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![Niels Möller](/assets/img/avatar_default.png)
Cleanup and integration by Diego Biurrun. Signed-off-by: Diego Biurrun <diego@biurrun.de>
1586 lines
57 KiB
C
1586 lines
57 KiB
C
/*
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* DCA compatible decoder
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* Copyright (C) 2004 Gildas Bazin
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* Copyright (C) 2004 Benjamin Zores
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* Copyright (C) 2006 Benjamin Larsson
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* Copyright (C) 2007 Konstantin Shishkov
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* Copyright (C) 2012 Paul B Mahol
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* Copyright (C) 2014 Niels Möller
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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 <math.h>
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#include <stddef.h>
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#include <stdio.h>
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#include "libavutil/attributes.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/internal.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/mathematics.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "avcodec.h"
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#include "dca.h"
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#include "dca_syncwords.h"
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#include "dcadata.h"
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#include "dcadsp.h"
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#include "dcahuff.h"
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#include "fft.h"
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#include "fmtconvert.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 "put_bits.h"
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#include "synth_filter.h"
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#if ARCH_ARM
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# include "arm/dca.h"
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#endif
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enum DCAMode {
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DCA_MONO = 0,
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DCA_CHANNEL,
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DCA_STEREO,
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DCA_STEREO_SUMDIFF,
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DCA_STEREO_TOTAL,
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DCA_3F,
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DCA_2F1R,
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DCA_3F1R,
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DCA_2F2R,
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DCA_3F2R,
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DCA_4F2R
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};
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/* -1 are reserved or unknown */
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static const int dca_ext_audio_descr_mask[] = {
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DCA_EXT_XCH,
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-1,
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DCA_EXT_X96,
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DCA_EXT_XCH | DCA_EXT_X96,
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-1,
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-1,
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DCA_EXT_XXCH,
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-1,
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};
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/* Tables for mapping dts channel configurations to libavcodec multichannel api.
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* Some compromises have been made for special configurations. Most configurations
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* are never used so complete accuracy is not needed.
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*
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* L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead.
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* S -> side, when both rear and back are configured move one of them to the side channel
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* OV -> center back
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* All 2 channel configurations -> AV_CH_LAYOUT_STEREO
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*/
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static const uint64_t dca_core_channel_layout[] = {
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AV_CH_FRONT_CENTER, ///< 1, A
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AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
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AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
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AV_CH_LAYOUT_STEREO, ///< 2, (L + R) + (L - R) (sum-difference)
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AV_CH_LAYOUT_STEREO, ///< 2, LT + RT (left and right total)
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AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER, ///< 3, C + L + R
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AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER, ///< 3, L + R + S
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AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 4, C + L + R + S
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AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 4, L + R + SL + SR
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AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |
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AV_CH_SIDE_RIGHT, ///< 5, C + L + R + SL + SR
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AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
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AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR
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AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |
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AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 6, C + L + R + LR + RR + OV
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AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
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AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER |
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AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 6, CF + CR + LF + RF + LR + RR
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AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
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AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
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AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR
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AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
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AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
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AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + SR2
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AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
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AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
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AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER | AV_CH_SIDE_RIGHT, ///< 8, CL + C + CR + L + R + SL + S + SR
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};
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#define DCA_DOLBY 101 /* FIXME */
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#define DCA_CHANNEL_BITS 6
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#define DCA_CHANNEL_MASK 0x3F
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#define DCA_LFE 0x80
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#define HEADER_SIZE 14
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#define DCA_NSYNCAUX 0x9A1105A0
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/** Bit allocation */
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typedef struct BitAlloc {
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int offset; ///< code values offset
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int maxbits[8]; ///< max bits in VLC
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int wrap; ///< wrap for get_vlc2()
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VLC vlc[8]; ///< actual codes
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} BitAlloc;
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static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select
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static BitAlloc dca_tmode; ///< transition mode VLCs
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static BitAlloc dca_scalefactor; ///< scalefactor VLCs
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static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
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static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,
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int idx)
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{
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return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) +
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ba->offset;
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}
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static av_cold void dca_init_vlcs(void)
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{
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static int vlcs_initialized = 0;
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int i, j, c = 14;
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static VLC_TYPE dca_table[23622][2];
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if (vlcs_initialized)
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return;
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dca_bitalloc_index.offset = 1;
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dca_bitalloc_index.wrap = 2;
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for (i = 0; i < 5; i++) {
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dca_bitalloc_index.vlc[i].table = &dca_table[ff_dca_vlc_offs[i]];
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dca_bitalloc_index.vlc[i].table_allocated = ff_dca_vlc_offs[i + 1] - ff_dca_vlc_offs[i];
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init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
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bitalloc_12_bits[i], 1, 1,
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bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
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}
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dca_scalefactor.offset = -64;
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dca_scalefactor.wrap = 2;
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for (i = 0; i < 5; i++) {
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dca_scalefactor.vlc[i].table = &dca_table[ff_dca_vlc_offs[i + 5]];
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dca_scalefactor.vlc[i].table_allocated = ff_dca_vlc_offs[i + 6] - ff_dca_vlc_offs[i + 5];
