.0;
|
|
} else if(blk == 0) {
|
|
s->dynamic_range[i] = 1.0f;
|
|
}
|
|
} while(i--);
|
|
|
|
/* spectral extension strategy */
|
|
if (s->eac3 && (!blk || get_bits1(gbc))) {
|
|
s->spx_in_use = get_bits1(gbc);
|
|
if (s->spx_in_use) {
|
|
int dst_start_freq, dst_end_freq, src_start_freq,
|
|
start_subband, end_subband;
|
|
|
|
/* determine which channels use spx */
|
|
if (s->channel_mode == AC3_CHMODE_MONO) {
|
|
s->channel_uses_spx[1] = 1;
|
|
} else {
|
|
for (ch = 1; ch <= fbw_channels; ch++)
|
|
s->channel_uses_spx[ch] = get_bits1(gbc);
|
|
}
|
|
|
|
/* get the frequency bins of the spx copy region and the spx start
|
|
and end subbands */
|
|
dst_start_freq = get_bits(gbc, 2);
|
|
start_subband = get_bits(gbc, 3) + 2;
|
|
if (start_subband > 7)
|
|
start_subband += start_subband - 7;
|
|
end_subband = get_bits(gbc, 3) + 5;
|
|
if (end_subband > 7)
|
|
end_subband += end_subband - 7;
|
|
dst_start_freq = dst_start_freq * 12 + 25;
|
|
src_start_freq = start_subband * 12 + 25;
|
|
dst_end_freq = end_subband * 12 + 25;
|
|
|
|
/* check validity of spx ranges */
|
|
if (start_subband >= end_subband) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "invalid spectral extension "
|
|
"range (%d >= %d)\n", start_subband, end_subband);
|
|
return -1;
|
|
}
|
|
if (dst_start_freq >= src_start_freq) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "invalid spectral extension "
|
|
"copy start bin (%d >= %d)\n", dst_start_freq, src_start_freq);
|
|
return -1;
|
|
}
|
|
|
|
s->spx_dst_start_freq = dst_start_freq;
|
|
s->spx_src_start_freq = src_start_freq;
|
|
s->spx_dst_end_freq = dst_end_freq;
|
|
|
|
decode_band_structure(gbc, blk, s->eac3, 0,
|
|
start_subband, end_subband,
|
|
ff_eac3_default_spx_band_struct,
|
|
&s->num_spx_bands,
|
|
s->spx_band_sizes);
|
|
} else {
|
|
for (ch = 1; ch <= fbw_channels; ch++) {
|
|
s->channel_uses_spx[ch] = 0;
|
|
s->first_spx_coords[ch] = 1;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* spectral extension coordinates */
|
|
if (s->spx_in_use) {
|
|
for (ch = 1; ch <= fbw_channels; ch++) {
|
|
if (s->channel_uses_spx[ch]) {
|
|
if (s->first_spx_coords[ch] || get_bits1(gbc)) {
|
|
float spx_blend;
|
|
int bin, master_spx_coord;
|
|
|
|
s->first_spx_coords[ch] = 0;
|
|
spx_blend = get_bits(gbc, 5) * (1.0f/32);
|
|
master_spx_coord = get_bits(gbc, 2) * 3;
|
|
|
|
bin = s->spx_src_start_freq;
|
|
for (bnd = 0; bnd < s->num_spx_bands; bnd++) {
|
|
int bandsize;
|
|
int spx_coord_exp, spx_coord_mant;
|
|
float nratio, sblend, nblend, spx_coord;
|
|
|
|
/* calculate blending factors */
|
|
bandsize = s->spx_band_sizes[bnd];
|
|
nratio = ((float)((bin + (bandsize >> 1))) / s->spx_dst_end_freq) - spx_blend;
|
|
nratio = av_clipf(nratio, 0.0f, 1.0f);
|
|
nblend = sqrtf(3.0f * nratio); // noise is scaled by sqrt(3) to give unity variance
|
|
sblend = sqrtf(1.0f - nratio);
|
|
bin += bandsize;
|
|
|
|
