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Merge commit 'origin/master' into ordered_chapters
[FFMpeg-mirror/ordered_chapters.git] / libavcodec / qdm2.c
blobdf1479c25ed46485c031111fbf09721044ad71ea
1 /*
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
7 *
8 * This file is part of FFmpeg.
9 *
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.
14 *
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.
19 *
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
23 */
24
25 /**
26 * @file libavcodec/qdm2.c
27 * QDM2 decoder
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.
31 */
32
33 #include <math.h>
34 #include <stddef.h>
35 #include <stdio.h>
36
37 #define ALT_BITSTREAM_READER_LE
38 #include "avcodec.h"
39 #include "get_bits.h"
40 #include "dsputil.h"
41 #include "mpegaudio.h"
42
43 #include "qdm2data.h"
44
45 #undef NDEBUG
46 #include <assert.h>
47
48
49 #define SOFTCLIP_THRESHOLD 27600
50 #define HARDCLIP_THRESHOLD 35716
51
52
53 #define QDM2_LIST_ADD(list, size, packet) \
54 do { \
55 if (size > 0) { \
56 list[size - 1].next = &list[size]; \
57 } \
58 list[size].packet = packet; \
59 list[size].next = NULL; \
60 size++; \
61 } while(0)
62
63 // Result is 8, 16 or 30
64 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
65
66 #define FIX_NOISE_IDX(noise_idx) \
67 if ((noise_idx) >= 3840) \
68 (noise_idx) -= 3840; \
69
70 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
71
72 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
73
74 #define SAMPLES_NEEDED \
75 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
76
77 #define SAMPLES_NEEDED_2(why) \
78 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
79
80
81 typedef int8_t sb_int8_array[2][30][64];
82
83 /**
84 * Subpacket
85 */
86 typedef struct {
87 int type; ///< subpacket type
88 unsigned int size; ///< subpacket size
89 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
90 } QDM2SubPacket;
91
92 /**
93 * A node in the subpacket list
94 */
95 typedef struct QDM2SubPNode {
96 QDM2SubPacket *packet; ///< packet
97 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
98 } QDM2SubPNode;
99
100 typedef struct {
101 float re;
102 float im;
103 } QDM2Complex;
104
105 typedef struct {
106 float level;
107 QDM2Complex *complex;
108 const float *table;
109 int phase;
110 int phase_shift;
111 int duration;
112 short time_index;
113 short cutoff;
114 } FFTTone;
115
116 typedef struct {
117 int16_t sub_packet;
118 uint8_t channel;
119 int16_t offset;
120 int16_t exp;
121 uint8_t phase;
122 } FFTCoefficient;
123
124 typedef struct {
125 DECLARE_ALIGNED_16(QDM2Complex, complex[MPA_MAX_CHANNELS][256]);
126 } QDM2FFT;
127
128 /**
129 * QDM2 decoder context
130 */
131 typedef struct {
132 /// Parameters from codec header, do not change during playback
133 int nb_channels; ///< number of channels
134 int channels; ///< number of channels
135 int group_size; ///< size of frame group (16 frames per group)
136 int fft_size; ///< size of FFT, in complex numbers
137 int checksum_size; ///< size of data block, used also for checksum
138
139 /// Parameters built from header parameters, do not change during playback
140 int group_order; ///< order of frame group
141 int fft_order; ///< order of FFT (actually fftorder+1)
142 int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
143 int frame_size; ///< size of data frame
144 int frequency_range;
145 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
146 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
147 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
148
149 /// Packets and packet lists
150 QDM2SubPacket sub_packets[16]; ///< the packets themselves
151 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
152 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
153 int sub_packets_B; ///< number of packets on 'B' list
154 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
155 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
156
157 /// FFT and tones
158 FFTTone fft_tones[1000];
159 int fft_tone_start;
160 int fft_tone_end;
161 FFTCoefficient fft_coefs[1000];
162 int fft_coefs_index;
163 int fft_coefs_min_index[5];
164 int fft_coefs_max_index[5];
165 int fft_level_exp[6];
166 RDFTContext rdft_ctx;
167 QDM2FFT fft;
168
169 /// I/O data
170 const uint8_t *compressed_data;
171 int compressed_size;
172 float output_buffer[1024];
173
174 /// Synthesis filter
175 DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]);
176 int synth_buf_offset[MPA_MAX_CHANNELS];
177 DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]);
178
179 /// Mixed temporary data used in decoding
180 float tone_level[MPA_MAX_CHANNELS][30][64];
181 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
182 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
183 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
184 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
185 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
186 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
187 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
188 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
189
190 // Flags
191 int has_errors; ///< packet has errors
192 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
193 int do_synth_filter; ///< used to perform or skip synthesis filter
194
195 int sub_packet;
196 int noise_idx; ///< index for dithering noise table
197 } QDM2Context;
198
199
200 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
201
202 static VLC vlc_tab_level;
203 static VLC vlc_tab_diff;
204 static VLC vlc_tab_run;
205 static VLC fft_level_exp_alt_vlc;
206 static VLC fft_level_exp_vlc;
207 static VLC fft_stereo_exp_vlc;
208 static VLC fft_stereo_phase_vlc;
209 static VLC vlc_tab_tone_level_idx_hi1;
210 static VLC vlc_tab_tone_level_idx_mid;
211 static VLC vlc_tab_tone_level_idx_hi2;
212 static VLC vlc_tab_type30;
213 static VLC vlc_tab_type34;
214 static VLC vlc_tab_fft_tone_offset[5];
215
216 static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
217 static float noise_table[4096];
218 static uint8_t random_dequant_index[256][5];
219 static uint8_t random_dequant_type24[128][3];
220 static float noise_samples[128];
221
222 static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);
223
224
225 static av_cold void softclip_table_init(void) {
226 int i;
227 double dfl = SOFTCLIP_THRESHOLD - 32767;
228 float delta = 1.0 / -dfl;
229 for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
230 softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
231 }
232
233
234 // random generated table
235 static av_cold void rnd_table_init(void) {
236 int i,j;
237 uint32_t ldw,hdw;
238 uint64_t tmp64_1;
239 uint64_t random_seed = 0;
240 float delta = 1.0 / 16384.0;
241 for(i = 0; i < 4096 ;i++) {
242 random_seed = random_seed * 214013 + 2531011;
243 noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
244 }
245
246 for (i = 0; i < 256 ;i++) {
247 random_seed = 81;
248 ldw = i;
249 for (j = 0; j < 5 ;j++) {
250 random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
251 ldw = (uint32_t)ldw % (uint32_t)random_seed;
252 tmp64_1 = (random_seed * 0x55555556);
253 hdw = (uint32_t)(tmp64_1 >> 32);
254 random_seed = (uint64_t)(hdw + (ldw >> 31));
255 }
256 }
257 for (i = 0; i < 128 ;i++) {
258 random_seed = 25;
259 ldw = i;
260 for (j = 0; j < 3 ;j++) {
261 random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
262 ldw = (uint32_t)ldw % (uint32_t)random_seed;
263 tmp64_1 = (random_seed * 0x66666667);
264 hdw = (uint32_t)(tmp64_1 >> 33);
265 random_seed = hdw + (ldw >> 31);
266 }
267 }
268 }
269
270
271 static av_cold void init_noise_samples(void) {
272 int i;
273 int random_seed = 0;
274 float delta = 1.0 / 16384.0;
275 for (i = 0; i < 128;i++) {
276 random_seed = random_seed * 214013 + 2531011;
277 noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
278 }
279 }
280
281 static const uint16_t qdm2_vlc_offs[] = {
282 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
283 };
284
285 static av_cold void qdm2_init_vlc(void)
286 {
287 static int vlcs_initialized = 0;
288 static VLC_TYPE qdm2_table[3838][2];
289
290 if (!vlcs_initialized) {
291
292 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
293 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
