release/src/router/ffmpeg/libavcodec/atrac1.c

   1 /*
   2  * Atrac 1 compatible decoder
   3  * Copyright (c) 2009 Maxim Poliakovski
   4  * Copyright (c) 2009 Benjamin Larsson
   5  *
   6  * This file is part of FFmpeg.
   7  *
   8  * FFmpeg is free software; you can redistribute it and/or
   9  * modify it under the terms of the GNU Lesser General Public
  10  * License as published by the Free Software Foundation; either
  11  * version 2.1 of the License, or (at your option) any later version.
  12  *
  13  * FFmpeg is distributed in the hope that it will be useful,
  14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  16  * Lesser General Public License for more details.
  17  *
  18  * You should have received a copy of the GNU Lesser General Public
  19  * License along with FFmpeg; if not, write to the Free Software
  20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  21  */
  22
  23 /**
  24  * @file
  25  * Atrac 1 compatible decoder.
  26  * This decoder handles raw ATRAC1 data and probably SDDS data.
  27  */
  28
  29 /* Many thanks to Tim Craig for all the help! */
  30
  31 #include <math.h>
  32 #include <stddef.h>
  33 #include <stdio.h>
  34
  35 #include "avcodec.h"
  36 #include "get_bits.h"
  37 #include "dsputil.h"
  38 #include "fft.h"
  39
  40 #include "atrac.h"
  41 #include "atrac1data.h"
  42
  43 #define AT1_MAX_BFU      52                 ///< max number of block floating units in a sound unit
  44 #define AT1_SU_SIZE      212                ///< number of bytes in a sound unit
  45 #define AT1_SU_SAMPLES   512                ///< number of samples in a sound unit
  46 #define AT1_FRAME_SIZE   AT1_SU_SIZE * 2
  47 #define AT1_SU_MAX_BITS  AT1_SU_SIZE * 8
  48 #define AT1_MAX_CHANNELS 2
  49
  50 #define AT1_QMF_BANDS    3
  51 #define IDX_LOW_BAND     0
  52 #define IDX_MID_BAND     1
  53 #define IDX_HIGH_BAND    2
  54
  55 /**
  56  * Sound unit struct, one unit is used per channel
  57  */
  58 typedef struct {
  59     int                 log2_block_count[AT1_QMF_BANDS];    ///< log2 number of blocks in a band
  60     int                 num_bfus;                           ///< number of Block Floating Units
  61     float*              spectrum[2];
  62     DECLARE_ALIGNED(16, float, spec1)[AT1_SU_SAMPLES];     ///< mdct buffer
  63     DECLARE_ALIGNED(16, float, spec2)[AT1_SU_SAMPLES];     ///< mdct buffer
  64     DECLARE_ALIGNED(16, float, fst_qmf_delay)[46];         ///< delay line for the 1st stacked QMF filter
  65     DECLARE_ALIGNED(16, float, snd_qmf_delay)[46];         ///< delay line for the 2nd stacked QMF filter
  66     DECLARE_ALIGNED(16, float, last_qmf_delay)[256+23];    ///< delay line for the last stacked QMF filter
  67 } AT1SUCtx;
  68
  69 /**
  70  * The atrac1 context, holds all needed parameters for decoding
  71  */
  72 typedef struct {
  73     AT1SUCtx            SUs[AT1_MAX_CHANNELS];              ///< channel sound unit
  74     DECLARE_ALIGNED(16, float, spec)[AT1_SU_SAMPLES];      ///< the mdct spectrum buffer
  75
  76     DECLARE_ALIGNED(16, float,  low)[256];
  77     DECLARE_ALIGNED(16, float,  mid)[256];
  78     DECLARE_ALIGNED(16, float, high)[512];
  79     float*              bands[3];
  80     DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
  81     FFTContext          mdct_ctx[3];
  82     int                 channels;
  83     DSPContext          dsp;
  84 } AT1Ctx;
  85
  86 /** size of the transform in samples in the long mode for each QMF band */
  87 static const uint16_t samples_per_band[3] = {128, 128, 256};
  88 static const uint8_t   mdct_long_nbits[3] = {7, 7, 8};
  89
  90
  91 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
  92                       int rev_spec)
  93 {
  94     FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
  95     int transf_size = 1 << nbits;
  96
  97     if (rev_spec) {
  98         int i;
  99         for (i = 0; i < transf_size / 2; i++)
 100             FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
 101     }
 102     ff_imdct_half(mdct_context, out, spec);
 103 }
 104
 105
 106 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
 107 {
 108     int          band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
 109     unsigned int start_pos, ref_pos = 0, pos = 0;
 110
 111     for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
 112         float *prev_buf;
 113         int j;
 114
 115         band_samples = samples_per_band[band_num];
 116         log2_block_count = su->log2_block_count[band_num];
 117
 118         /* number of mdct blocks in the current QMF band: 1 - for long mode */
