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
26 * @file libavcodec/qdm2.c
28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
29 * The decoder is not perfect yet, there are still some distortions
30 * especially on files encoded with 16 or 8 subbands.
37 #define ALT_BITSTREAM_READER_LE
42 #include "mpegaudio.h"
50 #define SOFTCLIP_THRESHOLD 27600
51 #define HARDCLIP_THRESHOLD 35716
54 #define QDM2_LIST_ADD(list, size, packet) \
57 list[size - 1].next = &list[size]; \
59 list[size].packet = packet; \
60 list[size].next = NULL; \
64 // Result is 8, 16 or 30
65 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
67 #define FIX_NOISE_IDX(noise_idx) \
68 if ((noise_idx) >= 3840) \
69 (noise_idx) -= 3840; \
71 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
73 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
75 #define SAMPLES_NEEDED \
76 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
78 #define SAMPLES_NEEDED_2(why) \
79 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
82 typedef int8_t sb_int8_array[2][30][64];
88 int type; ///< subpacket type
89 unsigned int size; ///< subpacket size
90 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
94 * A node in the subpacket list
96 typedef struct QDM2SubPNode {
97 QDM2SubPacket *packet; ///< packet
98 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
108 QDM2Complex *complex;
126 DECLARE_ALIGNED(16, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
130 * QDM2 decoder context
133 /// Parameters from codec header, do not change during playback
134 int nb_channels; ///< number of channels
135 int channels; ///< number of channels
136 int group_size; ///< size of frame group (16 frames per group)
137 int fft_size; ///< size of FFT, in complex numbers
138 int checksum_size; ///< size of data block, used also for checksum
140 /// Parameters built from header parameters, do not change during playback
141 int group_order; ///< order of frame group
142 int fft_order; ///< order of FFT (actually fftorder+1)
143 int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
144 int frame_size; ///< size of data frame
146 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
147 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
148 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
150 /// Packets and packet lists
151 QDM2SubPacket sub_packets[16]; ///< the packets themselves
152 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
153 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
154 int sub_packets_B; ///< number of packets on 'B' list
155 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
156 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
159 FFTTone fft_tones[1000];
162 FFTCoefficient fft_coefs[1000];
164 int fft_coefs_min_index[5];
165 int fft_coefs_max_index[5];
166 int fft_level_exp[6];
167 RDFTContext rdft_ctx;
171 const uint8_t *compressed_data;
173 float output_buffer[1024];
176 DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2];
177 int synth_buf_offset[MPA_MAX_CHANNELS];
178 DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
180 /// Mixed temporary data used in decoding
181 float tone_level[MPA_MAX_CHANNELS][30][64];
182 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
183 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
184 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
185 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
186 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
187 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
188 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
189 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
192 int has_errors; ///< packet has errors
193 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
194 int do_synth_filter; ///< used to perform or skip synthesis filter
197 int noise_idx; ///< index for dithering noise table
201 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
203 static VLC vlc_tab_level;
204 static VLC vlc_tab_diff;
205 static VLC vlc_tab_run;
206 static VLC fft_level_exp_alt_vlc;
207 static VLC fft_level_exp_vlc;
208 static VLC fft_stereo_exp_vlc;
209 static VLC fft_stereo_phase_vlc;
210 static VLC vlc_tab_tone_level_idx_hi1;
211 static VLC vlc_tab_tone_level_idx_mid;
212 static VLC vlc_tab_tone_level_idx_hi2;
213 static VLC vlc_tab_type30;
214 static VLC vlc_tab_type34;
215 static VLC vlc_tab_fft_tone_offset[5];
217 static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
218 static float noise_table[4096];
219 static uint8_t random_dequant_index[256][5];
220 static uint8_t random_dequant_type24[128][3];
221 static float noise_samples[128];
224 static av_cold void softclip_table_init(void) {
226 double dfl = SOFTCLIP_THRESHOLD - 32767;
227 float delta = 1.0 / -dfl;
228 for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
229 softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
233 // random generated table
234 static av_cold void rnd_table_init(void) {
238 uint64_t random_seed = 0;
239 float delta = 1.0 / 16384.0;
240 for(i = 0; i < 4096 ;i++) {
241 random_seed = random_seed * 214013 + 2531011;
242 noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
245 for (i = 0; i < 256 ;i++) {
248 for (j = 0; j < 5 ;j++) {
249 random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
250 ldw = (uint32_t)ldw % (uint32_t)random_seed;
251 tmp64_1 = (random_seed * 0x55555556);
252 hdw = (uint32_t)(tmp64_1 >> 32);
253 random_seed = (uint64_t)(hdw + (ldw >> 31));
256 for (i = 0; i < 128 ;i++) {
259 for (j = 0; j < 3 ;j++) {
260 random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
261 ldw = (uint32_t)ldw % (uint32_t)random_seed;
262 tmp64_1 = (random_seed * 0x66666667);
263 hdw = (uint32_t)(tmp64_1 >> 33);
264 random_seed = hdw + (ldw >> 31);
270 static av_cold void init_noise_samples(void) {
273 float delta = 1.0 / 16384.0;
274 for (i = 0; i < 128;i++) {
275 random_seed = random_seed * 214013 + 2531011;
276 noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
280 static const uint16_t qdm2_vlc_offs[] = {
281 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
284 static av_cold void qdm2_init_vlc(void)
286 static int vlcs_initialized = 0;
287 static VLC_TYPE qdm2_table[3838][2];
289 if (!vlcs_initialized) {
291 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
292 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
293 init_vlc (&vlc_tab_level, 8, 24,
294 vlc_tab_level_huffbits, 1, 1,
295 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
297 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
298 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
299 init_vlc (&vlc_tab_diff, 8, 37,
300 vlc_tab_diff_huffbits, 1, 1,
301 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
303 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
304 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
305 init_vlc (&vlc_tab_run, 5, 6,
306 vlc_tab_run_huffbits, 1, 1,