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_NEW_STATIC | INIT_VLC_LE);
297
298 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
299 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
300 init_vlc (&vlc_tab_diff, 8, 37,
301 vlc_tab_diff_huffbits, 1, 1,
302 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
303
304 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
305 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
306 init_vlc (&vlc_tab_run, 5, 6,
307 vlc_tab_run_huffbits, 1, 1,
308 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
309
310 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
311 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
312 init_vlc (&fft_level_exp_alt_vlc, 8, 28,
313 fft_level_exp_alt_huffbits, 1, 1,
314 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
315
316
317 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
318 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
319 init_vlc (&fft_level_exp_vlc, 8, 20,
320 fft_level_exp_huffbits, 1, 1,
321 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
322
323 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
324 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
325 init_vlc (&fft_stereo_exp_vlc, 6, 7,
326 fft_stereo_exp_huffbits, 1, 1,
327 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
328
329 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
330 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
331 init_vlc (&fft_stereo_phase_vlc, 6, 9,
332 fft_stereo_phase_huffbits, 1, 1,
333 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
334
335 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
336 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
337 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
338 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
339 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
340
341 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
342 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
343 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
344 vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
345 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
346
347 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
348 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
349 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
350 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
351 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
352
353 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
354 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
355 init_vlc (&vlc_tab_type30, 6, 9,
356 vlc_tab_type30_huffbits, 1, 1,
357 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
358
359 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
360 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
361 init_vlc (&vlc_tab_type34, 5, 10,
362 vlc_tab_type34_huffbits, 1, 1,
363 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
364
365 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
366 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
367 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
368 vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
369 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
370
371 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
372 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
373 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
374 vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
375 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
376
377 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
378 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
379 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
380 vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
381 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
382
383 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
384 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
385 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
386 vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
387 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
388
389 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
390 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
391 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
392 vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
393 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
394
395 vlcs_initialized=1;
396 }
397 }
398
399
400 /* for floating point to fixed point conversion */
401 static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
402
403
404 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
405 {
406 int value;
407
408 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
409
410 /* stage-2, 3 bits exponent escape sequence */
411 if (value-- == 0)
412 value = get_bits (gb, get_bits (gb, 3) + 1);
413
414 /* stage-3, optional */
415 if (flag) {
416 int tmp = vlc_stage3_values[value];
417
418 if ((value & ~3) > 0)
419 tmp += get_bits (gb, (value >> 2));
420 value = tmp;
421 }
422
423 return value;
424 }
425
426
427 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
428 {
429 int value = qdm2_get_vlc (gb, vlc, 0, depth);
430
431 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
432 }
433
434
435 /**
436 * QDM2 checksum
437 *
438 * @param data pointer to data to be checksum'ed
439 * @param length data length
440 * @param value checksum value
441 *
442 * @return 0 if checksum is OK
443 */
444 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
445 int i;
446
447 for (i=0; i < length; i++)
448 value -= data[i];
449
450 return (uint16_t)(value & 0xffff);
451 }
452
453
454 /**
455 * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
456 *
457 * @param gb bitreader context
458 * @param sub_packet packet under analysis
459 */
460 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
461 {
462 sub_packet->type = get_bits (gb, 8);
463
464 if (sub_packet->type == 0) {
465 sub_packet->size = 0;
466 sub_packet->data = NULL;
467 } else {
468 sub_packet->size = get_bits (gb, 8);
469
470 if (sub_packet->type & 0x80) {
471 sub_packet->size <<= 8;
472 sub_packet->size |= get_bits (gb, 8);
473 sub_packet->type &= 0x7f;
474 }
475
476 if (sub_packet->type == 0x7f)
477 sub_packet->type |= (get_bits (gb, 8) << 8);
478
479 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
480 }
481
482 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
483 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
484 }
485
486
487 /**
488 * Return node pointer to first packet of requested type in list.
489 *
490 * @param list list of subpackets to be scanned
491 * @param type type of searched subpacket
492 * @return node pointer for subpacket if found, else NULL
493 */
494 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
495 {
496 while (list != NULL && list->packet != NULL) {
497 if (list->packet->type == type)
498 return list;
499 list = list->next;
500 }
501 return NULL;
502 }
503
504
505 /**
506 * Replaces 8 elements with their average value.
507 * Called by qdm2_decode_superblock before starting subblock decoding.
508 *
509 * @param q context
510 */
511 static void average_quantized_coeffs (QDM2Context *q)
512 {
513 int i, j, n, ch, sum;
514
515 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
516
517 for (ch = 0; ch < q->nb_channels; ch++)
518 for (i = 0; i < n; i++) {
519 sum = 0;
520
521 for (j = 0; j < 8; j++)
522 sum += q->quantized_coeffs[ch][i][j];
523
524 sum /= 8;
525 if (sum > 0)
526 sum--;
527
528 for (j=0; j < 8; j++)
529 q->quantized_coeffs[ch][i][j] = sum;
530 }
531 }
532
533
534 /**
535 * Build subband samples with noise weighted by q->tone_level.
536 * Called by synthfilt_build_sb_samples.
537 *
538 * @param q context
539 * @param sb subband index
540 */
541 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
542 {
543 int ch, j;
544
545 FIX_NOISE_IDX(q->noise_idx);
546
547 if (!q->nb_channels)
548 return;
549
550 for (ch = 0; ch < q->nb_channels; ch++)
551 for (j = 0; j < 64; j++) {
552 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);
553 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);
554 }
555 }
556
557
558 /**
559 * Called while processing data from subpackets 11 and 12.
560 * Used after making changes to coding_method array.