/* decode spx coordinates */
|
|
spx_coord_exp = get_bits(gbc, 4);
|
|
spx_coord_mant = get_bits(gbc, 2);
|
|
if (spx_coord_exp == 15) spx_coord_mant <<= 1;
|
|
else spx_coord_mant += 4;
|
|
spx_coord_mant <<= (25 - spx_coord_exp - master_spx_coord);
|
|
spx_coord = spx_coord_mant * (1.0f/(1<<23));
|
|
|
|
/* multiply noise and signal blending factors by spx coordinate */
|
|
s->spx_noise_blend [ch][bnd] = nblend * spx_coord;
|
|
s->spx_signal_blend[ch][bnd] = sblend * spx_coord;
|
|
}
|
|
}
|
|
} else {
|
|
s->first_spx_coords[ch] = 1;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* coupling strategy */
|
|
if (s->eac3 ? s->cpl_strategy_exists[blk] : get_bits1(gbc)) {
|
|
memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
|
|
if (!s->eac3)
|
|
s->cpl_in_use[blk] = get_bits1(gbc);
|
|
if (s->cpl_in_use[blk]) {
|
|
/* coupling in use */
|
|
int cpl_start_subband, cpl_end_subband;
|
|
|
|
if (channel_mode < AC3_CHMODE_STEREO) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "coupling not allowed in mono or dual-mono\n");
|
|
return -1;
|
|
}
|
|
|
|
/* check for enhanced coupling */
|
|
if (s->eac3 && get_bits1(gbc)) {
|
|
/* TODO: parse enhanced coupling strategy info */
|
|
av_log_missing_feature(s->avctx, "Enhanced coupling", 1);
|
|
return -1;
|
|
}
|
|
|
|
/* determine which channels are coupled */
|
|
if (s->eac3 && s->channel_mode == AC3_CHMODE_STEREO) {
|
|
s->channel_in_cpl[1] = 1;
|
|
s->channel_in_cpl[2] = 1;
|
|
} else {
|
|
for (ch = 1; ch <= fbw_channels; ch++)
|
|
s->channel_in_cpl[ch] = get_bits1(gbc);
|
|
}
|
|
|
|
/* phase flags in use */
|
|
if (channel_mode == AC3_CHMODE_STEREO)
|
|
s->phase_flags_in_use = get_bits1(gbc);
|
|
|
|
/* coupling frequency range */
|
|
cpl_start_subband = get_bits(gbc, 4);
|
|
cpl_end_subband = s->spx_in_use ? (s->spx_src_start_freq - 37) / 12 :
|
|
get_bits(gbc, 4) + 3;
|
|
if (cpl_start_subband >= cpl_end_subband) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "invalid coupling range (%d >= %d)\n",
|
|
cpl_start_subband, cpl_end_subband);
|
|
return -1;
|
|
}
|
|
s->start_freq[CPL_CH] = cpl_start_subband * 12 + 37;
|
|
s->end_freq[CPL_CH] = cpl_end_subband * 12 + 37;
|
|
|
|
decode_band_structure(gbc, blk, s->eac3, 0, cpl_start_subband,
|
|
cpl_end_subband,
|
|
ff_eac3_default_cpl_band_struct,
|
|
&s->num_cpl_bands, s->cpl_band_sizes);
|
|
} else {
|
|
/* coupling not in use */
|
|
for (ch = 1; ch <= fbw_channels; ch++) {
|
|
s->channel_in_cpl[ch] = 0;
|
|
s->first_cpl_coords[ch] = 1;
|
|
}
|
|
s->first_cpl_leak = s->eac3;
|
|
s->phase_flags_in_use = 0;
|
|
}
|
|
} else if (!s->eac3) {
|
|
if(!blk) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "new coupling strategy must be present in block 0\n");
|
|
return -1;
|
|
} else {
|
|
s->cpl_in_use[blk] = s->cpl_in_use[blk-1];
|
|
}
|
|
}
|
|
cpl_in_use = s->cpl_in_use[blk];
|
|
|
|
/* coupling coordinates */
|
|
if (cpl_in_use) {
|
|
