2 * QDM2 compatible decoder
3 * Copyright (c) 2003 Ewald Snel
4 * Copyright (c) 2005 Benjamin Larsson
5 * Copyright (c) 2005 Alex Beregszaszi
6 * Copyright (c) 2005 Roberto Togni
8 * This file is part of FFmpeg.
10 * FFmpeg is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Lesser General Public
12 * License as published by the Free Software Foundation; either
13 * version 2.1 of the License, or (at your option) any later version.
15 * FFmpeg is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Lesser General Public License for more details.
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with FFmpeg; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
29 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
30 * The decoder is not perfect yet, there are still some distortions
31 * especially on files encoded with 16 or 8 subbands.
38 #define ALT_BITSTREAM_READER_LE
40 #include "bitstream.h"
43 #ifdef CONFIG_MPEGAUDIO_HP
44 #define USE_HIGHPRECISION
47 #include "mpegaudio.h"
55 #define SOFTCLIP_THRESHOLD 27600
56 #define HARDCLIP_THRESHOLD 35716
59 #define QDM2_LIST_ADD(list, size, packet) \
62 list[size - 1].next = &list[size]; \
64 list[size].packet = packet; \
65 list[size].next = NULL; \
69 // Result is 8, 16 or 30
70 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
72 #define FIX_NOISE_IDX(noise_idx) \
73 if ((noise_idx) >= 3840) \
74 (noise_idx) -= 3840; \
76 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
78 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
80 #define SAMPLES_NEEDED \
81 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
83 #define SAMPLES_NEEDED_2(why) \
84 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
87 typedef int8_t sb_int8_array[2][30][64];
93 int type; ///< subpacket type
94 unsigned int size; ///< subpacket size
95 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
99 * A node in the subpacket list
101 typedef struct _QDM2SubPNode {
102 QDM2SubPacket *packet; ///< packet
103 struct _QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
132 QDM2Complex complex[256 + 1] __attribute__((aligned(16)));
133 float samples_im[MPA_MAX_CHANNELS][256];
134 float samples_re[MPA_MAX_CHANNELS][256];
138 * QDM2 decoder context
141 /// Parameters from codec header, do not change during playback
142 int nb_channels; ///< number of channels
143 int channels; ///< number of channels
144 int group_size; ///< size of frame group (16 frames per group)
145 int fft_size; ///< size of FFT, in complex numbers
146 int checksum_size; ///< size of data block, used also for checksum
148 /// Parameters built from header parameters, do not change during playback
149 int group_order; ///< order of frame group
150 int fft_order; ///< order of FFT (actually fftorder+1)
151 int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
152 int frame_size; ///< size of data frame
154 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
155 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
156 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
158 /// Packets and packet lists
159 QDM2SubPacket sub_packets[16]; ///< the packets themselves
160 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
161 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
162 int sub_packets_B; ///< number of packets on 'B' list
163 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
164 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
167 FFTTone fft_tones[1000];
170 FFTCoefficient fft_coefs[1000];
172 int fft_coefs_min_index[5];
173 int fft_coefs_max_index[5];
174 int fft_level_exp[6];
176 FFTComplex exptab[128];
180 uint8_t *compressed_data;
182 float output_buffer[1024];
185 MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16)));
186 int synth_buf_offset[MPA_MAX_CHANNELS];
187 int32_t sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT] __attribute__((aligned(16)));
189 /// Mixed temporary data used in decoding
190 float tone_level[MPA_MAX_CHANNELS][30][64];
191 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
192 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
193 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
194 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
195 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
196 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
197 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
198 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
201 int has_errors; ///< packet has errors
202 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
203 int do_synth_filter; ///< used to perform or skip synthesis filter
206 int noise_idx; ///< index for dithering noise table
210 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
212 static VLC vlc_tab_level;
213 static VLC vlc_tab_diff;
214 static VLC vlc_tab_run;
215 static VLC fft_level_exp_alt_vlc;
216 static VLC fft_level_exp_vlc;
217 static VLC fft_stereo_exp_vlc;
218 static VLC fft_stereo_phase_vlc;
219 static VLC vlc_tab_tone_level_idx_hi1;
220 static VLC vlc_tab_tone_level_idx_mid;
221 static VLC vlc_tab_tone_level_idx_hi2;
222 static VLC vlc_tab_type30;
223 static VLC vlc_tab_type34;
224 static VLC vlc_tab_fft_tone_offset[5];
226 static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
227 static float noise_table[4096];
228 static uint8_t random_dequant_index[256][5];
229 static uint8_t random_dequant_type24[128][3];
230 static float noise_samples[128];
232 static MPA_INT mpa_window[512] __attribute__((aligned(16)));
235 static void softclip_table_init(void) {
237 double dfl = SOFTCLIP_THRESHOLD - 32767;
238 float delta = 1.0 / -dfl;
239 for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
240 softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
244 // random generated table
245 static void rnd_table_init(void) {
249 uint64_t random_seed = 0;
250 float delta = 1.0 / 16384.0;
251 for(i = 0; i < 4096 ;i++) {
252 random_seed = random_seed * 214013 + 2531011;
253 noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
256 for (i = 0; i < 256 ;i++) {
259 for (j = 0; j < 5 ;j++) {
260 random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
261 ldw = (uint32_t)ldw % (uint32_t)random_seed;
262 tmp64_1 = (random_seed * 0x55555556);
263 hdw = (uint32_t)(tmp64_1 >> 32);
