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9aa8193a23
decoders. Based on patches by clsid2 in ffdshow-tryout.
1934 lines
70 KiB
C
1934 lines
70 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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*
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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/common.h"
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#include "libavutil/intmath.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/audioconvert.h"
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#include "avcodec.h"
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#include "dsputil.h"
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#include "fft.h"
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#include "get_bits.h"
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#include "put_bits.h"
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#include "dcadata.h"
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#include "dcahuff.h"
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#include "dca.h"
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#include "synth_filter.h"
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#include "dcadsp.h"
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#include "fmtconvert.h"
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//#define TRACE
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#define DCA_PRIM_CHANNELS_MAX (7)
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#define DCA_SUBBANDS (32)
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#define DCA_ABITS_MAX (32) /* Should be 28 */
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#define DCA_SUBSUBFRAMES_MAX (4)
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#define DCA_SUBFRAMES_MAX (16)
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#define DCA_BLOCKS_MAX (16)
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#define DCA_LFE_MAX (3)
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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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/* these are unconfirmed but should be mostly correct */
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enum DCAExSSSpeakerMask {
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DCA_EXSS_FRONT_CENTER = 0x0001,
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DCA_EXSS_FRONT_LEFT_RIGHT = 0x0002,
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DCA_EXSS_SIDE_REAR_LEFT_RIGHT = 0x0004,
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DCA_EXSS_LFE = 0x0008,
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DCA_EXSS_REAR_CENTER = 0x0010,
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DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,
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DCA_EXSS_REAR_LEFT_RIGHT = 0x0040,
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DCA_EXSS_FRONT_HIGH_CENTER = 0x0080,
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DCA_EXSS_OVERHEAD = 0x0100,
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DCA_EXSS_CENTER_LEFT_RIGHT = 0x0200,
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DCA_EXSS_WIDE_LEFT_RIGHT = 0x0400,
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DCA_EXSS_SIDE_LEFT_RIGHT = 0x0800,
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DCA_EXSS_LFE2 = 0x1000,
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DCA_EXSS_SIDE_HIGH_LEFT_RIGHT = 0x2000,
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DCA_EXSS_REAR_HIGH_CENTER = 0x4000,
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DCA_EXSS_REAR_HIGH_LEFT_RIGHT = 0x8000,
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};
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enum DCAExtensionMask {
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DCA_EXT_CORE = 0x001, ///< core in core substream
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DCA_EXT_XXCH = 0x002, ///< XXCh channels extension in core substream
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DCA_EXT_X96 = 0x004, ///< 96/24 extension in core substream
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DCA_EXT_XCH = 0x008, ///< XCh channel extension in core substream
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DCA_EXT_EXSS_CORE = 0x010, ///< core in ExSS (extension substream)
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DCA_EXT_EXSS_XBR = 0x020, ///< extended bitrate extension in ExSS
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DCA_EXT_EXSS_XXCH = 0x040, ///< XXCh channels extension in ExSS
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DCA_EXT_EXSS_X96 = 0x080, ///< 96/24 extension in ExSS
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DCA_EXT_EXSS_LBR = 0x100, ///< low bitrate component in ExSS
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DCA_EXT_EXSS_XLL = 0x200, ///< lossless extension in ExSS
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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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/* extensions that reside in core substream */
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#define DCA_CORE_EXTS (DCA_EXT_XCH | DCA_EXT_XXCH | DCA_EXT_X96)
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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 int64_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|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|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|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|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_BACK_CENTER|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|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_LAYOUT_STEREO|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|AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT|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|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_LAYOUT_STEREO|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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static const int8_t dca_lfe_index[] = {
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1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
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};
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static const int8_t dca_channel_reorder_lfe[][9] = {
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{ 0, -1, -1, -1, -1, -1, -1, -1, -1},
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
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{ 2, 0, 1, -1, -1, -1, -1, -1, -1},
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{ 0, 1, 3, -1, -1, -1, -1, -1, -1},
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{ 2, 0, 1, 4, -1, -1, -1, -1, -1},
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{ 0, 1, 3, 4, -1, -1, -1, -1, -1},
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{ 2, 0, 1, 4, 5, -1, -1, -1, -1},
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{ 3, 4, 0, 1, 5, 6, -1, -1, -1},
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{ 2, 0, 1, 4, 5, 6, -1, -1, -1},
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{ 0, 6, 4, 5, 2, 3, -1, -1, -1},
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{ 4, 2, 5, 0, 1, 6, 7, -1, -1},