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init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
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scales_bits[i], 1, 1,
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scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
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}
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dca_tmode.offset = 0;
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dca_tmode.wrap = 1;
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for (i = 0; i < 4; i++) {
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dca_tmode.vlc[i].table = &dca_table[ff_dca_vlc_offs[i + 10]];
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dca_tmode.vlc[i].table_allocated = ff_dca_vlc_offs[i + 11] - ff_dca_vlc_offs[i + 10];
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init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
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tmode_bits[i], 1, 1,
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tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
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}
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for (i = 0; i < 10; i++)
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for (j = 0; j < 7; j++) {
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if (!bitalloc_codes[i][j])
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break;
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dca_smpl_bitalloc[i + 1].offset = bitalloc_offsets[i];
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dca_smpl_bitalloc[i + 1].wrap = 1 + (j > 4);
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dca_smpl_bitalloc[i + 1].vlc[j].table = &dca_table[ff_dca_vlc_offs[c]];
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dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = ff_dca_vlc_offs[c + 1] - ff_dca_vlc_offs[c];
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init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],
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bitalloc_sizes[i],
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bitalloc_bits[i][j], 1, 1,
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bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
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c++;
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}
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vlcs_initialized = 1;
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}
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static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
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{
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while (len--)
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*dst++ = get_bits(gb, bits);
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}
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static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
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{
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int i, j;
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static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
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static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
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static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
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s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
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s->prim_channels = s->total_channels;
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if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
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s->prim_channels = DCA_PRIM_CHANNELS_MAX;
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for (i = base_channel; i < s->prim_channels; i++) {
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s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
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if (s->subband_activity[i] > DCA_SUBBANDS)
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s->subband_activity[i] = DCA_SUBBANDS;
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}
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for (i = base_channel; i < s->prim_channels; i++) {
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s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
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if (s->vq_start_subband[i] > DCA_SUBBANDS)
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s->vq_start_subband[i] = DCA_SUBBANDS;
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}
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get_array(&s->gb, s->joint_intensity + base_channel, s->prim_channels - base_channel, 3);
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get_array(&s->gb, s->transient_huffman + base_channel, s->prim_channels - base_channel, 2);
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get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3);
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get_array(&s->gb, s->bitalloc_huffman + base_channel, s->prim_channels - base_channel, 3);
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/* Get codebooks quantization indexes */
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if (!base_channel)
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memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
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for (j = 1; j < 11; j++)
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for (i = base_channel; i < s->prim_channels; i++)
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s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
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/* Get scale factor adjustment */
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for (j = 0; j < 11; j++)
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for (i = base_channel; i < s->prim_channels; i++)
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s->scalefactor_adj[i][j] = 1;
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for (j = 1; j < 11; j++)
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for (i = base_channel; i < s->prim_channels; i++)
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if (s->quant_index_huffman[i][j] < thr[j])
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s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
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if (s->crc_present) {
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/* Audio header CRC check */
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get_bits(&s->gb, 16);
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}
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s->current_subframe = 0;
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s->current_subsubframe = 0;
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return 0;
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}
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static int dca_parse_frame_header(DCAContext *s)
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{
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init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
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/* Sync code */
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skip_bits_long(&s->gb, 32);
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/* Frame header */
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s->frame_type = get_bits(&s->gb, 1);
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s->samples_deficit = get_bits(&s->gb, 5) + 1;
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s->crc_present = get_bits(&s->gb, 1);
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s->sample_blocks = get_bits(&s->gb, 7) + 1;
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s->frame_size = get_bits(&s->gb, 14) + 1;
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if (s->frame_size < 95)
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return AVERROR_INVALIDDATA;
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s->amode = get_bits(&s->gb, 6);
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s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];
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if (!s->sample_rate)
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return AVERROR_INVALIDDATA;
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s->bit_rate_index = get_bits(&s->gb, 5);
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s->bit_rate = ff_dca_bit_rates[s->bit_rate_index];
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if (!s->bit_rate)
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return AVERROR_INVALIDDATA;
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skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)
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s->dynrange = get_bits(&s->gb, 1);
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s->timestamp = get_bits(&s->gb, 1);
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s->aux_data = get_bits(&s->gb, 1);
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s->hdcd = get_bits(&s->gb, 1);
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s->ext_descr = get_bits(&s->gb, 3);
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s->ext_coding = get_bits(&s->gb, 1);
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s->aspf = get_bits(&s->gb, 1);
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s->lfe = get_bits(&s->gb, 2);
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s->predictor_history = get_bits(&s->gb, 1);
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if (s->lfe > 2) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe);
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return AVERROR_INVALIDDATA;
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}
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/* TODO: check CRC */
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if (s->crc_present)
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s->header_crc = get_bits(&s->gb, 16);
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s->multirate_inter = get_bits(&s->gb, 1);
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s->version = get_bits(&s->gb, 4);
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s->copy_history = get_bits(&s->gb, 2);
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s->source_pcm_res = get_bits(&s->gb, 3);
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s->front_sum = get_bits(&s->gb, 1);
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s->surround_sum = get_bits(&s->gb, 1);
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s->dialog_norm = get_bits(&s->gb, 4);
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/* FIXME: channels mixing levels */
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s->output = s->amode;
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if (s->lfe)
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s->output |= DCA_LFE;
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/* Primary audio coding header */
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s->subframes = get_bits(&s->gb, 4) + 1;
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return dca_parse_audio_coding_header(s, 0);
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}
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static inline int get_scale(GetBitContext *gb, int level, int value, int log2range)
