Libav 0.7.1
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00001 /* 00002 * QDM2 compatible decoder 00003 * Copyright (c) 2003 Ewald Snel 00004 * Copyright (c) 2005 Benjamin Larsson 00005 * Copyright (c) 2005 Alex Beregszaszi 00006 * Copyright (c) 2005 Roberto Togni 00007 * 00008 * This file is part of Libav. 00009 * 00010 * Libav is free software; you can redistribute it and/or 00011 * modify it under the terms of the GNU Lesser General Public 00012 * License as published by the Free Software Foundation; either 00013 * version 2.1 of the License, or (at your option) any later version. 00014 * 00015 * Libav is distributed in the hope that it will be useful, 00016 * but WITHOUT ANY WARRANTY; without even the implied warranty of 00017 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 00018 * Lesser General Public License for more details. 00019 * 00020 * You should have received a copy of the GNU Lesser General Public 00021 * License along with Libav; if not, write to the Free Software 00022 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 00023 */ 00024 00033 #include <math.h> 00034 #include <stddef.h> 00035 #include <stdio.h> 00036 00037 #define ALT_BITSTREAM_READER_LE 00038 #include "avcodec.h" 00039 #include "get_bits.h" 00040 #include "dsputil.h" 00041 #include "rdft.h" 00042 #include "mpegaudiodsp.h" 00043 #include "mpegaudio.h" 00044 00045 #include "qdm2data.h" 00046 #include "qdm2_tablegen.h" 00047 00048 #undef NDEBUG 00049 #include <assert.h> 00050 00051 00052 #define QDM2_LIST_ADD(list, size, packet) \ 00053 do { \ 00054 if (size > 0) { \ 00055 list[size - 1].next = &list[size]; \ 00056 } \ 00057 list[size].packet = packet; \ 00058 list[size].next = NULL; \ 00059 size++; \ 00060 } while(0) 00061 00062 // Result is 8, 16 or 30 00063 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling)) 00064 00065 #define FIX_NOISE_IDX(noise_idx) \ 00066 if ((noise_idx) >= 3840) \ 00067 (noise_idx) -= 3840; \ 00068 00069 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)]) 00070 00071 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb))) 00072 00073 #define SAMPLES_NEEDED \ 00074 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); 00075 00076 #define SAMPLES_NEEDED_2(why) \ 00077 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); 00078 00079 #define QDM2_MAX_FRAME_SIZE 512 00080 00081 typedef int8_t sb_int8_array[2][30][64]; 00082 00086 typedef struct { 00087 int type; 00088 unsigned int size; 00089 const uint8_t *data; 00090 } QDM2SubPacket; 00091 00095 typedef struct QDM2SubPNode { 00096 QDM2SubPacket *packet; 00097 struct QDM2SubPNode *next; 00098 } QDM2SubPNode; 00099 00100 typedef struct { 00101 float re; 00102 float im; 00103 } QDM2Complex; 00104 00105 typedef struct { 00106 float level; 00107 QDM2Complex *complex; 00108 const float *table; 00109 int phase; 00110 int phase_shift; 00111 int duration; 00112 short time_index; 00113 short cutoff; 00114 } FFTTone; 00115 00116 typedef struct { 00117 int16_t sub_packet; 00118 uint8_t channel; 00119 int16_t offset; 00120 int16_t exp; 00121 uint8_t phase; 00122 } FFTCoefficient; 00123 00124 typedef struct { 00125 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; 00126 } QDM2FFT; 00127 00131 typedef struct { 00133 int nb_channels; 00134 int channels; 00135 int group_size; 00136 int fft_size; 00137 int checksum_size; 00138 00140 int group_order; 00141 int fft_order; 00142 int fft_frame_size; 00143 int frame_size; 00144 int frequency_range; 00145 int sub_sampling; 00146 int coeff_per_sb_select; 00147 int cm_table_select; 00148 00150 QDM2SubPacket sub_packets[16]; 00151 QDM2SubPNode sub_packet_list_A[16]; 00152 QDM2SubPNode sub_packet_list_B[16]; 00153 int sub_packets_B; 00154 QDM2SubPNode sub_packet_list_C[16]; 00155 QDM2SubPNode sub_packet_list_D[16]; 00156 00158 FFTTone fft_tones[1000]; 00159 int fft_tone_start; 00160 int fft_tone_end; 00161 FFTCoefficient fft_coefs[1000]; 00162 int fft_coefs_index; 00163 int fft_coefs_min_index[5]; 00164 int fft_coefs_max_index[5]; 00165 int fft_level_exp[6]; 00166 RDFTContext rdft_ctx; 00167 QDM2FFT fft; 00168 00170 const uint8_t *compressed_data; 00171 int compressed_size; 00172 float output_buffer[QDM2_MAX_FRAME_SIZE * 2]; 00173 00175 MPADSPContext mpadsp; 00176 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2]; 00177 int synth_buf_offset[MPA_MAX_CHANNELS]; 00178 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; 00179 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; 00180 00182 float tone_level[MPA_MAX_CHANNELS][30][64]; 00183 int8_t coding_method[MPA_MAX_CHANNELS][30][64]; 00184 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; 00185 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; 00186 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; 00187 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; 00188 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; 00189 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; 00190 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; 00191 00192 // Flags 00193 int has_errors; 00194 int superblocktype_2_3; 00195 int do_synth_filter; 00196 00197 int sub_packet; 00198 int noise_idx; 00199 } QDM2Context; 00200 00201 00202 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE]; 00203 00204 static VLC vlc_tab_level; 00205 static VLC vlc_tab_diff; 00206 static VLC vlc_tab_run; 00207 static VLC fft_level_exp_alt_vlc; 00208 static VLC fft_level_exp_vlc; 00209 static VLC fft_stereo_exp_vlc; 00210 static VLC fft_stereo_phase_vlc; 00211 static VLC vlc_tab_tone_level_idx_hi1; 00212 static VLC vlc_tab_tone_level_idx_mid; 00213 static VLC vlc_tab_tone_level_idx_hi2; 00214 static VLC vlc_tab_type30; 00215 static VLC vlc_tab_type34; 00216 static VLC vlc_tab_fft_tone_offset[5]; 00217 00218 static const uint16_t qdm2_vlc_offs[] = { 00219 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, 00220 }; 00221 00222 static av_cold void qdm2_init_vlc(void) 00223 { 00224 static int vlcs_initialized = 0; 00225 static VLC_TYPE qdm2_table[3838][2]; 00226 00227 if (!vlcs_initialized) { 00228 00229 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; 00230 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; 00231 init_vlc (&vlc_tab_level, 8, 24, 00232 vlc_tab_level_huffbits, 1, 1, 00233 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00234 00235 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; 00236 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; 00237 init_vlc (&vlc_tab_diff, 8, 37, 00238 vlc_tab_diff_huffbits, 1, 1, 00239 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00240 00241 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; 00242 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; 00243 init_vlc (&vlc_tab_run, 5, 6, 00244 vlc_tab_run_huffbits, 1, 1, 00245 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00246 00247 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; 00248 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3]; 00249 init_vlc (&fft_level_exp_alt_vlc, 8, 28, 00250 fft_level_exp_alt_huffbits, 1, 1, 00251 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00252 00253 00254 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; 00255 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; 00256 init_vlc (&fft_level_exp_vlc, 8, 20, 00257 fft_level_exp_huffbits, 1, 1, 00258 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00259 00260 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; 00261 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5]; 00262 init_vlc (&fft_stereo_exp_vlc, 6, 7, 00263 fft_stereo_exp_huffbits, 1, 1, 00264 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00265 00266 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; 00267 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6]; 00268 init_vlc (&fft_stereo_phase_vlc, 6, 9, 00269 fft_stereo_phase_huffbits, 1, 1, 00270 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00271 00272 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]]; 00273 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7]; 00274 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, 00275 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, 00276 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00277 00278 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]]; 00279 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8]; 00280 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, 00281 vlc_tab_tone_level_idx_mid_huffbits, 1, 1, 00282 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00283 00284 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]]; 00285 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9]; 00286 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, 00287 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, 00288 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00289 00290 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; 00291 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; 00292 init_vlc (&vlc_tab_type30, 6, 9, 00293 vlc_tab_type30_huffbits, 1, 1, 00294 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00295 00296 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; 00297 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; 00298 init_vlc (&vlc_tab_type34, 5, 10, 00299 vlc_tab_type34_huffbits, 1, 1, 00300 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00301 00302 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]]; 00303 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12]; 00304 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, 00305 vlc_tab_fft_tone_offset_0_huffbits, 1, 1, 00306 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00307 00308 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]]; 00309 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13]; 00310 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, 00311 vlc_tab_fft_tone_offset_1_huffbits, 1, 1, 00312 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00313 00314 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]]; 00315 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14]; 00316 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, 00317 vlc_tab_fft_tone_offset_2_huffbits, 1, 1, 00318 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00319 00320 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]]; 00321 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15]; 00322 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, 00323 vlc_tab_fft_tone_offset_3_huffbits, 1, 1, 00324 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00325 00326 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]]; 00327 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16]; 00328 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, 00329 vlc_tab_fft_tone_offset_4_huffbits, 1, 1, 00330 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00331 00332 vlcs_initialized=1; 00333 } 00334 } 00335 00336 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) 00337 { 00338 int value; 00339 00340 value = get_vlc2(gb, vlc->table, vlc->bits, depth); 00341 00342 /* stage-2, 3 bits exponent escape sequence */ 00343 if (value-- == 0) 00344 value = get_bits (gb, get_bits (gb, 3) + 1); 00345 00346 /* stage-3, optional */ 00347 if (flag) { 00348 int tmp = vlc_stage3_values[value]; 00349 00350 if ((value & ~3) > 0) 00351 tmp += get_bits (gb, (value >> 2)); 00352 value = tmp; 00353 } 00354 00355 return value; 00356 } 00357 00358 00359 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) 00360 { 00361 int value = qdm2_get_vlc (gb, vlc, 0, depth); 00362 00363 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); 00364 } 00365 00366 00376 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) { 00377 int i; 00378 00379 for (i=0; i < length; i++) 00380 value -= data[i]; 00381 00382 return (uint16_t)(value & 0xffff); 00383 } 00384 00385 00392 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) 00393 { 00394 sub_packet->type = get_bits (gb, 8); 00395 00396 if (sub_packet->type == 0) { 00397 sub_packet->size = 0; 00398 sub_packet->data = NULL; 00399 } else { 00400 sub_packet->size = get_bits (gb, 8); 00401 00402 if (sub_packet->type & 0x80) { 00403 sub_packet->size <<= 8; 00404 sub_packet->size |= get_bits (gb, 8); 00405 sub_packet->type &= 0x7f; 00406 } 00407 00408 if (sub_packet->type == 0x7f) 00409 sub_packet->type |= (get_bits (gb, 8) << 8); 00410 00411 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data 00412 } 00413 00414 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", 00415 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); 00416 } 00417 00418 00426 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) 00427 { 00428 while (list != NULL && list->packet != NULL) { 00429 if (list->packet->type == type) 00430 return list; 00431 list = list->next; 00432 } 00433 return NULL; 00434 } 00435 00436 00443 static void average_quantized_coeffs (QDM2Context *q) 00444 { 00445 int i, j, n, ch, sum; 00446 00447 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; 00448 00449 for (ch = 0; ch < q->nb_channels; ch++) 00450 for (i = 0; i < n; i++) { 00451 sum = 0; 00452 00453 for (j = 0; j < 8; j++) 00454 sum += q->quantized_coeffs[ch][i][j]; 00455 00456 sum /= 8; 00457 if (sum > 0) 00458 sum--; 00459 00460 for (j=0; j < 8; j++) 00461 q->quantized_coeffs[ch][i][j] = sum; 00462 } 00463 } 00464 00465 00473 static void build_sb_samples_from_noise (QDM2Context *q, int sb) 00474 { 00475 int ch, j; 00476 00477 FIX_NOISE_IDX(q->noise_idx); 00478 00479 if (!q->nb_channels) 00480 return; 00481 00482 for (ch = 0; ch < q->nb_channels; ch++) 00483 for (j = 0; j < 64; j++) { 00484 q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; 00485 q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; 00486 } 00487 } 00488 00489 00498 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) 00499 { 00500 int j,k; 00501 int ch; 00502 int run, case_val; 00503 int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4}; 