1/* 2 * ATRAC1 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 * ATRAC1 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 "libavutil/float_dsp.h" 36#include "libavutil/mem_internal.h" 37 38#include "avcodec.h" 39#include "codec_internal.h" 40#include "get_bits.h" 41#include "fft.h" 42#include "internal.h" 43#include "sinewin.h" 44 45#include "atrac.h" 46#include "atrac1data.h" 47 48#define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit 49#define AT1_SU_SIZE 212 ///< number of bytes in a sound unit 50#define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit 51#define AT1_FRAME_SIZE AT1_SU_SIZE * 2 52#define AT1_SU_MAX_BITS AT1_SU_SIZE * 8 53#define AT1_MAX_CHANNELS 2 54 55#define AT1_QMF_BANDS 3 56#define IDX_LOW_BAND 0 57#define IDX_MID_BAND 1 58#define IDX_HIGH_BAND 2 59 60/** 61 * Sound unit struct, one unit is used per channel 62 */ 63typedef struct AT1SUCtx { 64 int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band 65 int num_bfus; ///< number of Block Floating Units 66 float* spectrum[2]; 67 DECLARE_ALIGNED(32, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer 68 DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer 69 DECLARE_ALIGNED(32, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter 70 DECLARE_ALIGNED(32, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter 71 DECLARE_ALIGNED(32, float, last_qmf_delay)[256+39]; ///< delay line for the last stacked QMF filter 72} AT1SUCtx; 73 74/** 75 * The atrac1 context, holds all needed parameters for decoding 76 */ 77typedef struct AT1Ctx { 78 AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit 79 DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer 80 81 DECLARE_ALIGNED(32, float, low)[256]; 82 DECLARE_ALIGNED(32, float, mid)[256]; 83 DECLARE_ALIGNED(32, float, high)[512]; 84 float* bands[3]; 85 FFTContext mdct_ctx[3]; 86 void (*vector_fmul_window)(float *dst, const float *src0, 87 const float *src1, const float *win, int len); 88} AT1Ctx; 89 90/** size of the transform in samples in the long mode for each QMF band */ 91static const uint16_t samples_per_band[3] = {128, 128, 256}; 92static const uint8_t mdct_long_nbits[3] = {7, 7, 8}; 93 94 95static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, 96 int rev_spec) 97{ 98 FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)]; 99 int transf_size = 1 << nbits; 100 101 if (rev_spec) { 102 int i; 103 for (i = 0; i < transf_size / 2; i++) 104 FFSWAP(float, spec[i], spec[transf_size - 1 - i]); 105 } 106 mdct_context->imdct_half(mdct_context, out, spec); 107} 108 109 110static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q) 111{ 112 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size; 113 unsigned int start_pos, ref_pos = 0, pos = 0; 114 115 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { 116 float *prev_buf; 117 int j; 118 119 band_samples = samples_per_band[band_num]; 120 log2_block_count = su->log2_block_count[band_num]; 121 122 /* number of mdct blocks in the current QMF band: 1 - for long mode */ 123 /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/ 124 num_blocks = 1 << log2_block_count; 125 126 if (num_blocks == 1) { 127 /* mdct block size in samples: 128 (long mode, low & mid bands), */ 128 /* 256 (long mode, high band) and 32 (short mode, all bands) */ 129 block_size = band_samples >> log2_block_count; 130 131 /* calc transform size in bits according to the block_size_mode */ 132 nbits = mdct_long_nbits[band_num] - log2_block_count; 133 134 if (nbits != 5 && nbits != 7 && nbits != 8) 135 return AVERROR_INVALIDDATA; 136 } else { 137 block_size = 32; 138 nbits = 5; 139 } 140 141 start_pos = 0; 142 prev_buf = &su->spectrum[1][ref_pos + band_samples - 16]; 143 for (j=0; j < num_blocks; j++) { 144 