264 random_seed = (uint64_t)(hdw + (ldw >> 31));
267 for (i = 0; i < 128 ;i++) {
270 for (j = 0; j < 3 ;j++) {
271 random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
272 ldw = (uint32_t)ldw % (uint32_t)random_seed;
273 tmp64_1 = (random_seed * 0x66666667);
274 hdw = (uint32_t)(tmp64_1 >> 33);
275 random_seed = hdw + (ldw >> 31);
281 static void init_noise_samples(void) {
284 float delta = 1.0 / 16384.0;
285 for (i = 0; i < 128;i++) {
286 random_seed = random_seed * 214013 + 2531011;
287 noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
292 static void qdm2_init_vlc(void)
294 init_vlc (&vlc_tab_level, 8, 24,
295 vlc_tab_level_huffbits, 1, 1,
296 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
298 init_vlc (&vlc_tab_diff, 8, 37,
299 vlc_tab_diff_huffbits, 1, 1,
300 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
302 init_vlc (&vlc_tab_run, 5, 6,
303 vlc_tab_run_huffbits, 1, 1,
304 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
306 init_vlc (&fft_level_exp_alt_vlc, 8, 28,
307 fft_level_exp_alt_huffbits, 1, 1,
308 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
310 init_vlc (&fft_level_exp_vlc, 8, 20,
311 fft_level_exp_huffbits, 1, 1,
312 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
314 init_vlc (&fft_stereo_exp_vlc, 6, 7,
315 fft_stereo_exp_huffbits, 1, 1,
316 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
318 init_vlc (&fft_stereo_phase_vlc, 6, 9,
319 fft_stereo_phase_huffbits, 1, 1,
320 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
322 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
323 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
324 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
326 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
327 vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
328 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
330 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
331 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
332 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
334 init_vlc (&vlc_tab_type30, 6, 9,
335 vlc_tab_type30_huffbits, 1, 1,
336 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
338 init_vlc (&vlc_tab_type34, 5, 10,
339 vlc_tab_type34_huffbits, 1, 1,
340 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
342 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
343 vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
344 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
346 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
347 vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
348 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
350 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
351 vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
352 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
354 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
355 vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
356 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
358 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
359 vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
360 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
364 /* for floating point to fixed point conversion */
365 static float f2i_scale = (float) (1 << (FRAC_BITS - 15));
368 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
372 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
374 /* stage-2, 3 bits exponent escape sequence */
376 value = get_bits (gb, get_bits (gb, 3) + 1);
378 /* stage-3, optional */
380 int tmp = vlc_stage3_values[value];
382 if ((value & ~3) > 0)
383 tmp += get_bits (gb, (value >> 2));
391 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
393 int value = qdm2_get_vlc (gb, vlc, 0, depth);
395 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
402 * @param data pointer to data to be checksum'ed
403 * @param length data length
404 * @param value checksum value
406 * @return 0 if checksum is OK
408 static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) {
411 for (i=0; i < length; i++)
414 return (uint16_t)(value & 0xffff);
419 * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
421 * @param gb bitreader context
422 * @param sub_packet packet under analysis
424 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
426 sub_packet->type = get_bits (gb, 8);
428 if (sub_packet->type == 0) {
429 sub_packet->size = 0;
430 sub_packet->data = NULL;
432 sub_packet->size = get_bits (gb, 8);
434 if (sub_packet->type & 0x80) {
435 sub_packet->size <<= 8;
436 sub_packet->size |= get_bits (gb, 8);
437 sub_packet->type &= 0x7f;
440 if (sub_packet->type == 0x7f)
441 sub_packet->type |= (get_bits (gb, 8) << 8);
443 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
446 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
447 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
452 * Return node pointer to first packet of requested type in list.
454 * @param list list of subpackets to be scanned
455 * @param type type of searched subpacket
456 * @return node pointer for subpacket if found, else NULL
458 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
460 while (list != NULL && list->packet != NULL) {
461 if (list->packet->type == type)
470 * Replaces 8 elements with their average value.
471 * Called by qdm2_decode_superblock before starting subblock decoding.
475 static void average_quantized_coeffs (QDM2Context *q)
477 int i, j, n, ch, sum;
479 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
481 for (ch = 0; ch < q->nb_channels; ch++)
482 for (i = 0; i < n; i++) {
485 for (j = 0; j < 8; j++)
486 sum += q->quantized_coeffs[ch][i][j];
492 for (j=0; j < 8; j++)
493 q->quantized_coeffs[ch][i][j] = sum;
499 * Build subband samples with noise weighted by q->tone_level.
500 * Called by synthfilt_build_sb_samples.
503 * @param sb subband index
505 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
509 FIX_NOISE_IDX(q->noise_idx);
514 for (ch = 0; ch < q->nb_channels; ch++)
515 for (j = 0; j < 64; j++) {
516 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);
517 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);
523 * Called while processing data from subpackets 11 and 12.
524 * Used after making changes to coding_method array.