00504 00505 for (ch = 0; ch < channels; ch++) { 00506 for (j = 0; j < 64; ) { 00507 if((coding_method[ch][sb][j] - 8) > 22) { 00508 run = 1; 00509 case_val = 8; 00510 } else { 00511 switch (switchtable[coding_method[ch][sb][j]-8]) { 00512 case 0: run = 10; case_val = 10; break; 00513 case 1: run = 1; case_val = 16; break; 00514 case 2: run = 5; case_val = 24; break; 00515 case 3: run = 3; case_val = 30; break; 00516 case 4: run = 1; case_val = 30; break; 00517 case 5: run = 1; case_val = 8; break; 00518 default: run = 1; case_val = 8; break; 00519 } 00520 } 00521 for (k = 0; k < run; k++) 00522 if (j + k < 128) 00523 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) 00524 if (k > 0) { 00525 SAMPLES_NEEDED 00526 //not debugged, almost never used 00527 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t)); 00528 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); 00529 } 00530 j += run; 00531 } 00532 } 00533 } 00534 00535 00543 static void fill_tone_level_array (QDM2Context *q, int flag) 00544 { 00545 int i, sb, ch, sb_used; 00546 int tmp, tab; 00547 00548 // This should never happen 00549 if (q->nb_channels <= 0) 00550 return; 00551 00552 for (ch = 0; ch < q->nb_channels; ch++) 00553 for (sb = 0; sb < 30; sb++) 00554 for (i = 0; i < 8; i++) { 00555 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) 00556 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ 00557 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 00558 else 00559 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 00560 if(tmp < 0) 00561 tmp += 0xff; 00562 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; 00563 } 00564 00565 sb_used = QDM2_SB_USED(q->sub_sampling); 00566 00567 if ((q->superblocktype_2_3 != 0) && !flag) { 00568 for (sb = 0; sb < sb_used; sb++) 00569 for (ch = 0; ch < q->nb_channels; ch++) 00570 for (i = 0; i < 64; i++) { 00571 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 00572 if (q->tone_level_idx[ch][sb][i] < 0) 00573 q->tone_level[ch][sb][i] = 0; 00574 else 00575 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; 00576 } 00577 } else { 00578 tab = q->superblocktype_2_3 ? 0 : 1; 00579 for (sb = 0; sb < sb_used; sb++) { 00580 if ((sb >= 4) && (sb <= 23)) { 00581 for (ch = 0; ch < q->nb_channels; ch++) 00582 for (i = 0; i < 64; i++) { 00583 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 00584 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - 00585 q->tone_level_idx_mid[ch][sb - 4][i / 8] - 00586 q->tone_level_idx_hi2[ch][sb - 4]; 00587 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 00588 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 00589 q->tone_level[ch][sb][i] = 0; 00590 else 00591 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 00592 } 00593 } else { 00594 if (sb > 4) { 00595 for (ch = 0; ch < q->nb_channels; ch++) 00596 for (i = 0; i < 64; i++) { 00597 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 00598 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - 00599 q->tone_level_idx_hi2[ch][sb - 4]; 00600 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 00601 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 00602 q->tone_level[ch][sb][i] = 0; 00603 else 00604 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 00605 } 00606 } else { 00607 for (ch = 0; ch < q->nb_channels; ch++) 00608 for (i = 0; i < 64; i++) { 00609 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 00610 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 00611 q->tone_level[ch][sb][i] = 0; 00612 else 00613 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 00614 } 00615 } 00616 } 00617 } 00618 } 00619 00620 return; 00621 } 00622 00623 00638 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, 00639 sb_int8_array coding_method, int nb_channels, 00640 int c, int superblocktype_2_3, int cm_table_select) 00641 { 00642 int ch, sb, j; 00643 int tmp, acc, esp_40, comp; 00644 int add1, add2, add3, add4; 00645 int64_t multres; 00646 00647 // This should never happen 00648 if (nb_channels <= 0) 00649 return; 00650 00651 if (!superblocktype_2_3) { 00652 /* This case is untested, no samples available */ 00653 SAMPLES_NEEDED 00654 for (ch = 0; ch < nb_channels; ch++) 00655 for (sb = 0; sb < 30; sb++) { 00656 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer 00657 add1 = tone_level_idx[ch][sb][j] - 10; 00658 if (add1 < 0) 00659 add1 = 0; 00660 add2 = add3 = add4 = 0; 00661 if (sb > 1) { 00662 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; 00663 if (add2 < 0) 00664 add2 = 0; 00665 } 00666 if (sb > 0) { 00667 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; 00668 if (add3 < 0) 00669 add3 = 0; 00670 } 00671 if (sb < 29) { 00672 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; 00673 if (add4 < 0) 00674 add4 = 0; 00675 } 00676 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; 00677 if (tmp < 0) 00678 tmp = 0; 00679 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; 00680 } 00681 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; 00682 } 00683 acc = 0; 00684 for (ch = 0; ch < nb_channels; ch++) 00685 for (sb = 0; sb < 30; sb++) 00686 for (j = 0; j < 64; j++) 00687 acc += tone_level_idx_temp[ch][sb][j]; 00688 00689 multres = 0x66666667 * (acc * 10); 00690 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); 00691 for (ch = 0; ch < nb_channels; ch++) 00692 for (sb = 0; sb < 30; sb++) 00693 for (j = 0; j < 64; j++) { 00694 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; 00695 if (comp < 0) 00696 comp += 0xff; 00697 comp /= 256; // signed shift 00698 switch(sb) { 00699 case 0: 00700 if (comp < 30) 00701 comp = 30; 00702 comp += 15; 00703 break; 00704 case 1: 00705 if (comp < 24) 00706 comp = 24; 00707 comp += 10; 00708 break; 00709 case 2: 00710 case 3: 00711 case 4: 00712 if (comp < 16) 00713 comp = 16; 00714 } 00715 if (comp <= 5) 00716 tmp = 0; 00717 else if (comp <= 10) 00718 tmp = 10; 00719 else if (comp <= 16) 00720 tmp = 16; 00721 else if (comp <= 24) 00722 tmp = -1; 00723 else 00724 tmp = 0; 00725 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; 00726 } 00727 for (sb = 0; sb < 30; sb++) 00728 fix_coding_method_array(sb, nb_channels, coding_method); 00729 for (ch = 0; ch < nb_channels; ch++) 00730 for (sb = 0; sb < 30; sb++) 00731 for (j = 0; j < 64; j++) 00732 if (sb >= 10) { 00733 if (coding_method[ch][sb][j] < 10) 00734 coding_method[ch][sb][j] = 10; 00735 } else { 00736 if (sb >= 2) { 00737 if (coding_method[ch][sb][j] < 16) 00738 coding_method[ch][sb][j] = 16; 00739 } else { 00740 if (coding_method[ch][sb][j] < 30) 00741 coding_method[ch][sb][j] = 30; 00742 } 00743 } 00744 } else { // superblocktype_2_3 != 0 00745 for (ch = 0; ch < nb_channels; ch++) 00746 for (sb = 0; sb < 30; sb++) 00747 for (j = 0; j < 64; j++) 00748 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; 00749 } 00750 00751 return; 00752 } 00753 00754 00766 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) 00767 { 00768 int sb, j, k, n, ch, run, channels; 00769 int joined_stereo, zero_encoding, chs; 