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 00079 typedef int8_t sb_int8_array[2][30][64]; 00080 00084 typedef struct { 00085 int type; 00086 unsigned int size; 00087 const uint8_t *data; 00088 } QDM2SubPacket; 00089 00093 typedef struct QDM2SubPNode { 00094 QDM2SubPacket *packet; 00095 struct QDM2SubPNode *next; 00096 } QDM2SubPNode; 00097 00098 typedef struct { 00099 float re; 00100 float im; 00101 } QDM2Complex; 00102 00103 typedef struct { 00104 float level; 00105 QDM2Complex *complex; 00106 const float *table; 00107 int phase; 00108 int phase_shift; 00109 int duration; 00110 short time_index; 00111 short cutoff; 00112 } FFTTone; 00113 00114 typedef struct { 00115 int16_t sub_packet; 00116 uint8_t channel; 00117 int16_t offset; 00118 int16_t exp; 00119 uint8_t phase; 00120 } FFTCoefficient; 00121 00122 typedef struct { 00123 DECLARE_ALIGNED(16, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; 00124 } QDM2FFT; 00125 00129 typedef struct { 00131 int nb_channels; 00132 int channels; 00133 int group_size; 00134 int fft_size; 00135 int checksum_size; 00136 00138 int group_order; 00139 int fft_order; 00140 int fft_frame_size; 00141 int frame_size; 00142 int frequency_range; 00143 int sub_sampling; 00144 int coeff_per_sb_select; 00145 int cm_table_select; 00146 00148 QDM2SubPacket sub_packets[16]; 00149 QDM2SubPNode sub_packet_list_A[16]; 00150 QDM2SubPNode sub_packet_list_B[16]; 00151 int sub_packets_B; 00152 QDM2SubPNode sub_packet_list_C[16]; 00153 QDM2SubPNode sub_packet_list_D[16]; 00154 00156 FFTTone fft_tones[1000]; 00157 int fft_tone_start; 00158 int fft_tone_end; 00159 FFTCoefficient fft_coefs[1000]; 00160 int fft_coefs_index; 00161 int fft_coefs_min_index[5]; 00162 int fft_coefs_max_index[5]; 00163 int fft_level_exp[6]; 00164 RDFTContext rdft_ctx; 00165 QDM2FFT fft; 00166 00168 const uint8_t *compressed_data; 00169 int compressed_size; 00170 float output_buffer[1024]; 00171 00173 DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2]; 00174 int synth_buf_offset[MPA_MAX_CHANNELS]; 00175 DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; 00176 00178 float tone_level[MPA_MAX_CHANNELS][30][64]; 00179 int8_t coding_method[MPA_MAX_CHANNELS][30][64]; 00180 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; 00181 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; 00182 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; 00183 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; 00184 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; 00185 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; 00186 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; 00187 00188 // Flags 00189 int has_errors; 00190 int superblocktype_2_3; 00191 int do_synth_filter; 00192 00193 int sub_packet; 00194 int noise_idx; 00195 } QDM2Context; 00196 00197 00198 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE]; 00199 00200 static VLC vlc_tab_level; 00201 static VLC vlc_tab_diff; 00202 static VLC vlc_tab_run; 00203 static VLC fft_level_exp_alt_vlc; 00204 static VLC fft_level_exp_vlc; 00205 static VLC fft_stereo_exp_vlc; 00206 static VLC fft_stereo_phase_vlc; 00207 static VLC vlc_tab_tone_level_idx_hi1; 00208 static VLC vlc_tab_tone_level_idx_mid; 00209 static VLC vlc_tab_tone_level_idx_hi2; 00210 static VLC vlc_tab_type30; 00211 static VLC vlc_tab_type34; 00212 static VLC vlc_tab_fft_tone_offset[5]; 00213 00214 static const uint16_t qdm2_vlc_offs[] = { 00215 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, 00216 }; 00217 00218 static av_cold void qdm2_init_vlc(void) 00219 { 00220 static int vlcs_initialized = 0; 00221 static VLC_TYPE qdm2_table[3838][2]; 00222 00223 if (!vlcs_initialized) { 00224 00225 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; 00226 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; 00227 init_vlc (&vlc_tab_level, 8, 24, 00228 vlc_tab_level_huffbits, 1, 1, 00229 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00230 00231 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; 00232 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; 00233 init_vlc (&vlc_tab_diff, 8, 37, 00234 vlc_tab_diff_huffbits, 1, 1, 00235 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00236 00237 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; 00238 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; 00239 init_vlc (&vlc_tab_run, 5, 6, 00240 vlc_tab_run_huffbits, 1, 1, 00241 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00242 00243 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; 00244 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3]; 00245 init_vlc (&fft_level_exp_alt_vlc, 8, 28, 00246 fft_level_exp_alt_huffbits, 1, 1, 00247 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00248 00249 00250 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; 00251 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; 00252 init_vlc (&fft_level_exp_vlc, 8, 20, 00253 fft_level_exp_huffbits, 1, 1, 00254 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00255 00256 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; 00257 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5]; 00258 init_vlc (&fft_stereo_exp_vlc, 6, 7, 00259 fft_stereo_exp_huffbits, 1, 1, 00260 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00261 00262 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; 00263 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6]; 00264 init_vlc (&fft_stereo_phase_vlc, 6, 9, 00265 fft_stereo_phase_huffbits, 1, 1, 00266 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00267 00268 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]]; 00269 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7]; 00270 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, 00271 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, 00272 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00273 00274 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]]; 00275 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8]; 00276 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, 00277 vlc_tab_tone_level_idx_mid_huffbits, 1, 1, 00278 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00279 00280 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]]; 00281 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9]; 00282 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, 00283 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, 00284 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00285 00286 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; 00287 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; 