int cpl_coords_exist = 0;
|
|
|
|
for (ch = 1; ch <= fbw_channels; ch++) {
|
|
if (s->channel_in_cpl[ch]) {
|
|
if ((s->eac3 && s->first_cpl_coords[ch]) || get_bits1(gbc)) {
|
|
int master_cpl_coord, cpl_coord_exp, cpl_coord_mant;
|
|
s->first_cpl_coords[ch] = 0;
|
|
cpl_coords_exist = 1;
|
|
master_cpl_coord = 3 * get_bits(gbc, 2);
|
|
for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
|
|
cpl_coord_exp = get_bits(gbc, 4);
|
|
cpl_coord_mant = get_bits(gbc, 4);
|
|
if (cpl_coord_exp == 15)
|
|
s->cpl_coords[ch][bnd] = cpl_coord_mant << 22;
|
|
else
|
|
s->cpl_coords[ch][bnd] = (cpl_coord_mant + 16) << 21;
|
|
s->cpl_coords[ch][bnd] >>= (cpl_coord_exp + master_cpl_coord);
|
|
}
|
|
} else if (!blk) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "new coupling coordinates must be present in block 0\n");
|
|
return -1;
|
|
}
|
|
} else {
|
|
/* channel not in coupling */
|
|
s->first_cpl_coords[ch] = 1;
|
|
}
|
|
}
|
|
/* phase flags */
|
|
if (channel_mode == AC3_CHMODE_STEREO && cpl_coords_exist) {
|
|
for (bnd = 0; bnd < s->num_cpl_bands; bnd++) {
|
|
s->phase_flags[bnd] = s->phase_flags_in_use? get_bits1(gbc) : 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* stereo rematrixing strategy and band structure */
|
|
if (channel_mode == AC3_CHMODE_STEREO) {
|
|
if ((s->eac3 && !blk) || get_bits1(gbc)) {
|
|
s->num_rematrixing_bands = 4;
|
|
if (cpl_in_use && s->start_freq[CPL_CH] <= 61) {
|
|
s->num_rematrixing_bands -= 1 + (s->start_freq[CPL_CH] == 37);
|
|
} else if (s->spx_in_use && s->spx_src_start_freq <= 61) {
|
|
s->num_rematrixing_bands--;
|
|
}
|
|
for(bnd=0; bnd<s->num_rematrixing_bands; bnd++)
|
|
s->rematrixing_flags[bnd] = get_bits1(gbc);
|
|
} else if (!blk) {
|
|
av_log(s->avctx, AV_LOG_WARNING, "Warning: new rematrixing strategy not present in block 0\n");
|
|
s->num_rematrixing_bands = 0;
|
|
}
|
|
}
|
|
|
|
/* exponent strategies for each channel */
|
|
for (ch = !cpl_in_use; ch <= s->channels; ch++) {
|
|
if (!s->eac3)
|
|
s->exp_strategy[blk][ch] = get_bits(gbc, 2 - (ch == s->lfe_ch));
|
|
if(s->exp_strategy[blk][ch] != EXP_REUSE)
|
|
bit_alloc_stages[ch] = 3;
|
|
}
|
|
|
|
/* channel bandwidth */
|
|
for (ch = 1; ch <= fbw_channels; ch++) {
|
|
s->start_freq[ch] = 0;
|
|
if (s->exp_strategy[blk][ch] != EXP_REUSE) {
|
|
int group_size;
|
|
int prev = s->end_freq[ch];
|
|
if (s->channel_in_cpl[ch])
|
|
s->end_freq[ch] = s->start_freq[CPL_CH];
|
|
else if (s->channel_uses_spx[ch])
|
|
s->end_freq[ch] = s->spx_src_start_freq;
|
|
else {
|
|
int bandwidth_code = get_bits(gbc, 6);
|
|
if (bandwidth_code > 60) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "bandwidth code = %d > 60\n", bandwidth_code);
|
|
return -1;
|
|
}
|
|
s->end_freq[ch] = bandwidth_code * 3 + 73;
|
|
}
|
|
group_size = 3 << (s->exp_strategy[blk][ch] - 1);
|
|
s->num_exp_groups[ch] = (s->end_freq[ch]+group_size-4) / group_size;
|
|
if(blk > 0 && s->end_freq[ch] != prev)
