| 1 | /* |
| 2 | * QDM2 compatible decoder |
| 3 | * Copyright (c) 2003 Ewald Snel |
| 4 | * Copyright (c) 2005 Benjamin Larsson |
| 5 | * Copyright (c) 2005 Alex Beregszaszi |
| 6 | * Copyright (c) 2005 Roberto Togni |
| 7 | * |
| 8 | * This file is part of FFmpeg. |
| 9 | * |
| 10 | * FFmpeg is free software; you can redistribute it and/or |
| 11 | * modify it under the terms of the GNU Lesser General Public |
| 12 | * License as published by the Free Software Foundation; either |
| 13 | * version 2.1 of the License, or (at your option) any later version. |
| 14 | * |
| 15 | * FFmpeg is distributed in the hope that it will be useful, |
| 16 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| 17 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| 18 | * Lesser General Public License for more details. |
| 19 | * |
| 20 | * You should have received a copy of the GNU Lesser General Public |
| 21 | * License along with FFmpeg; if not, write to the Free Software |
| 22 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| 23 | */ |
| 24 | |
| 25 | /** |
| 26 | * @file |
| 27 | * QDM2 decoder |
| 28 | * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni |
| 29 | * |
| 30 | * The decoder is not perfect yet, there are still some distortions |
| 31 | * especially on files encoded with 16 or 8 subbands. |
| 32 | */ |
| 33 | |
| 34 | #include <math.h> |
| 35 | #include <stddef.h> |
| 36 | #include <stdio.h> |
| 37 | |
| 38 | #define BITSTREAM_READER_LE |
| 39 | #include "libavutil/channel_layout.h" |
| 40 | #include "avcodec.h" |
| 41 | #include "get_bits.h" |
| 42 | #include "internal.h" |
| 43 | #include "rdft.h" |
| 44 | #include "mpegaudiodsp.h" |
| 45 | #include "mpegaudio.h" |
| 46 | |
| 47 | #include "qdm2data.h" |
| 48 | #include "qdm2_tablegen.h" |
| 49 | |
| 50 | #undef NDEBUG |
| 51 | #include <assert.h> |
| 52 | |
| 53 | |
| 54 | #define QDM2_LIST_ADD(list, size, packet) \ |
| 55 | do { \ |
| 56 | if (size > 0) { \ |
| 57 | list[size - 1].next = &list[size]; \ |
| 58 | } \ |
| 59 | list[size].packet = packet; \ |
| 60 | list[size].next = NULL; \ |
| 61 | size++; \ |
| 62 | } while(0) |
| 63 | |
| 64 | // Result is 8, 16 or 30 |
| 65 | #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling)) |
| 66 | |
| 67 | #define FIX_NOISE_IDX(noise_idx) \ |
| 68 | if ((noise_idx) >= 3840) \ |
| 69 | (noise_idx) -= 3840; \ |
| 70 | |
| 71 | #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)]) |
| 72 | |
| 73 | #define SAMPLES_NEEDED \ |
| 74 | av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); |
| 75 | |
| 76 | #define SAMPLES_NEEDED_2(why) \ |
| 77 | av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); |
| 78 | |
| 79 | #define QDM2_MAX_FRAME_SIZE 512 |
| 80 | |
| 81 | typedef int8_t sb_int8_array[2][30][64]; |
| 82 | |
| 83 | /** |
| 84 | * Subpacket |
| 85 | */ |
| 86 | typedef struct { |
| 87 | int type; ///< subpacket type |
| 88 | unsigned int size; ///< subpacket size |
| 89 | const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) |
| 90 | } QDM2SubPacket; |
| 91 | |
| 92 | /** |
| 93 | * A node in the subpacket list |
| 94 | */ |
| 95 | typedef struct QDM2SubPNode { |
| 96 | QDM2SubPacket *packet; ///< packet |
| 97 | struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node |
| 98 | } QDM2SubPNode; |
| 99 | |
| 100 | typedef struct { |
| 101 | float re; |
| 102 | float im; |
| 103 | } QDM2Complex; |
| 104 | |
| 105 | typedef struct { |
| 106 | float level; |
| 107 | QDM2Complex *complex; |
| 108 | const float *table; |
| 109 | int phase; |
| 110 | int phase_shift; |
| 111 | int duration; |
| 112 | short time_index; |
| 113 | short cutoff; |
| 114 | } FFTTone; |
| 115 | |
| 116 | typedef struct { |
| 117 | int16_t sub_packet; |
| 118 | uint8_t channel; |
| 119 | int16_t offset; |
| 120 | int16_t exp; |
| 121 | uint8_t phase; |
| 122 | } FFTCoefficient; |
| 123 | |
| 124 | typedef struct { |
| 125 | DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; |
| 126 | } QDM2FFT; |
| 127 | |
| 128 | /** |
| 129 | * QDM2 decoder context |
| 130 | */ |
| 131 | typedef struct { |
| 132 | /// Parameters from codec header, do not change during playback |
| 133 | int nb_channels; ///< number of channels |
| 134 | int channels; ///< number of channels |
| 135 | int group_size; ///< size of frame group (16 frames per group) |
| 136 | int fft_size; ///< size of FFT, in complex numbers |
| 137 | int checksum_size; ///< size of data block, used also for checksum |
| 138 | |
| 139 | /// Parameters built from header parameters, do not change during playback |
| 140 | int group_order; ///< order of frame group |
| 141 | int fft_order; ///< order of FFT (actually fftorder+1) |
| 142 | int frame_size; ///< size of data frame |
| 143 | int frequency_range; |
| 144 | int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ |
| 145 | int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 |
| 146 | int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) |
| 147 | |
| 148 | /// Packets and packet lists |
| 149 | QDM2SubPacket sub_packets[16]; ///< the packets themselves |
| 150 | QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets |
| 151 | QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list |
| 152 | int sub_packets_B; ///< number of packets on 'B' list |
| 153 | QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? |
| 154 | QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets |
| 155 | |
| 156 | /// FFT and tones |
| 157 | FFTTone fft_tones[1000]; |
| 158 | int fft_tone_start; |
| 159 | int fft_tone_end; |
| 160 | FFTCoefficient fft_coefs[1000]; |
| 161 | int fft_coefs_index; |
| 162 | int fft_coefs_min_index[5]; |
| 163 | int fft_coefs_max_index[5]; |
| 164 | int fft_level_exp[6]; |
| 165 | RDFTContext rdft_ctx; |
| 166 | QDM2FFT fft; |
| 167 | |
| 168 | /// I/O data |
| 169 | const uint8_t *compressed_data; |
| 170 | int compressed_size; |
| 171 | float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2]; |
| 172 | |
| 173 | /// Synthesis filter |
| 174 | MPADSPContext mpadsp; |
| 175 | DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2]; |
| 176 | int synth_buf_offset[MPA_MAX_CHANNELS]; |
| 177 | DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; |
| 178 | DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; |
| 179 | |
| 180 | /// Mixed temporary data used in decoding |
| 181 | float tone_level[MPA_MAX_CHANNELS][30][64]; |
| 182 | int8_t coding_method[MPA_MAX_CHANNELS][30][64]; |
| 183 | int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; |
| 184 | int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; |
| 185 | int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; |
| 186 | int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; |
| 187 | int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; |
| 188 | int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; |
| 189 | int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; |
| 190 | |
| 191 | // Flags |
| 192 | int has_errors; ///< packet has errors |
| 193 | int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type |
| 194 | int do_synth_filter; ///< used to perform or skip synthesis filter |
| 195 | |
| 196 | int sub_packet; |
| 197 | int noise_idx; ///< index for dithering noise table |
| 198 | } QDM2Context; |
| 199 | |
| 200 | |
| 201 | static VLC vlc_tab_level; |
| 202 | static VLC vlc_tab_diff; |
| 203 | static VLC vlc_tab_run; |
| 204 | static VLC fft_level_exp_alt_vlc; |
| 205 | static VLC fft_level_exp_vlc; |
| 206 | static VLC fft_stereo_exp_vlc; |
| 207 | static VLC fft_stereo_phase_vlc; |
| 208 | static VLC vlc_tab_tone_level_idx_hi1; |
| 209 | static VLC vlc_tab_tone_level_idx_mid; |
| 210 | static VLC vlc_tab_tone_level_idx_hi2; |
| 211 | static VLC vlc_tab_type30; |
| 212 | static VLC vlc_tab_type34; |
| 213 | static VLC vlc_tab_fft_tone_offset[5]; |
| 214 | |
| 215 | static const uint16_t qdm2_vlc_offs[] = { |
| 216 | 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, |
| 217 | }; |
| 218 | |
| 219 | static const int switchtable[23] = { |
| 220 | 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4 |
| 221 | }; |
| 222 | |
| 223 | static av_cold void qdm2_init_vlc(void) |
| 224 | { |
| 225 | static VLC_TYPE qdm2_table[3838][2]; |
| 226 | |
| 227 | vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; |
| 228 | vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; |
| 229 | init_vlc(&vlc_tab_level, 8, 24, |
| 230 | vlc_tab_level_huffbits, 1, 1, |
| 231 | vlc_tab_level_huffcodes, 2, 2, |
| 232 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 233 | |
| 234 | vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; |
| 235 | vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; |
| 236 | init_vlc(&vlc_tab_diff, 8, 37, |
| 237 | vlc_tab_diff_huffbits, 1, 1, |
| 238 | vlc_tab_diff_huffcodes, 2, 2, |
| 239 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 240 | |
| 241 | vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; |
| 242 | vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; |
