* Copyright (C) 2006 Benjamin Larsson
* Copyright (C) 2007 Konstantin Shishkov
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/**
- * @file dca.c
- */
-
#include <math.h>
#include <stddef.h>
#include <stdio.h>
+#include "libavutil/common.h"
+#include "libavutil/intmath.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/audioconvert.h"
#include "avcodec.h"
#include "dsputil.h"
-#include "bitstream.h"
+#include "fft.h"
+#include "get_bits.h"
+#include "put_bits.h"
#include "dcadata.h"
#include "dcahuff.h"
#include "dca.h"
+#include "dca_parser.h"
+#include "synth_filter.h"
+#include "dcadsp.h"
+#include "fmtconvert.h"
+
+#if ARCH_ARM
+# include "arm/dca.h"
+#endif
//#define TRACE
-#define DCA_PRIM_CHANNELS_MAX (5)
-#define DCA_SUBBANDS (32)
-#define DCA_ABITS_MAX (32) /* Should be 28 */
-#define DCA_SUBSUBFAMES_MAX (4)
-#define DCA_LFE_MAX (3)
+#define DCA_PRIM_CHANNELS_MAX (7)
+#define DCA_SUBBANDS (32)
+#define DCA_ABITS_MAX (32) /* Should be 28 */
+#define DCA_SUBSUBFRAMES_MAX (4)
+#define DCA_SUBFRAMES_MAX (16)
+#define DCA_BLOCKS_MAX (16)
+#define DCA_LFE_MAX (3)
enum DCAMode {
DCA_MONO = 0,
DCA_4F2R
};
+/* these are unconfirmed but should be mostly correct */
+enum DCAExSSSpeakerMask {
+ DCA_EXSS_FRONT_CENTER = 0x0001,
+ DCA_EXSS_FRONT_LEFT_RIGHT = 0x0002,
+ DCA_EXSS_SIDE_REAR_LEFT_RIGHT = 0x0004,
+ DCA_EXSS_LFE = 0x0008,
+ DCA_EXSS_REAR_CENTER = 0x0010,
+ DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,
+ DCA_EXSS_REAR_LEFT_RIGHT = 0x0040,
+ DCA_EXSS_FRONT_HIGH_CENTER = 0x0080,
+ DCA_EXSS_OVERHEAD = 0x0100,
+ DCA_EXSS_CENTER_LEFT_RIGHT = 0x0200,
+ DCA_EXSS_WIDE_LEFT_RIGHT = 0x0400,
+ DCA_EXSS_SIDE_LEFT_RIGHT = 0x0800,
+ DCA_EXSS_LFE2 = 0x1000,
+ DCA_EXSS_SIDE_HIGH_LEFT_RIGHT = 0x2000,
+ DCA_EXSS_REAR_HIGH_CENTER = 0x4000,
+ DCA_EXSS_REAR_HIGH_LEFT_RIGHT = 0x8000,
+};
+
+enum DCAExtensionMask {
+ DCA_EXT_CORE = 0x001, ///< core in core substream
+ DCA_EXT_XXCH = 0x002, ///< XXCh channels extension in core substream
+ DCA_EXT_X96 = 0x004, ///< 96/24 extension in core substream
+ DCA_EXT_XCH = 0x008, ///< XCh channel extension in core substream
+ DCA_EXT_EXSS_CORE = 0x010, ///< core in ExSS (extension substream)
+ DCA_EXT_EXSS_XBR = 0x020, ///< extended bitrate extension in ExSS
+ DCA_EXT_EXSS_XXCH = 0x040, ///< XXCh channels extension in ExSS
+ DCA_EXT_EXSS_X96 = 0x080, ///< 96/24 extension in ExSS
+ DCA_EXT_EXSS_LBR = 0x100, ///< low bitrate component in ExSS
+ DCA_EXT_EXSS_XLL = 0x200, ///< lossless extension in ExSS
+};
+
+/* -1 are reserved or unknown */
+static const int dca_ext_audio_descr_mask[] = {
+ DCA_EXT_XCH,
+ -1,
+ DCA_EXT_X96,
+ DCA_EXT_XCH | DCA_EXT_X96,
+ -1,
+ -1,
+ DCA_EXT_XXCH,
+ -1,
+};
+
+/* extensions that reside in core substream */
+#define DCA_CORE_EXTS (DCA_EXT_XCH | DCA_EXT_XXCH | DCA_EXT_X96)
+
/* Tables for mapping dts channel configurations to libavcodec multichannel api.
* Some compromises have been made for special configurations. Most configurations
* are never used so complete accuracy is not needed.
*
* L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead.