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num); 145 146 /* overlap and window */ 147 q->vector_fmul_window(&q->bands[band_num][start_pos], prev_buf, 148 &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 16); 149 150 prev_buf = &su->spectrum[0][ref_pos+start_pos + 16]; 151 start_pos += block_size; 152 pos += block_size; 153 } 154 155 if (num_blocks == 1) 156 memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float)); 157 158 ref_pos += band_samples; 159 } 160 161 /* Swap buffers so the mdct overlap works */ 162 FFSWAP(float*, su->spectrum[0], su->spectrum[1]); 163 164 return 0; 165} 166 167/** 168 * Parse the block size mode byte 169 */ 170 171static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS]) 172{ 173 int log2_block_count_tmp, i; 174 175 for (i = 0; i < 2; i++) { 176 /* low and mid band */ 177 log2_block_count_tmp = get_bits(gb, 2); 178 if (log2_block_count_tmp & 1) 179 return AVERROR_INVALIDDATA; 180 log2_block_cnt[i] = 2 - log2_block_count_tmp; 181 } 182 183 /* high band */ 184 log2_block_count_tmp = get_bits(gb, 2); 185 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3) 186 return AVERROR_INVALIDDATA; 187 log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; 188 189 skip_bits(gb, 2); 190 return 0; 191} 192 193 194static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, 195 float spec[AT1_SU_SAMPLES]) 196{ 197 int bits_used, band_num, bfu_num, i; 198 uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU 199 uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU 200 201 /* parse the info byte (2nd byte) telling how much BFUs were coded */ 202 su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)]; 203 204 /* calc number of consumed bits: 205 num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits) 206 + info_byte_copy(8bits) + log2_block_count_copy(8bits) */ 207 bits_used = su->num_bfus * 10 + 32 + 208 bfu_amount_tab2[get_bits(gb, 2)] + 209 (bfu_amount_tab3[get_bits(gb, 3)] << 1); 210 211 /* get word length index (idwl) for each BFU */ 212 for (i = 0; i < su->num_bfus; i++) 213 idwls[i] = get_bits(gb, 4); 214 215 /* get scalefactor index (idsf) for each BFU */ 216 for (i = 0; i < su->num_bfus; i++) 217 idsfs[i] = get_bits(gb, 6); 218 219 /* zero idwl/idsf for empty BFUs */ 220 for (i = su->num_bfus; i < AT1_MAX_BFU; i++) 221 idwls[i] = idsfs[i] = 0; 222 223 /* read in the spectral data and reconstruct MDCT spectrum of this channel */ 224 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { 225 for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) { 226 int pos; 227 228 int num_specs = specs_per_bfu[bfu_num]; 229 int word_len = !!idwls[bfu_num] + idwls[bfu_num]; 230 float scale_factor = ff_atrac_sf_table[idsfs[bfu_num]]; 231 bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */ 232 233 /* check for bitstream overflow */ 234 if (bits_used > AT1_SU_MAX_BITS) 235 return AVERROR_INVALIDDATA; 236 237 /* get the position of the 1st spec according to the block size mode */ 238 pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num]; 239 240 if (word_len) { 241 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1); 242 243 for (i = 0; i < num_specs; i++) { 244 /* read in a quantized spec and convert it to 245 * signed int and then inverse quantization 246 */ 247 spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant; 248 } 249 } else { /* word_len = 0 -> empty BFU, zero all specs in the empty BFU */ 250 memset(&spec[pos], 0, num_specs * sizeof(float)); 251 } 252 } 253 } 254 255 return 0; 256} 257 258 259static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) 260{ 261 float temp[256]; 262 float iqmf_temp[512 + 46]; 263 264 /* combine low and middle bands */ 265 ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp); 266 267 /* delay the signal of the high band