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{ 5, 6, 0, 1, 7, 3, 8, 4, -1},
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{ 4, 2, 5, 0, 1, 6, 8, 7, -1},
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};
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static const int8_t dca_channel_reorder_lfe_xch[][9] = {
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{ 0, 2, -1, -1, -1, -1, -1, -1, -1},
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{ 0, 1, 3, -1, -1, -1, -1, -1, -1},
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{ 0, 1, 3, -1, -1, -1, -1, -1, -1},
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{ 0, 1, 3, -1, -1, -1, -1, -1, -1},
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{ 0, 1, 3, -1, -1, -1, -1, -1, -1},
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{ 2, 0, 1, 4, -1, -1, -1, -1, -1},
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{ 0, 1, 3, 4, -1, -1, -1, -1, -1},
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{ 2, 0, 1, 4, 5, -1, -1, -1, -1},
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{ 0, 1, 4, 5, 3, -1, -1, -1, -1},
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{ 2, 0, 1, 5, 6, 4, -1, -1, -1},
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{ 3, 4, 0, 1, 6, 7, 5, -1, -1},
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{ 2, 0, 1, 4, 5, 6, 7, -1, -1},
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{ 0, 6, 4, 5, 2, 3, 7, -1, -1},
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{ 4, 2, 5, 0, 1, 7, 8, 6, -1},
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{ 5, 6, 0, 1, 8, 3, 9, 4, 7},
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{ 4, 2, 5, 0, 1, 6, 9, 8, 7},
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};
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static const int8_t dca_channel_reorder_nolfe[][9] = {
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{ 0, -1, -1, -1, -1, -1, -1, -1, -1},
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
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{ 2, 0, 1, -1, -1, -1, -1, -1, -1},
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{ 0, 1, 2, -1, -1, -1, -1, -1, -1},
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{ 2, 0, 1, 3, -1, -1, -1, -1, -1},
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{ 0, 1, 2, 3, -1, -1, -1, -1, -1},
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{ 2, 0, 1, 3, 4, -1, -1, -1, -1},
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{ 2, 3, 0, 1, 4, 5, -1, -1, -1},
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{ 2, 0, 1, 3, 4, 5, -1, -1, -1},
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{ 0, 5, 3, 4, 1, 2, -1, -1, -1},
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{ 3, 2, 4, 0, 1, 5, 6, -1, -1},
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{ 4, 5, 0, 1, 6, 2, 7, 3, -1},
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{ 3, 2, 4, 0, 1, 5, 7, 6, -1},
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};
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static const int8_t dca_channel_reorder_nolfe_xch[][9] = {
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
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{ 0, 1, 2, -1, -1, -1, -1, -1, -1},
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{ 0, 1, 2, -1, -1, -1, -1, -1, -1},
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{ 0, 1, 2, -1, -1, -1, -1, -1, -1},
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{ 0, 1, 2, -1, -1, -1, -1, -1, -1},
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{ 2, 0, 1, 3, -1, -1, -1, -1, -1},
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{ 0, 1, 2, 3, -1, -1, -1, -1, -1},
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{ 2, 0, 1, 3, 4, -1, -1, -1, -1},
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{ 0, 1, 3, 4, 2, -1, -1, -1, -1},
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{ 2, 0, 1, 4, 5, 3, -1, -1, -1},
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{ 2, 3, 0, 1, 5, 6, 4, -1, -1},
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{ 2, 0, 1, 3, 4, 5, 6, -1, -1},
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{ 0, 5, 3, 4, 1, 2, 6, -1, -1},
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{ 3, 2, 4, 0, 1, 6, 7, 5, -1},
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{ 4, 5, 0, 1, 7, 2, 8, 3, 6},
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{ 3, 2, 4, 0, 1, 5, 8, 7, 6},
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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_MAX_FRAME_SIZE 16384
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#define DCA_MAX_EXSS_HEADER_SIZE 4096
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#define DCA_BUFFER_PADDING_SIZE 1024
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/** Bit allocation */
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typedef struct {
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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, int idx)
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{
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return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
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}
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typedef struct {
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AVCodecContext *avctx;
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/* Frame header */
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int frame_type; ///< type of the current frame
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int samples_deficit; ///< deficit sample count
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int crc_present; ///< crc is present in the bitstream
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int sample_blocks; ///< number of PCM sample blocks
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int frame_size; ///< primary frame byte size
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int amode; ///< audio channels arrangement
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int sample_rate; ///< audio sampling rate
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int bit_rate; ///< transmission bit rate
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int bit_rate_index; ///< transmission bit rate index
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int downmix; ///< embedded downmix enabled
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int dynrange; ///< embedded dynamic range flag
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int timestamp; ///< embedded time stamp flag
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int aux_data; ///< auxiliary data flag
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int hdcd; ///< source material is mastered in HDCD
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int ext_descr; ///< extension audio descriptor flag
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int ext_coding; ///< extended coding flag
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int aspf; ///< audio sync word insertion flag
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int lfe; ///< low frequency effects flag
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int predictor_history; ///< predictor history flag