00288 init_vlc (&vlc_tab_type30, 6, 9, 00289 vlc_tab_type30_huffbits, 1, 1, 00290 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00291 00292 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; 00293 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; 00294 init_vlc (&vlc_tab_type34, 5, 10, 00295 vlc_tab_type34_huffbits, 1, 1, 00296 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00297 00298 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]]; 00299 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12]; 00300 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, 00301 vlc_tab_fft_tone_offset_0_huffbits, 1, 1, 00302 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00303 00304 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]]; 00305 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13]; 00306 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, 00307 vlc_tab_fft_tone_offset_1_huffbits, 1, 1, 00308 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00309 00310 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]]; 00311 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14]; 00312 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, 00313 vlc_tab_fft_tone_offset_2_huffbits, 1, 1, 00314 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00315 00316 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]]; 00317 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15]; 00318 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, 00319 vlc_tab_fft_tone_offset_3_huffbits, 1, 1, 00320 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00321 00322 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]]; 00323 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16]; 00324 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, 00325 vlc_tab_fft_tone_offset_4_huffbits, 1, 1, 00326 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); 00327 00328 vlcs_initialized=1; 00329 } 00330 } 00331 00332 00333 /* for floating point to fixed point conversion */ 00334 static const float f2i_scale = (float) (1 << (FRAC_BITS - 15)); 00335 00336 00337 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) 00338 { 00339 int value; 00340 00341 value = get_vlc2(gb, vlc->table, vlc->bits, depth); 00342 00343 /* stage-2, 3 bits exponent escape sequence */ 00344 if (value-- == 0) 00345 value = get_bits (gb, get_bits (gb, 3) + 1); 00346 00347 /* stage-3, optional */ 00348 if (flag) { 00349 int tmp = vlc_stage3_values[value]; 00350 00351 if ((value & ~3) > 0) 00352 tmp += get_bits (gb, (value >> 2)); 00353 value = tmp; 00354 } 00355 00356 return value; 00357 } 00358 00359 00360 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) 00361 { 00362 int value = qdm2_get_vlc (gb, vlc, 0, depth); 00363 00364 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); 00365 } 00366 00367 00377 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) { 00378 int i; 00379 00380 for (i=0; i < length; i++) 00381 value -= data[i]; 00382 00383 return (uint16_t)(value & 0xffff); 00384 } 00385 00386 00393 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) 00394 { 00395 sub_packet->type = get_bits (gb, 8); 00396 00397 if (sub_packet->type == 0) { 00398 sub_packet->size = 0; 00399 sub_packet->data = NULL; 00400 } else { 00401 sub_packet->size = get_bits (gb, 8); 00402 00403 if (sub_packet->type & 0x80) { 00404 sub_packet->size <<= 8; 00405 sub_packet->size |= get_bits (gb, 8); 00406 sub_packet->type &= 0x7f; 00407 } 00408 00409 if (sub_packet->type == 0x7f) 00410 sub_packet->type |= (get_bits (gb, 8) << 8); 00411 00412 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data 00413 } 00414 00415 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", 00416 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); 00417 } 00418 00419 00427 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) 00428 { 00429 while (list != NULL && list->packet != NULL) { 00430 if (list->packet->type == type) 00431 return list; 00432 list = list->next; 00433 } 00434 return NULL; 00435 } 00436 00437 00444 static void average_quantized_coeffs (QDM2Context *q) 00445 { 00446 int i, j, n, ch, sum; 00447 00448 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; 00449 00450 for (ch = 0; ch < q->nb_channels; ch++) 00451 for (i = 0; i < n; i++) { 00452 sum = 0; 00453 00454 for (j = 0; j < 8; j++) 00455 sum += q->quantized_coeffs[ch][i][j]; 00456 00457 sum /= 8; 00458 if (sum > 0) 00459 sum--; 00460 00461 for (j=0; j < 8; j++) 00462 q->quantized_coeffs[ch][i][j] = sum; 00463 } 00464 } 00465 00466 00474 static void build_sb_samples_from_noise (QDM2Context *q, int sb) 00475 { 00476 int ch, j; 00477 00478 FIX_NOISE_IDX(q->noise_idx); 00479 00480 if (!q->nb_channels) 00481 return; 00482 00483 for (ch = 0; ch < q->nb_channels; ch++) 00484 for (j = 0; j < 64; j++) { 00485 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); 00486 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); 00487 } 00488 } 00489 00490 00499 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) 00500 { 00501 int j,k; 00502 int ch; 00503 int run, case_val; 00504 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}; 00505 00506 for (ch = 0; ch < channels; ch++) { 00507 for (j = 0; j < 64; ) { 00508 if((coding_method[ch][sb][j] - 8) > 22) { 00509 run = 1; 00510 case_val = 8; 00511 } else { 00512 switch (switchtable[coding_method[ch][sb][j]-8]) { 00513 case 0: run = 10; case_val = 10; break; 00514 case 1: run = 1; case_val = 16; break; 00515 case 2: run = 5; case_val = 24; break; 00516 case 3: run = 3; case_val = 30; break; 00517 case 4: run = 1; case_val = 30; break; 00518 case 5: run = 1; case_val = 8; break; 00519 default: run = 1; case_val = 8; break; 00520 } 00521 } 00522 for (k = 0; k < run; k++) 00523 if (j + k < 128) 00524 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) 00525 if (k > 0) { 00526 SAMPLES_NEEDED 00527 //not debugged, almost never used 00528 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t)); 00529 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); 00530 } 00531 j += run; 00532 } 00533 } 00534 } 00535 00536 00544 static void fill_tone_level_array (QDM2Context *q, int flag) 00545 { 00546 int i, sb, ch, sb_used; 00547 int tmp, tab; 00548 00549 // This should never happen 00550 if (q->nb_channels <= 0) 00551 return; 00552 00553 for (ch = 0; ch < q->nb_channels; ch++) 00554 for (sb = 0; sb < 30; sb++) 00555 for (i = 0; i < 8; i++) { 00556 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) 00557 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ 00558 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 00559 else 00560 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; 00561 if(tmp < 0) 00562 tmp += 0xff; 00563 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; 00564 } 00565 00566 sb_used = QDM2_SB_USED(q->sub_sampling); 00567 00568 if ((q->superblocktype_2_3 != 0) && !flag) { 00569 for (sb = 0; sb < sb_used; sb++) 00570 for (ch = 0; ch < q->nb_channels; ch++) 00571 for (i = 0; i < 64; i++) { 00572 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 00573 if (q->tone_level_idx[ch][sb][i] < 0) 00574 q->tone_level[ch][sb][i] = 0; 00575 else 00576 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; 00577 } 00578 } else { 00579 tab = q->superblocktype_2_3 ? 