- * S -> back, when both rear and back are configured move one of them to the side channel
+ * S -> side, when both rear and back are configured move one of them to the side channel
* OV -> center back
- * All 2 channel configurations -> CH_LAYOUT_STEREO
+ * All 2 channel configurations -> AV_CH_LAYOUT_STEREO
*/
+static const uint64_t dca_core_channel_layout[] = {
+ AV_CH_FRONT_CENTER, ///< 1, A
+ AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
+ AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
+ AV_CH_LAYOUT_STEREO, ///< 2, (L + R) + (L - R) (sum-difference)
+ AV_CH_LAYOUT_STEREO, ///< 2, LT + RT (left and right total)
+ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER, ///< 3, C + L + R
+ AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER, ///< 3, L + R + S
+ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 4, C + L + R + S
+ AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 4, L + R + SL + SR
+
+ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |
+ AV_CH_SIDE_RIGHT, ///< 5, C + L + R + SL + SR
+
+ 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
+
+ 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
+
+ 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
+
+ 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
+
+ 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
+
+ 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
+};
-static const int64_t dca_core_channel_layout[] = {
- CH_FRONT_CENTER, ///< 1, A
- CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
- CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
- CH_LAYOUT_STEREO, ///< 2, (L+R) + (L-R) (sum-difference)
- CH_LAYOUT_STEREO, ///< 2, LT +RT (left and right total)
- CH_LAYOUT_STEREO|CH_FRONT_CENTER, ///< 3, C+L+R
- CH_LAYOUT_STEREO|CH_BACK_CENTER, ///< 3, L+R+S
- CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_BACK_CENTER, ///< 4, C + L + R+ S
- CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT, ///< 4, L + R+ SL+SR
- CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_BACK_LEFT|CH_BACK_RIGHT, ///< 5, C + L + R+ SL+SR
- CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT|CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR
- CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT|CH_FRONT_CENTER|CH_BACK_CENTER, ///< 6, C + L + R+ LR + RR + OV
- CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_FRONT_LEFT_OF_CENTER|CH_BACK_CENTER|CH_BACK_LEFT|CH_BACK_RIGHT, ///< 6, CF+ CR+LF+ RF+LR + RR
- CH_FRONT_LEFT_OF_CENTER|CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR
- CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT|CH_BACK_LEFT|CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2+ SR1 + SR2
- CH_FRONT_LEFT_OF_CENTER|CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_CENTER|CH_BACK_RIGHT, ///< 8, CL + C+ CR + L + R + SL + S+ SR
-
- /* The following entries adds the LFE layouts, this way we can reuse the table for the AVCodec channel_layouts member*/
- CH_FRONT_CENTER|CH_LOW_FREQUENCY,
- CH_LAYOUT_STEREO|CH_LOW_FREQUENCY,
- CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_LOW_FREQUENCY,
- CH_LAYOUT_STEREO|CH_BACK_CENTER|CH_LOW_FREQUENCY,
- CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_BACK_CENTER|CH_LOW_FREQUENCY,
- CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT|CH_LOW_FREQUENCY,
- CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_BACK_LEFT|CH_BACK_RIGHT|CH_LOW_FREQUENCY,
- CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT|CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LOW_FREQUENCY,
- CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT|CH_FRONT_CENTER|CH_BACK_CENTER|CH_LOW_FREQUENCY,
- CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_FRONT_LEFT_OF_CENTER|CH_BACK_CENTER|CH_BACK_LEFT|CH_BACK_RIGHT|CH_LOW_FREQUENCY,
- CH_FRONT_LEFT_OF_CENTER|CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT|CH_LOW_FREQUENCY,
- CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT|CH_BACK_LEFT|CH_BACK_RIGHT|CH_LOW_FREQUENCY,
- CH_FRONT_LEFT_OF_CENTER|CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_CENTER|CH_BACK_RIGHT|CH_LOW_FREQUENCY,
- 0,
+static const int8_t dca_lfe_index[] = {
+ 1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3
};
+static const int8_t dca_channel_reorder_lfe[][9] = {
+ { 0, -1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 4, -1, -1, -1, -1, -1},
+ { 0, 1, 3, 4, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 4, 5, -1, -1, -1, -1},
+ { 3, 4, 0, 1, 5, 6, -1, -1, -1},
+ { 2, 0, 1, 4, 5, 6, -1, -1, -1},
+ { 0, 6, 4, 5, 2, 3, -1, -1, -1},
+ { 4, 2, 5, 0, 1, 6, 7, -1, -1},
+ { 5, 6, 0, 1, 7, 3, 8, 4, -1},
+ { 4, 2, 5, 0, 1, 6, 8, 7, -1},
+};
-#define DCA_DOLBY 101 /* FIXME */
+static const int8_t dca_channel_reorder_lfe_xch[][9] = {
+ { 0, 2, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 3, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 4, -1, -1, -1, -1, -1},
+ { 0, 1, 3, 4, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 4, 5, -1, -1, -1, -1},
+ { 0, 1, 4, 5, 3, -1, -1, -1, -1},
+ { 2, 0, 1, 5, 6, 4, -1, -1, -1},
+ { 3, 4, 0, 1, 6, 7, 5, -1, -1},
+ { 2, 0, 1, 4, 5, 6, 7, -1, -1},
+ { 0, 6, 4, 5, 2, 3, 7, -1, -1},
+ { 4, 2, 5, 0, 1, 7, 8, 6, -1},
+ { 5, 6, 0, 1, 8, 3, 9, 4, 7},
+ { 4, 2, 5, 0, 1, 6, 9, 8, 7},
+};
-#define DCA_CHANNEL_BITS 6
-#define DCA_CHANNEL_MASK 0x3F
+static const int8_t dca_channel_reorder_nolfe[][9] = {
