| Line | Branch | Exec | Source |
|---|---|---|---|
| 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 | |||
| 37 | #include "libavutil/attributes.h" | ||
| 38 | #include "libavutil/channel_layout.h" | ||
| 39 | #include "libavutil/mem_internal.h" | ||
| 40 | #include "libavutil/thread.h" | ||
| 41 | #include "libavutil/tx.h" | ||
| 42 | |||
| 43 | #define BITSTREAM_READER_LE | ||
| 44 | #include "avcodec.h" | ||
| 45 | #include "get_bits.h" | ||
| 46 | #include "bytestream.h" | ||
| 47 | #include "codec_internal.h" | ||
| 48 | #include "decode.h" | ||
| 49 | #include "mpegaudio.h" | ||
| 50 | #include "mpegaudiodsp.h" | ||
| 51 | |||
| 52 | #include "qdm2_tablegen.h" | ||
| 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 QDM2SubPacket { | ||
| 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 FFTTone { | ||
| 101 | float level; | ||
| 102 | AVComplexFloat *complex; | ||
| 103 | const float *table; | ||
| 104 | int phase; | ||
| 105 | int phase_shift; | ||
| 106 | int duration; | ||
| 107 | short time_index; | ||
| 108 | short cutoff; | ||
| 109 | } FFTTone; | ||
| 110 | |||
| 111 | typedef struct FFTCoefficient { | ||
| 112 | int16_t sub_packet; | ||
| 113 | uint8_t channel; | ||
| 114 | int16_t offset; | ||
| 115 | int16_t exp; | ||
| 116 | uint8_t phase; | ||
| 117 | } FFTCoefficient; | ||
| 118 | |||
| 119 | typedef struct QDM2FFT { | ||
| 120 | DECLARE_ALIGNED(32, AVComplexFloat, complex)[MPA_MAX_CHANNELS][256 + 1]; | ||
| 121 | DECLARE_ALIGNED(32, AVComplexFloat, temp)[MPA_MAX_CHANNELS][256]; | ||
| 122 | } QDM2FFT; | ||
| 123 | |||
| 124 | /** | ||
| 125 | * QDM2 decoder context | ||
| 126 | */ | ||
| 127 | typedef struct QDM2Context { | ||
| 128 | /// Parameters from codec header, do not change during playback | ||
| 129 | int nb_channels; ///< number of channels | ||
| 130 | int channels; ///< number of channels | ||
| 131 | int group_size; ///< size of frame group (16 frames per group) | ||
| 132 | int fft_size; ///< size of FFT, in complex numbers | ||
| 133 | int checksum_size; ///< size of data block, used also for checksum | ||
| 134 | |||
| 135 | /// Parameters built from header parameters, do not change during playback | ||
| 136 | int group_order; ///< order of frame group | ||
| 137 | int fft_order; ///< order of FFT (actually fftorder+1) | ||
| 138 | int frame_size; ///< size of data frame | ||
| 139 | int frequency_range; | ||
| 140 | int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ | ||
| 141 | int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 | ||
| 142 | int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) | ||
| 143 | |||
| 144 | /// Packets and packet lists | ||
| 145 | QDM2SubPacket sub_packets[16]; ///< the packets themselves | ||
| 146 | QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets | ||
| 147 | QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list | ||
| 148 | int sub_packets_B; ///< number of packets on 'B' list | ||
| 149 | QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? | ||
| 150 | QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets | ||
| 151 | |||
| 152 | /// FFT and tones | ||
| 153 | FFTTone fft_tones[1000]; | ||
| 154 | int fft_tone_start; | ||
| 155 | int fft_tone_end; | ||
| 156 | FFTCoefficient fft_coefs[1000]; | ||
| 157 | int fft_coefs_index; | ||
| 158 | int fft_coefs_min_index[5]; | ||
| 159 | int fft_coefs_max_index[5]; | ||
| 160 | int fft_level_exp[6]; | ||
| 161 | AVTXContext *rdft_ctx; | ||
| 162 | av_tx_fn rdft_fn; | ||
| 163 | QDM2FFT fft; | ||
| 164 | |||
| 165 | /// I/O data | ||
| 166 | const uint8_t *compressed_data; | ||
| 167 | int compressed_size; | ||
| 168 | float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2]; | ||
| 169 | |||
| 170 | /// Synthesis filter | ||
| 171 | MPADSPContext mpadsp; | ||
| 172 | DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2]; | ||
| 173 | int synth_buf_offset[MPA_MAX_CHANNELS]; | ||
| 174 | DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; | ||
| 175 | DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; | ||
| 176 | |||
| 177 | /// Mixed temporary data used in decoding | ||
| 178 | float tone_level[MPA_MAX_CHANNELS][30][64]; | ||
| 179 | int8_t coding_method[MPA_MAX_CHANNELS][30][64]; | ||
| 180 | int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; | ||
| 181 | int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; | ||
| 182 | int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; | ||
| 183 | int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; | ||
| 184 | int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; | ||
| 185 | int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; | ||
| 186 | int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; | ||
| 187 | |||
| 188 | // Flags | ||
| 189 | int has_errors; ///< packet has errors | ||
| 190 | int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type | ||
| 191 | int do_synth_filter; ///< used to perform or skip synthesis filter | ||
| 192 | |||
| 193 | int sub_packet; | ||
| 194 | int noise_idx; ///< index for dithering noise table | ||
| 195 | } QDM2Context; | ||
| 196 | |||
| 197 | static const int switchtable[23] = { | ||
| 198 | 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4 | ||
| 199 | }; | ||
| 200 | |||
| 201 | 67959 | static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth) | |
| 202 | { | ||
| 203 | 67959 | int value = get_vlc2(gb, vlc->table, vlc->bits, | |
| 204 | av_builtin_constant_p(depth) ? depth : 2); | ||
| 205 | |||
| 206 | /* stage-2, 3 bits exponent escape sequence */ | ||
| 207 |
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67959 | if (value < 0) |
| 208 | 558 | value = get_bits(gb, get_bits(gb, 3) + 1); | |
| 209 | |||
| 210 | /* stage-3, optional */ | ||
| 211 |
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67959 | if (flag) { |
| 212 | int tmp; | ||
| 213 | |||
| 214 |
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18368 | if (value >= 60) { |
| 215 | ✗ | av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value); | |
| 216 | ✗ | return 0; | |
| 217 | } | ||
| 218 | |||
| 219 | 18368 | tmp= vlc_stage3_values[value]; | |
| 220 | |||
| 221 |
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18368 | if ((value & ~3) > 0) |
| 222 | 14255 | tmp += get_bits(gb, (value >> 2)); | |
| 223 | 18368 | value = tmp; | |
| 224 | } | ||
| 225 | |||
| 226 | 67959 | return value; | |
| 227 | } | ||
| 228 | |||
| 229 | 11196 | static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth) | |
| 230 | { | ||
| 231 | 11196 | int value = qdm2_get_vlc(gb, vlc, 0, depth); | |
| 232 | |||
| 233 |
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11196 | return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); |
| 234 | } | ||
| 235 | |||
| 236 | /** | ||
| 237 | * QDM2 checksum | ||
| 238 | * | ||
| 239 | * @param data pointer to data to be checksummed | ||
| 240 | * @param length data length | ||
| 241 | * @param value checksum value | ||
| 242 | * | ||
| 243 | * @return 0 if checksum is OK | ||
| 244 | */ | ||
| 245 | 141 | static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value) | |
| 246 | { | ||
| 247 | int i; | ||
| 248 | |||
| 249 |
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52311 | for (i = 0; i < length; i++) |
| 250 | 52170 | value -= data[i]; | |
| 251 | |||
| 252 | 141 | return (uint16_t)(value & 0xffff); | |
| 253 | } | ||
| 254 | |||
| 255 | /** | ||
| 256 | * Fill a QDM2SubPacket structure with packet type, size, and data pointer. | ||
| 257 | * | ||
| 258 | * @param gb bitreader context | ||
| 259 | * @param sub_packet packet under analysis | ||
| 260 | */ | ||
| 261 | 1128 | static void qdm2_decode_sub_packet_header(GetBitContext *gb, | |
| 262 | QDM2SubPacket *sub_packet) | ||
| 263 | { | ||
| 264 | 1128 | sub_packet->type = get_bits(gb, 8); | |
| 265 | |||
| 266 |
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1128 | if (sub_packet->type == 0) { |
| 267 | 141 | sub_packet->size = 0; | |
| 268 | 141 | sub_packet->data = NULL; | |
| 269 | } else { | ||
| 270 | 987 | sub_packet->size = get_bits(gb, 8); | |
| 271 | |||
| 272 |
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987 | if (sub_packet->type & 0x80) { |
| 273 | 141 | sub_packet->size <<= 8; | |
| 274 | 141 | sub_packet->size |= get_bits(gb, 8); | |
| 275 | 141 | sub_packet->type &= 0x7f; | |
| 276 | } | ||
| 277 | |||
| 278 |
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987 | if (sub_packet->type == 0x7f) |
| 279 | ✗ | sub_packet->type |= (get_bits(gb, 8) << 8); | |
| 280 | |||
| 281 | // FIXME: this depends on bitreader-internal data | ||
| 282 | 987 | sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; | |
| 283 | } | ||
| 284 | |||
| 285 | 1128 | av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n", | |
| 286 | 1128 | sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); | |
| 287 | 1128 | } | |
| 288 | |||
| 289 | /** | ||
| 290 | * Return node pointer to first packet of requested type in list. | ||
| 291 | * | ||
| 292 | * @param list list of subpackets to be scanned | ||
| 293 | * @param type type of searched subpacket | ||
| 294 | * @return node pointer for subpacket if found, else NULL | ||
| 295 | */ | ||
| 296 | 564 | static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list, | |
| 297 | int type) | ||
| 298 | { | ||
| 299 |
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987 | while (list && list->packet) { |
| 300 |
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564 | if (list->packet->type == type) |
| 301 | 141 | return list; | |
| 302 | 423 | list = list->next; | |
| 303 | } | ||
| 304 | 423 | return NULL; | |
| 305 | } | ||
| 306 | |||
| 307 | /** | ||
| 308 | * Replace 8 elements with their average value. | ||
| 309 | * Called by qdm2_decode_superblock before starting subblock decoding. | ||
| 310 | * | ||
| 311 | * @param q context | ||
| 312 | */ | ||
| 313 | 141 | static void average_quantized_coeffs(QDM2Context *q) | |
| 314 | { | ||
| 315 | int i, j, n, ch, sum; | ||
| 316 | |||
| 317 |
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141 | n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; |
