Libav
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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 FFmpeg. 00009 * 00010 * FFmpeg 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 * FFmpeg 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 FFmpeg; 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 "fft.h" 00042 #include "mpegaudio.h" 00043 00044 #include "qdm2data.h" 00045 #include "qdm2_tablegen.h" 00046 00047 #undef NDEBUG 00048 #include <assert.h> 00049 00050 00051 #define QDM2_LIST_ADD(list, size, packet) \ 00052 do { \ 00053 if (size > 0) { \ 00054 list[size - 1].next = &list[size]; \ 00055 } \ 00056 list[size].packet = packet; \ 00057 list[size].next = NULL; \ 00058 size++; \ 00059 } while(0) 00060 00061 // Result is 8, 16 or 30 00062 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling)) 00063 00064 #define FIX_NOISE_IDX(noise_idx) \ 00065 if ((noise_idx) >= 3840) \ 00066 (noise_idx) -= 3840; \ 00067 00068 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)]) 00069 00070 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb))) 00071 00072 #define SAMPLES_NEEDED \ 00073 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); 00074 00075 #define SAMPLES_NEEDED_2(why) \ 00076 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); 00077 00078 #define QDM2_MAX_FRAME_SIZE 512 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(16, 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[QDM2_MAX_FRAME_SIZE * 2]; 00172 00174 DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2]; 00175 int synth_buf_offset[MPA_MAX_CHANNELS]; 00176 DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; 00177 00179 float tone_level[MPA_MAX_CHANNELS][30][64]; 00180 int8_t coding_method[MPA_MAX_CHANNELS][30][64]; 00181 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; 00182 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; 00183 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; 00184 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; 00185 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; 00186 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; 00187 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; 00188 00189 // Flags 00190 int has_errors; 00191 int superblocktype_2_3; 00192 int do_synth_filter; 00193 00194 int sub_packet; 00195 int noise_idx; 00196 } QDM2Context; 00197 00198 00199 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE]; 00200 00201 static VLC vlc_tab_level; 00202 static VLC vlc_tab_diff; 00203 static VLC vlc_tab_run; 00204 static VLC fft_level_exp_alt_vlc; 00205 static VLC fft_level_exp_vlc; 00206 static VLC fft_stereo_exp_vlc; 00207 static VLC fft_stereo_phase_vlc; 00208 static VLC vlc_tab_tone_level_idx_hi1; 00209 static VLC vlc_tab_tone_level_idx_mid; 00210 static VLC vlc_tab_tone_level_idx_hi2; 00211 static VLC vlc_tab_type30; 00212 static VLC vlc_tab_type34; 00213 static VLC vlc_tab_fft_tone_offset[5]; 00214 00215 static const uint16_t qdm2_vlc_offs[] = { 00216 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, 00217 }; 00218 00219 static av_cold void qdm2_init_vlc(void) 00220 { 00221 static int vlcs_initialized = 0; 00222 static VLC_TYPE qdm2_table[3838][2]; 00223 00224 if (!vlcs_initialized) { 00225 00226 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; 00227 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; 00228 init_vlc (&vlc_tab_level, 8, 24, 00229 vlc_tab_level_huffbits, 1, 1, 00230 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00231 00232 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; 00233 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; 00234 init_vlc (&vlc_tab_diff, 8, 37, 00235 vlc_tab_diff_huffbits, 1, 1, 00236 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00237 00238 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; 00239 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; 00240 init_vlc (&vlc_tab_run, 5, 6, 00241 vlc_tab_run_huffbits, 1, 1, 00242 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00243 00244 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; 00245 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3]; 00246 init_vlc (&fft_level_exp_alt_vlc, 8, 28, 00247 fft_level_exp_alt_huffbits, 1, 1, 00248 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00249 00250 00251 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; 00252 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; 00253 init_vlc (&fft_level_exp_vlc, 8, 20, 00254 fft_level_exp_huffbits, 1, 1, 00255 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00256 00257 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; 00258 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5]; 00259 init_vlc (&fft_stereo_exp_vlc, 6, 7, 00260 fft_stereo_exp_huffbits, 1, 1, 00261 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00262 00263 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; 00264 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6]; 00265 init_vlc (&fft_stereo_phase_vlc, 6, 9, 00266 fft_stereo_phase_huffbits, 1, 1, 00267 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00268 00269 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]]; 00270 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7]; 00271 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, 00272 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, 00273 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00274 00275 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]]; 00276 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8]; 00277 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, 00278 vlc_tab_tone_level_idx_mid_huffbits, 1, 1, 00279 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00280 00281 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]]; 00282 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9]; 00283 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, 00284 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, 00285 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00286 00287 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; 00288 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; 00289 init_vlc (&vlc_tab_type30, 6, 9, 00290 vlc_tab_type30_huffbits, 1, 1, 00291 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00292 00293 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; 00294 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; 00295 init_vlc (&vlc_tab_type34, 5, 10, 00296 vlc_tab_type34_huffbits, 1, 1, 00297 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00298 00299 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]]; 00300 