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int header_crc; ///< header crc check bytes
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int multirate_inter; ///< multirate interpolator switch
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int version; ///< encoder software revision
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int copy_history; ///< copy history
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int source_pcm_res; ///< source pcm resolution
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int front_sum; ///< front sum/difference flag
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int surround_sum; ///< surround sum/difference flag
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int dialog_norm; ///< dialog normalisation parameter
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/* Primary audio coding header */
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int subframes; ///< number of subframes
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int is_channels_set; ///< check for if the channel number is already set
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int total_channels; ///< number of channels including extensions
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int prim_channels; ///< number of primary audio channels
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int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count
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int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband
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int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index
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int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book
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int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
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int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select
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int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
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float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
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/* Primary audio coding side information */
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int subsubframes[DCA_SUBFRAMES_MAX]; ///< number of subsubframes
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int partial_samples[DCA_SUBFRAMES_MAX]; ///< partial subsubframe samples count
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int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not)
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int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs
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int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index
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int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients)
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int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient)
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int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook
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int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
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int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients
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int dynrange_coef; ///< dynamic range coefficient
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int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
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float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)]; ///< Low frequency effect data
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int lfe_scale_factor;
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/* Subband samples history (for ADPCM) */
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float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
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DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
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DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
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int hist_index[DCA_PRIM_CHANNELS_MAX];
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DECLARE_ALIGNED(32, float, raXin)[32];
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int output; ///< type of output
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float scale_bias; ///< output scale
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DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
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DECLARE_ALIGNED(32, float, samples)[(DCA_PRIM_CHANNELS_MAX+1)*256];
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const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX+1];
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uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE + DCA_BUFFER_PADDING_SIZE];
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int dca_buffer_size; ///< how much data is in the dca_buffer
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const int8_t* channel_order_tab; ///< channel reordering table, lfe and non lfe
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GetBitContext gb;
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/* Current position in DCA frame */
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int current_subframe;
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int current_subsubframe;
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int core_ext_mask; ///< present extensions in the core substream
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/* XCh extension information */
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int xch_present; ///< XCh extension present and valid
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int xch_base_channel; ///< index of first (only) channel containing XCH data
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/* ExSS header parser */
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int static_fields; ///< static fields present
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int mix_metadata; ///< mixing metadata present
|
|
int num_mix_configs; ///< number of mix out configurations
|
|
int mix_config_num_ch[4]; ///< number of channels in each mix out configuration
|
|
|
|
int profile;
|
|
|
|
int debug_flag; ///< used for suppressing repeated error messages output
|
|
DSPContext dsp;
|
|
FFTContext imdct;
|
|
SynthFilterContext synth;
|
|
DCADSPContext dcadsp;
|
|
FmtConvertContext fmt_conv;
|
|
} DCAContext;
|
|
|
|
static const uint16_t dca_vlc_offs[] = {
|
|
0, 512, 640, 768, 1282, 1794, 2436, 3080, 3770, 4454, 5364,
|
|
5372, 5380, 5388, 5392, 5396, 5412, 5420, 5428, 5460, 5492, 5508,
|
|
5572, 5604, 5668, 5796, 5860, 5892, 6412, 6668, 6796, 7308, 7564,
|
|
7820, 8076, 8620, 9132, 9388, 9910, 10166, 10680, 11196, 11726, 12240,
|
|
12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,
|
|
18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,
|
|
};
|
|
|
|
static av_cold void dca_init_vlcs(void)
|
|
{
|
|
static int vlcs_initialized = 0;
|
|
int i, j, c = 14;
|
|
static VLC_TYPE dca_table[23622][2];