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{
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if (level < 5) {
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/* huffman encoded */
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value += get_bitalloc(gb, &dca_scalefactor, level);
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value = av_clip(value, 0, (1 << log2range) - 1);
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} else if (level < 8) {
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if (level + 1 > log2range) {
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skip_bits(gb, level + 1 - log2range);
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value = get_bits(gb, log2range);
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} else {
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value = get_bits(gb, level + 1);
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}
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}
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return value;
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}
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static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
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{
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/* Primary audio coding side information */
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int j, k;
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if (get_bits_left(&s->gb) < 0)
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return AVERROR_INVALIDDATA;
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if (!base_channel) {
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s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
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s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
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}
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for (j = base_channel; j < s->prim_channels; j++) {
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for (k = 0; k < s->subband_activity[j]; k++)
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s->prediction_mode[j][k] = get_bits(&s->gb, 1);
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}
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/* Get prediction codebook */
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for (j = base_channel; j < s->prim_channels; j++) {
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for (k = 0; k < s->subband_activity[j]; k++) {
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if (s->prediction_mode[j][k] > 0) {
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/* (Prediction coefficient VQ address) */
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s->prediction_vq[j][k] = get_bits(&s->gb, 12);
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}
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}
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}
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|
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/* Bit allocation index */
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for (j = base_channel; j < s->prim_channels; j++) {
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for (k = 0; k < s->vq_start_subband[j]; k++) {
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if (s->bitalloc_huffman[j] == 6)
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s->bitalloc[j][k] = get_bits(&s->gb, 5);
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else if (s->bitalloc_huffman[j] == 5)
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s->bitalloc[j][k] = get_bits(&s->gb, 4);
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else if (s->bitalloc_huffman[j] == 7) {
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av_log(s->avctx, AV_LOG_ERROR,
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"Invalid bit allocation index\n");
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return AVERROR_INVALIDDATA;
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} else {
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s->bitalloc[j][k] =
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get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
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}
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if (s->bitalloc[j][k] > 26) {
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av_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
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j, k, s->bitalloc[j][k]);
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return AVERROR_INVALIDDATA;
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}
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}
|
|
}
|
|
|
|
/* Transition mode */
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
for (k = 0; k < s->subband_activity[j]; k++) {
|
|
s->transition_mode[j][k] = 0;
|
|
if (s->subsubframes[s->current_subframe] > 1 &&
|
|
k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
|
|
s->transition_mode[j][k] =
|
|
get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (get_bits_left(&s->gb) < 0)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
const uint32_t *scale_table;
|
|
int scale_sum, log_size;
|
|
|
|
memset(s->scale_factor[j], 0,
|
|
s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
|
|
|
|
if (s->scalefactor_huffman[j] == 6) {
|
|
scale_table = ff_dca_scale_factor_quant7;
|
|
log_size = 7;
|
|
} else {
|
|
scale_table = ff_dca_scale_factor_quant6;
|
|
log_size = 6;
|
|
}
|
|
|
|
/* When huffman coded, only the difference is encoded */
|
|
scale_sum = 0;
|
|
|
|
for (k = 0; k < s->subband_activity[j]; k++) {
|
|
if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
|
|
scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
|
|
s->scale_factor[j][k][0] = scale_table[scale_sum];
|
|
}
|
|
|
|
if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
|
|
/* Get second scale factor */
|
|
scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
|
|
s->scale_factor[j][k][1] = scale_table[scale_sum];
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Joint subband scale factor codebook select */
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
/* Transmitted only if joint subband coding enabled */
|
|
if (s->joint_intensity[j] > 0)
|
|
s->joint_huff[j] = get_bits(&s->gb, 3);
|
|
}
|
|
|
|
if (get_bits_left(&s->gb) < 0)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
/* Scale factors for joint subband coding */
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
int source_channel;
|
|
|
|
/* Transmitted only if joint subband coding enabled */
|
|
if (s->joint_intensity[j] > 0) {
|
|
int scale = 0;
|
|
source_channel = s->joint_intensity[j] - 1;
|
|
|
|
/* When huffman coded, only the difference is encoded
|
|
* (is this valid as well for joint scales ???) */
|
|
|
|
for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
|
|
scale = get_scale(&s->gb, s->joint_huff[j], 64 /* bias */, 7);
|
|
s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */
|
|
}
|
|
|
|
if (!(s->debug_flag & 0x02)) {
|
|
av_log(s->avctx, AV_LOG_DEBUG,
|
|
"Joint stereo coding not supported\n");
|
|
s->debug_flag |= 0x02;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Dynamic range coefficient */
|
|
if (!base_channel && s->dynrange)
|
|
s->dynrange_coef = get_bits(&s->gb, 8);
|
|
|
|
/* Side information CRC check word */
|
|
if (s->crc_present) {
|
|
get_bits(&s->gb, 16);
|
|
}
|
|
|
|
/*
|
|
* Primary audio data arrays
|
|
*/
|
|
|
|
/* VQ encoded high frequency subbands */
|
|
for (j = base_channel; j < s->prim_channels; j++)
|
|
for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
|
|
/* 1 vector -> 32 samples */
|
|
s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
|
|
|
|
/* Low frequency effect data */
|
|
if (!base_channel && s->lfe) {
|
|
/* LFE samples */
|
|
int lfe_samples = 2 * s->lfe * (4 + block_index);
|
|
int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
|
|
float lfe_scale;
|
|
|
|
for (j = lfe_samples; j < lfe_end_sample; j++) {
|
|
/* Signed 8 bits int */
|
|
s->lfe_data[j] = get_sbits(&s->gb, 8);
|
|
}
|
|
|
|
/* Scale factor index */
|
|
skip_bits(&s->gb, 1);
|
|
s->lfe_scale_factor = ff_dca_scale_factor_quant7[get_bits(&s->gb, 7)];
|
|
|
|
/* Quantization step size * scale factor */
|
|
lfe_scale = 0.035 * s->lfe_scale_factor;
|
|
|
|
for (j = lfe_samples; j < lfe_end_sample; j++)
|
|
s->lfe_data[j] *= lfe_scale;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void qmf_32_subbands(DCAContext *s, int chans,
|
|
float samples_in[32][8], float *samples_out,
|
|
float scale)
|
|
{
|
|
const float *prCoeff;
|
|
|
|
int sb_act = s->subband_activity[chans];
|
|
|
|
scale *= sqrt(1 / 8.0);
|
|
|
|
/* Select filter */
|
|
if (!s->multirate_inter) /* Non-perfect reconstruction */
|
|
prCoeff = ff_dca_fir_32bands_nonperfect;
|
|
else /* Perfect reconstruction */
|
|
prCoeff = ff_dca_fir_32bands_perfect;
|
|
|
|
s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
|
|
s->subband_fir_hist[chans],
|
|
&s->hist_index[chans],
|
|
s->subband_fir_noidea[chans], prCoeff,
|
|
samples_out, s->raXin, scale);
|
|
}
|
|
|
|
static QMF64_table *qmf64_precompute(void)
|
|
{
|
|
unsigned i, j;
|
|
QMF64_table *table = av_malloc(sizeof(*table));
|
|
if (!table)
|
|
return NULL;
|
|
|
|
for (i = 0; i < 32; i++)
|
|
for (j = 0; j < 32; j++)
|
|
table->dct4_coeff[i][j] = cos((2 * i + 1) * (2 * j + 1) * M_PI / 128);
|
|
for (i = 0; i < 32; i++)
|
|
for (j = 0; j < 32; j++)
|
|
table->dct2_coeff[i][j] = cos((2 * i + 1) * j * M_PI / 64);
|
|
|
|
/* FIXME: Is the factor 0.125 = 1/8 right? */
|
|
for (i = 0; i < 32; i++)
|
|
table->rcos[i] = 0.125 / cos((2 * i + 1) * M_PI / 256);
|
|
for (i = 0; i < 32; i++)
|
|
table->rsin[i] = -0.125 / sin((2 * i + 1) * M_PI / 256);
|
|
|
|
return table;
|
|
}
|
|
|
|
/* FIXME: Totally unoptimized. Based on the reference code and
|
|
* http://multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks
|
|
* for doubling the size. */
|
|
static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][8],
|
|
float *samples_out, float scale)
|
|
{
|
|
float raXin[64];
|
|
float A[32], B[32];
|
|
float *raX = s->subband_fir_hist[chans];
|
|
float *raZ = s->subband_fir_noidea[chans];
|
|
unsigned i, j, k, subindex;
|
|
|
|
for (i = s->subband_activity[chans]; i < 64; i++)
|
|
raXin[i] = 0.0;
|
|
for (subindex = 0; subindex < 8; subindex++) {
|
|
for (i = 0; i < s->subband_activity[chans]; i++)
|
|
raXin[i] = samples_in[i][subindex];
|
|
|
|
for (k = 0; k < 32; k++) {
|
|
A[k] = 0.0;
|
|
for (i = 0; i < 32; i++)
|
|
A[k] += (raXin[2 * i] + raXin[2 * i + 1]) * s->qmf64_table->dct4_coeff[k][i];
|
|
}
|
|
for (k = 0; k < 32; k++) {
|
|
B[k] = raXin[0] * s->qmf64_table->dct2_coeff[k][0];
|
|
for (i = 1; i < 32; i++)
|
|
B[k] += (raXin[2 * i] + raXin[2 * i - 1]) * s->qmf64_table->dct2_coeff[k][i];
|
|
}
|
|
for (k = 0; k < 32; k++) {
|
|
raX[k] = s->qmf64_table->rcos[k] * (A[k] + B[k]);
|
|
raX[63 - k] = s->qmf64_table->rsin[k] * (A[k] - B[k]);
|
|
}
|
|
|
|
for (i = 0; i < 64; i++) {
|
|
float out = raZ[i];
|
|
for (j = 0; j < 1024; j += 128)
|
|
out += ff_dca_fir_64bands[j + i] * (raX[j + i] - raX[j + 63 - i]);
|
|
*samples_out++ = out * scale;
|
|
}
|
|
|
|
for (i = 0; i < 64; i++) {
|
|
float hist = 0.0;
|
|
for (j = 0; j < 1024; j += 128)
|
|
hist += ff_dca_fir_64bands[64 + j + i] * (-raX[i + j] - raX[j + 63 - i]);
|
|
|
|
raZ[i] = hist;
|
|
}
|
|
|
|
/* FIXME: Make buffer circular, to avoid this move. */
|
|
memmove(raX + 64, raX, (1024 - 64) * sizeof(*raX));
|
|
}
|
|
}
|
|
|
|
static void lfe_interpolation_fir(DCAContext *s, const float *samples_in,
|
|
float *samples_out)
|
|
{
|
|
/* samples_in: An array holding decimated samples.
