Libav 0.7.1
|
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 00080 typedef int8_t sb_int8_array[2][30][64]; 00081 00085 typedef struct { 00086 int type; 00087 unsigned int size; 00088 const uint8_t *data; 00089 } QDM2SubPacket; 00090 00094 typedef struct QDM2SubPNode { 00095 QDM2SubPacket *packet; 00096 struct QDM2SubPNode *next; 00097 } QDM2SubPNode; 00098 00099 typedef struct { 00100 float re; 00101 float im; 00102 } QDM2Complex; 00103 00104 typedef struct { 00105 float level; 00106 QDM2Complex *complex; 00107 const float *table; 00108 int phase; 00109 int phase_shift; 00110 int duration; 00111 short time_index; 00112 short cutoff; 00113 } FFTTone; 00114 00115 typedef struct { 00116 int16_t sub_packet; 00117 uint8_t channel; 00118 int16_t offset; 00119 int16_t exp; 00120 uint8_t phase; 00121 } FFTCoefficient; 00122 00123 typedef struct { 00124 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; 00125 } QDM2FFT; 00126 00130 typedef struct { 00132 int nb_channels; 00133 int channels; 00134 int group_size; 00135 int fft_size; 00136 int checksum_size; 00137 00139 int group_order; 00140 int fft_order; 00141 int fft_frame_size; 00142 int frame_size; 00143 int frequency_range; 00144 int sub_sampling; 00145 int coeff_per_sb_select; 00146 int cm_table_select; 00147 00149 QDM2SubPacket sub_packets[16]; 00150 QDM2SubPNode sub_packet_list_A[16]; 00151 QDM2SubPNode sub_packet_list_B[16]; 00152 int sub_packets_B; 00153 QDM2SubPNode sub_packet_list_C[16]; 00154 QDM2SubPNode sub_packet_list_D[16]; 00155 00157 FFTTone fft_tones[1000]; 00158 int fft_tone_start; 00159 int fft_tone_end; 00160 FFTCoefficient fft_coefs[1000]; 00161 int fft_coefs_index; 00162 int fft_coefs_min_index[5]; 00163 int fft_coefs_max_index[5]; 00164 int fft_level_exp[6]; 00165 RDFTContext rdft_ctx; 00166 QDM2FFT fft; 00167 00169 const uint8_t *compressed_data; 00170 int compressed_size; 00171 float output_buffer[1024]; 00172 00174 MPADSPContext mpadsp; 00175 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2]; 00176 int synth_buf_offset[MPA_MAX_CHANNELS]; 00177 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; 00178 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; 00179 00181 float tone_level[MPA_MAX_CHANNELS][30][64]; 00182 int8_t coding_method[MPA_MAX_CHANNELS][30][64]; 00183 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; 00184 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; 00185 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; 00186 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; 00187 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; 00188 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; 00189 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; 00190 00191 // Flags 00192 int has_errors; 00193 int superblocktype_2_3; 00194 int do_synth_filter; 00195 00196 int sub_packet; 00197 int noise_idx; 00198 } QDM2Context; 00199 00200 00201 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE]; 00202 00203 static VLC vlc_tab_level; 00204 static VLC vlc_tab_diff; 00205 static VLC vlc_tab_run; 00206 static VLC fft_level_exp_alt_vlc; 00207 static VLC fft_level_exp_vlc; 00208 static VLC fft_stereo_exp_vlc; 00209 static VLC fft_stereo_phase_vlc; 00210 static VLC vlc_tab_tone_level_idx_hi1; 00211 static VLC vlc_tab_tone_level_idx_mid; 00212 static VLC vlc_tab_tone_level_idx_hi2; 00213 static VLC vlc_tab_type30; 00214 static VLC vlc_tab_type34; 00215 static VLC vlc_tab_fft_tone_offset[5]; 00216 00217 static const uint16_t qdm2_vlc_offs[] = { 00218 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, 00219 }; 00220 00221 static av_cold void qdm2_init_vlc(void) 00222 { 00223 static int vlcs_initialized = 0; 00224 static VLC_TYPE qdm2_table[3838][2]; 00225 00226 if (!vlcs_initialized) { 00227 00228 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; 00229 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; 00230 init_vlc (&vlc_tab_level, 8, 24, 00231 vlc_tab_level_huffbits, 1, 1, 00232 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00233 00234 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; 00235 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; 00236 init_vlc (&vlc_tab_diff, 8, 37, 00237 vlc_tab_diff_huffbits, 1, 1, 00238 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00239 00240 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; 00241 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; 00242 init_vlc (&vlc_tab_run, 5, 6, 00243 vlc_tab_run_huffbits, 1, 1, 00244 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00245 00246 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; 00247 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3]; 00248 init_vlc (&fft_level_exp_alt_vlc, 8, 28, 00249 fft_level_exp_alt_huffbits, 1, 1, 00250 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00251 00252 00253 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; 00254 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; 00255 init_vlc (&fft_level_exp_vlc, 8, 20, 00256 fft_level_exp_huffbits, 1, 1, 00257 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00258 00259 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; 00260 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5]; 00261 init_vlc (&fft_stereo_exp_vlc, 6, 7, 00262 fft_stereo_exp_huffbits, 1, 1, 00263 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00264 00265 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; 00266 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6]; 00267 init_vlc (&fft_stereo_phase_vlc, 6, 9, 00268 fft_stereo_phase_huffbits, 1, 1, 00269 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00270 00271 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]]; 00272 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7]; 00273 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, 00274 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, 00275 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00276 00277 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]]; 00278 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8]; 00279 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, 00280 vlc_tab_tone_level_idx_mid_huffbits, 1, 1, 00281 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00282 00283 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]]; 00284 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9]; 00285 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, 00286 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, 00287 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00288 00289 