| 318 | |||
| 319 |
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423 | for (ch = 0; ch < q->nb_channels; ch++) |
| 320 |
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3102 | for (i = 0; i < n; i++) { |
| 321 | 2820 | sum = 0; | |
| 322 | |||
| 323 |
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25380 | for (j = 0; j < 8; j++) |
| 324 | 22560 | sum += q->quantized_coeffs[ch][i][j]; | |
| 325 | |||
| 326 | 2820 | sum /= 8; | |
| 327 |
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2820 | if (sum > 0) |
| 328 | 2466 | sum--; | |
| 329 | |||
| 330 |
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25380 | for (j = 0; j < 8; j++) |
| 331 | 22560 | q->quantized_coeffs[ch][i][j] = sum; | |
| 332 | } | ||
| 333 | 141 | } | |
| 334 | |||
| 335 | /** | ||
| 336 | * Build subband samples with noise weighted by q->tone_level. | ||
| 337 | * Called by synthfilt_build_sb_samples. | ||
| 338 | * | ||
| 339 | * @param q context | ||
| 340 | * @param sb subband index | ||
| 341 | */ | ||
| 342 | 4230 | static void build_sb_samples_from_noise(QDM2Context *q, int sb) | |
| 343 | { | ||
| 344 | int ch, j; | ||
| 345 | |||
| 346 |
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4230 | FIX_NOISE_IDX(q->noise_idx); |
| 347 | |||
| 348 |
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4230 | if (!q->nb_channels) |
| 349 | ✗ | return; | |
| 350 | |||
| 351 |
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12690 | for (ch = 0; ch < q->nb_channels; ch++) { |
| 352 |
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549900 | for (j = 0; j < 64; j++) { |
| 353 | 541440 | q->sb_samples[ch][j * 2][sb] = | |
| 354 | 541440 | SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j]; | |
| 355 | 541440 | q->sb_samples[ch][j * 2 + 1][sb] = | |
| 356 | 541440 | SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j]; | |
| 357 | } | ||
| 358 | } | ||
| 359 | } | ||
| 360 | |||
| 361 | /** | ||
| 362 | * Called while processing data from subpackets 11 and 12. | ||
| 363 | * Used after making changes to coding_method array. | ||
| 364 | * | ||
| 365 | * @param sb subband index | ||
| 366 | * @param channels number of channels | ||
| 367 | * @param coding_method q->coding_method[0][0][0] | ||
| 368 | */ | ||
| 369 | ✗ | static int fix_coding_method_array(int sb, int channels, | |
| 370 | sb_int8_array coding_method) | ||
| 371 | { | ||
| 372 | int j, k; | ||
| 373 | int ch; | ||
| 374 | int run, case_val; | ||
| 375 | |||
| 376 | ✗ | for (ch = 0; ch < channels; ch++) { | |
| 377 | ✗ | for (j = 0; j < 64; ) { | |
| 378 | ✗ | if (coding_method[ch][sb][j] < 8) | |
| 379 | ✗ | return -1; | |
| 380 | ✗ | if ((coding_method[ch][sb][j] - 8) > 22) { | |
| 381 | ✗ | run = 1; | |
| 382 | ✗ | case_val = 8; | |
| 383 | } else { | ||
| 384 | ✗ | switch (switchtable[coding_method[ch][sb][j] - 8]) { | |
| 385 | ✗ | case 0: run = 10; | |
| 386 | ✗ | case_val = 10; | |
| 387 | ✗ | break; | |
| 388 | ✗ | case 1: run = 1; | |
| 389 | ✗ | case_val = 16; | |
| 390 | ✗ | break; | |
| 391 | ✗ | case 2: run = 5; | |
| 392 | ✗ | case_val = 24; | |
| 393 | ✗ | break; | |
| 394 | ✗ | case 3: run = 3; | |
| 395 | ✗ | case_val = 30; | |
| 396 | ✗ | break; | |
| 397 | ✗ | case 4: run = 1; | |
| 398 | ✗ | case_val = 30; | |
| 399 | ✗ | break; | |
| 400 | ✗ | case 5: run = 1; | |
| 401 | ✗ | case_val = 8; | |
| 402 | ✗ | break; | |
| 403 | ✗ | default: run = 1; | |
| 404 | ✗ | case_val = 8; | |
| 405 | ✗ | break; | |
| 406 | } | ||
| 407 | } | ||
| 408 | ✗ | for (k = 0; k < run; k++) { | |
| 409 | ✗ | if (j + k < 128) { | |
| 410 | ✗ | int sbjk = sb + (j + k) / 64; | |
| 411 | ✗ | if (sbjk > 29) { | |
| 412 | ✗ | SAMPLES_NEEDED | |
| 413 | ✗ | continue; | |
| 414 | } | ||
| 415 | ✗ | if (coding_method[ch][sbjk][(j + k) % 64] > coding_method[ch][sb][j]) { | |
| 416 | ✗ | if (k > 0) { | |
| 417 | ✗ | SAMPLES_NEEDED | |
| 418 | //not debugged, almost never used | ||
| 419 | ✗ | memset(&coding_method[ch][sb][j + k], case_val, | |
| 420 | k *sizeof(int8_t)); | ||
| 421 | ✗ | memset(&coding_method[ch][sb][j + k], case_val, | |
| 422 | 3 * sizeof(int8_t)); | ||
| 423 | } | ||
| 424 | } | ||
| 425 | } | ||
| 426 | } | ||
| 427 | ✗ | j += run; | |
| 428 | } | ||
| 429 | } | ||
| 430 | ✗ | return 0; | |
| 431 | } | ||
| 432 | |||
| 433 | /** | ||
| 434 | * Related to synthesis filter | ||
| 435 | * Called by process_subpacket_10 | ||
| 436 | * | ||
| 437 | * @param q context | ||
| 438 | * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 | ||
| 439 | */ | ||
| 440 | 141 | static void fill_tone_level_array(QDM2Context *q, int flag) | |
| 441 | { | ||
| 442 | int i, sb, ch, sb_used; | ||
| 443 | int tmp, tab; | ||
| 444 | |||
| 445 |
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423 | for (ch = 0; ch < q->nb_channels; ch++) |
| 446 |
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8742 | for (sb = 0; sb < 30; sb++) |
| 447 |
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76140 | for (i = 0; i < 8; i++) { |
| 448 |
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67680 | if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) |
| 449 | 54144 | tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ | |
| 450 | 54144 | q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; | |
| 451 | else | ||
| 452 | 13536 | tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; | |
| 453 |
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67680 | if(tmp < 0) |
| 454 | ✗ | tmp += 0xff; | |
| 455 | 67680 | q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; | |
| 456 | } | ||
| 457 | |||
| 458 |
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141 | sb_used = QDM2_SB_USED(q->sub_sampling); |
| 459 | |||
| 460 |
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141 | if ((q->superblocktype_2_3 != 0) && !flag) { |
| 461 |
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4371 | for (sb = 0; sb < sb_used; sb++) |
| 462 |
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12690 | for (ch = 0; ch < q->nb_channels; ch++) |
| 463 |
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549900 | for (i = 0; i < 64; i++) { |
| 464 | 541440 | q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; | |
| 465 |
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541440 | if (q->tone_level_idx[ch][sb][i] < 0) |
| 466 | ✗ | q->tone_level[ch][sb][i] = 0; | |
| 467 | else | ||
| 468 | 541440 | q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; | |
| 469 | } | ||
| 470 | } else { | ||
| 471 | ✗ | tab = q->superblocktype_2_3 ? 0 : 1; | |
| 472 | ✗ | for (sb = 0; sb < sb_used; sb++) { | |
| 473 | ✗ | if ((sb >= 4) && (sb <= 23)) { | |
| 474 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) | |
| 475 | ✗ | for (i = 0; i < 64; i++) { | |
| 476 | ✗ | tmp = q->tone_level_idx_base[ch][sb][i / 8] - | |
| 477 | ✗ | q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - | |
| 478 | ✗ | q->tone_level_idx_mid[ch][sb - 4][i / 8] - | |
| 479 | ✗ | q->tone_level_idx_hi2[ch][sb - 4]; | |
| 480 | ✗ | q->tone_level_idx[ch][sb][i] = tmp & 0xff; | |
| 481 | ✗ | if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
| 482 | ✗ | q->tone_level[ch][sb][i] = 0; | |
| 483 | else | ||
| 484 | ✗ | q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
| 485 | } | ||
| 486 | } else { | ||
| 487 | ✗ | if (sb > 4) { | |
| 488 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) | |
| 489 | ✗ | for (i = 0; i < 64; i++) { | |
| 490 | ✗ | tmp = q->tone_level_idx_base[ch][sb][i / 8] - | |
| 491 | ✗ | q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - | |
| 492 | ✗ | q->tone_level_idx_hi2[ch][sb - 4]; | |
| 493 | ✗ | q->tone_level_idx[ch][sb][i] = tmp & 0xff; | |
| 494 | ✗ | if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
| 495 | ✗ | q->tone_level[ch][sb][i] = 0; | |
| 496 | else | ||
| 497 | ✗ | q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
| 498 | } | ||
| 499 | } else { | ||
| 500 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) | |
| 501 | ✗ | for (i = 0; i < 64; i++) { | |
| 502 | ✗ | tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; | |
| 503 | ✗ | if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
| 504 | ✗ | q->tone_level[ch][sb][i] = 0; | |
| 505 | else | ||
| 506 | ✗ | q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
| 507 | } | ||
| 508 | } | ||
| 509 | } | ||
| 510 | } | ||
| 511 | } | ||
| 512 | 141 | } | |
| 513 | |||
| 514 | /** | ||
| 515 | * Related to synthesis filter | ||
| 516 | * Called by process_subpacket_11 | ||
| 517 | * c is built with data from subpacket 11 | ||
| 518 | * Most of this function is used only if superblock_type_2_3 == 0, | ||
| 519 | * never seen it in samples. | ||
| 520 | * | ||
| 521 | * @param tone_level_idx | ||
| 522 | * @param tone_level_idx_temp | ||
| 523 | * @param coding_method q->coding_method[0][0][0] | ||
| 524 | * @param nb_channels number of channels | ||
| 525 | * @param c coming from subpacket 11, passed as 8*c | ||
| 526 | * @param superblocktype_2_3 flag based on superblock packet type | ||
| 527 | * @param cm_table_select q->cm_table_select | ||
| 528 | */ | ||
| 529 | ✗ | static void fill_coding_method_array(sb_int8_array tone_level_idx, | |
| 530 | sb_int8_array tone_level_idx_temp, | ||
| 531 | sb_int8_array coding_method, | ||
| 532 | int nb_channels, | ||
| 533 | int c, int superblocktype_2_3, | ||
| 534 | int cm_table_select) | ||
| 535 | { | ||
| 536 | int ch, sb, j; | ||
| 537 | int tmp, acc, esp_40, comp; | ||
| 538 | int add1, add2, add3, add4; | ||
| 539 | int64_t multres; | ||
| 540 | |||
| 541 | ✗ | if (!superblocktype_2_3) { | |
| 542 | /* This case is untested, no samples available */ | ||
| 543 | ✗ | avpriv_request_sample(NULL, "!superblocktype_2_3"); | |
| 544 | ✗ | return; | |
| 545 | for (ch = 0; ch < nb_channels; ch++) { | ||
| 546 | for (sb = 0; sb < 30; sb++) { | ||
| 547 | for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer | ||
| 548 | add1 = tone_level_idx[ch][sb][j] - 10; | ||
| 549 | if (add1 < 0) | ||
| 550 | add1 = 0; | ||
| 551 | add2 = add3 = add4 = 0; | ||