+ { 0, -1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 3, -1, -1, -1, -1, -1},
+ { 0, 1, 2, 3, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 3, 4, -1, -1, -1, -1},
+ { 2, 3, 0, 1, 4, 5, -1, -1, -1},
+ { 2, 0, 1, 3, 4, 5, -1, -1, -1},
+ { 0, 5, 3, 4, 1, 2, -1, -1, -1},
+ { 3, 2, 4, 0, 1, 5, 6, -1, -1},
+ { 4, 5, 0, 1, 6, 2, 7, 3, -1},
+ { 3, 2, 4, 0, 1, 5, 7, 6, -1},
+};
-#define DCA_LFE 0x80
+static const int8_t dca_channel_reorder_nolfe_xch[][9] = {
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 3, -1, -1, -1, -1, -1},
+ { 0, 1, 2, 3, -1, -1, -1, -1, -1},
+ { 2, 0, 1, 3, 4, -1, -1, -1, -1},
+ { 0, 1, 3, 4, 2, -1, -1, -1, -1},
+ { 2, 0, 1, 4, 5, 3, -1, -1, -1},
+ { 2, 3, 0, 1, 5, 6, 4, -1, -1},
+ { 2, 0, 1, 3, 4, 5, 6, -1, -1},
+ { 0, 5, 3, 4, 1, 2, 6, -1, -1},
+ { 3, 2, 4, 0, 1, 6, 7, 5, -1},
+ { 4, 5, 0, 1, 7, 2, 8, 3, 6},
+ { 3, 2, 4, 0, 1, 5, 8, 7, 6},
+};
-#define HEADER_SIZE 14
+#define DCA_DOLBY 101 /* FIXME */
-#define DCA_MAX_FRAME_SIZE 16384
+#define DCA_CHANNEL_BITS 6
+#define DCA_CHANNEL_MASK 0x3F
+
+#define DCA_LFE 0x80
+
+#define HEADER_SIZE 14
+
+#define DCA_MAX_FRAME_SIZE 16384
+#define DCA_MAX_EXSS_HEADER_SIZE 4096
+
+#define DCA_BUFFER_PADDING_SIZE 1024
/** Bit allocation */
typedef struct {
static BitAlloc dca_scalefactor; ///< scalefactor VLCs
static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
-static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
+static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,
+ int idx)
{
- return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
+ return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) +
+ ba->offset;
}
typedef struct {
AVCodecContext *avctx;
+ AVFrame frame;
/* Frame header */
int frame_type; ///< type of the current frame
int samples_deficit; ///< deficit sample count
/* Primary audio coding header */
int subframes; ///< number of subframes
+ int is_channels_set; ///< check for if the channel number is already set
int total_channels; ///< number of channels including extensions
int prim_channels; ///< number of primary audio channels
int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count
float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
/* Primary audio coding side information */
- int subsubframes; ///< number of subsubframes
- int partial_samples; ///< partial subsubframe samples count
+ int subsubframes[DCA_SUBFRAMES_MAX]; ///< number of subsubframes
+ int partial_samples[DCA_SUBFRAMES_MAX]; ///< partial subsubframe samples count
int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not)
int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs
int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index
int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
- float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX *
- 2 /*history */ ]; ///< Low frequency effect data
+ float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)]; ///< Low frequency effect data
int lfe_scale_factor;
/* Subband samples history (for ADPCM) */
- float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
- DECLARE_ALIGNED_16(float, subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512]);
- float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][32];
+ DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
+ DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
+ DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
int hist_index[DCA_PRIM_CHANNELS_MAX];
+ DECLARE_ALIGNED(32, float, raXin)[32];
int output; ///< type of output
- float add_bias; ///< output bias
float scale_bias; ///< output scale
- DECLARE_ALIGNED_16(float, samples[1536]); /* 6 * 256 = 1536, might only need 5 */
- const float *samples_chanptr[6];
+ DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
+ DECLARE_ALIGNED(32, float, samples)[(DCA_PRIM_CHANNELS_MAX + 1) * 256];
+ const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
- uint8_t dca_buffer[DCA_MAX_FRAME_SIZE];
+ uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE + DCA_BUFFER_PADDING_SIZE];
int dca_buffer_size; ///< how much data is in the dca_buffer
+ const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
GetBitContext gb;
/* Current position in DCA frame */
int current_subframe;
int current_subsubframe;
+ int core_ext_mask; ///< present extensions in the core substream
+
+ /* XCh extension information */
+ int xch_present; ///< XCh extension present and valid
+ int xch_base_channel; ///< index of first (only) channel containing XCH data
+
+ /* ExSS header parser */
+ int static_fields; ///< static fields present
+ 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;
- MDCTContext imdct;
+ 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;
+ 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++)
+ 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, 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++)
+ 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, 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++)
+ 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, 1);
-
- 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);
- init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],
+ 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, 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--)
+ while (len--)
*dst++ = get_bits(gb, bits);