|
|
|
|
if (vlcs_initialized)
|
|
return;
|
|
|
|
dca_bitalloc_index.offset = 1;
|
|
dca_bitalloc_index.wrap = 2;
|
|
for (i = 0; i < 5; i++) {
|
|
dca_bitalloc_index.vlc[i].table = &dca_table[dca_vlc_offs[i]];
|
|
dca_bitalloc_index.vlc[i].table_allocated = dca_vlc_offs[i + 1] - dca_vlc_offs[i];
|
|
init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
|
|
bitalloc_12_bits[i], 1, 1,
|
|
bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
|
|
}
|
|
dca_scalefactor.offset = -64;
|
|
dca_scalefactor.wrap = 2;
|
|
for (i = 0; i < 5; i++) {
|
|
dca_scalefactor.vlc[i].table = &dca_table[dca_vlc_offs[i + 5]];
|
|
dca_scalefactor.vlc[i].table_allocated = dca_vlc_offs[i + 6] - dca_vlc_offs[i + 5];
|
|
init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
|
|
scales_bits[i], 1, 1,
|
|
scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
|
|
}
|
|
dca_tmode.offset = 0;
|
|
dca_tmode.wrap = 1;
|
|
for (i = 0; i < 4; i++) {
|
|
dca_tmode.vlc[i].table = &dca_table[dca_vlc_offs[i + 10]];
|
|
dca_tmode.vlc[i].table_allocated = dca_vlc_offs[i + 11] - dca_vlc_offs[i + 10];
|
|
init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
|
|
tmode_bits[i], 1, 1,
|
|
tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
|
|
}
|
|
|
|
for (i = 0; i < 10; i++)
|
|
for (j = 0; j < 7; j++){
|
|
if (!bitalloc_codes[i][j]) break;
|
|
dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
|
|
dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
|
|
dca_smpl_bitalloc[i+1].vlc[j].table = &dca_table[dca_vlc_offs[c]];
|
|
dca_smpl_bitalloc[i+1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
|
|
init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],
|
|
bitalloc_sizes[i],
|
|
bitalloc_bits[i][j], 1, 1,
|
|
bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
|
|
c++;
|
|
}
|
|
vlcs_initialized = 1;
|
|
}
|
|
|
|
static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
|
|
{
|
|
while(len--)
|
|
*dst++ = get_bits(gb, bits);
|
|
}
|
|
|
|
static int dca_parse_audio_coding_header(DCAContext * s, int base_channel)
|
|
{
|
|
int i, j;
|
|
static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
|
|
static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
|
|
static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
|
|
|
|
s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
|
|
s->prim_channels = s->total_channels;
|
|
|
|
if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
|
|
s->prim_channels = DCA_PRIM_CHANNELS_MAX;
|
|
|
|
|
|
for (i = base_channel; i < s->prim_channels; i++) {
|
|
s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
|
|
if (s->subband_activity[i] > DCA_SUBBANDS)
|
|
s->subband_activity[i] = DCA_SUBBANDS;
|
|
}
|
|
for (i = base_channel; i < s->prim_channels; i++) {
|
|
s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
|
|
if (s->vq_start_subband[i] > DCA_SUBBANDS)
|
|
s->vq_start_subband[i] = DCA_SUBBANDS;
|
|
}
|
|
get_array(&s->gb, s->joint_intensity + base_channel, s->prim_channels - base_channel, 3);
|
|
get_array(&s->gb, s->transient_huffman + base_channel, s->prim_channels - base_channel, 2);
|
|
get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3);
|
|
get_array(&s->gb, s->bitalloc_huffman + base_channel, s->prim_channels - base_channel, 3);
|
|
|
|
/* Get codebooks quantization indexes */
|
|
if (!base_channel)
|
|
memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
|
|
for (j = 1; j < 11; j++)
|
|
for (i = base_channel; i < s->prim_channels; i++)
|
|
s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
|
|
|
|
/* Get scale factor adjustment */
|
|
for (j = 0; j < 11; j++)
|
|
for (i = base_channel; i < s->prim_channels; i++)
|
|
s->scalefactor_adj[i][j] = 1;
|
|
|
|
for (j = 1; j < 11; j++)
|
|
for (i = base_channel; i < s->prim_channels; i++)
|
|
if (s->quant_index_huffman[i][j] < thr[j])
|
|
s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
|
|
|
|
if (s->crc_present) {
|
|
/* Audio header CRC check */
|
|
get_bits(&s->gb, 16);
|
|
}
|
|
|
|
s->current_subframe = 0;
|
|
s->current_subsubframe = 0;
|
|
|
|
#ifdef TRACE
|
|
av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
|
|
for (i = base_channel; i < s->prim_channels; i++){
|
|
av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
|
|
for (j = 0; j < 11; j++)
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %i",
|
|
s->quant_index_huffman[i][j]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
|
|
for (j = 0; j < 11; j++)
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
}
|
|
#endif
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int dca_parse_frame_header(DCAContext * s)
|
|
{
|
|
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
|
|
|
|
/* Sync code */
|
|
get_bits(&s->gb, 32);
|
|
|
|
/* Frame header */
|
|
s->frame_type = get_bits(&s->gb, 1);
|
|
s->samples_deficit = get_bits(&s->gb, 5) + 1;
|
|
s->crc_present = get_bits(&s->gb, 1);
|
|
s->sample_blocks = get_bits(&s->gb, 7) + 1;
|
|
s->frame_size = get_bits(&s->gb, 14) + 1;
|
|
if (s->frame_size < 95)
|
|
return -1;
|
|
s->amode = get_bits(&s->gb, 6);
|
|
s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)];
|
|
if (!s->sample_rate)
|
|
return -1;
|
|
s->bit_rate_index = get_bits(&s->gb, 5);
|
|
s->bit_rate = dca_bit_rates[s->bit_rate_index];
|
|
if (!s->bit_rate)
|
|
return -1;
|
|
|
|
s->downmix = get_bits(&s->gb, 1);
|
|
s->dynrange = get_bits(&s->gb, 1);
|
|
s->timestamp = get_bits(&s->gb, 1);
|
|
s->aux_data = get_bits(&s->gb, 1);
|
|
s->hdcd = get_bits(&s->gb, 1);
|
|
s->ext_descr = get_bits(&s->gb, 3);
|
|
s->ext_coding = get_bits(&s->gb, 1);
|
|
s->aspf = get_bits(&s->gb, 1);
|
|
s->lfe = get_bits(&s->gb, 2);
|
|
s->predictor_history = get_bits(&s->gb, 1);
|
|
|
|
/* TODO: check CRC */
|
|
if (s->crc_present)
|
|
s->header_crc = get_bits(&s->gb, 16);
|
|
|
|
s->multirate_inter = get_bits(&s->gb, 1);
|
|
s->version = get_bits(&s->gb, 4);
|
|
s->copy_history = get_bits(&s->gb, 2);
|
|
s->source_pcm_res = get_bits(&s->gb, 3);
|
|
s->front_sum = get_bits(&s->gb, 1);
|
|
s->surround_sum = get_bits(&s->gb, 1);
|
|
s->dialog_norm = get_bits(&s->gb, 4);
|
|
|
|
/* FIXME: channels mixing levels */
|
|
s->output = s->amode;
|
|
if (s->lfe) s->output |= DCA_LFE;
|
|
|
|
#ifdef TRACE
|
|
av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n",
|
|
s->sample_blocks, s->sample_blocks * 32);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n",
|
|
s->amode, dca_channels[s->amode]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i Hz\n",
|
|
s->sample_rate);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i bits/s\n",
|
|
s->bit_rate);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n",
|
|
s->predictor_history);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n",
|
|
s->multirate_inter);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history);
|
|
av_log(s->avctx, AV_LOG_DEBUG,
|
|
"source pcm resolution: %i (%i bits/sample)\n",
|
|
s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
#endif
|
|
|
|
/* Primary audio coding header */
|
|
s->subframes = get_bits(&s->gb, 4) + 1;
|
|
|
|
return dca_parse_audio_coding_header(s, 0);
|
|
}
|
|
|
|
|
|
static inline int get_scale(GetBitContext *gb, int level, int value)
|
|
{
|
|
if (level < 5) {
|
|
/* huffman encoded */
|
|
value += get_bitalloc(gb, &dca_scalefactor, level);
|
|
} else if (level < 8)
|
|
value = get_bits(gb, level + 1);
|
|
return value;
|
|
}
|
|
|
|
static int dca_subframe_header(DCAContext * s, int base_channel, int block_index)