00770 int type34_first; 00771 float type34_div = 0; 00772 float type34_predictor; 00773 float samples[10], sign_bits[16]; 00774 00775 if (length == 0) { 00776 // If no data use noise 00777 for (sb=sb_min; sb < sb_max; sb++) 00778 build_sb_samples_from_noise (q, sb); 00779 00780 return; 00781 } 00782 00783 for (sb = sb_min; sb < sb_max; sb++) { 00784 FIX_NOISE_IDX(q->noise_idx); 00785 00786 channels = q->nb_channels; 00787 00788 if (q->nb_channels <= 1 || sb < 12) 00789 joined_stereo = 0; 00790 else if (sb >= 24) 00791 joined_stereo = 1; 00792 else 00793 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0; 00794 00795 if (joined_stereo) { 00796 if (BITS_LEFT(length,gb) >= 16) 00797 for (j = 0; j < 16; j++) 00798 sign_bits[j] = get_bits1 (gb); 00799 00800 for (j = 0; j < 64; j++) 00801 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) 00802 q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; 00803 00804 fix_coding_method_array(sb, q->nb_channels, q->coding_method); 00805 channels = 1; 00806 } 00807 00808 for (ch = 0; ch < channels; ch++) { 00809 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; 00810 type34_predictor = 0.0; 00811 type34_first = 1; 00812 00813 for (j = 0; j < 128; ) { 00814 switch (q->coding_method[ch][sb][j / 2]) { 00815 case 8: 00816 if (BITS_LEFT(length,gb) >= 10) { 00817 if (zero_encoding) { 00818 for (k = 0; k < 5; k++) { 00819 if ((j + 2 * k) >= 128) 00820 break; 00821 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; 00822 } 00823 } else { 00824 n = get_bits(gb, 8); 00825 for (k = 0; k < 5; k++) 00826 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 00827 } 00828 for (k = 0; k < 5; k++) 00829 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); 00830 } else { 00831 for (k = 0; k < 10; k++) 00832 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 00833 } 00834 run = 10; 00835 break; 00836 00837 case 10: 00838 if (BITS_LEFT(length,gb) >= 1) { 00839 float f = 0.81; 00840 00841 if (get_bits1(gb)) 00842 f = -f; 00843 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; 00844 samples[0] = f; 00845 } else { 00846 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00847 } 00848 run = 1; 00849 break; 00850 00851 case 16: 00852 if (BITS_LEFT(length,gb) >= 10) { 00853 if (zero_encoding) { 00854 for (k = 0; k < 5; k++) { 00855 if ((j + k) >= 128) 00856 break; 00857 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; 00858 } 00859 } else { 00860 n = get_bits (gb, 8); 00861 for (k = 0; k < 5; k++) 00862 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 00863 } 00864 } else { 00865 for (k = 0; k < 5; k++) 00866 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 00867 } 00868 run = 5; 00869 break; 00870 00871 case 24: 00872 if (BITS_LEFT(length,gb) >= 7) { 00873 n = get_bits(gb, 7); 00874 for (k = 0; k < 3; k++) 00875 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; 00876 } else { 00877 for (k = 0; k < 3; k++) 00878 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 00879 } 00880 run = 3; 00881 break; 00882 00883 case 30: 00884 if (BITS_LEFT(length,gb) >= 4) { 00885 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1); 00886 if (index < FF_ARRAY_ELEMS(type30_dequant)) { 00887 samples[0] = type30_dequant[index]; 00888 } else 00889 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00890 } else 00891 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00892 00893 run = 1; 00894 break; 00895 00896 case 34: 00897 if (BITS_LEFT(length,gb) >= 7) { 00898 if (type34_first) { 00899 type34_div = (float)(1 << get_bits(gb, 2)); 00900 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; 00901 type34_predictor = samples[0]; 00902 type34_first = 0; 00903 } else { 00904 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1); 00905 if (index < FF_ARRAY_ELEMS(type34_delta)) { 00906 samples[0] = type34_delta[index] / type34_div + type34_predictor; 00907 type34_predictor = samples[0]; 00908 } else 00909 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00910 } 00911 } else { 00912 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00913 } 00914 run = 1; 00915 break; 00916 00917 default: 00918 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00919 run = 1; 00920 break; 00921 } 00922 00923 if (joined_stereo) { 00924 float tmp[10][MPA_MAX_CHANNELS]; 00925 00926 for (k = 0; k < run; k++) { 00927 tmp[k][0] = samples[k]; 00928 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; 00929 } 00930 for (chs = 0; chs < q->nb_channels; chs++) 00931 for (k = 0; k < run; k++) 00932 if ((j + k) < 128) 00933 q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs]; 00934 } else { 00935 for (k = 0; k < run; k++) 00936 if ((j + k) < 128) 00937 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k]; 00938 } 00939 00940 j += run; 00941 } // j loop 00942 } // channel loop 00943 } // subband loop 00944 } 00945 00946 00956 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) 00957 { 00958 int i, k, run, level, diff; 00959 00960 if (BITS_LEFT(length,gb) < 16) 00961 return; 00962 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); 00963 00964 quantized_coeffs[0] = level; 00965 00966 for (i = 0; i < 7; ) { 00967 if (BITS_LEFT(length,gb) < 16) 00968 break; 00969 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; 00970 00971 if (BITS_LEFT(length,gb) < 16) 00972 break; 00973 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); 00974 00975 for (k = 1; k <= run; k++) 00976 quantized_coeffs[i + k] = (level + ((k * diff) / run)); 00977 00978 level += diff; 00979 i += run; 00980 } 00981 } 00982 00983 00993 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) 00994 { 00995 int sb, j, k, n, ch; 00996 00997 for (ch = 0; ch < q->nb_channels; ch++) { 00998 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length); 00999 01000 if (BITS_LEFT(length,gb) < 16) { 01001 memset(q->quantized_coeffs[ch][0], 0, 8); 01002 break; 01003 } 01004 } 01005 01006 n = q->sub_sampling + 1; 01007 01008 for (sb = 0; sb < n; sb++) 01009 for (ch = 0; ch < q->nb_channels; ch++) 01010 for (j = 0; j < 8; j++) { 01011 if (BITS_LEFT(length,gb) < 1) 01012 break; 01013 if (get_bits1(gb)) { 01014 for (k=0; k < 8; k++) { 01015 if (BITS_LEFT(length,gb) < 16) 01016 break; 01017 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); 01018 } 01019 } else { 01020 for (k=0; k < 8; k++) 01021 q->tone_level_idx_hi1[ch][sb][j][k] = 0; 01022 } 01023 } 01024 01025 n = QDM2_SB_USED(q->sub_sampling) - 4; 01026 01027 for (sb = 0; sb < n; sb++) 01028 for (ch = 0; ch < q->nb_channels; ch++) { 01029 if (BITS_LEFT(length,gb) < 16) 01030 break; 01031 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); 01032 if (sb > 19) 01033 q->tone_level_idx_hi2[ch][sb] -= 16; 01034 else 01035 for (j = 0; j < 8; j++) 01036 q->tone_level_idx_mid[ch][sb][j] = -16; 01037 } 01038 01039 n = QDM2_SB_USED(q->sub_sampling) - 5; 01040 01041 for (sb = 0; sb < n; sb++) 01042 for (ch = 0; ch < q->nb_channels; ch++) 01043 for (j = 0; j < 8; j++) { 01044 if (BITS_LEFT(length,gb) < 16) 01045 