526 * @param sb subband index
527 * @param channels number of channels
528 * @param coding_method q->coding_method[0][0][0]
530 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
535 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};
537 for (ch = 0; ch < channels; ch++) {
538 for (j = 0; j < 64; ) {
539 if((coding_method[ch][sb][j] - 8) > 22) {
543 switch (switchtable[coding_method[ch][sb][j]-8]) {
544 case 0: run = 10; case_val = 10; break;
545 case 1: run = 1; case_val = 16; break;
546 case 2: run = 5; case_val = 24; break;
547 case 3: run = 3; case_val = 30; break;
548 case 4: run = 1; case_val = 30; break;
549 case 5: run = 1; case_val = 8; break;
550 default: run = 1; case_val = 8; break;
553 for (k = 0; k < run; k++)
555 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
558 //not debugged, almost never used
559 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
560 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
569 * Related to synthesis filter
570 * Called by process_subpacket_10
573 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
575 static void fill_tone_level_array (QDM2Context *q, int flag)
577 int i, sb, ch, sb_used;
580 // This should never happen
581 if (q->nb_channels <= 0)
584 for (ch = 0; ch < q->nb_channels; ch++)
585 for (sb = 0; sb < 30; sb++)
586 for (i = 0; i < 8; i++) {
587 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
588 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
589 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
591 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
594 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
597 sb_used = QDM2_SB_USED(q->sub_sampling);
599 if ((q->superblocktype_2_3 != 0) && !flag) {
600 for (sb = 0; sb < sb_used; sb++)
601 for (ch = 0; ch < q->nb_channels; ch++)
602 for (i = 0; i < 64; i++) {
603 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
604 if (q->tone_level_idx[ch][sb][i] < 0)
605 q->tone_level[ch][sb][i] = 0;
607 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
610 tab = q->superblocktype_2_3 ? 0 : 1;
611 for (sb = 0; sb < sb_used; sb++) {
612 if ((sb >= 4) && (sb <= 23)) {
613 for (ch = 0; ch < q->nb_channels; ch++)
614 for (i = 0; i < 64; i++) {
615 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
616 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
617 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
618 q->tone_level_idx_hi2[ch][sb - 4];
619 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
620 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
621 q->tone_level[ch][sb][i] = 0;
623 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
627 for (ch = 0; ch < q->nb_channels; ch++)
628 for (i = 0; i < 64; i++) {
629 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
630 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
631 q->tone_level_idx_hi2[ch][sb - 4];
632 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
633 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
634 q->tone_level[ch][sb][i] = 0;
636 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
639 for (ch = 0; ch < q->nb_channels; ch++)
640 for (i = 0; i < 64; i++) {
641 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
642 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
643 q->tone_level[ch][sb][i] = 0;
645 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
657 * Related to synthesis filter
658 * Called by process_subpacket_11
659 * c is built with data from subpacket 11
660 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
662 * @param tone_level_idx
663 * @param tone_level_idx_temp
664 * @param coding_method q->coding_method[0][0][0]
665 * @param nb_channels number of channels
666 * @param c coming from subpacket 11, passed as 8*c
667 * @param superblocktype_2_3 flag based on superblock packet type
668 * @param cm_table_select q->cm_table_select
670 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
671 sb_int8_array coding_method, int nb_channels,
672 int c, int superblocktype_2_3, int cm_table_select)
675 int tmp, acc, esp_40, comp;
676 int add1, add2, add3, add4;
679 // This should never happen
680 if (nb_channels <= 0)
683 if (!superblocktype_2_3) {
684 /* This case is untested, no samples available */
686 for (ch = 0; ch < nb_channels; ch++)
687 for (sb = 0; sb < 30; sb++) {
688 for (j = 1; j < 64; j++) {
689 add1 = tone_level_idx[ch][sb][j] - 10;
692 add2 = add3 = add4 = 0;
694 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
699 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
704 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
708 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
711 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
713 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
716 for (ch = 0; ch < nb_channels; ch++)
717 for (sb = 0; sb < 30; sb++)
718 for (j = 0; j < 64; j++)
719 acc += tone_level_idx_temp[ch][sb][j];
721 tmp = c * 256 / (acc & 0xffff);
722 multres = 0x66666667 * (acc * 10);
723 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
724 for (ch = 0; ch < nb_channels; ch++)
725 for (sb = 0; sb < 30; sb++)
726 for (j = 0; j < 64; j++) {
727 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
730 comp /= 256; // signed shift
758 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
760 for (sb = 0; sb < 30; sb++)
761 fix_coding_method_array(sb, nb_channels, coding_method);
762 for (ch = 0; ch < nb_channels; ch++)
763 for (sb = 0; sb < 30; sb++)
764 for (j = 0; j < 64; j++)
766 if (coding_method[ch][sb][j] < 10)
767 coding_method[ch][sb][j] = 10;
770 if (coding_method[ch][sb][j] < 16)
771 coding_method[ch][sb][j] = 16;
773 if (coding_method[ch][sb][j] < 30)
774 coding_method[ch][sb][j] = 30;
777 } else { // superblocktype_2_3 != 0
778 for (ch = 0; ch < nb_channels; ch++)
779 for (sb = 0; sb < 30; sb++)
780 for (j = 0; j < 64; j++)