| 552 | if (sb > 1) { | ||
| 553 | add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; | ||
| 554 | if (add2 < 0) | ||
| 555 | add2 = 0; | ||
| 556 | } | ||
| 557 | if (sb > 0) { | ||
| 558 | add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; | ||
| 559 | if (add3 < 0) | ||
| 560 | add3 = 0; | ||
| 561 | } | ||
| 562 | if (sb < 29) { | ||
| 563 | add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; | ||
| 564 | if (add4 < 0) | ||
| 565 | add4 = 0; | ||
| 566 | } | ||
| 567 | tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; | ||
| 568 | if (tmp < 0) | ||
| 569 | tmp = 0; | ||
| 570 | tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; | ||
| 571 | } | ||
| 572 | tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; | ||
| 573 | } | ||
| 574 | } | ||
| 575 | acc = 0; | ||
| 576 | for (ch = 0; ch < nb_channels; ch++) | ||
| 577 | for (sb = 0; sb < 30; sb++) | ||
| 578 | for (j = 0; j < 64; j++) | ||
| 579 | acc += tone_level_idx_temp[ch][sb][j]; | ||
| 580 | |||
| 581 | multres = 0x66666667LL * (acc * 10); | ||
| 582 | esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); | ||
| 583 | for (ch = 0; ch < nb_channels; ch++) | ||
| 584 | for (sb = 0; sb < 30; sb++) | ||
| 585 | for (j = 0; j < 64; j++) { | ||
| 586 | comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; | ||
| 587 | if (comp < 0) | ||
| 588 | comp += 0xff; | ||
| 589 | comp /= 256; // signed shift | ||
| 590 | switch(sb) { | ||
| 591 | case 0: | ||
| 592 | if (comp < 30) | ||
| 593 | comp = 30; | ||
| 594 | comp += 15; | ||
| 595 | break; | ||
| 596 | case 1: | ||
| 597 | if (comp < 24) | ||
| 598 | comp = 24; | ||
| 599 | comp += 10; | ||
| 600 | break; | ||
| 601 | case 2: | ||
| 602 | case 3: | ||
| 603 | case 4: | ||
| 604 | if (comp < 16) | ||
| 605 | comp = 16; | ||
| 606 | } | ||
| 607 | if (comp <= 5) | ||
| 608 | tmp = 0; | ||
| 609 | else if (comp <= 10) | ||
| 610 | tmp = 10; | ||
| 611 | else if (comp <= 16) | ||
| 612 | tmp = 16; | ||
| 613 | else if (comp <= 24) | ||
| 614 | tmp = -1; | ||
| 615 | else | ||
| 616 | tmp = 0; | ||
| 617 | coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; | ||
| 618 | } | ||
| 619 | for (sb = 0; sb < 30; sb++) | ||
| 620 | fix_coding_method_array(sb, nb_channels, coding_method); | ||
| 621 | for (ch = 0; ch < nb_channels; ch++) | ||
| 622 | for (sb = 0; sb < 30; sb++) | ||
| 623 | for (j = 0; j < 64; j++) | ||
| 624 | if (sb >= 10) { | ||
| 625 | if (coding_method[ch][sb][j] < 10) | ||
| 626 | coding_method[ch][sb][j] = 10; | ||
| 627 | } else { | ||
| 628 | if (sb >= 2) { | ||
| 629 | if (coding_method[ch][sb][j] < 16) | ||
| 630 | coding_method[ch][sb][j] = 16; | ||
| 631 | } else { | ||
| 632 | if (coding_method[ch][sb][j] < 30) | ||
| 633 | coding_method[ch][sb][j] = 30; | ||
| 634 | } | ||
| 635 | } | ||
| 636 | } else { // superblocktype_2_3 != 0 | ||
| 637 | ✗ | for (ch = 0; ch < nb_channels; ch++) | |
| 638 | ✗ | for (sb = 0; sb < 30; sb++) | |
| 639 | ✗ | for (j = 0; j < 64; j++) | |
| 640 | ✗ | coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; | |
| 641 | } | ||
| 642 | } | ||
| 643 | |||
| 644 | /** | ||
| 645 | * Called by process_subpacket_11 to process more data from subpacket 11 | ||
| 646 | * with sb 0-8. | ||
| 647 | * Called by process_subpacket_12 to process data from subpacket 12 with | ||
| 648 | * sb 8-sb_used. | ||
| 649 | * | ||
| 650 | * @param q context | ||
| 651 | * @param gb bitreader context | ||
| 652 | * @param length packet length in bits | ||
| 653 | * @param sb_min lower subband processed (sb_min included) | ||
| 654 | * @param sb_max higher subband processed (sb_max excluded) | ||
| 655 | */ | ||
| 656 | 282 | static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, | |
| 657 | int length, int sb_min, int sb_max) | ||
| 658 | { | ||
| 659 | int sb, j, k, n, ch, run, channels; | ||
| 660 | int joined_stereo, zero_encoding; | ||
| 661 | int type34_first; | ||
| 662 | 282 | float type34_div = 0; | |
| 663 | float type34_predictor; | ||
| 664 | float samples[10]; | ||
| 665 | 282 | int sign_bits[16] = {0}; | |
| 666 | |||
| 667 |
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282 | if (length == 0) { |
| 668 | // If no data use noise | ||
| 669 |
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4512 | for (sb=sb_min; sb < sb_max; sb++) |
| 670 | 4230 | build_sb_samples_from_noise(q, sb); | |
| 671 | |||
| 672 | 282 | return 0; | |
| 673 | } | ||
| 674 | |||
| 675 | ✗ | for (sb = sb_min; sb < sb_max; sb++) { | |
| 676 | ✗ | channels = q->nb_channels; | |
| 677 | |||
| 678 | ✗ | if (q->nb_channels <= 1 || sb < 12) | |
| 679 | ✗ | joined_stereo = 0; | |
| 680 | ✗ | else if (sb >= 24) | |
| 681 | ✗ | joined_stereo = 1; | |
| 682 | else | ||
| 683 | ✗ | joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0; | |
| 684 | |||
| 685 | ✗ | if (joined_stereo) { | |
| 686 | ✗ | if (get_bits_left(gb) >= 16) | |
| 687 | ✗ | for (j = 0; j < 16; j++) | |
| 688 | ✗ | sign_bits[j] = get_bits1(gb); | |
| 689 | |||
| 690 | ✗ | for (j = 0; j < 64; j++) | |
| 691 | ✗ | if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) | |
| 692 | ✗ | q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; | |
| 693 | |||
| 694 | ✗ | if (fix_coding_method_array(sb, q->nb_channels, | |
| 695 | ✗ | q->coding_method)) { | |
| 696 | ✗ | av_log(NULL, AV_LOG_ERROR, "coding method invalid\n"); | |
| 697 | ✗ | build_sb_samples_from_noise(q, sb); | |
| 698 | ✗ | continue; | |
| 699 | } | ||
| 700 | ✗ | channels = 1; | |
| 701 | } | ||
| 702 | |||
| 703 | ✗ | for (ch = 0; ch < channels; ch++) { | |
| 704 | ✗ | FIX_NOISE_IDX(q->noise_idx); | |
| 705 | ✗ | zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0; | |
| 706 | ✗ | type34_predictor = 0.0; | |
| 707 | ✗ | type34_first = 1; | |
| 708 | |||
| 709 | ✗ | for (j = 0; j < 128; ) { | |
| 710 | ✗ | switch (q->coding_method[ch][sb][j / 2]) { | |
| 711 | ✗ | case 8: | |
| 712 | ✗ | if (get_bits_left(gb) >= 10) { | |
| 713 | ✗ | if (zero_encoding) { | |
| 714 | ✗ | for (k = 0; k < 5; k++) { | |
| 715 | ✗ | if ((j + 2 * k) >= 128) | |
| 716 | ✗ | break; | |
| 717 | ✗ | samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; | |
| 718 | } | ||
| 719 | } else { | ||
| 720 | ✗ | n = get_bits(gb, 8); | |
| 721 | ✗ | if (n >= 243) { | |
| 722 | ✗ | av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); | |
| 723 | ✗ | return AVERROR_INVALIDDATA; | |
| 724 | } | ||
| 725 | |||
| 726 | ✗ | for (k = 0; k < 5; k++) | |
| 727 | ✗ | samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
| 728 | } | ||
| 729 | ✗ | for (k = 0; k < 5; k++) | |
| 730 | ✗ | samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
| 731 | } else { | ||
| 732 | ✗ | for (k = 0; k < 10; k++) | |
| 733 | ✗ | samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
| 734 | } | ||
| 735 | ✗ | run = 10; | |
| 736 | ✗ | break; | |
| 737 | |||
| 738 | ✗ | case 10: | |
| 739 | ✗ | if (get_bits_left(gb) >= 1) { | |
| 740 | ✗ | float f = 0.81; | |
| 741 | |||
| 742 | ✗ | if (get_bits1(gb)) | |
| 743 | ✗ | f = -f; | |
| 744 | ✗ | f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; | |
| 745 | ✗ | samples[0] = f; | |
| 746 | } else { | ||
| 747 | ✗ | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
| 748 | } | ||
| 749 | ✗ | run = 1; | |
| 750 | ✗ | break; | |
| 751 | |||
| 752 | ✗ | case 16: | |
| 753 | ✗ | if (get_bits_left(gb) >= 10) { | |
| 754 | ✗ | if (zero_encoding) { | |
| 755 | ✗ | for (k = 0; k < 5; k++) { | |
| 756 | ✗ | if ((j + k) >= 128) | |
| 757 | ✗ | break; | |
| 758 | ✗ | samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; | |
| 759 | } | ||
| 760 | } else { | ||
| 761 | ✗ | n = get_bits (gb, 8); | |
| 762 | ✗ | if (n >= 243) { | |
| 763 | ✗ | av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); | |
| 764 | ✗ | return AVERROR_INVALIDDATA; | |
| 765 | } | ||
| 766 | |||
| 767 | ✗ | for (k = 0; k < 5; k++) | |
| 768 | ✗ | samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
| 769 | } | ||
| 770 | } else { | ||
| 771 | ✗ | for (k = 0; k < 5; k++) | |
| 772 | ✗ | samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
| 773 | } | ||
| 774 | ✗ | run = 5; | |
| 775 | ✗ | break; | |
| 776 | |||
| 777 | ✗ | case 24: | |
| 778 | ✗ | if (get_bits_left(gb) >= 7) { | |
| 779 | ✗ | n = get_bits(gb, 7); | |
| 780 | ✗ | if (n >= 125) { | |
| 781 | ✗ | av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n"); | |
| 782 | ✗ | return AVERROR_INVALIDDATA; | |
| 783 | } | ||
| 784 | |||
| 785 | ✗ | for (k = 0; k < 3; k++) | |
| 786 | ✗ | samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; | |
| 787 | } else { | ||
| 788 | ✗ | for (k = 0; k < 3; k++) | |
| 789 | ✗ | samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
| 790 | } | ||
| 791 | ✗ | run = 3; | |
| 792 | ✗ | break; | |
| 793 | |||
| 794 | ✗ | case 30: | |
| 795 | ✗ | if (get_bits_left(gb) >= 4) { | |
| 796 | ✗ | unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1); | |
| 797 | ✗ | if (index >= FF_ARRAY_ELEMS(type30_dequant)) { | |
| 798 | ✗ | av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index); | |
| 799 | ✗ | return AVERROR_INVALIDDATA; | |
| 800 | } | ||
| 801 | ✗ | samples[0] = type30_dequant[index]; | |
| 802 | } else | ||
| 803 | ✗ | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
| 804 | |||
| 805 | ✗ | run = 1; | |
| 806 | ✗ | break; | |
| 807 | |||
| 808 | ✗ | case 34: | |
| 809 | ✗ | if (get_bits_left(gb) >= 7) { | |
| 810 | ✗ | if (type34_first) { | |
| 811 | ✗ | type34_div = (float)(1 << get_bits(gb, 2)); | |
| 812 | ✗ | samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; | |
| 813 | ✗ | type34_predictor = samples[0]; | |
| 814 | ✗ | type34_first = 0; | |
| 815 | } else { | ||
| 816 | ✗ | unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1); | |
| 817 | ✗ | if (index >= FF_ARRAY_ELEMS(type34_delta)) { | |
| 818 | ✗ | av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index); | |
| 819 | ✗ | return AVERROR_INVALIDDATA; | |
| 820 | } | ||
| 821 | ✗ | samples[0] = type34_delta[index] / type34_div + type34_predictor; | |
| 822 | ✗ | type34_predictor = samples[0]; | |
| 823 | } | ||
| 824 | } else { | ||