|
|
memset(bit_alloc_stages, 3, AC3_MAX_CHANNELS);
|
|
}
|
|
}
|
|
if (cpl_in_use && s->exp_strategy[blk][CPL_CH] != EXP_REUSE) {
|
|
s->num_exp_groups[CPL_CH] = (s->end_freq[CPL_CH] - s->start_freq[CPL_CH]) /
|
|
(3 << (s->exp_strategy[blk][CPL_CH] - 1));
|
|
}
|
|
|
|
/* decode exponents for each channel */
|
|
for (ch = !cpl_in_use; ch <= s->channels; ch++) {
|
|
if (s->exp_strategy[blk][ch] != EXP_REUSE) {
|
|
s->dexps[ch][0] = get_bits(gbc, 4) << !ch;
|
|
if (decode_exponents(gbc, s->exp_strategy[blk][ch],
|
|
s->num_exp_groups[ch], s->dexps[ch][0],
|
|
&s->dexps[ch][s->start_freq[ch]+!!ch])) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "exponent out-of-range\n");
|
|
return -1;
|
|
}
|
|
if(ch != CPL_CH && ch != s->lfe_ch)
|
|
skip_bits(gbc, 2); /* skip gainrng */
|
|
}
|
|
}
|
|
|
|
/* bit allocation information */
|
|
if (s->bit_allocation_syntax) {
|
|
if (get_bits1(gbc)) {
|
|
s->bit_alloc_params.slow_decay = ff_ac3_slow_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift;
|
|
s->bit_alloc_params.fast_decay = ff_ac3_fast_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift;
|
|
s->bit_alloc_params.slow_gain = ff_ac3_slow_gain_tab[get_bits(gbc, 2)];
|
|
s->bit_alloc_params.db_per_bit = ff_ac3_db_per_bit_tab[get_bits(gbc, 2)];
|
|
s->bit_alloc_params.floor = ff_ac3_floor_tab[get_bits(gbc, 3)];
|
|
for(ch=!cpl_in_use; ch<=s->channels; ch++)
|
|
bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
|
|
} else if (!blk) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "new bit allocation info must be present in block 0\n");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
/* signal-to-noise ratio offsets and fast gains (signal-to-mask ratios) */
|
|
if(!s->eac3 || !blk){
|
|
if(s->snr_offset_strategy && get_bits1(gbc)) {
|
|
int snr = 0;
|
|
int csnr;
|
|
csnr = (get_bits(gbc, 6) - 15) << 4;
|
|
for (i = ch = !cpl_in_use; ch <= s->channels; ch++) {
|
|
/* snr offset */
|
|
if (ch == i || s->snr_offset_strategy == 2)
|
|
snr = (csnr + get_bits(gbc, 4)) << 2;
|
|
/* run at least last bit allocation stage if snr offset changes */
|
|
if(blk && s->snr_offset[ch] != snr) {
|
|
bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 1);
|
|
}
|
|
s->snr_offset[ch] = snr;
|
|
|
|
/* fast gain (normal AC-3 only) */
|
|
if (!s->eac3) {
|
|
int prev = s->fast_gain[ch];
|
|
s->fast_gain[ch] = ff_ac3_fast_gain_tab[get_bits(gbc, 3)];
|
|
/* run last 2 bit allocation stages if fast gain changes */
|
|
if(blk && prev != s->fast_gain[ch])
|
|
bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
|
|
}
|
|
}
|
|
} else if (!s->eac3 && !blk) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "new snr offsets must be present in block 0\n");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
/* fast gain (E-AC-3 only) */
|
|
if (s->fast_gain_syntax && get_bits1(gbc)) {
|
|
for (ch = !cpl_in_use; ch <= s->channels; ch++) {