781 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
790 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
791 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
794 * @param gb bitreader context
795 * @param length packet length in bits
796 * @param sb_min lower subband processed (sb_min included)
797 * @param sb_max higher subband processed (sb_max excluded)
799 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
801 int sb, j, k, n, ch, run, channels;
802 int joined_stereo, zero_encoding, chs;
804 float type34_div = 0;
805 float type34_predictor;
806 float samples[10], sign_bits[16];
809 // If no data use noise
810 for (sb=sb_min; sb < sb_max; sb++)
811 build_sb_samples_from_noise (q, sb);
816 for (sb = sb_min; sb < sb_max; sb++) {
817 FIX_NOISE_IDX(q->noise_idx);
819 channels = q->nb_channels;
821 if (q->nb_channels <= 1 || sb < 12)
826 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
829 if (BITS_LEFT(length,gb) >= 16)
830 for (j = 0; j < 16; j++)
831 sign_bits[j] = get_bits1 (gb);
833 for (j = 0; j < 64; j++)
834 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
835 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
837 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
841 for (ch = 0; ch < channels; ch++) {
842 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
843 type34_predictor = 0.0;
846 for (j = 0; j < 128; ) {
847 switch (q->coding_method[ch][sb][j / 2]) {
849 if (BITS_LEFT(length,gb) >= 10) {
851 for (k = 0; k < 5; k++) {
852 if ((j + 2 * k) >= 128)
854 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
858 for (k = 0; k < 5; k++)
859 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
861 for (k = 0; k < 5; k++)
862 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
864 for (k = 0; k < 10; k++)
865 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
871 if (BITS_LEFT(length,gb) >= 1) {
876 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
879 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
885 if (BITS_LEFT(length,gb) >= 10) {
887 for (k = 0; k < 5; k++) {
890 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
893 n = get_bits (gb, 8);
894 for (k = 0; k < 5; k++)
895 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
898 for (k = 0; k < 5; k++)
899 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
905 if (BITS_LEFT(length,gb) >= 7) {
907 for (k = 0; k < 3; k++)
908 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
910 for (k = 0; k < 3; k++)
911 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
917 if (BITS_LEFT(length,gb) >= 4)
918 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
920 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
926 if (BITS_LEFT(length,gb) >= 7) {
928 type34_div = (float)(1 << get_bits(gb, 2));
929 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
930 type34_predictor = samples[0];
933 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
934 type34_predictor = samples[0];
937 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
943 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
949 float tmp[10][MPA_MAX_CHANNELS];
951 for (k = 0; k < run; k++) {
952 tmp[k][0] = samples[k];
953 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
955 for (chs = 0; chs < q->nb_channels; chs++)
956 for (k = 0; k < run; k++)
958 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
960 for (k = 0; k < run; k++)
962 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
973 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
974 * This is similar to process_subpacket_9, but for a single channel and for element [0]
975 * same VLC tables as process_subpacket_9 are used.
978 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
979 * @param gb bitreader context
980 * @param length packet length in bits
982 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
984 int i, k, run, level, diff;
986 if (BITS_LEFT(length,gb) < 16)
988 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
990 quantized_coeffs[0] = level;
992 for (i = 0; i < 7; ) {
993 if (BITS_LEFT(length,gb) < 16)
995 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
997 if (BITS_LEFT(length,gb) < 16)
999 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
1001 for (k = 1; k <= run; k++)
1002 quantized_coeffs[i + k] = (level + ((k * diff) / run));
1011 * Related to synthesis filter, process data from packet 10
1012 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1013 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1016 * @param gb bitreader context
1017 * @param length packet length in bits
1019 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1021 int sb, j, k, n, ch;
1023 for (ch = 0; ch < q->nb_channels; ch++) {
1024 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1026 if (BITS_LEFT(length,gb) < 16) {
1027 memset(q->quantized_coeffs[ch][0], 0, 8);
1032 n = q->sub_sampling + 1;
1034 for (sb = 0; sb < n; sb++)
1035 for (ch = 0; ch < q->nb_channels; ch++)
1036 for (j = 0; j < 8; j++) {
1037 if (BITS_LEFT(length,gb) < 1)
1039 if (get_bits1(gb)) {
1040 for (k=0; k < 8; k++) {
1041 if (BITS_LEFT(length,gb) < 16)
1043 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1046 for (k=0; k < 8; k++)
1047 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1051 n = QDM2_SB_USED(q->sub_sampling) - 4;
1053 for (sb = 0; sb < n; sb++)
1054 for (ch = 0; ch < q->nb_channels; ch++) {
1055 if (BITS_LEFT(length,gb) < 16)
1057 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1059 q->tone_level_idx_hi2[ch][sb] -= 16;
1061 for (j = 0; j < 8; j++)
1062 q->tone_level_idx_mid[ch][sb][j] = -16;
1065 n = QDM2_SB_USED(q->sub_sampling) - 5;
1067 for (sb = 0; sb < n; sb++)
1068 for (ch = 0; ch < q->nb_channels; ch++)
1069 for (j = 0; j < 8; j++) {
1070 if (BITS_LEFT(length,gb) < 16)
1072 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1077 * Process subpacket 9, init quantized_coeffs with data from it