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12]; 00301 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, 00302 vlc_tab_fft_tone_offset_0_huffbits, 1, 1, 00303 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00304 00305 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]]; 00306 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13]; 00307 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, 00308 vlc_tab_fft_tone_offset_1_huffbits, 1, 1, 00309 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00310 00311 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]]; 00312 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14]; 00313 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, 00314 vlc_tab_fft_tone_offset_2_huffbits, 1, 1, 00315 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00316 00317 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]]; 00318 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15]; 00319 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, 00320 vlc_tab_fft_tone_offset_3_huffbits, 1, 1, 00321 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00322 00323 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]]; 00324 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16]; 00325 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, 00326 vlc_tab_fft_tone_offset_4_huffbits, 1, 1, 00327 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00328 00329 vlcs_initialized=1; 00330 } 00331 } 00332 00333 00334 /* for floating point to fixed point conversion */ 00335 static const float f2i_scale = (float) (1 << (FRAC_BITS - 15)); 00336 00337 00338 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) 00339 { 00340 int value; 00341 00342 value = get_vlc2(gb, vlc->table, vlc->bits, depth); 00343 00344 /* stage-2, 3 bits exponent escape sequence */ 00345 if (value-- == 0) 00346 value = get_bits (gb, get_bits (gb, 3) + 1); 00347 00348 /* stage-3, optional */ 00349 if (flag) { 00350 int tmp = vlc_stage3_values[value]; 00351 00352 if ((value & ~3) > 0) 00353 tmp += get_bits (gb, (value >> 2)); 00354 value = tmp; 00355 } 00356 00357 return value; 00358 } 00359 00360 00361 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) 00362 { 00363 int value = qdm2_get_vlc (gb, vlc, 0, depth); 00364 00365 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); 00366 } 00367 00368 00378 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) { 00379 int i; 00380 00381 for (i=0; i < length; i++) 00382 value -= data[i]; 00383 00384 return (uint16_t)(value & 0xffff); 00385 } 00386 00387 00394 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) 00395 { 00396 sub_packet->type = get_bits (gb, 8); 00397 00398 if (sub_packet->type == 0) { 00399 sub_packet->size = 0; 00400 sub_packet->data = NULL; 00401 } else { 00402 sub_packet->size = get_bits (gb, 8); 00403 00404 if (sub_packet->type & 0x80) { 00405 sub_packet->size <<= 8; 00406 sub_packet->size |= get_bits (gb, 8); 00407 sub_packet->type &= 0x7f; 00408 } 00409 00410 if (sub_packet->type == 0x7f) 00411 sub_packet->type |= (get_bits (gb, 8) << 8); 00412 00413 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data 00414 } 00415 00416 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", 00417 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); 00418 } 00419 00420 00428 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) 00429 { 00430 while (list != NULL && list->packet != NULL) { 00431 if (list->packet->type == type) 00432 return list; 00433 list = list->next; 00434 } 00435 return NULL; 00436 } 00437 00438 00445 static void average_quantized_coeffs (QDM2Context *q) 00446 { 00447 int i, j, n, ch, sum; 00448 00449 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; 00450 00451 for (ch = 0; ch < q->nb_channels; ch++) 00452 for (i = 0; i < n; i++) { 00453 sum = 0; 00454 00455 for (j = 0; j < 8; j++) 00456 sum += q->quantized_coeffs[ch][i][j]; 00457 00458 sum /= 8; 00459 if (sum > 0) 00460 sum--; 00461 00462 for (j=0; j < 8; j++) 00463 q->quantized_coeffs[ch][i][j] = sum; 00464 } 00465 } 00466 00467 00475 static void build_sb_samples_from_noise (QDM2Context *q, int sb) 00476 { 00477 int ch, j; 00478 00479 FIX_NOISE_IDX(q->noise_idx); 00480 00481 if (!q->nb_channels) 00482 return; 00483 00484 for (ch = 0; ch < q->nb_channels; ch++) 00485 for (j = 0; j < 64; j++) { 00486 q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5); 00487 q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5); 00488 } 00489 } 00490 00491 00500 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) 00501 { 00502 int j,k; 00503 int ch; 00504 int run, case_val; 00505 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}; 00506 00507 for (ch = 0; ch < channels; ch++) { 00508 for (j = 0; j < 64; ) { 00509 if((coding_method[ch][sb][j] - 8) > 22) { 00510 run = 1; 00511 case_val = 8; 00512 } else { 00513 switch (switchtable[coding_method[ch][sb][j]-8]) { 00514 case 0: run = 10; case_val = 10; break; 00515 case 1: run = 1; case_val = 16; break; 00516 case 2: run = 5; case_val = 24; break; 00517 case 3: run = 3; case_val = 30; break; 00518 case 4: run = 1; case_val = 30; break; 00519 case 5: run = 1; case_val = 8; break; 00520 default: run = 1; case_val = 8; break; 00521 } 00522 } 00523 for (k = 0; k < run; k++) 00524 if (j + k < 128) 00525 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) 00526 if (k > 0) { 00527 SAMPLES_NEEDED 00528 //not debugged, almost never used 00529 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t)); 00530 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); 00531 } 00532 j += run; 00533 } 00534 } 00535 } 00536 00537 00545 static void fill_tone_level_array (QDM2Context *q, int flag) 00546 { 00547 int i, sb, ch, sb_used; 00548 int tmp, tab; 00549 00550 // This should never happen 00551 if (q->nb_channels <= 0) 00552 return; 00553 00554 for (ch = 0; ch < q->nb_channels; ch++) 00555 for (sb = 0; sb < 30; sb++) 00556 for (i = 0; i < 8; i++) { 00557 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) 00558 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ 00559 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 00560 else 00561 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 00562 if(tmp < 0) 00563 tmp += 0xff; 00564 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; 00565 } 00566 00567 sb_used = QDM2_SB_USED(q->sub_sampling); 00568 00569 if ((q->superblocktype_2_3 != 0) && !flag) { 00570 for (sb = 0; sb < sb_used; sb++) 00571 for (ch = 0; ch < q->nb_channels; ch++) 00572 for (i = 0; i < 64; i++) { 00573 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 00574 if (q->tone_level_idx[ch][sb][i] < 0) 00575 q->tone_level[ch][sb][i] = 0; 00576 else 00577 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; 00578 } 00579 } else { 00580 tab = q->superblocktype_2_3 ? 