| 825 | ✗ | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
| 826 | } | ||
| 827 | ✗ | run = 1; | |
| 828 | ✗ | break; | |
| 829 | |||
| 830 | ✗ | default: | |
| 831 | ✗ | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
| 832 | ✗ | run = 1; | |
| 833 | ✗ | break; | |
| 834 | } | ||
| 835 | |||
| 836 | ✗ | if (joined_stereo) { | |
| 837 | ✗ | for (k = 0; k < run && j + k < 128; k++) { | |
| 838 | ✗ | q->sb_samples[0][j + k][sb] = | |
| 839 | ✗ | q->tone_level[0][sb][(j + k) / 2] * samples[k]; | |
| 840 | ✗ | if (q->nb_channels == 2) { | |
| 841 | ✗ | if (sign_bits[(j + k) / 8]) | |
| 842 | ✗ | q->sb_samples[1][j + k][sb] = | |
| 843 | ✗ | q->tone_level[1][sb][(j + k) / 2] * -samples[k]; | |
| 844 | else | ||
| 845 | ✗ | q->sb_samples[1][j + k][sb] = | |
| 846 | ✗ | q->tone_level[1][sb][(j + k) / 2] * samples[k]; | |
| 847 | } | ||
| 848 | } | ||
| 849 | } else { | ||
| 850 | ✗ | for (k = 0; k < run; k++) | |
| 851 | ✗ | if ((j + k) < 128) | |
| 852 | ✗ | q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k]; | |
| 853 | } | ||
| 854 | |||
| 855 | ✗ | j += run; | |
| 856 | } // j loop | ||
| 857 | } // channel loop | ||
| 858 | } // subband loop | ||
| 859 | ✗ | return 0; | |
| 860 | } | ||
| 861 | |||
| 862 | /** | ||
| 863 | * Init the first element of a channel in quantized_coeffs with data | ||
| 864 | * from packet 10 (quantized_coeffs[ch][0]). | ||
| 865 | * This is similar to process_subpacket_9, but for a single channel | ||
| 866 | * and for element [0] | ||
| 867 | * same VLC tables as process_subpacket_9 are used. | ||
| 868 | * | ||
| 869 | * @param quantized_coeffs pointer to quantized_coeffs[ch][0] | ||
| 870 | * @param gb bitreader context | ||
| 871 | */ | ||
| 872 | ✗ | static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs, | |
| 873 | GetBitContext *gb) | ||
| 874 | { | ||
| 875 | int i, k, run, level, diff; | ||
| 876 | |||
| 877 | ✗ | if (get_bits_left(gb) < 16) | |
| 878 | ✗ | return -1; | |
| 879 | ✗ | level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); | |
| 880 | |||
| 881 | ✗ | quantized_coeffs[0] = level; | |
| 882 | |||
| 883 | ✗ | for (i = 0; i < 7; ) { | |
| 884 | ✗ | if (get_bits_left(gb) < 16) | |
| 885 | ✗ | return -1; | |
| 886 | ✗ | run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; | |
| 887 | |||
| 888 | ✗ | if (i + run >= 8) | |
| 889 | ✗ | return -1; | |
| 890 | |||
| 891 | ✗ | if (get_bits_left(gb) < 16) | |
| 892 | ✗ | return -1; | |
| 893 | ✗ | diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); | |
| 894 | |||
| 895 | ✗ | for (k = 1; k <= run; k++) | |
| 896 | ✗ | quantized_coeffs[i + k] = (level + ((k * diff) / run)); | |
| 897 | |||
| 898 | ✗ | level += diff; | |
| 899 | ✗ | i += run; | |
| 900 | } | ||
| 901 | ✗ | return 0; | |
| 902 | } | ||
| 903 | |||
| 904 | /** | ||
| 905 | * Related to synthesis filter, process data from packet 10 | ||
| 906 | * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 | ||
| 907 | * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with | ||
| 908 | * data from packet 10 | ||
| 909 | * | ||
| 910 | * @param q context | ||
| 911 | * @param gb bitreader context | ||
| 912 | */ | ||
| 913 | ✗ | static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb) | |
| 914 | { | ||
| 915 | int sb, j, k, n, ch; | ||
| 916 | |||
| 917 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) { | |
| 918 | ✗ | init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb); | |
| 919 | |||
| 920 | ✗ | if (get_bits_left(gb) < 16) { | |
| 921 | ✗ | memset(q->quantized_coeffs[ch][0], 0, 8); | |
| 922 | ✗ | break; | |
| 923 | } | ||
| 924 | } | ||
| 925 | |||
| 926 | ✗ | n = q->sub_sampling + 1; | |
| 927 | |||
| 928 | ✗ | for (sb = 0; sb < n; sb++) | |
| 929 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) | |
| 930 | ✗ | for (j = 0; j < 8; j++) { | |
| 931 | ✗ | if (get_bits_left(gb) < 1) | |
| 932 | ✗ | break; | |
| 933 | ✗ | if (get_bits1(gb)) { | |
| 934 | ✗ | for (k=0; k < 8; k++) { | |
| 935 | ✗ | if (get_bits_left(gb) < 16) | |
| 936 | ✗ | break; | |
| 937 | ✗ | q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); | |
| 938 | } | ||
| 939 | } else { | ||
| 940 | ✗ | for (k=0; k < 8; k++) | |
| 941 | ✗ | q->tone_level_idx_hi1[ch][sb][j][k] = 0; | |
| 942 | } | ||
| 943 | } | ||
| 944 | |||
| 945 | ✗ | n = QDM2_SB_USED(q->sub_sampling) - 4; | |
| 946 | |||
| 947 | ✗ | for (sb = 0; sb < n; sb++) | |
| 948 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) { | |
| 949 | ✗ | if (get_bits_left(gb) < 16) | |
| 950 | ✗ | break; | |
| 951 | ✗ | q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); | |
| 952 | ✗ | if (sb > 19) | |
| 953 | ✗ | q->tone_level_idx_hi2[ch][sb] -= 16; | |
| 954 | else | ||
| 955 | ✗ | for (j = 0; j < 8; j++) | |
| 956 | ✗ | q->tone_level_idx_mid[ch][sb][j] = -16; | |
| 957 | } | ||
| 958 | |||
| 959 | ✗ | n = QDM2_SB_USED(q->sub_sampling) - 5; | |
| 960 | |||
| 961 | ✗ | for (sb = 0; sb < n; sb++) | |
| 962 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) | |
| 963 | ✗ | for (j = 0; j < 8; j++) { | |
| 964 | ✗ | if (get_bits_left(gb) < 16) | |
| 965 | ✗ | break; | |
| 966 | ✗ | q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; | |
| 967 | } | ||
| 968 | ✗ | } | |
| 969 | |||
| 970 | /** | ||
| 971 | * Process subpacket 9, init quantized_coeffs with data from it | ||
| 972 | * | ||
| 973 | * @param q context | ||
| 974 | * @param node pointer to node with packet | ||
| 975 | */ | ||
| 976 | 141 | static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node) | |
| 977 | { | ||
| 978 | GetBitContext gb; | ||
| 979 | int i, j, k, n, ch, run, level, diff; | ||
| 980 | |||
| 981 | 141 | init_get_bits(&gb, node->packet->data, node->packet->size * 8); | |
| 982 | |||
| 983 |
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141 | n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; |
| 984 | |||
| 985 |
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1410 | for (i = 1; i < n; i++) |
| 986 |
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3807 | for (ch = 0; ch < q->nb_channels; ch++) { |
| 987 | 2538 | level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); | |
| 988 | 2538 | q->quantized_coeffs[ch][i][0] = level; | |
| 989 | |||
| 990 |
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13734 | for (j = 0; j < (8 - 1); ) { |
| 991 | 11196 | run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; | |
| 992 | 11196 | diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); | |
| 993 | |||
| 994 |
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11196 | if (j + run >= 8) |
| 995 | ✗ | return -1; | |
| 996 | |||
| 997 |
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28962 | for (k = 1; k <= run; k++) |
| 998 | 17766 | q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run)); | |
| 999 | |||
| 1000 | 11196 | level += diff; | |
| 1001 | 11196 | j += run; | |
| 1002 | } | ||
| 1003 | } | ||
| 1004 | |||
| 1005 |
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423 | for (ch = 0; ch < q->nb_channels; ch++) |
| 1006 |
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2538 | for (i = 0; i < 8; i++) |
| 1007 | 2256 | q->quantized_coeffs[ch][0][i] = 0; | |
| 1008 | |||
| 1009 | 141 | return 0; | |
| 1010 | } | ||
| 1011 | |||
| 1012 | /** | ||
| 1013 | * Process subpacket 10 if not null, else | ||
| 1014 | * | ||
| 1015 | * @param q context | ||
| 1016 | * @param node pointer to node with packet | ||
| 1017 | */ | ||
| 1018 | 141 | static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node) | |
| 1019 | { | ||
| 1020 | GetBitContext gb; | ||
| 1021 | |||
| 1022 |
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141 | if (node) { |
| 1023 | ✗ | init_get_bits(&gb, node->packet->data, node->packet->size * 8); | |
| 1024 | ✗ | init_tone_level_dequantization(q, &gb); | |
| 1025 | ✗ | fill_tone_level_array(q, 1); | |
| 1026 | } else { | ||
| 1027 | 141 | fill_tone_level_array(q, 0); | |
| 1028 | } | ||
| 1029 | 141 | } | |
| 1030 | |||
| 1031 | /** | ||
| 1032 | * Process subpacket 11 | ||
| 1033 | * | ||
| 1034 | * @param q context | ||
| 1035 | * @param node pointer to node with packet | ||
| 1036 | */ | ||
| 1037 | 141 | static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node) | |
| 1038 | { | ||
| 1039 | GetBitContext gb; | ||
| 1040 | 141 | int length = 0; | |
| 1041 | |||
| 1042 |
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141 | if (node) { |
| 1043 | ✗ | length = node->packet->size * 8; | |
| 1044 | ✗ | init_get_bits(&gb, node->packet->data, length); | |
| 1045 | } | ||
| 1046 | |||
| 1047 |
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141 | if (length >= 32) { |
| 1048 | ✗ | int c = get_bits(&gb, 13); | |
| 1049 | |||
| 1050 | ✗ | if (c > 3) | |
| 1051 | ✗ | fill_coding_method_array(q->tone_level_idx, | |
| 1052 | ✗ | q->tone_level_idx_temp, q->coding_method, | |
| 1053 | q->nb_channels, 8 * c, | ||
| 1054 | q->superblocktype_2_3, q->cm_table_select); | ||
| 1055 | } | ||
| 1056 | |||
| 1057 | 141 | synthfilt_build_sb_samples(q, &gb, length, 0, 8); | |
| 1058 | 141 | } | |
| 1059 | |||
| 1060 | /** | ||
| 1061 | * Process subpacket 12 | ||
| 1062 | * | ||
| 1063 | * @param q context | ||
| 1064 | * @param node pointer to node with packet | ||
| 1065 | */ | ||
| 1066 | 141 | static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node) | |
| 1067 | { | ||
| 1068 | GetBitContext gb; | ||
| 1069 | 141 | int length = 0; | |
| 1070 | |||
| 1071 |
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141 | if (node) { |
| 1072 | ✗ | length = node->packet->size * 8; | |
| 1073 | ✗ | init_get_bits(&gb, node->packet->data, length); | |
| 1074 | } | ||
| 1075 | |||
| 1076 |
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141 | synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); |
| 1077 | 141 | } | |
| 1078 | |||
| 1079 | /** | ||
| 1080 | * Process new subpackets for synthesis filter | ||
| 1081 | * | ||
| 1082 | * @param q context | ||
| 1083 | * @param list list with synthesis filter packets (list D) | ||
| 1084 | */ | ||
| 1085 | 141 | static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list) | |