1080 * @param node pointer to node with packet
1082 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1085 int i, j, k, n, ch, run, level, diff;
1087 init_get_bits(&gb, node->packet->data, node->packet->size*8);
1089 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1091 for (i = 1; i < n; i++)
1092 for (ch=0; ch < q->nb_channels; ch++) {
1093 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1094 q->quantized_coeffs[ch][i][0] = level;
1096 for (j = 0; j < (8 - 1); ) {
1097 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1098 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1100 for (k = 1; k <= run; k++)
1101 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1108 for (ch = 0; ch < q->nb_channels; ch++)
1109 for (i = 0; i < 8; i++)
1110 q->quantized_coeffs[ch][0][i] = 0;
1115 * Process subpacket 10 if not null, else
1118 * @param node pointer to node with packet
1119 * @param length packet length in bits
1121 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1125 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1128 init_tone_level_dequantization(q, &gb, length);
1129 fill_tone_level_array(q, 1);
1131 fill_tone_level_array(q, 0);
1137 * Process subpacket 11
1140 * @param node pointer to node with packet
1141 * @param length packet length in bit
1143 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1147 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1149 int c = get_bits (&gb, 13);
1152 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1153 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1156 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1161 * Process subpacket 12
1164 * @param node pointer to node with packet
1165 * @param length packet length in bits
1167 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1171 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1172 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1176 * Process new subpackets for synthesis filter
1179 * @param list list with synthesis filter packets (list D)
1181 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1183 QDM2SubPNode *nodes[4];
1185 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1186 if (nodes[0] != NULL)
1187 process_subpacket_9(q, nodes[0]);
1189 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1190 if (nodes[1] != NULL)
1191 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1193 process_subpacket_10(q, NULL, 0);
1195 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1196 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1197 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1199 process_subpacket_11(q, NULL, 0);
1201 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1202 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1203 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1205 process_subpacket_12(q, NULL, 0);
1210 * Decode superblock, fill packet lists.
1214 static void qdm2_decode_super_block (QDM2Context *q)
1217 QDM2SubPacket header, *packet;
1218 int i, packet_bytes, sub_packet_size, sub_packets_D;
1219 unsigned int next_index = 0;
1221 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1222 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1223 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1225 q->sub_packets_B = 0;
1228 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1230 init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1231 qdm2_decode_sub_packet_header(&gb, &header);
1233 if (header.type < 2 || header.type >= 8) {
1235 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1239 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1240 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1242 init_get_bits(&gb, header.data, header.size*8);
1244 if (header.type == 2 || header.type == 4 || header.type == 5) {
1245 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1247 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1251 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1256 q->sub_packet_list_B[0].packet = NULL;
1257 q->sub_packet_list_D[0].packet = NULL;
1259 for (i = 0; i < 6; i++)
1260 if (--q->fft_level_exp[i] < 0)
1261 q->fft_level_exp[i] = 0;
1263 for (i = 0; packet_bytes > 0; i++) {
1266 q->sub_packet_list_A[i].next = NULL;
1269 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1271 /* seek to next block */
1272 init_get_bits(&gb, header.data, header.size*8);
1273 skip_bits(&gb, next_index*8);
1275 if (next_index >= header.size)
1279 /* decode subpacket */
1280 packet = &q->sub_packets[i];
1281 qdm2_decode_sub_packet_header(&gb, packet);
1282 next_index = packet->size + get_bits_count(&gb) / 8;
1283 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1285 if (packet->type == 0)
1288 if (sub_packet_size > packet_bytes) {
1289 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1291 packet->size += packet_bytes - sub_packet_size;
1294 packet_bytes -= sub_packet_size;
1296 /* add subpacket to 'all subpackets' list */
1297 q->sub_packet_list_A[i].packet = packet;
1299 /* add subpacket to related list */
1300 if (packet->type == 8) {
1301 SAMPLES_NEEDED_2("packet type 8");
1303 } else if (packet->type >= 9 && packet->type <= 12) {
1304 /* packets for MPEG Audio like Synthesis Filter */
1305 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1306 } else if (packet->type == 13) {
1307 for (j = 0; j < 6; j++)
1308 q->fft_level_exp[j] = get_bits(&gb, 6);
1309 } else if (packet->type == 14) {
1310 for (j = 0; j < 6; j++)
1311 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1312 } else if (packet->type == 15) {
1313 SAMPLES_NEEDED_2("packet type 15")
1315 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1316 /* packets for FFT */
1317 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1319 } // Packet bytes loop
1321 /* **************************************************************** */
