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1924 lines
70 KiB
1924 lines
70 KiB
/* |
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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 FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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|
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#include <math.h> |
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#include <stddef.h> |
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#include <stdio.h> |
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|
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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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|
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//#define TRACE |
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|
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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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|
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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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|
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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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|
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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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|
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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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|
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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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|
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int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands |
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|
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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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|
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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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|
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int output; ///< type of output |
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float scale_bias; ///< output scale |
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|
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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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|
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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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|
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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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|
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int core_ext_mask; ///< present extensions in the core substream |
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|
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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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|
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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 |
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int num_mix_configs; ///< number of mix out configurations |
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int mix_config_num_ch[4]; ///< number of channels in each mix out configuration |
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|
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int profile; |
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|
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int debug_flag; ///< used for suppressing repeated error messages output |
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DSPContext dsp; |
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FFTContext imdct; |
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SynthFilterContext synth; |
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DCADSPContext dcadsp; |
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FmtConvertContext fmt_conv; |
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} DCAContext; |
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|
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static const uint16_t dca_vlc_offs[] = { |
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0, 512, 640, 768, 1282, 1794, 2436, 3080, 3770, 4454, 5364, |
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5372, 5380, 5388, 5392, 5396, 5412, 5420, 5428, 5460, 5492, 5508, |
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5572, 5604, 5668, 5796, 5860, 5892, 6412, 6668, 6796, 7308, 7564, |
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7820, 8076, 8620, 9132, 9388, 9910, 10166, 10680, 11196, 11726, 12240, |
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12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264, |
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18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622, |
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}; |
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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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|
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if (vlcs_initialized) |
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return; |
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|
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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[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 data_size_tmp; |
|
|
|
int lfe_samples; |
|
int num_core_channels = 0; |
|
int i; |
|
float *samples_flt = data; |
|
int16_t *samples = data; |
|
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 FFmpeg 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; |
|
} |
|
|
|
data_size_tmp = (s->sample_blocks / 8) * 256 * channels; |
|
data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(*samples_flt) : sizeof(*samples); |
|
if (*data_size < data_size_tmp) |
|
return -1; |
|
*data_size = data_size_tmp; |
|
|
|
/* 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; |
|
} |
|
} |
|
|
|
/* interleave samples */ |
|
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) { |
|
float_interleave(samples_flt, s->samples_chanptr, 256, channels); |
|
samples_flt += 256 * channels; |
|
} else { |
|
s->fmt_conv.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels); |
|
samples += 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; |
|
avctx->sample_fmt = avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT ? |
|
AV_SAMPLE_FMT_FLT : 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, |
|
.profiles = NULL_IF_CONFIG_SMALL(profiles), |
|
};
|
|
|