307 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
309 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
310 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
311 init_vlc (&fft_level_exp_alt_vlc, 8, 28,
312 fft_level_exp_alt_huffbits, 1, 1,
313 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
316 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
317 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
318 init_vlc (&fft_level_exp_vlc, 8, 20,
319 fft_level_exp_huffbits, 1, 1,
320 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
322 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
323 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
324 init_vlc (&fft_stereo_exp_vlc, 6, 7,
325 fft_stereo_exp_huffbits, 1, 1,
326 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
328 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
329 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
330 init_vlc (&fft_stereo_phase_vlc, 6, 9,
331 fft_stereo_phase_huffbits, 1, 1,
332 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
334 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
335 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
336 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
337 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
338 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
340 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
341 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
342 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
343 vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
344 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
346 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
347 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
348 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
349 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
350 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
352 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
353 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
354 init_vlc (&vlc_tab_type30, 6, 9,
355 vlc_tab_type30_huffbits, 1, 1,
356 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
358 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
359 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
360 init_vlc (&vlc_tab_type34, 5, 10,
361 vlc_tab_type34_huffbits, 1, 1,
362 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
364 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
365 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
366 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
367 vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
368 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
370 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
371 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
372 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
373 vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
374 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
376 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
377 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
378 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
379 vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
380 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
382 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
383 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
384 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
385 vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
386 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
388 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
389 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
390 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
391 vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
392 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
399 /* for floating point to fixed point conversion */
400 static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
403 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
407 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
409 /* stage-2, 3 bits exponent escape sequence */
411 value = get_bits (gb, get_bits (gb, 3) + 1);
413 /* stage-3, optional */
415 int tmp = vlc_stage3_values[value];
417 if ((value & ~3) > 0)
418 tmp += get_bits (gb, (value >> 2));
426 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
428 int value = qdm2_get_vlc (gb, vlc, 0, depth);
430 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
437 * @param data pointer to data to be checksum'ed
438 * @param length data length
439 * @param value checksum value
441 * @return 0 if checksum is OK
443 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
446 for (i=0; i < length; i++)
449 return (uint16_t)(value & 0xffff);
454 * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
456 * @param gb bitreader context
457 * @param sub_packet packet under analysis
459 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
461 sub_packet->type = get_bits (gb, 8);
463 if (sub_packet->type == 0) {
464 sub_packet->size = 0;
465 sub_packet->data = NULL;
467 sub_packet->size = get_bits (gb, 8);
469 if (sub_packet->type & 0x80) {
470 sub_packet->size <<= 8;
471 sub_packet->size |= get_bits (gb, 8);
472 sub_packet->type &= 0x7f;
475 if (sub_packet->type == 0x7f)
476 sub_packet->type |= (get_bits (gb, 8) << 8);
478 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
481 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
482 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
487 * Return node pointer to first packet of requested type in list.
489 * @param list list of subpackets to be scanned
490 * @param type type of searched subpacket
491 * @return node pointer for subpacket if found, else NULL
493 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
495 while (list != NULL && list->packet != NULL) {
496 if (list->packet->type == type)
505 * Replaces 8 elements with their average value.
506 * Called by qdm2_decode_superblock before starting subblock decoding.
510 static void average_quantized_coeffs (QDM2Context *q)
512 int i, j, n, ch, sum;
514 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
516 for (ch = 0; ch < q->nb_channels; ch++)
517 for (i = 0; i < n; i++) {
520 for (j = 0; j < 8; j++)
521 sum += q->quantized_coeffs[ch][i][j];
527 for (j=0; j < 8; j++)
528 q->quantized_coeffs[ch][i][j] = sum;
534 * Build subband samples with noise weighted by q->tone_level.
535 * Called by synthfilt_build_sb_samples.
538 * @param sb subband index
540 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
544 FIX_NOISE_IDX(q->noise_idx);
549 for (ch = 0; ch < q->nb_channels; ch++)
550 for (j = 0; j < 64; j++) {
551 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);
552 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);
558 * Called while processing data from subpackets 11 and 12.