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; 00290 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; 00291 init_vlc (&vlc_tab_type30, 6, 9, 00292 vlc_tab_type30_huffbits, 1, 1, 00293 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00294 00295 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; 00296 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; 00297 init_vlc (&vlc_tab_type34, 5, 10, 00298 vlc_tab_type34_huffbits, 1, 1, 00299 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00300 00301 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]]; 00302 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12]; 00303 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, 00304 vlc_tab_fft_tone_offset_0_huffbits, 1, 1, 00305 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00306 00307 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]]; 00308 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13]; 00309 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, 00310 vlc_tab_fft_tone_offset_1_huffbits, 1, 1, 00311 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00312 00313 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]]; 00314 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14]; 00315 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, 00316 vlc_tab_fft_tone_offset_2_huffbits, 1, 1, 00317 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00318 00319 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]]; 00320 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15]; 00321 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, 00322 vlc_tab_fft_tone_offset_3_huffbits, 1, 1, 00323 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00324 00325 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]]; 00326 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16]; 00327 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, 00328 vlc_tab_fft_tone_offset_4_huffbits, 1, 1, 00329 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00330 00331 vlcs_initialized=1; 00332 } 00333 } 00334 00335 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) 00336 { 00337 int value; 00338 00339 value = get_vlc2(gb, vlc->table, vlc->bits, depth); 00340 00341 /* stage-2, 3 bits exponent escape sequence */ 00342 if (value-- == 0) 00343 value = get_bits (gb, get_bits (gb, 3) + 1); 00344 00345 /* stage-3, optional */ 00346 if (flag) { 00347 int tmp = vlc_stage3_values[value]; 00348 00349 if ((value & ~3) > 0) 00350 tmp += get_bits (gb, (value >> 2)); 00351 value = tmp; 00352 } 00353 00354 return value; 00355 } 00356 00357 00358 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) 00359 { 00360 int value = qdm2_get_vlc (gb, vlc, 0, depth); 00361 00362 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); 00363 } 00364 00365 00375 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) { 00376 int i; 00377 00378 for (i=0; i < length; i++) 00379 value -= data[i]; 00380 00381 return (uint16_t)(value & 0xffff); 00382 } 00383 00384 00391 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) 00392 { 00393 sub_packet->type = get_bits (gb, 8); 00394 00395 if (sub_packet->type == 0) { 00396 sub_packet->size = 0; 00397 sub_packet->data = NULL; 00398 } else { 00399 sub_packet->size = get_bits (gb, 8); 00400 00401 if (sub_packet->type & 0x80) { 00402 sub_packet->size <<= 8; 00403 sub_packet->size |= get_bits (gb, 8); 00404 sub_packet->type &= 0x7f; 00405 } 00406 00407 if (sub_packet->type == 0x7f) 00408 sub_packet->type |= (get_bits (gb, 8) << 8); 00409 00410 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data 00411 } 00412 00413 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", 00414 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); 00415 } 00416 00417 00425 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) 00426 { 00427 while (list != NULL && list->packet != NULL) { 00428 if (list->packet->type == type) 00429 return list; 00430 list = list->next; 00431 } 00432 return NULL; 00433 } 00434 00435 00442 static void average_quantized_coeffs (QDM2Context *q) 00443 { 00444 int i, j, n, ch, sum; 00445 00446 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; 00447 00448 for (ch = 0; ch < q->nb_channels; ch++) 00449 for (i = 0; i < n; i++) { 00450 sum = 0; 00451 00452 for (j = 0; j < 8; j++) 00453 sum += q->quantized_coeffs[ch][i][j]; 00454 00455 sum /= 8; 00456 if (sum > 0) 00457 sum--; 00458 00459 for (j=0; j < 8; j++) 00460 q->quantized_coeffs[ch][i][j] = sum; 00461 } 00462 } 00463 00464 00472 static void build_sb_samples_from_noise (QDM2Context *q, int sb) 00473 { 00474 int ch, j; 00475 00476 FIX_NOISE_IDX(q->noise_idx); 00477 00478 if (!q->nb_channels) 00479 return; 00480 00481 for (ch = 0; ch < q->nb_channels; ch++) 00482 for (j = 0; j < 64; j++) { 00483 q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; 00484 q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; 00485 } 00486 } 00487 00488 00497 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) 00498 { 00499 int j,k; 00500 int ch; 00501 int run, case_val; 00502 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}; 00503 00504 for (ch = 0; ch < channels; ch++) { 00505 for (j = 0; j < 64; ) { 00506 if((coding_method[ch][sb][j] - 8) > 22) { 00507 run = 1; 00508 case_val = 8; 00509 } else { 00510 switch (switchtable[coding_method[ch][sb][j]-8]) { 00511 case 0: run = 10; case_val = 10; break; 00512 case 1: run = 1; case_val = 16; break; 00513 case 2: run = 5; case_val = 24; break; 00514 case 3: run = 3; case_val = 30; break; 00515 case 4: run = 1; case_val = 30; break; 00516 case 5: run = 1; case_val = 8; break; 00517 default: run = 1; case_val = 8; break; 00518 } 00519 } 00520 for (k = 0; k < run; k++) 00521 if (j + k < 128) 00522 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) 00523 if (k > 0) { 00524 SAMPLES_NEEDED 00525 //not debugged, almost never used 00526 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t)); 00527 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); 00528 } 00529 j += run; 00530 } 00531 } 00532 } 00533 00534 00542 static void fill_tone_level_array (QDM2Context *q, int flag) 00543 { 00544 int i, sb, ch, sb_used; 00545 int tmp, tab; 00546 00547 // This should never happen 00548 if (q->nb_channels <= 0) 00549 return; 00550 00551 for (ch = 0; ch < q->nb_channels; ch++) 00552 for (sb = 0; sb < 30; sb++) 00553 for (i = 0; i < 8; i++) { 00554 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) 00555 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ 00556 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 00557 else 00558 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 00559 if(tmp < 0) 00560 tmp += 0xff; 00561 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; 00562 } 00563 00564 sb_used = QDM2_SB_USED(q->sub_sampling); 00565 00566 if ((q->superblocktype_2_3 != 0) && !flag) { 00567 for (sb = 0; sb < sb_used; sb++) 00568 for (ch = 0; ch < q->nb_channels; ch++) 00569 for (i = 0; i < 64; i++) { 00570 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 00571 if (q->tone_level_idx[ch][sb][i] < 0) 00572 q->tone_level[ch][sb][i] = 0; 00573 else 00574 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; 00575 } 00576 } else { 00577 tab = q->superblocktype_2_3 ? 