| 1086 | { | ||
| 1087 | QDM2SubPNode *nodes[4]; | ||
| 1088 | |||
| 1089 | 141 | nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); | |
| 1090 |
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141 | if (nodes[0]) |
| 1091 | 141 | process_subpacket_9(q, nodes[0]); | |
| 1092 | |||
| 1093 | 141 | nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); | |
| 1094 |
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141 | if (nodes[1]) |
| 1095 | ✗ | process_subpacket_10(q, nodes[1]); | |
| 1096 | else | ||
| 1097 | 141 | process_subpacket_10(q, NULL); | |
| 1098 | |||
| 1099 | 141 | nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); | |
| 1100 |
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141 | if (nodes[0] && nodes[1] && nodes[2]) |
| 1101 | ✗ | process_subpacket_11(q, nodes[2]); | |
| 1102 | else | ||
| 1103 | 141 | process_subpacket_11(q, NULL); | |
| 1104 | |||
| 1105 | 141 | nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); | |
| 1106 |
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141 | if (nodes[0] && nodes[1] && nodes[3]) |
| 1107 | ✗ | process_subpacket_12(q, nodes[3]); | |
| 1108 | else | ||
| 1109 | 141 | process_subpacket_12(q, NULL); | |
| 1110 | 141 | } | |
| 1111 | |||
| 1112 | /** | ||
| 1113 | * Decode superblock, fill packet lists. | ||
| 1114 | * | ||
| 1115 | * @param q context | ||
| 1116 | */ | ||
| 1117 | 141 | static void qdm2_decode_super_block(QDM2Context *q) | |
| 1118 | { | ||
| 1119 | GetBitContext gb; | ||
| 1120 | QDM2SubPacket header, *packet; | ||
| 1121 | int i, packet_bytes, sub_packet_size, sub_packets_D; | ||
| 1122 | 141 | unsigned int next_index = 0; | |
| 1123 | |||
| 1124 | 141 | memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); | |
| 1125 | 141 | memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); | |
| 1126 | 141 | memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); | |
| 1127 | |||
| 1128 | 141 | q->sub_packets_B = 0; | |
| 1129 | 141 | sub_packets_D = 0; | |
| 1130 | |||
| 1131 | 141 | average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] | |
| 1132 | |||
| 1133 | 141 | init_get_bits(&gb, q->compressed_data, q->compressed_size * 8); | |
| 1134 | 141 | qdm2_decode_sub_packet_header(&gb, &header); | |
| 1135 | |||
| 1136 |
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141 | if (header.type < 2 || header.type >= 8) { |
| 1137 | ✗ | q->has_errors = 1; | |
| 1138 | ✗ | av_log(NULL, AV_LOG_ERROR, "bad superblock type\n"); | |
| 1139 | ✗ | return; | |
| 1140 | } | ||
| 1141 | |||
| 1142 |
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141 | q->superblocktype_2_3 = (header.type == 2 || header.type == 3); |
| 1143 | 141 | packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); | |
| 1144 | |||
| 1145 | 141 | init_get_bits(&gb, header.data, header.size * 8); | |
| 1146 | |||
| 1147 |
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141 | if (header.type == 2 || header.type == 4 || header.type == 5) { |
| 1148 | 141 | int csum = 257 * get_bits(&gb, 8); | |
| 1149 | 141 | csum += 2 * get_bits(&gb, 8); | |
| 1150 | |||
| 1151 | 141 | csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); | |
| 1152 | |||
| 1153 |
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141 | if (csum != 0) { |
| 1154 | ✗ | q->has_errors = 1; | |
| 1155 | ✗ | av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n"); | |
| 1156 | ✗ | return; | |
| 1157 | } | ||
| 1158 | } | ||
| 1159 | |||
| 1160 | 141 | q->sub_packet_list_B[0].packet = NULL; | |
| 1161 | 141 | q->sub_packet_list_D[0].packet = NULL; | |
| 1162 | |||
| 1163 |
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987 | for (i = 0; i < 6; i++) |
| 1164 |
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846 | if (--q->fft_level_exp[i] < 0) |
| 1165 | 846 | q->fft_level_exp[i] = 0; | |
| 1166 | |||
| 1167 |
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987 | for (i = 0; packet_bytes > 0; i++) { |
| 1168 | int j; | ||
| 1169 | |||
| 1170 |
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987 | if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) { |
| 1171 | ✗ | SAMPLES_NEEDED_2("too many packet bytes"); | |
| 1172 | ✗ | return; | |
| 1173 | } | ||
| 1174 | |||
| 1175 | 987 | q->sub_packet_list_A[i].next = NULL; | |
| 1176 | |||
| 1177 |
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987 | if (i > 0) { |
| 1178 | 846 | q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; | |
| 1179 | |||
| 1180 | /* seek to next block */ | ||
| 1181 | 846 | init_get_bits(&gb, header.data, header.size * 8); | |
| 1182 | 846 | skip_bits(&gb, next_index * 8); | |
| 1183 | |||
| 1184 |
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846 | if (next_index >= header.size) |
| 1185 | ✗ | break; | |
| 1186 | } | ||
| 1187 | |||
| 1188 | /* decode subpacket */ | ||
| 1189 | 987 | packet = &q->sub_packets[i]; | |
| 1190 | 987 | qdm2_decode_sub_packet_header(&gb, packet); | |
| 1191 | 987 | next_index = packet->size + get_bits_count(&gb) / 8; | |
| 1192 |
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987 | sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; |
| 1193 | |||
| 1194 |
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987 | if (packet->type == 0) |
| 1195 | 141 | break; | |
| 1196 | |||
| 1197 |
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846 | if (sub_packet_size > packet_bytes) { |
| 1198 | ✗ | if (packet->type != 10 && packet->type != 11 && packet->type != 12) | |
| 1199 | ✗ | break; | |
| 1200 | ✗ | packet->size += packet_bytes - sub_packet_size; | |
| 1201 | } | ||
| 1202 | |||
| 1203 | 846 | packet_bytes -= sub_packet_size; | |
| 1204 | |||
| 1205 | /* add subpacket to 'all subpackets' list */ | ||
| 1206 | 846 | q->sub_packet_list_A[i].packet = packet; | |
| 1207 | |||
| 1208 | /* add subpacket to related list */ | ||
| 1209 |
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846 | if (packet->type == 8) { |
| 1210 | ✗ | SAMPLES_NEEDED_2("packet type 8"); | |
| 1211 | ✗ | return; | |
| 1212 |
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846 | } else if (packet->type >= 9 && packet->type <= 12) { |
| 1213 | /* packets for MPEG Audio like Synthesis Filter */ | ||
| 1214 |
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141 | QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); |
| 1215 |
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705 | } else if (packet->type == 13) { |
| 1216 | ✗ | for (j = 0; j < 6; j++) | |
| 1217 | ✗ | q->fft_level_exp[j] = get_bits(&gb, 6); | |
| 1218 |
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705 | } else if (packet->type == 14) { |
| 1219 | ✗ | for (j = 0; j < 6; j++) | |
| 1220 | ✗ | q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); | |
| 1221 |
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705 | } else if (packet->type == 15) { |
| 1222 | ✗ | SAMPLES_NEEDED_2("packet type 15") | |
| 1223 | ✗ | return; | |
| 1224 |
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|
705 | } else if (packet->type >= 16 && packet->type < 48 && |
| 1225 |
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|
705 | !fft_subpackets[packet->type - 16]) { |
| 1226 | /* packets for FFT */ | ||
| 1227 |
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|
705 | QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); |
| 1228 | } | ||
| 1229 | } // Packet bytes loop | ||
| 1230 | |||
| 1231 |
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|
141 | if (q->sub_packet_list_D[0].packet) { |
| 1232 | 141 | process_synthesis_subpackets(q, q->sub_packet_list_D); | |
| 1233 | 141 | q->do_synth_filter = 1; | |
| 1234 | ✗ | } else if (q->do_synth_filter) { | |
| 1235 | ✗ | process_subpacket_10(q, NULL); | |
| 1236 | ✗ | process_subpacket_11(q, NULL); | |
| 1237 | ✗ | process_subpacket_12(q, NULL); | |
| 1238 | } | ||
| 1239 | } | ||
| 1240 | |||
| 1241 | 20217 | static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet, | |
| 1242 | int offset, int duration, int channel, | ||
| 1243 | int exp, int phase) | ||
| 1244 | { | ||
| 1245 |
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|
20217 | if (q->fft_coefs_min_index[duration] < 0) |
| 1246 | 551 | q->fft_coefs_min_index[duration] = q->fft_coefs_index; | |
| 1247 | |||
| 1248 |
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|
20217 | q->fft_coefs[q->fft_coefs_index].sub_packet = |
| 1249 | 296 | ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); | |
| 1250 | 20217 | q->fft_coefs[q->fft_coefs_index].channel = channel; | |
| 1251 | 20217 | q->fft_coefs[q->fft_coefs_index].offset = offset; | |
| 1252 | 20217 | q->fft_coefs[q->fft_coefs_index].exp = exp; | |
| 1253 | 20217 | q->fft_coefs[q->fft_coefs_index].phase = phase; | |
| 1254 | 20217 | q->fft_coefs_index++; | |
| 1255 | 20217 | } | |
| 1256 | |||
| 1257 | 564 | static void qdm2_fft_decode_tones(QDM2Context *q, int duration, | |
| 1258 | GetBitContext *gb, int b) | ||
| 1259 | { | ||
| 1260 | int channel, stereo, phase, exp; | ||
| 1261 | int local_int_4, local_int_8, stereo_phase, local_int_10; | ||
| 1262 | int local_int_14, stereo_exp, local_int_20, local_int_28; | ||
| 1263 | int n, offset; | ||
| 1264 | |||
| 1265 | 564 | local_int_4 = 0; | |
| 1266 | 564 | local_int_28 = 0; | |
| 1267 | 564 | local_int_20 = 2; | |
| 1268 | 564 | local_int_8 = (4 - duration); | |
| 1269 | 564 | local_int_10 = 1 << (q->group_order - duration - 1); | |
| 1270 | 564 | offset = 1; | |
| 1271 | |||
| 1272 |
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|
16337 | while (get_bits_left(gb)>0) { |
| 1273 |
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|
16337 | if (q->superblocktype_2_3) { |
| 1274 |
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|
18368 | while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { |
| 1275 |
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|
2098 | if (get_bits_left(gb)<0) { |
| 1276 |
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|
67 | if(local_int_4 < q->group_size) |
| 1277 | ✗ | av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n"); | |
| 1278 | 67 | return; | |