| 243 | init_vlc(&vlc_tab_run, 5, 6, |
| 244 | vlc_tab_run_huffbits, 1, 1, |
| 245 | vlc_tab_run_huffcodes, 1, 1, |
| 246 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 247 | |
| 248 | fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; |
| 249 | fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - |
| 250 | qdm2_vlc_offs[3]; |
| 251 | init_vlc(&fft_level_exp_alt_vlc, 8, 28, |
| 252 | fft_level_exp_alt_huffbits, 1, 1, |
| 253 | fft_level_exp_alt_huffcodes, 2, 2, |
| 254 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 255 | |
| 256 | fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; |
| 257 | fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; |
| 258 | init_vlc(&fft_level_exp_vlc, 8, 20, |
| 259 | fft_level_exp_huffbits, 1, 1, |
| 260 | fft_level_exp_huffcodes, 2, 2, |
| 261 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 262 | |
| 263 | fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; |
| 264 | fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - |
| 265 | qdm2_vlc_offs[5]; |
| 266 | init_vlc(&fft_stereo_exp_vlc, 6, 7, |
| 267 | fft_stereo_exp_huffbits, 1, 1, |
| 268 | fft_stereo_exp_huffcodes, 1, 1, |
| 269 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 270 | |
| 271 | fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; |
| 272 | fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - |
| 273 | qdm2_vlc_offs[6]; |
| 274 | init_vlc(&fft_stereo_phase_vlc, 6, 9, |
| 275 | fft_stereo_phase_huffbits, 1, 1, |
| 276 | fft_stereo_phase_huffcodes, 1, 1, |
| 277 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 278 | |
| 279 | vlc_tab_tone_level_idx_hi1.table = |
| 280 | &qdm2_table[qdm2_vlc_offs[7]]; |
| 281 | vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - |
| 282 | qdm2_vlc_offs[7]; |
| 283 | init_vlc(&vlc_tab_tone_level_idx_hi1, 8, 20, |
| 284 | vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, |
| 285 | vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, |
| 286 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 287 | |
| 288 | vlc_tab_tone_level_idx_mid.table = |
| 289 | &qdm2_table[qdm2_vlc_offs[8]]; |
| 290 | vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - |
| 291 | qdm2_vlc_offs[8]; |
| 292 | init_vlc(&vlc_tab_tone_level_idx_mid, 8, 24, |
| 293 | vlc_tab_tone_level_idx_mid_huffbits, 1, 1, |
| 294 | vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, |
| 295 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 296 | |
| 297 | vlc_tab_tone_level_idx_hi2.table = |
| 298 | &qdm2_table[qdm2_vlc_offs[9]]; |
| 299 | vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - |
| 300 | qdm2_vlc_offs[9]; |
| 301 | init_vlc(&vlc_tab_tone_level_idx_hi2, 8, 24, |
| 302 | vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, |
| 303 | vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, |
| 304 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 305 | |
| 306 | vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; |
| 307 | vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; |
| 308 | init_vlc(&vlc_tab_type30, 6, 9, |
| 309 | vlc_tab_type30_huffbits, 1, 1, |
| 310 | vlc_tab_type30_huffcodes, 1, 1, |
| 311 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 312 | |
| 313 | vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; |
| 314 | vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; |
| 315 | init_vlc(&vlc_tab_type34, 5, 10, |
| 316 | vlc_tab_type34_huffbits, 1, 1, |
| 317 | vlc_tab_type34_huffcodes, 1, 1, |
| 318 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 319 | |
| 320 | vlc_tab_fft_tone_offset[0].table = |
| 321 | &qdm2_table[qdm2_vlc_offs[12]]; |
| 322 | vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - |
| 323 | qdm2_vlc_offs[12]; |
| 324 | init_vlc(&vlc_tab_fft_tone_offset[0], 8, 23, |
| 325 | vlc_tab_fft_tone_offset_0_huffbits, 1, 1, |
| 326 | vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, |
| 327 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 328 | |
| 329 | vlc_tab_fft_tone_offset[1].table = |
| 330 | &qdm2_table[qdm2_vlc_offs[13]]; |
| 331 | vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - |
| 332 | qdm2_vlc_offs[13]; |
| 333 | init_vlc(&vlc_tab_fft_tone_offset[1], 8, 28, |
| 334 | vlc_tab_fft_tone_offset_1_huffbits, 1, 1, |
| 335 | vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, |
| 336 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 337 | |
| 338 | vlc_tab_fft_tone_offset[2].table = |
| 339 | &qdm2_table[qdm2_vlc_offs[14]]; |
| 340 | vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - |
| 341 | qdm2_vlc_offs[14]; |
| 342 | init_vlc(&vlc_tab_fft_tone_offset[2], 8, 32, |
| 343 | vlc_tab_fft_tone_offset_2_huffbits, 1, 1, |
| 344 | vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, |
| 345 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 346 | |
| 347 | vlc_tab_fft_tone_offset[3].table = |
| 348 | &qdm2_table[qdm2_vlc_offs[15]]; |
| 349 | vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - |
| 350 | qdm2_vlc_offs[15]; |
| 351 | init_vlc(&vlc_tab_fft_tone_offset[3], 8, 35, |
| 352 | vlc_tab_fft_tone_offset_3_huffbits, 1, 1, |
| 353 | vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, |
| 354 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 355 | |
| 356 | vlc_tab_fft_tone_offset[4].table = |
| 357 | &qdm2_table[qdm2_vlc_offs[16]]; |
| 358 | vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - |
| 359 | qdm2_vlc_offs[16]; |
| 360 | init_vlc(&vlc_tab_fft_tone_offset[4], 8, 38, |
| 361 | vlc_tab_fft_tone_offset_4_huffbits, 1, 1, |
| 362 | vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, |
| 363 | INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); |
| 364 | } |
| 365 | |
| 366 | static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth) |
| 367 | { |
| 368 | int value; |
| 369 | |
| 370 | value = get_vlc2(gb, vlc->table, vlc->bits, depth); |
| 371 | |
| 372 | /* stage-2, 3 bits exponent escape sequence */ |
| 373 | if (value-- == 0) |
| 374 | value = get_bits(gb, get_bits(gb, 3) + 1); |
| 375 | |
| 376 | /* stage-3, optional */ |
| 377 | if (flag) { |
| 378 | int tmp; |
| 379 | |
| 380 | if (value >= 60) { |
| 381 | av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value); |
| 382 | return 0; |
| 383 | } |
| 384 | |
| 385 | tmp= vlc_stage3_values[value]; |
| 386 | |
| 387 | if ((value & ~3) > 0) |
| 388 | tmp += get_bits(gb, (value >> 2)); |
| 389 | value = tmp; |
| 390 | } |
| 391 | |
| 392 | return value; |
| 393 | } |
| 394 | |
| 395 | static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth) |
| 396 | { |
| 397 | int value = qdm2_get_vlc(gb, vlc, 0, depth); |
| 398 | |
| 399 | return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); |
| 400 | } |
| 401 | |
| 402 | /** |
| 403 | * QDM2 checksum |
| 404 | * |
| 405 | * @param data pointer to data to be checksum'ed |
| 406 | * @param length data length |
| 407 | * @param value checksum value |
| 408 | * |
| 409 | * @return 0 if checksum is OK |
| 410 | */ |
| 411 | static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value) |
| 412 | { |
| 413 | int i; |
| 414 | |
| 415 | for (i = 0; i < length; i++) |
| 416 | value -= data[i]; |
| 417 | |
| 418 | return (uint16_t)(value & 0xffff); |
| 419 | } |
| 420 | |
| 421 | /** |
| 422 | * Fill a QDM2SubPacket structure with packet type, size, and data pointer. |
| 423 | * |
| 424 | * @param gb bitreader context |
| 425 | * @param sub_packet packet under analysis |
| 426 | */ |
| 427 | static void qdm2_decode_sub_packet_header(GetBitContext *gb, |
| 428 | QDM2SubPacket *sub_packet) |
| 429 | { |
| 430 | sub_packet->type = get_bits(gb, 8); |
| 431 | |
| 432 | if (sub_packet->type == 0) { |
| 433 | sub_packet->size = 0; |
| 434 | sub_packet->data = NULL; |
| 435 | } else { |
| 436 | sub_packet->size = get_bits(gb, 8); |
| 437 | |
| 438 | if (sub_packet->type & 0x80) { |
| 439 | sub_packet->size <<= 8; |
| 440 | sub_packet->size |= get_bits(gb, 8); |
| 441 | sub_packet->type &= 0x7f; |
| 442 | } |
| 443 | |
| 444 | if (sub_packet->type == 0x7f) |
| 445 | sub_packet->type |= (get_bits(gb, 8) << 8); |
| 446 | |
| 447 | // FIXME: this depends on bitreader-internal data |
| 448 | sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; |
| 449 | } |
| 450 | |
| 451 | av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n", |
| 452 | sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); |
| 453 | } |
| 454 | |
| 455 | /** |
| 456 | * Return node pointer to first packet of requested type in list. |
| 457 | * |
| 458 | * @param list list of subpackets to be scanned |
| 459 | * @param type type of searched subpacket |
| 460 | * @return node pointer for subpacket if found, else NULL |
| 461 | */ |
| 462 | static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list, |
| 463 | int type) |
| 464 | { |
| 465 | while (list && list->packet) { |
| 466 | if (list->packet->type == type) |
| 467 | return list; |
| 468 | list = list->next; |
| 469 | } |
| 470 | return NULL; |
| 471 | } |
| 472 | |
| 473 | /** |
| 474 | * Replace 8 elements with their average value. |