|
|
{
|
|
/* Primary audio coding side information */
|
|
int j, k;
|
|
|
|
if (get_bits_left(&s->gb) < 0)
|
|
return -1;
|
|
|
|
if (!base_channel) {
|
|
s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
|
|
s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
|
|
}
|
|
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
for (k = 0; k < s->subband_activity[j]; k++)
|
|
s->prediction_mode[j][k] = get_bits(&s->gb, 1);
|
|
}
|
|
|
|
/* Get prediction codebook */
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
for (k = 0; k < s->subband_activity[j]; k++) {
|
|
if (s->prediction_mode[j][k] > 0) {
|
|
/* (Prediction coefficient VQ address) */
|
|
s->prediction_vq[j][k] = get_bits(&s->gb, 12);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Bit allocation index */
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
for (k = 0; k < s->vq_start_subband[j]; k++) {
|
|
if (s->bitalloc_huffman[j] == 6)
|
|
s->bitalloc[j][k] = get_bits(&s->gb, 5);
|
|
else if (s->bitalloc_huffman[j] == 5)
|
|
s->bitalloc[j][k] = get_bits(&s->gb, 4);
|
|
else if (s->bitalloc_huffman[j] == 7) {
|
|
av_log(s->avctx, AV_LOG_ERROR,
|
|
"Invalid bit allocation index\n");
|
|
return -1;
|
|
} else {
|
|
s->bitalloc[j][k] =
|
|
get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
|
|
}
|
|
|
|
if (s->bitalloc[j][k] > 26) {
|
|
// av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n",
|
|
// j, k, s->bitalloc[j][k]);
|
|
return -1;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* 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 -1;
|
|
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
const uint32_t *scale_table;
|
|
int scale_sum;
|
|
|
|
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 = scale_factor_quant7;
|
|
else
|
|
scale_table = scale_factor_quant6;
|
|
|
|
/* 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);
|
|
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);
|
|
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 -1;
|
|
|
|
/* 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], 0);
|
|
scale += 64; /* bias */
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Stereo downmix coefficients */
|
|
if (!base_channel && s->prim_channels > 2) {
|
|
if (s->downmix) {
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
s->downmix_coef[j][0] = get_bits(&s->gb, 7);
|
|
s->downmix_coef[j][1] = get_bits(&s->gb, 7);
|
|
}
|
|
} else {
|
|
int am = s->amode & DCA_CHANNEL_MASK;
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];
|
|
s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];
|
|
}
|
|
}
|
|
}
|
|
|
|
/* 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 */
|
|
s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)];
|
|
|
|
/* 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;
|
|
}
|
|
|
|
#ifdef TRACE
|
|
av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes[s->current_subframe]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
|
|
s->partial_samples[s->current_subframe]);
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
|
|
for (k = 0; k < s->subband_activity[j]; k++)
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
}
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
for (k = 0; k < s->subband_activity[j]; k++)
|
|
av_log(s->avctx, AV_LOG_DEBUG,
|
|
"prediction coefs: %f, %f, %f, %f\n",
|
|
(float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
|
|
(float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
|
|
(float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
|
|
(float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
|
|
}
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
|
|
for (k = 0; k < s->vq_start_subband[j]; k++)
|
|
av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
}
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");
|
|
for (k = 0; k < s->subband_activity[j]; k++)
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
}
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");
|
|
for (k = 0; k < s->subband_activity[j]; k++) {
|
|
if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]);
|
|
if (k < s->vq_start_subband[j] && s->transition_mode[j][k])
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]);
|
|
}
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
}
|
|
for (j = base_channel; j < s->prim_channels; j++) {
|
|
if (s->joint_intensity[j] > 0) {
|
|
int source_channel = s->joint_intensity[j] - 1;
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");
|
|
for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
}
|
|
}
|
|
if (!base_channel && s->prim_channels > 2 && s->downmix) {
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]);
|
|
}
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
}
|
|
for (j = base_channel; j < s->prim_channels; j++)
|
|
for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
|
|
av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
|
|
if (!base_channel && s->lfe) {
|
|
int lfe_samples = 2 * s->lfe * (4 + block_index);
|
|
int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
|
|
for (j = lfe_samples; j < lfe_end_sample; j++)
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
}
|
|
#endif
|
|
|
|
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 i;
|
|
|
|
int sb_act = s->subband_activity[chans];
|
|
int subindex;
|
|
|
|
scale *= sqrt(1/8.0);
|
|
|
|
/* Select filter */
|
|
if (!s->multirate_inter) /* Non-perfect reconstruction */
|
|
prCoeff = fir_32bands_nonperfect;
|
|
else /* Perfect reconstruction */
|
|
prCoeff = fir_32bands_perfect;
|
|
|
|
/* Reconstructed channel sample index */
|
|
for (subindex = 0; subindex < 8; subindex++) {
|
|
/* Load in one sample from each subband and clear inactive subbands */
|
|
for (i = 0; i < sb_act; i++){
|
|
uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ ((i-1)&2)<<30;
|
|
AV_WN32A(&s->raXin[i], v);
|
|
}
|
|
for (; i < 32; i++)
|
|
s->raXin[i] = 0.0;
|
|
|
|
s->synth.synth_filter_float(&s->imdct,
|
|
s->subband_fir_hist[chans], &s->hist_index[chans],
|
|
s->subband_fir_noidea[chans], prCoeff,
|
|
samples_out, s->raXin, scale);
|
|
samples_out+= 32;
|
|
|
|
}
|
|
}
|
|
|
|
static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
|
|
int num_deci_sample, float *samples_in,
|
|
float *samples_out, float scale)
|
|
{
|
|
/* 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 decifactor;
|
|
const float *prCoeff;
|
|
int deciindex;
|
|
|
|
/* Select decimation filter */
|
|
if (decimation_select == 1) {
|
|
decifactor = 64;
|
|
prCoeff = lfe_fir_128;
|
|
} else {
|
|
decifactor = 32;
|
|
prCoeff = lfe_fir_64;
|
|
}
|
|
/* Interpolation */
|
|
for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
|
|
s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor,
|
|
scale);
|
|
samples_in++;
|
|
samples_out += 2 * decifactor;
|
|
}
|
|
}
|
|
|
|
/* downmixing routines */
|
|
#define MIX_REAR1(samples, si1, rs, coef) \
|
|
samples[i] += samples[si1] * coef[rs][0]; \
|
|
samples[i+256] += samples[si1] * coef[rs][1];
|
|
|
|
#define MIX_REAR2(samples, si1, si2, rs, coef) \
|
|
samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \
|
|
samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1];
|
|
|
|
#define MIX_FRONT3(samples, coef) \
|
|
t = samples[i+c]; \
|
|
u = samples[i+l]; \
|
|
v = samples[i+r]; \
|
|
samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
|
|
samples[i+256] = 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 downmix_coef[DCA_PRIM_CHANNELS_MAX][2],