 119         /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
 120         num_blocks = 1 << log2_block_count;
 121
 122         if (num_blocks == 1) {
 123             /* mdct block size in samples: 128 (long mode, low & mid bands), */
 124             /* 256 (long mode, high band) and 32 (short mode, all bands) */
 125             block_size = band_samples >> log2_block_count;
 126
 127             /* calc transform size in bits according to the block_size_mode */
 128             nbits = mdct_long_nbits[band_num] - log2_block_count;
 129
 130             if (nbits != 5 && nbits != 7 && nbits != 8)
 131                 return -1;
 132         } else {
 133             block_size = 32;
 134             nbits = 5;
 135         }
 136
 137         start_pos = 0;
 138         prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
 139         for (j=0; j < num_blocks; j++) {
 140             at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
 141
 142             /* overlap and window */
 143             q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
 144                                       &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 0, 16);
 145
 146             prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
 147             start_pos += block_size;
 148             pos += block_size;
 149         }
 150
 151         if (num_blocks == 1)
 152             memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
 153
 154         ref_pos += band_samples;
 155     }
 156
 157     /* Swap buffers so the mdct overlap works */
 158     FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
 159
 160     return 0;
 161 }
 162
 163 /**
 164  * Parse the block size mode byte
 165  */
 166
 167 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
 168 {
 169     int log2_block_count_tmp, i;
 170
 171     for (i = 0; i < 2; i++) {
 172         /* low and mid band */
 173         log2_block_count_tmp = get_bits(gb, 2);
 174         if (log2_block_count_tmp & 1)
 175             return -1;
 176         log2_block_cnt[i] = 2 - log2_block_count_tmp;
 177     }
 178
 179     /* high band */
 180     log2_block_count_tmp = get_bits(gb, 2);
 181     if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
 182         return -1;
 183     log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
 184
 185     skip_bits(gb, 2);
 186     return 0;
 187 }
 188
 189
 190 static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
 191                               float spec[AT1_SU_SAMPLES])
 192 {
 193     int bits_used, band_num, bfu_num, i;
 194     uint8_t idwls[AT1_MAX_BFU];                 ///< the word length indexes for each BFU
 195     uint8_t idsfs[AT1_MAX_BFU];                 ///< the scalefactor indexes for each BFU
 196
 197     /* parse the info byte (2nd byte) telling how much BFUs were coded */
 198     su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
 199
 200     /* calc number of consumed bits:
 201         num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
 202         + info_byte_copy(8bits) + log2_block_count_copy(8bits) */
 203     bits_used = su->num_bfus * 10 + 32 +
 204                 bfu_amount_tab2[get_bits(gb, 2)] +
 205                 (bfu_amount_tab3[get_bits(gb, 3)] << 1);
 206
 207     /* get word length index (idwl) for each BFU */
 208     for (i = 0; i < su->num_bfus; i++)
 209         idwls[i] = get_bits(gb, 4);
 210
 211     /* get scalefactor index (idsf) for each BFU */
 212     for (i = 0; i < su->num_bfus; i++)
 213         idsfs[i] = get_bits(gb, 6);
 214
 215     /* zero idwl/idsf for empty BFUs */
 216     for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
 217         idwls[i] = idsfs[i] = 0;
 218
 219     /* read in the spectral data and reconstruct MDCT spectrum of this channel */
 220     for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
 221         for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {
 222             int pos;
 223
 224             int num_specs = specs_per_bfu[bfu_num];
 225             int word_len  = !!idwls[bfu_num] + idwls[bfu_num];
 226             float scale_factor = sf_table[idsfs[bfu_num]];
 227             bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
 228
 229             /* check for bitstream overflow */
 230             if (bits_used > AT1_SU_MAX_BITS)
 231                 return -1;
 232
 233             /* get the position of the 1st spec according to the block size mode */
 234             pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