|
|
* Samples in current subframe starts from samples_in[0],
|
|
* while samples_in[-1], samples_in[-2], ..., stores samples
|
|
* from last subframe as history.
|
|
*
|
|
* samples_out: An array holding interpolated samples
|
|
*/
|
|
|
|
int idx;
|
|
const float *prCoeff;
|
|
int deciindex;
|
|
|
|
/* Select decimation filter */
|
|
if (s->lfe == 1) {
|
|
idx = 1;
|
|
prCoeff = ff_dca_lfe_fir_128;
|
|
} else {
|
|
idx = 0;
|
|
if (s->exss_ext_mask & DCA_EXT_EXSS_XLL)
|
|
prCoeff = ff_dca_lfe_xll_fir_64;
|
|
else
|
|
prCoeff = ff_dca_lfe_fir_64;
|
|
}
|
|
/* Interpolation */
|
|
for (deciindex = 0; deciindex < 2 * s->lfe; deciindex++) {
|
|
s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff);
|
|
samples_in++;
|
|
samples_out += 2 * 32 * (1 + idx);
|
|
}
|
|
}
|
|
|
|
/* downmixing routines */
|
|
#define MIX_REAR1(samples, s1, rs, coef) \
|
|
samples[0][i] += samples[s1][i] * coef[rs][0]; \
|
|
samples[1][i] += samples[s1][i] * coef[rs][1];
|
|
|
|
#define MIX_REAR2(samples, s1, s2, rs, coef) \
|
|
samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \
|
|
samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];
|
|
|
|
#define MIX_FRONT3(samples, coef) \
|
|
t = samples[c][i]; \
|
|
u = samples[l][i]; \
|
|
v = samples[r][i]; \
|
|
samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
|
|
samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
|
|
|
|
#define DOWNMIX_TO_STEREO(op1, op2) \
|
|
for (i = 0; i < 256; i++) { \
|
|
op1 \
|
|
op2 \
|
|
}
|
|
|
|
static void dca_downmix(float **samples, int srcfmt, int lfe_present,
|
|
float coef[DCA_PRIM_CHANNELS_MAX + 1][2],
|
|
const int8_t *channel_mapping)
|
|
{
|
|
int c, l, r, sl, sr, s;
|
|
int i;
|
|
float t, u, v;
|
|
|
|
switch (srcfmt) {
|
|
case DCA_MONO:
|
|
case DCA_4F2R:
|
|
av_log(NULL, 0, "Not implemented!\n");
|
|
break;
|
|
case DCA_CHANNEL:
|
|
case DCA_STEREO:
|
|
case DCA_STEREO_TOTAL:
|
|
case DCA_STEREO_SUMDIFF:
|
|
break;
|
|
case DCA_3F:
|
|
c = channel_mapping[0];
|
|
l = channel_mapping[1];
|
|
r = channel_mapping[2];
|
|
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
|
|
break;
|
|
case DCA_2F1R:
|
|
s = channel_mapping[2];
|
|
DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );
|
|
break;
|
|
case DCA_3F1R:
|
|
c = channel_mapping[0];
|
|
l = channel_mapping[1];
|
|
r = channel_mapping[2];
|
|
s = channel_mapping[3];
|
|
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
|
|
MIX_REAR1(samples, s, 3, coef));
|
|
break;
|
|
case DCA_2F2R:
|
|
sl = channel_mapping[2];
|
|
sr = channel_mapping[3];
|
|
DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );
|
|
break;
|
|
case DCA_3F2R:
|
|
c = channel_mapping[0];
|
|
l = channel_mapping[1];
|
|
r = channel_mapping[2];
|
|
sl = channel_mapping[3];
|
|
sr = channel_mapping[4];
|
|
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
|
|
MIX_REAR2(samples, sl, sr, 3, coef));
|
|
break;
|
|
}
|
|
if (lfe_present) {
|
|
int lf_buf = ff_dca_lfe_index[srcfmt];
|
|
int lf_idx = ff_dca_channels[srcfmt];
|
|
for (i = 0; i < 256; i++) {
|
|
samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];
|
|
samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];
|
|
}
|
|
}
|
|
}
|
|
|
|
#ifndef decode_blockcodes
|
|
/* Very compact version of the block code decoder that does not use table
|
|
* look-up but is slightly slower */
|
|
static int decode_blockcode(int code, int levels, int32_t *values)
|
|
{
|
|
int i;
|
|
int offset = (levels - 1) >> 1;
|
|
|
|
for (i = 0; i < 4; i++) {
|
|
int div = FASTDIV(code, levels);
|
|
values[i] = code - offset - div * levels;
|
|
code = div;
|
|
}
|
|
|
|
return code;
|
|
}
|
|
|
|
static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)
|
|
{
|
|
return decode_blockcode(code1, levels, values) |
|
|
decode_blockcode(code2, levels, values + 4);
|
|
}
|
|
#endif
|
|
|
|
static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
|
|
static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
|
|
|
|
static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
|
|
{
|
|
int k, l;
|
|
int subsubframe = s->current_subsubframe;
|
|
|
|
const float *quant_step_table;
|
|
|
|
/* FIXME */
|
|
float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
|
|
LOCAL_ALIGNED_16(int32_t, block, [8 * DCA_SUBBANDS]);
|
|
|
|
/*
|
|
* Audio data
|
|
*/
|
|
|
|
/* Select quantization step size table */
|
|
if (s->bit_rate_index == 0x1f)
|
|
quant_step_table = ff_dca_lossless_quant_d;
|
|
else
|
|
quant_step_table = ff_dca_lossy_quant_d;
|
|
|
|
for (k = base_channel; k < s->prim_channels; k++) {
|
|
float rscale[DCA_SUBBANDS];
|
|
|
|
if (get_bits_left(&s->gb) < 0)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
for (l = 0; l < s->vq_start_subband[k]; l++) {
|
|
int m;
|
|
|
|
/* Select the mid-tread linear quantizer */
|
|
int abits = s->bitalloc[k][l];
|
|
|
|
float quant_step_size = quant_step_table[abits];
|
|
|
|
/*