559 * Used after making changes to coding_method array.
561 * @param sb subband index
562 * @param channels number of channels
563 * @param coding_method q->coding_method[0][0][0]
565 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
570 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};
572 for (ch = 0; ch < channels; ch++) {
573 for (j = 0; j < 64; ) {
574 if((coding_method[ch][sb][j] - 8) > 22) {
578 switch (switchtable[coding_method[ch][sb][j]-8]) {
579 case 0: run = 10; case_val = 10; break;
580 case 1: run = 1; case_val = 16; break;
581 case 2: run = 5; case_val = 24; break;
582 case 3: run = 3; case_val = 30; break;
583 case 4: run = 1; case_val = 30; break;
584 case 5: run = 1; case_val = 8; break;
585 default: run = 1; case_val = 8; break;
588 for (k = 0; k < run; k++)
590 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
593 //not debugged, almost never used
594 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
595 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
604 * Related to synthesis filter
605 * Called by process_subpacket_10
608 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
610 static void fill_tone_level_array (QDM2Context *q, int flag)
612 int i, sb, ch, sb_used;
615 // This should never happen
616 if (q->nb_channels <= 0)
619 for (ch = 0; ch < q->nb_channels; ch++)
620 for (sb = 0; sb < 30; sb++)
621 for (i = 0; i < 8; i++) {
622 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
623 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
624 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
626 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
629 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
632 sb_used = QDM2_SB_USED(q->sub_sampling);
634 if ((q->superblocktype_2_3 != 0) && !flag) {
635 for (sb = 0; sb < sb_used; sb++)
636 for (ch = 0; ch < q->nb_channels; ch++)
637 for (i = 0; i < 64; i++) {
638 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
639 if (q->tone_level_idx[ch][sb][i] < 0)
640 q->tone_level[ch][sb][i] = 0;
642 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
645 tab = q->superblocktype_2_3 ? 0 : 1;
646 for (sb = 0; sb < sb_used; sb++) {
647 if ((sb >= 4) && (sb <= 23)) {
648 for (ch = 0; ch < q->nb_channels; ch++)
649 for (i = 0; i < 64; i++) {
650 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
651 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
652 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
653 q->tone_level_idx_hi2[ch][sb - 4];
654 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
655 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
656 q->tone_level[ch][sb][i] = 0;
658 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
662 for (ch = 0; ch < q->nb_channels; ch++)
663 for (i = 0; i < 64; i++) {
664 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
665 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
666 q->tone_level_idx_hi2[ch][sb - 4];
667 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
668 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
669 q->tone_level[ch][sb][i] = 0;
671 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
674 for (ch = 0; ch < q->nb_channels; ch++)
675 for (i = 0; i < 64; i++) {
676 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
677 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
678 q->tone_level[ch][sb][i] = 0;
680 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
692 * Related to synthesis filter
693 * Called by process_subpacket_11
694 * c is built with data from subpacket 11
695 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
697 * @param tone_level_idx
698 * @param tone_level_idx_temp
699 * @param coding_method q->coding_method[0][0][0]
700 * @param nb_channels number of channels
701 * @param c coming from subpacket 11, passed as 8*c
702 * @param superblocktype_2_3 flag based on superblock packet type
703 * @param cm_table_select q->cm_table_select
705 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
706 sb_int8_array coding_method, int nb_channels,
707 int c, int superblocktype_2_3, int cm_table_select)
710 int tmp, acc, esp_40, comp;
711 int add1, add2, add3, add4;
714 // This should never happen
715 if (nb_channels <= 0)
718 if (!superblocktype_2_3) {
719 /* This case is untested, no samples available */
721 for (ch = 0; ch < nb_channels; ch++)
722 for (sb = 0; sb < 30; sb++) {
723 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
724 add1 = tone_level_idx[ch][sb][j] - 10;
727 add2 = add3 = add4 = 0;
729 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
734 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
739 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
743 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
746 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
748 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
751 for (ch = 0; ch < nb_channels; ch++)
752 for (sb = 0; sb < 30; sb++)
753 for (j = 0; j < 64; j++)
754 acc += tone_level_idx_temp[ch][sb][j];
756 multres = 0x66666667 * (acc * 10);
757 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
758 for (ch = 0; ch < nb_channels; ch++)
759 for (sb = 0; sb < 30; sb++)
760 for (j = 0; j < 64; j++) {
761 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
764 comp /= 256; // signed shift
792 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
794 for (sb = 0; sb < 30; sb++)
795 fix_coding_method_array(sb, nb_channels, coding_method);
796 for (ch = 0; ch < nb_channels; ch++)
797 for (sb = 0; sb < 30; sb++)
798 for (j = 0; j < 64; j++)
800 if (coding_method[ch][sb][j] < 10)
801 coding_method[ch][sb][j] = 10;
804 if (coding_method[ch][sb][j] < 16)
805 coding_method[ch][sb][j] = 16;