561 *
562 * @param sb subband index
563 * @param channels number of channels
564 * @param coding_method q->coding_method[0][0][0]
565 */
566 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
567 {
568 int j,k;
569 int ch;
570 int run, case_val;
571 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
573 for (ch = 0; ch < channels; ch++) {
574 for (j = 0; j < 64; ) {
575 if((coding_method[ch][sb][j] - 8) > 22) {
576 run = 1;
577 case_val = 8;
578 } else {
579 switch (switchtable[coding_method[ch][sb][j]-8]) {
580 case 0: run = 10; case_val = 10; break;
581 case 1: run = 1; case_val = 16; break;
582 case 2: run = 5; case_val = 24; break;
583 case 3: run = 3; case_val = 30; break;
584 case 4: run = 1; case_val = 30; break;
585 case 5: run = 1; case_val = 8; break;
586 default: run = 1; case_val = 8; break;
587 }
588 }
589 for (k = 0; k < run; k++)
590 if (j + k < 128)
591 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
592 if (k > 0) {
593 SAMPLES_NEEDED
594 //not debugged, almost never used
595 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
596 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
597 }
598 j += run;
599 }
600 }
601 }
602
603
604 /**
605 * Related to synthesis filter
606 * Called by process_subpacket_10
607 *
608 * @param q context
609 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
610 */
611 static void fill_tone_level_array (QDM2Context *q, int flag)
612 {
613 int i, sb, ch, sb_used;
614 int tmp, tab;
615
616 // This should never happen
617 if (q->nb_channels <= 0)
618 return;
619
620 for (ch = 0; ch < q->nb_channels; ch++)
621 for (sb = 0; sb < 30; sb++)
622 for (i = 0; i < 8; i++) {
623 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
624 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
625 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
626 else
627 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
628 if(tmp < 0)
629 tmp += 0xff;
630 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
631 }
632
633 sb_used = QDM2_SB_USED(q->sub_sampling);
634
635 if ((q->superblocktype_2_3 != 0) && !flag) {
636 for (sb = 0; sb < sb_used; sb++)
637 for (ch = 0; ch < q->nb_channels; ch++)
638 for (i = 0; i < 64; i++) {
639 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
640 if (q->tone_level_idx[ch][sb][i] < 0)
641 q->tone_level[ch][sb][i] = 0;
642 else
643 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
644 }
645 } else {
646 tab = q->superblocktype_2_3 ? 0 : 1;
647 for (sb = 0; sb < sb_used; sb++) {
648 if ((sb >= 4) && (sb <= 23)) {
649 for (ch = 0; ch < q->nb_channels; ch++)
650 for (i = 0; i < 64; i++) {
651 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
652 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
653 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
654 q->tone_level_idx_hi2[ch][sb - 4];
655 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
656 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
657 q->tone_level[ch][sb][i] = 0;
658 else
659 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
660 }
661 } else {
662 if (sb > 4) {
663 for (ch = 0; ch < q->nb_channels; ch++)
664 for (i = 0; i < 64; i++) {
665 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
666 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
667 q->tone_level_idx_hi2[ch][sb - 4];
668 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
669 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
670 q->tone_level[ch][sb][i] = 0;
671 else
672 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
673 }
674 } else {
675 for (ch = 0; ch < q->nb_channels; ch++)
676 for (i = 0; i < 64; i++) {
677 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
678 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
679 q->tone_level[ch][sb][i] = 0;
680 else
681 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
682 }
683 }
684 }
685 }
686 }
687
688 return;
689 }
690
691
692 /**
693 * Related to synthesis filter
694 * Called by process_subpacket_11
695 * c is built with data from subpacket 11
696 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
697 *
698 * @param tone_level_idx
699 * @param tone_level_idx_temp
700 * @param coding_method q->coding_method[0][0][0]
701 * @param nb_channels number of channels
702 * @param c coming from subpacket 11, passed as 8*c
703 * @param superblocktype_2_3 flag based on superblock packet type
704 * @param cm_table_select q->cm_table_select
705 */
706 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
707 sb_int8_array coding_method, int nb_channels,
708 int c, int superblocktype_2_3, int cm_table_select)
709 {
710 int ch, sb, j;
711 int tmp, acc, esp_40, comp;
712 int add1, add2, add3, add4;
713 int64_t multres;
714
715 // This should never happen
716 if (nb_channels <= 0)
717 return;
718
719 if (!superblocktype_2_3) {
720 /* This case is untested, no samples available */
721 SAMPLES_NEEDED
722 for (ch = 0; ch < nb_channels; ch++)
723 for (sb = 0; sb < 30; sb++) {
724 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
725 add1 = tone_level_idx[ch][sb][j] - 10;
726 if (add1 < 0)
727 add1 = 0;
728 add2 = add3 = add4 = 0;
729 if (sb > 1) {
730 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
731 if (add2 < 0)
732 add2 = 0;
733 }
734 if (sb > 0) {
735 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
736 if (add3 < 0)
737 add3 = 0;
738 }
739 if (sb < 29) {
740 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
741 if (add4 < 0)
742 add4 = 0;
743 }
744 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
745 if (tmp < 0)
746 tmp = 0;
747 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
748 }
749 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
750 }
751 acc = 0;
752 for (ch = 0; ch < nb_channels; ch++)
753 for (sb = 0; sb < 30; sb++)
754 for (j = 0; j < 64; j++)
755 acc += tone_level_idx_temp[ch][sb][j];
756
757 multres = 0x66666667 * (acc * 10);
758 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
759 for (ch = 0; ch < nb_channels; ch++)
760 for (sb = 0; sb < 30; sb++)
761 for (j = 0; j < 64; j++) {
762 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
763 if (comp < 0)