| 1279 | } | ||
| 1280 | 2031 | offset = 1; | |
| 1281 |
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|
2031 | if (n == 0) { |
| 1282 | 2019 | local_int_4 += local_int_10; | |
| 1283 | 2019 | local_int_28 += (1 << local_int_8); | |
| 1284 | } else { | ||
| 1285 | 12 | local_int_4 += 8 * local_int_10; | |
| 1286 | 12 | local_int_28 += (8 << local_int_8); | |
| 1287 | } | ||
| 1288 | } | ||
| 1289 | 16270 | offset += (n - 2); | |
| 1290 | } else { | ||
| 1291 | ✗ | if (local_int_10 <= 2) { | |
| 1292 | ✗ | av_log(NULL, AV_LOG_ERROR, "qdm2_fft_decode_tones() stuck\n"); | |
| 1293 | ✗ | return; | |
| 1294 | } | ||
| 1295 | ✗ | offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); | |
| 1296 | ✗ | while (offset >= (local_int_10 - 1)) { | |
| 1297 | ✗ | offset += (1 - (local_int_10 - 1)); | |
| 1298 | ✗ | local_int_4 += local_int_10; | |
| 1299 | ✗ | local_int_28 += (1 << local_int_8); | |
| 1300 | } | ||
| 1301 | } | ||
| 1302 | |||
| 1303 |
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|
16270 | if (local_int_4 >= q->group_size) |
| 1304 | 497 | return; | |
| 1305 | |||
| 1306 | 15773 | local_int_14 = (offset >> local_int_8); | |
| 1307 |
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|
15773 | if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table)) |
| 1308 | ✗ | return; | |
| 1309 | |||
| 1310 |
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|
15773 | if (q->nb_channels > 1) { |
| 1311 | 15773 | channel = get_bits1(gb); | |
| 1312 | 15773 | stereo = get_bits1(gb); | |
| 1313 | } else { | ||
| 1314 | ✗ | channel = 0; | |
| 1315 | ✗ | stereo = 0; | |
| 1316 | } | ||
| 1317 | |||
| 1318 |
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|
15773 | exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); |
| 1319 | 15773 | exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; | |
| 1320 | 15773 | exp = (exp < 0) ? 0 : exp; | |
| 1321 | |||
| 1322 | 15773 | phase = get_bits(gb, 3); | |
| 1323 | 15773 | stereo_exp = 0; | |
| 1324 | 15773 | stereo_phase = 0; | |
| 1325 | |||
| 1326 |
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15773 | if (stereo) { |
| 1327 | 4444 | stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); | |
| 1328 | 4444 | stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); | |
| 1329 |
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4444 | if (stereo_phase < 0) |
| 1330 | 497 | stereo_phase += 8; | |
| 1331 | } | ||
| 1332 | |||
| 1333 |
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15773 | if (q->frequency_range > (local_int_14 + 1)) { |
| 1334 | 15773 | int sub_packet = (local_int_20 + local_int_28); | |
| 1335 | |||
| 1336 |
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|
15773 | if (q->fft_coefs_index + stereo >= FF_ARRAY_ELEMS(q->fft_coefs)) |
| 1337 | ✗ | return; | |
| 1338 | |||
| 1339 | 15773 | qdm2_fft_init_coefficient(q, sub_packet, offset, duration, | |
| 1340 | channel, exp, phase); | ||
| 1341 |
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|
15773 | if (stereo) |
| 1342 | 4444 | qdm2_fft_init_coefficient(q, sub_packet, offset, duration, | |
| 1343 | 1 - channel, | ||
| 1344 | stereo_exp, stereo_phase); | ||
| 1345 | } | ||
| 1346 | 15773 | offset++; | |
| 1347 | } | ||
| 1348 | } | ||
| 1349 | |||
| 1350 | 141 | static void qdm2_decode_fft_packets(QDM2Context *q) | |
| 1351 | { | ||
| 1352 | int i, j, min, max, value, type, unknown_flag; | ||
| 1353 | GetBitContext gb; | ||
| 1354 | |||
| 1355 |
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141 | if (!q->sub_packet_list_B[0].packet) |
| 1356 | ✗ | return; | |
| 1357 | |||
| 1358 | /* reset minimum indexes for FFT coefficients */ | ||
| 1359 | 141 | q->fft_coefs_index = 0; | |
| 1360 |
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846 | for (i = 0; i < 5; i++) |
| 1361 | 705 | q->fft_coefs_min_index[i] = -1; | |
| 1362 | |||
| 1363 | /* process subpackets ordered by type, largest type first */ | ||
| 1364 |
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846 | for (i = 0, max = 256; i < q->sub_packets_B; i++) { |
| 1365 | 705 | QDM2SubPacket *packet = NULL; | |
| 1366 | |||
| 1367 | /* find subpacket with largest type less than max */ | ||
| 1368 |
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4230 | for (j = 0, min = 0; j < q->sub_packets_B; j++) { |
| 1369 | 3525 | value = q->sub_packet_list_B[j].packet->type; | |
| 1370 |
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3525 | if (value > min && value < max) { |
| 1371 | 705 | min = value; | |
| 1372 | 705 | packet = q->sub_packet_list_B[j].packet; | |
| 1373 | } | ||
| 1374 | } | ||
| 1375 | |||
| 1376 | 705 | max = min; | |
| 1377 | |||
| 1378 | /* check for errors (?) */ | ||
| 1379 |
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|
705 | if (!packet) |
| 1380 | ✗ | return; | |
| 1381 | |||
| 1382 |
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705 | if (i == 0 && |
| 1383 |
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141 | (packet->type < 16 || packet->type >= 48 || |
| 1384 |
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141 | fft_subpackets[packet->type - 16])) |
| 1385 | ✗ | return; | |
| 1386 | |||
| 1387 | /* decode FFT tones */ | ||
| 1388 | 705 | init_get_bits(&gb, packet->data, packet->size * 8); | |
| 1389 | |||
| 1390 |
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|
705 | if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) |
| 1391 | ✗ | unknown_flag = 1; | |
| 1392 | else | ||
| 1393 | 705 | unknown_flag = 0; | |
| 1394 | |||
| 1395 | 705 | type = packet->type; | |
| 1396 | |||
| 1397 |
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1410 | if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { |
| 1398 | 705 | int duration = q->sub_sampling + 5 - (type & 15); | |
| 1399 | |||
| 1400 |
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705 | if (duration >= 0 && duration < 4) |
| 1401 | 564 | qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); | |
| 1402 | ✗ | } else if (type == 31) { | |
| 1403 | ✗ | for (j = 0; j < 4; j++) | |
| 1404 | ✗ | qdm2_fft_decode_tones(q, j, &gb, unknown_flag); | |
| 1405 | ✗ | } else if (type == 46) { | |
| 1406 | ✗ | for (j = 0; j < 6; j++) | |
| 1407 | ✗ | q->fft_level_exp[j] = get_bits(&gb, 6); | |
| 1408 | ✗ | for (j = 0; j < 4; j++) | |
| 1409 | ✗ | qdm2_fft_decode_tones(q, j, &gb, unknown_flag); | |
| 1410 | } | ||
| 1411 | } // Loop on B packets | ||
| 1412 | |||
| 1413 | /* calculate maximum indexes for FFT coefficients */ | ||
| 1414 |
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846 | for (i = 0, j = -1; i < 5; i++) |
| 1415 |
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|
705 | if (q->fft_coefs_min_index[i] >= 0) { |
| 1416 |
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551 | if (j >= 0) |
| 1417 | 410 | q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; | |
| 1418 | 551 | j = i; | |
| 1419 | } | ||
| 1420 |
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141 | if (j >= 0) |
| 1421 | 141 | q->fft_coefs_max_index[j] = q->fft_coefs_index; | |
| 1422 | } | ||
| 1423 | |||
| 1424 | 341696 | static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone) | |
| 1425 | { | ||
| 1426 | float level, f[6]; | ||
| 1427 | int i; | ||
| 1428 | AVComplexFloat c; | ||
| 1429 | 341696 | const double iscale = 2.0 * M_PI / 512.0; | |
| 1430 | |||
| 1431 | 341696 | tone->phase += tone->phase_shift; | |
| 1432 | |||
| 1433 | /* calculate current level (maximum amplitude) of tone */ | ||
| 1434 | 341696 | level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; | |
| 1435 | 341696 | c.im = level * sin(tone->phase * iscale); | |
| 1436 | 341696 | c.re = level * cos(tone->phase * iscale); | |
| 1437 | |||
| 1438 | /* generate FFT coefficients for tone */ | ||
| 1439 |
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|
341696 | if (tone->duration >= 3 || tone->cutoff >= 3) { |
| 1440 | 46655 | tone->complex[0].im += c.im; | |
| 1441 | 46655 | tone->complex[0].re += c.re; | |
| 1442 | 46655 | tone->complex[1].im -= c.im; | |
| 1443 | 46655 | tone->complex[1].re -= c.re; | |
| 1444 | } else { | ||
| 1445 | 295041 | f[1] = -tone->table[4]; | |
| 1446 | 295041 | f[0] = tone->table[3] - tone->table[0]; | |
| 1447 | 295041 | f[2] = 1.0 - tone->table[2] - tone->table[3]; | |
| 1448 | 295041 | f[3] = tone->table[1] + tone->table[4] - 1.0; | |
| 1449 | 295041 | f[4] = tone->table[0] - tone->table[1]; | |
| 1450 | 295041 | f[5] = tone->table[2]; | |
| 1451 |
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885123 | for (i = 0; i < 2; i++) { |
| 1452 | 590082 | tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += | |
| 1453 | 590082 | c.re * f[i]; | |
| 1454 | 1180164 | tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += | |
| 1455 |
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590082 | c.im * ((tone->cutoff <= i) ? -f[i] : f[i]); |
| 1456 | } | ||
| 1457 |
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1475205 | for (i = 0; i < 4; i++) { |
| 1458 | 1180164 | tone->complex[i].re += c.re * f[i + 2]; | |
| 1459 | 1180164 | tone->complex[i].im += c.im * f[i + 2]; | |
| 1460 | } | ||
| 1461 | } | ||
| 1462 | |||
| 1463 | /* copy the tone if it has not yet died out */ | ||
| 1464 |
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341696 | if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { |
| 1465 | 321878 | memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); | |
| 1466 | 321878 | q->fft_tone_end = (q->fft_tone_end + 1) % 1000; | |
| 1467 | } | ||
| 1468 | 341696 | } | |
| 1469 | |||
| 1470 | 2256 | static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet) | |
| 1471 | { | ||
| 1472 | int i, j, ch; | ||
| 1473 | 2256 | const double iscale = 0.25 * M_PI; | |
| 1474 | |||
| 1475 |
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6768 | for (ch = 0; ch < q->channels; ch++) { |
| 1476 | 4512 | memset(q->fft.complex[ch], 0, q->fft_size * sizeof(AVComplexFloat)); | |
| 1477 | } | ||
| 1478 | |||
| 1479 | |||
| 1480 | /* apply FFT tones with duration 4 (1 FFT period) */ | ||
| 1481 |
