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1
/*
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 * DCA compatible decoder
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 * Copyright (C) 2004 Gildas Bazin
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 * Copyright (C) 2004 Benjamin Zores
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 * Copyright (C) 2006 Benjamin Larsson
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 * Copyright (C) 2007 Konstantin Shishkov
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 *
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
11
 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
24

    
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/**
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 * @file dca.c
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 */
28

    
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#include <math.h>
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#include <stddef.h>
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#include <stdio.h>
32

    
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#include "avcodec.h"
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#include "dsputil.h"
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#include "bitstream.h"
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#include "dcadata.h"
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#include "dcahuff.h"
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#include "dca.h"
39

    
40
//#define TRACE
41

    
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#define DCA_PRIM_CHANNELS_MAX (5)
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#define DCA_SUBBANDS (32)
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#define DCA_ABITS_MAX (32)      /* Should be 28 */
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#define DCA_SUBSUBFAMES_MAX (4)
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#define DCA_LFE_MAX (3)
47

    
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enum DCAMode {
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    DCA_MONO = 0,
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    DCA_CHANNEL,
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    DCA_STEREO,
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    DCA_STEREO_SUMDIFF,
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    DCA_STEREO_TOTAL,
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    DCA_3F,
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    DCA_2F1R,
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    DCA_3F1R,
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    DCA_2F2R,
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    DCA_3F2R,
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    DCA_4F2R
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};
61

    
62
#define DCA_DOLBY 101           /* FIXME */
63

    
64
#define DCA_CHANNEL_BITS 6
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#define DCA_CHANNEL_MASK 0x3F
66

    
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#define DCA_LFE 0x80
68

    
69
#define HEADER_SIZE 14
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#define CONVERT_BIAS 384
71

    
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#define DCA_MAX_FRAME_SIZE 16383
73

    
74
/** Bit allocation */
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typedef struct {
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    int offset;                 ///< code values offset
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    int maxbits[8];             ///< max bits in VLC
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    int wrap;                   ///< wrap for get_vlc2()
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    VLC vlc[8];                 ///< actual codes
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} BitAlloc;
81

    
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static BitAlloc dca_bitalloc_index;    ///< indexes for samples VLC select
83
static BitAlloc dca_tmode;             ///< transition mode VLCs
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static BitAlloc dca_scalefactor;       ///< scalefactor VLCs
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static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
86

    
87
/** Pre-calculated cosine modulation coefs for the QMF */
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static float cos_mod[544];
89

    
90
static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
91
{
92
    return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
93
}
94

    
95
typedef struct {
96
    AVCodecContext *avctx;
97
    /* Frame header */
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    int frame_type;             ///< type of the current frame
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    int samples_deficit;        ///< deficit sample count
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    int crc_present;            ///< crc is present in the bitstream
101
    int sample_blocks;          ///< number of PCM sample blocks
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    int frame_size;             ///< primary frame byte size
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    int amode;                  ///< audio channels arrangement
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    int sample_rate;            ///< audio sampling rate
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    int bit_rate;               ///< transmission bit rate
106

    
107
    int downmix;                ///< embedded downmix enabled
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    int dynrange;               ///< embedded dynamic range flag
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    int timestamp;              ///< embedded time stamp flag
110
    int aux_data;               ///< auxiliary data flag
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    int hdcd;                   ///< source material is mastered in HDCD
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    int ext_descr;              ///< extension audio descriptor flag
113
    int ext_coding;             ///< extended coding flag
114
    int aspf;                   ///< audio sync word insertion flag
115
    int lfe;                    ///< low frequency effects flag
116
    int predictor_history;      ///< predictor history flag
117
    int header_crc;             ///< header crc check bytes
118
    int multirate_inter;        ///< multirate interpolator switch
119
    int version;                ///< encoder software revision
120
    int copy_history;           ///< copy history
121
    int source_pcm_res;         ///< source pcm resolution
122
    int front_sum;              ///< front sum/difference flag
123
    int surround_sum;           ///< surround sum/difference flag
124
    int dialog_norm;            ///< dialog normalisation parameter
125

    
126
    /* Primary audio coding header */
127
    int subframes;              ///< number of subframes
128
    int prim_channels;          ///< number of primary audio channels
129
    int subband_activity[DCA_PRIM_CHANNELS_MAX];    ///< subband activity count
130
    int vq_start_subband[DCA_PRIM_CHANNELS_MAX];    ///< high frequency vq start subband
131
    int joint_intensity[DCA_PRIM_CHANNELS_MAX];     ///< joint intensity coding index
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    int transient_huffman[DCA_PRIM_CHANNELS_MAX];   ///< transient mode code book
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    int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
134
    int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX];    ///< bit allocation quantizer select
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    int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
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    float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX];   ///< scale factor adjustment
137

    
138
    /* Primary audio coding side information */
139
    int subsubframes;           ///< number of subsubframes
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    int partial_samples;        ///< partial subsubframe samples count
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    int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< prediction mode (ADPCM used or not)
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    int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];      ///< prediction VQ coefs
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    int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];           ///< bit allocation index
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    int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];    ///< transition mode (transients)
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    int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2];    ///< scale factors (2 if transient)
146
    int joint_huff[DCA_PRIM_CHANNELS_MAX];                       ///< joint subband scale factors codebook
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    int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
148
    int downmix_coef[DCA_PRIM_CHANNELS_MAX][2];                  ///< stereo downmix coefficients
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    int dynrange_coef;                                           ///< dynamic range coefficient
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151
    int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS];       ///< VQ encoded high frequency subbands
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153
    float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX *
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                   2 /*history */ ];    ///< Low frequency effect data
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    int lfe_scale_factor;
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157
    /* Subband samples history (for ADPCM) */
158
    float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
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    float subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512];
160
    float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][64];
161