break; 01046 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; 01047 } 01048 } 01049 01056 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) 01057 { 01058 GetBitContext gb; 01059 int i, j, k, n, ch, run, level, diff; 01060 01061 init_get_bits(&gb, node->packet->data, node->packet->size*8); 01062 01063 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function 01064 01065 for (i = 1; i < n; i++) 01066 for (ch=0; ch < q->nb_channels; ch++) { 01067 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); 01068 q->quantized_coeffs[ch][i][0] = level; 01069 01070 for (j = 0; j < (8 - 1); ) { 01071 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; 01072 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); 01073 01074 for (k = 1; k <= run; k++) 01075 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); 01076 01077 level += diff; 01078 j += run; 01079 } 01080 } 01081 01082 for (ch = 0; ch < q->nb_channels; ch++) 01083 for (i = 0; i < 8; i++) 01084 q->quantized_coeffs[ch][0][i] = 0; 01085 } 01086 01087 01095 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) 01096 { 01097 GetBitContext gb; 01098 01099 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 01100 01101 if (length != 0) { 01102 init_tone_level_dequantization(q, &gb, length); 01103 fill_tone_level_array(q, 1); 01104 } else { 01105 fill_tone_level_array(q, 0); 01106 } 01107 } 01108 01109 01117 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length) 01118 { 01119 GetBitContext gb; 01120 01121 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 01122 if (length >= 32) { 01123 int c = get_bits (&gb, 13); 01124 01125 if (c > 3) 01126 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, 01127 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select); 01128 } 01129 01130 synthfilt_build_sb_samples(q, &gb, length, 0, 8); 01131 } 01132 01133 01141 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) 01142 { 01143 GetBitContext gb; 01144 01145 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 01146 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); 01147 } 01148 01149 /* 01150 * Process new subpackets for synthesis filter 01151 * 01152 * @param q context 01153 * @param list list with synthesis filter packets (list D) 01154 */ 01155 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) 01156 { 01157 QDM2SubPNode *nodes[4]; 01158 01159 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); 01160 if (nodes[0] != NULL) 01161 process_subpacket_9(q, nodes[0]); 01162 01163 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); 01164 if (nodes[1] != NULL) 01165 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3); 01166 else 01167 process_subpacket_10(q, NULL, 0); 01168 01169 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); 01170 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) 01171 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3)); 01172 else 01173 process_subpacket_11(q, NULL, 0); 01174 01175 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); 01176 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) 01177 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3)); 01178 else 01179 process_subpacket_12(q, NULL, 0); 01180 } 01181 01182 01183 /* 01184 * Decode superblock, fill packet lists. 01185 * 01186 * @param q context 01187 */ 01188 static void qdm2_decode_super_block (QDM2Context *q) 01189 { 01190 GetBitContext gb; 01191 QDM2SubPacket header, *packet; 01192 int i, packet_bytes, sub_packet_size, sub_packets_D; 01193 unsigned int next_index = 0; 01194 01195 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); 01196 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); 01197 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); 01198 01199 q->sub_packets_B = 0; 01200 sub_packets_D = 0; 01201 01202 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] 01203 01204 init_get_bits(&gb, q->compressed_data, q->compressed_size*8); 01205 qdm2_decode_sub_packet_header(&gb, &header); 01206 01207 if (header.type < 2 || header.type >= 8) { 01208 q->has_errors = 1; 01209 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); 01210 return; 01211 } 01212 01213 q->superblocktype_2_3 = (header.type == 2 || header.type == 3); 01214 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); 01215 01216 init_get_bits(&gb, header.data, header.size*8); 01217 01218 if (header.type == 2 || header.type == 4 || header.type == 5) { 01219 int csum = 257 * get_bits(&gb, 8); 01220 csum += 2 * get_bits(&gb, 8); 01221 01222 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); 01223 01224 if (csum != 0) { 01225 q->has_errors = 1; 01226 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); 01227 return; 01228 } 01229 } 01230 01231 q->sub_packet_list_B[0].packet = NULL; 01232 q->sub_packet_list_D[0].packet = NULL; 01233 01234 for (i = 0; i < 6; i++) 01235 if (--q->fft_level_exp[i] < 0) 01236 q->fft_level_exp[i] = 0; 01237 01238 for (i = 0; packet_bytes > 0; i++) { 01239 int j; 01240 01241 q->sub_packet_list_A[i].next = NULL; 01242 01243 if (i > 0) { 01244 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; 01245 01246 /* seek to next block */ 01247 init_get_bits(&gb, header.data, header.size*8); 01248 skip_bits(&gb, next_index*8); 01249 01250 if (next_index >= header.size) 01251 break; 01252 } 01253 01254 /* decode subpacket */ 01255 packet = &q->sub_packets[i]; 01256 qdm2_decode_sub_packet_header(&gb, packet); 01257 next_index = packet->size + get_bits_count(&gb) / 8; 01258 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; 01259 01260 if (packet->type == 0) 01261 break; 01262 01263 if (sub_packet_size > packet_bytes) { 01264 if (packet->type != 10 && packet->type != 11 && packet->type != 12) 01265 break; 01266 packet->size += packet_bytes - sub_packet_size; 01267 } 01268 01269 packet_bytes -= sub_packet_size; 01270 01271 /* add subpacket to 'all subpackets' list */ 01272 q->sub_packet_list_A[i].packet = packet; 01273 01274 /* add subpacket to related list */ 01275 if (packet->type == 8) { 01276 SAMPLES_NEEDED_2("packet type 8"); 01277 return; 01278 } else if (packet->type >= 9 && packet->type <= 12) { 01279 /* packets for MPEG Audio like Synthesis Filter */ 01280 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); 01281 } else if (packet->type == 13) { 01282 for (j = 0; j < 6; j++) 01283 q->fft_level_exp[j] = get_bits(&gb, 6); 01284 } else if (packet->type == 14) { 01285 for (j = 0; j < 6; j++) 01286 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); 01287 } else if (packet->type == 15) { 01288 SAMPLES_NEEDED_2("packet type 15") 01289 return; 01290 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { 01291 /* packets for FFT */ 01292 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); 01293 } 01294 } // Packet bytes loop 01295 01296 /* **************************************************************** */ 01297 if (q->sub_packet_list_D[0].packet != NULL) { 01298 process_synthesis_subpackets(q, q->sub_packet_list_D); 01299 q->do_synth_filter = 1; 