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2256 | if (q->fft_coefs_min_index[4] >= 0) |
| 1482 |
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8 | for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { |
| 1483 | float level; | ||
| 1484 | AVComplexFloat c; | ||
| 1485 | |||
| 1486 | ✗ | if (q->fft_coefs[i].sub_packet != sub_packet) | |
| 1487 | ✗ | break; | |
| 1488 | |||
| 1489 | ✗ | ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; | |
| 1490 | ✗ | 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]; | |
| 1491 | |||
| 1492 | ✗ | c.re = level * cos(q->fft_coefs[i].phase * iscale); | |
| 1493 | ✗ | c.im = level * sin(q->fft_coefs[i].phase * iscale); | |
| 1494 | ✗ | q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; | |
| 1495 | ✗ | q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; | |
| 1496 | ✗ | q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; | |
| 1497 | ✗ | q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; | |
| 1498 | } | ||
| 1499 | |||
| 1500 | /* generate existing FFT tones */ | ||
| 1501 |
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323747 | for (i = q->fft_tone_end; i != q->fft_tone_start; ) { |
| 1502 | 321491 | qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); | |
| 1503 | 321491 | q->fft_tone_start = (q->fft_tone_start + 1) % 1000; | |
| 1504 | } | ||
| 1505 | |||
| 1506 | /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ | ||
| 1507 |
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11280 | for (i = 0; i < 4; i++) |
| 1508 |
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|
9024 | if (q->fft_coefs_min_index[i] >= 0) { |
| 1509 |
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|
29029 | for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { |
| 1510 | int offset, four_i; | ||
| 1511 | FFTTone tone; | ||
| 1512 | |||
| 1513 |
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|
24549 | if (q->fft_coefs[j].sub_packet != sub_packet) |
| 1514 | 4344 | break; | |
| 1515 | |||
| 1516 | 20205 | four_i = (4 - i); | |
| 1517 | 20205 | offset = q->fft_coefs[j].offset >> four_i; | |
| 1518 |
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20205 | ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; |
| 1519 | |||
| 1520 |
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|
20205 | if (offset < q->frequency_range) { |
| 1521 |
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20205 | if (offset < 2) |
| 1522 | 2795 | tone.cutoff = offset; | |
| 1523 | else | ||
| 1524 |
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17410 | tone.cutoff = (offset >= 60) ? 3 : 2; |
| 1525 | |||
| 1526 |
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20205 | 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]; |
| 1527 | 20205 | tone.complex = &q->fft.complex[ch][offset]; | |
| 1528 | 20205 | tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; | |
| 1529 | 20205 | tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; | |
| 1530 | 20205 | tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); | |
| 1531 | 20205 | tone.duration = i; | |
| 1532 | 20205 | tone.time_index = 0; | |
| 1533 | |||
| 1534 | 20205 | qdm2_fft_generate_tone(q, &tone); | |
| 1535 | } | ||
| 1536 | } | ||
| 1537 | 8824 | q->fft_coefs_min_index[i] = j; | |
| 1538 | } | ||
| 1539 | 2256 | } | |
| 1540 | |||
| 1541 | 4512 | static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet) | |
| 1542 | { | ||
| 1543 |
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4512 | const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; |
| 1544 | 4512 | float *out = q->output_buffer + channel; | |
| 1545 | |||
| 1546 | 4512 | q->fft.complex[channel][0].re *= 2.0f; | |
| 1547 | 4512 | q->fft.complex[channel][0].im = 0.0f; | |
| 1548 | 4512 | q->fft.complex[channel][q->fft_size].re = 0.0f; | |
| 1549 | 4512 | q->fft.complex[channel][q->fft_size].im = 0.0f; | |
| 1550 | |||
| 1551 | 4512 | q->rdft_fn(q->rdft_ctx, q->fft.temp[channel], q->fft.complex[channel], | |
| 1552 | sizeof(AVComplexFloat)); | ||
| 1553 | |||
| 1554 | /* add samples to output buffer */ | ||
| 1555 |
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1159584 | for (int i = 0; i < FFALIGN(q->fft_size, 8); i++) { |
| 1556 | 1155072 | out[0] += q->fft.temp[channel][i].re * gain; | |
| 1557 | 1155072 | out[q->channels] += q->fft.temp[channel][i].im * gain; | |
| 1558 | 1155072 | out += 2 * q->channels; | |
| 1559 | } | ||
| 1560 | 4512 | } | |
| 1561 | |||
| 1562 | /** | ||
| 1563 | * @param q context | ||
| 1564 | * @param index subpacket number | ||
| 1565 | */ | ||
| 1566 | 2256 | static void qdm2_synthesis_filter(QDM2Context *q, int index) | |
| 1567 | { | ||
| 1568 | 2256 | int i, k, ch, sb_used, sub_sampling, dither_state = 0; | |
| 1569 | |||
| 1570 | /* copy sb_samples */ | ||
| 1571 |
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|
2256 | sb_used = QDM2_SB_USED(q->sub_sampling); |
| 1572 | |||
| 1573 |
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|
6768 | for (ch = 0; ch < q->channels; ch++) |
| 1574 |
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40608 | for (i = 0; i < 8; i++) |
| 1575 |
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108288 | for (k = sb_used; k < SBLIMIT; k++) |
| 1576 | 72192 | q->sb_samples[ch][(8 * index) + i][k] = 0; | |
| 1577 | |||
| 1578 |
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|
6768 | for (ch = 0; ch < q->nb_channels; ch++) { |
| 1579 | 4512 | float *samples_ptr = q->samples + ch; | |
| 1580 | |||
| 1581 |
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|
40608 | for (i = 0; i < 8; i++) { |
| 1582 | 36096 | ff_mpa_synth_filter_float(&q->mpadsp, | |
| 1583 | 36096 | q->synth_buf[ch], &(q->synth_buf_offset[ch]), | |
| 1584 | ff_mpa_synth_window_float, &dither_state, | ||
| 1585 | 36096 | samples_ptr, q->nb_channels, | |
| 1586 | 36096 | q->sb_samples[ch][(8 * index) + i]); | |
| 1587 | 36096 | samples_ptr += 32 * q->nb_channels; | |
| 1588 | } | ||
| 1589 | } | ||
| 1590 | |||
| 1591 | /* add samples to output buffer */ | ||
| 1592 | 2256 | sub_sampling = (4 >> q->sub_sampling); | |
| 1593 | |||
| 1594 |
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|
6768 | for (ch = 0; ch < q->channels; ch++) |
| 1595 |
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|
1159584 | for (i = 0; i < q->frame_size; i++) |
| 1596 | 1155072 | q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch]; | |
| 1597 | 2256 | } | |
| 1598 | |||
| 1599 | /** | ||
| 1600 | * Init static data (does not depend on specific file) | ||
| 1601 | */ | ||
| 1602 | 3 | static av_cold void qdm2_init_static_data(void) { | |
| 1603 | 3 | qdm2_init_vlc(); | |
| 1604 | 3 | softclip_table_init(); | |
| 1605 | 3 | rnd_table_init(); | |
| 1606 | 3 | init_noise_samples(); | |
| 1607 | |||
| 1608 | 3 | ff_mpa_synth_init_float(); | |
| 1609 | 3 | } | |
| 1610 | |||
| 1611 | /** | ||
| 1612 | * Init parameters from codec extradata | ||
| 1613 | */ | ||
| 1614 | 4 | static av_cold int qdm2_decode_init(AVCodecContext *avctx) | |
| 1615 | { | ||
| 1616 | static AVOnce init_static_once = AV_ONCE_INIT; | ||
| 1617 | 4 | QDM2Context *s = avctx->priv_data; | |
| 1618 | int ret, tmp_val, tmp, size; | ||
| 1619 | 4 | float scale = 1.0f / 2.0f; | |
| 1620 | GetByteContext gb; | ||
| 1621 | |||
| 1622 | /* extradata parsing | ||
| 1623 | |||
| 1624 | Structure: | ||
| 1625 | wave { | ||
| 1626 | frma (QDM2) | ||
| 1627 | QDCA | ||
| 1628 | QDCP | ||
| 1629 | } | ||
| 1630 | |||
| 1631 | 32 size (including this field) | ||
| 1632 | 32 tag (=frma) | ||
| 1633 | 32 type (=QDM2 or QDMC) | ||
| 1634 | |||
| 1635 | 32 size (including this field, in bytes) | ||
| 1636 | 32 tag (=QDCA) // maybe mandatory parameters | ||
| 1637 | 32 unknown (=1) | ||
| 1638 | 32 channels (=2) | ||
| 1639 | 32 samplerate (=44100) | ||
| 1640 | 32 bitrate (=96000) | ||
| 1641 | 32 block size (=4096) | ||
| 1642 | 32 frame size (=256) (for one channel) | ||
| 1643 | 32 packet size (=1300) | ||
| 1644 | |||
| 1645 | 32 size (including this field, in bytes) | ||
| 1646 | 32 tag (=QDCP) // maybe some tuneable parameters | ||
| 1647 | 32 float1 (=1.0) | ||
| 1648 | 32 zero ? | ||
| 1649 | 32 float2 (=1.0) | ||
| 1650 | 32 float3 (=1.0) | ||
| 1651 | 32 unknown (27) | ||
| 1652 | 32 unknown (8) | ||
| 1653 | 32 zero ? | ||
| 1654 | */ | ||
| 1655 | |||
| 1656 |
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4 | if (!avctx->extradata || (avctx->extradata_size < 48)) { |
| 1657 | ✗ | av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); | |
| 1658 | ✗ | return AVERROR_INVALIDDATA; | |
| 1659 | } | ||
| 1660 | |||
| 1661 | 4 | bytestream2_init(&gb, avctx->extradata, avctx->extradata_size); | |
| 1662 | |||
| 1663 |
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20 | while (bytestream2_get_bytes_left(&gb) > 8) { |
| 1664 |
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|
20 | if (bytestream2_peek_be64u(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) | |
| 1665 | (uint64_t)MKBETAG('Q','D','M','2'))) | ||
| 1666 | 4 | break; | |
| 1667 | 16 | bytestream2_skipu(&gb, 1); | |
| 1668 | } | ||
| 1669 | |||
| 1670 |
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4 | if (bytestream2_get_bytes_left(&gb) < 44) { |
| 1671 | ✗ | av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", | |
| 1672 | bytestream2_get_bytes_left(&gb)); | ||
| 1673 | ✗ | return AVERROR_INVALIDDATA; | |
| 1674 | } | ||
| 1675 | |||
| 1676 | 4 | bytestream2_skipu(&gb, 8); | |
| 1677 | 4 | size = bytestream2_get_be32u(&gb); | |
| 1678 | |||
| 1679 |
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4 | if (size > bytestream2_get_bytes_left(&gb)) { |
| 1680 | ✗ | av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", | |
| 1681 | bytestream2_get_bytes_left(&gb), size); | ||
| 1682 | ✗ | return AVERROR_INVALIDDATA; | |
| 1683 | } | ||
| 1684 | |||
| 1685 | 4 | av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); | |
| 1686 |
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|
4 | if (bytestream2_get_be32u(&gb) != MKBETAG('Q','D','C','A')) { |
| 1687 | ✗ | av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); | |
