| 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 "avcodec.h" |
| 37 | #include "get_bits.h" |
| 38 | #include "fft.h" |
| 39 | #include "internal.h" |
| 40 | #include "sinewin.h" |
| 41 | |
| 42 | #include "atrac.h" |
| 43 | #include "atrac1data.h" |
| 44 | |
| 45 | #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit |
| 46 | #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit |
| 47 | #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit |
| 48 | #define AT1_FRAME_SIZE AT1_SU_SIZE * 2 |
| 49 | #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8 |
| 50 | #define AT1_MAX_CHANNELS 2 |
| 51 | |
| 52 | #define AT1_QMF_BANDS 3 |
| 53 | #define IDX_LOW_BAND 0 |
| 54 | #define IDX_MID_BAND 1 |
| 55 | #define IDX_HIGH_BAND 2 |
| 56 | |
| 57 | /** |
| 58 | * Sound unit struct, one unit is used per channel |
| 59 | */ |
| 60 | typedef struct { |
| 61 | int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band |
| 62 | int num_bfus; ///< number of Block Floating Units |
| 63 | float* spectrum[2]; |
| 64 | DECLARE_ALIGNED(32, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer |
| 65 | DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer |
| 66 | DECLARE_ALIGNED(32, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter |
| 67 | DECLARE_ALIGNED(32, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter |
| 68 | DECLARE_ALIGNED(32, float, last_qmf_delay)[256+23]; ///< delay line for the last stacked QMF filter |
| 69 | } AT1SUCtx; |
| 70 | |
| 71 | /** |
| 72 | * The atrac1 context, holds all needed parameters for decoding |
| 73 | */ |
| 74 | typedef struct { |
| 75 | AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit |
| 76 | DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer |
| 77 | |
| 78 | DECLARE_ALIGNED(32, float, low)[256]; |
| 79 | DECLARE_ALIGNED(32, float, mid)[256]; |
| 80 | DECLARE_ALIGNED(32, float, high)[512]; |
| 81 | float* bands[3]; |
| 82 | FFTContext mdct_ctx[3]; |
| 83 | AVFloatDSPContext *fdsp; |
| 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 | mdct_context->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 AVERROR_INVALIDDATA; |
| 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->fdsp->vector_fmul_window(&q->bands[band_num][start_pos], prev_buf, |
| 144 | &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 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 AVERROR_INVALIDDATA; |
| 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 AVERROR_INVALIDDATA; |
| 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 = ff_atrac_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 AVERROR_INVALIDDATA; |
| 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 empty 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 | ff_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 | ff_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 *got_frame_ptr, AVPacket *avpkt) |
| 274 | { |
| 275 | AVFrame *frame = data; |
| 276 | const uint8_t *buf = avpkt->data; |
| 277 | int buf_size = avpkt->size; |
| 278 | AT1Ctx *q = avctx->priv_data; |
| 279 | int ch, ret; |
| 280 | GetBitContext gb; |
| 281 | |
| 282 | |
| 283 | if (buf_size < 212 * avctx->channels) { |
| 284 | av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n"); |
| 285 | return AVERROR_INVALIDDATA; |
| 286 | } |
| 287 | |
| 288 | /* get output buffer */ |
| 289 | frame->nb_samples = AT1_SU_SAMPLES; |
| 290 | if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
| 291 | return ret; |
| 292 | |
| 293 | for (ch = 0; ch < avctx->channels; ch++) { |
| 294 | AT1SUCtx* su = &q->SUs[ch]; |
| 295 | |
| 296 | init_get_bits(&gb, &buf[212 * ch], 212 * 8); |
| 297 | |
| 298 | /* parse block_size_mode, 1st byte */ |
| 299 | ret = at1_parse_bsm(&gb, su->log2_block_count); |
| 300 | if (ret < 0) |
| 301 | return ret; |
| 302 | |
| 303 | ret = at1_unpack_dequant(&gb, su, q->spec); |
| 304 | if (ret < 0) |
| 305 | return ret; |
| 306 | |
| 307 | ret = at1_imdct_block(su, q); |
| 308 | if (ret < 0) |
| 309 | return ret; |
| 310 | at1_subband_synthesis(q, su, (float *)frame->extended_data[ch]); |
| 311 | } |
| 312 | |
| 313 | *got_frame_ptr = 1; |
| 314 | |
| 315 | return avctx->block_align; |
| 316 | } |
| 317 | |
| 318 | |
| 319 | static av_cold int atrac1_decode_end(AVCodecContext * avctx) |
| 320 | { |
| 321 | AT1Ctx *q = avctx->priv_data; |
| 322 | |
| 323 | ff_mdct_end(&q->mdct_ctx[0]); |
| 324 | ff_mdct_end(&q->mdct_ctx[1]); |
| 325 | ff_mdct_end(&q->mdct_ctx[2]); |
| 326 | |
| 327 | av_freep(&q->fdsp); |
| 328 | |
| 329 | return 0; |
| 330 | } |
| 331 | |
| 332 | |
| 333 | static av_cold int atrac1_decode_init(AVCodecContext *avctx) |
| 334 | { |
| 335 | AT1Ctx *q = avctx->priv_data; |
| 336 | int ret; |
| 337 | |
| 338 | avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; |
| 339 | |
| 340 | if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) { |
| 341 | av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", |
| 342 | avctx->channels); |
| 343 | return AVERROR(EINVAL); |
| 344 | } |
| 345 | |
| 346 | if (avctx->block_align <= 0) { |
| 347 | av_log(avctx, AV_LOG_ERROR, "Unsupported block align."); |
| 348 | return AVERROR_PATCHWELCOME; |
| 349 | } |
| 350 | |
| 351 | /* Init the mdct transforms */ |
| 352 | if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) || |
| 353 | (ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) || |
| 354 | (ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) { |
| 355 | av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n"); |
| 356 | atrac1_decode_end(avctx); |
| 357 | return ret; |
| 358 | } |
| 359 | |
| 360 | ff_init_ff_sine_windows(5); |
| 361 | |
| 362 | ff_atrac_generate_tables(); |
| 363 | |
| 364 | q->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT); |
| 365 | |
| 366 | q->bands[0] = q->low; |
| 367 | q->bands[1] = q->mid; |
| 368 | q->bands[2] = q->high; |
| 369 | |
| 370 | /* Prepare the mdct overlap buffers */ |
| 371 | q->SUs[0].spectrum[0] = q->SUs[0].spec1; |
| 372 | q->SUs[0].spectrum[1] = q->SUs[0].spec2; |
| 373 | q->SUs[1].spectrum[0] = q->SUs[1].spec1; |
| 374 | q->SUs[1].spectrum[1] = q->SUs[1].spec2; |
| 375 | |
| 376 | return 0; |
| 377 | } |
| 378 | |
| 379 | |
| 380 | AVCodec ff_atrac1_decoder = { |
| 381 | .name = "atrac1", |
| 382 | .long_name = NULL_IF_CONFIG_SMALL("ATRAC1 (Adaptive TRansform Acoustic Coding)"), |
| 383 | .type = AVMEDIA_TYPE_AUDIO, |
| 384 | .id = AV_CODEC_ID_ATRAC1, |
| 385 | .priv_data_size = sizeof(AT1Ctx), |
| 386 | .init = atrac1_decode_init, |
| 387 | .close = atrac1_decode_end, |
| 388 | .decode = atrac1_decode_frame, |
| 389 | .capabilities = CODEC_CAP_DR1, |
| 390 | .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, |
| 391 | AV_SAMPLE_FMT_NONE }, |
| 392 | }; |