1322 if (q->sub_packet_list_D[0].packet != NULL) {
1323 process_synthesis_subpackets(q, q->sub_packet_list_D);
1324 q->do_synth_filter = 1;
1325 } else if (q->do_synth_filter) {
1326 process_subpacket_10(q, NULL, 0);
1327 process_subpacket_11(q, NULL, 0);
1328 process_subpacket_12(q, NULL, 0);
1330 /* **************************************************************** */
1334 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1335 int offset, int duration, int channel,
1338 if (q->fft_coefs_min_index[duration] < 0)
1339 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1341 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1342 q->fft_coefs[q->fft_coefs_index].channel = channel;
1343 q->fft_coefs[q->fft_coefs_index].offset = offset;
1344 q->fft_coefs[q->fft_coefs_index].exp = exp;
1345 q->fft_coefs[q->fft_coefs_index].phase = phase;
1346 q->fft_coefs_index++;
1350 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1352 int channel, stereo, phase, exp;
1353 int local_int_4, local_int_8, stereo_phase, local_int_10;
1354 int local_int_14, stereo_exp, local_int_20, local_int_28;
1360 local_int_8 = (4 - duration);
1361 local_int_10 = 1 << (q->group_order - duration - 1);
1365 if (q->superblocktype_2_3) {
1366 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1369 local_int_4 += local_int_10;
1370 local_int_28 += (1 << local_int_8);
1372 local_int_4 += 8*local_int_10;
1373 local_int_28 += (8 << local_int_8);
1378 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1379 while (offset >= (local_int_10 - 1)) {
1380 offset += (1 - (local_int_10 - 1));
1381 local_int_4 += local_int_10;
1382 local_int_28 += (1 << local_int_8);
1386 if (local_int_4 >= q->group_size)
1389 local_int_14 = (offset >> local_int_8);
1391 if (q->nb_channels > 1) {
1392 channel = get_bits1(gb);
1393 stereo = get_bits1(gb);
1399 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1400 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1401 exp = (exp < 0) ? 0 : exp;
1403 phase = get_bits(gb, 3);
1408 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1409 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1410 if (stereo_phase < 0)
1414 if (q->frequency_range > (local_int_14 + 1)) {
1415 int sub_packet = (local_int_20 + local_int_28);
1417 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1419 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1427 static void qdm2_decode_fft_packets (QDM2Context *q)
1429 int i, j, min, max, value, type, unknown_flag;
1432 if (q->sub_packet_list_B[0].packet == NULL)
1435 /* reset minimum indices for FFT coefficients */
1436 q->fft_coefs_index = 0;
1437 for (i=0; i < 5; i++)
1438 q->fft_coefs_min_index[i] = -1;
1440 /* process subpackets ordered by type, largest type first */
1441 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1442 QDM2SubPacket *packet;
1444 /* find subpacket with largest type less than max */
1445 for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) {
1446 value = q->sub_packet_list_B[j].packet->type;
1447 if (value > min && value < max) {
1449 packet = q->sub_packet_list_B[j].packet;
1455 /* check for errors (?) */
1456 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1459 /* decode FFT tones */
1460 init_get_bits (&gb, packet->data, packet->size*8);
1462 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1467 type = packet->type;
1469 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1470 int duration = q->sub_sampling + 5 - (type & 15);
1472 if (duration >= 0 && duration < 4)
1473 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1474 } else if (type == 31) {
1475 for (j=0; j < 4; j++)
1476 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1477 } else if (type == 46) {
1478 for (j=0; j < 6; j++)
1479 q->fft_level_exp[j] = get_bits(&gb, 6);
1480 for (j=0; j < 4; j++)
1481 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1483 } // Loop on B packets
1485 /* calculate maximum indices for FFT coefficients */
1486 for (i = 0, j = -1; i < 5; i++)
1487 if (q->fft_coefs_min_index[i] >= 0) {
1489 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1493 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1497 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1502 const double iscale = 2.0*M_PI / 512.0;
1504 tone->phase += tone->phase_shift;
1506 /* calculate current level (maximum amplitude) of tone */
1507 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1508 c.im = level * sin(tone->phase*iscale);
1509 c.re = level * cos(tone->phase*iscale);
1511 /* generate FFT coefficients for tone */
1512 if (tone->duration >= 3 || tone->cutoff >= 3) {
1513 tone->samples_im[0] += c.im;
1514 tone->samples_re[0] += c.re;
1515 tone->samples_im[1] -= c.im;
1516 tone->samples_re[1] -= c.re;
1518 f[1] = -tone->table[4];
1519 f[0] = tone->table[3] - tone->table[0];
1520 f[2] = 1.0 - tone->table[2] - tone->table[3];
1521 f[3] = tone->table[1] + tone->table[4] - 1.0;
1522 f[4] = tone->table[0] - tone->table[1];
1523 f[5] = tone->table[2];
1524 for (i = 0; i < 2; i++) {
1525 tone->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i];
1526 tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1528 for (i = 0; i < 4; i++) {
1529 tone->samples_re[i] += c.re * f[i+2];
1530 tone->samples_im[i] += c.im * f[i+2];
1534 /* copy the tone if it has not yet died out */
1535 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1536 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1537 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1542 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1545 const double iscale = 0.25 * M_PI;
1547 for (ch = 0; ch < q->channels; ch++) {
1548 memset(q->fft.samples_im[ch], 0, q->fft_size * sizeof(float));
1549 memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float));
1553 /* apply FFT tones with duration 4 (1 FFT period) */
1554 if (q->fft_coefs_min_index[4] >= 0)