|
|
const int8_t *channel_mapping)
|
|
{
|
|
int c,l,r,sl,sr,s;
|
|
int i;
|
|
float t, u, v;
|
|
float coef[DCA_PRIM_CHANNELS_MAX][2];
|
|
|
|
for (i=0; i<DCA_PRIM_CHANNELS_MAX; i++) {
|
|
coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];
|
|
coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];
|
|
}
|
|
|
|
switch (srcfmt) {
|
|
case DCA_MONO:
|
|
case DCA_CHANNEL:
|
|
case DCA_STEREO_TOTAL:
|
|
case DCA_STEREO_SUMDIFF:
|
|
case DCA_4F2R:
|
|
av_log(NULL, 0, "Not implemented!\n");
|
|
break;
|
|
case DCA_STEREO:
|
|
break;
|
|
case DCA_3F:
|
|
c = channel_mapping[0] * 256;
|
|
l = channel_mapping[1] * 256;
|
|
r = channel_mapping[2] * 256;
|
|
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);
|
|
break;
|
|
case DCA_2F1R:
|
|
s = channel_mapping[2] * 256;
|
|
DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef),);
|
|
break;
|
|
case DCA_3F1R:
|
|
c = channel_mapping[0] * 256;
|
|
l = channel_mapping[1] * 256;
|
|
r = channel_mapping[2] * 256;
|
|
s = channel_mapping[3] * 256;
|
|
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
|
|
MIX_REAR1(samples, i + s, 3, coef));
|
|
break;
|
|
case DCA_2F2R:
|
|
sl = channel_mapping[2] * 256;
|
|
sr = channel_mapping[3] * 256;
|
|
DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef),);
|
|
break;
|
|
case DCA_3F2R:
|
|
c = channel_mapping[0] * 256;
|
|
l = channel_mapping[1] * 256;
|
|
r = channel_mapping[2] * 256;
|
|
sl = channel_mapping[3] * 256;
|
|
sr = channel_mapping[4] * 256;
|
|
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
|
|
MIX_REAR2(samples, i + sl, i + sr, 3, coef));
|
|
break;
|
|
}
|
|
}
|
|
|
|
|
|
/* 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, int *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;
|
|
}
|
|
|
|
if (code == 0)
|
|
return 0;
|
|
else {
|
|
av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
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(int, block, [8]);
|
|
|
|
/*
|
|
* Audio data
|
|
*/
|
|
|
|
/* Select quantization step size table */
|
|
if (s->bit_rate_index == 0x1f)
|
|
quant_step_table = lossless_quant_d;
|
|
else
|
|
quant_step_table = lossy_quant_d;
|
|
|
|
for (k = base_channel; k < s->prim_channels; k++) {
|
|
if (get_bits_left(&s->gb) < 0)
|
|
return -1;
|
|
|
|
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){
|
|
memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
|
|
} else {
|
|
/* Deal with transients */
|
|
int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
|
|
float rscale = 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;
|
|
|
|
size = abits_sizes[abits-1];
|
|
levels = abits_levels[abits-1];
|
|
|
|
block_code1 = get_bits(&s->gb, size);
|
|
/* FIXME Should test return value */
|
|
decode_blockcode(block_code1, levels, block);
|
|
block_code2 = get_bits(&s->gb, size);
|
|
decode_blockcode(block_code2, levels, &block[4]);
|
|
}else{
|
|
/* no coding */
|
|
for (m = 0; m < 8; m++)
|
|
block[m] = get_sbits(&s->gb, abits - 3);
|
|
}
|
|
}else{
|
|
/* Huffman coded */
|
|
for (m = 0; m < 8; m++)
|
|
block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
|
|
}
|
|
|
|
s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],
|
|
block, rscale, 8);
|
|
}
|
|
|
|
/*
|
|
* Inverse ADPCM if in prediction mode
|
|
*/
|
|
if (s->prediction_mode[k][l]) {
|
|
int n;
|
|
for (m = 0; m < 8; m++) {
|
|
for (n = 1; n <= 4; n++)
|
|
if (m >= n)
|
|
subband_samples[k][l][m] +=
|
|
(adpcm_vb[s->prediction_vq[k][l]][n - 1] *
|
|
subband_samples[k][l][m - n] / 8192);
|
|
else if (s->predictor_history)
|
|
subband_samples[k][l][m] +=
|
|
(adpcm_vb[s->prediction_vq[k][l]][n - 1] *
|
|
s->subband_samples_hist[k][l][m - n +
|
|
4] / 8192);
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Decode VQ encoded high frequencies
|
|
*/
|
|
for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {
|
|
/* 1 vector -> 32 samples but we only need the 8 samples
|
|
* for this subsubframe. */
|
|
int m;
|
|
|
|
if (!s->debug_flag & 0x01) {
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");
|
|
s->debug_flag |= 0x01;
|
|
}
|
|
|
|
for (m = 0; m < 8; m++) {
|
|
subband_samples[k][l][m] =
|
|
high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 +
|
|
m]
|
|
* (float) s->scale_factor[k][l][0] / 16.0;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Check for DSYNC after subsubframe */
|
|
if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
|
|
if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */
|
|
#ifdef TRACE
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
|
|
#endif
|
|
} else {
|
|
av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
|
|
}
|
|
}
|
|
|
|
/* Backup predictor history for adpcm */
|
|
for (k = base_channel; k < s->prim_channels; k++)
|
|
for (l = 0; l < s->vq_start_subband[k]; l++)
|
|
memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],
|
|
4 * sizeof(subband_samples[0][0][0]));
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int dca_filter_channels(DCAContext * s, int block_index)
|
|
{
|
|
float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
|
|
int k;
|
|
|
|
/* 32 subbands QMF */
|
|
for (k = 0; k < s->prim_channels; k++) {
|
|
/* static float pcm_to_double[8] =
|
|
{32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
|
|
qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]],
|
|
M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ );
|
|
}
|
|
|
|
/* Down mixing */
|
|
if (s->avctx->request_channels == 2 && s->prim_channels > 2) {
|
|
dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);
|
|
}
|
|
|
|
/* Generate LFE samples for this subsubframe FIXME!!! */
|
|
if (s->output & DCA_LFE) {
|
|
lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
|
|
s->lfe_data + 2 * s->lfe * (block_index + 4),
|
|
&s->samples[256 * dca_lfe_index[s->amode]],
|
|
(1.0/256.0)*s->scale_bias);
|
|
/* Outputs 20bits pcm samples */
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
static int dca_subframe_footer(DCAContext * s, int base_channel)
|
|
{
|
|
int aux_data_count = 0, i;
|
|
|
|
/*
|
|
* Unpack optional information
|
|
*/
|
|
|
|
/* presumably optional information only appears in the core? */
|
|
if (!base_channel) {
|
|
if (s->timestamp)
|
|
get_bits(&s->gb, 32);
|
|
|
|
if (s->aux_data)
|
|
aux_data_count = get_bits(&s->gb, 6);
|
|
|
|
for (i = 0; i < aux_data_count; i++)
|
|
get_bits(&s->gb, 8);
|
|
|
|
if (s->crc_present && (s->downmix || 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)
|
|
{
|
|
|
|
/* 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 -1;
|
|
}
|
|
|
|
if (!s->current_subsubframe) {
|
|
#ifdef TRACE
|
|
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
|
|
#endif
|
|
/* Read subframe header */
|
|
if (dca_subframe_header(s, base_channel, block_index))
|
|
return -1;
|
|
}
|
|
|
|
/* Read subsubframe */
|
|
#ifdef TRACE
|
|
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
|
|
#endif
|
|
if (dca_subsubframe(s, base_channel, block_index))
|
|
return -1;
|
|
|
|
/* 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) {
|
|
#ifdef TRACE
|
|
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
|
|
#endif
|
|
/* Read subframe footer */
|
|
if (dca_subframe_footer(s, base_channel))
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Convert bitstream to one representation based on sync marker
|
|
*/
|
|
static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,
|
|
int max_size)