0 : 1; 00581 for (sb = 0; sb < sb_used; sb++) { 00582 if ((sb >= 4) && (sb <= 23)) { 00583 for (ch = 0; ch < q->nb_channels; ch++) 00584 for (i = 0; i < 64; i++) { 00585 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 00586 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - 00587 q->tone_level_idx_mid[ch][sb - 4][i / 8] - 00588 q->tone_level_idx_hi2[ch][sb - 4]; 00589 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 00590 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 00591 q->tone_level[ch][sb][i] = 0; 00592 else 00593 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 00594 } 00595 } else { 00596 if (sb > 4) { 00597 for (ch = 0; ch < q->nb_channels; ch++) 00598 for (i = 0; i < 64; i++) { 00599 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 00600 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - 00601 q->tone_level_idx_hi2[ch][sb - 4]; 00602 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 00603 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 00604 q->tone_level[ch][sb][i] = 0; 00605 else 00606 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 00607 } 00608 } else { 00609 for (ch = 0; ch < q->nb_channels; ch++) 00610 for (i = 0; i < 64; i++) { 00611 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 00612 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 00613 q->tone_level[ch][sb][i] = 0; 00614 else 00615 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 00616 } 00617 } 00618 } 00619 } 00620 } 00621 00622 return; 00623 } 00624 00625 00640 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, 00641 sb_int8_array coding_method, int nb_channels, 00642 int c, int superblocktype_2_3, int cm_table_select) 00643 { 00644 int ch, sb, j; 00645 int tmp, acc, esp_40, comp; 00646 int add1, add2, add3, add4; 00647 int64_t multres; 00648 00649 // This should never happen 00650 if (nb_channels <= 0) 00651 return; 00652 00653 if (!superblocktype_2_3) { 00654 /* This case is untested, no samples available */ 00655 SAMPLES_NEEDED 00656 for (ch = 0; ch < nb_channels; ch++) 00657 for (sb = 0; sb < 30; sb++) { 00658 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer 00659 add1 = tone_level_idx[ch][sb][j] - 10; 00660 if (add1 < 0) 00661 add1 = 0; 00662 add2 = add3 = add4 = 0; 00663 if (sb > 1) { 00664 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; 00665 if (add2 < 0) 00666 add2 = 0; 00667 } 00668 if (sb > 0) { 00669 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; 00670 if (add3 < 0) 00671 add3 = 0; 00672 } 00673 if (sb < 29) { 00674 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; 00675 if (add4 < 0) 00676 add4 = 0; 00677 } 00678 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; 00679 if (tmp < 0) 00680 tmp = 0; 00681 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; 00682 } 00683 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; 00684 } 00685 acc = 0; 00686 for (ch = 0; ch < nb_channels; ch++) 00687 for (sb = 0; sb < 30; sb++) 00688 for (j = 0; j < 64; j++) 00689 acc += tone_level_idx_temp[ch][sb][j]; 00690 00691 multres = 0x66666667 * (acc * 10); 00692 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); 00693 for (ch = 0; ch < nb_channels; ch++) 00694 for (sb = 0; sb < 30; sb++) 00695 for (j = 0; j < 64; j++) { 00696 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; 00697 if (comp < 0) 00698 comp += 0xff; 00699 comp /= 256; // signed shift 00700 switch(sb) { 00701 case 0: 00702 if (comp < 30) 00703 comp = 30; 00704 comp += 15; 00705 break; 00706 case 1: 00707 if (comp < 24) 00708 comp = 24; 00709 comp += 10; 00710 break; 00711 case 2: 00712 case 3: 00713 case 4: 00714 if (comp < 16) 00715 comp = 16; 00716 } 00717 if (comp <= 5) 00718 tmp = 0; 00719 else if (comp <= 10) 00720 tmp = 10; 00721 else if (comp <= 16) 00722 tmp = 16; 00723 else if (comp <= 24) 00724 tmp = -1; 00725 else 00726 tmp = 0; 00727 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; 00728 } 00729 for (sb = 0; sb < 30; sb++) 00730 fix_coding_method_array(sb, nb_channels, coding_method); 00731 for (ch = 0; ch < nb_channels; ch++) 00732 for (sb = 0; sb < 30; sb++) 00733 for (j = 0; j < 64; j++) 00734 if (sb >= 10) { 00735 if (coding_method[ch][sb][j] < 10) 00736 coding_method[ch][sb][j] = 10; 00737 } else { 00738 if (sb >= 2) { 00739 if (coding_method[ch][sb][j] < 16) 00740 coding_method[ch][sb][j] = 16; 00741 } else { 00742 if (coding_method[ch][sb][j] < 30) 00743 coding_method[ch][sb][j] = 30; 00744 } 00745 } 00746 } else { // superblocktype_2_3 != 0 00747 for (ch = 0; ch < nb_channels; ch++) 00748 for (sb = 0; sb < 30; sb++) 00749 for (j = 0; j < 64; j++) 00750 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; 00751 } 00752 00753 return; 00754 } 00755 00756 00768 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) 00769 { 00770 int sb, j, k, n, ch, run, channels; 00771 int joined_stereo, zero_encoding, chs; 00772 int type34_first; 00773 float type34_div = 0; 00774 float type34_predictor; 00775 float samples[10], sign_bits[16]; 00776 00777 if (length == 0) { 00778 // If no data use noise 00779 for (sb=sb_min; sb < sb_max; sb++) 00780 build_sb_samples_from_noise (q, sb); 00781 00782 return; 00783 } 00784 00785 for (sb = sb_min; sb < sb_max; sb++) { 00786 FIX_NOISE_IDX(q->noise_idx); 00787 00788 channels = q->nb_channels; 00789 00790 if (q->nb_channels <= 1 || sb < 12) 00791 joined_stereo = 0; 00792 else if (sb >= 24) 00793 joined_stereo = 1; 00794 else 00795 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0; 00796 00797 if (joined_stereo) { 00798 if (BITS_LEFT(length,gb) >= 16) 00799 for (j = 0; j < 16; j++) 00800 sign_bits[j] = get_bits1 (gb); 00801 00802 for (j = 0; j < 64; j++) 00803 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) 00804 q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; 00805 00806 fix_coding_method_array(sb, q->nb_channels, q->coding_method); 00807 channels = 1; 00808 } 00809 00810 for (ch = 0; ch < channels; ch++) { 00811 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; 00812 type34_predictor = 0.0; 00813 type34_first = 1; 00814 00815 for (j = 0; j < 128; ) { 00816 switch (q->coding_method[ch][sb][j / 2]) { 00817 case 8: 00818 if (BITS_LEFT(length,gb) >= 10) { 00819 if (zero_encoding) { 00820 for (k = 0; k < 5; k++) { 00821 if ((j + 2 * k) >= 128) 00822 break; 00823 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; 00824 } 00825 } else { 00826 n = get_bits(gb, 8); 00827 for (k = 0; k < 5; k++) 00828 