807 if (coding_method[ch][sb][j] < 30)
808 coding_method[ch][sb][j] = 30;
811 } else { // superblocktype_2_3 != 0
812 for (ch = 0; ch < nb_channels; ch++)
813 for (sb = 0; sb < 30; sb++)
814 for (j = 0; j < 64; j++)
815 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
824 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
825 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
828 * @param gb bitreader context
829 * @param length packet length in bits
830 * @param sb_min lower subband processed (sb_min included)
831 * @param sb_max higher subband processed (sb_max excluded)
833 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
835 int sb, j, k, n, ch, run, channels;
836 int joined_stereo, zero_encoding, chs;
838 float type34_div = 0;
839 float type34_predictor;
840 float samples[10], sign_bits[16];
843 // If no data use noise
844 for (sb=sb_min; sb < sb_max; sb++)
845 build_sb_samples_from_noise (q, sb);
850 for (sb = sb_min; sb < sb_max; sb++) {
851 FIX_NOISE_IDX(q->noise_idx);
853 channels = q->nb_channels;
855 if (q->nb_channels <= 1 || sb < 12)
860 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
863 if (BITS_LEFT(length,gb) >= 16)
864 for (j = 0; j < 16; j++)
865 sign_bits[j] = get_bits1 (gb);
867 for (j = 0; j < 64; j++)
868 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
869 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
871 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
875 for (ch = 0; ch < channels; ch++) {
876 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
877 type34_predictor = 0.0;
880 for (j = 0; j < 128; ) {
881 switch (q->coding_method[ch][sb][j / 2]) {
883 if (BITS_LEFT(length,gb) >= 10) {
885 for (k = 0; k < 5; k++) {
886 if ((j + 2 * k) >= 128)
888 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
892 for (k = 0; k < 5; k++)
893 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
895 for (k = 0; k < 5; k++)
896 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
898 for (k = 0; k < 10; k++)
899 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
905 if (BITS_LEFT(length,gb) >= 1) {
910 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
913 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
919 if (BITS_LEFT(length,gb) >= 10) {
921 for (k = 0; k < 5; k++) {
924 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
927 n = get_bits (gb, 8);
928 for (k = 0; k < 5; k++)
929 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
932 for (k = 0; k < 5; k++)
933 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
939 if (BITS_LEFT(length,gb) >= 7) {
941 for (k = 0; k < 3; k++)
942 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
944 for (k = 0; k < 3; k++)
945 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
951 if (BITS_LEFT(length,gb) >= 4)
952 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
954 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
960 if (BITS_LEFT(length,gb) >= 7) {
962 type34_div = (float)(1 << get_bits(gb, 2));
963 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
964 type34_predictor = samples[0];
967 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
968 type34_predictor = samples[0];
971 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
977 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
983 float tmp[10][MPA_MAX_CHANNELS];
985 for (k = 0; k < run; k++) {
986 tmp[k][0] = samples[k];
987 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
989 for (chs = 0; chs < q->nb_channels; chs++)
990 for (k = 0; k < run; k++)
992 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
994 for (k = 0; k < run; k++)
996 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
1007 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
1008 * This is similar to process_subpacket_9, but for a single channel and for element [0]
1009 * same VLC tables as process_subpacket_9 are used.
1012 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
1013 * @param gb bitreader context
1014 * @param length packet length in bits
1016 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
1018 int i, k, run, level, diff;
1020 if (BITS_LEFT(length,gb) < 16)
1022 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
1024 quantized_coeffs[0] = level;
1026 for (i = 0; i < 7; ) {
1027 if (BITS_LEFT(length,gb) < 16)
1029 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
1031 if (BITS_LEFT(length,gb) < 16)
1033 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
1035 for (k = 1; k <= run; k++)
1036 quantized_coeffs[i + k] = (level + ((k * diff) / run));
1045 * Related to synthesis filter, process data from packet 10
1046 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1047 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1050 * @param gb bitreader context
1051 * @param length packet length in bits
1053 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1055 int sb, j, k, n, ch;
1057 for (ch = 0; ch < q->nb_channels; ch++) {
1058 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1060 if (BITS_LEFT(length,gb) < 16) {
1061 memset(q->quantized_coeffs[ch][0], 0, 8);
1066 n = q->sub_sampling + 1;
1068 for (sb = 0; sb < n; sb++)
1069 for (ch = 0; ch < q->nb_channels; ch++)
1070 for (j = 0; j < 8; j++) {
1071 if (BITS_LEFT(length,gb) < 1)
1073 if (get_bits1(gb)) {
1074 for (k=0; k < 8; k++) {
1075 if (BITS_LEFT(length,gb) < 16)
1077 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1080 for (k=0; k < 8; k++)
1081 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1085 n = QDM2_SB_USED(q->sub_sampling) - 4;
1087 for (sb = 0; sb < n; sb++)
1088 for (ch = 0; ch < q->nb_channels; ch++) {
1089 if (BITS_LEFT(length,gb) < 16)