01300 } else if (q->do_synth_filter) { 01301 process_subpacket_10(q, NULL, 0); 01302 process_subpacket_11(q, NULL, 0); 01303 process_subpacket_12(q, NULL, 0); 01304 } 01305 /* **************************************************************** */ 01306 } 01307 01308 01309 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, 01310 int offset, int duration, int channel, 01311 int exp, int phase) 01312 { 01313 if (q->fft_coefs_min_index[duration] < 0) 01314 q->fft_coefs_min_index[duration] = q->fft_coefs_index; 01315 01316 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); 01317 q->fft_coefs[q->fft_coefs_index].channel = channel; 01318 q->fft_coefs[q->fft_coefs_index].offset = offset; 01319 q->fft_coefs[q->fft_coefs_index].exp = exp; 01320 q->fft_coefs[q->fft_coefs_index].phase = phase; 01321 q->fft_coefs_index++; 01322 } 01323 01324 01325 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) 01326 { 01327 int channel, stereo, phase, exp; 01328 int local_int_4, local_int_8, stereo_phase, local_int_10; 01329 int local_int_14, stereo_exp, local_int_20, local_int_28; 01330 int n, offset; 01331 01332 local_int_4 = 0; 01333 local_int_28 = 0; 01334 local_int_20 = 2; 01335 local_int_8 = (4 - duration); 01336 local_int_10 = 1 << (q->group_order - duration - 1); 01337 offset = 1; 01338 01339 while (1) { 01340 if (q->superblocktype_2_3) { 01341 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { 01342 offset = 1; 01343 if (n == 0) { 01344 local_int_4 += local_int_10; 01345 local_int_28 += (1 << local_int_8); 01346 } else { 01347 local_int_4 += 8*local_int_10; 01348 local_int_28 += (8 << local_int_8); 01349 } 01350 } 01351 offset += (n - 2); 01352 } else { 01353 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); 01354 while (offset >= (local_int_10 - 1)) { 01355 offset += (1 - (local_int_10 - 1)); 01356 local_int_4 += local_int_10; 01357 local_int_28 += (1 << local_int_8); 01358 } 01359 } 01360 01361 if (local_int_4 >= q->group_size) 01362 return; 01363 01364 local_int_14 = (offset >> local_int_8); 01365 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table)) 01366 return; 01367 01368 if (q->nb_channels > 1) { 01369 channel = get_bits1(gb); 01370 stereo = get_bits1(gb); 01371 } else { 01372 channel = 0; 01373 stereo = 0; 01374 } 01375 01376 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); 01377 exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; 01378 exp = (exp < 0) ? 0 : exp; 01379 01380 phase = get_bits(gb, 3); 01381 stereo_exp = 0; 01382 stereo_phase = 0; 01383 01384 if (stereo) { 01385 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); 01386 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); 01387 if (stereo_phase < 0) 01388 stereo_phase += 8; 01389 } 01390 01391 if (q->frequency_range > (local_int_14 + 1)) { 01392 int sub_packet = (local_int_20 + local_int_28); 01393 01394 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); 01395 if (stereo) 01396 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); 01397 } 01398 01399 offset++; 01400 } 01401 } 01402 01403 01404 static void qdm2_decode_fft_packets (QDM2Context *q) 01405 { 01406 int i, j, min, max, value, type, unknown_flag; 01407 GetBitContext gb; 01408 01409 if (q->sub_packet_list_B[0].packet == NULL) 01410 return; 01411 01412 /* reset minimum indexes for FFT coefficients */ 01413 q->fft_coefs_index = 0; 01414 for (i=0; i < 5; i++) 01415 q->fft_coefs_min_index[i] = -1; 01416 01417 /* process subpackets ordered by type, largest type first */ 01418 for (i = 0, max = 256; i < q->sub_packets_B; i++) { 01419 QDM2SubPacket *packet= NULL; 01420 01421 /* find subpacket with largest type less than max */ 01422 for (j = 0, min = 0; j < q->sub_packets_B; j++) { 01423 value = q->sub_packet_list_B[j].packet->type; 01424 if (value > min && value < max) { 01425 min = value; 01426 packet = q->sub_packet_list_B[j].packet; 01427 } 01428 } 01429 01430 max = min; 01431 01432 /* check for errors (?) */ 01433 if (!packet) 01434 return; 01435 01436 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) 01437 return; 01438 01439 /* decode FFT tones */ 01440 init_get_bits (&gb, packet->data, packet->size*8); 01441 01442 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) 01443 unknown_flag = 1; 01444 else 01445 unknown_flag = 0; 01446 01447 type = packet->type; 01448 01449 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { 01450 int duration = q->sub_sampling + 5 - (type & 15); 01451 01452 if (duration >= 0 && duration < 4) 01453 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); 01454 } else if (type == 31) { 01455 for (j=0; j < 4; j++) 01456 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 01457 } else if (type == 46) { 01458 for (j=0; j < 6; j++) 01459 q->fft_level_exp[j] = get_bits(&gb, 6); 01460 for (j=0; j < 4; j++) 01461 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 01462 } 01463 } // Loop on B packets 01464 01465 /* calculate maximum indexes for FFT coefficients */ 01466 for (i = 0, j = -1; i < 5; i++) 01467 if (q->fft_coefs_min_index[i] >= 0) { 01468 if (j >= 0) 01469 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; 01470 j = i; 01471 } 01472 if (j >= 0) 01473 q->fft_coefs_max_index[j] = q->fft_coefs_index; 01474 } 01475 01476 01477 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) 01478 { 01479 float level, f[6]; 01480 int i; 01481 QDM2Complex c; 01482 const double iscale = 2.0*M_PI / 512.0; 01483 01484 tone->phase += tone->phase_shift; 01485 01486 /* calculate current level (maximum amplitude) of tone */ 01487 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; 01488 c.im = level * sin(tone->phase*iscale); 01489 c.re = level * cos(tone->phase*iscale); 01490 01491 /* generate FFT coefficients for tone */ 01492 if (tone->duration >= 3 || tone->cutoff >= 3) { 01493 tone->complex[0].im += c.im; 01494 tone->complex[0].re += c.re; 01495 tone->complex[1].im -= c.im; 01496 tone->complex[1].re -= c.re; 01497 } else { 01498 f[1] = -tone->table[4]; 01499 f[0] = tone->table[3] - tone->table[0]; 01500 f[2] = 1.0 - tone->table[2] - tone->table[3]; 01501 f[3] = tone->table[1] + tone->table[4] - 1.0; 01502 f[4] = tone->table[0] - tone->table[1]; 01503 f[5] = tone->table[2]; 01504 for (i = 0; i < 2; i++) { 01505 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; 01506 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); 01507 } 01508 for (i = 0; i < 4; i++) { 01509 tone->complex[i].re += c.re * f[i+2]; 01510 tone->complex[i].im += c.im * f[i+2]; 01511 } 01512 } 01513 01514 /* copy the tone if it has not yet died out */ 01515 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { 01516 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); 01517 q->fft_tone_end = (q->fft_tone_end + 1) % 1000; 01518 } 01519 } 01520 01521 01522 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) 01523 { 01524 int i, j, ch; 01525 const double iscale = 0.25 * M_PI; 01526 01527 for (ch = 0; ch < q->channels; ch++) { 01528 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); 01529 } 01530 01531 01532 /* apply FFT tones with duration 4 (1 FFT period) */ 01533 if (q->fft_coefs_min_index[4] >= 0) 01534 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { 01535 float level; 01536 QDM2Complex c; 01537 01538 if (q->fft_coefs[i].sub_packet != sub_packet) 01539 break; 01540 01541 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; 01542 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]; 01543 01544 c.re = level * cos(q->fft_coefs[i].phase * iscale); 01545 c.im = level * sin(q->fft_coefs[i].phase * iscale); 01546 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; 01547 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; 01548 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; 01549 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; 01550 } 01551 01552 /* generate existing FFT tones */ 01553 for (i = q->fft_tone_end; i != q->fft_tone_start; ) { 01554 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); 01555 q->fft_tone_start = (q->fft_tone_start + 1) % 1000; 01556 } 01557 01558 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ 01559 for (i = 0; i < 4; i++) 01560 if (q->fft_coefs_min_index[i] >= 0) { 01561 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { 01562 int offset, four_i; 01563 FFTTone tone; 01564 01565 if (q->fft_coefs[j].sub_packet != sub_packet) 01566 break; 01567 01568 four_i = (4 - i); 01569 offset = q->fft_coefs[j].offset >> four_i; 01570 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; 01571 01572 if (offset < q->frequency_range) { 01573 if (offset < 2) 01574 tone.cutoff = offset; 01575 else 01576 tone.cutoff = (offset >= 60) ? 3 : 2; 01577 01578 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]; 01579 tone.complex = &q->fft.complex[ch][offset]; 01580 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; 01581 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; 01582 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); 01583 tone.duration = i; 01584 tone.time_index = 0; 01585 01586 qdm2_fft_generate_tone(q, &tone); 01587 } 01588 } 01589 q->fft_coefs_min_index[i] = j; 01590 } 01591 } 01592 01593 01594 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) 01595 { 01596 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; 01597 int i; 01598 q->fft.complex[channel][0].re *= 2.0f; 01599 q->fft.complex[channel][0].im = 0.0f; 01600 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); 01601 /* add samples to output buffer */ 01602 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) 01603 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; 01604 } 01605 01606 01611 static void qdm2_synthesis_filter (QDM2Context *q, int index) 01612 { 01613 int i, k, ch, sb_used, sub_sampling, dither_state = 0; 01614 01615 /* copy sb_samples */ 01616 sb_used = QDM2_SB_USED(q->sub_sampling); 01617 01618 for (ch = 0; ch < q->channels; ch++) 01619 for (i = 0; i < 8; i++) 01620 for (k=sb_used; k < SBLIMIT; k++) 01621 q->sb_samples[ch][(8 * index) + i][k] = 0; 01622 01623 for (ch = 0; ch < q->nb_channels; ch++) { 01624 float *samples_ptr = q->samples + ch; 01625 01626 for (i = 0; i < 8; i++) { 01627 ff_mpa_synth_filter_float(&q->mpadsp, 01628 q->synth_buf[ch], &(q->synth_buf_offset[ch]), 01629 ff_mpa_synth_window_float, &dither_state, 01630 samples_ptr, q->nb_channels, 01631 q->sb_samples[ch][(8 * index) + i]); 01632 samples_ptr += 32 * q->nb_channels; 01633 } 01634 } 01635 01636 /* add samples to output buffer */ 01637 sub_sampling = (4 >> q->sub_sampling); 01638 01639 for (ch = 0; ch < q->channels; ch++) 01640 for (i = 0; i < q->frame_size; i++) 01641 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch]; 01642 } 01643 01644 01650 static av_cold void qdm2_init(QDM2Context *q) { 01651 static int initialized = 0; 01652 01653 if (initialized != 0) 01654 return; 01655 initialized = 1; 01656 01657 qdm2_init_vlc(); 01658 ff_mpa_synth_init_float(ff_mpa_synth_window_float); 01659 softclip_table_init(); 01660 rnd_table_init(); 01661 init_noise_samples(); 01662 01663 av_log(NULL, AV_LOG_DEBUG, "init done\n"); 01664 } 01665 01666 01667 #if 0 01668 static void dump_context(QDM2Context *q) 01669 { 01670 int i; 01671 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b); 01672 PRINT("compressed_data",q->compressed_data); 01673 PRINT("compressed_size",q->compressed_size); 01674 PRINT("frame_size",q->frame_size); 01675 PRINT("checksum_size",q->checksum_size); 01676 PRINT("channels",q->channels); 01677 PRINT("nb_channels",q->nb_channels); 01678 PRINT("fft_frame_size",q->fft_frame_size); 01679 PRINT("fft_size",q->fft_size); 01680 PRINT("sub_sampling",q->sub_sampling); 01681 PRINT("fft_order",q->fft_order); 01682 PRINT("group_order",q->group_order); 01683 PRINT("group_size",q->group_size); 01684 PRINT("sub_packet",q->sub_packet); 01685 PRINT("frequency_range",q->frequency_range); 01686 PRINT("has_errors",q->has_errors); 01687 PRINT("fft_tone_end",q->fft_tone_end); 01688 PRINT("fft_tone_start",q->fft_tone_start); 01689 PRINT("fft_coefs_index",q->fft_coefs_index); 01690 PRINT("coeff_per_sb_select",q->coeff_per_sb_select); 01691 PRINT("cm_table_select",q->cm_table_select); 01692 PRINT("noise_idx",q->noise_idx); 01693 01694 for (i = q->fft_tone_start; i < q->fft_tone_end; i++) 01695 { 01696 FFTTone *t = &q->fft_tones[i]; 01697 01698 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i); 01699 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level); 01700 // PRINT(" level", t->level); 01701 PRINT(" phase", t->phase); 01702 PRINT(" phase_shift", t->phase_shift); 01703 PRINT(" duration", t->duration); 01704 PRINT(" samples_im", t->samples_im); 01705 PRINT(" samples_re", t->samples_re); 01706 PRINT(" table", t->table); 01707 } 01708 01709 } 01710 #endif 01711 01712 01716 static av_cold int qdm2_decode_init(AVCodecContext *avctx) 01717 { 01718 QDM2Context *s = avctx->priv_data; 01719 uint8_t *extradata; 01720 int extradata_size; 01721 int tmp_val, tmp, size; 01722 01723 /* extradata parsing 01724 01725 Structure: 01726 wave { 01727 frma (QDM2) 01728 QDCA 01729 QDCP 01730 } 01731 01732 32 size (including this field) 01733 32 tag (=frma) 01734 32 type (=QDM2 or QDMC) 01735 01736 32 size (including this field, in bytes) 01737 32 tag (=QDCA) // maybe mandatory parameters 01738 32 unknown (=1) 01739 32 channels (=2) 01740 32 samplerate (=44100) 01741 32 bitrate (=96000) 01742 32 block size (=4096) 01743 32 frame size (=256) (for one channel) 01744 32 packet size (=1300) 01745 01746 32 size (including this field, in bytes) 01747 32 tag (=QDCP) // maybe some tuneable parameters 01748 32 float1 (=1.0) 01749 32 zero ? 01750 32 float2 (=1.0) 01751 32 float3 (=1.0) 01752 32 unknown (27) 01753 32 unknown (8) 01754 32 zero ? 