0 : 1; 00578 for (sb = 0; sb < sb_used; sb++) { 00579 if ((sb >= 4) && (sb <= 23)) { 00580 for (ch = 0; ch < q->nb_channels; ch++) 00581 for (i = 0; i < 64; i++) { 00582 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 00583 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - 00584 q->tone_level_idx_mid[ch][sb - 4][i / 8] - 00585 q->tone_level_idx_hi2[ch][sb - 4]; 00586 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 00587 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 00588 q->tone_level[ch][sb][i] = 0; 00589 else 00590 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 00591 } 00592 } else { 00593 if (sb > 4) { 00594 for (ch = 0; ch < q->nb_channels; ch++) 00595 for (i = 0; i < 64; i++) { 00596 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 00597 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - 00598 q->tone_level_idx_hi2[ch][sb - 4]; 00599 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 00600 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 00601 q->tone_level[ch][sb][i] = 0; 00602 else 00603 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 00604 } 00605 } else { 00606 for (ch = 0; ch < q->nb_channels; ch++) 00607 for (i = 0; i < 64; i++) { 00608 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 00609 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 00610 q->tone_level[ch][sb][i] = 0; 00611 else 00612 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 00613 } 00614 } 00615 } 00616 } 00617 } 00618 00619 return; 00620 } 00621 00622 00637 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, 00638 sb_int8_array coding_method, int nb_channels, 00639 int c, int superblocktype_2_3, int cm_table_select) 00640 { 00641 int ch, sb, j; 00642 int tmp, acc, esp_40, comp; 00643 int add1, add2, add3, add4; 00644 int64_t multres; 00645 00646 // This should never happen 00647 if (nb_channels <= 0) 00648 return; 00649 00650 if (!superblocktype_2_3) { 00651 /* This case is untested, no samples available */ 00652 SAMPLES_NEEDED 00653 for (ch = 0; ch < nb_channels; ch++) 00654 for (sb = 0; sb < 30; sb++) { 00655 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer 00656 add1 = tone_level_idx[ch][sb][j] - 10; 00657 if (add1 < 0) 00658 add1 = 0; 00659 add2 = add3 = add4 = 0; 00660 if (sb > 1) { 00661 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; 00662 if (add2 < 0) 00663 add2 = 0; 00664 } 00665 if (sb > 0) { 00666 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; 00667 if (add3 < 0) 00668 add3 = 0; 00669 } 00670 if (sb < 29) { 00671 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; 00672 if (add4 < 0) 00673 add4 = 0; 00674 } 00675 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; 00676 if (tmp < 0) 00677 tmp = 0; 00678 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; 00679 } 00680 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; 00681 } 00682 acc = 0; 00683 for (ch = 0; ch < nb_channels; ch++) 00684 for (sb = 0; sb < 30; sb++) 00685 for (j = 0; j < 64; j++) 00686 acc += tone_level_idx_temp[ch][sb][j]; 00687 00688 multres = 0x66666667 * (acc * 10); 00689 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); 00690 for (ch = 0; ch < nb_channels; ch++) 00691 for (sb = 0; sb < 30; sb++) 00692 for (j = 0; j < 64; j++) { 00693 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; 00694 if (comp < 0) 00695 comp += 0xff; 00696 comp /= 256; // signed shift 00697 switch(sb) { 00698 case 0: 00699 if (comp < 30) 00700 comp = 30; 00701 comp += 15; 00702 break; 00703 case 1: 00704 if (comp < 24) 00705 comp = 24; 00706 comp += 10; 00707 break; 00708 case 2: 00709 case 3: 00710 case 4: 00711 if (comp < 16) 00712 comp = 16; 00713 } 00714 if (comp <= 5) 00715 tmp = 0; 00716 else if (comp <= 10) 00717 tmp = 10; 00718 else if (comp <= 16) 00719 tmp = 16; 00720 else if (comp <= 24) 00721 tmp = -1; 00722 else 00723 tmp = 0; 00724 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; 00725 } 00726 for (sb = 0; sb < 30; sb++) 00727 fix_coding_method_array(sb, nb_channels, coding_method); 00728 for (ch = 0; ch < nb_channels; ch++) 00729 for (sb = 0; sb < 30; sb++) 00730 for (j = 0; j < 64; j++) 00731 if (sb >= 10) { 00732 if (coding_method[ch][sb][j] < 10) 00733 coding_method[ch][sb][j] = 10; 00734 } else { 00735 if (sb >= 2) { 00736 if (coding_method[ch][sb][j] < 16) 00737 coding_method[ch][sb][j] = 16; 00738 } else { 00739 if (coding_method[ch][sb][j] < 30) 00740 coding_method[ch][sb][j] = 30; 00741 } 00742 } 00743 } else { // superblocktype_2_3 != 0 00744 for (ch = 0; ch < nb_channels; ch++) 00745 for (sb = 0; sb < 30; sb++) 00746 for (j = 0; j < 64; j++) 00747 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; 00748 } 00749 00750 return; 00751 } 00752 00753 00765 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) 00766 { 00767 int sb, j, k, n, ch, run, channels; 00768 int joined_stereo, zero_encoding, chs; 00769 int type34_first; 00770 float type34_div = 0; 00771 float type34_predictor; 00772 float samples[10], sign_bits[16]; 00773 00774 if (length == 0) { 00775 // If no data use noise 00776 for (sb=sb_min; sb < sb_max; sb++) 00777 build_sb_samples_from_noise (q, sb); 00778 00779 return; 00780 } 00781 00782 for (sb = sb_min; sb < sb_max; sb++) { 00783 FIX_NOISE_IDX(q->noise_idx); 00784 00785 channels = q->nb_channels; 00786 00787 if (q->nb_channels <= 1 || sb < 12) 00788 joined_stereo = 0; 00789 else if (sb >= 24) 00790 joined_stereo = 1; 00791 else 00792 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0; 00793 00794 if (joined_stereo) { 00795 if (BITS_LEFT(length,gb) >= 16) 00796 for (j = 0; j < 16; j++) 00797 sign_bits[j] = get_bits1 (gb); 00798 00799 for (j = 0; j < 64; j++) 00800 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) 00801 q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; 00802 00803 fix_coding_method_array(sb, q->nb_channels, q->coding_method); 00804 channels = 1; 00805 } 00806 00807 for (ch = 0; ch < channels; ch++) { 