    
162
    int output;                 ///< type of output
163
    int bias;                   ///< output bias
164

    
165
    DECLARE_ALIGNED_16(float, samples[1536]);  /* 6 * 256 = 1536, might only need 5 */
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    DECLARE_ALIGNED_16(int16_t, tsamples[1536]);
167

    
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    uint8_t dca_buffer[DCA_MAX_FRAME_SIZE];
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    int dca_buffer_size;        ///< how much data is in the dca_buffer
170

    
171
    GetBitContext gb;
172
    /* Current position in DCA frame */
173
    int current_subframe;
174
    int current_subsubframe;
175

    
176
    int debug_flag;             ///< used for suppressing repeated error messages output
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    DSPContext dsp;
178
} DCAContext;
179

    
180
static void dca_init_vlcs(void)
181
{
182
    static int vlcs_inited = 0;
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    int i, j;
184

    
185
    if (vlcs_inited)
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        return;
187

    
188
    dca_bitalloc_index.offset = 1;
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    dca_bitalloc_index.wrap = 2;
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    for (i = 0; i < 5; i++)
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        init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
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                 bitalloc_12_bits[i], 1, 1,
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                 bitalloc_12_codes[i], 2, 2, 1);
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    dca_scalefactor.offset = -64;
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    dca_scalefactor.wrap = 2;
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    for (i = 0; i < 5; i++)
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        init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
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                 scales_bits[i], 1, 1,
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                 scales_codes[i], 2, 2, 1);
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    dca_tmode.offset = 0;
201
    dca_tmode.wrap = 1;
202
    for (i = 0; i < 4; i++)
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        init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
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                 tmode_bits[i], 1, 1,
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                 tmode_codes[i], 2, 2, 1);
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207
    for(i = 0; i < 10; i++)
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        for(j = 0; j < 7; j++){
209
            if(!bitalloc_codes[i][j]) break;
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            dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
211
            dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
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            init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],
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                     bitalloc_sizes[i],
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                     bitalloc_bits[i][j], 1, 1,
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                     bitalloc_codes[i][j], 2, 2, 1);
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        }
217
    vlcs_inited = 1;
218
}
219

    
220
static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
221
{
222
    while(len--)
223
        *dst++ = get_bits(gb, bits);
224
}
225

    
226
static int dca_parse_frame_header(DCAContext * s)
227
{
228
    int i, j;
229
    static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
230
    static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
231
    static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
232

    
233
    s->bias = CONVERT_BIAS;
234

    
235
    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
236

    
237
    /* Sync code */
238
    get_bits(&s->gb, 32);
239

    
240
    /* Frame header */
241
    s->frame_type        = get_bits(&s->gb, 1);
242
    s->samples_deficit   = get_bits(&s->gb, 5) + 1;
243
    s->crc_present       = get_bits(&s->gb, 1);
244
    s->sample_blocks     = get_bits(&s->gb, 7) + 1;
245
    s->frame_size        = get_bits(&s->gb, 14) + 1;
246
    if (s->frame_size < 95)
247
        return -1;
248
    s->amode             = get_bits(&s->gb, 6);
249
    s->sample_rate       = dca_sample_rates[get_bits(&s->gb, 4)];
250
    if (!s->sample_rate)
251
        return -1;
252
    s->bit_rate          = dca_bit_rates[get_bits(&s->gb, 5)];
253
    if (!s->bit_rate)
254
        return -1;
255

    
256
    s->downmix           = get_bits(&s->gb, 1);
257
    s->dynrange          = get_bits(&s->gb, 1);
258
    s->timestamp         = get_bits(&s->gb, 1);
259
    s->aux_data          = get_bits(&s->gb, 1);
260
    s->hdcd              = get_bits(&s->gb, 1);
261
    s->ext_descr         = get_bits(&s->gb, 3);
262
    s->ext_coding        = get_bits(&s->gb, 1);
263
    s->aspf              = get_bits(&s->gb, 1);
264
    s->lfe               = get_bits(&s->gb, 2);
265
    s->predictor_history = get_bits(&s->gb, 1);
266

    
267
    /* TODO: check CRC */
268
    if (s->crc_present)
269
        s->header_crc    = get_bits(&s->gb, 16);
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271
    s->multirate_inter   = get_bits(&s->gb, 1);
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    s->version           = get_bits(&s->gb, 4);
273
    s->copy_history      = get_bits(&s->gb, 2);
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    s->source_pcm_res    = get_bits(&s->gb, 3);
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    s->front_sum         = get_bits(&s->gb, 1);
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    s->surround_sum      = get_bits(&s->gb, 1);
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    s->dialog_norm       = get_bits(&s->gb, 4);
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279
    /* FIXME: channels mixing levels */
280
    s->output = s->amode;
281
    if(s->lfe) s->output |= DCA_LFE;
282