1091 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1093 q->tone_level_idx_hi2[ch][sb] -= 16;
1095 for (j = 0; j < 8; j++)
1096 q->tone_level_idx_mid[ch][sb][j] = -16;
1099 n = QDM2_SB_USED(q->sub_sampling) - 5;
1101 for (sb = 0; sb < n; sb++)
1102 for (ch = 0; ch < q->nb_channels; ch++)
1103 for (j = 0; j < 8; j++) {
1104 if (BITS_LEFT(length,gb) < 16)
1106 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1111 * Process subpacket 9, init quantized_coeffs with data from it
1114 * @param node pointer to node with packet
1116 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1119 int i, j, k, n, ch, run, level, diff;
1121 init_get_bits(&gb, node->packet->data, node->packet->size*8);
1123 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1125 for (i = 1; i < n; i++)
1126 for (ch=0; ch < q->nb_channels; ch++) {
1127 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1128 q->quantized_coeffs[ch][i][0] = level;
1130 for (j = 0; j < (8 - 1); ) {
1131 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1132 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1134 for (k = 1; k <= run; k++)
1135 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1142 for (ch = 0; ch < q->nb_channels; ch++)
1143 for (i = 0; i < 8; i++)
1144 q->quantized_coeffs[ch][0][i] = 0;
1149 * Process subpacket 10 if not null, else
1152 * @param node pointer to node with packet
1153 * @param length packet length in bits
1155 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1159 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1162 init_tone_level_dequantization(q, &gb, length);
1163 fill_tone_level_array(q, 1);
1165 fill_tone_level_array(q, 0);
1171 * Process subpacket 11
1174 * @param node pointer to node with packet
1175 * @param length packet length in bit
1177 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1181 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1183 int c = get_bits (&gb, 13);
1186 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1187 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1190 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1195 * Process subpacket 12
1198 * @param node pointer to node with packet
1199 * @param length packet length in bits
1201 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1205 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1206 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1210 * Process new subpackets for synthesis filter
1213 * @param list list with synthesis filter packets (list D)
1215 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1217 QDM2SubPNode *nodes[4];
1219 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1220 if (nodes[0] != NULL)
1221 process_subpacket_9(q, nodes[0]);
1223 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1224 if (nodes[1] != NULL)
1225 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1227 process_subpacket_10(q, NULL, 0);
1229 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1230 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1231 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1233 process_subpacket_11(q, NULL, 0);
1235 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1236 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1237 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1239 process_subpacket_12(q, NULL, 0);
1244 * Decode superblock, fill packet lists.
1248 static void qdm2_decode_super_block (QDM2Context *q)
1251 QDM2SubPacket header, *packet;
1252 int i, packet_bytes, sub_packet_size, sub_packets_D;
1253 unsigned int next_index = 0;
1255 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1256 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1257 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1259 q->sub_packets_B = 0;
1262 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1264 init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1265 qdm2_decode_sub_packet_header(&gb, &header);
1267 if (header.type < 2 || header.type >= 8) {
1269 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1273 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1274 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1276 init_get_bits(&gb, header.data, header.size*8);
1278 if (header.type == 2 || header.type == 4 || header.type == 5) {
1279 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1281 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1285 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1290 q->sub_packet_list_B[0].packet = NULL;
1291 q->sub_packet_list_D[0].packet = NULL;
1293 for (i = 0; i < 6; i++)
1294 if (--q->fft_level_exp[i] < 0)
1295 q->fft_level_exp[i] = 0;
1297 for (i = 0; packet_bytes > 0; i++) {
1300 q->sub_packet_list_A[i].next = NULL;
1303 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1305 /* seek to next block */
1306 init_get_bits(&gb, header.data, header.size*8);
1307 skip_bits(&gb, next_index*8);
1309 if (next_index >= header.size)
1313 /* decode subpacket */
1314 packet = &q->sub_packets[i];
1315 qdm2_decode_sub_packet_header(&gb, packet);
1316 next_index = packet->size + get_bits_count(&gb) / 8;
1317 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1319 if (packet->type == 0)
1322 if (sub_packet_size > packet_bytes) {
1323 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1325 packet->size += packet_bytes - sub_packet_size;
1328 packet_bytes -= sub_packet_size;
1330 /* add subpacket to 'all subpackets' list */
1331 q->sub_packet_list_A[i].packet = packet;
1333 /* add subpacket to related list */
1334 if (packet->type == 8) {
1335 SAMPLES_NEEDED_2("packet type 8");
1337 } else if (packet->type >= 9 && packet->type <= 12) {
1338 /* packets for MPEG Audio like Synthesis Filter */