764 comp += 0xff;
765 comp /= 256; // signed shift
766 switch(sb) {
767 case 0:
768 if (comp < 30)
769 comp = 30;
770 comp += 15;
771 break;
772 case 1:
773 if (comp < 24)
774 comp = 24;
775 comp += 10;
776 break;
777 case 2:
778 case 3:
779 case 4:
780 if (comp < 16)
781 comp = 16;
782 }
783 if (comp <= 5)
784 tmp = 0;
785 else if (comp <= 10)
786 tmp = 10;
787 else if (comp <= 16)
788 tmp = 16;
789 else if (comp <= 24)
790 tmp = -1;
791 else
792 tmp = 0;
793 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
794 }
795 for (sb = 0; sb < 30; sb++)
796 fix_coding_method_array(sb, nb_channels, coding_method);
797 for (ch = 0; ch < nb_channels; ch++)
798 for (sb = 0; sb < 30; sb++)
799 for (j = 0; j < 64; j++)
800 if (sb >= 10) {
801 if (coding_method[ch][sb][j] < 10)
802 coding_method[ch][sb][j] = 10;
803 } else {
804 if (sb >= 2) {
805 if (coding_method[ch][sb][j] < 16)
806 coding_method[ch][sb][j] = 16;
807 } else {
808 if (coding_method[ch][sb][j] < 30)
809 coding_method[ch][sb][j] = 30;
810 }
811 }
812 } else { // superblocktype_2_3 != 0
813 for (ch = 0; ch < nb_channels; ch++)
814 for (sb = 0; sb < 30; sb++)
815 for (j = 0; j < 64; j++)
816 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
817 }
818
819 return;
820 }
821
822
823 /**
824 *
825 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
826 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
827 *
828 * @param q context
829 * @param gb bitreader context
830 * @param length packet length in bits
831 * @param sb_min lower subband processed (sb_min included)
832 * @param sb_max higher subband processed (sb_max excluded)
833 */
834 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
835 {
836 int sb, j, k, n, ch, run, channels;
837 int joined_stereo, zero_encoding, chs;
838 int type34_first;
839 float type34_div = 0;
840 float type34_predictor;
841 float samples[10], sign_bits[16];
842
843 if (length == 0) {
844 // If no data use noise
845 for (sb=sb_min; sb < sb_max; sb++)
846 build_sb_samples_from_noise (q, sb);
847
848 return;
849 }
850
851 for (sb = sb_min; sb < sb_max; sb++) {
852 FIX_NOISE_IDX(q->noise_idx);
853
854 channels = q->nb_channels;
855
856 if (q->nb_channels <= 1 || sb < 12)
857 joined_stereo = 0;
858 else if (sb >= 24)
859 joined_stereo = 1;
860 else
861 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
862
863 if (joined_stereo) {
864 if (BITS_LEFT(length,gb) >= 16)
865 for (j = 0; j < 16; j++)
866 sign_bits[j] = get_bits1 (gb);
867
868 for (j = 0; j < 64; j++)
869 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
870 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
871
872 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
873 channels = 1;
874 }
875
876 for (ch = 0; ch < channels; ch++) {
877 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
878 type34_predictor = 0.0;
879 type34_first = 1;
880
881 for (j = 0; j < 128; ) {
882 switch (q->coding_method[ch][sb][j / 2]) {
883 case 8:
884 if (BITS_LEFT(length,gb) >= 10) {
885 if (zero_encoding) {
886 for (k = 0; k < 5; k++) {
887 if ((j + 2 * k) >= 128)
888 break;
889 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
890 }
891 } else {
892 n = get_bits(gb, 8);
893 for (k = 0; k < 5; k++)
894 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
895 }
896 for (k = 0; k < 5; k++)
897 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
898 } else {
899 for (k = 0; k < 10; k++)
900 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
901 }
902 run = 10;
903 break;
904
905 case 10:
906 if (BITS_LEFT(length,gb) >= 1) {
907 float f = 0.81;
908
909 if (get_bits1(gb))
910 f = -f;
911 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
912 samples[0] = f;
913 } else {
914 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
915 }
916 run = 1;
917 break;
918
919 case 16:
920 if (BITS_LEFT(length,gb) >= 10) {
921 if (zero_encoding) {
922 for (k = 0; k < 5; k++) {
923 if ((j + k) >= 128)
924 break;
925 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
926 }
927 } else {
928 n = get_bits (gb, 8);
929 for (k = 0; k < 5; k++)
930 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
931 }
932 } else {
933 for (k = 0; k < 5; k++)
934 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
935 }
936 run = 5;
937 break;
938
939 case 24:
940 if (BITS_LEFT(length,gb) >= 7) {
941 n = get_bits(gb, 7);
942 for (k = 0; k < 3; k++)
943 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
944 } else {
945 for (k = 0; k < 3; k++)
946 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
947 }
948 run = 3;
949 break;
950
951 case 30:
952 if (BITS_LEFT(length,gb) >= 4)
953 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
954 else
955 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
956
957 run = 1;
958 break;
959
960 case 34:
961 if (BITS_LEFT(length,gb) >= 7) {
962 if (type34_first) {
963 type34_div = (float)(1 << get_bits(gb, 2));
964 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
965 type34_predictor = samples[0];
966 type34_first = 0;
967 } else {
968 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
969 type34_predictor = samples[0];
970 }
971 } else {
972 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
973 }
974 run = 1;
975 break;
976
977 default:
978 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
979 run = 1;
980 break;
981 }
982
983 if (joined_stereo) {
984 float tmp[10][MPA_MAX_CHANNELS];
985
986 for (k = 0; k < run; k++) {
987 tmp[k][0] = samples[k];
988 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
989 }
990 for (chs = 0; chs < q->nb_channels; chs++)
991 for (k = 0; k < run; k++)
992 if ((j + k) < 128)
993 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
994 } else {
995 for (k = 0; k < run; k++)
996 if ((j + k) < 128)
997 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
998 }
999
1000 j += run;
1001 } // j loop
1002 } // channel loop
1003 } // subband loop
1004 }
1005
1006
1007 /**
1008 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
1009 * This is similar to process_subpacket_9, but for a single channel and for element [0]
1010 * same VLC tables as process_subpacket_9 are used.