    
283
#ifdef TRACE
284
    av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
285
    av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit);
286
    av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present);
287
    av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n",
288
           s->sample_blocks, s->sample_blocks * 32);
289
    av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size);
290
    av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n",
291
           s->amode, dca_channels[s->amode]);
292
    av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i (%i Hz)\n",
293
           s->sample_rate, dca_sample_rates[s->sample_rate]);
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    av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i (%i bits/s)\n",
295
           s->bit_rate, dca_bit_rates[s->bit_rate]);
296
    av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix);
297
    av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
298
    av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
299
    av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);
300
    av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd);
301
    av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr);
302
    av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding);
303
    av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf);
304
    av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe);
305
    av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n",
306
           s->predictor_history);
307
    av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc);
308
    av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n",
309
           s->multirate_inter);
310
    av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version);
311
    av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history);
312
    av_log(s->avctx, AV_LOG_DEBUG,
313
           "source pcm resolution: %i (%i bits/sample)\n",
314
           s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]);
315
    av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum);
316
    av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum);
317
    av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm);
318
    av_log(s->avctx, AV_LOG_DEBUG, "\n");
319
#endif
320

    
321
    /* Primary audio coding header */
322
    s->subframes         = get_bits(&s->gb, 4) + 1;
323
    s->prim_channels     = get_bits(&s->gb, 3) + 1;
324

    
325

    
326
    for (i = 0; i < s->prim_channels; i++) {
327
        s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
328
        if (s->subband_activity[i] > DCA_SUBBANDS)
329
            s->subband_activity[i] = DCA_SUBBANDS;
330
    }
331
    for (i = 0; i < s->prim_channels; i++) {
332
        s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
333
        if (s->vq_start_subband[i] > DCA_SUBBANDS)
334
            s->vq_start_subband[i] = DCA_SUBBANDS;
335
    }
336
    get_array(&s->gb, s->joint_intensity,     s->prim_channels, 3);
337
    get_array(&s->gb, s->transient_huffman,   s->prim_channels, 2);
338
    get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);
339
    get_array(&s->gb, s->bitalloc_huffman,    s->prim_channels, 3);
340

    
341
    /* Get codebooks quantization indexes */
342
    memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
343
    for (j = 1; j < 11; j++)
344
        for (i = 0; i < s->prim_channels; i++)
345
            s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
346

    
347
    /* Get scale factor adjustment */
348
    for (j = 0; j < 11; j++)
349
        for (i = 0; i < s->prim_channels; i++)
350
            s->scalefactor_adj[i][j] = 1;
351

    
352
    for (j = 1; j < 11; j++)
353
        for (i = 0; i < s->prim_channels; i++)
354
            if (s->quant_index_huffman[i][j] < thr[j])
355
                s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
356

    
357
    if (s->crc_present) {
358
        /* Audio header CRC check */
359
        get_bits(&s->gb, 16);
360
    }
361

    
362
    s->current_subframe = 0;
363
    s->current_subsubframe = 0;
364

    
365
#ifdef TRACE
366
    av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
367
    av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
368
    for(i = 0; i < s->prim_channels; i++){
369
        av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
370
        av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
371
        av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
372
        av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
373
        av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
374
        av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
375
        av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
376
        for (j = 0; j < 11; j++)
377
            av_log(s->avctx, AV_LOG_DEBUG, " %i",
378
                   s->quant_index_huffman[i][j]);
379
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
380
        av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
381
        for (j = 0; j < 11; j++)
382
            av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
383
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
384
    }
385
#endif
386

    
387
    return 0;
388
}
389

    
390

    
391
static inline int get_scale(GetBitContext *gb, int level, int value)
392
{
393
   if (level < 5) {
394
       /* huffman encoded */
395
       value += get_bitalloc(gb, &dca_scalefactor, level);
396
   } else if(level < 8)
397
       value = get_bits(gb, level + 1);
398
   return value;
399
}
400

    
401
static int dca_subframe_header(DCAContext * s)
402
{
403
    /* Primary audio coding side information */
404
    int j, k;
405

    
406
    s->subsubframes = get_bits(&s->gb, 2) + 1;
407
    s->partial_samples = get_bits(&s->gb, 3);
408
    for (j = 0; j < s->prim_channels; j++) {
409
        for (k = 0; k < s->subband_activity[j]; k++)
410
            s->prediction_mode[j][k] = get_bits(&s->gb, 1);
411
    }
412

    
413
    /* Get prediction codebook */
414
    for (j = 0; j < s->prim_channels; j++) {
415
        for (k = 0; k < s->subband_activity[j]; k++) {
416
            if (s->prediction_mode[j][k] > 0) {
417
                /* (Prediction coefficient VQ address) */
418
                s->prediction_vq[j][k] = get_bits(&s->gb, 12);
419
            }
420
        }
421
    }
422

    
423
    /* Bit allocation index */
424
    for (j = 0; j < s->prim_channels; j++) {
425
        for (k = 0; k < s->vq_start_subband[j]; k++) {
426
            if (s->bitalloc_huffman[j] == 6)
427
                s->bitalloc[j][k] = get_bits(&s->gb, 5);
428
            else if (s->bitalloc_huffman[j] == 5)
429
                s->bitalloc[j][k] = get_bits(&s->gb, 4);
430
            else {
431
                s->bitalloc[j][k] =
432
                    get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
433
            }
434

    
435
            if (s->bitalloc[j][k] > 26) {
436
//                 av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n",
437
//                          j, k, s->bitalloc[j][k]);
438
                return -1;
439
            }
440
        }
441
    }
442

    
443
    /* Transition mode */
444
    for (j = 0; j < s->prim_channels; j++) {
445
        for (k = 0; k < s->subband_activity[j]; k++) {
446
            s->transition_mode[j][k] = 0;
447
            if (s->subsubframes > 1 &&
448
                k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
449
                s->transition_mode[j][k] =
450
                    get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
451
            }
452
        }
453
    }
454