}
-static int dca_parse_frame_header(DCAContext * s)
+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 };
+ 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);
+ skip_bits_long(&s->gb, 32);
/* Frame header */
s->frame_type = 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;
+ return AVERROR_INVALIDDATA;
s->amode = get_bits(&s->gb, 6);
s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)];
if (!s->sample_rate)
- return -1;
+ return AVERROR_INVALIDDATA;
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;
+ return AVERROR_INVALIDDATA;
s->downmix = get_bits(&s->gb, 1);
s->dynrange = get_bits(&s->gb, 1);
/* FIXME: channels mixing levels */
s->output = s->amode;
- if(s->lfe) s->output |= DCA_LFE;
+ if (s->lfe)
+ s->output |= DCA_LFE;
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
/* Primary audio coding header */
s->subframes = get_bits(&s->gb, 4) + 1;
- s->total_channels = get_bits(&s->gb, 3) + 1;
- s->prim_channels = s->total_channels;
- if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
- s->prim_channels = DCA_PRIM_CHANNELS_MAX; /* We only support DTS core */
-
-
- for (i = 0; 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 = 0; 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, s->prim_channels, 3);
- get_array(&s->gb, s->transient_huffman, s->prim_channels, 2);
- get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);
- get_array(&s->gb, s->bitalloc_huffman, s->prim_channels, 3);
-
- /* Get codebooks quantization indexes */
- memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
- for (j = 1; j < 11; j++)
- for (i = 0; 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 = 0; i < s->prim_channels; i++)
- s->scalefactor_adj[i][j] = 1;
-
- for (j = 1; j < 11; j++)
- for (i = 0; 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 = 0; 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;
+ return dca_parse_audio_coding_header(s, 0);
}
-static inline int get_scale(GetBitContext *gb, int level, int value)
+static inline int get_scale(GetBitContext *gb, int level, int value, int log2range)
{
- if (level < 5) {
- /* huffman encoded */
- value += get_bitalloc(gb, &dca_scalefactor, level);
- } else if(level < 8)
- value = get_bits(gb, level + 1);
- return value;
+ if (level < 5) {
+ /* huffman encoded */
+ value += get_bitalloc(gb, &dca_scalefactor, level);
+ value = av_clip(value, 0, (1 << log2range) - 1);
+ } else if (level < 8) {
+ if (level + 1 > log2range) {
+ skip_bits(gb, level + 1 - log2range);
+ value = get_bits(gb, log2range);
+ } else {
+ value = get_bits(gb, level + 1);
+ }
+ }
+ return value;
}
-static int dca_subframe_header(DCAContext * s)
+static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
{
/* Primary audio coding side information */
int j, k;
- s->subsubframes = get_bits(&s->gb, 2) + 1;
- s->partial_samples = get_bits(&s->gb, 3);
- for (j = 0; j < s->prim_channels; j++) {
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ 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 = 0; j < s->prim_channels; j++) {
+ 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) */
}
/* Bit allocation index */
- for (j = 0; j < s->prim_channels; j++) {
+ 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] == 7) {
av_log(s->avctx, AV_LOG_ERROR,
"Invalid bit allocation index\n");
- return -1;
+ return AVERROR_INVALIDDATA;
} 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;
+ // av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index [%i][%i] too big (%i)\n",
+ // j, k, s->bitalloc[j][k]);
+ return AVERROR_INVALIDDATA;
}
}
}
/* Transition mode */
- for (j = 0; j < s->prim_channels; j++) {
+ 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 > 1 &&
+ 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]);
}
}
- for (j = 0; j < s->prim_channels; j++) {
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ for (j = base_channel; j < s->prim_channels; j++) {
const uint32_t *scale_table;
- int scale_sum;
+ int scale_sum, log_size;
- memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
+ memset(s->scale_factor[j], 0,
+ s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
- if (s->scalefactor_huffman[j] == 6)
+ if (s->scalefactor_huffman[j] == 6) {
scale_table = scale_factor_quant7;
- else
+ log_size = 7;
+ } else {
scale_table = scale_factor_quant6;
+ log_size = 6;
+ }
/* When huffman coded, only the difference is encoded */
scale_sum = 0;
for (k = 0; k < s->subband_activity[j]; k++) {
if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
- scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
+ scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
s->scale_factor[j][k][0] = scale_table[scale_sum];
}
if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
/* Get second scale factor */
- scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
+ scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
s->scale_factor[j][k][1] = scale_table[scale_sum];
}
}
}
/* Joint subband scale factor codebook select */
- for (j = 0; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->prim_channels; j++) {
/* Transmitted only if joint subband coding enabled */
if (s->joint_intensity[j] > 0)
s->joint_huff[j] = get_bits(&s->gb, 3);
}
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
/* Scale factors for joint subband coding */
- for (j = 0; j < s->prim_channels; j++) {