|
|
* Determine quantization index code book and its type
|
|
*/
|
|
|
|
/* Select quantization index code book */
|
|
int sel = s->quant_index_huffman[k][abits];
|
|
|
|
/*
|
|
* Extract bits from the bit stream
|
|
*/
|
|
if (!abits) {
|
|
rscale[l] = 0;
|
|
memset(block + 8 * l, 0, 8 * sizeof(block[0]));
|
|
} else {
|
|
/* Deal with transients */
|
|
int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
|
|
rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] *
|
|
s->scalefactor_adj[k][sel];
|
|
|
|
if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
|
|
if (abits <= 7) {
|
|
/* Block code */
|
|
int block_code1, block_code2, size, levels, err;
|
|
|
|
size = abits_sizes[abits - 1];
|
|
levels = abits_levels[abits - 1];
|
|
|
|
block_code1 = get_bits(&s->gb, size);
|
|
block_code2 = get_bits(&s->gb, size);
|
|
err = decode_blockcodes(block_code1, block_code2,
|
|
levels, block + 8 * l);
|
|
if (err) {
|
|
av_log(s->avctx, AV_LOG_ERROR,
|
|
"ERROR: block code look-up failed\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
} else {
|
|
/* no coding */
|
|
for (m = 0; m < 8; m++)
|
|
block[8 * l + m] = get_sbits(&s->gb, abits - 3);
|
|
}
|
|
} else {
|
|
/* Huffman coded */
|
|
for (m = 0; m < 8; m++)
|
|
block[8 * l + m] = get_bitalloc(&s->gb,
|
|
&dca_smpl_bitalloc[abits], sel);
|
|
}
|
|
}
|
|
}
|
|
|
|
s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],
|
|
block, rscale, 8 * s->vq_start_subband[k]);
|
|
|
|
for (l = 0; l < s->vq_start_subband[k]; l++) {
|
|
int m;
|
|
/*
|
|
* Inverse ADPCM if in prediction mode
|
|
*/
|
|
if (s->prediction_mode[k][l]) {
|
|
int n;
|
|
if (s->predictor_history)
|
|
subband_samples[k][l][0] += (ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] *
|
|
s->subband_samples_hist[k][l][3] +
|
|
ff_dca_adpcm_vb[s->prediction_vq[k][l]][1] *
|
|
s->subband_samples_hist[k][l][2] +
|
|
ff_dca_adpcm_vb[s->prediction_vq[k][l]][2] *
|
|
s->subband_samples_hist[k][l][1] +
|
|
ff_dca_adpcm_vb[s->prediction_vq[k][l]][3] *
|
|
s->subband_samples_hist[k][l][0]) *
|
|
(1.0f / 8192);
|
|
for (m = 1; m < 8; m++) {
|
|
float sum = ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] *
|
|
subband_samples[k][l][m - 1];
|
|
for (n = 2; n <= 4; n++)
|
|
if (m >= n)
|
|
sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] *
|
|
subband_samples[k][l][m - n];
|
|
else if (s->predictor_history)
|
|
sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] *
|
|
s->subband_samples_hist[k][l][m - n + 4];
|
|
subband_samples[k][l][m] += sum * 1.0f / 8192;
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Decode VQ encoded high frequencies
|
|
*/
|
|
if (s->subband_activity[k] > s->vq_start_subband[k]) {
|
|
if (!s->debug_flag & 0x01) {
|
|
av_log(s->avctx, AV_LOG_DEBUG,
|
|
"Stream with high frequencies VQ coding\n");
|
|
s->debug_flag |= 0x01;
|
|
}
|
|
s->dcadsp.decode_hf(subband_samples[k], s->high_freq_vq[k],
|
|
ff_dca_high_freq_vq, subsubframe * 8,
|
|
s->scale_factor[k], s->vq_start_subband[k],
|
|
s->subband_activity[k]);
|
|
}
|
|
}
|
|
|
|
/* Check for DSYNC after subsubframe */
|
|
if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
|
|
if (get_bits(&s->gb, 16) != 0xFFFF) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
}
|
|
|
|
/* Backup predictor history for adpcm */
|
|
for (k = base_channel; k < s->prim_channels; k++)
|
|
for (l = 0; l < s->vq_start_subband[k]; l++)
|
|
AV_COPY128(s->subband_samples_hist[k][l], &subband_samples[k][l][4]);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
|
|
{
|
|
float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
|
|
int k;
|
|
|
|
if (upsample) {
|
|
if (!s->qmf64_table) {
|
|
s->qmf64_table = qmf64_precompute();
|
|
if (!s->qmf64_table)
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
/* 64 subbands QMF */
|
|
for (k = 0; k < s->prim_channels; k++) {
|
|
if (s->channel_order_tab[k] >= 0)
|
|
qmf_64_subbands(s, k, subband_samples[k],
|
|
s->samples_chanptr[s->channel_order_tab[k]],
|
|
/* Upsampling needs a factor 2 here. */
|
|
M_SQRT2 / 32768.0);
|
|
}
|
|
} else {
|
|
/* 32 subbands QMF */
|
|
for (k = 0; k < s->prim_channels; k++) {
|
|
if (s->channel_order_tab[k] >= 0)
|
|
qmf_32_subbands(s, k, subband_samples[k],
|
|
s->samples_chanptr[s->channel_order_tab[k]],
|
|
M_SQRT1_2 / 32768.0);
|
|
}
|
|
}
|
|
|
|
/* Generate LFE samples for this subsubframe FIXME!!! */
|
|
if (s->lfe) {
|
|
float *samples = s->samples_chanptr[ff_dca_lfe_index[s->amode]];
|
|
lfe_interpolation_fir(s,
|
|
s->lfe_data + 2 * s->lfe * (block_index + 4),
|
|
samples);
|
|
if (upsample) {
|
|
unsigned i;
|
|
/* Should apply the filter in Table 6-11 when upsampling. For
|
|
* now, just duplicate. */
|
|
for (i = 511; i > 0; i--) {
|
|
samples[2 * i] =
|
|
samples[2 * i + 1] = samples[i];
|
|
}
|
|
samples[1] = samples[0];
|
|
}
|
|
}
|
|
|
|
/* FIXME: This downmixing is probably broken with upsample.