    
455
    for (j = 0; j < s->prim_channels; j++) {
456
        const uint32_t *scale_table;
457
        int scale_sum;
458

    
459
        memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
460

    
461
        if (s->scalefactor_huffman[j] == 6)
462
            scale_table = scale_factor_quant7;
463
        else
464
            scale_table = scale_factor_quant6;
465

    
466
        /* When huffman coded, only the difference is encoded */
467
        scale_sum = 0;
468

    
469
        for (k = 0; k < s->subband_activity[j]; k++) {
470
            if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
471
                scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
472
                s->scale_factor[j][k][0] = scale_table[scale_sum];
473
            }
474

    
475
            if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
476
                /* Get second scale factor */
477
                scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
478
                s->scale_factor[j][k][1] = scale_table[scale_sum];
479
            }
480
        }
481
    }
482

    
483
    /* Joint subband scale factor codebook select */
484
    for (j = 0; j < s->prim_channels; j++) {
485
        /* Transmitted only if joint subband coding enabled */
486
        if (s->joint_intensity[j] > 0)
487
            s->joint_huff[j] = get_bits(&s->gb, 3);
488
    }
489

    
490
    /* Scale factors for joint subband coding */
491
    for (j = 0; j < s->prim_channels; j++) {
492
        int source_channel;
493

    
494
        /* Transmitted only if joint subband coding enabled */
495
        if (s->joint_intensity[j] > 0) {
496
            int scale = 0;
497
            source_channel = s->joint_intensity[j] - 1;
498

    
499
            /* When huffman coded, only the difference is encoded
500
             * (is this valid as well for joint scales ???) */
501

    
502
            for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
503
                scale = get_scale(&s->gb, s->joint_huff[j], 0);
504
                scale += 64;    /* bias */
505
                s->joint_scale_factor[j][k] = scale;    /*joint_scale_table[scale]; */
506
            }
507

    
508
            if (!s->debug_flag & 0x02) {
509
                av_log(s->avctx, AV_LOG_DEBUG,
510
                       "Joint stereo coding not supported\n");
511
                s->debug_flag |= 0x02;
512
            }
513
        }
514
    }
515

    
516
    /* Stereo downmix coefficients */
517
    if (s->prim_channels > 2) {
518
        if(s->downmix) {
519
            for (j = 0; j < s->prim_channels; j++) {
520
                s->downmix_coef[j][0] = get_bits(&s->gb, 7);
521
                s->downmix_coef[j][1] = get_bits(&s->gb, 7);
522
            }
523
        } else {
524
            int am = s->amode & DCA_CHANNEL_MASK;
525
            for (j = 0; j < s->prim_channels; j++) {
526
                s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];
527
                s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];
528
            }
529
        }
530
    }
531

    
532
    /* Dynamic range coefficient */
533
    if (s->dynrange)
534
        s->dynrange_coef = get_bits(&s->gb, 8);
535

    
536
    /* Side information CRC check word */
537
    if (s->crc_present) {
538
        get_bits(&s->gb, 16);
539
    }
540

    
541
    /*
542
     * Primary audio data arrays
543
     */
544

    
545
    /* VQ encoded high frequency subbands */
546
    for (j = 0; j < s->prim_channels; j++)
547
        for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
548
            /* 1 vector -> 32 samples */
549
            s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
550

    
551
    /* Low frequency effect data */
552
    if (s->lfe) {
553
        /* LFE samples */
554
        int lfe_samples = 2 * s->lfe * s->subsubframes;
555
        float lfe_scale;
556

    
557
        for (j = lfe_samples; j < lfe_samples * 2; j++) {
558
            /* Signed 8 bits int */
559
            s->lfe_data[j] = get_sbits(&s->gb, 8);
560
        }
561

    
562
        /* Scale factor index */
563
        s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)];
564

    
565
        /* Quantization step size * scale factor */
566
        lfe_scale = 0.035 * s->lfe_scale_factor;
567

    
568
        for (j = lfe_samples; j < lfe_samples * 2; j++)
569
            s->lfe_data[j] *= lfe_scale;
570
    }
571

    
572
#ifdef TRACE
573
    av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes);
574
    av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
575
           s->partial_samples);
576
    for (j = 0; j < s->prim_channels; j++) {
577
        av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
578
        for (k = 0; k < s->subband_activity[j]; k++)
579
            av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
580
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
581
    }
582
    for (j = 0; j < s->prim_channels; j++) {
583
        for (k = 0; k < s->subband_activity[j]; k++)
584
                av_log(s->avctx, AV_LOG_DEBUG,
585
                       "prediction coefs: %f, %f, %f, %f\n",
586
                       (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
587
                       (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
588
                       (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
589
                       (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
590
    }
591
    for (j = 0; j < s->prim_channels; j++) {
592
        av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
593
        for (k = 0; k < s->vq_start_subband[j]; k++)
594
            av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);
595
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
596
    }
597
    for (j = 0; j < s->prim_channels; j++) {
598
        av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");
599
        for (k = 0; k < s->subband_activity[j]; k++)
600
            av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);
601
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
602
    }
603
    for (j = 0; j < s->prim_channels; j++) {
604
        av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");
605
        for (k = 0; k < s->subband_activity[j]; k++) {
606
            if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)
607
                av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]);
608
            if (k < s->vq_start_subband[j] && s->transition_mode[j][k])
609
                av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]);
610
        }
611
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
612
    }
613
    for (j = 0; j < s->prim_channels; j++) {
614
        if (s->joint_intensity[j] > 0) {
615
            int source_channel = s->joint_intensity[j] - 1;
616
            av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");
617
            for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)
618
                av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);
619
            av_log(s->avctx, AV_LOG_DEBUG, "\n");
620
        }
621
    }
622
    if (s->prim_channels > 2 && s->downmix) {
623
        av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");
624
        for (j = 0; j < s->prim_channels; j++) {
625
            av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]);
626
            av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]);
627
        }
628
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
629
    }
630
    for (j = 0; j < s->prim_channels; j++)
631
        for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
632
            av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
633
    if(s->lfe){
634
        int lfe_samples = 2 * s->lfe * s->subsubframes;
635
        av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
636
        for (j = lfe_samples; j < lfe_samples * 2; j++)
637
            av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
638
        av_log(s->avctx, AV_LOG_DEBUG, "\n");
639
    }
640
#endif
641