1011 *
1012 * @param q context
1013 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
1014 * @param gb bitreader context
1015 * @param length packet length in bits
1016 */
1017 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
1018 {
1019 int i, k, run, level, diff;
1020
1021 if (BITS_LEFT(length,gb) < 16)
1022 return;
1023 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
1024
1025 quantized_coeffs[0] = level;
1026
1027 for (i = 0; i < 7; ) {
1028 if (BITS_LEFT(length,gb) < 16)
1029 break;
1030 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
1031
1032 if (BITS_LEFT(length,gb) < 16)
1033 break;
1034 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
1035
1036 for (k = 1; k <= run; k++)
1037 quantized_coeffs[i + k] = (level + ((k * diff) / run));
1038
1039 level += diff;
1040 i += run;
1041 }
1042 }
1043
1044
1045 /**
1046 * Related to synthesis filter, process data from packet 10
1047 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1048 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1049 *
1050 * @param q context
1051 * @param gb bitreader context
1052 * @param length packet length in bits
1053 */
1054 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1055 {
1056 int sb, j, k, n, ch;
1057
1058 for (ch = 0; ch < q->nb_channels; ch++) {
1059 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1060
1061 if (BITS_LEFT(length,gb) < 16) {
1062 memset(q->quantized_coeffs[ch][0], 0, 8);
1063 break;
1064 }
1065 }
1066
1067 n = q->sub_sampling + 1;
1068
1069 for (sb = 0; sb < n; sb++)
1070 for (ch = 0; ch < q->nb_channels; ch++)
1071 for (j = 0; j < 8; j++) {
1072 if (BITS_LEFT(length,gb) < 1)
1073 break;
1074 if (get_bits1(gb)) {
1075 for (k=0; k < 8; k++) {
1076 if (BITS_LEFT(length,gb) < 16)
1077 break;
1078 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1079 }
1080 } else {
1081 for (k=0; k < 8; k++)
1082 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1083 }
1084 }
1085
1086 n = QDM2_SB_USED(q->sub_sampling) - 4;
1087
1088 for (sb = 0; sb < n; sb++)
1089 for (ch = 0; ch < q->nb_channels; ch++) {
1090 if (BITS_LEFT(length,gb) < 16)
1091 break;
1092 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1093 if (sb > 19)
1094 q->tone_level_idx_hi2[ch][sb] -= 16;
1095 else
1096 for (j = 0; j < 8; j++)
1097 q->tone_level_idx_mid[ch][sb][j] = -16;
1098 }
1099
1100 n = QDM2_SB_USED(q->sub_sampling) - 5;
1101
1102 for (sb = 0; sb < n; sb++)
1103 for (ch = 0; ch < q->nb_channels; ch++)
1104 for (j = 0; j < 8; j++) {
1105 if (BITS_LEFT(length,gb) < 16)
1106 break;
1107 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1108 }
1109 }
1110
1111 /**
1112 * Process subpacket 9, init quantized_coeffs with data from it
1113 *
1114 * @param q context
1115 * @param node pointer to node with packet
1116 */
1117 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1118 {
1119 GetBitContext gb;
1120 int i, j, k, n, ch, run, level, diff;
1121
1122 init_get_bits(&gb, node->packet->data, node->packet->size*8);
1123
1124 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1125
1126 for (i = 1; i < n; i++)
1127 for (ch=0; ch < q->nb_channels; ch++) {
1128 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1129 q->quantized_coeffs[ch][i][0] = level;
1130
1131 for (j = 0; j < (8 - 1); ) {
1132 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1133 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1134
1135 for (k = 1; k <= run; k++)
1136 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1137
1138 level += diff;
1139 j += run;
1140 }
1141 }
1142
1143 for (ch = 0; ch < q->nb_channels; ch++)
1144 for (i = 0; i < 8; i++)
1145 q->quantized_coeffs[ch][0][i] = 0;
1146 }
1147
1148
1149 /**
1150 * Process subpacket 10 if not null, else
1151 *
1152 * @param q context
1153 * @param node pointer to node with packet
1154 * @param length packet length in bits
1155 */
1156 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1157 {
1158 GetBitContext gb;
1159
1160 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1161
1162 if (length != 0) {
1163 init_tone_level_dequantization(q, &gb, length);
1164 fill_tone_level_array(q, 1);
1165 } else {
1166 fill_tone_level_array(q, 0);
1167 }
1168 }
1169
1170
1171 /**
1172 * Process subpacket 11
1173 *
1174 * @param q context
1175 * @param node pointer to node with packet
1176 * @param length packet length in bit
1177 */
1178 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1179 {
1180 GetBitContext gb;
1181
1182 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1183 if (length >= 32) {
1184 int c = get_bits (&gb, 13);
1185
1186 if (c > 3)
1187 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1188 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1189 }
1190
1191 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1192 }
1193
1194
1195 /**
1196 * Process subpacket 12
1197 *
1198 * @param q context
1199 * @param node pointer to node with packet
1200 * @param length packet length in bits
1201 */
1202 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1203 {
1204 GetBitContext gb;
1205
1206 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1207 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1208 }
1209
1210 /*
1211 * Process new subpackets for synthesis filter
1212 *
1213 * @param q context
1214 * @param list list with synthesis filter packets (list D)
1215 */
1216 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1217 {
1218 QDM2SubPNode *nodes[4];
1219
1220 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1221 if (nodes[0] != NULL)
1222 process_subpacket_9(q, nodes[0]);
1223
1224 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1225 if (nodes[1] != NULL)
1226 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1227 else
1228 process_subpacket_10(q, NULL, 0);
1229
1230 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1231 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1232 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1233 else
1234 process_subpacket_11(q, NULL, 0);
1235
1236 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1237 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1238 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1239 else
1240 process_subpacket_12(q, NULL, 0);
1241 }
1242
1243
1244 /*
1245 * Decode superblock, fill packet lists.
1246 *
1247 * @param q context
1248 */
1249 static void qdm2_decode_super_block (QDM2Context *q)
1250 {
1251 GetBitContext gb;
1252 QDM2SubPacket header, *packet;
1253 int i, packet_bytes, sub_packet_size, sub_packets_D;
1254 unsigned int next_index = 0;
1255
1256 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1257 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1258 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1259
1260 q->sub_packets_B = 0;
1261 sub_packets_D = 0;
1262
1263 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1264
1265 init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1266 qdm2_decode_sub_packet_header(&gb, &header);
1267
1268 if (header.type < 2 || header.type >= 8) {
1269 q->has_errors = 1;
1270 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1271 return;
1272 }
1273
1274 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1275 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1276
1277 init_get_bits(&gb, header.data, header.size*8);
1278
1279 if (header.type == 2 || header.type == 4 || header.type == 5) {
1280 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1281
1282 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1283
1284 if (csum != 0) {
1285 q->has_errors = 1;
1286 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1287 return;
1288 }
1289 }
1290
1291 q->sub_packet_list_B[0].packet = NULL;