0 : 1; 00580 for (sb = 0; sb < sb_used; sb++) { 00581 if ((sb >= 4) && (sb <= 23)) { 00582 for (ch = 0; ch < q->nb_channels; ch++) 00583 for (i = 0; i < 64; i++) { 00584 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 00585 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - 00586 q->tone_level_idx_mid[ch][sb - 4][i / 8] - 00587 q->tone_level_idx_hi2[ch][sb - 4]; 00588 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 00589 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 00590 q->tone_level[ch][sb][i] = 0; 00591 else 00592 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 00593 } 00594 } else { 00595 if (sb > 4) { 00596 for (ch = 0; ch < q->nb_channels; ch++) 00597 for (i = 0; i < 64; i++) { 00598 tmp = q->tone_level_idx_base[ch][sb][i / 8] - 00599 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - 00600 q->tone_level_idx_hi2[ch][sb - 4]; 00601 q->tone_level_idx[ch][sb][i] = tmp & 0xff; 00602 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 00603 q->tone_level[ch][sb][i] = 0; 00604 else 00605 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 00606 } 00607 } else { 00608 for (ch = 0; ch < q->nb_channels; ch++) 00609 for (i = 0; i < 64; i++) { 00610 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; 00611 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) 00612 q->tone_level[ch][sb][i] = 0; 00613 else 00614 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; 00615 } 00616 } 00617 } 00618 } 00619 } 00620 00621 return; 00622 } 00623 00624 00639 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, 00640 sb_int8_array coding_method, int nb_channels, 00641 int c, int superblocktype_2_3, int cm_table_select) 00642 { 00643 int ch, sb, j; 00644 int tmp, acc, esp_40, comp; 00645 int add1, add2, add3, add4; 00646 int64_t multres; 00647 00648 // This should never happen 00649 if (nb_channels <= 0) 00650 return; 00651 00652 if (!superblocktype_2_3) { 00653 /* This case is untested, no samples available */ 00654 SAMPLES_NEEDED 00655 for (ch = 0; ch < nb_channels; ch++) 00656 for (sb = 0; sb < 30; sb++) { 00657 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer 00658 add1 = tone_level_idx[ch][sb][j] - 10; 00659 if (add1 < 0) 00660 add1 = 0; 00661 add2 = add3 = add4 = 0; 00662 if (sb > 1) { 00663 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; 00664 if (add2 < 0) 00665 add2 = 0; 00666 } 00667 if (sb > 0) { 00668 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; 00669 if (add3 < 0) 00670 add3 = 0; 00671 } 00672 if (sb < 29) { 00673 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; 00674 if (add4 < 0) 00675 add4 = 0; 00676 } 00677 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; 00678 if (tmp < 0) 00679 tmp = 0; 00680 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; 00681 } 00682 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; 00683 } 00684 acc = 0; 00685 for (ch = 0; ch < nb_channels; ch++) 00686 for (sb = 0; sb < 30; sb++) 00687 for (j = 0; j < 64; j++) 00688 acc += tone_level_idx_temp[ch][sb][j]; 00689 00690 multres = 0x66666667 * (acc * 10); 00691 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); 00692 for (ch = 0; ch < nb_channels; ch++) 00693 for (sb = 0; sb < 30; sb++) 00694 for (j = 0; j < 64; j++) { 00695 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; 00696 if (comp < 0) 00697 comp += 0xff; 00698 comp /= 256; // signed shift 00699 switch(sb) { 00700 case 0: 00701 if (comp < 30) 00702 comp = 30; 00703 comp += 15; 00704 break; 00705 case 1: 00706 if (comp < 24) 00707 comp = 24; 00708 comp += 10; 00709 break; 00710 case 2: 00711 case 3: 00712 case 4: 00713 if (comp < 16) 00714 comp = 16; 00715 } 00716 if (comp <= 5) 00717 tmp = 0; 00718 else if (comp <= 10) 00719 tmp = 10; 00720 else if (comp <= 16) 00721 tmp = 16; 00722 else if (comp <= 24) 00723 tmp = -1; 00724 else 00725 tmp = 0; 00726 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; 00727 } 00728 for (sb = 0; sb < 30; sb++) 00729 fix_coding_method_array(sb, nb_channels, coding_method); 00730 for (ch = 0; ch < nb_channels; ch++) 00731 for (sb = 0; sb < 30; sb++) 00732 for (j = 0; j < 64; j++) 00733 if (sb >= 10) { 00734 if (coding_method[ch][sb][j] < 10) 00735 coding_method[ch][sb][j] = 10; 00736 } else { 00737 if (sb >= 2) { 00738 if (coding_method[ch][sb][j] < 16) 00739 coding_method[ch][sb][j] = 16; 00740 } else { 00741 if (coding_method[ch][sb][j] < 30) 00742 coding_method[ch][sb][j] = 30; 00743 } 00744 } 00745 } else { // superblocktype_2_3 != 0 00746 for (ch = 0; ch < nb_channels; ch++) 00747 for (sb = 0; sb < 30; sb++) 00748 for (j = 0; j < 64; j++) 00749 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; 00750 } 00751 00752 return; 00753 } 00754 00755 00767 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) 00768 { 00769 int sb, j, k, n, ch, run, channels; 00770 int joined_stereo, zero_encoding, chs; 00771 int type34_first; 00772 float type34_div = 0; 00773 float type34_predictor; 00774 float samples[10], sign_bits[16]; 00775 00776 if (length == 0) { 00777 // If no data use noise 00778 for (sb=sb_min; sb < sb_max; sb++) 00779 build_sb_samples_from_noise (q, sb); 00780 00781 return; 00782 } 00783 00784 for (sb = sb_min; sb < sb_max; sb++) { 00785 FIX_NOISE_IDX(q->noise_idx); 00786 00787 channels = q->nb_channels; 00788 00789 if (q->nb_channels <= 1 || sb < 12) 00790 joined_stereo = 0; 00791 else if (sb >= 24) 00792 joined_stereo = 1; 00793 else 00794 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0; 00795 00796 if (joined_stereo) { 00797 if (BITS_LEFT(length,gb) >= 16) 00798 for (j = 0; j < 16; j++) 00799 sign_bits[j] = get_bits1 (gb); 00800 00801 for (j = 0; j < 64; j++) 00802 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) 00803 q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; 00804 00805 fix_coding_method_array(sb, q->nb_channels, q->coding_method); 00806 