    
642
    return 0;
643
}
644

    
645
static void qmf_32_subbands(DCAContext * s, int chans,
646
                            float samples_in[32][8], float *samples_out,
647
                            float scale, float bias)
648
{
649
    const float *prCoeff;
650
    int i, j, k;
651
    float praXin[33], *raXin = &praXin[1];
652

    
653
    float *subband_fir_hist = s->subband_fir_hist[chans];
654
    float *subband_fir_hist2 = s->subband_fir_noidea[chans];
655

    
656
    int chindex = 0, subindex;
657

    
658
    praXin[0] = 0.0;
659

    
660
    /* Select filter */
661
    if (!s->multirate_inter)    /* Non-perfect reconstruction */
662
        prCoeff = fir_32bands_nonperfect;
663
    else                        /* Perfect reconstruction */
664
        prCoeff = fir_32bands_perfect;
665

    
666
    /* Reconstructed channel sample index */
667
    for (subindex = 0; subindex < 8; subindex++) {
668
        float t1, t2, sum[16], diff[16];
669

    
670
        /* Load in one sample from each subband and clear inactive subbands */
671
        for (i = 0; i < s->subband_activity[chans]; i++)
672
            raXin[i] = samples_in[i][subindex];
673
        for (; i < 32; i++)
674
            raXin[i] = 0.0;
675

    
676
        /* Multiply by cosine modulation coefficients and
677
         * create temporary arrays SUM and DIFF */
678
        for (j = 0, k = 0; k < 16; k++) {
679
            t1 = 0.0;
680
            t2 = 0.0;
681
            for (i = 0; i < 16; i++, j++){
682
                t1 += (raXin[2 * i] + raXin[2 * i + 1]) * cos_mod[j];
683
                t2 += (raXin[2 * i] + raXin[2 * i - 1]) * cos_mod[j + 256];
684
            }
685
            sum[k] = t1 + t2;
686
            diff[k] = t1 - t2;
687
        }
688

    
689
        j = 512;
690
        /* Store history */
691
        for (k = 0; k < 16; k++)
692
            subband_fir_hist[k] = cos_mod[j++] * sum[k];
693
        for (k = 0; k < 16; k++)
694
            subband_fir_hist[32-k-1] = cos_mod[j++] * diff[k];
695

    
696
        /* Multiply by filter coefficients */
697
        for (k = 31, i = 0; i < 32; i++, k--)
698
            for (j = 0; j < 512; j += 64){
699
                subband_fir_hist2[i]    += prCoeff[i+j]  * ( subband_fir_hist[i+j] - subband_fir_hist[j+k]);
700
                subband_fir_hist2[i+32] += prCoeff[i+j+32]*(-subband_fir_hist[i+j] - subband_fir_hist[j+k]);
701
            }
702

    
703
        /* Create 32 PCM output samples */
704
        for (i = 0; i < 32; i++)
705
            samples_out[chindex++] = subband_fir_hist2[i] * scale + bias;
706

    
707
        /* Update working arrays */
708
        memmove(&subband_fir_hist[32], &subband_fir_hist[0], (512 - 32) * sizeof(float));
709
        memmove(&subband_fir_hist2[0], &subband_fir_hist2[32], 32 * sizeof(float));
710
        memset(&subband_fir_hist2[32], 0, 32 * sizeof(float));
711
    }
712
}
713

    
714
static void lfe_interpolation_fir(int decimation_select,
715
                                  int num_deci_sample, float *samples_in,
716
                                  float *samples_out, float scale,
717
                                  float bias)
718
{
719
    /* samples_in: An array holding decimated samples.
720
     *   Samples in current subframe starts from samples_in[0],
721
     *   while samples_in[-1], samples_in[-2], ..., stores samples
722
     *   from last subframe as history.
723
     *
724
     * samples_out: An array holding interpolated samples
725
     */
726

    
727
    int decifactor, k, j;
728
    const float *prCoeff;
729

    
730
    int interp_index = 0;       /* Index to the interpolated samples */
731
    int deciindex;
732

    
733
    /* Select decimation filter */
734
    if (decimation_select == 1) {
735
        decifactor = 128;
736
        prCoeff = lfe_fir_128;
737
    } else {
738
        decifactor = 64;
739
        prCoeff = lfe_fir_64;
740
    }
741
    /* Interpolation */
742
    for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
743
        /* One decimated sample generates decifactor interpolated ones */
744
        for (k = 0; k < decifactor; k++) {
745
            float rTmp = 0.0;
746
            //FIXME the coeffs are symetric, fix that
747
            for (j = 0; j < 512 / decifactor; j++)
748
                rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor];
749
            samples_out[interp_index++] = rTmp / scale + bias;
750
        }
751
    }
752
}
753