| 475 | * Called by qdm2_decode_superblock before starting subblock decoding. |
| 476 | * |
| 477 | * @param q context |
| 478 | */ |
| 479 | static void average_quantized_coeffs(QDM2Context *q) |
| 480 | { |
| 481 | int i, j, n, ch, sum; |
| 482 | |
| 483 | n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; |
| 484 | |
| 485 | for (ch = 0; ch < q->nb_channels; ch++) |
| 486 | for (i = 0; i < n; i++) { |
| 487 | sum = 0; |
| 488 | |
| 489 | for (j = 0; j < 8; j++) |
| 490 | sum += q->quantized_coeffs[ch][i][j]; |
| 491 | |
| 492 | sum /= 8; |
| 493 | if (sum > 0) |
| 494 | sum--; |
| 495 | |
| 496 | for (j = 0; j < 8; j++) |
| 497 | q->quantized_coeffs[ch][i][j] = sum; |
| 498 | } |
| 499 | } |
| 500 | |
| 501 | /** |
| 502 | * Build subband samples with noise weighted by q->tone_level. |
| 503 | * Called by synthfilt_build_sb_samples. |
| 504 | * |
| 505 | * @param q context |
| 506 | * @param sb subband index |
| 507 | */ |
| 508 | static void build_sb_samples_from_noise(QDM2Context *q, int sb) |
| 509 | { |
| 510 | int ch, j; |
| 511 | |
| 512 | FIX_NOISE_IDX(q->noise_idx); |
| 513 | |
| 514 | if (!q->nb_channels) |
| 515 | return; |
| 516 | |
| 517 | for (ch = 0; ch < q->nb_channels; ch++) { |
| 518 | for (j = 0; j < 64; j++) { |
| 519 | q->sb_samples[ch][j * 2][sb] = |
| 520 | SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j]; |
| 521 | q->sb_samples[ch][j * 2 + 1][sb] = |
| 522 | SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j]; |
| 523 | } |
| 524 | } |
| 525 | } |
| 526 | |
| 527 | /** |
| 528 | * Called while processing data from subpackets 11 and 12. |
| 529 | * Used after making changes to coding_method array. |
| 530 | * |
| 531 | * @param sb subband index |
| 532 | * @param channels number of channels |
| 533 | * @param coding_method q->coding_method[0][0][0] |
| 534 | */ |
| 535 | static int fix_coding_method_array(int sb, int channels, |
| 536 | sb_int8_array coding_method) |
| 537 | { |
| 538 | int j, k; |
| 539 | int ch; |
| 540 | int run, case_val; |
| 541 | |
| 542 | for (ch = 0; ch < channels; ch++) { |
| 543 | for (j = 0; j < 64; ) { |
| 544 | if (coding_method[ch][sb][j] < 8) |
| 545 | return -1; |
| 546 | if ((coding_method[ch][sb][j] - 8) > 22) { |
| 547 | run = 1; |
| 548 | case_val = 8; |
| 549 | } else { |
| 550 | switch (switchtable[coding_method[ch][sb][j] - 8]) { |
| 551 | case 0: run = 10; |
| 552 | case_val = 10; |
| 553 | break; |
| 554 | case 1: run = 1; |
| 555 | case_val = 16; |
| 556 | break; |
| 557 | case 2: run = 5; |
| 558 | case_val = 24; |
| 559 | break; |
| 560 | case 3: run = 3; |
| 561 | case_val = 30; |
| 562 | break; |
| 563 | case 4: run = 1; |
| 564 | case_val = 30; |
| 565 | break; |
| 566 | case 5: run = 1; |
| 567 | case_val = 8; |
| 568 | break; |
| 569 | default: run = 1; |
| 570 | case_val = 8; |
| 571 | break; |
| 572 | } |
| 573 | } |
| 574 | for (k = 0; k < run; k++) { |
| 575 | if (j + k < 128) { |
| 576 | if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) { |
| 577 | if (k > 0) { |
| 578 | SAMPLES_NEEDED |
| 579 | //not debugged, almost never used |
| 580 | memset(&coding_method[ch][sb][j + k], case_val, |
| 581 | k *sizeof(int8_t)); |
| 582 | memset(&coding_method[ch][sb][j + k], case_val, |
| 583 | 3 * sizeof(int8_t)); |
| 584 | } |
| 585 | } |
| 586 | } |
| 587 | } |
| 588 | j += run; |
| 589 | } |
| 590 | } |
| 591 | return 0; |
| 592 | } |
| 593 | |
| 594 | /** |
| 595 | * Related to synthesis filter |
| 596 | * Called by process_subpacket_10 |
| 597 | * |
| 598 | * @param q context |
| 599 | * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 |
| 600 | */ |
| 601 | static void fill_tone_level_array(QDM2Context *q, int flag) |
| 602 | { |
| 603 | int i, sb, ch, sb_used; |
| 604 | int tmp, tab; |
| 605 | |
| 606 | for (ch = 0; ch < q->nb_channels; ch++) |
| 607 | for (sb = 0; sb < 30; sb++) |
| 608 | for (i = 0; i < 8; i++) { |
| 609 | if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) |
| 610 | tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ |
| 611 | q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; |
| 612 | else |
| 613 | tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; |
| 614 | if(tmp < 0) |
| 615 | tmp += 0xff; |
| 616 | q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; |
| 617 | } |
| 618 | |
| 619 | sb_used = QDM2_SB_USED(q->sub_sampling); |
| 620 | |
| 621 | if ((q->superblocktype_2_3 != 0) && !flag) { |
| 622 | for (sb = 0; sb < sb_used; sb++) |
| 623 | for (ch = 0; ch < q->nb_channels; ch++) |
| 624 | for (i = 0; i < 64; i++) { |
| 625 | q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; |
| 626 | if (q->tone_level_idx[ch][sb][i] < 0) |
| 627 | q->tone_level[ch][sb][i] = 0; |
| 628 | else |
| 629 | q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; |
| 630 | } |
| 631 | } else { |
| 632 | tab = q->superblocktype_2_3 ? 0 : 1; |
| 633 | for (sb = 0; sb < sb_used; sb++) { |
| 634 | if ((sb >= 4) && (sb <= 23)) { |
| 635 | for (ch = 0; ch < q->nb_channels; ch++) |
| 636 | for (i = 0; i < 64; i++) { |
| 637 | tmp = q->tone_level_idx_base[ch][sb][i / 8] - |
| 638 | q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - |
| 639 | q->tone_level_idx_mid[ch][sb - 4][i / 8] - |
| 640 | q->tone_level_idx_hi2[ch][sb - 4]; |
| 641 | q->tone_level_idx[ch][sb][i] = tmp & 0xff; |
| 642 | if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) |
| 643 | q->tone_level[ch][sb][i] = 0; |
| 644 | else |
| 645 | q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; |
| 646 | } |
| 647 | } else { |
| 648 | if (sb > 4) { |
| 649 | for (ch = 0; ch < q->nb_channels; ch++) |
| 650 | for (i = 0; i < 64; i++) { |
| 651 | tmp = q->tone_level_idx_base[ch][sb][i / 8] - |
| 652 | q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - |
| 653 | q->tone_level_idx_hi2[ch][sb - 4]; |
| 654 | q->tone_level_idx[ch][sb][i] = tmp & 0xff; |
| 655 | if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) |
| 656 | q->tone_level[ch][sb][i] = 0; |
| 657 | else |
| 658 | q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; |
| 659 | } |
| 660 | } else { |
| 661 | for (ch = 0; ch < q->nb_channels; ch++) |
| 662 | for (i = 0; i < 64; i++) { |
| 663 | tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; |
| 664 | if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) |
| 665 | q->tone_level[ch][sb][i] = 0; |
| 666 | else |
| 667 | q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; |
| 668 | } |
| 669 | } |
| 670 | } |
| 671 | } |
| 672 | } |
| 673 | } |
| 674 | |
| 675 | /** |
| 676 | * Related to synthesis filter |
| 677 | * Called by process_subpacket_11 |
| 678 | * c is built with data from subpacket 11 |
| 679 | * Most of this function is used only if superblock_type_2_3 == 0, |
| 680 | * never seen it in samples. |
| 681 | * |
| 682 | * @param tone_level_idx |
| 683 | * @param tone_level_idx_temp |
| 684 | * @param coding_method q->coding_method[0][0][0] |
| 685 | * @param nb_channels number of channels |
| 686 | * @param c coming from subpacket 11, passed as 8*c |
| 687 | * @param superblocktype_2_3 flag based on superblock packet type |
| 688 | * @param cm_table_select q->cm_table_select |
| 689 | */ |
| 690 | static void fill_coding_method_array(sb_int8_array tone_level_idx, |
| 691 | sb_int8_array tone_level_idx_temp, |
| 692 | sb_int8_array coding_method, |
| 693 | int nb_channels, |
| 694 | int c, int superblocktype_2_3, |
| 695 | int cm_table_select) |
| 696 | { |
| 697 | int ch, sb, j; |
| 698 | int tmp, acc, esp_40, comp; |
| 699 | int add1, add2, add3, add4; |
| 700 | int64_t multres; |
| 701 | |
| 702 | if (!superblocktype_2_3) { |
| 703 | /* This case is untested, no samples available */ |
| 704 | avpriv_request_sample(NULL, "!superblocktype_2_3"); |
| 705 | return; |
| 706 | for (ch = 0; ch < nb_channels; ch++) |
| 707 | for (sb = 0; sb < 30; sb++) { |
| 708 | for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer |
| 709 | add1 = tone_level_idx[ch][sb][j] - 10; |
| 710 | if (add1 < 0) |
| 711 | add1 = 0; |
| 712 | add2 = add3 = add4 = 0; |
| 713 | if (sb > 1) { |
| 714 | add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; |
| 715 | if (add2 < 0) |
| 716 | add2 = 0; |
| 717 | } |
| 718 | if (sb > 0) { |
| 719 | add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; |
| 720 | if (add3 < 0) |
| 721 | add3 = 0; |
| 722 | } |
| 723 | if (sb < 29) { |
| 724 | add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; |
| 725 | if (add4 < 0) |
| 726 | add4 = 0; |
| 727 | } |
| 728 | tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; |
| 729 | if (tmp < 0) |
| 730 | tmp = 0; |
| 731 | tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; |
| 732 | } |
| 733 | tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; |
| 734 | } |
| 735 | acc = 0; |
| 736 | for (ch = 0; ch < nb_channels; ch++) |
| 737 | for (sb = 0; sb < 30; sb++) |
| 738 | for (j = 0; j < 64; j++) |
| 739 | acc += tone_level_idx_temp[ch][sb][j]; |
| 740 | |
| 741 | multres = 0x66666667LL * (acc * 10); |
| 742 | esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); |