00808 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; 00809 type34_predictor = 0.0; 00810 type34_first = 1; 00811 00812 for (j = 0; j < 128; ) { 00813 switch (q->coding_method[ch][sb][j / 2]) { 00814 case 8: 00815 if (BITS_LEFT(length,gb) >= 10) { 00816 if (zero_encoding) { 00817 for (k = 0; k < 5; k++) { 00818 if ((j + 2 * k) >= 128) 00819 break; 00820 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; 00821 } 00822 } else { 00823 n = get_bits(gb, 8); 00824 for (k = 0; k < 5; k++) 00825 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 00826 } 00827 for (k = 0; k < 5; k++) 00828 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); 00829 } else { 00830 for (k = 0; k < 10; k++) 00831 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 00832 } 00833 run = 10; 00834 break; 00835 00836 case 10: 00837 if (BITS_LEFT(length,gb) >= 1) { 00838 float f = 0.81; 00839 00840 if (get_bits1(gb)) 00841 f = -f; 00842 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; 00843 samples[0] = f; 00844 } else { 00845 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00846 } 00847 run = 1; 00848 break; 00849 00850 case 16: 00851 if (BITS_LEFT(length,gb) >= 10) { 00852 if (zero_encoding) { 00853 for (k = 0; k < 5; k++) { 00854 if ((j + k) >= 128) 00855 break; 00856 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; 00857 } 00858 } else { 00859 n = get_bits (gb, 8); 00860 for (k = 0; k < 5; k++) 00861 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 00862 } 00863 } else { 00864 for (k = 0; k < 5; k++) 00865 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 00866 } 00867 run = 5; 00868 break; 00869 00870 case 24: 00871 if (BITS_LEFT(length,gb) >= 7) { 00872 n = get_bits(gb, 7); 00873 for (k = 0; k < 3; k++) 00874 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; 00875 } else { 00876 for (k = 0; k < 3; k++) 00877 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 00878 } 00879 run = 3; 00880 break; 00881 00882 case 30: 00883 if (BITS_LEFT(length,gb) >= 4) 00884 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)]; 00885 else 00886 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00887 00888 run = 1; 00889 break; 00890 00891 case 34: 00892 if (BITS_LEFT(length,gb) >= 7) { 00893 if (type34_first) { 00894 type34_div = (float)(1 << get_bits(gb, 2)); 00895 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; 00896 type34_predictor = samples[0]; 00897 type34_first = 0; 00898 } else { 00899 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor; 00900 type34_predictor = samples[0]; 00901 } 00902 } else { 00903 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00904 } 00905 run = 1; 00906 break; 00907 00908 default: 00909 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00910 run = 1; 00911 break; 00912 } 00913 00914 if (joined_stereo) { 00915 float tmp[10][MPA_MAX_CHANNELS]; 00916 00917 for (k = 0; k < run; k++) { 00918 tmp[k][0] = samples[k]; 00919 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; 00920 } 00921 for (chs = 0; chs < q->nb_channels; chs++) 00922 for (k = 0; k < run; k++) 00923 if ((j + k) < 128) 00924 q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs]; 00925 } else { 00926 for (k = 0; k < run; k++) 00927 if ((j + k) < 128) 00928 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k]; 00929 } 00930 00931 j += run; 00932 } // j loop 00933 } // channel loop 00934 } // subband loop 00935 } 00936 00937 00947 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) 00948 { 00949 int i, k, run, level, diff; 00950 00951 if (BITS_LEFT(length,gb) < 16) 00952 return; 00953 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); 00954 00955 quantized_coeffs[0] = level; 00956 00957 for (i = 0; i < 7; ) { 00958 if (BITS_LEFT(length,gb) < 16) 00959 break; 00960 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; 00961 00962 if (BITS_LEFT(length,gb) < 16) 00963 break; 00964 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); 00965 00966 for (k = 1; k <= run; k++) 00967 quantized_coeffs[i + k] = (level + ((k * diff) / run)); 00968 00969 level += diff; 00970 i += run; 00971 } 00972 } 00973 00974 00984 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) 00985 { 00986 int sb, j, k, n, ch; 00987 00988 for (ch = 0; ch < q->nb_channels; ch++) { 00989 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length); 00990 00991 if (BITS_LEFT(length,gb) < 16) { 00992 memset(q->quantized_coeffs[ch][0], 0, 8); 00993 break; 00994 } 00995 } 00996 00997 n = q->sub_sampling + 1; 00998 00999 for (sb = 0; sb < n; sb++) 01000 for (ch = 0; ch < q->nb_channels; ch++) 01001 for (j = 0; j < 8; j++) { 01002 if (BITS_LEFT(length,gb) < 1) 01003 break; 01004 if (get_bits1(gb)) { 01005 for (k=0; k < 8; k++) { 01006 if (BITS_LEFT(length,gb) < 16) 01007 break; 01008 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); 01009 } 01010 } else { 01011 for (k=0; k < 8; k++) 01012 q->tone_level_idx_hi1[ch][sb][j][k] = 0; 01013 } 01014 } 01015 01016 n = QDM2_SB_USED(q->sub_sampling) - 4; 01017 01018 for (sb = 0; sb < n; sb++) 01019 for (ch = 0; ch < q->nb_channels; ch++) { 01020 if (BITS_LEFT(length,gb) < 16) 01021 break; 01022 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); 01023 if (sb > 19) 01024 q->tone_level_idx_hi2[ch][sb] -= 16; 01025 else 01026 for (j = 0; j < 8; j++) 01027 q->tone_level_idx_mid[ch][sb][j] = -16; 01028 } 01029 01030 n = QDM2_SB_USED(q->sub_sampling) - 5; 01031 01032 for (sb = 0; sb < n; sb++) 01033 for (ch = 0; ch < q->nb_channels; ch++) 01034 for (j = 0; j < 8; j++) { 01035 if (BITS_LEFT(length,gb) < 16) 01036 break; 01037 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; 01038 } 01039 } 01040 01047 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) 01048 { 01049 GetBitContext gb; 01050 int i, j, k, n, ch, run, level, diff; 01051 01052 init_get_bits(&gb, node->packet->data, node->packet->size*8); 01053 01054 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function 01055 01056 for (i = 1; i < n; i++) 01057 for (ch=0; ch < q->nb_channels; ch++) { 01058 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); 01059 q->quantized_coeffs[ch][i][0] = level; 01060 01061 for (j = 0; j < (8 - 1); ) { 01062 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; 01063 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); 01064 01065 for (k = 1; k <= run; k++) 01066 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); 01067 01068 level += diff; 01069 j += run; 01070 } 01071 } 01072 01073 for (ch = 0; ch < q->nb_channels; ch++) 01074 for (i = 0; i < 8; i++) 01075 q->quantized_coeffs[ch][0][i] = 0; 01076 } 