    
754
/* downmixing routines */
755
#define MIX_REAR1(samples, si1, rs, coef) \
756
     samples[i]     += samples[si1] * coef[rs][0]; \
757
     samples[i+256] += samples[si1] * coef[rs][1];
758

    
759
#define MIX_REAR2(samples, si1, si2, rs, coef) \
760
     samples[i]     += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \
761
     samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1];
762

    
763
#define MIX_FRONT3(samples, coef) \
764
    t = samples[i]; \
765
    samples[i]     = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \
766
    samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * coef[2][1];
767

    
768
#define DOWNMIX_TO_STEREO(op1, op2) \
769
    for(i = 0; i < 256; i++){ \
770
        op1 \
771
        op2 \
772
    }
773

    
774
static void dca_downmix(float *samples, int srcfmt,
775
                        int downmix_coef[DCA_PRIM_CHANNELS_MAX][2])
776
{
777
    int i;
778
    float t;
779
    float coef[DCA_PRIM_CHANNELS_MAX][2];
780

    
781
    for(i=0; i<DCA_PRIM_CHANNELS_MAX; i++) {
782
        coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];
783
        coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];
784
    }
785

    
786
    switch (srcfmt) {
787
    case DCA_MONO:
788
    case DCA_CHANNEL:
789
    case DCA_STEREO_TOTAL:
790
    case DCA_STEREO_SUMDIFF:
791
    case DCA_4F2R:
792
        av_log(NULL, 0, "Not implemented!\n");
793
        break;
794
    case DCA_STEREO:
795
        break;
796
    case DCA_3F:
797
        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);
798
        break;
799
    case DCA_2F1R:
800
        DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),);
801
        break;
802
    case DCA_3F1R:
803
        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
804
                          MIX_REAR1(samples, i + 768, 3, coef));
805
        break;
806
    case DCA_2F2R:
807
        DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),);
808
        break;
809
    case DCA_3F2R:
810
        DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
811
                          MIX_REAR2(samples, i + 768, i + 1024, 3, coef));
812
        break;
813
    }
814
}
815

    
816

    
817
/* Very compact version of the block code decoder that does not use table
818
 * look-up but is slightly slower */
819
static int decode_blockcode(int code, int levels, int *values)
820
{
821
    int i;
822
    int offset = (levels - 1) >> 1;
823

    
824
    for (i = 0; i < 4; i++) {
825
        values[i] = (code % levels) - offset;
826
        code /= levels;
827
    }
828

    
829
    if (code == 0)
830
        return 0;
831
    else {
832
        av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");
833
        return -1;
834
    }
835
}
836

    
837
static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
838
static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
839

    
840
static int dca_subsubframe(DCAContext * s)
841
{
842
    int k, l;
843
    int subsubframe = s->current_subsubframe;
844

    
845
    const float *quant_step_table;
846

    
847
    /* FIXME */
848
    float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
849

    
850
    /*
851
     * Audio data
852
     */
853

    
854
    /* Select quantization step size table */
855
    if (s->bit_rate == 0x1f)
856
        quant_step_table = lossless_quant_d;
857
    else
858
        quant_step_table = lossy_quant_d;
859

    
860
    for (k = 0; k < s->prim_channels; k++) {
861
        for (l = 0; l < s->vq_start_subband[k]; l++) {
862
            int m;
863

    
864
            /* Select the mid-tread linear quantizer */
865
            int abits = s->bitalloc[k][l];
866

    
867
            float quant_step_size = quant_step_table[abits];
868
            float rscale;
869

    
870
            /*
871
             * Determine quantization index code book and its type
872
             */
873

    
874
            /* Select quantization index code book */
875
            int sel = s->quant_index_huffman[k][abits];
876

    
877
            /*
878
             * Extract bits from the bit stream
879
             */
880
            if(!abits){
881
                memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
882
            }else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
883
                if(abits <= 7){
884
                    /* Block code */
885
                    int block_code1, block_code2, size, levels;
886
                    int block[8];
887

    
888
                    size = abits_sizes[abits-1];
889
                    levels = abits_levels[abits-1];
890

    
891
                    block_code1 = get_bits(&s->gb, size);
892
                    /* FIXME Should test return value */
893
                    decode_blockcode(block_code1, levels, block);
894
                    block_code2 = get_bits(&s->gb, size);
895
                    decode_blockcode(block_code2, levels, &block[4]);
896
                    for (m = 0; m < 8; m++)
897
                        subband_samples[k][l][m] = block[m];
898
                }else{
899
                    /* no coding */
900
                    for (m = 0; m < 8; m++)
901
                        subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3);
902
                }
903
            }else{
904
                /* Huffman coded */
905
                for (m = 0; m < 8; m++)
906
                    subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
907
            }
908

    
909
            /* Deal with transients */
910
            if (s->transition_mode[k][l] &&
911
                subsubframe >= s->transition_mode[k][l])
912
                rscale = quant_step_size * s->scale_factor[k][l][1];
913
            else
914
                rscale = quant_step_size * s->scale_factor[k][l][0];
915

    
916
            rscale *= s->scalefactor_adj[k][sel];
917

    
918
            for (m = 0; m < 8; m++)
919
                subband_samples[k][l][m] *= rscale;
920