1292 q->sub_packet_list_D[0].packet = NULL;
1293
1294 for (i = 0; i < 6; i++)
1295 if (--q->fft_level_exp[i] < 0)
1296 q->fft_level_exp[i] = 0;
1297
1298 for (i = 0; packet_bytes > 0; i++) {
1299 int j;
1300
1301 q->sub_packet_list_A[i].next = NULL;
1302
1303 if (i > 0) {
1304 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1305
1306 /* seek to next block */
1307 init_get_bits(&gb, header.data, header.size*8);
1308 skip_bits(&gb, next_index*8);
1309
1310 if (next_index >= header.size)
1311 break;
1312 }
1313
1314 /* decode subpacket */
1315 packet = &q->sub_packets[i];
1316 qdm2_decode_sub_packet_header(&gb, packet);
1317 next_index = packet->size + get_bits_count(&gb) / 8;
1318 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1319
1320 if (packet->type == 0)
1321 break;
1322
1323 if (sub_packet_size > packet_bytes) {
1324 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1325 break;
1326 packet->size += packet_bytes - sub_packet_size;
1327 }
1328
1329 packet_bytes -= sub_packet_size;
1330
1331 /* add subpacket to 'all subpackets' list */
1332 q->sub_packet_list_A[i].packet = packet;
1333
1334 /* add subpacket to related list */
1335 if (packet->type == 8) {
1336 SAMPLES_NEEDED_2("packet type 8");
1337 return;
1338 } else if (packet->type >= 9 && packet->type <= 12) {
1339 /* packets for MPEG Audio like Synthesis Filter */
1340 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1341 } else if (packet->type == 13) {
1342 for (j = 0; j < 6; j++)
1343 q->fft_level_exp[j] = get_bits(&gb, 6);
1344 } else if (packet->type == 14) {
1345 for (j = 0; j < 6; j++)
1346 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1347 } else if (packet->type == 15) {
1348 SAMPLES_NEEDED_2("packet type 15")
1349 return;
1350 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1351 /* packets for FFT */
1352 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1353 }
1354 } // Packet bytes loop
1355
1356 /* **************************************************************** */
1357 if (q->sub_packet_list_D[0].packet != NULL) {
1358 process_synthesis_subpackets(q, q->sub_packet_list_D);
1359 q->do_synth_filter = 1;
1360 } else if (q->do_synth_filter) {
1361 process_subpacket_10(q, NULL, 0);
1362 process_subpacket_11(q, NULL, 0);
1363 process_subpacket_12(q, NULL, 0);
1364 }
1365 /* **************************************************************** */
1366 }
1367
1368
1369 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1370 int offset, int duration, int channel,
1371 int exp, int phase)
1372 {
1373 if (q->fft_coefs_min_index[duration] < 0)
1374 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1375
1376 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1377 q->fft_coefs[q->fft_coefs_index].channel = channel;
1378 q->fft_coefs[q->fft_coefs_index].offset = offset;
1379 q->fft_coefs[q->fft_coefs_index].exp = exp;
1380 q->fft_coefs[q->fft_coefs_index].phase = phase;
1381 q->fft_coefs_index++;
1382 }
1383
1384
1385 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1386 {
1387 int channel, stereo, phase, exp;
1388 int local_int_4, local_int_8, stereo_phase, local_int_10;
1389 int local_int_14, stereo_exp, local_int_20, local_int_28;
1390 int n, offset;
1391
1392 local_int_4 = 0;
1393 local_int_28 = 0;
1394 local_int_20 = 2;
1395 local_int_8 = (4 - duration);
1396 local_int_10 = 1 << (q->group_order - duration - 1);
1397 offset = 1;
1398
1399 while (1) {
1400 if (q->superblocktype_2_3) {
1401 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1402 offset = 1;
1403 if (n == 0) {
1404 local_int_4 += local_int_10;
1405 local_int_28 += (1 << local_int_8);
1406 } else {
1407 local_int_4 += 8*local_int_10;
1408 local_int_28 += (8 << local_int_8);
1409 }
1410 }
1411 offset += (n - 2);
1412 } else {
1413 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1414 while (offset >= (local_int_10 - 1)) {
1415 offset += (1 - (local_int_10 - 1));
1416 local_int_4 += local_int_10;
1417 local_int_28 += (1 << local_int_8);
1418 }
1419 }
1420
1421 if (local_int_4 >= q->group_size)
1422 return;
1423
1424 local_int_14 = (offset >> local_int_8);
1425
1426 if (q->nb_channels > 1) {
1427 channel = get_bits1(gb);
1428 stereo = get_bits1(gb);
1429 } else {
1430 channel = 0;
1431 stereo = 0;
1432 }
1433
1434 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1435 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1436 exp = (exp < 0) ? 0 : exp;
1437
1438 phase = get_bits(gb, 3);
1439 stereo_exp = 0;
1440 stereo_phase = 0;
1441
1442 if (stereo) {
1443 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1444 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1445 if (stereo_phase < 0)
1446 stereo_phase += 8;
1447 }
1448
1449 if (q->frequency_range > (local_int_14 + 1)) {
1450 int sub_packet = (local_int_20 + local_int_28);
1451
1452 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1453 if (stereo)
1454 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1455 }
1456
1457 offset++;
1458 }
1459 }
1460
1461
1462 static void qdm2_decode_fft_packets (QDM2Context *q)
1463 {
1464 int i, j, min, max, value, type, unknown_flag;
1465 GetBitContext gb;
1466
1467 if (q->sub_packet_list_B[0].packet == NULL)
1468 return;
1469
1470 /* reset minimum indexes for FFT coefficients */
1471 q->fft_coefs_index = 0;
1472 for (i=0; i < 5; i++)
1473 q->fft_coefs_min_index[i] = -1;
1474
1475 /* process subpackets ordered by type, largest type first */
1476 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1477 QDM2SubPacket *packet= NULL;
1478
1479 /* find subpacket with largest type less than max */
1480 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1481 value = q->sub_packet_list_B[j].packet->type;
1482 if (value > min && value < max) {
1483 min = value;
1484 packet = q->sub_packet_list_B[j].packet;
1485 }
1486 }
1487
1488 max = min;
1489
1490 /* check for errors (?) */
1491 if (!packet)
1492 return;
1493
1494 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1495 return;
1496
1497 /* decode FFT tones */
1498 init_get_bits (&gb, packet->data, packet->size*8);
1499
1500 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1501 unknown_flag = 1;
1502 else
1503 unknown_flag = 0;
1504
1505 type = packet->type;
1506
1507 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1508 int duration = q->sub_sampling + 5 - (type & 15);
1509
1510 if (duration >= 0 && duration < 4)
1511 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1512 } else if (type == 31) {
1513 for (j=0; j < 4; j++)
1514 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1515 } else if (type == 46) {
1516 for (j=0; j < 6; j++)
1517 q->fft_level_exp[j] = get_bits(&gb, 6);
1518 for (j=0; j < 4; j++)
1519 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1520 }
1521 } // Loop on B packets
1522
1523 /* calculate maximum indexes for FFT coefficients */
1524 for (i = 0, j = -1; i < 5; i++)
1525 if (q->fft_coefs_min_index[i] >= 0) {
1526 if (j >= 0)
1527 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1528 j = i;
1529 }
1530 if (j >= 0)
1531 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1532 }
1533
1534
1535 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1536 {
1537 float level, f[6];
1538 int i;
1539 QDM2Complex c;
1540 const double iscale = 2.0*M_PI / 512.0;
1541
1542 tone->phase += tone->phase_shift;
1543
1544 /* calculate current level (maximum amplitude) of tone */
1545 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1546 c.im = level * sin(tone->phase*iscale);
1547 c.re = level * cos(tone->phase*iscale);
1548
1549 /* generate FFT coefficients for tone */
1550 if (tone->duration >= 3 || tone->cutoff >= 3) {
1551 tone->complex[0].im += c.im;