01077 01078 01086 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) 01087 { 01088 GetBitContext gb; 01089 01090 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 01091 01092 if (length != 0) { 01093 init_tone_level_dequantization(q, &gb, length); 01094 fill_tone_level_array(q, 1); 01095 } else { 01096 fill_tone_level_array(q, 0); 01097 } 01098 } 01099 01100 01108 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length) 01109 { 01110 GetBitContext gb; 01111 01112 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 01113 if (length >= 32) { 01114 int c = get_bits (&gb, 13); 01115 01116 if (c > 3) 01117 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, 01118 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select); 01119 } 01120 01121 synthfilt_build_sb_samples(q, &gb, length, 0, 8); 01122 } 01123 01124 01132 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) 01133 { 01134 GetBitContext gb; 01135 01136 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 01137 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); 01138 } 01139 01140 /* 01141 * Process new subpackets for synthesis filter 01142 * 01143 * @param q context 01144 * @param list list with synthesis filter packets (list D) 01145 */ 01146 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) 01147 { 01148 QDM2SubPNode *nodes[4]; 01149 01150 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); 01151 if (nodes[0] != NULL) 01152 process_subpacket_9(q, nodes[0]); 01153 01154 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); 01155 if (nodes[1] != NULL) 01156 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3); 01157 else 01158 process_subpacket_10(q, NULL, 0); 01159 01160 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); 01161 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) 01162 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3)); 01163 else 01164 process_subpacket_11(q, NULL, 0); 01165 01166 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); 01167 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) 01168 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3)); 01169 else 01170 process_subpacket_12(q, NULL, 0); 01171 } 01172 01173 01174 /* 01175 * Decode superblock, fill packet lists. 01176 * 01177 * @param q context 01178 */ 01179 static void qdm2_decode_super_block (QDM2Context *q) 01180 { 01181 GetBitContext gb; 01182 QDM2SubPacket header, *packet; 01183 int i, packet_bytes, sub_packet_size, sub_packets_D; 01184 unsigned int next_index = 0; 01185 01186 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); 01187 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); 01188 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); 01189 01190 q->sub_packets_B = 0; 01191 sub_packets_D = 0; 01192 01193 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] 01194 01195 init_get_bits(&gb, q->compressed_data, q->compressed_size*8); 01196 qdm2_decode_sub_packet_header(&gb, &header); 01197 01198 if (header.type < 2 || header.type >= 8) { 01199 q->has_errors = 1; 01200 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); 01201 return; 01202 } 01203 01204 q->superblocktype_2_3 = (header.type == 2 || header.type == 3); 01205 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); 01206 01207 init_get_bits(&gb, header.data, header.size*8); 01208 01209 if (header.type == 2 || header.type == 4 || header.type == 5) { 01210 int csum = 257 * get_bits(&gb, 8); 01211 csum += 2 * get_bits(&gb, 8); 01212 01213 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); 01214 01215 if (csum != 0) { 01216 q->has_errors = 1; 01217 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); 01218 return; 01219 } 01220 } 01221 01222 q->sub_packet_list_B[0].packet = NULL; 01223 q->sub_packet_list_D[0].packet = NULL; 01224 01225 for (i = 0; i < 6; i++) 01226 if (--q->fft_level_exp[i] < 0) 01227 q->fft_level_exp[i] = 0; 01228 01229 for (i = 0; packet_bytes > 0; i++) { 01230 int j; 01231 01232 q->sub_packet_list_A[i].next = NULL; 01233 01234 if (i > 0) { 01235 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; 01236 01237 /* seek to next block */ 01238 init_get_bits(&gb, header.data, header.size*8); 01239 skip_bits(&gb, next_index*8); 01240 01241 if (next_index >= header.size) 01242 break; 01243 } 01244 01245 /* decode subpacket */ 01246 packet = &q->sub_packets[i]; 01247 qdm2_decode_sub_packet_header(&gb, packet); 01248 next_index = packet->size + get_bits_count(&gb) / 8; 01249 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; 01250 01251 if (packet->type == 0) 01252 break; 01253 01254 if (sub_packet_size > packet_bytes) { 01255 if (packet->type != 10 && packet->type != 11 && packet->type != 12) 01256 break; 01257 packet->size += packet_bytes - sub_packet_size; 01258 } 01259 01260 packet_bytes -= sub_packet_size; 01261 01262 /* add subpacket to 'all subpackets' list */ 01263 q->sub_packet_list_A[i].packet = packet; 01264 01265 /* add subpacket to related list */ 01266 if (packet->type == 8) { 01267 SAMPLES_NEEDED_2("packet type 8"); 01268 return; 01269 } else if (packet->type >= 9 && packet->type <= 12) { 01270 /* packets for MPEG Audio like Synthesis Filter */ 01271 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); 01272 } else if (packet->type == 13) { 01273 for (j = 0; j < 6; j++) 01274 q->fft_level_exp[j] = get_bits(&gb, 6); 01275 } else if (packet->type == 14) { 01276 for (j = 0; j < 6; j++) 01277 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); 01278 } else if (packet->type == 15) { 01279 SAMPLES_NEEDED_2("packet type 15") 01280 return; 01281 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { 01282 /* packets for FFT */ 01283 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); 01284 } 01285 } // Packet bytes loop 01286 01287 /* **************************************************************** */ 01288 if (q->sub_packet_list_D[0].packet != NULL) { 01289 process_synthesis_subpackets(q, q->sub_packet_list_D); 01290 q->do_synth_filter = 1; 01291 } else if (q->do_synth_filter) { 01292 process_subpacket_10(q, NULL, 0); 01293 process_subpacket_11(q, NULL, 0); 01294 process_subpacket_12(q, NULL, 0); 01295 } 01296 /* **************************************************************** */ 01297 } 01298 01299 01300 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, 01301 int offset, int duration, int channel, 01302 int exp, int phase) 01303 { 01304 if (q->fft_coefs_min_index[duration] < 0) 01305 q->fft_coefs_min_index[duration] = q->fft_coefs_index; 01306 01307 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); 01308 q->fft_coefs[q->fft_coefs_index].channel = channel; 01309 q->fft_coefs[q->fft_coefs_index].offset = offset; 01310 q->fft_coefs[q->fft_coefs_index].exp = exp; 01311 q->fft_coefs[q->fft_coefs_index].phase = phase; 01312 q->fft_coefs_index++; 01313 } 01314 01315 01316 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) 01317 { 01318 int channel, stereo, phase, exp; 01319 int local_int_4, local_int_8, stereo_phase, local_int_10; 01320 int local_int_14, stereo_exp, local_int_20, local_int_28; 01321 int n, offset; 01322 01323 local_int_4 = 0; 01324 local_int_28 = 0; 01325 local_int_20 = 2; 01326 local_int_8 = (4 - duration); 01327 local_int_10 = 1 << (q->group_order - duration - 1); 01328 offset = 1; 01329 01330 while (1) { 01331 if (q->superblocktype_2_3) { 01332 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { 01333 offset = 1; 01334 if (n == 0) { 01335 local_int_4 += local_int_10; 01336 local_int_28 += (1 << local_int_8); 01337 } else { 01338 local_int_4 += 8*local_int_10; 01339 local_int_28 += (8 << local_int_8); 01340 } 01341 } 01342 offset += (n - 2); 01343 } else { 01344 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); 01345 while (offset >= (local_int_10 - 1)) { 01346 offset += (1 - (local_int_10 - 1)); 01347 local_int_4 += local_int_10; 01348 local_int_28 += (1 << local_int_8); 01349 } 01350 } 01351 01352 if (local_int_4 >= q->group_size) 01353 return; 01354 01355 local_int_14 = (offset >> local_int_8); 01356 01357 if (q->nb_channels > 1) { 01358 channel = get_bits1(gb); 01359 stereo = get_bits1(gb); 01360 } else { 01361 channel = 0; 01362 stereo = 0; 01363 } 01364 01365 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); 01366 exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; 01367 exp = (exp < 0) ? 0 : exp; 01368 01369 phase = get_bits(gb, 3); 01370 stereo_exp = 0; 01371 stereo_phase = 0; 01372 01373 if (stereo) { 01374 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); 01375 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); 01376 if (stereo_phase < 0) 01377 stereo_phase += 8; 01378 } 01379 01380 if (q->frequency_range > (local_int_14 + 1)) { 01381 int sub_packet = (local_int_20 + local_int_28); 01382 01383 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); 01384 if (stereo) 01385 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); 01386 } 01387 01388 offset++; 01389 } 01390 } 01391 01392 01393 static void qdm2_decode_fft_packets (QDM2Context *q) 01394 { 01395 int i, j, min, max, value, type, unknown_flag; 01396 GetBitContext gb; 01397 01398 if (q->sub_packet_list_B[0].packet == NULL) 01399 return; 01400 01401 /* reset minimum indexes for FFT coefficients */ 01402 q->fft_coefs_index = 0; 01403 for (i=0; i < 5; i++) 01404 q->fft_coefs_min_index[i] = -1; 01405 01406 /* process subpackets ordered by type, largest type first */ 01407 for (i = 0, max = 256; i < q->sub_packets_B; i++) { 01408 QDM2SubPacket *packet= NULL; 01409 01410 /* find subpacket with largest type less than max */ 01411 for (j = 0, min = 0; j < q->sub_packets_B; j++) { 01412 value = q->sub_packet_list_B[j].packet->type; 01413 if (value > min && value < max) { 01414 min = value; 01415 packet = q->sub_packet_list_B[j].packet; 01416 } 01417 } 01418 01419 max = min; 01420 01421 /* check for errors (?) */ 01422 if (!packet) 01423 return; 01424 01425 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) 01426 return; 01427 01428 /* decode FFT tones */ 01429 init_get_bits (&gb, packet->data, packet->size*8); 01430 01431 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) 01432 unknown_flag = 1; 01433 else 01434 unknown_flag = 0; 01435 01436 type = packet->type; 01437 01438 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { 01439 int duration = q->sub_sampling + 5 - (type & 15); 01440 01441 if (duration >= 0 && duration < 4) 01442 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); 01443 } else if (type == 31) { 01444 for (j=0; j < 4; j++) 01445 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 01446 } else if (type == 46) { 01447 for (j=0; j < 6; j++) 01448 q->fft_level_exp[j] = get_bits(&gb, 6); 01449 for (j=0; j < 4; j++) 01450 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 01451 } 01452 } // Loop on B packets 01453 01454 /* calculate maximum indexes for FFT coefficients */ 01455 for (i = 0, j = -1; i < 5; i++) 01456 if (q->fft_coefs_min_index[i] >= 0) { 01457 if (j >= 0) 01458 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; 01459 j = i; 01460 } 01461 if (j >= 0) 01462 q->fft_coefs_max_index[j] = q->fft_coefs_index; 01463 } 01464 01465 01466 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) 01467 { 01468 float level, f[6]; 01469 int i; 01470 QDM2Complex c; 01471 const double iscale = 2.0*M_PI / 512.0; 01472 01473 tone->phase += tone->phase_shift; 01474 01475 /* calculate current level (maximum amplitude) of tone */ 01476 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; 01477 c.im = level * sin(tone->phase*iscale); 01478 c.re = level * cos(tone->phase*iscale); 01479 01480 /* generate FFT coefficients for tone */ 01481 if (tone->duration >= 3 || tone->cutoff >= 3) { 01482 tone->complex[0].im += c.im; 01483 tone->complex[0].re += c.re; 01484 tone->complex[1].im -= c.im; 01485 tone->complex[1].re -= c.re; 01486 } else { 01487 f[1] = -tone->table[4]; 01488 f[0] = tone->table[3] - tone->table[0]; 01489 f[2] = 1.0 - tone->table[2] - tone->table[3]; 01490 f[3] = tone->table[1] + tone->table[4] - 1.0; 01491 f[4] = tone->table[0] - tone->table[1]; 01492 f[5] = tone->table[2]; 01493 for (i = 0; i < 2; i++) { 01494 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; 01495 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); 01496 } 01497 for (i = 0; i < 4; i++) { 01498 tone->complex[i].re += c.re * f[i+2]; 01499 tone->complex[i].im += c.im * f[i+2]; 01500 } 01501 } 01502 01503 /* copy the tone if it has not yet died out */ 01504 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { 01505 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); 01506 q->fft_tone_end = (q->fft_tone_end + 1) % 1000; 01507 } 01508 } 01509 01510 01511 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) 01512 { 01513 int i, j, ch; 01514 const double iscale = 0.25 * M_PI; 01515 01516 for (ch = 0; ch < q->channels; ch++) { 01517 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); 01518 } 01519 01520 01521 /* apply FFT tones with duration 4 (1 FFT period) */ 01522 if (q->fft_coefs_min_index[4] >= 0) 01523 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { 01524 float level; 01525 QDM2Complex c; 01526 01527 if (q->fft_coefs[i].sub_packet != sub_packet) 01528 