|
|
* Probably totally broken also with XLL in general. */
|
|
/* Downmixing to Stereo */
|
|
if (s->prim_channels + !!s->lfe > 2 &&
|
|
s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
|
|
dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
|
|
s->channel_order_tab);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int dca_subframe_footer(DCAContext *s, int base_channel)
|
|
{
|
|
int in, out, aux_data_count, aux_data_end, reserved;
|
|
uint32_t nsyncaux;
|
|
|
|
/*
|
|
* Unpack optional information
|
|
*/
|
|
|
|
/* presumably optional information only appears in the core? */
|
|
if (!base_channel) {
|
|
if (s->timestamp)
|
|
skip_bits_long(&s->gb, 32);
|
|
|
|
if (s->aux_data) {
|
|
aux_data_count = get_bits(&s->gb, 6);
|
|
|
|
// align (32-bit)
|
|
skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
|
|
|
|
aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);
|
|
|
|
if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",
|
|
nsyncaux);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if (get_bits1(&s->gb)) { // bAUXTimeStampFlag
|
|
avpriv_request_sample(s->avctx,
|
|
"Auxiliary Decode Time Stamp Flag");
|
|
// align (4-bit)
|
|
skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);
|
|
// 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)
|
|
skip_bits_long(&s->gb, 44);
|
|
}
|
|
|
|
if ((s->core_downmix = get_bits1(&s->gb))) {
|
|
int am = get_bits(&s->gb, 3);
|
|
switch (am) {
|
|
case 0:
|
|
s->core_downmix_amode = DCA_MONO;
|
|
break;
|
|
case 1:
|
|
s->core_downmix_amode = DCA_STEREO;
|
|
break;
|
|
case 2:
|
|
s->core_downmix_amode = DCA_STEREO_TOTAL;
|
|
break;
|
|
case 3:
|
|
s->core_downmix_amode = DCA_3F;
|
|
break;
|
|
case 4:
|
|
s->core_downmix_amode = DCA_2F1R;
|
|
break;
|
|
case 5:
|
|
s->core_downmix_amode = DCA_2F2R;
|
|
break;
|
|
case 6:
|
|
s->core_downmix_amode = DCA_3F1R;
|
|
break;
|
|
default:
|
|
av_log(s->avctx, AV_LOG_ERROR,
|
|
"Invalid mode %d for embedded downmix coefficients\n",
|
|
am);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) {
|
|
for (in = 0; in < s->prim_channels + !!s->lfe; in++) {
|
|
uint16_t tmp = get_bits(&s->gb, 9);
|
|
if ((tmp & 0xFF) > 241) {
|
|
av_log(s->avctx, AV_LOG_ERROR,
|
|
"Invalid downmix coefficient code %"PRIu16"\n",
|
|
tmp);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
s->core_downmix_codes[in][out] = tmp;
|
|
}
|
|
}
|
|
}
|
|
|
|
align_get_bits(&s->gb); // byte align
|
|
skip_bits(&s->gb, 16); // nAUXCRC16
|
|
|
|
// additional data (reserved, cf. ETSI TS 102 114 V1.4.1)
|
|
if ((reserved = (aux_data_end - get_bits_count(&s->gb))) < 0) {
|
|
av_log(s->avctx, AV_LOG_ERROR,
|
|
"Overread auxiliary data by %d bits\n", -reserved);
|
|
return AVERROR_INVALIDDATA;
|
|
} else if (reserved) {
|
|
avpriv_request_sample(s->avctx,
|
|
"Core auxiliary data reserved content");
|
|
skip_bits_long(&s->gb, reserved);
|
|
}
|
|
}
|
|
|
|
if (s->crc_present && s->dynrange)
|
|
get_bits(&s->gb, 16);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Decode a dca frame block
|
|
*
|
|
* @param s pointer to the DCAContext
|
|
*/
|
|
|
|
static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
|
|
{
|
|
int ret;
|
|
|
|
/* Sanity check */
|
|
if (s->current_subframe >= s->subframes) {
|
|
av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
|
|
s->current_subframe, s->subframes);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if (!s->current_subsubframe) {
|
|
/* Read subframe header */
|
|
if ((ret = dca_subframe_header(s, base_channel, block_index)))
|
|
return ret;
|
|
}
|
|
|
|
/* Read subsubframe */
|
|
if ((ret = dca_subsubframe(s, base_channel, block_index)))
|
|
return ret;
|
|
|
|
/* Update state */
|
|
s->current_subsubframe++;
|
|
if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {
|
|
s->current_subsubframe = 0;
|
|
s->current_subframe++;
|
|
}
|
|
if (s->current_subframe >= s->subframes) {
|
|
/* Read subframe footer */
|
|
if ((ret = dca_subframe_footer(s, base_channel)))
|
|
return ret;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static float dca_dmix_code(unsigned code)
|
|
{
|
|
int sign = (code >> 8) - 1;
|
|
code &= 0xff;
|
|
return ((ff_dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1U << 15));
|
|
}
|
|
|
|
/**
|
|
* Main frame decoding function
|
|
* FIXME add arguments
|
|
*/
|
|
static int dca_decode_frame(AVCodecContext *avctx, void *data,
|
|
int *got_frame_ptr, AVPacket *avpkt)
|
|
{
|
|
AVFrame *frame = data;
|
|
const uint8_t *buf = avpkt->data;
|
|
int buf_size = avpkt->size;
|
|
|
|
int lfe_samples;
|
|
int num_core_channels = 0;
|
|
int i, ret;
|
|
float **samples_flt;
|
|
DCAContext *s = avctx->priv_data;
|
|
int channels, full_channels;
|
|
int core_ss_end;
|
|
int upsample = 0;
|
|
|
|
s->exss_ext_mask = 0;
|
|
s->xch_present = 0;
|
|
|
|
s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,
|
|
DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
|
|
if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
|
|
av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if ((ret = dca_parse_frame_header(s)) < 0) {
|
|
// seems like the frame is corrupt, try with the next one
|
|
return ret;
|
|
}
|
|
// set AVCodec values with parsed data
|
|
avctx->sample_rate = s->sample_rate;
|
|
avctx->bit_rate = s->bit_rate;
|
|
|
|
s->profile = FF_PROFILE_DTS;
|
|
|
|
for (i = 0; i < (s->sample_blocks / 8); i++) {
|
|
if ((ret = dca_decode_block(s, 0, i))) {
|
|
av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
/* record number of core channels incase less than max channels are requested */
|
|
num_core_channels = s->prim_channels;
|
|
|
|
if (s->ext_coding)
|
|
s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
|
|
else
|
|
s->core_ext_mask = 0;
|
|
|
|
core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;
|
|
|
|
/* only scan for extensions if ext_descr was unknown or indicated a
|
|
* supported XCh extension */
|
|
if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {
|
|
/* if ext_descr was unknown, clear s->core_ext_mask so that the
|
|
* extensions scan can fill it up */
|
|
s->core_ext_mask = FFMAX(s->core_ext_mask, 0);
|
|
|
|
/* extensions start at 32-bit boundaries into bitstream */
|
|
skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
|
|
|
|
while (core_ss_end - get_bits_count(&s->gb) >= 32) {
|
|
uint32_t bits = get_bits_long(&s->gb, 32);
|
|
|
|
switch (bits) {
|
|
case DCA_SYNCWORD_XCH: {
|
|
int ext_amode, xch_fsize;
|
|
|
|
s->xch_base_channel = s->prim_channels;
|
|
|
|
/* validate sync word using XCHFSIZE field */
|
|
xch_fsize = show_bits(&s->gb, 10);
|
|
if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
|
|
(s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
