/*
- * DCA compatible decoder
- * Copyright (C) 2004 Gildas Bazin
- * Copyright (C) 2004 Benjamin Zores
- * Copyright (C) 2006 Benjamin Larsson
- * Copyright (C) 2007 Konstantin Shishkov
+ * DCA compatible decoder data
*
- * 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
*/
-#include <math.h>
-#include <stddef.h>
-#include <stdio.h>
+#include <stdint.h>
-#include "libavutil/intmath.h"
-#include "libavutil/intreadwrite.h"
-#include "libavcore/audioconvert.h"
-#include "avcodec.h"
-#include "dsputil.h"
-#include "fft.h"
-#include "get_bits.h"
-#include "put_bits.h"
-#include "dcadata.h"
-#include "dcahuff.h"
#include "dca.h"
-#include "synth_filter.h"
-#include "dcadsp.h"
-//#define TRACE
-
-#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_CHANNEL,
- DCA_STEREO,
- DCA_STEREO_SUMDIFF,
- DCA_STEREO_TOTAL,
- DCA_3F,
- DCA_2F1R,
- DCA_3F1R,
- DCA_2F2R,
- DCA_3F2R,
- DCA_4F2R
-};
-
-/* 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 -> 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
- */
-
-static const int64_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|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 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},
-};
-
-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},
-};
-
-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},
-};
-
-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 DCA_DOLBY 101 /* FIXME */
-
-#define DCA_CHANNEL_BITS 6
-#define DCA_CHANNEL_MASK 0x3F
-
-#define DCA_LFE 0x80
-
-#define HEADER_SIZE 14
-
-#define DCA_MAX_FRAME_SIZE 16384
-
-/** Bit allocation */
-typedef struct {
- int offset; ///< code values offset
- int maxbits[8]; ///< max bits in VLC
- int wrap; ///< wrap for get_vlc2()
- VLC vlc[8]; ///< actual codes
-} BitAlloc;
-
-static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select
-static BitAlloc dca_tmode; ///< transition mode VLCs
-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)
-{
- return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
-}
-
-typedef struct {
- AVCodecContext *avctx;
- /* Frame header */
- int frame_type; ///< type of the current frame
- int samples_deficit; ///< deficit sample count
- int crc_present; ///< crc is present in the bitstream
- int sample_blocks; ///< number of PCM sample blocks
- int frame_size; ///< primary frame byte size
- int amode; ///< audio channels arrangement
- int sample_rate; ///< audio sampling rate
- int bit_rate; ///< transmission bit rate
- int bit_rate_index; ///< transmission bit rate index
-
- int downmix; ///< embedded downmix enabled
- int dynrange; ///< embedded dynamic range flag
- int timestamp; ///< embedded time stamp flag
- int aux_data; ///< auxiliary data flag
- int hdcd; ///< source material is mastered in HDCD
- int ext_descr; ///< extension audio descriptor flag
- int ext_coding; ///< extended coding flag
- int aspf; ///< audio sync word insertion flag
- int lfe; ///< low frequency effects flag
- int predictor_history; ///< predictor history flag
- int header_crc; ///< header crc check bytes
- int multirate_inter; ///< multirate interpolator switch
- int version; ///< encoder software revision
- int copy_history; ///< copy history
- int source_pcm_res; ///< source pcm resolution
- int front_sum; ///< front sum/difference flag
- int surround_sum; ///< surround sum/difference flag
- int dialog_norm; ///< dialog normalisation parameter
-
- /* Primary audio coding header */
- int subframes; ///< number of subframes
- 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
- int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband
- int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index
- int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book
- int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
- int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select
- int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
- float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
-
- /* Primary audio coding side information */
- 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 transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients)
- int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient)
- int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook
- int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
- int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients
- int dynrange_coef; ///< dynamic range coefficient
-
- int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
-
- 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];
- DECLARE_ALIGNED(16, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
- int hist_index[DCA_PRIM_CHANNELS_MAX];
- DECLARE_ALIGNED(16, float, raXin)[32];
-
- int output; ///< type of output
- float add_bias; ///< output bias
- float scale_bias; ///< output scale
-
- DECLARE_ALIGNED(16, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
- DECLARE_ALIGNED(16, 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];
- 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;
-
- /* XCh extension information */
- int xch_present;
- int xch_base_channel; ///< index of first (only) channel containing XCH data
-
- int debug_flag; ///< used for suppressing repeated error messages output
- DSPContext dsp;
- FFTContext imdct;
- SynthFilterContext synth;
- DCADSPContext dcadsp;
-} DCAContext;
-
-static const uint16_t dca_vlc_offs[] = {
- 0, 512, 640, 768, 1282, 1794, 2436, 3080, 3770, 4454, 5364,
- 5372, 5380, 5388, 5392, 5396, 5412, 5420, 5428, 5460, 5492, 5508,
- 5572, 5604, 5668, 5796, 5860, 5892, 6412, 6668, 6796, 7308, 7564,
- 7820, 8076, 8620, 9132, 9388, 9910, 10166, 10680, 11196, 11726, 12240,
- 12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,
- 18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,
-};
-
-static av_cold void dca_init_vlcs(void)
-{
- static int vlcs_initialized = 0;