break; 01529 01530 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; 01531 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]; 01532 01533 c.re = level * cos(q->fft_coefs[i].phase * iscale); 01534 c.im = level * sin(q->fft_coefs[i].phase * iscale); 01535 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; 01536 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; 01537 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; 01538 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; 01539 } 01540 01541 /* generate existing FFT tones */ 01542 for (i = q->fft_tone_end; i != q->fft_tone_start; ) { 01543 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); 01544 q->fft_tone_start = (q->fft_tone_start + 1) % 1000; 01545 } 01546 01547 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ 01548 for (i = 0; i < 4; i++) 01549 if (q->fft_coefs_min_index[i] >= 0) { 01550 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { 01551 int offset, four_i; 01552 FFTTone tone; 01553 01554 if (q->fft_coefs[j].sub_packet != sub_packet) 01555 break; 01556 01557 four_i = (4 - i); 01558 offset = q->fft_coefs[j].offset >> four_i; 01559 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; 01560 01561 if (offset < q->frequency_range) { 01562 if (offset < 2) 01563 tone.cutoff = offset; 01564 else 01565 tone.cutoff = (offset >= 60) ? 3 : 2; 01566 01567 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]; 01568 tone.complex = &q->fft.complex[ch][offset]; 01569 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; 01570 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; 01571 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); 01572 tone.duration = i; 01573 tone.time_index = 0; 01574 01575 qdm2_fft_generate_tone(q, &tone); 01576 } 01577 } 01578 q->fft_coefs_min_index[i] = j; 01579 } 01580 } 01581 01582 01583 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) 01584 { 01585 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; 01586 int i; 01587 q->fft.complex[channel][0].re *= 2.0f; 01588 q->fft.complex[channel][0].im = 0.0f; 01589 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); 01590 /* add samples to output buffer */ 01591 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) 01592 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; 01593 } 01594 01595 01600 static void qdm2_synthesis_filter (QDM2Context *q, int index) 01601 { 01602 int i, k, ch, sb_used, sub_sampling, dither_state = 0; 01603 01604 /* copy sb_samples */ 01605 sb_used = QDM2_SB_USED(q->sub_sampling); 01606 01607 for (ch = 0; ch < q->channels; ch++) 01608 for (i = 0; i < 8; i++) 01609 for (k=sb_used; k < SBLIMIT; k++) 01610 q->sb_samples[ch][(8 * index) + i][k] = 0; 01611 01612 for (ch = 0; ch < q->nb_channels; ch++) { 01613 float *samples_ptr = q->samples + ch; 01614 01615 for (i = 0; i < 8; i++) { 01616 ff_mpa_synth_filter_float(&q->mpadsp, 01617 q->synth_buf[ch], &(q->synth_buf_offset[ch]), 01618 ff_mpa_synth_window_float, &dither_state, 01619 samples_ptr, q->nb_channels, 01620 q->sb_samples[ch][(8 * index) + i]); 01621 samples_ptr += 32 * q->nb_channels; 01622 } 01623 } 01624 01625 /* add samples to output buffer */ 01626 sub_sampling = (4 >> q->sub_sampling); 01627 01628 for (ch = 0; ch < q->channels; ch++) 01629 for (i = 0; i < q->frame_size; i++) 01630 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch]; 01631 } 01632 01633 01639 static av_cold void qdm2_init(QDM2Context *q) { 01640 static int initialized = 0; 01641 01642 if (initialized != 0) 01643 return; 01644 initialized = 1; 01645 01646 qdm2_init_vlc(); 01647 ff_mpa_synth_init_float(ff_mpa_synth_window_float); 01648 softclip_table_init(); 01649 rnd_table_init(); 01650 init_noise_samples(); 01651 01652 av_log(NULL, AV_LOG_DEBUG, "init done\n"); 01653 } 01654 01655 01656 #if 0 01657 static void dump_context(QDM2Context *q) 01658 { 01659 int i; 01660 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b); 01661 PRINT("compressed_data",q->compressed_data); 01662 PRINT("compressed_size",q->compressed_size); 01663 PRINT("frame_size",q->frame_size); 01664 PRINT("checksum_size",q->checksum_size); 01665 PRINT("channels",q->channels); 01666 PRINT("nb_channels",q->nb_channels); 01667 PRINT("fft_frame_size",q->fft_frame_size); 01668 PRINT("fft_size",q->fft_size); 01669 PRINT("sub_sampling",q->sub_sampling); 01670 PRINT("fft_order",q->fft_order); 01671 PRINT("group_order",q->group_order); 01672 PRINT("group_size",q->group_size); 01673 PRINT("sub_packet",q->sub_packet); 01674 PRINT("frequency_range",q->frequency_range); 01675 PRINT("has_errors",q->has_errors); 01676 PRINT("fft_tone_end",q->fft_tone_end); 01677 PRINT("fft_tone_start",q->fft_tone_start); 01678 PRINT("fft_coefs_index",q->fft_coefs_index); 01679 PRINT("coeff_per_sb_select",q->coeff_per_sb_select); 01680 PRINT("cm_table_select",q->cm_table_select); 01681 PRINT("noise_idx",q->noise_idx); 01682 01683 for (i = q->fft_tone_start; i < q->fft_tone_end; i++) 01684 { 01685 FFTTone *t = &q->fft_tones[i]; 01686 01687 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i); 01688 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level); 01689 // PRINT(" level", t->level); 01690 PRINT(" phase", t->phase); 01691 PRINT(" phase_shift", t->phase_shift); 01692 PRINT(" duration", t->duration); 01693 PRINT(" samples_im", t->samples_im); 01694 PRINT(" samples_re", t->samples_re); 01695 PRINT(" table", t->table); 01696 } 01697 01698 } 01699 #endif 01700 01701 01705 static av_cold int qdm2_decode_init(AVCodecContext *avctx) 01706 { 01707 QDM2Context *s = avctx->priv_data; 01708 uint8_t *extradata; 01709 int extradata_size; 01710 int tmp_val, tmp, size; 01711 01712 /* extradata parsing 01713 01714 Structure: 01715 wave { 01716 frma (QDM2) 01717 QDCA 01718 QDCP 01719 } 01720 01721 32 size (including this field) 01722 32 tag (=frma) 01723 32 type (=QDM2 or QDMC) 01724 01725 32 size (including this field, in bytes) 01726 32 tag (=QDCA) // maybe mandatory parameters 01727 32 unknown (=1) 01728 32 channels (=2) 01729 32 samplerate (=44100) 01730 32 bitrate (=96000) 01731 32 block size (=4096) 01732 32 frame size (=256) (for one channel) 01733 32 packet size (=1300) 01734 01735 32 size (including this field, in bytes) 01736 32 tag (=QDCP) // maybe some tuneable parameters 01737 32 float1 (=1.0) 01738 32 zero ? 01739 32 float2 (=1.0) 01740 32 float3 (=1.0) 01741 32 unknown (27) 01742 32 unknown (8) 01743 32 zero ? 