+ for (j = base_channel; j < s->prim_channels; j++) {
int source_channel;
/* Transmitted only if joint subband coding enabled */
* (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 */
+ scale = get_scale(&s->gb, s->joint_huff[j], 64 /* bias */, 7);
s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */
}
- if (!s->debug_flag & 0x02) {
+ 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 (s->prim_channels > 2) {
- if(s->downmix) {
- for (j = 0; j < s->prim_channels; j++) {
+ 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 = 0; j < s->prim_channels; j++) {
+ if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "Invalid channel mode %d\n", am);
+ return AVERROR_INVALIDDATA;
+ }
+ 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 (s->dynrange)
+ if (!base_channel && s->dynrange)
s->dynrange_coef = get_bits(&s->gb, 8);
/* Side information CRC check word */
*/
/* VQ encoded high frequency subbands */
- for (j = 0; j < s->prim_channels; j++)
+ 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 (s->lfe) {
+ if (!base_channel && s->lfe) {
/* LFE samples */
- int lfe_samples = 2 * s->lfe * s->subsubframes;
+ 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_samples * 2; j++) {
+ 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)];
+ skip_bits(&s->gb, 1);
+ s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 7)];
/* Quantization step size * scale factor */
lfe_scale = 0.035 * s->lfe_scale_factor;
- for (j = lfe_samples; j < lfe_samples * 2; j++)
+ 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);
+ 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);
- for (j = 0; j < s->prim_channels; j++) {
+ 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 = 0; j < s->prim_channels; j++) {
+ 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);
+ 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 = 0; j < s->prim_channels; j++) {
+ 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 = 0; j < s->prim_channels; j++) {
+ 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 = 0; j < s->prim_channels; j++) {
+ 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, "\n");
}
- for (j = 0; j < s->prim_channels; j++) {
+ 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");
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
}
- if (s->prim_channels > 2 && s->downmix) {
+ 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, "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 = 0; j < s->prim_channels; j++)
+ 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(s->lfe){
- int lfe_samples = 2 * s->lfe * s->subsubframes;
+ 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_samples * 2; j++)
+ 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");
}
return 0;
}
-static void qmf_32_subbands(DCAContext * s, int chans,
+static void qmf_32_subbands(DCAContext *s, int chans,
float samples_in[32][8], float *samples_out,
- float scale, float bias)
+ float scale)
{
const float *prCoeff;
- int i, j;
- DECLARE_ALIGNED_16(float, raXin[32]);
-
- int hist_index= s->hist_index[chans];
- float *subband_fir_hist2 = s->subband_fir_noidea[chans];
+ int i;
+ int sb_act = s->subband_activity[chans];
int subindex;
- scale *= sqrt(1/8.0);
+ scale *= sqrt(1 / 8.0);
/* Select filter */
if (!s->multirate_inter) /* Non-perfect reconstruction */
else /* Perfect reconstruction */
prCoeff = fir_32bands_perfect;
+ for (i = sb_act; i < 32; i++)
+ s->raXin[i] = 0.0;
+
/* Reconstructed channel sample index */
for (subindex = 0; subindex < 8; subindex++) {
- float *subband_fir_hist = s->subband_fir_hist[chans] + hist_index;
/* Load in one sample from each subband and clear inactive subbands */
- for (i = 0; i < s->subband_activity[chans]; i++){
- if((i-1)&2) raXin[i] = -samples_in[i][subindex];
- else raXin[i] = samples_in[i][subindex];
- }
- for (; i < 32; i++)
- raXin[i] = 0.0;
-
- ff_imdct_half(&s->imdct, subband_fir_hist, raXin);
-
- /* Multiply by filter coefficients */
- for (i = 0; i < 16; i++){
- float a= subband_fir_hist2[i ];
- float b= subband_fir_hist2[i+16];
- float c= 0;
- float d= 0;
- for (j = 0; j < 512-hist_index; j += 64){
- a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j]);
- b += prCoeff[i+j+16]*( subband_fir_hist[ i+j]);
- c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j]);
- d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j]);
- }
- for ( ; j < 512; j += 64){
- a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j-512]);
- b += prCoeff[i+j+16]*( subband_fir_hist[ i+j-512]);
- c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j-512]);
- d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j-512]);
- }
- samples_out[i ] = a * scale + bias;
- samples_out[i+16] = b * scale + bias;
- subband_fir_hist2[i ] = c;
- subband_fir_hist2[i+16] = d;
+ for (i = 0; i < sb_act; i++) {
+ unsigned sign = (i - 1) & 2;
+ uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
+ AV_WN32A(&s->raXin[i], v);
}
- samples_out+= 32;
- hist_index = (hist_index-32)&511;
+ 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;
}
- s->hist_index[chans]= hist_index;
}
-static void lfe_interpolation_fir(int decimation_select,
+static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
int num_deci_sample, float *samples_in,
- float *samples_out, float scale,
- float bias)
+ float *samples_out, float scale)
{
/* samples_in: An array holding decimated samples.