1339 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1340 } else if (packet->type == 13) {
1341 for (j = 0; j < 6; j++)
1342 q->fft_level_exp[j] = get_bits(&gb, 6);
1343 } else if (packet->type == 14) {
1344 for (j = 0; j < 6; j++)
1345 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1346 } else if (packet->type == 15) {
1347 SAMPLES_NEEDED_2("packet type 15")
1349 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1350 /* packets for FFT */
1351 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1353 } // Packet bytes loop
1355 /* **************************************************************** */
1356 if (q->sub_packet_list_D[0].packet != NULL) {
1357 process_synthesis_subpackets(q, q->sub_packet_list_D);
1358 q->do_synth_filter = 1;
1359 } else if (q->do_synth_filter) {
1360 process_subpacket_10(q, NULL, 0);
1361 process_subpacket_11(q, NULL, 0);
1362 process_subpacket_12(q, NULL, 0);
1364 /* **************************************************************** */
1368 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1369 int offset, int duration, int channel,
1372 if (q->fft_coefs_min_index[duration] < 0)
1373 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1375 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1376 q->fft_coefs[q->fft_coefs_index].channel = channel;
1377 q->fft_coefs[q->fft_coefs_index].offset = offset;
1378 q->fft_coefs[q->fft_coefs_index].exp = exp;
1379 q->fft_coefs[q->fft_coefs_index].phase = phase;
1380 q->fft_coefs_index++;
1384 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1386 int channel, stereo, phase, exp;
1387 int local_int_4, local_int_8, stereo_phase, local_int_10;
1388 int local_int_14, stereo_exp, local_int_20, local_int_28;
1394 local_int_8 = (4 - duration);
1395 local_int_10 = 1 << (q->group_order - duration - 1);
1399 if (q->superblocktype_2_3) {
1400 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1403 local_int_4 += local_int_10;
1404 local_int_28 += (1 << local_int_8);
1406 local_int_4 += 8*local_int_10;
1407 local_int_28 += (8 << local_int_8);
1412 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1413 while (offset >= (local_int_10 - 1)) {
1414 offset += (1 - (local_int_10 - 1));
1415 local_int_4 += local_int_10;
1416 local_int_28 += (1 << local_int_8);
1420 if (local_int_4 >= q->group_size)
1423 local_int_14 = (offset >> local_int_8);
1425 if (q->nb_channels > 1) {
1426 channel = get_bits1(gb);
1427 stereo = get_bits1(gb);
1433 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1434 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1435 exp = (exp < 0) ? 0 : exp;
1437 phase = get_bits(gb, 3);
1442 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1443 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1444 if (stereo_phase < 0)
1448 if (q->frequency_range > (local_int_14 + 1)) {
1449 int sub_packet = (local_int_20 + local_int_28);
1451 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1453 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1461 static void qdm2_decode_fft_packets (QDM2Context *q)
1463 int i, j, min, max, value, type, unknown_flag;
1466 if (q->sub_packet_list_B[0].packet == NULL)
1469 /* reset minimum indexes for FFT coefficients */
1470 q->fft_coefs_index = 0;
1471 for (i=0; i < 5; i++)
1472 q->fft_coefs_min_index[i] = -1;
1474 /* process subpackets ordered by type, largest type first */
1475 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1476 QDM2SubPacket *packet= NULL;
1478 /* find subpacket with largest type less than max */
1479 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1480 value = q->sub_packet_list_B[j].packet->type;
1481 if (value > min && value < max) {
1483 packet = q->sub_packet_list_B[j].packet;
1489 /* check for errors (?) */
1493 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1496 /* decode FFT tones */
1497 init_get_bits (&gb, packet->data, packet->size*8);
1499 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1504 type = packet->type;
1506 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1507 int duration = q->sub_sampling + 5 - (type & 15);
1509 if (duration >= 0 && duration < 4)
1510 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1511 } else if (type == 31) {
1512 for (j=0; j < 4; j++)
1513 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1514 } else if (type == 46) {
1515 for (j=0; j < 6; j++)
1516 q->fft_level_exp[j] = get_bits(&gb, 6);
1517 for (j=0; j < 4; j++)
1518 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1520 } // Loop on B packets
1522 /* calculate maximum indexes for FFT coefficients */
1523 for (i = 0, j = -1; i < 5; i++)
1524 if (q->fft_coefs_min_index[i] >= 0) {
1526 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1530 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1534 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1539 const double iscale = 2.0*M_PI / 512.0;
1541 tone->phase += tone->phase_shift;
1543 /* calculate current level (maximum amplitude) of tone */
1544 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1545 c.im = level * sin(tone->phase*iscale);
1546 c.re = level * cos(tone->phase*iscale);
1548 /* generate FFT coefficients for tone */
1549 if (tone->duration >= 3 || tone->cutoff >= 3) {
1550 tone->complex[0].im += c.im;
1551 tone->complex[0].re += c.re;
1552 tone->complex[1].im -= c.im;
1553 tone->complex[1].re -= c.re;
1555 f[1] = -tone->table[4];
1556 f[0] = tone->table[3] - tone->table[0];
1557 f[2] = 1.0 - tone->table[2] - tone->table[3];
1558 f[3] = tone->table[1] + tone->table[4] - 1.0;
1559 f[4] = tone->table[0] - tone->table[1];
1560 f[5] = tone->table[2];
1561 for (i = 0; i < 2; i++) {