| 743 | for (ch = 0; ch < nb_channels; ch++) |
| 744 | for (sb = 0; sb < 30; sb++) |
| 745 | for (j = 0; j < 64; j++) { |
| 746 | comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; |
| 747 | if (comp < 0) |
| 748 | comp += 0xff; |
| 749 | comp /= 256; // signed shift |
| 750 | switch(sb) { |
| 751 | case 0: |
| 752 | if (comp < 30) |
| 753 | comp = 30; |
| 754 | comp += 15; |
| 755 | break; |
| 756 | case 1: |
| 757 | if (comp < 24) |
| 758 | comp = 24; |
| 759 | comp += 10; |
| 760 | break; |
| 761 | case 2: |
| 762 | case 3: |
| 763 | case 4: |
| 764 | if (comp < 16) |
| 765 | comp = 16; |
| 766 | } |
| 767 | if (comp <= 5) |
| 768 | tmp = 0; |
| 769 | else if (comp <= 10) |
| 770 | tmp = 10; |
| 771 | else if (comp <= 16) |
| 772 | tmp = 16; |
| 773 | else if (comp <= 24) |
| 774 | tmp = -1; |
| 775 | else |
| 776 | tmp = 0; |
| 777 | coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; |
| 778 | } |
| 779 | for (sb = 0; sb < 30; sb++) |
| 780 | fix_coding_method_array(sb, nb_channels, coding_method); |
| 781 | for (ch = 0; ch < nb_channels; ch++) |
| 782 | for (sb = 0; sb < 30; sb++) |
| 783 | for (j = 0; j < 64; j++) |
| 784 | if (sb >= 10) { |
| 785 | if (coding_method[ch][sb][j] < 10) |
| 786 | coding_method[ch][sb][j] = 10; |
| 787 | } else { |
| 788 | if (sb >= 2) { |
| 789 | if (coding_method[ch][sb][j] < 16) |
| 790 | coding_method[ch][sb][j] = 16; |
| 791 | } else { |
| 792 | if (coding_method[ch][sb][j] < 30) |
| 793 | coding_method[ch][sb][j] = 30; |
| 794 | } |
| 795 | } |
| 796 | } else { // superblocktype_2_3 != 0 |
| 797 | for (ch = 0; ch < nb_channels; ch++) |
| 798 | for (sb = 0; sb < 30; sb++) |
| 799 | for (j = 0; j < 64; j++) |
| 800 | coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; |
| 801 | } |
| 802 | } |
| 803 | |
| 804 | /** |
| 805 | * |
| 806 | * Called by process_subpacket_11 to process more data from subpacket 11 |
| 807 | * with sb 0-8. |
| 808 | * Called by process_subpacket_12 to process data from subpacket 12 with |
| 809 | * sb 8-sb_used. |
| 810 | * |
| 811 | * @param q context |
| 812 | * @param gb bitreader context |
| 813 | * @param length packet length in bits |
| 814 | * @param sb_min lower subband processed (sb_min included) |
| 815 | * @param sb_max higher subband processed (sb_max excluded) |
| 816 | */ |
| 817 | static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, |
| 818 | int length, int sb_min, int sb_max) |
| 819 | { |
| 820 | int sb, j, k, n, ch, run, channels; |
| 821 | int joined_stereo, zero_encoding; |
| 822 | int type34_first; |
| 823 | float type34_div = 0; |
| 824 | float type34_predictor; |
| 825 | float samples[10]; |
| 826 | int sign_bits[16] = {0}; |
| 827 | |
| 828 | if (length == 0) { |
| 829 | // If no data use noise |
| 830 | for (sb=sb_min; sb < sb_max; sb++) |
| 831 | build_sb_samples_from_noise(q, sb); |
| 832 | |
| 833 | return 0; |
| 834 | } |
| 835 | |
| 836 | for (sb = sb_min; sb < sb_max; sb++) { |
| 837 | channels = q->nb_channels; |
| 838 | |
| 839 | if (q->nb_channels <= 1 || sb < 12) |
| 840 | joined_stereo = 0; |
| 841 | else if (sb >= 24) |
| 842 | joined_stereo = 1; |
| 843 | else |
| 844 | joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0; |
| 845 | |
| 846 | if (joined_stereo) { |
| 847 | if (get_bits_left(gb) >= 16) |
| 848 | for (j = 0; j < 16; j++) |
| 849 | sign_bits[j] = get_bits1(gb); |
| 850 | |
| 851 | for (j = 0; j < 64; j++) |
| 852 | if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) |
| 853 | q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; |
| 854 | |
| 855 | if (fix_coding_method_array(sb, q->nb_channels, |
| 856 | q->coding_method)) { |
| 857 | av_log(NULL, AV_LOG_ERROR, "coding method invalid\n"); |
| 858 | build_sb_samples_from_noise(q, sb); |
| 859 | continue; |
| 860 | } |
| 861 | channels = 1; |
| 862 | } |
| 863 | |
| 864 | for (ch = 0; ch < channels; ch++) { |
| 865 | FIX_NOISE_IDX(q->noise_idx); |
| 866 | zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0; |
| 867 | type34_predictor = 0.0; |
| 868 | type34_first = 1; |
| 869 | |
| 870 | for (j = 0; j < 128; ) { |
| 871 | switch (q->coding_method[ch][sb][j / 2]) { |
| 872 | case 8: |
| 873 | if (get_bits_left(gb) >= 10) { |
| 874 | if (zero_encoding) { |
| 875 | for (k = 0; k < 5; k++) { |
| 876 | if ((j + 2 * k) >= 128) |
| 877 | break; |
| 878 | samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; |
| 879 | } |
| 880 | } else { |
| 881 | n = get_bits(gb, 8); |
| 882 | if (n >= 243) { |
| 883 | av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); |
| 884 | return AVERROR_INVALIDDATA; |
| 885 | } |
| 886 | |
| 887 | for (k = 0; k < 5; k++) |
| 888 | samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; |
| 889 | } |
| 890 | for (k = 0; k < 5; k++) |
| 891 | samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); |
| 892 | } else { |
| 893 | for (k = 0; k < 10; k++) |
| 894 | samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); |
| 895 | } |
| 896 | run = 10; |
| 897 | break; |
| 898 | |
| 899 | case 10: |
| 900 | if (get_bits_left(gb) >= 1) { |
| 901 | float f = 0.81; |
| 902 | |
| 903 | if (get_bits1(gb)) |
| 904 | f = -f; |
| 905 | f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; |
| 906 | samples[0] = f; |
| 907 | } else { |
| 908 | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); |
| 909 | } |
| 910 | run = 1; |
| 911 | break; |
| 912 | |
| 913 | case 16: |
| 914 | if (get_bits_left(gb) >= 10) { |
| 915 | if (zero_encoding) { |
| 916 | for (k = 0; k < 5; k++) { |
| 917 | if ((j + k) >= 128) |
| 918 | break; |
| 919 | samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; |
| 920 | } |
| 921 | } else { |
| 922 | n = get_bits (gb, 8); |
| 923 | if (n >= 243) { |
| 924 | av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); |
| 925 | return AVERROR_INVALIDDATA; |
| 926 | } |
| 927 | |
| 928 | for (k = 0; k < 5; k++) |
| 929 | samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; |
| 930 | } |
| 931 | } else { |
| 932 | for (k = 0; k < 5; k++) |
| 933 | samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); |
| 934 | } |
| 935 | run = 5; |
| 936 | break; |
| 937 | |
| 938 | case 24: |
| 939 | if (get_bits_left(gb) >= 7) { |
| 940 | n = get_bits(gb, 7); |
| 941 | if (n >= 125) { |
| 942 | av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n"); |
| 943 | return AVERROR_INVALIDDATA; |
| 944 | } |
| 945 | |
| 946 | for (k = 0; k < 3; k++) |
| 947 | samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; |
| 948 | } else { |
| 949 | for (k = 0; k < 3; k++) |
| 950 | samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); |
| 951 | } |
| 952 | run = 3; |
| 953 | break; |
| 954 | |
| 955 | case 30: |
| 956 | if (get_bits_left(gb) >= 4) { |
| 957 | unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1); |
| 958 | if (index >= FF_ARRAY_ELEMS(type30_dequant)) { |
| 959 | av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index); |
| 960 | return AVERROR_INVALIDDATA; |
| 961 | } |
| 962 | samples[0] = type30_dequant[index]; |
| 963 | } else |
| 964 | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); |
| 965 | |
| 966 | run = 1; |
| 967 | break; |
| 968 | |
| 969 | case 34: |
| 970 | if (get_bits_left(gb) >= 7) { |
| 971 | if (type34_first) { |
| 972 | type34_div = (float)(1 << get_bits(gb, 2)); |
| 973 | samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; |
| 974 | type34_predictor = samples[0]; |
| 975 | type34_first = 0; |
| 976 | } else { |
| 977 | unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1); |
| 978 | if (index >= FF_ARRAY_ELEMS(type34_delta)) { |
| 979 | av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index); |
| 980 | return AVERROR_INVALIDDATA; |
| 981 | } |
| 982 | samples[0] = type34_delta[index] / type34_div + type34_predictor; |
| 983 | type34_predictor = samples[0]; |
| 984 | } |
| 985 | } else { |
| 986 | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); |
| 987 | } |
| 988 | run = 1; |
| 989 | break; |
| 990 | |
| 991 | default: |
| 992 | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); |
| 993 | run = 1; |
| 994 | break; |
| 995 | } |
| 996 | |
| 997 | if (joined_stereo) { |
| 998 | for (k = 0; k < run && j + k < 128; k++) { |
| 999 | q->sb_samples[0][j + k][sb] = |
| 1000 | q->tone_level[0][sb][(j + k) / 2] * samples[k]; |
| 1001 | if (q->nb_channels == 2) { |
| 1002 | if (sign_bits[(j + k) / 8]) |
| 1003 | q->sb_samples[1][j + k][sb] = |
| 1004 | q->tone_level[1][sb][(j + k) / 2] * -samples[k]; |
| 1005 | else |
| 1006 | q->sb_samples[1][j + k][sb] = |
| 1007 | q->tone_level[1][sb][(j + k) / 2] * samples[k]; |
| 1008 | } |
| 1009 | } |
| 1010 | } else { |
| 1011 | for (k = 0; k < run; k++) |
| 1012 | if ((j + k) < 128) |
| 1013 | q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k]; |
| 1014 | } |
| 1015 | |
| 1016 | j += run; |
| 1017 | } // j loop |
| 1018 | } // channel loop |
| 1019 | } // subband loop |
| 1020 | return 0; |
| 1021 | } |
| 1022 | |
| 1023 | /** |
| 1024 | * Init the first element of a channel in quantized_coeffs with data |