    
921
            /*
922
             * Inverse ADPCM if in prediction mode
923
             */
924
            if (s->prediction_mode[k][l]) {
925
                int n;
926
                for (m = 0; m < 8; m++) {
927
                    for (n = 1; n <= 4; n++)
928
                        if (m >= n)
929
                            subband_samples[k][l][m] +=
930
                                (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
931
                                 subband_samples[k][l][m - n] / 8192);
932
                        else if (s->predictor_history)
933
                            subband_samples[k][l][m] +=
934
                                (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
935
                                 s->subband_samples_hist[k][l][m - n +
936
                                                               4] / 8192);
937
                }
938
            }
939
        }
940

    
941
        /*
942
         * Decode VQ encoded high frequencies
943
         */
944
        for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {
945
            /* 1 vector -> 32 samples but we only need the 8 samples
946
             * for this subsubframe. */
947
            int m;
948

    
949
            if (!s->debug_flag & 0x01) {
950
                av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");
951
                s->debug_flag |= 0x01;
952
            }
953

    
954
            for (m = 0; m < 8; m++) {
955
                subband_samples[k][l][m] =
956
                    high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 +
957
                                                        m]
958
                    * (float) s->scale_factor[k][l][0] / 16.0;
959
            }
960
        }
961
    }
962

    
963
    /* Check for DSYNC after subsubframe */
964
    if (s->aspf || subsubframe == s->subsubframes - 1) {
965
        if (0xFFFF == get_bits(&s->gb, 16)) {   /* 0xFFFF */
966
#ifdef TRACE
967
            av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
968
#endif
969
        } else {
970
            av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
971
        }
972
    }
973

    
974
    /* Backup predictor history for adpcm */
975
    for (k = 0; k < s->prim_channels; k++)
976
        for (l = 0; l < s->vq_start_subband[k]; l++)
977
            memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],
978
                        4 * sizeof(subband_samples[0][0][0]));
979

    
980
    /* 32 subbands QMF */
981
    for (k = 0; k < s->prim_channels; k++) {
982
/*        static float pcm_to_double[8] =
983
            {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
984
         qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * k],
985
                            2.0 / 3 /*pcm_to_double[s->source_pcm_res] */ ,
986
                            0 /*s->bias */ );
987
    }
988

    
989
    /* Down mixing */
990

    
991
    if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) {
992
        dca_downmix(s->samples, s->amode, s->downmix_coef);
993
    }
994

    
995
    /* Generate LFE samples for this subsubframe FIXME!!! */
996
    if (s->output & DCA_LFE) {
997
        int lfe_samples = 2 * s->lfe * s->subsubframes;
998
        int i_channels = dca_channels[s->output & DCA_CHANNEL_MASK];
999

    
1000
        lfe_interpolation_fir(s->lfe, 2 * s->lfe,
1001
                              s->lfe_data + lfe_samples +
1002
                              2 * s->lfe * subsubframe,
1003
                              &s->samples[256 * i_channels],
1004
                              256.0, 0 /* s->bias */);
1005
        /* Outputs 20bits pcm samples */
1006
    }
1007

    
1008
    return 0;
1009
}
1010

    
1011

    
1012
static int dca_subframe_footer(DCAContext * s)
1013
{
1014
    int aux_data_count = 0, i;
1015
    int lfe_samples;
1016

    
1017
    /*
1018
     * Unpack optional information
1019
     */
1020

    
1021
    if (s->timestamp)
1022
        get_bits(&s->gb, 32);
1023

    
1024
    if (s->aux_data)
1025
        aux_data_count = get_bits(&s->gb, 6);
1026

    
1027
    for (i = 0; i < aux_data_count; i++)
1028
        get_bits(&s->gb, 8);
1029

    
1030
    if (s->crc_present && (s->downmix || s->dynrange))
1031
        get_bits(&s->gb, 16);
1032

    
1033
    lfe_samples = 2 * s->lfe * s->subsubframes;
1034
    for (i = 0; i < lfe_samples; i++) {
1035
        s->lfe_data[i] = s->lfe_data[i + lfe_samples];
1036
    }
1037

    
1038
    return 0;
1039
}
1040

    
1041
/**
1042
 * Decode a dca frame block
1043
 *
1044
 * @param s     pointer to the DCAContext
1045
 */
1046

    
1047
static int dca_decode_block(DCAContext * s)
1048
{
1049

    
1050
    /* Sanity check */
1051
    if (s->current_subframe >= s->subframes) {
1052
        av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
1053
               s->current_subframe, s->subframes);
1054
        return -1;
1055
    }
1056

    
1057
    if (!s->current_subsubframe) {
1058
#ifdef TRACE
1059
        av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
1060
#endif
1061
        /* Read subframe header */
1062
        if (dca_subframe_header(s))
1063
            return -1;
1064
    }
1065

    
1066
    /* Read subsubframe */
1067
#ifdef TRACE
1068
    av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
1069
#endif
1070
    if (dca_subsubframe(s))
1071
        return -1;
1072

    
1073
    /* Update state */
1074
    s->current_subsubframe++;
1075
    if (s->current_subsubframe >= s->subsubframes) {
1076
        s->current_subsubframe = 0;
1077
        s->current_subframe++;
1078
    }
1079
    if (s->current_subframe >= s->subframes) {
1080
#ifdef TRACE
1081
        av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
1082
#endif
1083
        /* Read subframe footer */
1084
        if (dca_subframe_footer(s))
1085
            return -1;
1086
    }
1087

    
1088
    return 0;
1089
}
1090

    
1091
/**
1092
 * Convert bitstream to one representation based on sync marker
1093
 */
1094
static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,
1095
                          int max_size)
1096
{
1097
    uint32_t mrk;
1098
    int i, tmp;
1099
    const uint16_t *ssrc = (const uint16_t *) src;
1100
    uint16_t *sdst = (uint16_t *) dst;
1101
    PutBitContext pb;
1102