| 1688 | ✗ | return AVERROR_INVALIDDATA; | |
| 1689 | } | ||
| 1690 | |||
| 1691 | 4 | bytestream2_skipu(&gb, 4); | |
| 1692 | |||
| 1693 | 4 | s->nb_channels = s->channels = bytestream2_get_be32u(&gb); | |
| 1694 |
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4 | if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) { |
| 1695 | ✗ | av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n"); | |
| 1696 | ✗ | return AVERROR_INVALIDDATA; | |
| 1697 | } | ||
| 1698 | 4 | av_channel_layout_uninit(&avctx->ch_layout); | |
| 1699 | 4 | av_channel_layout_default(&avctx->ch_layout, s->channels); | |
| 1700 | |||
| 1701 | 4 | avctx->sample_rate = bytestream2_get_be32u(&gb); | |
| 1702 | 4 | avctx->bit_rate = bytestream2_get_be32u(&gb); | |
| 1703 | 4 | s->group_size = bytestream2_get_be32u(&gb); | |
| 1704 | 4 | s->fft_size = bytestream2_get_be32u(&gb); | |
| 1705 | 4 | s->checksum_size = bytestream2_get_be32u(&gb); | |
| 1706 |
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4 | if (s->checksum_size >= 1U << 28 || s->checksum_size <= 1) { |
| 1707 | ✗ | av_log(avctx, AV_LOG_ERROR, "data block size invalid (%u)\n", s->checksum_size); | |
| 1708 | ✗ | return AVERROR_INVALIDDATA; | |
| 1709 | } | ||
| 1710 | |||
| 1711 | 4 | s->fft_order = av_log2(s->fft_size) + 1; | |
| 1712 | |||
| 1713 | // Fail on unknown fft order | ||
| 1714 |
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4 | if ((s->fft_order < 7) || (s->fft_order > 9)) { |
| 1715 | ✗ | avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order); | |
| 1716 | ✗ | return AVERROR_PATCHWELCOME; | |
| 1717 | } | ||
| 1718 | |||
| 1719 | // something like max decodable tones | ||
| 1720 | 4 | s->group_order = av_log2(s->group_size) + 1; | |
| 1721 | 4 | s->frame_size = s->group_size / 16; // 16 iterations per super block | |
| 1722 | |||
| 1723 |
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4 | if (s->frame_size > QDM2_MAX_FRAME_SIZE) |
| 1724 | ✗ | return AVERROR_INVALIDDATA; | |
| 1725 | |||
| 1726 | 4 | s->sub_sampling = s->fft_order - 7; | |
| 1727 | 4 | s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); | |
| 1728 | |||
| 1729 |
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4 | if (s->frame_size * 4 >> s->sub_sampling > MPA_FRAME_SIZE) { |
| 1730 | ✗ | avpriv_request_sample(avctx, "large frames"); | |
| 1731 | ✗ | return AVERROR_PATCHWELCOME; | |
| 1732 | } | ||
| 1733 | |||
| 1734 |
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4 | switch ((s->sub_sampling * 2 + s->channels - 1)) { |
| 1735 | ✗ | case 0: tmp = 40; break; | |
| 1736 | ✗ | case 1: tmp = 48; break; | |
| 1737 | ✗ | case 2: tmp = 56; break; | |
| 1738 | ✗ | case 3: tmp = 72; break; | |
| 1739 | ✗ | case 4: tmp = 80; break; | |
| 1740 | 4 | case 5: tmp = 100;break; | |
| 1741 | ✗ | default: tmp=s->sub_sampling; break; | |
| 1742 | } | ||
| 1743 | 4 | tmp_val = 0; | |
| 1744 |
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4 | if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; |
| 1745 |
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4 | if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; |
| 1746 |
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4 | if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; |
| 1747 |
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4 | if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; |
| 1748 | 4 | s->cm_table_select = tmp_val; | |
| 1749 | |||
| 1750 |
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4 | if (avctx->bit_rate <= 8000) |
| 1751 | ✗ | s->coeff_per_sb_select = 0; | |
| 1752 |
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4 | else if (avctx->bit_rate < 16000) |
| 1753 | ✗ | s->coeff_per_sb_select = 1; | |
| 1754 | else | ||
| 1755 | 4 | s->coeff_per_sb_select = 2; | |
| 1756 | |||
| 1757 |
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4 | if (s->fft_size != (1 << (s->fft_order - 1))) { |
| 1758 | ✗ | av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size); | |
| 1759 | ✗ | return AVERROR_INVALIDDATA; | |
| 1760 | } | ||
| 1761 | |||
| 1762 | 4 | ret = av_tx_init(&s->rdft_ctx, &s->rdft_fn, AV_TX_FLOAT_RDFT, 1, 2*s->fft_size, &scale, 0); | |
| 1763 |
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4 | if (ret < 0) |
| 1764 | ✗ | return ret; | |
| 1765 | |||
| 1766 | 4 | ff_mpadsp_init(&s->mpadsp); | |
| 1767 | |||
| 1768 | 4 | avctx->sample_fmt = AV_SAMPLE_FMT_S16; | |
| 1769 | |||
| 1770 | 4 | ff_thread_once(&init_static_once, qdm2_init_static_data); | |
| 1771 | |||
| 1772 | 4 | return 0; | |
| 1773 | } | ||
| 1774 | |||
| 1775 | 4 | static av_cold int qdm2_decode_close(AVCodecContext *avctx) | |
| 1776 | { | ||
| 1777 | 4 | QDM2Context *s = avctx->priv_data; | |
| 1778 | |||
| 1779 | 4 | av_tx_uninit(&s->rdft_ctx); | |
| 1780 | |||
| 1781 | 4 | return 0; | |
| 1782 | } | ||
| 1783 | |||
| 1784 | 2256 | static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out) | |
| 1785 | { | ||
| 1786 | int ch, i; | ||
| 1787 | 2256 | const int frame_size = (q->frame_size * q->channels); | |
| 1788 | |||
| 1789 |
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2256 | if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2) |
| 1790 | ✗ | return -1; | |
| 1791 | |||
| 1792 | /* select input buffer */ | ||
| 1793 | 2256 | q->compressed_data = in; | |
| 1794 | 2256 | q->compressed_size = q->checksum_size; | |
| 1795 | |||
| 1796 | /* copy old block, clear new block of output samples */ | ||
| 1797 | 2256 | memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); | |
| 1798 | 2256 | memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); | |
| 1799 | |||
| 1800 | /* decode block of QDM2 compressed data */ | ||
| 1801 |
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2256 | if (q->sub_packet == 0) { |
| 1802 | 141 | q->has_errors = 0; // zero it for a new super block | |
| 1803 | 141 | av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); | |
| 1804 | 141 | qdm2_decode_super_block(q); | |
| 1805 | } | ||
| 1806 | |||
| 1807 | /* parse subpackets */ | ||
| 1808 |
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2256 | if (!q->has_errors) { |
| 1809 |
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2256 | if (q->sub_packet == 2) |
| 1810 | 141 | qdm2_decode_fft_packets(q); | |
| 1811 | |||
| 1812 | 2256 | qdm2_fft_tone_synthesizer(q, q->sub_packet); | |
| 1813 | } | ||
| 1814 | |||
| 1815 | /* sound synthesis stage 1 (FFT) */ | ||
| 1816 |
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6768 | for (ch = 0; ch < q->channels; ch++) { |
| 1817 | 4512 | qdm2_calculate_fft(q, ch, q->sub_packet); | |
| 1818 | |||
| 1819 |
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4512 | if (!q->has_errors && q->sub_packet_list_C[0].packet) { |
| 1820 | ✗ | SAMPLES_NEEDED_2("has errors, and C list is not empty") | |
| 1821 | ✗ | return -1; | |
| 1822 | } | ||
| 1823 | } | ||
| 1824 | |||
| 1825 | /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ | ||
| 1826 |
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2256 | if (!q->has_errors && q->do_synth_filter) |
| 1827 | 2256 | qdm2_synthesis_filter(q, q->sub_packet); | |
| 1828 | |||
| 1829 | 2256 | q->sub_packet = (q->sub_packet + 1) % 16; | |
| 1830 | |||
| 1831 | /* clip and convert output float[] to 16-bit signed samples */ | ||
| 1832 |
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1157328 | for (i = 0; i < frame_size; i++) { |
| 1833 | 1155072 | int value = (int)q->output_buffer[i]; | |
| 1834 | |||
| 1835 |
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1155072 | if (value > SOFTCLIP_THRESHOLD) |
| 1836 |
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245 | value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; |
| 1837 |
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1154827 | else if (value < -SOFTCLIP_THRESHOLD) |
| 1838 |
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839 | value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; |
| 1839 | |||
| 1840 | 1155072 | out[i] = value; | |
| 1841 | } | ||
| 1842 | |||
| 1843 | 2256 | return 0; | |
| 1844 | } | ||
| 1845 | |||
| 1846 | 141 | static int qdm2_decode_frame(AVCodecContext *avctx, AVFrame *frame, | |
| 1847 | int *got_frame_ptr, AVPacket *avpkt) | ||
| 1848 | { | ||
| 1849 | 141 | const uint8_t *buf = avpkt->data; | |
| 1850 | 141 | int buf_size = avpkt->size; | |
| 1851 | 141 | QDM2Context *s = avctx->priv_data; | |
| 1852 | int16_t *out; | ||
| 1853 | int i, ret; | ||
| 1854 | |||
| 1855 |
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141 | if(!buf) |
| 1856 | ✗ | return 0; | |
| 1857 |
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141 | if(buf_size < s->checksum_size) |
| 1858 | ✗ | return -1; | |
| 1859 | |||
| 1860 | /* get output buffer */ | ||
| 1861 | 141 | frame->nb_samples = 16 * s->frame_size; | |
| 1862 |
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141 | if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
| 1863 | ✗ | return ret; | |
| 1864 | 141 | out = (int16_t *)frame->data[0]; | |
| 1865 | |||
| 1866 |
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2397 | for (i = 0; i < 16; i++) { |
| 1867 |
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✓ Branch 2 taken 2256 times.
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2256 | if ((ret = qdm2_decode(s, buf, out)) < 0) |
| 1868 | ✗ | return ret; | |
| 1869 | 2256 | out += s->channels * s->frame_size; | |
| 1870 | } | ||
| 1871 | |||
| 1872 | 141 | *got_frame_ptr = 1; | |
| 1873 | |||
| 1874 | 141 | return s->checksum_size; | |
| 1875 | } | ||
| 1876 | |||
| 1877 | const FFCodec ff_qdm2_decoder = { | ||
| 1878 | .p.name = "qdm2", | ||
| 1879 | CODEC_LONG_NAME("QDesign Music Codec 2"), | ||
| 1880 | .p.type = AVMEDIA_TYPE_AUDIO, | ||
| 1881 | .p.id = AV_CODEC_ID_QDM2, | ||
| 1882 | .priv_data_size = sizeof(QDM2Context), | ||
| 1883 | .init = qdm2_decode_init, | ||
| 1884 | .close = qdm2_decode_close, | ||
| 1885 | FF_CODEC_DECODE_CB(qdm2_decode_frame), | ||
| 1886 | .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, | ||
| 1887 | }; | ||
| 1888 |