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 00829 } 00830 for (k = 0; k < 5; k++) 00831 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); 00832 } else { 00833 for (k = 0; k < 10; k++) 00834 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 00835 } 00836 run = 10; 00837 break; 00838 00839 case 10: 00840 if (BITS_LEFT(length,gb) >= 1) { 00841 float f = 0.81; 00842 00843 if (get_bits1(gb)) 00844 f = -f; 00845 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; 00846 samples[0] = f; 00847 } else { 00848 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00849 } 00850 run = 1; 00851 break; 00852 00853 case 16: 00854 if (BITS_LEFT(length,gb) >= 10) { 00855 if (zero_encoding) { 00856 for (k = 0; k < 5; k++) { 00857 if ((j + k) >= 128) 00858 break; 00859 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; 00860 } 00861 } else { 00862 n = get_bits (gb, 8); 00863 for (k = 0; k < 5; k++) 00864 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 00865 } 00866 } else { 00867 for (k = 0; k < 5; k++) 00868 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 00869 } 00870 run = 5; 00871 break; 00872 00873 case 24: 00874 if (BITS_LEFT(length,gb) >= 7) { 00875 n = get_bits(gb, 7); 00876 for (k = 0; k < 3; k++) 00877 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; 00878 } else { 00879 for (k = 0; k < 3; k++) 00880 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 00881 } 00882 run = 3; 00883 break; 00884 00885 case 30: 00886 if (BITS_LEFT(length,gb) >= 4) { 00887 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1); 00888 if (index < FF_ARRAY_ELEMS(type30_dequant)) { 00889 samples[0] = type30_dequant[index]; 00890 } else 00891 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00892 } else 00893 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00894 00895 run = 1; 00896 break; 00897 00898 case 34: 00899 if (BITS_LEFT(length,gb) >= 7) { 00900 if (type34_first) { 00901 type34_div = (float)(1 << get_bits(gb, 2)); 00902 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; 00903 type34_predictor = samples[0]; 00904 type34_first = 0; 00905 } else { 00906 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1); 00907 if (index < FF_ARRAY_ELEMS(type34_delta)) { 00908 samples[0] = type34_delta[index] / type34_div + type34_predictor; 00909 type34_predictor = samples[0]; 00910 } else 00911 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00912 } 00913 } else { 00914 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00915 } 00916 run = 1; 00917 break; 00918 00919 default: 00920 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00921 run = 1; 00922 break; 00923 } 00924 00925 if (joined_stereo) { 00926 float tmp[10][MPA_MAX_CHANNELS]; 00927 00928 for (k = 0; k < run; k++) { 00929 tmp[k][0] = samples[k]; 00930 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; 00931 } 00932 for (chs = 0; chs < q->nb_channels; chs++) 00933 for (k = 0; k < run; k++) 00934 if ((j + k) < 128) 00935 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5); 00936 } else { 00937 for (k = 0; k < run; k++) 00938 if ((j + k) < 128) 00939 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5); 00940 } 00941 00942 j += run; 00943 } // j loop 00944 } // channel loop 00945 } // subband loop 00946 } 00947 00948 00959 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) 00960 { 00961 int i, k, run, level, diff; 00962 00963 if (BITS_LEFT(length,gb) < 16) 00964 return; 00965 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); 00966 00967 quantized_coeffs[0] = level; 00968 00969 for (i = 0; i < 7; ) { 00970 if (BITS_LEFT(length,gb) < 16) 00971 break; 00972 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; 00973 00974 if (BITS_LEFT(length,gb) < 16) 00975 break; 00976 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); 00977 00978 for (k = 1; k <= run; k++) 00979 quantized_coeffs[i + k] = (level + ((k * diff) / run)); 00980 00981 level += diff; 00982 i += run; 00983 } 00984 } 00985 00986 00996 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) 00997 { 00998 int sb, j, k, n, ch; 00999 01000 for (ch = 0; ch < q->nb_channels; ch++) { 01001 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length); 01002 01003 if (BITS_LEFT(length,gb) < 16) { 01004 memset(q->quantized_coeffs[ch][0], 0, 8); 01005 break; 01006 } 01007 } 01008 01009 n = q->sub_sampling + 1; 01010 01011 for (sb = 0; sb < n; sb++) 01012 for (ch = 0; ch < q->nb_channels; ch++) 01013 for (j = 0; j < 8; j++) { 01014 if (BITS_LEFT(length,gb) < 1) 01015 break; 01016 if (get_bits1(gb)) { 01017 for (k=0; k < 8; k++) { 01018 if (BITS_LEFT(length,gb) < 16) 01019 break; 01020 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); 01021 } 01022 } else { 01023 for (k=0; k < 8; k++) 01024 q->tone_level_idx_hi1[ch][sb][j][k] = 0; 01025 } 01026 } 01027 01028 n = QDM2_SB_USED(q->sub_sampling) - 4; 01029 01030 for (sb = 0; sb < n; sb++) 01031 for (ch = 0; ch < q->nb_channels; ch++) { 01032 if (BITS_LEFT(length,gb) < 16) 01033 break; 01034 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); 01035 if (sb > 19) 01036 q->tone_level_idx_hi2[ch][sb] -= 16; 01037 else 01038 for (j = 0; j < 8; j++) 01039 q->tone_level_idx_mid[ch][sb][j] = -16; 01040 } 01041 01042 n = QDM2_SB_USED(q->sub_sampling) - 5; 01043 01044 for (sb = 0; sb < n; sb++) 01045 for (ch = 0; ch < q->nb_channels; ch++) 01046 for (j = 0; j < 8; j++) { 01047 if (BITS_LEFT(length,gb) < 16) 01048 break; 01049 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; 01050 } 01051 } 01052 01059 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) 01060 { 01061 GetBitContext gb; 01062 int i, j, k, n, ch, run, level, diff; 01063 01064 init_get_bits(&gb, node->packet->data, node->packet->size*8); 01065 01066 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function 01067 01068 for (i = 1; i < n; i++) 01069 for (ch=0; ch < q->nb_channels; ch++) { 01070 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); 01071 q->quantized_coeffs[ch][i][0] = level; 01072 01073 for (j = 0; j < (8 - 1); ) { 01074 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; 01075 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); 01076 01077 for (k = 1; k <= run; k++) 01078 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); 01079 01080 level += diff; 01081 j += run; 01082 } 01083 } 01084 01085 for (ch = 0; ch < q->nb_channels; ch++) 01086 for (i = 0; i < 8; i++) 01087 q->quantized_coeffs[ch][0][i] = 0; 01088 } 01089 01090 01098 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) 01099 { 01100 GetBitContext gb; 01101 01102 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 01103 01104 if (length != 0) { 01105 