|
|
int prev = s->fast_gain[ch];
|
|
s->fast_gain[ch] = ff_ac3_fast_gain_tab[get_bits(gbc, 3)];
|
|
/* run last 2 bit allocation stages if fast gain changes */
|
|
if(blk && prev != s->fast_gain[ch])
|
|
bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
|
|
}
|
|
} else if (s->eac3 && !blk) {
|
|
for (ch = !cpl_in_use; ch <= s->channels; ch++)
|
|
s->fast_gain[ch] = ff_ac3_fast_gain_tab[4];
|
|
}
|
|
|
|
/* E-AC-3 to AC-3 converter SNR offset */
|
|
if (s->frame_type == EAC3_FRAME_TYPE_INDEPENDENT && get_bits1(gbc)) {
|
|
skip_bits(gbc, 10); // skip converter snr offset
|
|
}
|
|
|
|
/* coupling leak information */
|
|
if (cpl_in_use) {
|
|
if (s->first_cpl_leak || get_bits1(gbc)) {
|
|
int fl = get_bits(gbc, 3);
|
|
int sl = get_bits(gbc, 3);
|
|
/* run last 2 bit allocation stages for coupling channel if
|
|
coupling leak changes */
|
|
if(blk && (fl != s->bit_alloc_params.cpl_fast_leak ||
|
|
sl != s->bit_alloc_params.cpl_slow_leak)) {
|
|
bit_alloc_stages[CPL_CH] = FFMAX(bit_alloc_stages[CPL_CH], 2);
|
|
}
|
|
s->bit_alloc_params.cpl_fast_leak = fl;
|
|
s->bit_alloc_params.cpl_slow_leak = sl;
|
|
} else if (!s->eac3 && !blk) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "new coupling leak info must be present in block 0\n");
|
|
return -1;
|
|
}
|
|
s->first_cpl_leak = 0;
|
|
}
|
|
|
|
/* delta bit allocation information */
|
|
if (s->dba_syntax && get_bits1(gbc)) {
|
|
/* delta bit allocation exists (strategy) */
|
|
for (ch = !cpl_in_use; ch <= fbw_channels; ch++) {
|
|
s->dba_mode[ch] = get_bits(gbc, 2);
|
|
if (s->dba_mode[ch] == DBA_RESERVED) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "delta bit allocation strategy reserved\n");
|
|
return -1;
|
|
}
|
|
bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
|
|
}
|
|
/* channel delta offset, len and bit allocation */
|
|
for (ch = !cpl_in_use; ch <= fbw_channels; ch++) {
|
|
if (s->dba_mode[ch] == DBA_NEW) {
|
|
s->dba_nsegs[ch] = get_bits(gbc, 3) + 1;
|
|
for (seg = 0; seg < s->dba_nsegs[ch]; seg++) {
|
|
s->dba_offsets[ch][seg] = get_bits(gbc, 5);
|
|
s->dba_lengths[ch][seg] = get_bits(gbc, 4);
|
|
s->dba_values[ch][seg] = get_bits(gbc, 3);
|
|
}
|
|
/* run last 2 bit allocation stages if new dba values */
|
|
bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
|
|
}
|
|
}
|
|
} else if(blk == 0) {
|
|
for(ch=0; ch<=s->channels; ch++) {
|
|
s->dba_mode[ch] = DBA_NONE;
|
|
}
|
|
}
|
|
|
|
/* Bit allocation */
|
|
for(ch=!cpl_in_use; ch<=s->channels; ch++) {
|
|
if(bit_alloc_stages[ch] > 2) {
|
|
/* Exponent mapping into PSD and PSD integration */
|
|
ff_ac3_bit_alloc_calc_psd(s->dexps[ch],
|
|
s->start_freq[ch], s->end_freq[ch],
|
|
s->psd[ch], s->band_psd[ch]);
|
|
}
|
|
if(bit_alloc_stages[ch] > 1) {
|
|
/* Compute excitation function, Compute masking curve, and
|
|
Apply delta bit allocation */
|
|
if (ff_ac3_bit_alloc_calc_mask(&s->bit_alloc_params, s->band_psd[ch],