1552 tone->complex[0].re += c.re;
1553 tone->complex[1].im -= c.im;
1554 tone->complex[1].re -= c.re;
1555 } else {
1556 f[1] = -tone->table[4];
1557 f[0] = tone->table[3] - tone->table[0];
1558 f[2] = 1.0 - tone->table[2] - tone->table[3];
1559 f[3] = tone->table[1] + tone->table[4] - 1.0;
1560 f[4] = tone->table[0] - tone->table[1];
1561 f[5] = tone->table[2];
1562 for (i = 0; i < 2; i++) {
1563 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1564 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1565 }
1566 for (i = 0; i < 4; i++) {
1567 tone->complex[i].re += c.re * f[i+2];
1568 tone->complex[i].im += c.im * f[i+2];
1569 }
1570 }
1571
1572 /* copy the tone if it has not yet died out */
1573 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1574 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1575 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1576 }
1577 }
1578
1579
1580 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1581 {
1582 int i, j, ch;
1583 const double iscale = 0.25 * M_PI;
1584
1585 for (ch = 0; ch < q->channels; ch++) {
1586 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1587 }
1588
1589
1590 /* apply FFT tones with duration 4 (1 FFT period) */
1591 if (q->fft_coefs_min_index[4] >= 0)
1592 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1593 float level;
1594 QDM2Complex c;
1595
1596 if (q->fft_coefs[i].sub_packet != sub_packet)
1597 break;
1598
1599 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1600 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
1602 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1603 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1604 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1605 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1606 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1607 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1608 }
1609
1610 /* generate existing FFT tones */
1611 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1612 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1613 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1614 }
1615
1616 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1617 for (i = 0; i < 4; i++)
1618 if (q->fft_coefs_min_index[i] >= 0) {
1619 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1620 int offset, four_i;
1621 FFTTone tone;
1622
1623 if (q->fft_coefs[j].sub_packet != sub_packet)
1624 break;
1625
1626 four_i = (4 - i);
1627 offset = q->fft_coefs[j].offset >> four_i;
1628 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1629
1630 if (offset < q->frequency_range) {
1631 if (offset < 2)
1632 tone.cutoff = offset;
1633 else
1634 tone.cutoff = (offset >= 60) ? 3 : 2;
1635
1636 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];
1637 tone.complex = &q->fft.complex[ch][offset];
1638 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1639 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1640 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1641 tone.duration = i;
1642 tone.time_index = 0;
1643
1644 qdm2_fft_generate_tone(q, &tone);
1645 }
1646 }
1647 q->fft_coefs_min_index[i] = j;
1648 }
1649 }
1650
1651
1652 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1653 {
1654 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1655 int i;
1656 q->fft.complex[channel][0].re *= 2.0f;
1657 q->fft.complex[channel][0].im = 0.0f;
1658 ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1659 /* add samples to output buffer */
1660 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1661 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
1662 }
1663
1664
1665 /**
1666 * @param q context
1667 * @param index subpacket number
1668 */
1669 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1670 {
1671 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
1672 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1673
1674 /* copy sb_samples */
1675 sb_used = QDM2_SB_USED(q->sub_sampling);
1676
1677 for (ch = 0; ch < q->channels; ch++)
1678 for (i = 0; i < 8; i++)
1679 for (k=sb_used; k < SBLIMIT; k++)
1680 q->sb_samples[ch][(8 * index) + i][k] = 0;
1681
1682 for (ch = 0; ch < q->nb_channels; ch++) {
1683 OUT_INT *samples_ptr = samples + ch;
1684
1685 for (i = 0; i < 8; i++) {
1686 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1687 mpa_window, &dither_state,
1688 samples_ptr, q->nb_channels,
1689 q->sb_samples[ch][(8 * index) + i]);
1690 samples_ptr += 32 * q->nb_channels;
1691 }
1692 }
1693
1694 /* add samples to output buffer */
1695 sub_sampling = (4 >> q->sub_sampling);
1696
1697 for (ch = 0; ch < q->channels; ch++)
1698 for (i = 0; i < q->frame_size; i++)
1699 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
1700 }
1701
1702
1703 /**
1704 * Init static data (does not depend on specific file)
1705 *
1706 * @param q context
1707 */
1708 static av_cold void qdm2_init(QDM2Context *q) {
1709 static int initialized = 0;
1710
1711 if (initialized != 0)
1712 return;
1713 initialized = 1;
1714
1715 qdm2_init_vlc();
1716 ff_mpa_synth_init(mpa_window);
1717 softclip_table_init();
1718 rnd_table_init();
1719 init_noise_samples();
1720
1721 av_log(NULL, AV_LOG_DEBUG, "init done\n");
1722 }
1723
1724
1725 #if 0
1726 static void dump_context(QDM2Context *q)
1727 {
1728 int i;
1729 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1730 PRINT("compressed_data",q->compressed_data);
1731 PRINT("compressed_size",q->compressed_size);
1732 PRINT("frame_size",q->frame_size);
1733 PRINT("checksum_size",q->checksum_size);
1734 PRINT("channels",q->channels);
1735 PRINT("nb_channels",q->nb_channels);
1736 PRINT("fft_frame_size",q->fft_frame_size);
1737 PRINT("fft_size",q->fft_size);
1738 PRINT("sub_sampling",q->sub_sampling);
1739 PRINT("fft_order",q->fft_order);
1740 PRINT("group_order",q->group_order);
1741 PRINT("group_size",q->group_size);
1742 PRINT("sub_packet",q->sub_packet);
1743 PRINT("frequency_range",q->frequency_range);
1744 PRINT("has_errors",q->has_errors);
1745 PRINT("fft_tone_end",q->fft_tone_end);
1746 PRINT("fft_tone_start",q->fft_tone_start);
1747 PRINT("fft_coefs_index",q->fft_coefs_index);
1748 PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1749 PRINT("cm_table_select",q->cm_table_select);
1750 PRINT("noise_idx",q->noise_idx);
1751
1752 for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1753 {
1754 FFTTone *t = &q->fft_tones[i];
1755
1756 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1757 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1758 // PRINT(" level", t->level);
1759 PRINT(" phase", t->phase);
1760 PRINT(" phase_shift", t->phase_shift);
1761 PRINT(" duration", t->duration);
1762 PRINT(" samples_im", t->samples_im);
1763 PRINT(" samples_re", t->samples_re);
1764 PRINT(" table", t->table);
1765 }
1766
1767 }
1768 #endif
1769
1770
1771 /**
1772 * Init parameters from codec extradata
1773 */
1774 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1775 {
1776 QDM2Context *s = avctx->priv_data;
1777 uint8_t *extradata;
1778 int extradata_size;
1779 int tmp_val, tmp, size;
1780
1781 /* extradata parsing
1782
1783 Structure:
1784 wave {
1785 frma (QDM2)
1786 QDCA
1787 QDCP
1788 }
1789
1790 32 size (including this field)
1791 32 tag (=frma)
1792 32 type (=QDM2 or QDMC)
1793
1794 32 size (including this field, in bytes)
1795 32 tag (=QDCA) // maybe mandatory parameters
1796 32 unknown (=1)
1797 32 channels (=2)
1798 32 samplerate (=44100)
1799 32 bitrate (=96000)
1800 32 block size (=4096)
1801 32 frame size (=256) (for one channel)
1802 32 packet size (=1300)
1803
1804 32 size (including this field, in bytes)
1805 32 tag (=QDCP) // maybe some tuneable parameters
1806 32 float1 (=1.0)
1807 32 zero ?