* Samples in current subframe starts from samples_in[0],
* samples_out: An array holding interpolated samples
*/
- int decifactor, k, j;
+ int decifactor;
const float *prCoeff;
-
- int interp_index = 0; /* Index to the interpolated samples */
int deciindex;
/* Select decimation filter */
if (decimation_select == 1) {
- decifactor = 128;
+ decifactor = 64;
prCoeff = lfe_fir_128;
} else {
- decifactor = 64;
+ decifactor = 32;
prCoeff = lfe_fir_64;
}
/* Interpolation */
for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
- /* One decimated sample generates decifactor interpolated ones */
- for (k = 0; k < decifactor; k++) {
- float rTmp = 0.0;
- //FIXME the coeffs are symetric, fix that
- for (j = 0; j < 512 / decifactor; j++)
- rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor];
- samples_out[interp_index++] = (rTmp * scale) + bias;
- }
+ 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]; \
- samples[i] = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \
- samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * coef[2][1];
-
-#define DOWNMIX_TO_STEREO(op1, op2) \
- for(i = 0; i < 256; i++){ \
- op1 \
- op2 \
+#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])
+ int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],
+ const int8_t *channel_mapping)
{
+ int c, l, r, sl, sr, s;
int i;
- float t;
+ float t, u, v;
float coef[DCA_PRIM_CHANNELS_MAX][2];
- for(i=0; i<DCA_PRIM_CHANNELS_MAX; i++) {
+ 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]];
}
case DCA_STEREO:
break;
case DCA_3F:
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);
+ 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:
- DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),);
+ 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 + 768, 3, coef));
+ MIX_REAR1(samples, i + s, 3, coef));
break;
case DCA_2F2R:
- DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),);
+ 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 + 768, i + 1024, 3, coef));
+ MIX_REAR2(samples, i + sl, i + sr, 3, coef));
break;
}
}
+#ifndef decode_blockcodes
/* Very compact version of the block code decoder that does not use table
* look-up but is slightly slower */
static int decode_blockcode(int code, int levels, int *values)
int offset = (levels - 1) >> 1;
for (i = 0; i < 4; i++) {
- values[i] = (code % levels) - offset;
- code /= levels;
+ 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;
- }
+ return code;
}
-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 decode_blockcodes(int code1, int code2, int levels, int *values)
+{
+ return decode_blockcode(code1, levels, values) |
+ decode_blockcode(code2, levels, values + 4);
+}
+#endif
+
+static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
+static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
+
+#ifndef int8x8_fmul_int32
+static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
+{
+ float fscale = scale / 16.0;
+ int i;
+ for (i = 0; i < 8; i++)
+ dst[i] = src[i] * fscale;
+}
+#endif
-static int dca_subsubframe(DCAContext * s)
+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_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
+ float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
+ LOCAL_ALIGNED_16(int, block, [8]);
/*
* Audio data
else
quant_step_table = lossy_quant_d;
- for (k = 0; k < s->prim_channels; k++) {
+ for (k = base_channel; k < s->prim_channels; k++) {
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
for (l = 0; l < s->vq_start_subband[k]; l++) {
int m;
int abits = s->bitalloc[k][l];
float quant_step_size = quant_step_table[abits];
- float rscale;
/*
* Determine quantization index code book and its type
/*
* Extract bits from the bit stream
*/
- if(!abits){
+ if (!abits) {
memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
- }else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
- if(abits <= 7){
- /* Block code */
- int block_code1, block_code2, size, levels;
- int block[8];
-
- 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]);
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] = block[m];
- }else{
- /* no coding */
+ } 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, err;
+
+ size = abits_sizes[abits - 1];
+ levels = abits_levels[abits - 1];
+
+ block_code1 = get_bits(&s->gb, size);
+ block_code2 = get_bits(&s->gb, size);
+ err = decode_blockcodes(block_code1, block_code2,
+ levels, block);
+ if (err) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "ERROR: block code look-up failed\n");
+ return AVERROR_INVALIDDATA;
+ }
+ } else {
+ /* no coding */
+ for (m = 0; m < 8; m++)
+ block[m] = get_sbits(&s->gb, abits - 3);
+ }
+ } else {
+ /* Huffman coded */
for (m = 0; m < 8; m++)
- subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3);
+ block[m] = get_bitalloc(&s->gb,
+ &dca_smpl_bitalloc[abits], sel);
}
- }else{
- /* Huffman coded */
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
- }
- /* Deal with transients */
- if (s->transition_mode[k][l] &&
- subsubframe >= s->transition_mode[k][l])
- rscale = quant_step_size * s->scale_factor[k][l][1];
- else
- rscale = quant_step_size * s->scale_factor[k][l][0];
-