|
|
s->start_freq[ch], s->end_freq[ch],
|
|
s->fast_gain[ch], (ch == s->lfe_ch),
|
|
s->dba_mode[ch], s->dba_nsegs[ch],
|
|
s->dba_offsets[ch], s->dba_lengths[ch],
|
|
s->dba_values[ch], s->mask[ch])) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "error in bit allocation\n");
|
|
return -1;
|
|
}
|
|
}
|
|
if(bit_alloc_stages[ch] > 0) {
|
|
/* Compute bit allocation */
|
|
const uint8_t *bap_tab = s->channel_uses_aht[ch] ?
|
|
ff_eac3_hebap_tab : ff_ac3_bap_tab;
|
|
s->ac3dsp.bit_alloc_calc_bap(s->mask[ch], s->psd[ch],
|
|
s->start_freq[ch], s->end_freq[ch],
|
|
s->snr_offset[ch],
|
|
s->bit_alloc_params.floor,
|
|
bap_tab, s->bap[ch]);
|
|
}
|
|
}
|
|
|
|
/* unused dummy data */
|
|
if (s->skip_syntax && get_bits1(gbc)) {
|
|
int skipl = get_bits(gbc, 9);
|
|
while(skipl--)
|
|
skip_bits(gbc, 8);
|
|
}
|
|
|
|
/* unpack the transform coefficients
|
|
this also uncouples channels if coupling is in use. */
|
|
decode_transform_coeffs(s, blk);
|
|
|
|
/* TODO: generate enhanced coupling coordinates and uncouple */
|
|
|
|
/* recover coefficients if rematrixing is in use */
|
|
if(s->channel_mode == AC3_CHMODE_STEREO)
|
|
do_rematrixing(s);
|
|
|
|
/* apply scaling to coefficients (headroom, dynrng) */
|
|
for(ch=1; ch<=s->channels; ch++) {
|
|
float gain = s->mul_bias / 4194304.0f;
|
|
if(s->channel_mode == AC3_CHMODE_DUALMONO) {
|
|
gain *= s->dynamic_range[2-ch];
|
|
} else {
|
|
gain *= s->dynamic_range[0];
|
|
}
|
|
s->fmt_conv.int32_to_float_fmul_scalar(s->transform_coeffs[ch], s->fixed_coeffs[ch], gain, 256);
|
|
}
|
|
|
|
/* apply spectral extension to high frequency bins */
|
|
if (s->spx_in_use && CONFIG_EAC3_DECODER) {
|
|
ff_eac3_apply_spectral_extension(s);
|
|
}
|
|
|
|
/* downmix and MDCT. order depends on whether block switching is used for
|
|
any channel in this block. this is because coefficients for the long
|
|
and short transforms cannot be mixed. */
|
|
downmix_output = s->channels != s->out_channels &&
|
|
!((s->output_mode & AC3_OUTPUT_LFEON) &&
|
|
s->fbw_channels == s->out_channels);
|
|
if(different_transforms) {
|
|
/* the delay samples have already been downmixed, so we upmix the delay
|
|
samples in order to reconstruct all channels before downmixing. */
|
|
if(s->downmixed) {
|
|
s->downmixed = 0;
|
|
ac3_upmix_delay(s);
|
|
}
|
|
|
|
do_imdct(s, s->channels);
|
|
|
|
if(downmix_output) {
|
|
s->dsp.ac3_downmix(s->output, s->downmix_coeffs, s->out_channels, s->fbw_channels, 256);
|
|
}
|
|
} else {
|
|
if(downmix_output) {
|
|
s->dsp.ac3_downmix(s->transform_coeffs+1, s->downmix_coeffs, s->out_channels, s->fbw_channels, 256);
|
|
}
|
|
|
|
if(downmix_output && !s->downmixed) {
|
|
s->downmixed = 1;
|
|
s->dsp.ac3_downmix(s->delay, s->downmix_coeffs, s->out_channels, s->fbw_channels, 128);
|
|
}
|
|
|
|
do_imdct(s, s->out_channels);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Decode a single AC-3 frame.