| 1025 | * from packet 10 (quantized_coeffs[ch][0]). |
| 1026 | * This is similar to process_subpacket_9, but for a single channel |
| 1027 | * and for element [0] |
| 1028 | * same VLC tables as process_subpacket_9 are used. |
| 1029 | * |
| 1030 | * @param quantized_coeffs pointer to quantized_coeffs[ch][0] |
| 1031 | * @param gb bitreader context |
| 1032 | */ |
| 1033 | static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs, |
| 1034 | GetBitContext *gb) |
| 1035 | { |
| 1036 | int i, k, run, level, diff; |
| 1037 | |
| 1038 | if (get_bits_left(gb) < 16) |
| 1039 | return -1; |
| 1040 | level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); |
| 1041 | |
| 1042 | quantized_coeffs[0] = level; |
| 1043 | |
| 1044 | for (i = 0; i < 7; ) { |
| 1045 | if (get_bits_left(gb) < 16) |
| 1046 | return -1; |
| 1047 | run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; |
| 1048 | |
| 1049 | if (i + run >= 8) |
| 1050 | return -1; |
| 1051 | |
| 1052 | if (get_bits_left(gb) < 16) |
| 1053 | return -1; |
| 1054 | diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); |
| 1055 | |
| 1056 | for (k = 1; k <= run; k++) |
| 1057 | quantized_coeffs[i + k] = (level + ((k * diff) / run)); |
| 1058 | |
| 1059 | level += diff; |
| 1060 | i += run; |
| 1061 | } |
| 1062 | return 0; |
| 1063 | } |
| 1064 | |
| 1065 | /** |
| 1066 | * Related to synthesis filter, process data from packet 10 |
| 1067 | * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 |
| 1068 | * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with |
| 1069 | * data from packet 10 |
| 1070 | * |
| 1071 | * @param q context |
| 1072 | * @param gb bitreader context |
| 1073 | */ |
| 1074 | static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb) |
| 1075 | { |
| 1076 | int sb, j, k, n, ch; |
| 1077 | |
| 1078 | for (ch = 0; ch < q->nb_channels; ch++) { |
| 1079 | init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb); |
| 1080 | |
| 1081 | if (get_bits_left(gb) < 16) { |
| 1082 | memset(q->quantized_coeffs[ch][0], 0, 8); |
| 1083 | break; |
| 1084 | } |
| 1085 | } |
| 1086 | |
| 1087 | n = q->sub_sampling + 1; |
| 1088 | |
| 1089 | for (sb = 0; sb < n; sb++) |
| 1090 | for (ch = 0; ch < q->nb_channels; ch++) |
| 1091 | for (j = 0; j < 8; j++) { |
| 1092 | if (get_bits_left(gb) < 1) |
| 1093 | break; |
| 1094 | if (get_bits1(gb)) { |
| 1095 | for (k=0; k < 8; k++) { |
| 1096 | if (get_bits_left(gb) < 16) |
| 1097 | break; |
| 1098 | q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); |
| 1099 | } |
| 1100 | } else { |
| 1101 | for (k=0; k < 8; k++) |
| 1102 | q->tone_level_idx_hi1[ch][sb][j][k] = 0; |
| 1103 | } |
| 1104 | } |
| 1105 | |
| 1106 | n = QDM2_SB_USED(q->sub_sampling) - 4; |
| 1107 | |
| 1108 | for (sb = 0; sb < n; sb++) |
| 1109 | for (ch = 0; ch < q->nb_channels; ch++) { |
| 1110 | if (get_bits_left(gb) < 16) |
| 1111 | break; |
| 1112 | q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); |
| 1113 | if (sb > 19) |
| 1114 | q->tone_level_idx_hi2[ch][sb] -= 16; |
| 1115 | else |
| 1116 | for (j = 0; j < 8; j++) |
| 1117 | q->tone_level_idx_mid[ch][sb][j] = -16; |
| 1118 | } |
| 1119 | |
| 1120 | n = QDM2_SB_USED(q->sub_sampling) - 5; |
| 1121 | |
| 1122 | for (sb = 0; sb < n; sb++) |
| 1123 | for (ch = 0; ch < q->nb_channels; ch++) |
| 1124 | for (j = 0; j < 8; j++) { |
| 1125 | if (get_bits_left(gb) < 16) |
| 1126 | break; |
| 1127 | q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; |
| 1128 | } |
| 1129 | } |
| 1130 | |
| 1131 | /** |
| 1132 | * Process subpacket 9, init quantized_coeffs with data from it |
| 1133 | * |
| 1134 | * @param q context |
| 1135 | * @param node pointer to node with packet |
| 1136 | */ |
| 1137 | static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node) |
| 1138 | { |
| 1139 | GetBitContext gb; |
| 1140 | int i, j, k, n, ch, run, level, diff; |
| 1141 | |
| 1142 | init_get_bits(&gb, node->packet->data, node->packet->size * 8); |
| 1143 | |
| 1144 | n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; |
| 1145 | |
| 1146 | for (i = 1; i < n; i++) |
| 1147 | for (ch = 0; ch < q->nb_channels; ch++) { |
| 1148 | level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); |
| 1149 | q->quantized_coeffs[ch][i][0] = level; |
| 1150 | |
| 1151 | for (j = 0; j < (8 - 1); ) { |
| 1152 | run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; |
| 1153 | diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); |
| 1154 | |
| 1155 | if (j + run >= 8) |
| 1156 | return -1; |
| 1157 | |
| 1158 | for (k = 1; k <= run; k++) |
| 1159 | q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run)); |
| 1160 | |
| 1161 | level += diff; |
| 1162 | j += run; |
| 1163 | } |
| 1164 | } |
| 1165 | |
| 1166 | for (ch = 0; ch < q->nb_channels; ch++) |
| 1167 | for (i = 0; i < 8; i++) |
| 1168 | q->quantized_coeffs[ch][0][i] = 0; |
| 1169 | |
| 1170 | return 0; |
| 1171 | } |
| 1172 | |
| 1173 | /** |
| 1174 | * Process subpacket 10 if not null, else |
| 1175 | * |
| 1176 | * @param q context |
| 1177 | * @param node pointer to node with packet |
| 1178 | */ |
| 1179 | static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node) |
| 1180 | { |
| 1181 | GetBitContext gb; |
| 1182 | |
| 1183 | if (node) { |
| 1184 | init_get_bits(&gb, node->packet->data, node->packet->size * 8); |
| 1185 | init_tone_level_dequantization(q, &gb); |
| 1186 | fill_tone_level_array(q, 1); |
| 1187 | } else { |
| 1188 | fill_tone_level_array(q, 0); |
| 1189 | } |
| 1190 | } |
| 1191 | |
| 1192 | /** |
| 1193 | * Process subpacket 11 |
| 1194 | * |
| 1195 | * @param q context |
| 1196 | * @param node pointer to node with packet |
| 1197 | */ |
| 1198 | static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node) |
| 1199 | { |
| 1200 | GetBitContext gb; |
| 1201 | int length = 0; |
| 1202 | |
| 1203 | if (node) { |
| 1204 | length = node->packet->size * 8; |
| 1205 | init_get_bits(&gb, node->packet->data, length); |
| 1206 | } |
| 1207 | |
| 1208 | if (length >= 32) { |
| 1209 | int c = get_bits(&gb, 13); |
| 1210 | |
| 1211 | if (c > 3) |
| 1212 | fill_coding_method_array(q->tone_level_idx, |
| 1213 | q->tone_level_idx_temp, q->coding_method, |
| 1214 | q->nb_channels, 8 * c, |
| 1215 | q->superblocktype_2_3, q->cm_table_select); |
| 1216 | } |
| 1217 | |
| 1218 | synthfilt_build_sb_samples(q, &gb, length, 0, 8); |
| 1219 | } |
| 1220 | |
| 1221 | /** |
| 1222 | * Process subpacket 12 |
| 1223 | * |
| 1224 | * @param q context |
| 1225 | * @param node pointer to node with packet |
| 1226 | */ |
| 1227 | static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node) |
| 1228 | { |
| 1229 | GetBitContext gb; |
| 1230 | int length = 0; |
| 1231 | |
| 1232 | if (node) { |
| 1233 | length = node->packet->size * 8; |
| 1234 | init_get_bits(&gb, node->packet->data, length); |
| 1235 | } |
| 1236 | |
| 1237 | synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); |
| 1238 | } |
| 1239 | |
| 1240 | /** |
| 1241 | * Process new subpackets for synthesis filter |
| 1242 | * |
| 1243 | * @param q context |
| 1244 | * @param list list with synthesis filter packets (list D) |
| 1245 | */ |
| 1246 | static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list) |
| 1247 | { |
| 1248 | QDM2SubPNode *nodes[4]; |
| 1249 | |
| 1250 | nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); |
| 1251 | if (nodes[0]) |
| 1252 | process_subpacket_9(q, nodes[0]); |
| 1253 | |
| 1254 | nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); |
| 1255 | if (nodes[1]) |
| 1256 | process_subpacket_10(q, nodes[1]); |
| 1257 | else |
| 1258 | process_subpacket_10(q, NULL); |
| 1259 | |
| 1260 | nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); |
| 1261 | if (nodes[0] && nodes[1] && nodes[2]) |
| 1262 | process_subpacket_11(q, nodes[2]); |
| 1263 | else |
| 1264 | process_subpacket_11(q, NULL); |
| 1265 | |
| 1266 | nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); |
| 1267 | if (nodes[0] && nodes[1] && nodes[3]) |
| 1268 | process_subpacket_12(q, nodes[3]); |
| 1269 | else |
| 1270 | process_subpacket_12(q, NULL); |
| 1271 | } |
| 1272 | |
| 1273 | /** |
| 1274 | * Decode superblock, fill packet lists. |
| 1275 | * |
| 1276 | * @param q context |
| 1277 | */ |
| 1278 | static void qdm2_decode_super_block(QDM2Context *q) |
| 1279 | { |
| 1280 | GetBitContext gb; |
| 1281 | QDM2SubPacket header, *packet; |
| 1282 | int i, packet_bytes, sub_packet_size, sub_packets_D; |
| 1283 | unsigned int next_index = 0; |
| 1284 | |
| 1285 | memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); |
| 1286 | memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); |
| 1287 | memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); |
| 1288 | |
| 1289 | q->sub_packets_B = 0; |
| 1290 | sub_packets_D = 0; |
| 1291 | |
| 1292 | average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] |
| 1293 | |
| 1294 | init_get_bits(&gb, q->compressed_data, q->compressed_size * 8); |
| 1295 | qdm2_decode_sub_packet_header(&gb, &header); |
| 1296 | |
| 1297 | if (header.type < 2 || header.type >= 8) { |