- int i, j, c = 14;
- static VLC_TYPE dca_table[23622][2];
-
- if (vlcs_initialized)
- return;
-
- dca_bitalloc_index.offset = 1;
- dca_bitalloc_index.wrap = 2;
- for (i = 0; i < 5; i++) {
- dca_bitalloc_index.vlc[i].table = &dca_table[dca_vlc_offs[i]];
- dca_bitalloc_index.vlc[i].table_allocated = dca_vlc_offs[i + 1] - dca_vlc_offs[i];
- init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
- bitalloc_12_bits[i], 1, 1,
- bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
- }
- dca_scalefactor.offset = -64;
- dca_scalefactor.wrap = 2;
- for (i = 0; i < 5; i++) {
- dca_scalefactor.vlc[i].table = &dca_table[dca_vlc_offs[i + 5]];
- dca_scalefactor.vlc[i].table_allocated = dca_vlc_offs[i + 6] - dca_vlc_offs[i + 5];
- init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
- scales_bits[i], 1, 1,
- scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
- }
- dca_tmode.offset = 0;
- dca_tmode.wrap = 1;
- for (i = 0; i < 4; i++) {
- dca_tmode.vlc[i].table = &dca_table[dca_vlc_offs[i + 10]];
- dca_tmode.vlc[i].table_allocated = dca_vlc_offs[i + 11] - dca_vlc_offs[i + 10];
- init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
- tmode_bits[i], 1, 1,
- tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
- }
-
- for (i = 0; i < 10; i++)
- for (j = 0; j < 7; j++){
- if (!bitalloc_codes[i][j]) break;
- dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
- dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
- dca_smpl_bitalloc[i+1].vlc[j].table = &dca_table[dca_vlc_offs[c]];
- dca_smpl_bitalloc[i+1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
- init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],
- bitalloc_sizes[i],
- bitalloc_bits[i][j], 1, 1,
- bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
- c++;
- }
- vlcs_initialized = 1;
-}
-
-static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
-{
- while(len--)
- *dst++ = get_bits(gb, bits);
-}
-
-static int dca_parse_audio_coding_header(DCAContext * s, int base_channel)
+const uint32_t avpriv_dca_sample_rates[16] =
{
- 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 (!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]);
- }
- }
- }
-
- 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);
- }
-
- /* 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, float bias)
-{
- 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, bias);
- 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,
- float bias)
-{
- /* 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, bias);
- samples_in++;
- samples_out += 2 * decifactor;
- }
-}
-
-/* downmixing routines */
-#define MIX_REAR1(samples, si1, rs, coef) \
- samples[i] += (samples[si1] - add_bias) * coef[rs][0]; \
- samples[i+256] += (samples[si1] - add_bias) * coef[rs][1];
-
-#define MIX_REAR2(samples, si1, si2, rs, coef) \
- samples[i] += (samples[si1] - add_bias) * coef[rs][0] + (samples[si2] - add_bias) * coef[rs+1][0]; \
- samples[i+256] += (samples[si1] - add_bias) * coef[rs][1] + (samples[si2] - add_bias) * coef[rs+1][1];
-
-#define MIX_FRONT3(samples, coef) \
- t = samples[i+c] - add_bias; \
- u = samples[i+l] - add_bias; \
- v = samples[i+r] - add_bias; \
- samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0] + add_bias; \
- samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1] + add_bias;
-
-#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, float add_bias)
-{
- 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++) {
- 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->dsp.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] */ ,
- s->add_bias );
- }
-
- /* 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, s->add_bias);
- }
-
- /* 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, s->add_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;
- }
-}
-
-/**
- * Main frame decoding function
- * FIXME add arguments
- */
-static int dca_decode_frame(AVCodecContext * avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
-{
- const uint8_t *buf = avpkt->data;
- int buf_size = avpkt->size;
-
- int lfe_samples;
- int num_core_channels = 0;
- int i;
- int16_t *samples = data;
- DCAContext *s = avctx->priv_data;
- int channels;
-
-
- 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) {
- 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;
-
- 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;
-
- /* extensions start at 32-bit boundaries into bitstream */
- skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
-
- while(get_bits_left(&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);
-
- /* 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 0x1d95f262:
- av_log(avctx, AV_LOG_DEBUG, "Possible X96 extension found at %d bits\n", get_bits_count(&s->gb));
- av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", get_bits(&s->gb, 12)+1);
- av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
- break;
- }
-
- skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
- }
-
- 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.*/
- avctx->channels = channels;
-
- 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;
-
- /* 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] - s->add_bias) * M_SQRT1_2;
- rt_chan[j] -= (back_chan[j] - s->add_bias) * M_SQRT1_2;
- }
- }
-
- s->dsp.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);
-
- for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
- s->samples_chanptr[i] = s->samples + i * 256;
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
-
- if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
- s->add_bias = 385.0f;
- s->scale_bias = 1.0 / 32768.0;
- } else {
- s->add_bias = 0.0f;
- 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;
-}
-
-AVCodec 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,
+ 0, 8000, 16000, 32000, 0, 0, 11025, 22050, 44100, 0, 0,
+ 12000, 24000, 48000, 96000, 192000
};
projects / ffmpeg / blobdiff