|
|
*/
|
|
static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
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AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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AC3DecodeContext *s = avctx->priv_data;
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float *out_samples_flt = data;
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int16_t *out_samples_s16 = data;
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int blk, ch, err;
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int data_size_orig, data_size_tmp;
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const uint8_t *channel_map;
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const float *output[AC3_MAX_CHANNELS];
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/* copy input buffer to decoder context to avoid reading past the end
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of the buffer, which can be caused by a damaged input stream. */
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if (buf_size >= 2 && AV_RB16(buf) == 0x770B) {
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// seems to be byte-swapped AC-3
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int cnt = FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE) >> 1;
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s->dsp.bswap16_buf((uint16_t *)s->input_buffer, (const uint16_t *)buf, cnt);
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} else
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memcpy(s->input_buffer, buf, FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE));
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buf = s->input_buffer;
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/* initialize the GetBitContext with the start of valid AC-3 Frame */
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init_get_bits(&s->gbc, buf, buf_size * 8);
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/* parse the syncinfo */
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data_size_orig = *data_size;
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*data_size = 0;
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err = parse_frame_header(s);
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if (err) {
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switch(err) {
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case AAC_AC3_PARSE_ERROR_SYNC:
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av_log(avctx, AV_LOG_ERROR, "frame sync error\n");
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return -1;
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case AAC_AC3_PARSE_ERROR_BSID:
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av_log(avctx, AV_LOG_ERROR, "invalid bitstream id\n");
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break;
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case AAC_AC3_PARSE_ERROR_SAMPLE_RATE:
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av_log(avctx, AV_LOG_ERROR, "invalid sample rate\n");
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break;
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case AAC_AC3_PARSE_ERROR_FRAME_SIZE:
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av_log(avctx, AV_LOG_ERROR, "invalid frame size\n");
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break;
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case AAC_AC3_PARSE_ERROR_FRAME_TYPE:
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/* skip frame if CRC is ok. otherwise use error concealment. */
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/* TODO: add support for substreams and dependent frames */
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if(s->frame_type == EAC3_FRAME_TYPE_DEPENDENT || s->substreamid) {
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av_log(avctx, AV_LOG_ERROR, "unsupported frame type : skipping frame\n");
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return s->frame_size;
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} else {
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av_log(avctx, AV_LOG_ERROR, "invalid frame type\n");
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}
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break;
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default:
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av_log(avctx, AV_LOG_ERROR, "invalid header\n");
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break;
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}
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} else {
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/* check that reported frame size fits in input buffer */
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if (s->frame_size > buf_size) {
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av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
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err = AAC_AC3_PARSE_ERROR_FRAME_SIZE;
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} else if (avctx->error_recognition >= FF_ER_CAREFUL) {
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/* check for crc mismatch */
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if (av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2], s->frame_size-2)) {
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av_log(avctx, AV_LOG_ERROR, "frame CRC mismatch\n");
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err = AAC_AC3_PARSE_ERROR_CRC;
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}
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}
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}
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/* if frame is ok, set audio parameters */
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if (!err) {
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avctx->sample_rate = s->sample_rate;
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avctx->bit_rate = s->bit_rate;
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/* channel config */
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s->out_channels = s->channels;
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s->output_mode = s->channel_mode;
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if(s->lfe_on)
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s->output_mode |= AC3_OUTPUT_LFEON;
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if (avctx->request_channels > 0 && avctx->request_channels <= 2 &&
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avctx->request_channels < s->channels) {
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s->out_channels = avctx->request_channels;