|
|
{
|
|
uint32_t mrk;
|
|
int i, tmp;
|
|
const uint16_t *ssrc = (const uint16_t *) src;
|
|
uint16_t *sdst = (uint16_t *) dst;
|
|
PutBitContext pb;
|
|
|
|
if ((unsigned)src_size > (unsigned)max_size) {
|
|
// av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");
|
|
// return -1;
|
|
src_size = max_size;
|
|
}
|
|
|
|
mrk = AV_RB32(src);
|
|
switch (mrk) {
|
|
case DCA_MARKER_RAW_BE:
|
|
memcpy(dst, src, src_size);
|
|
return src_size;
|
|
case DCA_MARKER_RAW_LE:
|
|
for (i = 0; i < (src_size + 1) >> 1; i++)
|
|
*sdst++ = av_bswap16(*ssrc++);
|
|
return src_size;
|
|
case DCA_MARKER_14B_BE:
|
|
case DCA_MARKER_14B_LE:
|
|
init_put_bits(&pb, dst, max_size);
|
|
for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {
|
|
tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;
|
|
put_bits(&pb, 14, tmp);
|
|
}
|
|
flush_put_bits(&pb);
|
|
return (put_bits_count(&pb) + 7) >> 3;
|
|
default:
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Return the number of channels in an ExSS speaker mask (HD)
|
|
*/
|
|
static int dca_exss_mask2count(int mask)
|
|
{
|
|
/* count bits that mean speaker pairs twice */
|
|
return av_popcount(mask)
|
|
+ av_popcount(mask & (
|
|
DCA_EXSS_CENTER_LEFT_RIGHT
|
|
| DCA_EXSS_FRONT_LEFT_RIGHT
|
|
| DCA_EXSS_FRONT_HIGH_LEFT_RIGHT
|
|
| DCA_EXSS_WIDE_LEFT_RIGHT
|
|
| DCA_EXSS_SIDE_LEFT_RIGHT
|
|
| DCA_EXSS_SIDE_HIGH_LEFT_RIGHT
|
|
| DCA_EXSS_SIDE_REAR_LEFT_RIGHT
|
|
| DCA_EXSS_REAR_LEFT_RIGHT
|
|
| DCA_EXSS_REAR_HIGH_LEFT_RIGHT
|
|
));
|
|
}
|
|
|
|
/**
|
|
* Skip mixing coefficients of a single mix out configuration (HD)
|
|
*/
|
|
static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < channels; i++) {
|
|
int mix_map_mask = get_bits(gb, out_ch);
|
|
int num_coeffs = av_popcount(mix_map_mask);
|
|
skip_bits_long(gb, num_coeffs * 6);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Parse extension substream asset header (HD)
|
|
*/
|
|
static int dca_exss_parse_asset_header(DCAContext *s)
|
|
{
|
|
int header_pos = get_bits_count(&s->gb);
|
|
int header_size;
|
|
int channels;
|
|
int embedded_stereo = 0;
|
|
int embedded_6ch = 0;
|
|
int drc_code_present;
|
|
int extensions_mask;
|
|
int i, j;
|
|
|
|
if (get_bits_left(&s->gb) < 16)
|
|
return -1;
|
|
|
|
/* We will parse just enough to get to the extensions bitmask with which
|
|
* we can set the profile value. */
|
|
|
|
header_size = get_bits(&s->gb, 9) + 1;
|
|
skip_bits(&s->gb, 3); // asset index
|
|
|
|
if (s->static_fields) {
|
|
if (get_bits1(&s->gb))
|
|
skip_bits(&s->gb, 4); // asset type descriptor
|
|
if (get_bits1(&s->gb))
|
|
skip_bits_long(&s->gb, 24); // language descriptor
|
|
|
|
if (get_bits1(&s->gb)) {
|
|
/* How can one fit 1024 bytes of text here if the maximum value
|
|
* for the asset header size field above was 512 bytes? */
|
|
int text_length = get_bits(&s->gb, 10) + 1;
|
|
if (get_bits_left(&s->gb) < text_length * 8)
|
|
return -1;
|
|
skip_bits_long(&s->gb, text_length * 8); // info text
|
|
}
|
|
|
|
skip_bits(&s->gb, 5); // bit resolution - 1
|
|
skip_bits(&s->gb, 4); // max sample rate code
|
|
channels = get_bits(&s->gb, 8) + 1;
|
|
|
|
if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers
|
|
int spkr_remap_sets;
|
|
int spkr_mask_size = 16;
|
|
int num_spkrs[7];
|
|
|
|
if (channels > 2)
|
|
embedded_stereo = get_bits1(&s->gb);
|
|
if (channels > 6)
|
|
embedded_6ch = get_bits1(&s->gb);
|
|
|
|
if (get_bits1(&s->gb)) {
|
|
spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;
|
|
skip_bits(&s->gb, spkr_mask_size); // spkr activity mask
|
|
}
|
|
|
|
spkr_remap_sets = get_bits(&s->gb, 3);
|
|
|
|
for (i = 0; i < spkr_remap_sets; i++) {
|
|
/* std layout mask for each remap set */
|
|
num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));
|
|
}
|
|
|
|
for (i = 0; i < spkr_remap_sets; i++) {
|
|
int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;
|
|
if (get_bits_left(&s->gb) < 0)
|
|
return -1;
|
|
|
|
for (j = 0; j < num_spkrs[i]; j++) {
|
|
int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);
|
|
int num_dec_ch = av_popcount(remap_dec_ch_mask);
|
|
skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes
|
|
}
|
|
}
|
|
|
|
} else {
|
|
skip_bits(&s->gb, 3); // representation type
|
|
}
|
|
}
|
|
|
|
drc_code_present = get_bits1(&s->gb);
|
|
if (drc_code_present)
|
|
get_bits(&s->gb, 8); // drc code
|
|
|
|
if (get_bits1(&s->gb))
|
|
skip_bits(&s->gb, 5); // dialog normalization code
|
|
|
|
if (drc_code_present && embedded_stereo)
|
|
get_bits(&s->gb, 8); // drc stereo code
|
|
|
|
if (s->mix_metadata && get_bits1(&s->gb)) {
|
|
skip_bits(&s->gb, 1); // external mix
|
|
skip_bits(&s->gb, 6); // post mix gain code
|
|
|
|
if (get_bits(&s->gb, 2) != 3) // mixer drc code
|
|
skip_bits(&s->gb, 3); // drc limit
|
|
else
|
|
skip_bits(&s->gb, 8); // custom drc code
|
|
|
|
if (get_bits1(&s->gb)) // channel specific scaling
|
|
for (i = 0; i < s->num_mix_configs; i++)
|
|
skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes
|
|
else
|
|
skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes
|
|
|
|
for (i = 0; i < s->num_mix_configs; i++) {
|
|
if (get_bits_left(&s->gb) < 0)
|
|
return -1;
|
|
dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);
|
|
if (embedded_6ch)
|
|
dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);
|
|
if (embedded_stereo)
|
|
dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);
|
|
}
|
|
}
|
|
|
|
switch (get_bits(&s->gb, 2)) {
|
|
case 0: extensions_mask = get_bits(&s->gb, 12); break;
|
|
case 1: extensions_mask = DCA_EXT_EXSS_XLL; break;
|
|
case 2: extensions_mask = DCA_EXT_EXSS_LBR; break;
|
|
case 3: extensions_mask = 0; /* aux coding */ break;
|
|
}
|
|
|
|
/* not parsed further, we were only interested in the extensions mask */
|
|
|
|
if (get_bits_left(&s->gb) < 0)
|
|
return -1;
|
|
|
|
if (get_bits_count(&s->gb) - header_pos > header_size * 8) {
|
|
av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");
|
|
return -1;
|
|
}
|
|
skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));
|
|
|
|
if (extensions_mask & DCA_EXT_EXSS_XLL)
|
|
s->profile = FF_PROFILE_DTS_HD_MA;
|
|
else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |
|
|
DCA_EXT_EXSS_XXCH))
|
|
s->profile = FF_PROFILE_DTS_HD_HRA;
|
|
|
|
if (!(extensions_mask & DCA_EXT_CORE))
|
|
av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");
|
|
if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)
|
|
av_log(s->avctx, AV_LOG_WARNING, "DTS extensions detection mismatch (%d, %d)\n",
|
|
extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Parse extension substream header (HD)
|
|
*/
|
|
static void dca_exss_parse_header(DCAContext *s)
|
|
{
|
|
int ss_index;
|
|
int blownup;
|
|
int header_size;
|
|
int hd_size;
|
|
int num_audiop = 1;
|
|
int num_assets = 1;
|
|
int active_ss_mask[8];
|
|
int i, j;
|
|
|
|
if (get_bits_left(&s->gb) < 52)
|
|
return;
|
|
|
|
skip_bits(&s->gb, 8); // user data
|
|
ss_index = get_bits(&s->gb, 2);
|
|
|
|
blownup = get_bits1(&s->gb);
|
|
header_size = get_bits(&s->gb, 8 + 4 * blownup) + 1;
|
|
hd_size = get_bits_long(&s->gb, 16 + 4 * blownup) + 1;
|
|
|
|
s->static_fields = get_bits1(&s->gb);
|
|
if (s->static_fields) {
|
|
skip_bits(&s->gb, 2); // reference clock code
|
|
skip_bits(&s->gb, 3); // frame duration code
|
|
|
|
if (get_bits1(&s->gb))
|
|
skip_bits_long(&s->gb, 36); // timestamp
|
|
|
|
/* a single stream can contain multiple audio assets that can be
|
|