| 1298 | q->has_errors = 1; |
| 1299 | av_log(NULL, AV_LOG_ERROR, "bad superblock type\n"); |
| 1300 | return; |
| 1301 | } |
| 1302 | |
| 1303 | q->superblocktype_2_3 = (header.type == 2 || header.type == 3); |
| 1304 | packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); |
| 1305 | |
| 1306 | init_get_bits(&gb, header.data, header.size * 8); |
| 1307 | |
| 1308 | if (header.type == 2 || header.type == 4 || header.type == 5) { |
| 1309 | int csum = 257 * get_bits(&gb, 8); |
| 1310 | csum += 2 * get_bits(&gb, 8); |
| 1311 | |
| 1312 | csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); |
| 1313 | |
| 1314 | if (csum != 0) { |
| 1315 | q->has_errors = 1; |
| 1316 | av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n"); |
| 1317 | return; |
| 1318 | } |
| 1319 | } |
| 1320 | |
| 1321 | q->sub_packet_list_B[0].packet = NULL; |
| 1322 | q->sub_packet_list_D[0].packet = NULL; |
| 1323 | |
| 1324 | for (i = 0; i < 6; i++) |
| 1325 | if (--q->fft_level_exp[i] < 0) |
| 1326 | q->fft_level_exp[i] = 0; |
| 1327 | |
| 1328 | for (i = 0; packet_bytes > 0; i++) { |
| 1329 | int j; |
| 1330 | |
| 1331 | if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) { |
| 1332 | SAMPLES_NEEDED_2("too many packet bytes"); |
| 1333 | return; |
| 1334 | } |
| 1335 | |
| 1336 | q->sub_packet_list_A[i].next = NULL; |
| 1337 | |
| 1338 | if (i > 0) { |
| 1339 | q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; |
| 1340 | |
| 1341 | /* seek to next block */ |
| 1342 | init_get_bits(&gb, header.data, header.size * 8); |
| 1343 | skip_bits(&gb, next_index * 8); |
| 1344 | |
| 1345 | if (next_index >= header.size) |
| 1346 | break; |
| 1347 | } |
| 1348 | |
| 1349 | /* decode subpacket */ |
| 1350 | packet = &q->sub_packets[i]; |
| 1351 | qdm2_decode_sub_packet_header(&gb, packet); |
| 1352 | next_index = packet->size + get_bits_count(&gb) / 8; |
| 1353 | sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; |
| 1354 | |
| 1355 | if (packet->type == 0) |
| 1356 | break; |
| 1357 | |
| 1358 | if (sub_packet_size > packet_bytes) { |
| 1359 | if (packet->type != 10 && packet->type != 11 && packet->type != 12) |
| 1360 | break; |
| 1361 | packet->size += packet_bytes - sub_packet_size; |
| 1362 | } |
| 1363 | |
| 1364 | packet_bytes -= sub_packet_size; |
| 1365 | |
| 1366 | /* add subpacket to 'all subpackets' list */ |
| 1367 | q->sub_packet_list_A[i].packet = packet; |
| 1368 | |
| 1369 | /* add subpacket to related list */ |
| 1370 | if (packet->type == 8) { |
| 1371 | SAMPLES_NEEDED_2("packet type 8"); |
| 1372 | return; |
| 1373 | } else if (packet->type >= 9 && packet->type <= 12) { |
| 1374 | /* packets for MPEG Audio like Synthesis Filter */ |
| 1375 | QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); |
| 1376 | } else if (packet->type == 13) { |
| 1377 | for (j = 0; j < 6; j++) |
| 1378 | q->fft_level_exp[j] = get_bits(&gb, 6); |
| 1379 | } else if (packet->type == 14) { |
| 1380 | for (j = 0; j < 6; j++) |
| 1381 | q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); |
| 1382 | } else if (packet->type == 15) { |
| 1383 | SAMPLES_NEEDED_2("packet type 15") |
| 1384 | return; |
| 1385 | } else if (packet->type >= 16 && packet->type < 48 && |
| 1386 | !fft_subpackets[packet->type - 16]) { |
| 1387 | /* packets for FFT */ |
| 1388 | QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); |
| 1389 | } |
| 1390 | } // Packet bytes loop |
| 1391 | |
| 1392 | if (q->sub_packet_list_D[0].packet) { |
| 1393 | process_synthesis_subpackets(q, q->sub_packet_list_D); |
| 1394 | q->do_synth_filter = 1; |
| 1395 | } else if (q->do_synth_filter) { |
| 1396 | process_subpacket_10(q, NULL); |
| 1397 | process_subpacket_11(q, NULL); |
| 1398 | process_subpacket_12(q, NULL); |
| 1399 | } |
| 1400 | } |
| 1401 | |
| 1402 | static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet, |
| 1403 | int offset, int duration, int channel, |
| 1404 | int exp, int phase) |
| 1405 | { |
| 1406 | if (q->fft_coefs_min_index[duration] < 0) |
| 1407 | q->fft_coefs_min_index[duration] = q->fft_coefs_index; |
| 1408 | |
| 1409 | q->fft_coefs[q->fft_coefs_index].sub_packet = |
| 1410 | ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); |
| 1411 | q->fft_coefs[q->fft_coefs_index].channel = channel; |
| 1412 | q->fft_coefs[q->fft_coefs_index].offset = offset; |
| 1413 | q->fft_coefs[q->fft_coefs_index].exp = exp; |
| 1414 | q->fft_coefs[q->fft_coefs_index].phase = phase; |
| 1415 | q->fft_coefs_index++; |
| 1416 | } |
| 1417 | |
| 1418 | static void qdm2_fft_decode_tones(QDM2Context *q, int duration, |
| 1419 | GetBitContext *gb, int b) |
| 1420 | { |
| 1421 | int channel, stereo, phase, exp; |
| 1422 | int local_int_4, local_int_8, stereo_phase, local_int_10; |
| 1423 | int local_int_14, stereo_exp, local_int_20, local_int_28; |
| 1424 | int n, offset; |
| 1425 | |
| 1426 | local_int_4 = 0; |
| 1427 | local_int_28 = 0; |
| 1428 | local_int_20 = 2; |
| 1429 | local_int_8 = (4 - duration); |
| 1430 | local_int_10 = 1 << (q->group_order - duration - 1); |
| 1431 | offset = 1; |
| 1432 | |
| 1433 | while (get_bits_left(gb)>0) { |
| 1434 | if (q->superblocktype_2_3) { |
| 1435 | while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { |
| 1436 | if (get_bits_left(gb)<0) { |
| 1437 | if(local_int_4 < q->group_size) |
| 1438 | av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n"); |
| 1439 | return; |
| 1440 | } |
| 1441 | offset = 1; |
| 1442 | if (n == 0) { |
| 1443 | local_int_4 += local_int_10; |
| 1444 | local_int_28 += (1 << local_int_8); |
| 1445 | } else { |
| 1446 | local_int_4 += 8 * local_int_10; |
| 1447 | local_int_28 += (8 << local_int_8); |
| 1448 | } |
| 1449 | } |
| 1450 | offset += (n - 2); |
| 1451 | } else { |
| 1452 | offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); |
| 1453 | while (offset >= (local_int_10 - 1)) { |
| 1454 | offset += (1 - (local_int_10 - 1)); |
| 1455 | local_int_4 += local_int_10; |
| 1456 | local_int_28 += (1 << local_int_8); |
| 1457 | } |
| 1458 | } |
| 1459 | |
| 1460 | if (local_int_4 >= q->group_size) |
| 1461 | return; |
| 1462 | |
| 1463 | local_int_14 = (offset >> local_int_8); |
| 1464 | if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table)) |
| 1465 | return; |
| 1466 | |
| 1467 | if (q->nb_channels > 1) { |
| 1468 | channel = get_bits1(gb); |
| 1469 | stereo = get_bits1(gb); |
| 1470 | } else { |
| 1471 | channel = 0; |
| 1472 | stereo = 0; |
| 1473 | } |
| 1474 | |
| 1475 | exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); |
| 1476 | exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; |
| 1477 | exp = (exp < 0) ? 0 : exp; |
| 1478 | |
| 1479 | phase = get_bits(gb, 3); |
| 1480 | stereo_exp = 0; |
| 1481 | stereo_phase = 0; |
| 1482 | |
| 1483 | if (stereo) { |
| 1484 | stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); |
| 1485 | stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); |
| 1486 | if (stereo_phase < 0) |
| 1487 | stereo_phase += 8; |
| 1488 | } |
| 1489 | |
| 1490 | if (q->frequency_range > (local_int_14 + 1)) { |
| 1491 | int sub_packet = (local_int_20 + local_int_28); |
| 1492 | |
| 1493 | qdm2_fft_init_coefficient(q, sub_packet, offset, duration, |
| 1494 | channel, exp, phase); |
| 1495 | if (stereo) |
| 1496 | qdm2_fft_init_coefficient(q, sub_packet, offset, duration, |
| 1497 | 1 - channel, |
| 1498 | stereo_exp, stereo_phase); |
| 1499 | } |
| 1500 | offset++; |
| 1501 | } |
| 1502 | } |
| 1503 | |
| 1504 | static void qdm2_decode_fft_packets(QDM2Context *q) |
| 1505 | { |
| 1506 | int i, j, min, max, value, type, unknown_flag; |
| 1507 | GetBitContext gb; |
| 1508 | |
| 1509 | if (!q->sub_packet_list_B[0].packet) |
| 1510 | return; |
| 1511 | |
| 1512 | /* reset minimum indexes for FFT coefficients */ |
| 1513 | q->fft_coefs_index = 0; |
| 1514 | for (i = 0; i < 5; i++) |
| 1515 | q->fft_coefs_min_index[i] = -1; |
| 1516 | |
| 1517 | /* process subpackets ordered by type, largest type first */ |
| 1518 | for (i = 0, max = 256; i < q->sub_packets_B; i++) { |
| 1519 | QDM2SubPacket *packet = NULL; |
| 1520 | |
| 1521 | /* find subpacket with largest type less than max */ |
| 1522 | for (j = 0, min = 0; j < q->sub_packets_B; j++) { |
| 1523 | value = q->sub_packet_list_B[j].packet->type; |
| 1524 | if (value > min && value < max) { |
| 1525 | min = value; |
| 1526 | packet = q->sub_packet_list_B[j].packet; |
| 1527 | } |
| 1528 | } |
| 1529 | |
| 1530 | max = min; |
| 1531 | |
| 1532 | /* check for errors (?) */ |
| 1533 | if (!packet) |
| 1534 | return; |
| 1535 | |
| 1536 | if (i == 0 && |
| 1537 | (packet->type < 16 || packet->type >= 48 || |
| 1538 | fft_subpackets[packet->type - 16])) |
| 1539 | return; |
| 1540 | |
| 1541 | /* decode FFT tones */ |
| 1542 | init_get_bits(&gb, packet->data, packet->size * 8); |
| 1543 | |
| 1544 | if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) |
| 1545 | unknown_flag = 1; |
| 1546 | else |
| 1547 | unknown_flag = 0; |
| 1548 | |
| 1549 | type = packet->type; |
| 1550 | |