1555 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1559 if (q->fft_coefs[i].sub_packet != sub_packet)
1562 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1563 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];
1565 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1566 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1567 q->fft.samples_re[ch][q->fft_coefs[i].offset + 0] += c.re;
1568 q->fft.samples_im[ch][q->fft_coefs[i].offset + 0] += c.im;
1569 q->fft.samples_re[ch][q->fft_coefs[i].offset + 1] -= c.re;
1570 q->fft.samples_im[ch][q->fft_coefs[i].offset + 1] -= c.im;
1573 /* generate existing FFT tones */
1574 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1575 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1576 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1579 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1580 for (i = 0; i < 4; i++)
1581 if (q->fft_coefs_min_index[i] >= 0) {
1582 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1586 if (q->fft_coefs[j].sub_packet != sub_packet)
1590 offset = q->fft_coefs[j].offset >> four_i;
1591 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1593 if (offset < q->frequency_range) {
1595 tone.cutoff = offset;
1597 tone.cutoff = (offset >= 60) ? 3 : 2;
1599 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];
1600 tone.samples_im = &q->fft.samples_im[ch][offset];
1601 tone.samples_re = &q->fft.samples_re[ch][offset];
1602 tone.table = (float*)fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1603 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1604 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1606 tone.time_index = 0;
1608 qdm2_fft_generate_tone(q, &tone);
1611 q->fft_coefs_min_index[i] = j;
1616 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1618 const int n = 1 << (q->fft_order - 1);
1619 const int n2 = n >> 1;
1620 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f;
1621 float c, s, f0, f1, f2, f3;
1624 /* prerotation (or something like that) */
1625 for (i=1; i < n2; i++) {
1627 c = q->exptab[i].re;
1628 s = -q->exptab[i].im;
1629 f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain;
1630 f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain;
1631 f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain;
1632 f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain;
1633 q->fft.complex[i].re = s * f0 - c * f1 + f2;
1634 q->fft.complex[i].im = c * f0 + s * f1 + f3;
1635 q->fft.complex[j].re = -s * f0 + c * f1 + f2;
1636 q->fft.complex[j].im = c * f0 + s * f1 - f3;
1639 q->fft.complex[ 0].re = q->fft.samples_re[channel][ 0] * gain * 2.0;
1640 q->fft.complex[ 0].im = q->fft.samples_re[channel][ 0] * gain * 2.0;
1641 q->fft.complex[n2].re = q->fft.samples_re[channel][n2] * gain * 2.0;
1642 q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0;
1644 ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex);
1645 ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex);
1646 /* add samples to output buffer */
1647 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1648 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex)[i];
1654 * @param index subpacket number
1656 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1658 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
1659 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1661 /* copy sb_samples */
1662 sb_used = QDM2_SB_USED(q->sub_sampling);
1664 for (ch = 0; ch < q->channels; ch++)
1665 for (i = 0; i < 8; i++)
1666 for (k=sb_used; k < SBLIMIT; k++)
1667 q->sb_samples[ch][(8 * index) + i][k] = 0;
1669 for (ch = 0; ch < q->nb_channels; ch++) {
1670 OUT_INT *samples_ptr = samples + ch;
1672 for (i = 0; i < 8; i++) {
1673 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1674 mpa_window, &dither_state,
1675 samples_ptr, q->nb_channels,
1676 q->sb_samples[ch][(8 * index) + i]);
1677 samples_ptr += 32 * q->nb_channels;
1681 /* add samples to output buffer */
1682 sub_sampling = (4 >> q->sub_sampling);
1684 for (ch = 0; ch < q->channels; ch++)
1685 for (i = 0; i < q->frame_size; i++)
1686 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
1691 * Init static data (does not depend on specific file)
1695 static void qdm2_init(QDM2Context *q) {
1696 static int inited = 0;
1703 ff_mpa_synth_init(mpa_window);
1704 softclip_table_init();
1706 init_noise_samples();
1708 av_log(NULL, AV_LOG_DEBUG, "init done\n");
1713 static void dump_context(QDM2Context *q)
1716 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1717 PRINT("compressed_data",q->compressed_data);
1718 PRINT("compressed_size",q->compressed_size);
1719 PRINT("frame_size",q->frame_size);
1720 PRINT("checksum_size",q->checksum_size);
1721 PRINT("channels",q->channels);
1722 PRINT("nb_channels",q->nb_channels);
1723 PRINT("fft_frame_size",q->fft_frame_size);
1724 PRINT("fft_size",q->fft_size);
1725 PRINT("sub_sampling",q->sub_sampling);
1726 PRINT("fft_order",q->fft_order);
1727 PRINT("group_order",q->group_order);
1728 PRINT("group_size",q->group_size);
1729 PRINT("sub_packet",q->sub_packet);
1730 PRINT("frequency_range",q->frequency_range);
1731 PRINT("has_errors",q->has_errors);
1732 PRINT("fft_tone_end",q->fft_tone_end);
1733 PRINT("fft_tone_start",q->fft_tone_start);
1734 PRINT("fft_coefs_index",q->fft_coefs_index);
1735 PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1736 PRINT("cm_table_select",q->cm_table_select);
1737 PRINT("noise_idx",q->noise_idx);
1739 for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1741 FFTTone *t = &q->fft_tones[i];
1743 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1744 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1745 // PRINT(" level", t->level);
1746 PRINT(" phase", t->phase);
1747 PRINT(" phase_shift", t->phase_shift);
1748 PRINT(" duration", t->duration);
1749 PRINT(" samples_im", t->samples_im);
1750 PRINT(" samples_re", t->samples_re);