* combined to form multiple audio presentations */
|
|
|
|
num_audiop = get_bits(&s->gb, 3) + 1;
|
|
if (num_audiop > 1) {
|
|
av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations.");
|
|
/* ignore such streams for now */
|
|
return;
|
|
}
|
|
|
|
num_assets = get_bits(&s->gb, 3) + 1;
|
|
if (num_assets > 1) {
|
|
av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets.");
|
|
/* ignore such streams for now */
|
|
return;
|
|
}
|
|
|
|
for (i = 0; i < num_audiop; i++)
|
|
active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);
|
|
|
|
for (i = 0; i < num_audiop; i++)
|
|
for (j = 0; j <= ss_index; j++)
|
|
if (active_ss_mask[i] & (1 << j))
|
|
skip_bits(&s->gb, 8); // active asset mask
|
|
|
|
s->mix_metadata = get_bits1(&s->gb);
|
|
if (s->mix_metadata) {
|
|
int mix_out_mask_size;
|
|
|
|
skip_bits(&s->gb, 2); // adjustment level
|
|
mix_out_mask_size = (get_bits(&s->gb, 2) + 1) << 2;
|
|
s->num_mix_configs = get_bits(&s->gb, 2) + 1;
|
|
|
|
for (i = 0; i < s->num_mix_configs; i++) {
|
|
int mix_out_mask = get_bits(&s->gb, mix_out_mask_size);
|
|
s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);
|
|
}
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < num_assets; i++)
|
|
skip_bits_long(&s->gb, 16 + 4 * blownup); // asset size
|
|
|
|
for (i = 0; i < num_assets; i++) {
|
|
if (dca_exss_parse_asset_header(s))
|
|
return;
|
|
}
|
|
|
|
/* not parsed further, we were only interested in the extensions mask
|
|
* from the asset header */
|
|
}
|
|
|
|
/**
|
|
* Main frame decoding function
|
|
* FIXME add arguments
|
|
*/
|
|
static int dca_decode_frame(AVCodecContext * avctx,
|
|
void *data, int *data_size,
|
|
AVPacket *avpkt)
|
|
{
|
|
const uint8_t *buf = avpkt->data;
|
|
int buf_size = avpkt->size;
|
|
|
|
int lfe_samples;
|
|
int num_core_channels = 0;
|
|
int i;
|
|
float *samples_flt = data;
|
|
int16_t *samples_s16 = data;
|
|
int out_size;
|
|
DCAContext *s = avctx->priv_data;
|
|
int channels;
|
|
int core_ss_end;
|
|
|
|
|
|
s->xch_present = 0;
|
|
|
|
s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer,
|
|
DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
|
|
if (s->dca_buffer_size == -1) {
|
|
av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
|
|
return -1;
|
|
}
|
|
|
|
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
|
|
if (dca_parse_frame_header(s) < 0) {
|
|
//seems like the frame is corrupt, try with the next one
|
|
*data_size=0;
|
|
return buf_size;
|
|
}
|
|
//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++) {
|
|
dca_decode_block(s, 0, i);
|
|
}
|
|
|
|
/* 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 0x5a5a5a5a: {
|
|
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 should == 1, number of channels in extension */
|
|
/* 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++) {
|
|
dca_decode_block(s, s->xch_base_channel, i);
|
|
}
|
|
|
|
s->xch_present = 1;
|
|
break;
|
|
}
|
|
case 0x47004a03:
|
|
/* 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_HD_MARKER)
|
|
dca_exss_parse_header(s);
|
|
|
|
avctx->profile = s->profile;
|
|
|
|
channels = s->prim_channels + !!s->lfe;
|
|
|
|
if (s->amode<16) {
|
|
avctx->channel_layout = dca_core_channel_layout[s->amode];
|
|
|
|
if (s->xch_present && (!avctx->request_channels ||
|
|
avctx->request_channels > num_core_channels + !!s->lfe)) {
|
|
avctx->channel_layout |= AV_CH_BACK_CENTER;
|
|
if (s->lfe) {
|
|
avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
|
|
s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];
|
|
} else {
|
|
s->channel_order_tab = 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 = dca_channel_reorder_lfe[s->amode];
|
|
} else
|
|
s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];
|
|
}
|
|
|
|
if (channels > !!s->lfe &&
|
|
s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
|
|
return -1;
|
|
|
|
if (avctx->request_channels == 2 && s->prim_channels > 2) {
|
|
channels = 2;
|
|
s->output = DCA_STEREO;
|
|
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
|
|
}
|
|
} else {
|
|
av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);
|
|
return -1;
|
|
}
|
|
|
|
|
|
/* There is nothing that prevents a dts frame to change channel configuration
|
|
but Libav doesn't support that so only set the channels if it is previously
|
|
unset. Ideally during the first probe for channels the crc should be checked
|
|
and only set avctx->channels when the crc is ok. Right now the decoder could
|
|
set the channels based on a broken first frame.*/
|
|
if (s->is_channels_set == 0) {
|
|
s->is_channels_set = 1;
|
|
avctx->channels = channels;
|
|
}
|
|
if (avctx->channels != channels) {
|
|
av_log(avctx, AV_LOG_ERROR, "DCA decoder does not support number of "
|
|
"channels changing in stream. Skipping frame.\n");
|
|
return -1;
|
|
}
|
|
|
|
out_size = 256 / 8 * s->sample_blocks * channels *
|
|
(av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
|
|
if (*data_size < out_size)
|
|
return -1;
|
|
*data_size = out_size;
|
|
|
|
/* filter to get final output */
|
|
for (i = 0; i < (s->sample_blocks / 8); i++) {
|
|
dca_filter_channels(s, i);
|
|
|
|
/* 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 + s->channel_order_tab[s->xch_base_channel] * 256;
|
|
float* lt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;
|
|
float* rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
|
|
int j;
|
|
for(j = 0; j < 256; ++j) {
|
|
lt_chan[j] -= back_chan[j] * M_SQRT1_2;
|
|
rt_chan[j] -= back_chan[j] * M_SQRT1_2;
|
|
}
|
|
}
|
|
|
|
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
|
|
s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,
|
|
channels);
|
|
samples_flt += 256 * channels;
|
|
} else {
|
|
s->fmt_conv.float_to_int16_interleave(samples_s16,
|
|
s->samples_chanptr, 256,
|
|
channels);
|
|
samples_s16 += 256 * channels;
|
|
}
|
|
}
|
|
|
|
/* 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];
|
|
}
|
|
|
|
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;
|
|
int i;
|
|
|
|
s->avctx = avctx;
|
|
dca_init_vlcs();
|
|
|
|
dsputil_init(&s->dsp, avctx);
|
|
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);
|
|
|
|
for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
|
|
s->samples_chanptr[i] = s->samples + i * 256;
|
|
|
|
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
|
|
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
|
|
s->scale_bias = 1.0 / 32768.0;
|
|
} else {
|
|
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
|
|
s->scale_bias = 1.0;
|
|
}
|
|
|
|
/* allow downmixing to stereo */
|
|
if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
|
|
avctx->request_channels == 2) {
|
|
avctx->channels = avctx->request_channels;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int dca_decode_end(AVCodecContext * avctx)
|
|
{
|
|
DCAContext *s = avctx->priv_data;
|
|
ff_mdct_end(&s->imdct);
|
|
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 },
|
|
};
|
|
|
|
AVCodec ff_dca_decoder = {
|
|
.name = "dca",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = CODEC_ID_DTS,
|
|
.priv_data_size = sizeof(DCAContext),
|
|
.init = dca_decode_init,
|
|
.decode = dca_decode_frame,
|
|
.close = dca_decode_end,
|
|
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
|
|
.capabilities = CODEC_CAP_CHANNEL_CONF,
|
|
.sample_fmts = (const enum AVSampleFormat[]) {
|
|
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
|
|
},
|
|
.profiles = NULL_IF_CONFIG_SMALL(profiles),
|
|
};
|