1808 32 float2 (=1.0)
1809 32 float3 (=1.0)
1810 32 unknown (27)
1811 32 unknown (8)
1812 32 zero ?
1813 */
1814
1815 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1816 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1817 return -1;
1818 }
1819
1820 extradata = avctx->extradata;
1821 extradata_size = avctx->extradata_size;
1822
1823 while (extradata_size > 7) {
1824 if (!memcmp(extradata, "frmaQDM", 7))
1825 break;
1826 extradata++;
1827 extradata_size--;
1828 }
1829
1830 if (extradata_size < 12) {
1831 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1832 extradata_size);
1833 return -1;
1834 }
1835
1836 if (memcmp(extradata, "frmaQDM", 7)) {
1837 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1838 return -1;
1839 }
1840
1841 if (extradata[7] == 'C') {
1842 // s->is_qdmc = 1;
1843 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1844 return -1;
1845 }
1846
1847 extradata += 8;
1848 extradata_size -= 8;
1849
1850 size = AV_RB32(extradata);
1851
1852 if(size > extradata_size){
1853 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1854 extradata_size, size);
1855 return -1;
1856 }
1857
1858 extradata += 4;
1859 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1860 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1861 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1862 return -1;
1863 }
1864
1865 extradata += 8;
1866
1867 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1868 extradata += 4;
1869
1870 avctx->sample_rate = AV_RB32(extradata);
1871 extradata += 4;
1872
1873 avctx->bit_rate = AV_RB32(extradata);
1874 extradata += 4;
1875
1876 s->group_size = AV_RB32(extradata);
1877 extradata += 4;
1878
1879 s->fft_size = AV_RB32(extradata);
1880 extradata += 4;
1881
1882 s->checksum_size = AV_RB32(extradata);
1883
1884 s->fft_order = av_log2(s->fft_size) + 1;
1885 s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1886
1887 // something like max decodable tones
1888 s->group_order = av_log2(s->group_size) + 1;
1889 s->frame_size = s->group_size / 16; // 16 iterations per super block
1890
1891 s->sub_sampling = s->fft_order - 7;
1892 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1893
1894 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1895 case 0: tmp = 40; break;
1896 case 1: tmp = 48; break;
1897 case 2: tmp = 56; break;
1898 case 3: tmp = 72; break;
1899 case 4: tmp = 80; break;
1900 case 5: tmp = 100;break;
1901 default: tmp=s->sub_sampling; break;
1902 }
1903 tmp_val = 0;
1904 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1905 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1906 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1907 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1908 s->cm_table_select = tmp_val;
1909
1910 if (s->sub_sampling == 0)
1911 tmp = 7999;
1912 else
1913 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1914 /*
1915 0: 7999 -> 0
1916 1: 20000 -> 2
1917 2: 28000 -> 2
1918 */
1919 if (tmp < 8000)
1920 s->coeff_per_sb_select = 0;
1921 else if (tmp <= 16000)
1922 s->coeff_per_sb_select = 1;
1923 else
1924 s->coeff_per_sb_select = 2;
1925
1926 // Fail on unknown fft order
1927 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1928 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1929 return -1;
1930 }
1931
1932 ff_rdft_init(&s->rdft_ctx, s->fft_order, IRDFT);
1933
1934 qdm2_init(s);
1935
1936 avctx->sample_fmt = SAMPLE_FMT_S16;
1937
1938 // dump_context(s);
1939 return 0;
1940 }
1941
1942
1943 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1944 {
1945 QDM2Context *s = avctx->priv_data;
1946
1947 ff_rdft_end(&s->rdft_ctx);
1948
1949 return 0;
1950 }
1951
1952
1953 static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1954 {
1955 int ch, i;
1956 const int frame_size = (q->frame_size * q->channels);
1957
1958 /* select input buffer */
1959 q->compressed_data = in;
1960 q->compressed_size = q->checksum_size;
1961
1962 // dump_context(q);
1963
1964 /* copy old block, clear new block of output samples */
1965 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1966 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1967
1968 /* decode block of QDM2 compressed data */
1969 if (q->sub_packet == 0) {
1970 q->has_errors = 0; // zero it for a new super block
1971 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1972 qdm2_decode_super_block(q);
1973 }
1974
1975 /* parse subpackets */
1976 if (!q->has_errors) {
1977 if (q->sub_packet == 2)
1978 qdm2_decode_fft_packets(q);
1979
1980 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1981 }
1982
1983 /* sound synthesis stage 1 (FFT) */
1984 for (ch = 0; ch < q->channels; ch++) {
1985 qdm2_calculate_fft(q, ch, q->sub_packet);
1986
1987 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1988 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1989 return;
1990 }
1991 }
1992
1993 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1994 if (!q->has_errors && q->do_synth_filter)
1995 qdm2_synthesis_filter(q, q->sub_packet);
1996
1997 q->sub_packet = (q->sub_packet + 1) % 16;
1998
1999 /* clip and convert output float[] to 16bit signed samples */
2000 for (i = 0; i < frame_size; i++) {
2001 int value = (int)q->output_buffer[i];
2002
2003 if (value > SOFTCLIP_THRESHOLD)
2004 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
2005 else if (value < -SOFTCLIP_THRESHOLD)
2006 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
2007
2008 out[i] = value;
2009 }
2010 }
2011
2012
2013 static int qdm2_decode_frame(AVCodecContext *avctx,
2014 void *data, int *data_size,
2015 AVPacket *avpkt)
2016 {
2017 const uint8_t *buf = avpkt->data;
2018 int buf_size = avpkt->size;
2019 QDM2Context *s = avctx->priv_data;
2020
2021 if(!buf)
2022 return 0;
2023 if(buf_size < s->checksum_size)
2024 return -1;
2025
2026 *data_size = s->channels * s->frame_size * sizeof(int16_t);
2027
2028 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
2029 buf_size, buf, s->checksum_size, data, *data_size);
2030
2031 qdm2_decode(s, buf, data);
2032
2033 // reading only when next superblock found
2034 if (s->sub_packet == 0) {
2035 return s->checksum_size;
2036 }
2037
2038 return 0;
2039 }
2040
2041 AVCodec qdm2_decoder =
2042 {
2043 .name = "qdm2",
2044 .type = CODEC_TYPE_AUDIO,
2045 .id = CODEC_ID_QDM2,
2046 .priv_data_size = sizeof(QDM2Context),
2047 .init = qdm2_decode_init,
2048 .close = qdm2_decode_close,
2049 .decode = qdm2_decode_frame,
2050 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
2051 };
ffmpeg with support for matroska ordered chapters