init_tone_level_dequantization(q, &gb, length); 01106 fill_tone_level_array(q, 1); 01107 } else { 01108 fill_tone_level_array(q, 0); 01109 } 01110 } 01111 01112 01120 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length) 01121 { 01122 GetBitContext gb; 01123 01124 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 01125 if (length >= 32) { 01126 int c = get_bits (&gb, 13); 01127 01128 if (c > 3) 01129 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, 01130 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select); 01131 } 01132 01133 synthfilt_build_sb_samples(q, &gb, length, 0, 8); 01134 } 01135 01136 01144 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) 01145 { 01146 GetBitContext gb; 01147 01148 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 01149 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); 01150 } 01151 01152 /* 01153 * Process new subpackets for synthesis filter 01154 * 01155 * @param q context 01156 * @param list list with synthesis filter packets (list D) 01157 */ 01158 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) 01159 { 01160 QDM2SubPNode *nodes[4]; 01161 01162 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); 01163 if (nodes[0] != NULL) 01164 process_subpacket_9(q, nodes[0]); 01165 01166 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); 01167 if (nodes[1] != NULL) 01168 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3); 01169 else 01170 process_subpacket_10(q, NULL, 0); 01171 01172 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); 01173 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) 01174 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3)); 01175 else 01176 process_subpacket_11(q, NULL, 0); 01177 01178 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); 01179 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) 01180 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3)); 01181 else 01182 process_subpacket_12(q, NULL, 0); 01183 } 01184 01185 01186 /* 01187 * Decode superblock, fill packet lists. 01188 * 01189 * @param q context 01190 */ 01191 static void qdm2_decode_super_block (QDM2Context *q) 01192 { 01193 GetBitContext gb; 01194 QDM2SubPacket header, *packet; 01195 int i, packet_bytes, sub_packet_size, sub_packets_D; 01196 unsigned int next_index = 0; 01197 01198 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); 01199 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); 01200 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); 01201 01202 q->sub_packets_B = 0; 01203 sub_packets_D = 0; 01204 01205 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] 01206 01207 init_get_bits(&gb, q->compressed_data, q->compressed_size*8); 01208 qdm2_decode_sub_packet_header(&gb, &header); 01209 01210 if (header.type < 2 || header.type >= 8) { 01211 q->has_errors = 1; 01212 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); 01213 return; 01214 } 01215 01216 q->superblocktype_2_3 = (header.type == 2 || header.type == 3); 01217 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); 01218 01219 init_get_bits(&gb, header.data, header.size*8); 01220 01221 if (header.type == 2 || header.type == 4 || header.type == 5) { 01222 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8); 01223 01224 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); 01225 01226 if (csum != 0) { 01227 q->has_errors = 1; 01228 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); 01229 return; 01230 } 01231 } 01232 01233 q->sub_packet_list_B[0].packet = NULL; 01234 q->sub_packet_list_D[0].packet = NULL; 01235 01236 for (i = 0; i < 6; i++) 01237 if (--q->fft_level_exp[i] < 0) 01238 q->fft_level_exp[i] = 0; 01239 01240 for (i = 0; packet_bytes > 0; i++) { 01241 int j; 01242 01243 q->sub_packet_list_A[i].next = NULL; 01244 01245 if (i > 0) { 01246 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; 01247 01248 /* seek to next block */ 01249 init_get_bits(&gb, header.data, header.size*8); 01250 skip_bits(&gb, next_index*8); 01251 01252 if (next_index >= header.size) 01253 break; 01254 } 01255 01256 /* decode subpacket */ 01257 packet = &q->sub_packets[i]; 01258 qdm2_decode_sub_packet_header(&gb, packet); 01259 next_index = packet->size + get_bits_count(&gb) / 8; 01260 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; 01261 01262 if (packet->type == 0) 01263 break; 01264 01265 if (sub_packet_size > packet_bytes) { 01266 if (packet->type != 10 && packet->type != 11 && packet->type != 12) 01267 break; 01268 packet->size += packet_bytes - sub_packet_size; 01269 } 01270 01271 packet_bytes -= sub_packet_size; 01272 01273 /* add subpacket to 'all subpackets' list */ 01274 q->sub_packet_list_A[i].packet = packet; 01275 01276 /* add subpacket to related list */ 01277 if (packet->type == 8) { 01278 SAMPLES_NEEDED_2("packet type 8"); 01279 return; 01280 } else if (packet->type >= 9 && packet->type <= 12) { 01281 /* packets for MPEG Audio like Synthesis Filter */ 01282 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); 01283 } else if (packet->type == 13) { 01284 for (j = 0; j < 6; j++) 01285 q->fft_level_exp[j] = get_bits(&gb, 6); 01286 } else if (packet->type == 14) { 01287 for (j = 0; j < 6; j++) 01288 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); 01289 } else if (packet->type == 15) { 01290 SAMPLES_NEEDED_2("packet type 15") 01291 return; 01292 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { 01293 /* packets for FFT */ 01294 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); 01295 } 01296 } // Packet bytes loop 01297 01298 /* **************************************************************** */ 01299 if (q->sub_packet_list_D[0].packet != NULL) { 01300 process_synthesis_subpackets(q, q->sub_packet_list_D); 01301 q->do_synth_filter = 1; 01302 } else if (q->do_synth_filter) { 01303 process_subpacket_10(q, NULL, 0); 01304 process_subpacket_11(q, NULL, 0); 01305 process_subpacket_12(q, NULL, 0); 01306 } 01307 /* **************************************************************** */ 01308 } 01309 01310 01311 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, 01312 int offset, int duration, int channel, 01313 int exp, int phase) 01314 { 01315 if (q->fft_coefs_min_index[duration] < 0) 01316 q->fft_coefs_min_index[duration] = q->fft_coefs_index; 01317 01318 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); 01319 q->fft_coefs[q->fft_coefs_index].channel = channel; 01320 q->fft_coefs[q->fft_coefs_index].offset = offset; 01321 q->fft_coefs[q->fft_coefs_index].exp = exp; 01322 q->fft_coefs[q->fft_coefs_index].phase = phase; 01323 q->fft_coefs_index++; 01324 } 01325 01326 01327 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) 