 235
 236             if (word_len) {
 237                 float   max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
 238
 239                 for (i = 0; i < num_specs; i++) {
 240                     /* read in a quantized spec and convert it to
 241                      * signed int and then inverse quantization
 242                      */
 243                     spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
 244                 }
 245             } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
 246                 memset(&spec[pos], 0, num_specs * sizeof(float));
 247             }
 248         }
 249     }
 250
 251     return 0;
 252 }
 253
 254
 255 static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
 256 {
 257     float temp[256];
 258     float iqmf_temp[512 + 46];
 259
 260     /* combine low and middle bands */
 261     atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
 262
 263     /* delay the signal of the high band by 23 samples */
 264     memcpy( su->last_qmf_delay,    &su->last_qmf_delay[256], sizeof(float) *  23);
 265     memcpy(&su->last_qmf_delay[23], q->bands[2],             sizeof(float) * 256);
 266
 267     /* combine (low + middle) and high bands */
 268     atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
 269 }
 270
 271
 272 static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
 273                                int *data_size, AVPacket *avpkt)
 274 {
 275     const uint8_t *buf = avpkt->data;
 276     int buf_size       = avpkt->size;
 277     AT1Ctx *q          = avctx->priv_data;
 278     int ch, ret, i;
 279     GetBitContext gb;
 280     float* samples = data;
 281
 282
 283     if (buf_size < 212 * q->channels) {
 284         av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
 285         return -1;
 286     }
 287
 288     for (ch = 0; ch < q->channels; ch++) {
 289         AT1SUCtx* su = &q->SUs[ch];
 290
 291         init_get_bits(&gb, &buf[212 * ch], 212 * 8);
 292
 293         /* parse block_size_mode, 1st byte */
 294         ret = at1_parse_bsm(&gb, su->log2_block_count);
 295         if (ret < 0)
 296             return ret;
 297
 298         ret = at1_unpack_dequant(&gb, su, q->spec);
 299         if (ret < 0)
 300             return ret;
 301
 302         ret = at1_imdct_block(su, q);
 303         if (ret < 0)
 304             return ret;
 305         at1_subband_synthesis(q, su, q->out_samples[ch]);
 306     }
 307
 308     /* interleave; FIXME, should create/use a DSP function */
 309     if (q->channels == 1) {
 310         /* mono */
 311         memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4);
 312     } else {
 313         /* stereo */
 314         for (i = 0; i < AT1_SU_SAMPLES; i++) {
 315             samples[i * 2]     = q->out_samples[0][i];
 316             samples[i * 2 + 1] = q->out_samples[1][i];
 317         }
 318     }
 319
 320     *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
 321     return avctx->block_align;
 322 }
 323
 324
 325 static av_cold int atrac1_decode_init(AVCodecContext *avctx)
 326 {
 327     AT1Ctx *q = avctx->priv_data;
 328
 329     avctx->sample_fmt = SAMPLE_FMT_FLT;
 330
 331     q->channels = avctx->channels;
 332
 333     /* Init the mdct transforms */
 334     ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
 335     ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
 336     ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
 337
 338     ff_init_ff_sine_windows(5);
 339
 340     atrac_generate_tables();
 341
 342     dsputil_init(&q->dsp, avctx);
 343
 344     q->bands[0] = q->low;
 345     q->bands[1] = q->mid;
 346     q->bands[2] = q->high;
 347
 348     /* Prepare the mdct overlap buffers */
 349     q->SUs[0].spectrum[0] = q->SUs[0].spec1;
 350     q->SUs[0].spectrum[1] = q->SUs[0].spec2;
 351     q->SUs[1].spectrum[0] = q->SUs[1].spec1;
 352     q->SUs[1].spectrum[1] = q->SUs[1].spec2;
 353
 354     return 0;
 355 }
 356
 357
 358 static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
 359     AT1Ctx *q = avctx->priv_data;
 360
 361     ff_mdct_end(&q->mdct_ctx[0]);
 362     ff_mdct_end(&q->mdct_ctx[1]);
 363     ff_mdct_end(&q->mdct_ctx[2]);
 364     return 0;
 365 }
 366
 367
 368 AVCodec atrac1_decoder = {
 369     .name = "atrac1",
 370     .type = AVMEDIA_TYPE_AUDIO,
 371     .id = CODEC_ID_ATRAC1,
 372     .priv_data_size = sizeof(AT1Ctx),
 373     .init = atrac1_decode_init,
 374     .close = atrac1_decode_end,
 375     .decode = atrac1_decode_frame,
 376     .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
 377 };