channels = 1; 00807 } 00808 00809 for (ch = 0; ch < channels; ch++) { 00810 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; 00811 type34_predictor = 0.0; 00812 type34_first = 1; 00813 00814 for (j = 0; j < 128; ) { 00815 switch (q->coding_method[ch][sb][j / 2]) { 00816 case 8: 00817 if (BITS_LEFT(length,gb) >= 10) { 00818 if (zero_encoding) { 00819 for (k = 0; k < 5; k++) { 00820 if ((j + 2 * k) >= 128) 00821 break; 00822 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; 00823 } 00824 } else { 00825 n = get_bits(gb, 8); 00826 for (k = 0; k < 5; k++) 00827 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 00828 } 00829 for (k = 0; k < 5; k++) 00830 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); 00831 } else { 00832 for (k = 0; k < 10; k++) 00833 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 00834 } 00835 run = 10; 00836 break; 00837 00838 case 10: 00839 if (BITS_LEFT(length,gb) >= 1) { 00840 float f = 0.81; 00841 00842 if (get_bits1(gb)) 00843 f = -f; 00844 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; 00845 samples[0] = f; 00846 } else { 00847 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00848 } 00849 run = 1; 00850 break; 00851 00852 case 16: 00853 if (BITS_LEFT(length,gb) >= 10) { 00854 if (zero_encoding) { 00855 for (k = 0; k < 5; k++) { 00856 if ((j + k) >= 128) 00857 break; 00858 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; 00859 } 00860 } else { 00861 n = get_bits (gb, 8); 00862 for (k = 0; k < 5; k++) 00863 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; 00864 } 00865 } else { 00866 for (k = 0; k < 5; k++) 00867 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 00868 } 00869 run = 5; 00870 break; 00871 00872 case 24: 00873 if (BITS_LEFT(length,gb) >= 7) { 00874 n = get_bits(gb, 7); 00875 for (k = 0; k < 3; k++) 00876 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; 00877 } else { 00878 for (k = 0; k < 3; k++) 00879 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); 00880 } 00881 run = 3; 00882 break; 00883 00884 case 30: 00885 if (BITS_LEFT(length,gb) >= 4) 00886 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)]; 00887 else 00888 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00889 00890 run = 1; 00891 break; 00892 00893 case 34: 00894 if (BITS_LEFT(length,gb) >= 7) { 00895 if (type34_first) { 00896 type34_div = (float)(1 << get_bits(gb, 2)); 00897 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; 00898 type34_predictor = samples[0]; 00899 type34_first = 0; 00900 } else { 00901 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor; 00902 type34_predictor = samples[0]; 00903 } 00904 } else { 00905 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00906 } 00907 run = 1; 00908 break; 00909 00910 default: 00911 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); 00912 run = 1; 00913 break; 00914 } 00915 00916 if (joined_stereo) { 00917 float tmp[10][MPA_MAX_CHANNELS]; 00918 00919 for (k = 0; k < run; k++) { 00920 tmp[k][0] = samples[k]; 00921 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; 00922 } 00923 for (chs = 0; chs < q->nb_channels; chs++) 00924 for (k = 0; k < run; k++) 00925 if ((j + k) < 128) 00926 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5); 00927 } else { 00928 for (k = 0; k < run; k++) 00929 if ((j + k) < 128) 00930 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5); 00931 } 00932 00933 j += run; 00934 } // j loop 00935 } // channel loop 00936 } // subband loop 00937 } 00938 00939 00950 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) 00951 { 00952 int i, k, run, level, diff; 00953 00954 if (BITS_LEFT(length,gb) < 16) 00955 return; 00956 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); 00957 00958 quantized_coeffs[0] = level; 00959 00960 for (i = 0; i < 7; ) { 00961 if (BITS_LEFT(length,gb) < 16) 00962 break; 00963 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; 00964 00965 if (BITS_LEFT(length,gb) < 16) 00966 break; 00967 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); 00968 00969 for (k = 1; k <= run; k++) 00970 quantized_coeffs[i + k] = (level + ((k * diff) / run)); 00971 00972 level += diff; 00973 i += run; 00974 } 00975 } 00976 00977 00987 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) 00988 { 00989 int sb, j, k, n, ch; 00990 00991 for (ch = 0; ch < q->nb_channels; ch++) { 00992 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length); 00993 00994 if (BITS_LEFT(length,gb) < 16) { 00995 memset(q->quantized_coeffs[ch][0], 0, 8); 00996 break; 00997 } 00998 } 00999 01000 n = q->sub_sampling + 1; 01001 01002 for (sb = 0; sb < n; sb++) 01003 for (ch = 0; ch < q->nb_channels; ch++) 01004 for (j = 0; j < 8; j++) { 01005 if (BITS_LEFT(length,gb) < 1) 01006 break; 01007 if (get_bits1(gb)) { 01008 for (k=0; k < 8; k++) { 01009 if (BITS_LEFT(length,gb) < 16) 01010 break; 01011 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); 01012 } 01013 } else { 01014 for (k=0; k < 8; k++) 01015 q->tone_level_idx_hi1[ch][sb][j][k] = 0; 01016 } 01017 } 01018 01019 n = QDM2_SB_USED(q->sub_sampling) - 4; 01020 01021 for (sb = 0; sb < n; sb++) 01022 for (ch = 0; ch < q->nb_channels; ch++) { 01023 if (BITS_LEFT(length,gb) < 16) 01024 break; 01025 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); 01026 if (sb > 19) 01027 q->tone_level_idx_hi2[ch][sb] -= 16; 01028 else 01029 for (j = 0; j < 8; j++) 01030 q->tone_level_idx_mid[ch][sb][j] = -16; 01031 } 01032 01033 n = QDM2_SB_USED(q->sub_sampling) - 5; 01034 01035 for (sb = 0; sb < n; sb++) 01036 for (ch = 0; ch < q->nb_channels; ch++) 01037 for (j = 0; j < 8; j++) { 01038 if (BITS_LEFT(length,gb) < 16) 01039 break; 01040 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; 01041 } 01042 } 01043 01050 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) 01051 { 01052 GetBitContext gb; 01053 int i, j, k, n, ch, run, level, diff; 01054 01055 init_get_bits(&gb, node->packet->data, node->packet->size*8); 01056 01057 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function 01058 01059 for (i = 1; i < n; i++) 01060 for (ch=0; ch < q->nb_channels; ch++) { 01061 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); 01062 q->quantized_coeffs[ch][i][0] = level; 01063 01064 for (j = 0; j < (8 - 1); ) { 01065 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; 01066 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); 01067 01068 for (k = 1; k <= run; k++) 01069 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); 01070 01071 level += diff; 01072 j += run; 01073 } 01074 } 01075 01076 for (ch = 0; ch < q->nb_channels; ch++) 01077 for (i = 0; i < 8; i++) 01078 q->quantized_coeffs[ch][0][i] = 0; 01079 } 01080 01081 01089 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) 01090 { 01091 GetBitContext gb; 01092 01093 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 01094 01095 if (length != 0) { 01096 init_tone_level_dequantization(q, &gb, length); 01097 fill_tone_level_array(q, 1); 01098 } else { 01099 fill_tone_level_array(q, 0); 01100 } 01101 } 01102 01103 01111 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length) 01112 { 01113 GetBitContext gb; 01114 01115 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 01116 if (length >= 32) { 01117 int c = get_bits (&gb, 13); 01118 01119 if (c > 3) 01120 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, 01121 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select); 01122 } 01123 01124 synthfilt_build_sb_samples(q, &gb, length, 0, 8); 01125 } 01126 01127 01135 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) 01136 { 01137 GetBitContext gb; 01138 01139 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); 01140 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); 01141 } 01142 01143 /* 01144 * Process new subpackets for synthesis filter 01145 * 01146 * @param q context 01147 * @param list list with synthesis filter packets (list D) 01148 */ 01149 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) 01150 { 01151 QDM2SubPNode *nodes[4]; 01152 01153 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); 01154 if (nodes[0] != NULL) 01155 process_subpacket_9(q, nodes[0]); 01156 01157 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); 01158 if (nodes[1] != NULL) 01159 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3); 01160 else 01161 process_subpacket_10(q, NULL, 0); 01162 01163 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); 01164 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) 01165 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3)); 01166 else 01167 process_subpacket_11(q, NULL, 0); 01168 01169 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); 01170 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) 01171 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3)); 01172 else 01173 process_subpacket_12(q, NULL, 0); 01174 } 01175 01176 01177 /* 01178 * Decode superblock, fill packet lists. 01179 * 01180 * @param q context 01181 */ 01182 static void qdm2_decode_super_block (QDM2Context *q) 01183 { 01184 GetBitContext gb; 01185 QDM2SubPacket header, *packet; 01186 int i, packet_bytes, sub_packet_size, sub_packets_D; 01187 unsigned int next_index = 0; 01188 01189 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); 01190 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); 01191 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); 01192 01193 q->sub_packets_B = 0; 01194 sub_packets_D = 0; 01195 01196 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] 01197 01198 init_get_bits(&gb, q->compressed_data, q->compressed_size*8); 01199 qdm2_decode_sub_packet_header(&gb, &header); 01200 01201 if (header.type < 2 || header.type >= 8) { 01202 q->has_errors = 1; 01203 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); 01204 return; 01205 } 01206 01207 q->superblocktype_2_3 = (header.type == 2 || header.type == 3); 01208 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); 01209 01210 init_get_bits(&gb, header.data, header.size*8); 01211 01212 if (header.type == 2 || header.type == 4 || header.type == 5) { 01213 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8); 01214 01215 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); 01216 01217 if (csum != 0) { 01218 q->has_errors = 1; 01219 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); 01220 return; 01221 } 01222 } 01223 01224 q->sub_packet_list_B[0].packet = NULL; 01225 q->sub_packet_list_D[0].packet = NULL; 01226 01227 for (i = 0; i < 6; i++) 01228 if (--q->fft_level_exp[i] < 0) 01229 q->fft_level_exp[i] = 0; 01230 01231 for (i = 0; packet_bytes > 0; i++) { 01232 int j; 01233 01234 q->sub_packet_list_A[i].next = NULL; 01235 01236 if (i > 0) { 01237 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; 01238 01239 /* seek to next block */ 01240 init_get_bits(&gb, header.data, header.size*8); 01241 skip_bits(&gb, next_index*8); 01242 01243 if (next_index >= header.size) 01244 break; 01245 } 01246 01247 /* decode subpacket */ 01248 packet = &q->sub_packets[i]; 01249 qdm2_decode_sub_packet_header(&gb, packet); 01250 next_index = packet->size + get_bits_count(&gb) / 8; 01251 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; 01252 01253 if (packet->type == 0) 01254 break; 01255 01256 if (sub_packet_size > packet_bytes) { 01257 if (packet->type != 10 && packet->type != 11 && packet->type != 12) 01258 break; 01259 packet->size += packet_bytes - sub_packet_size; 01260 } 01261 01262 packet_bytes -= sub_packet_size; 01263 01264 /* add subpacket to 'all subpackets' list */ 01265 q->sub_packet_list_A[i].packet = packet; 01266 01267 /* add subpacket to related list */ 01268 if (packet->type == 8) { 01269 SAMPLES_NEEDED_2("packet type 8"); 01270 return; 01271 } else if (packet->type >= 9 && packet->type <= 12) { 01272 /* packets for MPEG Audio like Synthesis Filter */ 01273 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); 01274 } else if (packet->type == 13) { 01275 for (j = 0; j < 6; j++) 01276 q->fft_level_exp[j] = get_bits(&gb, 6); 01277 } else if (packet->type == 14) { 01278 for (j = 0; j < 6; j++) 01279 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); 01280 } else if (packet->type == 15) { 01281 SAMPLES_NEEDED_2("packet type 15") 01282 return; 01283 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { 01284 /* packets for FFT */ 01285 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); 01286 } 01287 } // Packet bytes loop 01288 01289 /* **************************************************************** */ 01290 if (q->sub_packet_list_D[0].packet != NULL) { 01291 process_synthesis_subpackets(q, q->sub_packet_list_D); 01292 q->do_synth_filter = 1; 01293 } else if (q->do_synth_filter) { 01294 process_subpacket_10(q, NULL, 0); 01295 process_subpacket_11(q, NULL, 0); 01296 process_subpacket_12(q, NULL, 0); 01297 } 01298 /* **************************************************************** */ 01299 } 01300 01301 01302 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, 01303 int offset, int duration, int channel, 01304 int exp, int phase) 01305 { 01306 if (q->fft_coefs_min_index[duration] < 0) 01307 q->fft_coefs_min_index[duration] = q->fft_coefs_index; 01308 01309 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); 