- rscale *= s->scalefactor_adj[k][sel];
-
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] *= rscale;
+ s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],
+ block, rscale, 8);
+ }
/*
* Inverse ADPCM if in prediction mode
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);
+ s->subband_samples_hist[k][l][m - n + 4] / 8192);
}
}
}
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;
+ int hfvq = s->high_freq_vq[k][l];
if (!s->debug_flag & 0x01) {
- av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");
+ 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;
- }
+ int8x8_fmul_int32(subband_samples[k][l],
+ &high_freq_vq[hfvq][subsubframe * 8],
+ s->scale_factor[k][l][0]);
}
}
/* Check for DSYNC after subsubframe */
- if (s->aspf || subsubframe == s->subsubframes - 1) {
+ 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");
}
/* Backup predictor history for adpcm */
- for (k = 0; k < s->prim_channels; k++)
+ 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]));
+ 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 * k],
- M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ ,
- s->add_bias );
+/* 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->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) {
- dca_downmix(s->samples, s->amode, s->downmix_coef);
+ 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) {
- int lfe_samples = 2 * s->lfe * s->subsubframes;
- int i_channels = dca_channels[s->output & DCA_CHANNEL_MASK];
-
- lfe_interpolation_fir(s->lfe, 2 * s->lfe,
- s->lfe_data + lfe_samples +
- 2 * s->lfe * subsubframe,
- &s->samples[256 * i_channels],
- (1.0/256.0)*s->scale_bias, s->add_bias);
+ 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 */
}
}
-static int dca_subframe_footer(DCAContext * s)
+static int dca_subframe_footer(DCAContext *s, int base_channel)
{
int aux_data_count = 0, i;
- int lfe_samples;
/*
* Unpack optional information
*/
- if (s->timestamp)
- get_bits(&s->gb, 32);
+ /* presumably optional information only appears in the core? */
+ if (!base_channel) {
+ if (s->timestamp)
+ skip_bits_long(&s->gb, 32);
- if (s->aux_data)
- aux_data_count = get_bits(&s->gb, 6);
+ 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);
+ 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);
-
- lfe_samples = 2 * s->lfe * s->subsubframes;
- for (i = 0; i < lfe_samples; i++) {
- s->lfe_data[i] = s->lfe_data[i + lfe_samples];
+ if (s->crc_present && (s->downmix || s->dynrange))
+ get_bits(&s->gb, 16);
}
return 0;
* @param s pointer to the DCAContext
*/
-static int dca_decode_block(DCAContext * s)
+static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
{
+ int ret;
/* Sanity check */
if (s->current_subframe >= s->subframes) {
av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
s->current_subframe, s->subframes);
- return -1;
+ return AVERROR_INVALIDDATA;
}
if (!s->current_subsubframe) {
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
#endif
/* Read subframe header */
- if (dca_subframe_header(s))
- return -1;
+ if ((ret = dca_subframe_header(s, base_channel, block_index)))
+ return ret;
}
/* Read subsubframe */
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
#endif
- if (dca_subsubframe(s))
- return -1;
+ if ((ret = dca_subsubframe(s, base_channel, block_index)))
+ return ret;
/* Update state */
s->current_subsubframe++;
- if (s->current_subsubframe >= s->subsubframes) {
+ if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {
s->current_subsubframe = 0;
s->current_subframe++;
}
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
#endif
/* Read subframe footer */
- if (dca_subframe_footer(s))
- return -1;
+ if ((ret = dca_subframe_footer(s, base_channel)))
+ return ret;
}
return 0;
}
/**
- * Convert bitstream to one representation based on sync marker
+ * Return the number of channels in an ExSS speaker mask (HD)
*/
-static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,
- int max_size)
+static int dca_exss_mask2count(int mask)
{
- 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");
+ /* 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
+ }
}
- 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++ = bswap_16(*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);
+ 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]);
}
- flush_put_bits(&pb);
- return (put_bits_count(&pb) + 7) >> 3;
- default:
+ }
+
+ 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 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);
+ skip_bits(&s->gb, 8 + 4 * blownup); // header_size
+ skip_bits(&s->gb, 16 + 4 * blownup); // hd_size
+
+ 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,
- const uint8_t * buf, int buf_size)
+static int dca_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
- int i;
- int16_t *samples = data;
+ int lfe_samples;
+ int num_core_channels = 0;
+ int i, ret;
+ float *samples_flt;
+ int16_t *samples_s16;
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);
- if (s->dca_buffer_size == -1) {
+ s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,
+ DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
+ if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