1562 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1563 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1565 for (i = 0; i < 4; i++) {
1566 tone->complex[i].re += c.re * f[i+2];
1567 tone->complex[i].im += c.im * f[i+2];
1571 /* copy the tone if it has not yet died out */
1572 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1573 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1574 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1579 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1582 const double iscale = 0.25 * M_PI;
1584 for (ch = 0; ch < q->channels; ch++) {
1585 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1589 /* apply FFT tones with duration 4 (1 FFT period) */
1590 if (q->fft_coefs_min_index[4] >= 0)
1591 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1595 if (q->fft_coefs[i].sub_packet != sub_packet)
1598 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1599 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];
1601 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1602 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1603 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1604 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1605 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1606 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1609 /* generate existing FFT tones */
1610 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1611 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1612 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1615 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1616 for (i = 0; i < 4; i++)
1617 if (q->fft_coefs_min_index[i] >= 0) {
1618 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1622 if (q->fft_coefs[j].sub_packet != sub_packet)
1626 offset = q->fft_coefs[j].offset >> four_i;
1627 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1629 if (offset < q->frequency_range) {
1631 tone.cutoff = offset;
1633 tone.cutoff = (offset >= 60) ? 3 : 2;
1635 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];
1636 tone.complex = &q->fft.complex[ch][offset];
1637 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1638 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1639 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1641 tone.time_index = 0;
1643 qdm2_fft_generate_tone(q, &tone);
1646 q->fft_coefs_min_index[i] = j;
1651 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1653 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1655 q->fft.complex[channel][0].re *= 2.0f;
1656 q->fft.complex[channel][0].im = 0.0f;
1657 ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1658 /* add samples to output buffer */
1659 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1660 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
1666 * @param index subpacket number
1668 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1670 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
1671 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1673 /* copy sb_samples */
1674 sb_used = QDM2_SB_USED(q->sub_sampling);
1676 for (ch = 0; ch < q->channels; ch++)
1677 for (i = 0; i < 8; i++)
1678 for (k=sb_used; k < SBLIMIT; k++)
1679 q->sb_samples[ch][(8 * index) + i][k] = 0;
1681 for (ch = 0; ch < q->nb_channels; ch++) {
1682 OUT_INT *samples_ptr = samples + ch;
1684 for (i = 0; i < 8; i++) {
1685 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1686 ff_mpa_synth_window, &dither_state,
1687 samples_ptr, q->nb_channels,
1688 q->sb_samples[ch][(8 * index) + i]);
1689 samples_ptr += 32 * q->nb_channels;
1693 /* add samples to output buffer */
1694 sub_sampling = (4 >> q->sub_sampling);
1696 for (ch = 0; ch < q->channels; ch++)
1697 for (i = 0; i < q->frame_size; i++)
1698 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
1703 * Init static data (does not depend on specific file)
1707 static av_cold void qdm2_init(QDM2Context *q) {
1708 static int initialized = 0;
1710 if (initialized != 0)
1715 ff_mpa_synth_init(ff_mpa_synth_window);
1716 softclip_table_init();
1718 init_noise_samples();
1720 av_log(NULL, AV_LOG_DEBUG, "init done\n");
1725 static void dump_context(QDM2Context *q)
1728 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1729 PRINT("compressed_data",q->compressed_data);
1730 PRINT("compressed_size",q->compressed_size);
1731 PRINT("frame_size",q->frame_size);
1732 PRINT("checksum_size",q->checksum_size);
1733 PRINT("channels",q->channels);
1734 PRINT("nb_channels",q->nb_channels);
1735 PRINT("fft_frame_size",q->fft_frame_size);
1736 PRINT("fft_size",q->fft_size);
1737 PRINT("sub_sampling",q->sub_sampling);
1738 PRINT("fft_order",q->fft_order);
1739 PRINT("group_order",q->group_order);
1740 PRINT("group_size",q->group_size);
1741 PRINT("sub_packet",q->sub_packet);
1742 PRINT("frequency_range",q->frequency_range);
1743 PRINT("has_errors",q->has_errors);
1744 PRINT("fft_tone_end",q->fft_tone_end);
1745 PRINT("fft_tone_start",q->fft_tone_start);
1746 PRINT("fft_coefs_index",q->fft_coefs_index);
1747 PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1748 PRINT("cm_table_select",q->cm_table_select);
1749 PRINT("noise_idx",q->noise_idx);
1751 for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1753 FFTTone *t = &q->fft_tones[i];
1755 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1756 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1757 // PRINT(" level", t->level);
1758 PRINT(" phase", t->phase);
1759 PRINT(" phase_shift", t->phase_shift);
1760 PRINT(" duration", t->duration);
1761 PRINT(" samples_im", t->samples_im);
1762 PRINT(" samples_re", t->samples_re);
1763 PRINT(" table", t->table);
1771 * Init parameters from codec extradata