|
|
continue;
|
|
|
|
/* skip length-to-end-of-frame field for the moment */
|
|
skip_bits(&s->gb, 10);
|
|
|
|
s->core_ext_mask |= DCA_EXT_XCH;
|
|
|
|
/* extension amode(number of channels in extension) should be 1 */
|
|
/* AFAIK XCh is not used for more channels */
|
|
if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
|
|
av_log(avctx, AV_LOG_ERROR,
|
|
"XCh extension amode %d not supported!\n",
|
|
ext_amode);
|
|
continue;
|
|
}
|
|
|
|
/* much like core primary audio coding header */
|
|
dca_parse_audio_coding_header(s, s->xch_base_channel);
|
|
|
|
for (i = 0; i < (s->sample_blocks / 8); i++)
|
|
if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
|
|
av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
|
|
continue;
|
|
}
|
|
|
|
s->xch_present = 1;
|
|
break;
|
|
}
|
|
case DCA_SYNCWORD_XXCH:
|
|
/* XXCh: extended channels */
|
|
/* usually found either in core or HD part in DTS-HD HRA streams,
|
|
* but not in DTS-ES which contains XCh extensions instead */
|
|
s->core_ext_mask |= DCA_EXT_XXCH;
|
|
break;
|
|
|
|
case 0x1d95f262: {
|
|
int fsize96 = show_bits(&s->gb, 12) + 1;
|
|
if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
|
|
continue;
|
|
|
|
av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",
|
|
get_bits_count(&s->gb));
|
|
skip_bits(&s->gb, 12);
|
|
av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
|
|
av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
|
|
|
|
s->core_ext_mask |= DCA_EXT_X96;
|
|
break;
|
|
}
|
|
}
|
|
|
|
skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
|
|
}
|
|
} else {
|
|
/* no supported extensions, skip the rest of the core substream */
|
|
skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));
|
|
}
|
|
|
|
if (s->core_ext_mask & DCA_EXT_X96)
|
|
s->profile = FF_PROFILE_DTS_96_24;
|
|
else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))
|
|
s->profile = FF_PROFILE_DTS_ES;
|
|
|
|
/* check for ExSS (HD part) */
|
|
if (s->dca_buffer_size - s->frame_size > 32 &&
|
|
get_bits_long(&s->gb, 32) == DCA_SYNCWORD_SUBSTREAM)
|
|
ff_dca_exss_parse_header(s);
|
|
|
|
avctx->profile = s->profile;
|
|
|
|
full_channels = channels = s->prim_channels + !!s->lfe;
|
|
|
|
if (s->amode < 16) {
|
|
avctx->channel_layout = dca_core_channel_layout[s->amode];
|
|
|
|
if (s->prim_channels + !!s->lfe > 2 &&
|
|
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
|
|
/*
|
|
* Neither the core's auxiliary data nor our default tables contain
|
|
* downmix coefficients for the additional channel coded in the XCh
|
|
* extension, so when we're doing a Stereo downmix, don't decode it.
|
|
*/
|
|
s->xch_disable = 1;
|
|
}
|
|
|
|
#if FF_API_REQUEST_CHANNELS
|
|
FF_DISABLE_DEPRECATION_WARNINGS
|
|
if (s->xch_present && !s->xch_disable &&
|
|
(!avctx->request_channels ||
|
|
avctx->request_channels > num_core_channels + !!s->lfe)) {
|
|
FF_ENABLE_DEPRECATION_WARNINGS
|
|
#else
|
|
if (s->xch_present && !s->xch_disable) {
|
|
#endif
|
|
avctx->channel_layout |= AV_CH_BACK_CENTER;
|
|
if (s->lfe) {
|
|
avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
|
|
s->channel_order_tab = ff_dca_channel_reorder_lfe_xch[s->amode];
|
|
} else {
|
|
s->channel_order_tab = ff_dca_channel_reorder_nolfe_xch[s->amode];
|
|
}
|
|
} else {
|
|
channels = num_core_channels + !!s->lfe;
|
|
s->xch_present = 0; /* disable further xch processing */
|
|
if (s->lfe) {
|
|
avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
|
|
s->channel_order_tab = ff_dca_channel_reorder_lfe[s->amode];
|
|
} else
|
|
s->channel_order_tab = ff_dca_channel_reorder_nolfe[s->amode];
|
|
}
|
|
|
|
if (channels > !!s->lfe &&
|
|
s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
if (num_core_channels + !!s->lfe > 2 &&
|
|
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
|
|
channels = 2;
|
|
s->output = s->prim_channels == 2 ? s->amode : DCA_STEREO;
|
|
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
|
|
|
|
/* Stereo downmix coefficients
|
|
*
|
|
* The decoder can only downmix to 2-channel, so we need to ensure
|
|
* embedded downmix coefficients are actually targeting 2-channel.
|
|
*/
|
|
if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||
|
|
s->core_downmix_amode == DCA_STEREO_TOTAL)) {
|
|
for (i = 0; i < num_core_channels + !!s->lfe; i++) {
|
|
/* Range checked earlier */
|
|
s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]);
|
|
s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]);
|
|
}
|
|
s->output = s->core_downmix_amode;
|
|
} else {
|
|
int am = s->amode & DCA_CHANNEL_MASK;
|
|
if (am >= FF_ARRAY_ELEMS(ff_dca_default_coeffs)) {
|
|
av_log(s->avctx, AV_LOG_ERROR,
|
|
"Invalid channel mode %d\n", am);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
if (num_core_channels + !!s->lfe >
|
|
FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) {
|
|
avpriv_request_sample(s->avctx, "Downmixing %d channels",
|
|
s->prim_channels + !!s->lfe);
|
|
return AVERROR_PATCHWELCOME;
|
|
}
|
|
for (i = 0; i < num_core_channels + !!s->lfe; i++) {
|
|
s->downmix_coef[i][0] = ff_dca_default_coeffs[am][i][0];
|
|
s->downmix_coef[i][1] = ff_dca_default_coeffs[am][i][1];
|
|
}
|
|
}
|
|
av_dlog(s->avctx, "Stereo downmix coeffs:\n");
|
|
for (i = 0; i < num_core_channels + !!s->lfe; i++) {
|
|
av_dlog(s->avctx, "L, input channel %d = %f\n", i,
|
|
s->downmix_coef[i][0]);
|
|
av_dlog(s->avctx, "R, input channel %d = %f\n", i,
|
|
s->downmix_coef[i][1]);
|
|
}
|
|
av_dlog(s->avctx, "\n");
|
|
}
|
|
} else {
|
|
av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
avctx->channels = channels;
|
|
|
|
/* get output buffer */
|
|
frame->nb_samples = 256 * (s->sample_blocks / 8);
|
|
if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
|
|
int xll_nb_samples = s->xll_segments * s->xll_smpl_in_seg;
|
|
/* Check for invalid/unsupported conditions first */
|
|
if (s->xll_residual_channels > channels) {
|
|
av_log(s->avctx, AV_LOG_WARNING,
|
|
"DCA: too many residual channels (%d, core channels %d). Disabling XLL\n",
|
|
s->xll_residual_channels, channels);
|
|
s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
|
|
} else if (xll_nb_samples != frame->nb_samples &&
|
|
2 * frame->nb_samples != xll_nb_samples) {
|
|
av_log(s->avctx, AV_LOG_WARNING,
|
|
"DCA: unsupported upsampling (%d XLL samples, %d core samples). Disabling XLL\n",
|
|
xll_nb_samples, frame->nb_samples);
|
|
s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
|
|
} else {
|
|
if (2 * frame->nb_samples == xll_nb_samples) {
|
|
av_log(s->avctx, AV_LOG_INFO,
|
|
"XLL: upsampling core channels by a factor of 2\n");
|
|
upsample = 1;
|
|
|
|
frame->nb_samples = xll_nb_samples;
|
|
// FIXME: Is it good enough to copy from the first channel set?