by 39 samples */ 268 memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 39); 269 memcpy(&su->last_qmf_delay[39], q->bands[2], sizeof(float) * 256); 270 271 /* combine (low + middle) and high bands */ 272 ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp); 273} 274 275 276static int atrac1_decode_frame(AVCodecContext *avctx, AVFrame *frame, 277 int *got_frame_ptr, AVPacket *avpkt) 278{ 279 const uint8_t *buf = avpkt->data; 280 int buf_size = avpkt->size; 281 AT1Ctx *q = avctx->priv_data; 282 int channels = avctx->ch_layout.nb_channels; 283 int ch, ret; 284 GetBitContext gb; 285 286 287 if (buf_size < 212 * channels) { 288 av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n"); 289 return AVERROR_INVALIDDATA; 290 } 291 292 /* get output buffer */ 293 frame->nb_samples = AT1_SU_SAMPLES; 294 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) 295 return ret; 296 297 for (ch = 0; ch < channels; ch++) { 298 AT1SUCtx* su = &q->SUs[ch]; 299 300 init_get_bits(&gb, &buf[212 * ch], 212 * 8); 301 302 /* parse block_size_mode, 1st byte */ 303 ret = at1_parse_bsm(&gb, su->log2_block_count); 304 if (ret < 0) 305 return ret; 306 307 ret = at1_unpack_dequant(&gb, su, q->spec); 308 if (ret < 0) 309 return ret; 310 311 ret = at1_imdct_block(su, q); 312 if (ret < 0) 313 return ret; 314 at1_subband_synthesis(q, su, (float *)frame->extended_data[ch]); 315 } 316 317 *got_frame_ptr = 1; 318 319 return avctx->block_align; 320} 321 322 323static av_cold int atrac1_decode_end(AVCodecContext * avctx) 324{ 325 AT1Ctx *q = avctx->priv_data; 326 327 ff_mdct_end(&q->mdct_ctx[0]); 328 ff_mdct_end(&q->mdct_ctx[1]); 329 ff_mdct_end(&q->mdct_ctx[2]); 330 331 return 0; 332} 333 334 335static av_cold int atrac1_decode_init(AVCodecContext *avctx) 336{ 337 AT1Ctx *q = avctx->priv_data; 338 AVFloatDSPContext *fdsp; 339 int channels = avctx->ch_layout.nb_channels; 340 int ret; 341 342 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; 343 344 if (channels < 1 || channels > AT1_MAX_CHANNELS) { 345 av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", 346 channels); 347 return AVERROR(EINVAL); 348 } 349 350 if (avctx->block_align <= 0) { 351 av_log(avctx, AV_LOG_ERROR, "Unsupported block align."); 352 return AVERROR_PATCHWELCOME; 353 } 354 355 /* Init the mdct transforms */ 356 if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) || 357 (ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) || 358 (ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) { 359 av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n"); 360 return ret; 361 } 362 363 ff_init_ff_sine_windows(5); 364 365 ff_atrac_generate_tables(); 366 367 fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT); 368 if (!fdsp) 369 return AVERROR(ENOMEM); 370 q->vector_fmul_window = fdsp->vector_fmul_window; 371 av_free(fdsp); 372 373 q->bands[0] = q->low; 374 q->bands[1] = q->mid; 375 q->bands[2] = q->high; 376 377 /* Prepare the mdct overlap buffers */ 378 q->SUs[0].spectrum[0] = q->SUs[0].spec1; 379 q->SUs[0].spectrum[1] = q->SUs[0].spec2; 380 q->SUs[1].spectrum[0] = q->SUs[1].spec1; 381 q->SUs[1].spectrum[1] = q->SUs[1].spec2; 382 383 return 0; 384} 385 386 387const FFCodec ff_atrac1_decoder = { 388 .p.name = "atrac1", 389 .p.long_name = NULL_IF_CONFIG_SMALL("ATRAC1 (Adaptive TRansform Acoustic Coding)"), 390 .p.type = AVMEDIA_TYPE_AUDIO, 391 .p.id = AV_CODEC_ID_ATRAC1, 392 .priv_data_size = sizeof(AT1Ctx), 393 .init = atrac1_decode_init, 394 .close = atrac1_decode_end, 395 FF_CODEC_DECODE_CB(atrac1_decode_frame), 396 .p.capabilities = AV_CODEC_CAP_DR1, 397 .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, 398 AV_SAMPLE_FMT_NONE }, 399 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, 400}; 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