1773 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1775 QDM2Context *s = avctx->priv_data;
1778 int tmp_val, tmp, size;
1780 /* extradata parsing
1789 32 size (including this field)
1791 32 type (=QDM2 or QDMC)
1793 32 size (including this field, in bytes)
1794 32 tag (=QDCA) // maybe mandatory parameters
1797 32 samplerate (=44100)
1799 32 block size (=4096)
1800 32 frame size (=256) (for one channel)
1801 32 packet size (=1300)
1803 32 size (including this field, in bytes)
1804 32 tag (=QDCP) // maybe some tuneable parameters
1814 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1815 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1819 extradata = avctx->extradata;
1820 extradata_size = avctx->extradata_size;
1822 while (extradata_size > 7) {
1823 if (!memcmp(extradata, "frmaQDM", 7))
1829 if (extradata_size < 12) {
1830 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1835 if (memcmp(extradata, "frmaQDM", 7)) {
1836 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1840 if (extradata[7] == 'C') {
1842 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1847 extradata_size -= 8;
1849 size = AV_RB32(extradata);
1851 if(size > extradata_size){
1852 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1853 extradata_size, size);
1858 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1859 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1860 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1866 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1869 avctx->sample_rate = AV_RB32(extradata);
1872 avctx->bit_rate = AV_RB32(extradata);
1875 s->group_size = AV_RB32(extradata);
1878 s->fft_size = AV_RB32(extradata);
1881 s->checksum_size = AV_RB32(extradata);
1883 s->fft_order = av_log2(s->fft_size) + 1;
1884 s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1886 // something like max decodable tones
1887 s->group_order = av_log2(s->group_size) + 1;
1888 s->frame_size = s->group_size / 16; // 16 iterations per super block
1890 s->sub_sampling = s->fft_order - 7;
1891 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1893 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1894 case 0: tmp = 40; break;
1895 case 1: tmp = 48; break;
1896 case 2: tmp = 56; break;
1897 case 3: tmp = 72; break;
1898 case 4: tmp = 80; break;
1899 case 5: tmp = 100;break;
1900 default: tmp=s->sub_sampling; break;
1903 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1904 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1905 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1906 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1907 s->cm_table_select = tmp_val;
1909 if (s->sub_sampling == 0)
1912 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1919 s->coeff_per_sb_select = 0;
1920 else if (tmp <= 16000)
1921 s->coeff_per_sb_select = 1;
1923 s->coeff_per_sb_select = 2;
1925 // Fail on unknown fft order
1926 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1927 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1931 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1935 avctx->sample_fmt = SAMPLE_FMT_S16;
1942 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1944 QDM2Context *s = avctx->priv_data;
1946 ff_rdft_end(&s->rdft_ctx);
1952 static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1955 const int frame_size = (q->frame_size * q->channels);
1957 /* select input buffer */
1958 q->compressed_data = in;
1959 q->compressed_size = q->checksum_size;
1963 /* copy old block, clear new block of output samples */
1964 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1965 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1967 /* decode block of QDM2 compressed data */
1968 if (q->sub_packet == 0) {
1969 q->has_errors = 0; // zero it for a new super block
1970 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1971 qdm2_decode_super_block(q);
1974 /* parse subpackets */
1975 if (!q->has_errors) {
1976 if (q->sub_packet == 2)
1977 qdm2_decode_fft_packets(q);
1979 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1982 /* sound synthesis stage 1 (FFT) */
1983 for (ch = 0; ch < q->channels; ch++) {
1984 qdm2_calculate_fft(q, ch, q->sub_packet);
1986 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1987 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1992 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1993 if (!q->has_errors && q->do_synth_filter)
1994 qdm2_synthesis_filter(q, q->sub_packet);
1996 q->sub_packet = (q->sub_packet + 1) % 16;
1998 /* clip and convert output float[] to 16bit signed samples */
1999 for (i = 0; i < frame_size; i++) {
2000 int value = (int)q->output_buffer[i];
2002 if (value > SOFTCLIP_THRESHOLD)
2003 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
2004 else if (value < -SOFTCLIP_THRESHOLD)
2005 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
2012 static int qdm2_decode_frame(AVCodecContext *avctx,
2013 void *data, int *data_size,
2016 const uint8_t *buf = avpkt->data;
2017 int buf_size = avpkt->size;
2018 QDM2Context *s = avctx->priv_data;
2022 if(buf_size < s->checksum_size)
2025 *data_size = s->channels * s->frame_size * sizeof(int16_t);
2027 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
2028 buf_size, buf, s->checksum_size, data, *data_size);
2030 qdm2_decode(s, buf, data);
2032 // reading only when next superblock found
2033 if (s->sub_packet == 0) {
2034 return s->checksum_size;
2040 AVCodec qdm2_decoder =
2043 .type = CODEC_TYPE_AUDIO,
2044 .id = CODEC_ID_QDM2,
2045 .priv_data_size = sizeof(QDM2Context),
2046 .init = qdm2_decode_init,
2047 .close = qdm2_decode_close,
2048 .decode = qdm2_decode_frame,
2049 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),