01328 { 01329 int channel, stereo, phase, exp; 01330 int local_int_4, local_int_8, stereo_phase, local_int_10; 01331 int local_int_14, stereo_exp, local_int_20, local_int_28; 01332 int n, offset; 01333 01334 local_int_4 = 0; 01335 local_int_28 = 0; 01336 local_int_20 = 2; 01337 local_int_8 = (4 - duration); 01338 local_int_10 = 1 << (q->group_order - duration - 1); 01339 offset = 1; 01340 01341 while (1) { 01342 if (q->superblocktype_2_3) { 01343 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { 01344 offset = 1; 01345 if (n == 0) { 01346 local_int_4 += local_int_10; 01347 local_int_28 += (1 << local_int_8); 01348 } else { 01349 local_int_4 += 8*local_int_10; 01350 local_int_28 += (8 << local_int_8); 01351 } 01352 } 01353 offset += (n - 2); 01354 } else { 01355 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); 01356 while (offset >= (local_int_10 - 1)) { 01357 offset += (1 - (local_int_10 - 1)); 01358 local_int_4 += local_int_10; 01359 local_int_28 += (1 << local_int_8); 01360 } 01361 } 01362 01363 if (local_int_4 >= q->group_size) 01364 return; 01365 01366 local_int_14 = (offset >> local_int_8); 01367 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table)) 01368 return; 01369 01370 if (q->nb_channels > 1) { 01371 channel = get_bits1(gb); 01372 stereo = get_bits1(gb); 01373 } else { 01374 channel = 0; 01375 stereo = 0; 01376 } 01377 01378 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); 01379 exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; 01380 exp = (exp < 0) ? 0 : exp; 01381 01382 phase = get_bits(gb, 3); 01383 stereo_exp = 0; 01384 stereo_phase = 0; 01385 01386 if (stereo) { 01387 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); 01388 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); 01389 if (stereo_phase < 0) 01390 stereo_phase += 8; 01391 } 01392 01393 if (q->frequency_range > (local_int_14 + 1)) { 01394 int sub_packet = (local_int_20 + local_int_28); 01395 01396 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); 01397 if (stereo) 01398 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); 01399 } 01400 01401 offset++; 01402 } 01403 } 01404 01405 01406 static void qdm2_decode_fft_packets (QDM2Context *q) 01407 { 01408 int i, j, min, max, value, type, unknown_flag; 01409 GetBitContext gb; 01410 01411 if (q->sub_packet_list_B[0].packet == NULL) 01412 return; 01413 01414 /* reset minimum indexes for FFT coefficients */ 01415 q->fft_coefs_index = 0; 01416 for (i=0; i < 5; i++) 01417 q->fft_coefs_min_index[i] = -1; 01418 01419 /* process subpackets ordered by type, largest type first */ 01420 for (i = 0, max = 256; i < q->sub_packets_B; i++) { 01421 QDM2SubPacket *packet= NULL; 01422 01423 /* find subpacket with largest type less than max */ 01424 for (j = 0, min = 0; j < q->sub_packets_B; j++) { 01425 value = q->sub_packet_list_B[j].packet->type; 01426 if (value > min && value < max) { 01427 min = value; 01428 packet = q->sub_packet_list_B[j].packet; 01429 } 01430 } 01431 01432 max = min; 01433 01434 /* check for errors (?) */ 01435 if (!packet) 01436 return; 01437 01438 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) 01439 return; 01440 01441 /* decode FFT tones */ 01442 init_get_bits (&gb, packet->data, packet->size*8); 01443 01444 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) 01445 unknown_flag = 1; 01446 else 01447 unknown_flag = 0; 01448 01449 type = packet->type; 01450 01451 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { 01452 int duration = q->sub_sampling + 5 - (type & 15); 01453 01454 if (duration >= 0 && duration < 4) 01455 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); 01456 } else if (type == 31) { 01457 for (j=0; j < 4; j++) 01458 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 01459 } else if (type == 46) { 01460 for (j=0; j < 6; j++) 01461 q->fft_level_exp[j] = get_bits(&gb, 6); 01462 for (j=0; j < 4; j++) 01463 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 01464 } 01465 } // Loop on B packets 01466 01467 /* calculate maximum indexes for FFT coefficients */ 01468 for (i = 0, j = -1; i < 5; i++) 01469 if (q->fft_coefs_min_index[i] >= 0) { 01470 if (j >= 0) 01471 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; 01472 j = i; 01473 } 01474 if (j >= 0) 01475 q->fft_coefs_max_index[j] = q->fft_coefs_index; 01476 } 01477 01478 01479 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) 01480 { 01481 float level, f[6]; 01482 int i; 01483 QDM2Complex c; 01484 const double iscale = 2.0*M_PI / 512.0; 01485 01486 tone->phase += tone->phase_shift; 01487 01488 /* calculate current level (maximum amplitude) of tone */ 01489 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; 01490 c.im = level * sin(tone->phase*iscale); 01491 c.re = level * cos(tone->phase*iscale); 01492 01493 /* generate FFT coefficients for tone */ 01494 if (tone->duration >= 3 || tone->cutoff >= 3) { 01495 tone->complex[0].im += c.im; 01496 tone->complex[0].re += c.re; 01497 tone->complex[1].im -= c.im; 01498 tone->complex[1].re -= c.re; 01499 } else { 01500 f[1] = -tone->table[4]; 01501 f[0] = tone->table[3] - tone->table[0]; 01502 f[2] = 1.0 - tone->table[2] - tone->table[3]; 01503 f[3] = tone->table[1] + tone->table[4] - 1.0; 01504 f[4] = tone->table[0] - tone->table[1]; 01505 f[5] = tone->table[2]; 01506 for (i = 0; i < 2; i++) { 01507 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; 01508 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); 01509 } 01510 for (i = 0; i < 4; i++) { 01511 tone->complex[i].re += c.re * f[i+2]; 01512 tone->complex[i].im += c.im * f[i+2]; 01513 } 01514 } 01515 01516 /* copy the tone if it has not yet died out */ 01517 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { 01518 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); 01519 q->fft_tone_end = (q->fft_tone_end + 1) % 1000; 01520 } 01521 } 01522 01523 01524 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) 01525 { 01526 int i, j, ch; 01527 const double iscale = 0.25 * M_PI; 01528 01529 for (ch = 0; ch < q->channels; ch++) { 01530 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); 01531 } 01532 01533 01534 /* apply FFT tones with duration 4 (1 FFT period) */ 01535 if (q->fft_coefs_min_index[4] >= 0) 01536 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { 01537 float level; 01538 QDM2Complex c; 01539 01540 if (q->fft_coefs[i].sub_packet != sub_packet) 01541 break; 01542 01543 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; 01544 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]; 01545 01546 c.re = level * cos(q->fft_coefs[i].phase * iscale); 01547 c.im = level * sin(q->fft_coefs[i].phase * iscale); 