|
|
avctx->sample_rate = s->xll_chsets[0].sampling_frequency;
|
|
}
|
|
/* If downmixing to stereo, don't decode additional channels.
|
|
* FIXME: Using the xch_disable flag for this doesn't seem right. */
|
|
if (!s->xch_disable)
|
|
avctx->channels += s->xll_channels - s->xll_residual_channels;
|
|
}
|
|
}
|
|
|
|
/* FIXME: This is an ugly hack, to just revert to the default
|
|
* layout if we have additional channels. Need to convert the XLL
|
|
* channel masks to libav channel_layout mask. */
|
|
if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels)
|
|
avctx->channel_layout = 0;
|
|
|
|
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
|
|
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
|
|
return ret;
|
|
}
|
|
samples_flt = (float **) frame->extended_data;
|
|
|
|
/* allocate buffer for extra channels if downmixing */
|
|
if (avctx->channels < full_channels) {
|
|
ret = av_samples_get_buffer_size(NULL, full_channels - channels,
|
|
frame->nb_samples,
|
|
avctx->sample_fmt, 0);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
av_fast_malloc(&s->extra_channels_buffer,
|
|
&s->extra_channels_buffer_size, ret);
|
|
if (!s->extra_channels_buffer)
|
|
return AVERROR(ENOMEM);
|
|
|
|
ret = av_samples_fill_arrays((uint8_t **) s->extra_channels, NULL,
|
|
s->extra_channels_buffer,
|
|
full_channels - channels,
|
|
frame->nb_samples, avctx->sample_fmt, 0);
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
|
|
/* filter to get final output */
|
|
for (i = 0; i < (s->sample_blocks / 8); i++) {
|
|
int ch;
|
|
unsigned block = upsample ? 512 : 256;
|
|
for (ch = 0; ch < channels; ch++)
|
|
s->samples_chanptr[ch] = samples_flt[ch] + i * block;
|
|
for (; ch < full_channels; ch++)
|
|
s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * block;
|
|
|
|
dca_filter_channels(s, i, upsample);
|
|
|
|
/* If this was marked as a DTS-ES stream we need to subtract back- */
|
|
/* channel from SL & SR to remove matrixed back-channel signal */
|
|
if ((s->source_pcm_res & 1) && s->xch_present) {
|
|
float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];
|
|
float *lt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];
|
|
float *rt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];
|
|
s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
|
|
s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
|
|
}
|
|
}
|
|
|
|
/* update lfe history */
|
|
lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);
|
|
for (i = 0; i < 2 * s->lfe * 4; i++)
|
|
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
|
|
|
|
if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
|
|
ret = ff_dca_xll_decode_audio(s, frame);
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
/* AVMatrixEncoding
|
|
*
|
|
* DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
|
|
ret = ff_side_data_update_matrix_encoding(frame,
|
|
(s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?
|
|
AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
*got_frame_ptr = 1;
|
|
|
|
return buf_size;
|
|
}
|
|
|
|
/**
|
|
* DCA initialization
|
|
*
|
|
* @param avctx pointer to the AVCodecContext
|
|
*/
|
|
|
|
static av_cold int dca_decode_init(AVCodecContext *avctx)
|
|
{
|
|
DCAContext *s = avctx->priv_data;
|
|
|
|
s->avctx = avctx;
|
|
dca_init_vlcs();
|
|
|
|
avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
|
|
ff_mdct_init(&s->imdct, 6, 1, 1.0);
|
|
ff_synth_filter_init(&s->synth);
|
|
ff_dcadsp_init(&s->dcadsp);
|
|
ff_fmt_convert_init(&s->fmt_conv, avctx);
|
|
|
|
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
|
|
|
|
/* allow downmixing to stereo */
|
|
#if FF_API_REQUEST_CHANNELS
|
|
FF_DISABLE_DEPRECATION_WARNINGS
|
|
if (avctx->request_channels == 2)
|
|
avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;
|
|
FF_ENABLE_DEPRECATION_WARNINGS
|
|
#endif
|
|
if (avctx->channels > 2 &&
|
|
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
|
|
avctx->channels = 2;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int dca_decode_end(AVCodecContext *avctx)
|
|
{
|
|
DCAContext *s = avctx->priv_data;
|
|
ff_mdct_end(&s->imdct);
|
|
av_freep(&s->extra_channels_buffer);
|
|
av_freep(&s->xll_sample_buf);
|
|
av_freep(&s->qmf64_table);
|
|
return 0;
|
|
}
|
|
|
|
static const AVProfile profiles[] = {
|
|
{ FF_PROFILE_DTS, "DTS" },
|
|
{ FF_PROFILE_DTS_ES, "DTS-ES" },
|
|
{ FF_PROFILE_DTS_96_24, "DTS 96/24" },
|
|
{ FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },
|
|
{ FF_PROFILE_DTS_HD_MA, "DTS-HD MA" },
|
|
{ FF_PROFILE_UNKNOWN },
|
|
};
|
|
|
|
static const AVOption options[] = {
|
|
{ "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
|
|
{ "disable_xll", "disable decoding of the XLL extension", offsetof(DCAContext, xll_disable), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
|
|
{ NULL },
|
|
};
|
|
|
|
static const AVClass dca_decoder_class = {
|
|
.class_name = "DCA decoder",
|
|
.item_name = av_default_item_name,
|
|
.option = options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
AVCodec ff_dca_decoder = {
|
|
.name = "dca",
|
|
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_DTS,
|
|
.priv_data_size = sizeof(DCAContext),
|
|
.init = dca_decode_init,
|
|
.decode = dca_decode_frame,
|
|
.close = dca_decode_end,
|
|
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
|
|
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
|
|
AV_SAMPLE_FMT_NONE },
|
|
.profiles = NULL_IF_CONFIG_SMALL(profiles),
|
|
.priv_class = &dca_decoder_class,
|
|
};
|