    
1103
    if((unsigned)src_size > (unsigned)max_size) {
1104
        av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");
1105
        return -1;
1106
    }
1107

    
1108
    mrk = AV_RB32(src);
1109
    switch (mrk) {
1110
    case DCA_MARKER_RAW_BE:
1111
        memcpy(dst, src, FFMIN(src_size, max_size));
1112
        return FFMIN(src_size, max_size);
1113
    case DCA_MARKER_RAW_LE:
1114
        for (i = 0; i < (FFMIN(src_size, max_size) + 1) >> 1; i++)
1115
            *sdst++ = bswap_16(*ssrc++);
1116
        return FFMIN(src_size, max_size);
1117
    case DCA_MARKER_14B_BE:
1118
    case DCA_MARKER_14B_LE:
1119
        init_put_bits(&pb, dst, max_size);
1120
        for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {
1121
            tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;
1122
            put_bits(&pb, 14, tmp);
1123
        }
1124
        flush_put_bits(&pb);
1125
        return (put_bits_count(&pb) + 7) >> 3;
1126
    default:
1127
        return -1;
1128
    }
1129
}
1130

    
1131
/**
1132
 * Main frame decoding function
1133
 * FIXME add arguments
1134
 */
1135
static int dca_decode_frame(AVCodecContext * avctx,
1136
                            void *data, int *data_size,
1137
                            const uint8_t * buf, int buf_size)
1138
{
1139

    
1140
    int i, j, k;
1141
    int16_t *samples = data;
1142
    DCAContext *s = avctx->priv_data;
1143
    int channels;
1144

    
1145

    
1146
    s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE);
1147
    if (s->dca_buffer_size == -1) {
1148
        av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
1149
        return -1;
1150
    }
1151

    
1152
    init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
1153
    if (dca_parse_frame_header(s) < 0) {
1154
        //seems like the frame is corrupt, try with the next one
1155
        *data_size=0;
1156
        return buf_size;
1157
    }
1158
    //set AVCodec values with parsed data
1159
    avctx->sample_rate = s->sample_rate;
1160
    avctx->bit_rate = s->bit_rate;
1161

    
1162
    channels = s->prim_channels + !!s->lfe;
1163
    if(avctx->request_channels == 2 && s->prim_channels > 2) {
1164
        channels = 2;
1165
        s->output = DCA_STEREO;
1166
    }
1167

    
1168
    avctx->channels = channels;
1169
    if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
1170
        return -1;
1171
    *data_size = 0;
1172
    for (i = 0; i < (s->sample_blocks / 8); i++) {
1173
        dca_decode_block(s);
1174
        s->dsp.float_to_int16(s->tsamples, s->samples, 256 * channels);
1175
        /* interleave samples */
1176
        for (j = 0; j < 256; j++) {
1177
            for (k = 0; k < channels; k++)
1178
                samples[k] = s->tsamples[j + k * 256];
1179
            samples += channels;
1180
        }
1181
        *data_size += 256 * sizeof(int16_t) * channels;
1182
    }
1183

    
1184
    return buf_size;
1185
}
1186

    
1187

    
1188

    
1189
/**
1190
 * Build the cosine modulation tables for the QMF
1191
 *
1192
 * @param s     pointer to the DCAContext
1193
 */
1194

    
1195
static void pre_calc_cosmod(DCAContext * s)
1196
{
1197
    int i, j, k;
1198
    static int cosmod_inited = 0;
1199

    
1200
    if(cosmod_inited) return;
1201
    for (j = 0, k = 0; k < 16; k++)
1202
        for (i = 0; i < 16; i++)
1203
            cos_mod[j++] = cos((2 * i + 1) * (2 * k + 1) * M_PI / 64);
1204

    
1205
    for (k = 0; k < 16; k++)
1206
        for (i = 0; i < 16; i++)
1207
            cos_mod[j++] = cos((i) * (2 * k + 1) * M_PI / 32);
1208

    
1209
    for (k = 0; k < 16; k++)
1210
        cos_mod[j++] = 0.25 / (2 * cos((2 * k + 1) * M_PI / 128));
1211

    
1212
    for (k = 0; k < 16; k++)
1213
        cos_mod[j++] = -0.25 / (2.0 * sin((2 * k + 1) * M_PI / 128));
1214

    
1215
    cosmod_inited = 1;
1216
}
1217

    
1218

    
1219
/**
1220
 * DCA initialization
1221
 *
1222
 * @param avctx     pointer to the AVCodecContext
1223
 */
1224

    
1225
static int dca_decode_init(AVCodecContext * avctx)
1226
{
1227
    DCAContext *s = avctx->priv_data;
1228

    
1229
    s->avctx = avctx;
1230
    dca_init_vlcs();
1231
    pre_calc_cosmod(s);
1232

    
1233
    dsputil_init(&s->dsp, avctx);
1234

    
1235
    /* allow downmixing to stereo */
1236
    if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
1237
            avctx->request_channels == 2) {
1238
        avctx->channels = avctx->request_channels;
1239
    }
1240

    
1241
    return 0;
1242
}
1243

    
1244

    
1245
AVCodec dca_decoder = {
1246
    .name = "dca",
1247
    .type = CODEC_TYPE_AUDIO,
1248
    .id = CODEC_ID_DTS,
1249
    .priv_data_size = sizeof(DCAContext),
1250
    .init = dca_decode_init,
1251
    .decode = dca_decode_frame,
1252
};