| 1551 | if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { |
| 1552 | int duration = q->sub_sampling + 5 - (type & 15); |
| 1553 | |
| 1554 | if (duration >= 0 && duration < 4) |
| 1555 | qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); |
| 1556 | } else if (type == 31) { |
| 1557 | for (j = 0; j < 4; j++) |
| 1558 | qdm2_fft_decode_tones(q, j, &gb, unknown_flag); |
| 1559 | } else if (type == 46) { |
| 1560 | for (j = 0; j < 6; j++) |
| 1561 | q->fft_level_exp[j] = get_bits(&gb, 6); |
| 1562 | for (j = 0; j < 4; j++) |
| 1563 | qdm2_fft_decode_tones(q, j, &gb, unknown_flag); |
| 1564 | } |
| 1565 | } // Loop on B packets |
| 1566 | |
| 1567 | /* calculate maximum indexes for FFT coefficients */ |
| 1568 | for (i = 0, j = -1; i < 5; i++) |
| 1569 | if (q->fft_coefs_min_index[i] >= 0) { |
| 1570 | if (j >= 0) |
| 1571 | q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; |
| 1572 | j = i; |
| 1573 | } |
| 1574 | if (j >= 0) |
| 1575 | q->fft_coefs_max_index[j] = q->fft_coefs_index; |
| 1576 | } |
| 1577 | |
| 1578 | static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone) |
| 1579 | { |
| 1580 | float level, f[6]; |
| 1581 | int i; |
| 1582 | QDM2Complex c; |
| 1583 | const double iscale = 2.0 * M_PI / 512.0; |
| 1584 | |
| 1585 | tone->phase += tone->phase_shift; |
| 1586 | |
| 1587 | /* calculate current level (maximum amplitude) of tone */ |
| 1588 | level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; |
| 1589 | c.im = level * sin(tone->phase * iscale); |
| 1590 | c.re = level * cos(tone->phase * iscale); |
| 1591 | |
| 1592 | /* generate FFT coefficients for tone */ |
| 1593 | if (tone->duration >= 3 || tone->cutoff >= 3) { |
| 1594 | tone->complex[0].im += c.im; |
| 1595 | tone->complex[0].re += c.re; |
| 1596 | tone->complex[1].im -= c.im; |
| 1597 | tone->complex[1].re -= c.re; |
| 1598 | } else { |
| 1599 | f[1] = -tone->table[4]; |
| 1600 | f[0] = tone->table[3] - tone->table[0]; |
| 1601 | f[2] = 1.0 - tone->table[2] - tone->table[3]; |
| 1602 | f[3] = tone->table[1] + tone->table[4] - 1.0; |
| 1603 | f[4] = tone->table[0] - tone->table[1]; |
| 1604 | f[5] = tone->table[2]; |
| 1605 | for (i = 0; i < 2; i++) { |
| 1606 | tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += |
| 1607 | c.re * f[i]; |
| 1608 | tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += |
| 1609 | c.im * ((tone->cutoff <= i) ? -f[i] : f[i]); |
| 1610 | } |
| 1611 | for (i = 0; i < 4; i++) { |
| 1612 | tone->complex[i].re += c.re * f[i + 2]; |
| 1613 | tone->complex[i].im += c.im * f[i + 2]; |
| 1614 | } |
| 1615 | } |
| 1616 | |
| 1617 | /* copy the tone if it has not yet died out */ |
| 1618 | if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { |
| 1619 | memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); |
| 1620 | q->fft_tone_end = (q->fft_tone_end + 1) % 1000; |
| 1621 | } |
| 1622 | } |
| 1623 | |
| 1624 | static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet) |
| 1625 | { |
| 1626 | int i, j, ch; |
| 1627 | const double iscale = 0.25 * M_PI; |
| 1628 | |
| 1629 | for (ch = 0; ch < q->channels; ch++) { |
| 1630 | memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); |
| 1631 | } |
| 1632 | |
| 1633 | |
| 1634 | /* apply FFT tones with duration 4 (1 FFT period) */ |
| 1635 | if (q->fft_coefs_min_index[4] >= 0) |
| 1636 | for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { |
| 1637 | float level; |
| 1638 | QDM2Complex c; |
| 1639 | |
| 1640 | if (q->fft_coefs[i].sub_packet != sub_packet) |
| 1641 | break; |
| 1642 | |
| 1643 | ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; |
| 1644 | level = (q->fft_coefs[i].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[i].exp & 63]; |
| 1645 | |
| 1646 | c.re = level * cos(q->fft_coefs[i].phase * iscale); |
| 1647 | c.im = level * sin(q->fft_coefs[i].phase * iscale); |
| 1648 | q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; |
| 1649 | q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; |
| 1650 | q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; |
| 1651 | q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; |
| 1652 | } |
| 1653 | |
| 1654 | /* generate existing FFT tones */ |
| 1655 | for (i = q->fft_tone_end; i != q->fft_tone_start; ) { |
| 1656 | qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); |
| 1657 | q->fft_tone_start = (q->fft_tone_start + 1) % 1000; |
| 1658 | } |
| 1659 | |
| 1660 | /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ |
| 1661 | for (i = 0; i < 4; i++) |
| 1662 | if (q->fft_coefs_min_index[i] >= 0) { |
| 1663 | for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { |
| 1664 | int offset, four_i; |
| 1665 | FFTTone tone; |
| 1666 | |
| 1667 | if (q->fft_coefs[j].sub_packet != sub_packet) |
| 1668 | break; |
| 1669 | |
| 1670 | four_i = (4 - i); |
| 1671 | offset = q->fft_coefs[j].offset >> four_i; |
| 1672 | ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; |
| 1673 | |
| 1674 | if (offset < q->frequency_range) { |
| 1675 | if (offset < 2) |
| 1676 | tone.cutoff = offset; |
| 1677 | else |
| 1678 | tone.cutoff = (offset >= 60) ? 3 : 2; |
| 1679 | |
| 1680 | tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63]; |
| 1681 | tone.complex = &q->fft.complex[ch][offset]; |
| 1682 | tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; |
| 1683 | tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; |
| 1684 | tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); |
| 1685 | tone.duration = i; |
| 1686 | tone.time_index = 0; |
| 1687 | |
| 1688 | qdm2_fft_generate_tone(q, &tone); |
| 1689 | } |
| 1690 | } |
| 1691 | q->fft_coefs_min_index[i] = j; |
| 1692 | } |
| 1693 | } |
| 1694 | |
| 1695 | static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet) |
| 1696 | { |
| 1697 | const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; |
| 1698 | float *out = q->output_buffer + channel; |
| 1699 | int i; |
| 1700 | q->fft.complex[channel][0].re *= 2.0f; |
| 1701 | q->fft.complex[channel][0].im = 0.0f; |
| 1702 | q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); |
| 1703 | /* add samples to output buffer */ |
| 1704 | for (i = 0; i < FFALIGN(q->fft_size, 8); i++) { |
| 1705 | out[0] += q->fft.complex[channel][i].re * gain; |
| 1706 | out[q->channels] += q->fft.complex[channel][i].im * gain; |
| 1707 | out += 2 * q->channels; |
| 1708 | } |
| 1709 | } |
| 1710 | |
| 1711 | /** |
| 1712 | * @param q context |
| 1713 | * @param index subpacket number |
| 1714 | */ |
| 1715 | static void qdm2_synthesis_filter(QDM2Context *q, int index) |
| 1716 | { |
| 1717 | int i, k, ch, sb_used, sub_sampling, dither_state = 0; |
| 1718 | |
| 1719 | /* copy sb_samples */ |
| 1720 | sb_used = QDM2_SB_USED(q->sub_sampling); |
| 1721 | |
| 1722 | for (ch = 0; ch < q->channels; ch++) |
| 1723 | for (i = 0; i < 8; i++) |
| 1724 | for (k = sb_used; k < SBLIMIT; k++) |
| 1725 | q->sb_samples[ch][(8 * index) + i][k] = 0; |
| 1726 | |
| 1727 | for (ch = 0; ch < q->nb_channels; ch++) { |
| 1728 | float *samples_ptr = q->samples + ch; |
| 1729 | |
| 1730 | for (i = 0; i < 8; i++) { |
| 1731 | ff_mpa_synth_filter_float(&q->mpadsp, |
| 1732 | q->synth_buf[ch], &(q->synth_buf_offset[ch]), |
| 1733 | ff_mpa_synth_window_float, &dither_state, |
| 1734 | samples_ptr, q->nb_channels, |
| 1735 | q->sb_samples[ch][(8 * index) + i]); |
| 1736 | samples_ptr += 32 * q->nb_channels; |
| 1737 | } |
| 1738 | } |
| 1739 | |
| 1740 | /* add samples to output buffer */ |
| 1741 | sub_sampling = (4 >> q->sub_sampling); |
| 1742 | |
| 1743 | for (ch = 0; ch < q->channels; ch++) |
| 1744 | for (i = 0; i < q->frame_size; i++) |
| 1745 | q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch]; |
| 1746 | } |
| 1747 | |
| 1748 | /** |
| 1749 | * Init static data (does not depend on specific file) |
| 1750 | * |
| 1751 | * @param q context |
| 1752 | */ |
| 1753 | static av_cold void qdm2_init_static_data(void) { |
| 1754 | static int done; |
| 1755 | |
| 1756 | if(done) |
| 1757 | return; |
| 1758 | |
| 1759 | qdm2_init_vlc(); |
| 1760 | ff_mpa_synth_init_float(ff_mpa_synth_window_float); |
| 1761 | softclip_table_init(); |
| 1762 | rnd_table_init(); |
| 1763 | init_noise_samples(); |
| 1764 | |
| 1765 | done = 1; |
| 1766 | } |
| 1767 | |
| 1768 | /** |
| 1769 | * Init parameters from codec extradata |
| 1770 | */ |
| 1771 | static av_cold int qdm2_decode_init(AVCodecContext *avctx) |
| 1772 | { |
| 1773 | QDM2Context *s = avctx->priv_data; |
| 1774 | uint8_t *extradata; |
| 1775 | int extradata_size; |
| 1776 | int tmp_val, tmp, size; |
| 1777 | |
| 1778 | qdm2_init_static_data(); |
| 1779 | |
| 1780 | /* extradata parsing |
| 1781 | |
| 1782 | Structure: |
| 1783 | wave { |
| 1784 | frma (QDM2) |
| 1785 | QDCA |
| 1786 | QDCP |
| 1787 | } |
| 1788 | |
| 1789 | 32 size (including this field) |
| 1790 | 32 tag (=frma) |
| 1791 | 32 type (=QDM2 or QDMC) |
| 1792 | |
| 1793 | 32 size (including this field, in bytes) |
| 1794 | 32 tag (=QDCA) // maybe mandatory parameters |
| 1795 | 32 unknown (=1) |
| 1796 | 32 channels (=2) |
| 1797 | 32 samplerate (=44100) |