1751 PRINT(" table", t->table);
1759 * Init parameters from codec extradata
1761 static int qdm2_decode_init(AVCodecContext *avctx)
1763 QDM2Context *s = avctx->priv_data;
1766 int tmp_val, tmp, size;
1770 /* extradata parsing
1779 32 size (including this field)
1781 32 type (=QDM2 or QDMC)
1783 32 size (including this field, in bytes)
1784 32 tag (=QDCA) // maybe mandatory parameters
1787 32 samplerate (=44100)
1789 32 block size (=4096)
1790 32 frame size (=256) (for one channel)
1791 32 packet size (=1300)
1793 32 size (including this field, in bytes)
1794 32 tag (=QDCP) // maybe some tuneable parameters
1804 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1805 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1809 extradata = avctx->extradata;
1810 extradata_size = avctx->extradata_size;
1812 while (extradata_size > 7) {
1813 if (!memcmp(extradata, "frmaQDM", 7))
1819 if (extradata_size < 12) {
1820 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1825 if (memcmp(extradata, "frmaQDM", 7)) {
1826 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1830 if (extradata[7] == 'C') {
1832 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1837 extradata_size -= 8;
1839 size = BE_32(extradata);
1841 if(size > extradata_size){
1842 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1843 extradata_size, size);
1848 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1849 if (BE_32(extradata) != MKBETAG('Q','D','C','A')) {
1850 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1856 avctx->channels = s->nb_channels = s->channels = BE_32(extradata);
1859 avctx->sample_rate = BE_32(extradata);
1862 avctx->bit_rate = BE_32(extradata);
1865 s->group_size = BE_32(extradata);
1868 s->fft_size = BE_32(extradata);
1871 s->checksum_size = BE_32(extradata);
1874 s->fft_order = av_log2(s->fft_size) + 1;
1875 s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1877 // something like max decodable tones
1878 s->group_order = av_log2(s->group_size) + 1;
1879 s->frame_size = s->group_size / 16; // 16 iterations per super block
1881 s->sub_sampling = s->fft_order - 7;
1882 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1884 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1885 case 0: tmp = 40; break;
1886 case 1: tmp = 48; break;
1887 case 2: tmp = 56; break;
1888 case 3: tmp = 72; break;
1889 case 4: tmp = 80; break;
1890 case 5: tmp = 100;break;
1891 default: tmp=s->sub_sampling; break;
1894 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1895 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1896 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1897 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1898 s->cm_table_select = tmp_val;
1900 if (s->sub_sampling == 0)
1903 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1910 s->coeff_per_sb_select = 0;
1911 else if (tmp <= 16000)
1912 s->coeff_per_sb_select = 1;
1914 s->coeff_per_sb_select = 2;
1916 // Fail on unknown fft order, if it's > 9 it can overflow s->exptab[]
1917 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1918 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1922 ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1);
1924 for (i = 1; i < (1 << (s->fft_order - 2)); i++) {
1925 alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1));
1926 s->exptab[i].re = cos(alpha);
1927 s->exptab[i].im = sin(alpha);
1937 static int qdm2_decode_close(AVCodecContext *avctx)
1939 QDM2Context *s = avctx->priv_data;
1941 ff_fft_end(&s->fft_ctx);
1947 static void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out)
1950 const int frame_size = (q->frame_size * q->channels);
1952 /* select input buffer */
1953 q->compressed_data = in;
1954 q->compressed_size = q->checksum_size;
1958 /* copy old block, clear new block of output samples */
1959 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1960 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1962 /* decode block of QDM2 compressed data */
1963 if (q->sub_packet == 0) {
1964 q->has_errors = 0; // zero it for a new super block
1965 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1966 qdm2_decode_super_block(q);
1969 /* parse subpackets */
1970 if (!q->has_errors) {
1971 if (q->sub_packet == 2)
1972 qdm2_decode_fft_packets(q);
1974 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1977 /* sound synthesis stage 1 (FFT) */
1978 for (ch = 0; ch < q->channels; ch++) {
1979 qdm2_calculate_fft(q, ch, q->sub_packet);
1981 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1982 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1987 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1988 if (!q->has_errors && q->do_synth_filter)
1989 qdm2_synthesis_filter(q, q->sub_packet);
1991 q->sub_packet = (q->sub_packet + 1) % 16;
1993 /* clip and convert output float[] to 16bit signed samples */
1994 for (i = 0; i < frame_size; i++) {
1995 int value = (int)q->output_buffer[i];
1997 if (value > SOFTCLIP_THRESHOLD)
1998 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1999 else if (value < -SOFTCLIP_THRESHOLD)
2000 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
2007 static int qdm2_decode_frame(AVCodecContext *avctx,
2008 void *data, int *data_size,
2009 uint8_t *buf, int buf_size)
2011 QDM2Context *s = avctx->priv_data;
2015 if(buf_size < s->checksum_size)
2018 *data_size = s->channels * s->frame_size * sizeof(int16_t);
2020 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
2021 buf_size, buf, s->checksum_size, data, *data_size);
2023 qdm2_decode(s, buf, data);
2025 // reading only when next superblock found
2026 if (s->sub_packet == 0) {
2027 return s->checksum_size;
2033 AVCodec qdm2_decoder =
2036 .type = CODEC_TYPE_AUDIO,
2037 .id = CODEC_ID_QDM2,
2038 .priv_data_size = sizeof(QDM2Context),
2039 .init = qdm2_decode_init,
2040 .close = qdm2_decode_close,
2041 .decode = qdm2_decode_frame,