01755 */ 01756 01757 if (!avctx->extradata || (avctx->extradata_size < 48)) { 01758 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); 01759 return -1; 01760 } 01761 01762 extradata = avctx->extradata; 01763 extradata_size = avctx->extradata_size; 01764 01765 while (extradata_size > 7) { 01766 if (!memcmp(extradata, "frmaQDM", 7)) 01767 break; 01768 extradata++; 01769 extradata_size--; 01770 } 01771 01772 if (extradata_size < 12) { 01773 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", 01774 extradata_size); 01775 return -1; 01776 } 01777 01778 if (memcmp(extradata, "frmaQDM", 7)) { 01779 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); 01780 return -1; 01781 } 01782 01783 if (extradata[7] == 'C') { 01784 // s->is_qdmc = 1; 01785 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); 01786 return -1; 01787 } 01788 01789 extradata += 8; 01790 extradata_size -= 8; 01791 01792 size = AV_RB32(extradata); 01793 01794 if(size > extradata_size){ 01795 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", 01796 extradata_size, size); 01797 return -1; 01798 } 01799 01800 extradata += 4; 01801 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); 01802 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { 01803 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); 01804 return -1; 01805 } 01806 01807 extradata += 8; 01808 01809 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); 01810 extradata += 4; 01811 if (s->channels > MPA_MAX_CHANNELS) 01812 return AVERROR_INVALIDDATA; 01813 01814 avctx->sample_rate = AV_RB32(extradata); 01815 extradata += 4; 01816 01817 avctx->bit_rate = AV_RB32(extradata); 01818 extradata += 4; 01819 01820 s->group_size = AV_RB32(extradata); 01821 extradata += 4; 01822 01823 s->fft_size = AV_RB32(extradata); 01824 extradata += 4; 01825 01826 s->checksum_size = AV_RB32(extradata); 01827 if (s->checksum_size >= 1U << 28) { 01828 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size); 01829 return AVERROR_INVALIDDATA; 01830 } 01831 01832 s->fft_order = av_log2(s->fft_size) + 1; 01833 s->fft_frame_size = 2 * s->fft_size; // complex has two floats 01834 01835 // something like max decodable tones 01836 s->group_order = av_log2(s->group_size) + 1; 01837 s->frame_size = s->group_size / 16; // 16 iterations per super block 01838 if (s->frame_size > QDM2_MAX_FRAME_SIZE) 01839 return AVERROR_INVALIDDATA; 01840 01841 s->sub_sampling = s->fft_order - 7; 01842 s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); 01843 01844 switch ((s->sub_sampling * 2 + s->channels - 1)) { 01845 case 0: tmp = 40; break; 01846 case 1: tmp = 48; break; 01847 case 2: tmp = 56; break; 01848 case 3: tmp = 72; break; 01849 case 4: tmp = 80; break; 01850 case 5: tmp = 100;break; 01851 default: tmp=s->sub_sampling; break; 01852 } 01853 tmp_val = 0; 01854 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; 01855 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; 01856 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; 01857 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; 01858 s->cm_table_select = tmp_val; 01859 01860 if (s->sub_sampling == 0) 01861 tmp = 7999; 01862 else 01863 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; 01864 /* 01865 0: 7999 -> 0 01866 1: 20000 -> 2 01867 2: 28000 -> 2 01868 */ 01869 if (tmp < 8000) 01870 s->coeff_per_sb_select = 0; 01871 else if (tmp <= 16000) 01872 s->coeff_per_sb_select = 1; 01873 else 01874 s->coeff_per_sb_select = 2; 01875 01876 // Fail on unknown fft order 01877 if ((s->fft_order < 7) || (s->fft_order > 9)) { 01878 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); 01879 return -1; 01880 } 01881 01882 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); 01883 ff_mpadsp_init(&s->mpadsp); 01884 01885 qdm2_init(s); 01886 01887 avctx->sample_fmt = AV_SAMPLE_FMT_S16; 01888 01889 // dump_context(s); 01890 return 0; 01891 } 01892 01893 01894 static av_cold int qdm2_decode_close(AVCodecContext *avctx) 01895 { 01896 QDM2Context *s = avctx->priv_data; 01897 01898 ff_rdft_end(&s->rdft_ctx); 01899 01900 return 0; 01901 } 01902 01903 01904 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) 01905 { 01906 int ch, i; 01907 const int frame_size = (q->frame_size * q->channels); 01908 01909 /* select input buffer */ 01910 q->compressed_data = in; 01911 q->compressed_size = q->checksum_size; 01912 01913 // dump_context(q); 01914 01915 /* copy old block, clear new block of output samples */ 01916 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); 01917 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); 01918 01919 /* decode block of QDM2 compressed data */ 01920 if (q->sub_packet == 0) { 01921 q->has_errors = 0; // zero it for a new super block 01922 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); 01923 qdm2_decode_super_block(q); 01924 } 01925 01926 /* parse subpackets */ 01927 if (!q->has_errors) { 01928 if (q->sub_packet == 2) 01929 qdm2_decode_fft_packets(q); 01930 01931 qdm2_fft_tone_synthesizer(q, q->sub_packet); 01932 } 01933 01934 /* sound synthesis stage 1 (FFT) */ 01935 for (ch = 0; ch < q->channels; ch++) { 01936 qdm2_calculate_fft(q, ch, q->sub_packet); 01937 01938 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { 01939 SAMPLES_NEEDED_2("has errors, and C list is not empty") 01940 return -1; 01941 } 01942 } 01943 01944 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ 01945 if (!q->has_errors && q->do_synth_filter) 01946 qdm2_synthesis_filter(q, q->sub_packet); 01947 01948 q->sub_packet = (q->sub_packet + 1) % 16; 01949 01950 /* clip and convert output float[] to 16bit signed samples */ 01951 for (i = 0; i < frame_size; i++) { 01952 int value = (int)q->output_buffer[i]; 01953 01954 if (value > SOFTCLIP_THRESHOLD) 01955 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; 01956 else if (value < -SOFTCLIP_THRESHOLD) 01957 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; 01958 01959 out[i] = value; 01960 } 01961 01962 return 0; 01963 } 01964 01965 01966 static int qdm2_decode_frame(AVCodecContext *avctx, 01967 void *data, int *data_size, 01968 AVPacket *avpkt) 01969 { 01970 const uint8_t *buf = avpkt->data; 01971 int buf_size = avpkt->size; 01972 QDM2Context *s = avctx->priv_data; 01973 int16_t *out = data; 01974 int i, out_size; 01975 01976 if(!buf) 01977 return 0; 01978 if(buf_size < s->checksum_size) 01979 return -1; 01980 01981 out_size = 16 * s->channels * s->frame_size * 01982 av_get_bytes_per_sample(avctx->sample_fmt); 01983 if (*data_size < out_size) { 01984 av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n"); 01985 return AVERROR(EINVAL); 01986 } 01987 01988 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", 01989 buf_size, buf, s->checksum_size, data, *data_size); 01990 01991 for (i = 0; i < 16; i++) { 01992 if (qdm2_decode(s, buf, out) < 0) 01993 return -1; 01994 out += s->channels * s->frame_size; 01995 } 01996 01997 *data_size = out_size; 01998 01999 return s->checksum_size; 02000 } 02001 02002 AVCodec ff_qdm2_decoder = 02003 { 02004 .name = "qdm2", 02005 .type = AVMEDIA_TYPE_AUDIO, 02006 .id = CODEC_ID_QDM2, 02007 .priv_data_size = sizeof(QDM2Context), 02008 .init = qdm2_decode_init, 02009 .close = qdm2_decode_close, 02010 .decode = qdm2_decode_frame, 02011 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), 02012 };