- if (dca_parse_frame_header(s) < 0) {
+ if ((ret = dca_parse_frame_header(s)) < 0) {
//seems like the frame is corrupt, try with the next one
- *data_size=0;
- return buf_size;
+ return ret;
}
//set AVCodec values with parsed data
avctx->sample_rate = s->sample_rate;
- avctx->bit_rate = s->bit_rate;
+ avctx->bit_rate = s->bit_rate;
- channels = s->prim_channels + !!s->lfe;
- if(avctx->request_channels == 2 && s->prim_channels > 2) {
- channels = 2;
- s->output = DCA_STEREO;
- avctx->channel_layout = CH_LAYOUT_STEREO;
+ s->profile = FF_PROFILE_DTS;
+
+ for (i = 0; i < (s->sample_blocks / 8); i++) {
+ if ((ret = dca_decode_block(s, 0, i))) {
+ av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
+ return ret;
+ }
+ }
+
+ /* record number of core channels incase less than max channels are requested */
+ num_core_channels = s->prim_channels;
+
+ if (s->ext_coding)
+ s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
+ else
+ s->core_ext_mask = 0;
+
+ core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;
+
+ /* only scan for extensions if ext_descr was unknown or indicated a
+ * supported XCh extension */
+ if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {
+
+ /* if ext_descr was unknown, clear s->core_ext_mask so that the
+ * extensions scan can fill it up */
+ s->core_ext_mask = FFMAX(s->core_ext_mask, 0);
+
+ /* extensions start at 32-bit boundaries into bitstream */
+ skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
+
+ while (core_ss_end - get_bits_count(&s->gb) >= 32) {
+ uint32_t bits = get_bits_long(&s->gb, 32);
+
+ switch (bits) {
+ case 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(number of channels in extension) should be 1 */
+ /* AFAIK XCh is not used for more channels */
+ if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
+ av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"
+ " supported!\n", ext_amode);
+ continue;
+ }
+
+ /* much like core primary audio coding header */
+ dca_parse_audio_coding_header(s, s->xch_base_channel);
+
+ for (i = 0; i < (s->sample_blocks / 8); i++)
+ if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
+ av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
+ continue;
+ }
+
+ s->xch_present = 1;
+ break;
+ }
+ case 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->amode<16)
+
+ 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->lfe) avctx->channel_layout |= CH_LOW_FREQUENCY;
+ 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 AVERROR_INVALIDDATA;
+
+ 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 AVERROR_INVALIDDATA;
+ }
+
/* 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
+ 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 (!avctx->channels)
+ 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 AVERROR_PATCHWELCOME;
+ }
- if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
- return -1;
- *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
+ /* get output buffer */
+ s->frame.nb_samples = 256 * (s->sample_blocks / 8);
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ samples_flt = (float *) s->frame.data[0];
+ samples_s16 = (int16_t *) s->frame.data[0];
+
+ /* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
- dca_decode_block(s);
- s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
- samples += 256 * channels;
+ 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;
+ s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
+ s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
+ }
+
+ 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];
+
+ *got_frame_ptr = 1;
+ *(AVFrame *) data = s->frame;
+
return buf_size;
}
* @param avctx pointer to the AVCodecContext
*/
-static av_cold int dca_decode_init(AVCodecContext * avctx)
+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);
+ ff_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 < 6; i++)
+ for (i = 0; i < DCA_PRIM_CHANNELS_MAX + 1; i++)
s->samples_chanptr[i] = s->samples + i * 256;
- avctx->sample_fmt = SAMPLE_FMT_S16;
- if(s->dsp.float_to_int16 == ff_float_to_int16_c) {
- s->add_bias = 385.0f;
- s->scale_bias = 1.0 / 32768.0;
+ if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ s->scale_bias = 1.0 / 32768.0;
} else {
- s->add_bias = 0.0f;
- s->scale_bias = 1.0;
+ 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;
- }
+ /* allow downmixing to stereo */
+ if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
+ avctx->request_channels == 2) {
+ avctx->channels = avctx->request_channels;
}
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
return 0;
}
-static av_cold int dca_decode_end(AVCodecContext * avctx)
+static av_cold int dca_decode_end(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
ff_mdct_end(&s->imdct);
return 0;
}
-AVCodec dca_decoder = {
- .name = "dca",
- .type = CODEC_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)"),
- .channel_layouts = dca_core_channel_layout,
+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 | CODEC_CAP_DR1,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
+ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
+ .profiles = NULL_IF_CONFIG_SMALL(profiles),
};
projects / ffmpeg / blobdiff