01310 q->fft_coefs[q->fft_coefs_index].channel = channel; 01311 q->fft_coefs[q->fft_coefs_index].offset = offset; 01312 q->fft_coefs[q->fft_coefs_index].exp = exp; 01313 q->fft_coefs[q->fft_coefs_index].phase = phase; 01314 q->fft_coefs_index++; 01315 } 01316 01317 01318 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) 01319 { 01320 int channel, stereo, phase, exp; 01321 int local_int_4, local_int_8, stereo_phase, local_int_10; 01322 int local_int_14, stereo_exp, local_int_20, local_int_28; 01323 int n, offset; 01324 01325 local_int_4 = 0; 01326 local_int_28 = 0; 01327 local_int_20 = 2; 01328 local_int_8 = (4 - duration); 01329 local_int_10 = 1 << (q->group_order - duration - 1); 01330 offset = 1; 01331 01332 while (1) { 01333 if (q->superblocktype_2_3) { 01334 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { 01335 offset = 1; 01336 if (n == 0) { 01337 local_int_4 += local_int_10; 01338 local_int_28 += (1 << local_int_8); 01339 } else { 01340 local_int_4 += 8*local_int_10; 01341 local_int_28 += (8 << local_int_8); 01342 } 01343 } 01344 offset += (n - 2); 01345 } else { 01346 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); 01347 while (offset >= (local_int_10 - 1)) { 01348 offset += (1 - (local_int_10 - 1)); 01349 local_int_4 += local_int_10; 01350 local_int_28 += (1 << local_int_8); 01351 } 01352 } 01353 01354 if (local_int_4 >= q->group_size) 01355 return; 01356 01357 local_int_14 = (offset >> local_int_8); 01358 01359 if (q->nb_channels > 1) { 01360 channel = get_bits1(gb); 01361 stereo = get_bits1(gb); 01362 } else { 01363 channel = 0; 01364 stereo = 0; 01365 } 01366 01367 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); 01368 exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; 01369 exp = (exp < 0) ? 0 : exp; 01370 01371 phase = get_bits(gb, 3); 01372 stereo_exp = 0; 01373 stereo_phase = 0; 01374 01375 if (stereo) { 01376 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); 01377 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); 01378 if (stereo_phase < 0) 01379 stereo_phase += 8; 01380 } 01381 01382 if (q->frequency_range > (local_int_14 + 1)) { 01383 int sub_packet = (local_int_20 + local_int_28); 01384 01385 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); 01386 if (stereo) 01387 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); 01388 } 01389 01390 offset++; 01391 } 01392 } 01393 01394 01395 static void qdm2_decode_fft_packets (QDM2Context *q) 01396 { 01397 int i, j, min, max, value, type, unknown_flag; 01398 GetBitContext gb; 01399 01400 if (q->sub_packet_list_B[0].packet == NULL) 01401 return; 01402 01403 /* reset minimum indexes for FFT coefficients */ 01404 q->fft_coefs_index = 0; 01405 for (i=0; i < 5; i++) 01406 q->fft_coefs_min_index[i] = -1; 01407 01408 /* process subpackets ordered by type, largest type first */ 01409 for (i = 0, max = 256; i < q->sub_packets_B; i++) { 01410 QDM2SubPacket *packet= NULL; 01411 01412 /* find subpacket with largest type less than max */ 01413 for (j = 0, min = 0; j < q->sub_packets_B; j++) { 01414 value = q->sub_packet_list_B[j].packet->type; 01415 if (value > min && value < max) { 01416 min = value; 01417 packet = q->sub_packet_list_B[j].packet; 01418 } 01419 } 01420 01421 max = min; 01422 01423 /* check for errors (?) */ 01424 if (!packet) 01425 return; 01426 01427 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) 01428 return; 01429 01430 /* decode FFT tones */ 01431 init_get_bits (&gb, packet->data, packet->size*8); 01432 01433 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) 01434 unknown_flag = 1; 01435 else 01436 unknown_flag = 0; 01437 01438 type = packet->type; 01439 01440 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { 01441 int duration = q->sub_sampling + 5 - (type & 15); 01442 01443 if (duration >= 0 && duration < 4) 01444 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); 01445 } else if (type == 31) { 01446 for (j=0; j < 4; j++) 01447 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 01448 } else if (type == 46) { 01449 for (j=0; j < 6; j++) 01450 q->fft_level_exp[j] = get_bits(&gb, 6); 01451 for (j=0; j < 4; j++) 01452 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); 01453 } 01454 } // Loop on B packets 01455 01456 /* calculate maximum indexes for FFT coefficients */ 01457 for (i = 0, j = -1; i < 5; i++) 01458 if (q->fft_coefs_min_index[i] >= 0) { 01459 if (j >= 0) 01460 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; 01461 j = i; 01462 } 01463 if (j >= 0) 01464 q->fft_coefs_max_index[j] = q->fft_coefs_index; 01465 } 01466 01467 01468 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) 01469 { 01470 float level, f[6]; 01471 int i; 01472 QDM2Complex c; 01473 const double iscale = 2.0*M_PI / 512.0; 01474 01475 tone->phase += tone->phase_shift; 01476 01477 /* calculate current level (maximum amplitude) of tone */ 01478 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; 01479 c.im = level * sin(tone->phase*iscale); 01480 c.re = level * cos(tone->phase*iscale); 01481 01482 /* generate FFT coefficients for tone */ 01483 if (tone->duration >= 3 || tone->cutoff >= 3) { 01484 tone->complex[0].im += c.im; 01485 tone->complex[0].re += c.re; 01486 tone->complex[1].im -= c.im; 01487 tone->complex[1].re -= c.re; 01488 } else { 01489 f[1] = -tone->table[4]; 01490 f[0] = tone->table[3] - tone->table[0]; 01491 f[2] = 1.0 - tone->table[2] - tone->table[3]; 01492 f[3] = tone->table[1] + tone->table[4] - 1.0; 01493 f[4] = tone->table[0] - tone->table[1]; 01494 f[5] = tone->table[2]; 01495 for (i = 0; i < 2; i++) { 01496 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; 01497 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); 01498 } 01499 for (i = 0; i < 4; i++) { 01500 tone->complex[i].re += c.re * f[i+2]; 01501 tone->complex[i].im += c.im * f[i+2]; 01502 } 01503 } 01504 01505 /* copy the tone if it has not yet died out */ 01506 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { 01507 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); 01508 q->fft_tone_end = (q->fft_tone_end + 1) % 1000; 01509 } 01510 } 01511 01512 01513 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) 01514 { 01515 int i, j, ch; 01516 const double iscale = 0.25 * M_PI; 01517 01518 for (ch = 0; ch < q->channels; ch++) { 01519 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); 01520 } 01521 01522 01523 /* apply FFT tones with duration 4 (1 FFT period) */ 01524 if (q->fft_coefs_min_index[4] >= 0) 01525 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { 01526 float level; 01527 QDM2Complex c; 