01548 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; 01549 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; 01550 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; 01551 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; 01552 } 01553 01554 /* generate existing FFT tones */ 01555 for (i = q->fft_tone_end; i != q->fft_tone_start; ) { 01556 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); 01557 q->fft_tone_start = (q->fft_tone_start + 1) % 1000; 01558 } 01559 01560 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ 01561 for (i = 0; i < 4; i++) 01562 if (q->fft_coefs_min_index[i] >= 0) { 01563 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { 01564 int offset, four_i; 01565 FFTTone tone; 01566 01567 if (q->fft_coefs[j].sub_packet != sub_packet) 01568 break; 01569 01570 four_i = (4 - i); 01571 offset = q->fft_coefs[j].offset >> four_i; 01572 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; 01573 01574 if (offset < q->frequency_range) { 01575 if (offset < 2) 01576 tone.cutoff = offset; 01577 else 01578 tone.cutoff = (offset >= 60) ? 3 : 2; 01579 01580 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]; 01581 tone.complex = &q->fft.complex[ch][offset]; 01582 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; 01583 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; 01584 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); 01585 tone.duration = i; 01586 tone.time_index = 0; 01587 01588 qdm2_fft_generate_tone(q, &tone); 01589 } 01590 } 01591 q->fft_coefs_min_index[i] = j; 01592 } 01593 } 01594 01595 01596 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) 01597 { 01598 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; 01599 int i; 01600 q->fft.complex[channel][0].re *= 2.0f; 01601 q->fft.complex[channel][0].im = 0.0f; 01602 ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); 01603 /* add samples to output buffer */ 01604 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) 01605 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; 01606 } 01607 01608 01613 static void qdm2_synthesis_filter (QDM2Context *q, int index) 01614 { 01615 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; 01616 int i, k, ch, sb_used, sub_sampling, dither_state = 0; 01617 01618 /* copy sb_samples */ 01619 sb_used = QDM2_SB_USED(q->sub_sampling); 01620 01621 for (ch = 0; ch < q->channels; ch++) 01622 for (i = 0; i < 8; i++) 01623 for (k=sb_used; k < SBLIMIT; k++) 01624 q->sb_samples[ch][(8 * index) + i][k] = 0; 01625 01626 for (ch = 0; ch < q->nb_channels; ch++) { 01627 OUT_INT *samples_ptr = samples + ch; 01628 01629 for (i = 0; i < 8; i++) { 01630 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]), 01631 ff_mpa_synth_window, &dither_state, 01632 samples_ptr, q->nb_channels, 01633 q->sb_samples[ch][(8 * index) + i]); 01634 samples_ptr += 32 * q->nb_channels; 01635 } 01636 } 01637 01638 /* add samples to output buffer */ 01639 sub_sampling = (4 >> q->sub_sampling); 01640 01641 for (ch = 0; ch < q->channels; ch++) 01642 for (i = 0; i < q->frame_size; i++) 01643 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16)); 01644 } 01645 01646 01652 static av_cold void qdm2_init(QDM2Context *q) { 01653 static int initialized = 0; 01654 01655 if (initialized != 0) 01656 return; 01657 initialized = 1; 01658 01659 qdm2_init_vlc(); 01660 ff_mpa_synth_init(ff_mpa_synth_window); 01661 softclip_table_init(); 01662 rnd_table_init(); 01663 init_noise_samples(); 01664 01665 av_log(NULL, AV_LOG_DEBUG, "init done\n"); 01666 } 01667 01668 01669 #if 0 01670 static void dump_context(QDM2Context *q) 01671 { 01672 int i; 01673 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b); 01674 PRINT("compressed_data",q->compressed_data); 01675 PRINT("compressed_size",q->compressed_size); 01676 PRINT("frame_size",q->frame_size); 01677 PRINT("checksum_size",q->checksum_size); 01678 PRINT("channels",q->channels); 01679 PRINT("nb_channels",q->nb_channels); 01680 PRINT("fft_frame_size",q->fft_frame_size); 01681 PRINT("fft_size",q->fft_size); 01682 PRINT("sub_sampling",q->sub_sampling); 01683 PRINT("fft_order",q->fft_order); 01684 PRINT("group_order",q->group_order); 01685 PRINT("group_size",q->group_size); 01686 PRINT("sub_packet",q->sub_packet); 01687 PRINT("frequency_range",q->frequency_range); 01688 PRINT("has_errors",q->has_errors); 01689 PRINT("fft_tone_end",q->fft_tone_end); 01690 PRINT("fft_tone_start",q->fft_tone_start); 01691 PRINT("fft_coefs_index",q->fft_coefs_index); 01692 PRINT("coeff_per_sb_select",q->coeff_per_sb_select); 01693 PRINT("cm_table_select",q->cm_table_select); 01694 PRINT("noise_idx",q->noise_idx); 01695 01696 for (i = q->fft_tone_start; i < q->fft_tone_end; i++) 01697 { 01698 FFTTone *t = &q->fft_tones[i]; 01699 01700 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i); 01701 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level); 01702 // PRINT(" level", t->level); 01703 PRINT(" phase", t->phase); 01704 PRINT(" phase_shift", t->phase_shift); 01705 PRINT(" duration", t->duration); 01706 PRINT(" samples_im", t->samples_im); 01707 PRINT(" samples_re", t->samples_re); 01708 PRINT(" table", t->table); 01709 } 01710 01711 } 01712 #endif 01713 01714 01718 static av_cold int qdm2_decode_init(AVCodecContext *avctx) 01719 { 01720 QDM2Context *s = avctx->priv_data; 01721 uint8_t *extradata; 01722 int extradata_size; 01723 int tmp_val, tmp, size; 01724 01725 /* extradata parsing 01726 01727 Structure: 01728 wave { 01729 frma (QDM2) 01730 QDCA 01731 QDCP 01732 } 01733 01734 32 size (including this field) 01735 32 tag (=frma) 01736 32 type (=QDM2 or QDMC) 01737 01738 32 size (including this field, in bytes) 01739 32 tag (=QDCA) // maybe mandatory parameters 01740 32 unknown (=1) 01741 32 channels (=2) 01742 32 samplerate (=44100) 01743 32 bitrate (=96000) 01744 32 block size (=4096) 01745 32 frame size (=256) (for one channel) 01746 32 packet size (=1300) 01747 01748 32 size (including this field, in bytes) 01749 32 tag (=QDCP) // maybe some tuneable parameters 01750 32 float1 (=1.0) 01751 32 zero ? 01752 32 float2 (=1.0) 01753 32 float3 (=1.0) 01754 32 unknown (27) 01755 32 unknown (8) 01756 32 zero ? 