| 1798 | 32 bitrate (=96000) |
| 1799 | 32 block size (=4096) |
| 1800 | 32 frame size (=256) (for one channel) |
| 1801 | 32 packet size (=1300) |
| 1802 | |
| 1803 | 32 size (including this field, in bytes) |
| 1804 | 32 tag (=QDCP) // maybe some tuneable parameters |
| 1805 | 32 float1 (=1.0) |
| 1806 | 32 zero ? |
| 1807 | 32 float2 (=1.0) |
| 1808 | 32 float3 (=1.0) |
| 1809 | 32 unknown (27) |
| 1810 | 32 unknown (8) |
| 1811 | 32 zero ? |
| 1812 | */ |
| 1813 | |
| 1814 | if (!avctx->extradata || (avctx->extradata_size < 48)) { |
| 1815 | av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); |
| 1816 | return -1; |
| 1817 | } |
| 1818 | |
| 1819 | extradata = avctx->extradata; |
| 1820 | extradata_size = avctx->extradata_size; |
| 1821 | |
| 1822 | while (extradata_size > 7) { |
| 1823 | if (!memcmp(extradata, "frmaQDM", 7)) |
| 1824 | break; |
| 1825 | extradata++; |
| 1826 | extradata_size--; |
| 1827 | } |
| 1828 | |
| 1829 | if (extradata_size < 12) { |
| 1830 | av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", |
| 1831 | extradata_size); |
| 1832 | return -1; |
| 1833 | } |
| 1834 | |
| 1835 | if (memcmp(extradata, "frmaQDM", 7)) { |
| 1836 | av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); |
| 1837 | return -1; |
| 1838 | } |
| 1839 | |
| 1840 | if (extradata[7] == 'C') { |
| 1841 | // s->is_qdmc = 1; |
| 1842 | av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); |
| 1843 | return -1; |
| 1844 | } |
| 1845 | |
| 1846 | extradata += 8; |
| 1847 | extradata_size -= 8; |
| 1848 | |
| 1849 | size = AV_RB32(extradata); |
| 1850 | |
| 1851 | if(size > extradata_size){ |
| 1852 | av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", |
| 1853 | extradata_size, size); |
| 1854 | return -1; |
| 1855 | } |
| 1856 | |
| 1857 | extradata += 4; |
| 1858 | av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); |
| 1859 | if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { |
| 1860 | av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); |
| 1861 | return -1; |
| 1862 | } |
| 1863 | |
| 1864 | extradata += 8; |
| 1865 | |
| 1866 | avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); |
| 1867 | extradata += 4; |
| 1868 | if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) { |
| 1869 | av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n"); |
| 1870 | return AVERROR_INVALIDDATA; |
| 1871 | } |
| 1872 | avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO : |
| 1873 | AV_CH_LAYOUT_MONO; |
| 1874 | |
| 1875 | avctx->sample_rate = AV_RB32(extradata); |
| 1876 | extradata += 4; |
| 1877 | |
| 1878 | avctx->bit_rate = AV_RB32(extradata); |
| 1879 | extradata += 4; |
| 1880 | |
| 1881 | s->group_size = AV_RB32(extradata); |
| 1882 | extradata += 4; |
| 1883 | |
| 1884 | s->fft_size = AV_RB32(extradata); |
| 1885 | extradata += 4; |
| 1886 | |
| 1887 | s->checksum_size = AV_RB32(extradata); |
| 1888 | if (s->checksum_size >= 1U << 28) { |
| 1889 | av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size); |
| 1890 | return AVERROR_INVALIDDATA; |
| 1891 | } |
| 1892 | |
| 1893 | s->fft_order = av_log2(s->fft_size) + 1; |
| 1894 | |
| 1895 | // something like max decodable tones |
| 1896 | s->group_order = av_log2(s->group_size) + 1; |
| 1897 | s->frame_size = s->group_size / 16; // 16 iterations per super block |
| 1898 | |
| 1899 | if (s->frame_size > QDM2_MAX_FRAME_SIZE) |
| 1900 | return AVERROR_INVALIDDATA; |
| 1901 | |
| 1902 | s->sub_sampling = s->fft_order - 7; |
| 1903 | s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); |
| 1904 | |
| 1905 | switch ((s->sub_sampling * 2 + s->channels - 1)) { |
| 1906 | case 0: tmp = 40; break; |
| 1907 | case 1: tmp = 48; break; |
| 1908 | case 2: tmp = 56; break; |
| 1909 | case 3: tmp = 72; break; |
| 1910 | case 4: tmp = 80; break; |
| 1911 | case 5: tmp = 100;break; |
| 1912 | default: tmp=s->sub_sampling; break; |
| 1913 | } |
| 1914 | tmp_val = 0; |
| 1915 | if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; |
| 1916 | if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; |
| 1917 | if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; |
| 1918 | if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; |
| 1919 | s->cm_table_select = tmp_val; |
| 1920 | |
| 1921 | if (avctx->bit_rate <= 8000) |
| 1922 | s->coeff_per_sb_select = 0; |
| 1923 | else if (avctx->bit_rate < 16000) |
| 1924 | s->coeff_per_sb_select = 1; |
| 1925 | else |
| 1926 | s->coeff_per_sb_select = 2; |
| 1927 | |
| 1928 | // Fail on unknown fft order |
| 1929 | if ((s->fft_order < 7) || (s->fft_order > 9)) { |
| 1930 | av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); |
| 1931 | return -1; |
| 1932 | } |
| 1933 | if (s->fft_size != (1 << (s->fft_order - 1))) { |
| 1934 | av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size); |
| 1935 | return AVERROR_INVALIDDATA; |
| 1936 | } |
| 1937 | |
| 1938 | ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); |
| 1939 | ff_mpadsp_init(&s->mpadsp); |
| 1940 | |
| 1941 | avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
| 1942 | |
| 1943 | return 0; |
| 1944 | } |
| 1945 | |
| 1946 | static av_cold int qdm2_decode_close(AVCodecContext *avctx) |
| 1947 | { |
| 1948 | QDM2Context *s = avctx->priv_data; |
| 1949 | |
| 1950 | ff_rdft_end(&s->rdft_ctx); |
| 1951 | |
| 1952 | return 0; |
| 1953 | } |
| 1954 | |
| 1955 | static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out) |
| 1956 | { |
| 1957 | int ch, i; |
| 1958 | const int frame_size = (q->frame_size * q->channels); |
| 1959 | |
| 1960 | if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2) |
| 1961 | return -1; |
| 1962 | |
| 1963 | /* select input buffer */ |
| 1964 | q->compressed_data = in; |
| 1965 | q->compressed_size = q->checksum_size; |
| 1966 | |
| 1967 | /* copy old block, clear new block of output samples */ |
| 1968 | memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); |
| 1969 | memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); |
| 1970 | |
| 1971 | /* decode block of QDM2 compressed data */ |
| 1972 | if (q->sub_packet == 0) { |
| 1973 | q->has_errors = 0; // zero it for a new super block |
| 1974 | av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); |
| 1975 | qdm2_decode_super_block(q); |
| 1976 | } |
| 1977 | |
| 1978 | /* parse subpackets */ |
| 1979 | if (!q->has_errors) { |
| 1980 | if (q->sub_packet == 2) |
| 1981 | qdm2_decode_fft_packets(q); |
| 1982 | |
| 1983 | qdm2_fft_tone_synthesizer(q, q->sub_packet); |
| 1984 | } |
| 1985 | |
| 1986 | /* sound synthesis stage 1 (FFT) */ |
| 1987 | for (ch = 0; ch < q->channels; ch++) { |
| 1988 | qdm2_calculate_fft(q, ch, q->sub_packet); |
| 1989 | |
| 1990 | if (!q->has_errors && q->sub_packet_list_C[0].packet) { |
| 1991 | SAMPLES_NEEDED_2("has errors, and C list is not empty") |
| 1992 | return -1; |
| 1993 | } |
| 1994 | } |
| 1995 | |
| 1996 | /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ |
| 1997 | if (!q->has_errors && q->do_synth_filter) |
| 1998 | qdm2_synthesis_filter(q, q->sub_packet); |
| 1999 | |
| 2000 | q->sub_packet = (q->sub_packet + 1) % 16; |
| 2001 | |
| 2002 | /* clip and convert output float[] to 16bit signed samples */ |
| 2003 | for (i = 0; i < frame_size; i++) { |
| 2004 | int value = (int)q->output_buffer[i]; |
| 2005 | |
| 2006 | if (value > SOFTCLIP_THRESHOLD) |
| 2007 | value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; |
| 2008 | else if (value < -SOFTCLIP_THRESHOLD) |
| 2009 | value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; |
| 2010 | |
| 2011 | out[i] = value; |
| 2012 | } |
| 2013 | |
| 2014 | return 0; |
| 2015 | } |
| 2016 | |
| 2017 | static int qdm2_decode_frame(AVCodecContext *avctx, void *data, |
| 2018 | int *got_frame_ptr, AVPacket *avpkt) |
| 2019 | { |
| 2020 | AVFrame *frame = data; |
| 2021 | const uint8_t *buf = avpkt->data; |
| 2022 | int buf_size = avpkt->size; |
| 2023 | QDM2Context *s = avctx->priv_data; |
| 2024 | int16_t *out; |
| 2025 | int i, ret; |
| 2026 | |
| 2027 | if(!buf) |
| 2028 | return 0; |
| 2029 | if(buf_size < s->checksum_size) |
| 2030 | return -1; |
| 2031 | |
| 2032 | /* get output buffer */ |
| 2033 | frame->nb_samples = 16 * s->frame_size; |
| 2034 | if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
| 2035 | return ret; |
| 2036 | out = (int16_t *)frame->data[0]; |
| 2037 | |
| 2038 | for (i = 0; i < 16; i++) { |
| 2039 | if (qdm2_decode(s, buf, out) < 0) |
| 2040 | return -1; |
| 2041 | out += s->channels * s->frame_size; |
| 2042 | } |
| 2043 | |
| 2044 | *got_frame_ptr = 1; |
| 2045 | |
| 2046 | return s->checksum_size; |
| 2047 | } |
| 2048 | |
| 2049 | AVCodec ff_qdm2_decoder = { |
| 2050 | .name = "qdm2", |
| 2051 | .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), |
| 2052 | .type = AVMEDIA_TYPE_AUDIO, |
| 2053 | .id = AV_CODEC_ID_QDM2, |
| 2054 | .priv_data_size = sizeof(QDM2Context), |
| 2055 | .init = qdm2_decode_init, |
| 2056 | .close = qdm2_decode_close, |
| 2057 | .decode = qdm2_decode_frame, |
| 2058 | .capabilities = CODEC_CAP_DR1, |
| 2059 | }; |