01528 01529 if (q->fft_coefs[i].sub_packet != sub_packet) 01530 break; 01531 01532 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; 01533 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]; 01534 01535 c.re = level * cos(q->fft_coefs[i].phase * iscale); 01536 c.im = level * sin(q->fft_coefs[i].phase * iscale); 01537 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; 01538 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; 01539 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; 01540 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; 01541 } 01542 01543 /* generate existing FFT tones */ 01544 for (i = q->fft_tone_end; i != q->fft_tone_start; ) { 01545 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); 01546 q->fft_tone_start = (q->fft_tone_start + 1) % 1000; 01547 } 01548 01549 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ 01550 for (i = 0; i < 4; i++) 01551 if (q->fft_coefs_min_index[i] >= 0) { 01552 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { 01553 int offset, four_i; 01554 FFTTone tone; 01555 01556 if (q->fft_coefs[j].sub_packet != sub_packet) 01557 break; 01558 01559 four_i = (4 - i); 01560 offset = q->fft_coefs[j].offset >> four_i; 01561 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; 01562 01563 if (offset < q->frequency_range) { 01564 if (offset < 2) 01565 tone.cutoff = offset; 01566 else 01567 tone.cutoff = (offset >= 60) ? 3 : 2; 01568 01569 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]; 01570 tone.complex = &q->fft.complex[ch][offset]; 01571 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; 01572 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; 01573 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); 01574 tone.duration = i; 01575 tone.time_index = 0; 01576 01577 qdm2_fft_generate_tone(q, &tone); 01578 } 01579 } 01580 q->fft_coefs_min_index[i] = j; 01581 } 01582 } 01583 01584 01585 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) 01586 { 01587 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; 01588 int i; 01589 q->fft.complex[channel][0].re *= 2.0f; 01590 q->fft.complex[channel][0].im = 0.0f; 01591 ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); 01592 /* add samples to output buffer */ 01593 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) 01594 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; 01595 } 01596 01597 01602 static void qdm2_synthesis_filter (QDM2Context *q, int index) 01603 { 01604 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; 01605 int i, k, ch, sb_used, sub_sampling, dither_state = 0; 01606 01607 /* copy sb_samples */ 01608 sb_used = QDM2_SB_USED(q->sub_sampling); 01609 01610 for (ch = 0; ch < q->channels; ch++) 01611 for (i = 0; i < 8; i++) 01612 for (k=sb_used; k < SBLIMIT; k++) 01613 q->sb_samples[ch][(8 * index) + i][k] = 0; 01614 01615 for (ch = 0; ch < q->nb_channels; ch++) { 01616 OUT_INT *samples_ptr = samples + ch; 01617 01618 for (i = 0; i < 8; i++) { 01619 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]), 01620 ff_mpa_synth_window, &dither_state, 01621 samples_ptr, q->nb_channels, 01622 q->sb_samples[ch][(8 * index) + i]); 01623 samples_ptr += 32 * q->nb_channels; 01624 } 01625 } 01626 01627 /* add samples to output buffer */ 01628 sub_sampling = (4 >> q->sub_sampling); 01629 01630 for (ch = 0; ch < q->channels; ch++) 01631 for (i = 0; i < q->frame_size; i++) 01632 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16)); 01633 } 01634 01635 01641 static av_cold void qdm2_init(QDM2Context *q) { 01642 static int initialized = 0; 01643 01644 if (initialized != 0) 01645 return; 01646 initialized = 1; 01647 01648 qdm2_init_vlc(); 01649 ff_mpa_synth_init(ff_mpa_synth_window); 01650 softclip_table_init(); 01651 rnd_table_init(); 01652 init_noise_samples(); 01653 01654 av_log(NULL, AV_LOG_DEBUG, "init done\n"); 01655 } 01656 01657 01658 #if 0 01659 static void dump_context(QDM2Context *q) 01660 { 01661 int i; 01662 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b); 01663 PRINT("compressed_data",q->compressed_data); 01664 PRINT("compressed_size",q->compressed_size); 01665 PRINT("frame_size",q->frame_size); 01666 PRINT("checksum_size",q->checksum_size); 01667 PRINT("channels",q->channels); 01668 PRINT("nb_channels",q->nb_channels); 01669 PRINT("fft_frame_size",q->fft_frame_size); 01670 PRINT("fft_size",q->fft_size); 01671 PRINT("sub_sampling",q->sub_sampling); 01672 PRINT("fft_order",q->fft_order); 01673 PRINT("group_order",q->group_order); 01674 PRINT("group_size",q->group_size); 01675 PRINT("sub_packet",q->sub_packet); 01676 PRINT("frequency_range",q->frequency_range); 01677 PRINT("has_errors",q->has_errors); 01678 PRINT("fft_tone_end",q->fft_tone_end); 01679 PRINT("fft_tone_start",q->fft_tone_start); 01680 PRINT("fft_coefs_index",q->fft_coefs_index); 01681 PRINT("coeff_per_sb_select",q->coeff_per_sb_select); 01682 PRINT("cm_table_select",q->cm_table_select); 01683 PRINT("noise_idx",q->noise_idx); 01684 01685 for (i = q->fft_tone_start; i < q->fft_tone_end; i++) 01686 { 01687 FFTTone *t = &q->fft_tones[i]; 01688 01689 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i); 01690 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level); 01691 // PRINT(" level", t->level); 01692 PRINT(" phase", t->phase); 01693 PRINT(" phase_shift", t->phase_shift); 01694 PRINT(" duration", t->duration); 01695 PRINT(" samples_im", t->samples_im); 01696 PRINT(" samples_re", t->samples_re); 01697 PRINT(" table", t->table); 01698 } 01699 01700 } 01701 #endif 01702 01703 01707 static av_cold int qdm2_decode_init(AVCodecContext *avctx) 01708 { 01709 QDM2Context *s = avctx->priv_data; 01710 uint8_t *extradata; 01711 int extradata_size; 01712 int tmp_val, tmp, size; 01713 01714 /* extradata parsing 01715 01716 Structure: 01717 wave { 01718 frma (QDM2) 01719 QDCA 01720 QDCP 01721 } 01722 01723 32 size (including this field) 01724 32 tag (=frma) 01725 32 type (=QDM2 or QDMC) 01726 01727 32 size (including this field, in bytes) 01728 32 tag (=QDCA) // maybe mandatory parameters 01729 32 unknown (=1) 01730 32 channels (=2) 01731 32 samplerate (=44100) 01732 32 bitrate (=96000) 01733 32 block size (=4096) 01734 32 frame size (=256) (for one channel) 01735 32 packet size (=1300) 01736 01737 32 size (including this field, in bytes) 01738 32 tag (=QDCP) // maybe some tuneable parameters 01739 32 float1 (=1.0) 01740 32 zero ? 01741 32 float2 (=1.0) 01742 32 float3 (=1.0) 01743 32 unknown (27) 01744 32 unknown (8) 01745 32 zero ? 