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 */ 58typedef 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 */ 72typedef 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 */ 87static const uint16_t samples_per_band[3] = {128, 128, 256}; 88static const uint8_t mdct_long_nbits[3] = {7, 7, 8}; 89 90 91static 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 106static 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 167static 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 190static 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 255static 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 272static 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 325static 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 358static 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 368AVCodec 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}; 378