01757 */ 01758 01759 if (!avctx->extradata || (avctx->extradata_size < 48)) { 01760 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); 01761 return -1; 01762 } 01763 01764 extradata = avctx->extradata; 01765 extradata_size = avctx->extradata_size; 01766 01767 while (extradata_size > 7) { 01768 if (!memcmp(extradata, "frmaQDM", 7)) 01769 break; 01770 extradata++; 01771 extradata_size--; 01772 } 01773 01774 if (extradata_size < 12) { 01775 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", 01776 extradata_size); 01777 return -1; 01778 } 01779 01780 if (memcmp(extradata, "frmaQDM", 7)) { 01781 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); 01782 return -1; 01783 } 01784 01785 if (extradata[7] == 'C') { 01786 // s->is_qdmc = 1; 01787 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); 01788 return -1; 01789 } 01790 01791 extradata += 8; 01792 extradata_size -= 8; 01793 01794 size = AV_RB32(extradata); 01795 01796 if(size > extradata_size){ 01797 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", 01798 extradata_size, size); 01799 return -1; 01800 } 01801 01802 extradata += 4; 01803 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); 01804 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { 01805 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); 01806 return -1; 01807 } 01808 01809 extradata += 8; 01810 01811 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); 01812 extradata += 4; 01813 if (s->channels > MPA_MAX_CHANNELS) 01814 return AVERROR_INVALIDDATA; 01815 01816 avctx->sample_rate = AV_RB32(extradata); 01817 extradata += 4; 01818 01819 avctx->bit_rate = AV_RB32(extradata); 01820 extradata += 4; 01821 01822 s->group_size = AV_RB32(extradata); 01823 extradata += 4; 01824 01825 s->fft_size = AV_RB32(extradata); 01826 extradata += 4; 01827 01828 s->checksum_size = AV_RB32(extradata); 01829 01830 s->fft_order = av_log2(s->fft_size) + 1; 01831 s->fft_frame_size = 2 * s->fft_size; // complex has two floats 01832 01833 // something like max decodable tones 01834 s->group_order = av_log2(s->group_size) + 1; 01835 s->frame_size = s->group_size / 16; // 16 iterations per super block 01836 if (s->frame_size > QDM2_MAX_FRAME_SIZE) 01837 return AVERROR_INVALIDDATA; 01838 01839 s->sub_sampling = s->fft_order - 7; 01840 s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); 01841 01842 switch ((s->sub_sampling * 2 + s->channels - 1)) { 01843 case 0: tmp = 40; break; 01844 case 1: tmp = 48; break; 01845 case 2: tmp = 56; break; 01846 case 3: tmp = 72; break; 01847 case 4: tmp = 80; break; 01848 case 5: tmp = 100;break; 01849 default: tmp=s->sub_sampling; break; 01850 } 01851 tmp_val = 0; 01852 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; 01853 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; 01854 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; 01855 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; 01856 s->cm_table_select = tmp_val; 01857 01858 if (s->sub_sampling == 0) 01859 tmp = 7999; 01860 else 01861 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; 01862 /* 01863 0: 7999 -> 0 01864 1: 20000 -> 2 01865 2: 28000 -> 2 01866 */ 01867 if (tmp < 8000) 01868 s->coeff_per_sb_select = 0; 01869 else if (tmp <= 16000) 01870 s->coeff_per_sb_select = 1; 01871 else 01872 s->coeff_per_sb_select = 2; 01873 01874 // Fail on unknown fft order 01875 if ((s->fft_order < 7) || (s->fft_order > 9)) { 01876 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); 01877 return -1; 01878 } 01879 01880 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); 01881 01882 qdm2_init(s); 01883 01884 avctx->sample_fmt = SAMPLE_FMT_S16; 01885 01886 // dump_context(s); 01887 return 0; 01888 } 01889 01890 01891 static av_cold int qdm2_decode_close(AVCodecContext *avctx) 01892 { 01893 QDM2Context *s = avctx->priv_data; 01894 01895 ff_rdft_end(&s->rdft_ctx); 01896 01897 return 0; 01898 } 01899 01900 01901 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) 01902 { 01903 int ch, i; 01904 const int frame_size = (q->frame_size * q->channels); 01905 01906 /* select input buffer */ 01907 q->compressed_data = in; 01908 q->compressed_size = q->checksum_size; 01909 01910 // dump_context(q); 01911 01912 /* copy old block, clear new block of output samples */ 01913 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); 01914 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); 01915 01916 /* decode block of QDM2 compressed data */ 01917 if (q->sub_packet == 0) { 01918 q->has_errors = 0; // zero it for a new super block 01919 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); 01920 qdm2_decode_super_block(q); 01921 } 01922 01923 /* parse subpackets */ 01924 if (!q->has_errors) { 01925 if (q->sub_packet == 2) 01926 qdm2_decode_fft_packets(q); 01927 01928 qdm2_fft_tone_synthesizer(q, q->sub_packet); 01929 } 01930 01931 /* sound synthesis stage 1 (FFT) */ 01932 for (ch = 0; ch < q->channels; ch++) { 01933 qdm2_calculate_fft(q, ch, q->sub_packet); 01934 01935 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { 01936 SAMPLES_NEEDED_2("has errors, and C list is not empty") 01937 return -1; 01938 } 01939 } 01940 01941 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ 01942 if (!q->has_errors && q->do_synth_filter) 01943 qdm2_synthesis_filter(q, q->sub_packet); 01944 01945 q->sub_packet = (q->sub_packet + 1) % 16; 01946 01947 /* clip and convert output float[] to 16bit signed samples */ 01948 for (i = 0; i < frame_size; i++) { 01949 int value = (int)q->output_buffer[i]; 01950 01951 if (value > SOFTCLIP_THRESHOLD) 01952 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; 01953 else if (value < -SOFTCLIP_THRESHOLD) 01954 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; 01955 01956 out[i] = value; 01957 } 01958 01959 return 0; 01960 } 01961 01962 01963 static int qdm2_decode_frame(AVCodecContext *avctx, 01964 void *data, int *data_size, 01965 AVPacket *avpkt) 01966 { 01967 const uint8_t *buf = avpkt->data; 01968 int buf_size = avpkt->size; 01969 QDM2Context *s = avctx->priv_data; 01970 int16_t *out = data; 01971 int i, out_size; 01972 01973 if(!buf) 01974 return 0; 01975 if(buf_size < s->checksum_size) 01976 return -1; 01977 01978 out_size = 16 * s->channels * s->frame_size * 01979 av_get_bits_per_sample_format(avctx->sample_fmt)/8; 01980 if (*data_size < out_size) { 01981 av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n"); 01982 return AVERROR(EINVAL); 01983 } 01984 01985 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", 01986 buf_size, buf, s->checksum_size, data, *data_size); 01987 01988 for (i = 0; i < 16; i++) { 01989 if (qdm2_decode(s, buf, out) < 0) 01990 return -1; 01991 out += s->channels * s->frame_size; 01992 } 01993 01994 *data_size = out_size; 01995 01996 return buf_size; 01997 } 01998 01999 AVCodec qdm2_decoder = 02000 { 02001 .name = "qdm2", 02002 .type = AVMEDIA_TYPE_AUDIO, 02003 .id = CODEC_ID_QDM2, 02004 .priv_data_size = sizeof(QDM2Context), 02005 .init = qdm2_decode_init, 02006 .close = qdm2_decode_close, 02007 .decode = qdm2_decode_frame, 02008 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), 02009 };