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s->output_mode = avctx->request_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO;
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s->channel_layout = ff_ac3_channel_layout_tab[s->output_mode];
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}
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avctx->channels = s->out_channels;
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avctx->channel_layout = s->channel_layout;
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/* set downmixing coefficients if needed */
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if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) &&
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s->fbw_channels == s->out_channels)) {
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set_downmix_coeffs(s);
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}
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} else if (!s->out_channels) {
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s->out_channels = avctx->channels;
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if(s->out_channels < s->channels)
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s->output_mode = s->out_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO;
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}
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/* set audio service type based on bitstream mode for AC-3 */
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avctx->audio_service_type = s->bitstream_mode;
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if (s->bitstream_mode == 0x7 && s->channels > 1)
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avctx->audio_service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE;
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/* decode the audio blocks */
|
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channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
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for (ch = 0; ch < s->out_channels; ch++)
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output[ch] = s->output[channel_map[ch]];
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data_size_tmp = s->num_blocks * 256 * avctx->channels;
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data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*out_samples_flt) : sizeof(*out_samples_s16);
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if (data_size_orig < data_size_tmp)
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return -1;
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*data_size = data_size_tmp;
|
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for (blk = 0; blk < s->num_blocks; blk++) {
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if (!err && decode_audio_block(s, blk)) {
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av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
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err = 1;
|
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}
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if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
|
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s->fmt_conv.float_interleave(out_samples_flt, output, 256,
|
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s->out_channels);
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out_samples_flt += 256 * s->out_channels;
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} else {
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s->fmt_conv.float_to_int16_interleave(out_samples_s16, output, 256,
|
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s->out_channels);
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out_samples_s16 += 256 * s->out_channels;
|
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}
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}
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*data_size = s->num_blocks * 256 * avctx->channels *
|
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(av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
|
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return FFMIN(buf_size, s->frame_size);
|
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}
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/**
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* Uninitialize the AC-3 decoder.
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*/
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static av_cold int ac3_decode_end(AVCodecContext *avctx)
|
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{
|
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AC3DecodeContext *s = avctx->priv_data;
|
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ff_mdct_end(&s->imdct_512);
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ff_mdct_end(&s->imdct_256);
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return 0;
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}
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|
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AVCodec ff_ac3_decoder = {
|
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.name = "ac3",
|
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.type = AVMEDIA_TYPE_AUDIO,
|
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.id = CODEC_ID_AC3,
|
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.priv_data_size = sizeof (AC3DecodeContext),
|
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.init = ac3_decode_init,
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.close = ac3_decode_end,
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.decode = ac3_decode_frame,
|
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.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
|
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.sample_fmts = (const enum AVSampleFormat[]) {
|
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AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
|
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},
|
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};
|
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|
|
#if CONFIG_EAC3_DECODER
|
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AVCodec ff_eac3_decoder = {
|
|
.name = "eac3",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = CODEC_ID_EAC3,
|
|
.priv_data_size = sizeof (AC3DecodeContext),
|
|
.init = ac3_decode_init,
|
|
.close = ac3_decode_end,
|
|
.decode = ac3_decode_frame,
|
|
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
|
|
.sample_fmts = (const enum AVSampleFormat[]) {
|
|
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
|
|
},
|
|
};
|
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#endif
|