01746 */ 01747 01748 if (!avctx->extradata || (avctx->extradata_size < 48)) { 01749 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); 01750 return -1; 01751 } 01752 01753 extradata = avctx->extradata; 01754 extradata_size = avctx->extradata_size; 01755 01756 while (extradata_size > 7) { 01757 if (!memcmp(extradata, "frmaQDM", 7)) 01758 break; 01759 extradata++; 01760 extradata_size--; 01761 } 01762 01763 if (extradata_size < 12) { 01764 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", 01765 extradata_size); 01766 return -1; 01767 } 01768 01769 if (memcmp(extradata, "frmaQDM", 7)) { 01770 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); 01771 return -1; 01772 } 01773 01774 if (extradata[7] == 'C') { 01775 // s->is_qdmc = 1; 01776 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); 01777 return -1; 01778 } 01779 01780 extradata += 8; 01781 extradata_size -= 8; 01782 01783 size = AV_RB32(extradata); 01784 01785 if(size > extradata_size){ 01786 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", 01787 extradata_size, size); 01788 return -1; 01789 } 01790 01791 extradata += 4; 01792 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); 01793 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { 01794 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); 01795 return -1; 01796 } 01797 01798 extradata += 8; 01799 01800 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); 01801 extradata += 4; 01802 01803 avctx->sample_rate = AV_RB32(extradata); 01804 extradata += 4; 01805 01806 avctx->bit_rate = AV_RB32(extradata); 01807 extradata += 4; 01808 01809 s->group_size = AV_RB32(extradata); 01810 extradata += 4; 01811 01812 s->fft_size = AV_RB32(extradata); 01813 extradata += 4; 01814 01815 s->checksum_size = AV_RB32(extradata); 01816 01817 s->fft_order = av_log2(s->fft_size) + 1; 01818 s->fft_frame_size = 2 * s->fft_size; // complex has two floats 01819 01820 // something like max decodable tones 01821 s->group_order = av_log2(s->group_size) + 1; 01822 s->frame_size = s->group_size / 16; // 16 iterations per super block 01823 01824 s->sub_sampling = s->fft_order - 7; 01825 s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); 01826 01827 switch ((s->sub_sampling * 2 + s->channels - 1)) { 01828 case 0: tmp = 40; break; 01829 case 1: tmp = 48; break; 01830 case 2: tmp = 56; break; 01831 case 3: tmp = 72; break; 01832 case 4: tmp = 80; break; 01833 case 5: tmp = 100;break; 01834 default: tmp=s->sub_sampling; break; 01835 } 01836 tmp_val = 0; 01837 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; 01838 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; 01839 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; 01840 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; 01841 s->cm_table_select = tmp_val; 01842 01843 if (s->sub_sampling == 0) 01844 tmp = 7999; 01845 else 01846 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; 01847 /* 01848 0: 7999 -> 0 01849 1: 20000 -> 2 01850 2: 28000 -> 2 01851 */ 01852 if (tmp < 8000) 01853 s->coeff_per_sb_select = 0; 01854 else if (tmp <= 16000) 01855 s->coeff_per_sb_select = 1; 01856 else 01857 s->coeff_per_sb_select = 2; 01858 01859 // Fail on unknown fft order 01860 if ((s->fft_order < 7) || (s->fft_order > 9)) { 01861 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); 01862 return -1; 01863 } 01864 01865 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); 01866 01867 qdm2_init(s); 01868 01869 avctx->sample_fmt = SAMPLE_FMT_S16; 01870 01871 // dump_context(s); 01872 return 0; 01873 } 01874 01875 01876 static av_cold int qdm2_decode_close(AVCodecContext *avctx) 01877 { 01878 QDM2Context *s = avctx->priv_data; 01879 01880 ff_rdft_end(&s->rdft_ctx); 01881 01882 return 0; 01883 } 01884 01885 01886 static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) 01887 { 01888 int ch, i; 01889 const int frame_size = (q->frame_size * q->channels); 01890 01891 /* select input buffer */ 01892 q->compressed_data = in; 01893 q->compressed_size = q->checksum_size; 01894 01895 // dump_context(q); 01896 01897 /* copy old block, clear new block of output samples */ 01898 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); 01899 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); 01900 01901 /* decode block of QDM2 compressed data */ 01902 if (q->sub_packet == 0) { 01903 q->has_errors = 0; // zero it for a new super block 01904 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); 01905 qdm2_decode_super_block(q); 01906 } 01907 01908 /* parse subpackets */ 01909 if (!q->has_errors) { 01910 if (q->sub_packet == 2) 01911 qdm2_decode_fft_packets(q); 01912 01913 qdm2_fft_tone_synthesizer(q, q->sub_packet); 01914 } 01915 01916 /* sound synthesis stage 1 (FFT) */ 01917 for (ch = 0; ch < q->channels; ch++) { 01918 qdm2_calculate_fft(q, ch, q->sub_packet); 01919 01920 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { 01921 SAMPLES_NEEDED_2("has errors, and C list is not empty") 01922 return; 01923 } 01924 } 01925 01926 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ 01927 if (!q->has_errors && q->do_synth_filter) 01928 qdm2_synthesis_filter(q, q->sub_packet); 01929 01930 q->sub_packet = (q->sub_packet + 1) % 16; 01931 01932 /* clip and convert output float[] to 16bit signed samples */ 01933 for (i = 0; i < frame_size; i++) { 01934 int value = (int)q->output_buffer[i]; 01935 01936 if (value > SOFTCLIP_THRESHOLD) 01937 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; 01938 else if (value < -SOFTCLIP_THRESHOLD) 01939 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; 01940 01941 out[i] = value; 01942 } 01943 } 01944 01945 01946 static int qdm2_decode_frame(AVCodecContext *avctx, 01947 void *data, int *data_size, 01948 AVPacket *avpkt) 01949 { 01950 const uint8_t *buf = avpkt->data; 01951 int buf_size = avpkt->size; 01952 QDM2Context *s = avctx->priv_data; 01953 01954 if(!buf) 01955 return 0; 01956 if(buf_size < s->checksum_size) 01957 return -1; 01958 01959 *data_size = s->channels * s->frame_size * sizeof(int16_t); 01960 01961 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", 01962 buf_size, buf, s->checksum_size, data, *data_size); 01963 01964 qdm2_decode(s, buf, data); 01965 01966 // reading only when next superblock found 01967 if (s->sub_packet == 0) { 01968 return s->checksum_size; 01969 } 01970 01971 return 0; 01972 } 01973 01974 AVCodec qdm2_decoder = 01975 { 01976 .name = "qdm2", 01977 .type = AVMEDIA_TYPE_AUDIO, 01978 .id = CODEC_ID_QDM2, 01979 .priv_data_size = sizeof(QDM2Context), 01980 .init = qdm2_decode_init, 01981 .close = qdm2_decode_close, 01982 .decode = qdm2_decode_frame, 01983 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), 01984 };