01744 */ 01745 01746 if (!avctx->extradata || (avctx->extradata_size < 48)) { 01747 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); 01748 return -1; 01749 } 01750 01751 extradata = avctx->extradata; 01752 extradata_size = avctx->extradata_size; 01753 01754 while (extradata_size > 7) { 01755 if (!memcmp(extradata, "frmaQDM", 7)) 01756 break; 01757 extradata++; 01758 extradata_size--; 01759 } 01760 01761 if (extradata_size < 12) { 01762 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", 01763 extradata_size); 01764 return -1; 01765 } 01766 01767 if (memcmp(extradata, "frmaQDM", 7)) { 01768 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); 01769 return -1; 01770 } 01771 01772 if (extradata[7] == 'C') { 01773 // s->is_qdmc = 1; 01774 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); 01775 return -1; 01776 } 01777 01778 extradata += 8; 01779 extradata_size -= 8; 01780 01781 size = AV_RB32(extradata); 01782 01783 if(size > extradata_size){ 01784 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", 01785 extradata_size, size); 01786 return -1; 01787 } 01788 01789 extradata += 4; 01790 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); 01791 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { 01792 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); 01793 return -1; 01794 } 01795 01796 extradata += 8; 01797 01798 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); 01799 extradata += 4; 01800 01801 avctx->sample_rate = AV_RB32(extradata); 01802 extradata += 4; 01803 01804 avctx->bit_rate = AV_RB32(extradata); 01805 extradata += 4; 01806 01807 s->group_size = AV_RB32(extradata); 01808 extradata += 4; 01809 01810 s->fft_size = AV_RB32(extradata); 01811 extradata += 4; 01812 01813 s->checksum_size = AV_RB32(extradata); 01814 01815 s->fft_order = av_log2(s->fft_size) + 1; 01816 s->fft_frame_size = 2 * s->fft_size; // complex has two floats 01817 01818 // something like max decodable tones 01819 s->group_order = av_log2(s->group_size) + 1; 01820 s->frame_size = s->group_size / 16; // 16 iterations per super block 01821 01822 s->sub_sampling = s->fft_order - 7; 01823 s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); 01824 01825 switch ((s->sub_sampling * 2 + s->channels - 1)) { 01826 case 0: tmp = 40; break; 01827 case 1: tmp = 48; break; 01828 case 2: tmp = 56; break; 01829 case 3: tmp = 72; break; 01830 case 4: tmp = 80; break; 01831 case 5: tmp = 100;break; 01832 default: tmp=s->sub_sampling; break; 01833 } 01834 tmp_val = 0; 01835 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; 01836 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; 01837 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; 01838 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; 01839 s->cm_table_select = tmp_val; 01840 01841 if (s->sub_sampling == 0) 01842 tmp = 7999; 01843 else 01844 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; 01845 /* 01846 0: 7999 -> 0 01847 1: 20000 -> 2 01848 2: 28000 -> 2 01849 */ 01850 if (tmp < 8000) 01851 s->coeff_per_sb_select = 0; 01852 else if (tmp <= 16000) 01853 s->coeff_per_sb_select = 1; 01854 else 01855 s->coeff_per_sb_select = 2; 01856 01857 // Fail on unknown fft order 01858 if ((s->fft_order < 7) || (s->fft_order > 9)) { 01859 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); 01860 return -1; 01861 } 01862 01863 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); 01864 ff_mpadsp_init(&s->mpadsp); 01865 01866 qdm2_init(s); 01867 01868 avctx->sample_fmt = AV_SAMPLE_FMT_S16; 01869 01870 // dump_context(s); 01871 return 0; 01872 } 01873 01874 01875 static av_cold int qdm2_decode_close(AVCodecContext *avctx) 01876 { 01877 QDM2Context *s = avctx->priv_data; 01878 01879 ff_rdft_end(&s->rdft_ctx); 01880 01881 return 0; 01882 } 01883 01884 01885 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) 01886 { 01887 int ch, i; 01888 const int frame_size = (q->frame_size * q->channels); 01889 01890 /* select input buffer */ 01891 q->compressed_data = in; 01892 q->compressed_size = q->checksum_size; 01893 01894 // dump_context(q); 01895 01896 /* copy old block, clear new block of output samples */ 01897 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); 01898 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); 01899 01900 /* decode block of QDM2 compressed data */ 01901 if (q->sub_packet == 0) { 01902 q->has_errors = 0; // zero it for a new super block 01903 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); 01904 qdm2_decode_super_block(q); 01905 } 01906 01907 /* parse subpackets */ 01908 if (!q->has_errors) { 01909 if (q->sub_packet == 2) 01910 qdm2_decode_fft_packets(q); 01911 01912 qdm2_fft_tone_synthesizer(q, q->sub_packet); 01913 } 01914 01915 /* sound synthesis stage 1 (FFT) */ 01916 for (ch = 0; ch < q->channels; ch++) { 01917 qdm2_calculate_fft(q, ch, q->sub_packet); 01918 01919 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { 01920 SAMPLES_NEEDED_2("has errors, and C list is not empty") 01921 return -1; 01922 } 01923 } 01924 01925 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ 01926 if (!q->has_errors && q->do_synth_filter) 01927 qdm2_synthesis_filter(q, q->sub_packet); 01928 01929 q->sub_packet = (q->sub_packet + 1) % 16; 01930 01931 /* clip and convert output float[] to 16bit signed samples */ 01932 for (i = 0; i < frame_size; i++) { 01933 int value = (int)q->output_buffer[i]; 01934 01935 if (value > SOFTCLIP_THRESHOLD) 01936 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; 01937 else if (value < -SOFTCLIP_THRESHOLD) 01938 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; 01939 01940 out[i] = value; 01941 } 01942 01943 return 0; 01944 } 01945 01946 01947 static int qdm2_decode_frame(AVCodecContext *avctx, 01948 void *data, int *data_size, 01949 AVPacket *avpkt) 01950 { 01951 const uint8_t *buf = avpkt->data; 01952 int buf_size = avpkt->size; 01953 QDM2Context *s = avctx->priv_data; 01954 int16_t *out = data; 01955 int i; 01956 01957 if(!buf) 01958 return 0; 01959 if(buf_size < s->checksum_size) 01960 return -1; 01961 01962 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", 01963 buf_size, buf, s->checksum_size, data, *data_size); 01964 01965 for (i = 0; i < 16; i++) { 01966 if (qdm2_decode(s, buf, out) < 0) 01967 return -1; 01968 out += s->channels * s->frame_size; 01969 } 01970 01971 *data_size = (uint8_t*)out - (uint8_t*)data; 01972 01973 return s->checksum_size; 01974 } 01975 01976 AVCodec ff_qdm2_decoder = 01977 { 01978 .name = "qdm2", 01979 .type = AVMEDIA_TYPE_AUDIO, 01980 .id = CODEC_ID_QDM2, 01981 .priv_data_size = sizeof(QDM2Context), 01982 .init = qdm2_decode_init, 01983 .close = qdm2_decode_close, 01984 .decode = qdm2_decode_frame, 01985 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), 01986 };