?  search:
summary | log | graphiclog1 | graphiclog2 | commit | commitdiff | tree | refs | edit | fork
blame | history | raw | HEAD
Add missing #includes for avutil.h, required for the AV_VERSION* macros.
[ffmpeg-lucabe.git] / libavcodec / qdm2.c
blob4dc76126da5ef682af2b322ff40081944c925dc9
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
223 static av_cold void softclip_table_init(void) {
224 int i;
225 double dfl = SOFTCLIP_THRESHOLD - 32767;
226 float delta = 1.0 / -dfl;
227 for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
228 softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
229 }
230
231
232 // random generated table
233 static av_cold void rnd_table_init(void) {
234 int i,j;
235 uint32_t ldw,hdw;
236 uint64_t tmp64_1;
237 uint64_t random_seed = 0;
238 float delta = 1.0 / 16384.0;
239 for(i = 0; i < 4096 ;i++) {
240 random_seed = random_seed * 214013 + 2531011;
241 noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
242 }
243
244 for (i = 0; i < 256 ;i++) {
245 random_seed = 81;
246 ldw = i;
247 for (j = 0; j < 5 ;j++) {
248 random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
249 ldw = (uint32_t)ldw % (uint32_t)random_seed;
250 tmp64_1 = (random_seed * 0x55555556);
251 hdw = (uint32_t)(tmp64_1 >> 32);
252 random_seed = (uint64_t)(hdw + (ldw >> 31));
253 }
254 }
255 for (i = 0; i < 128 ;i++) {
256 random_seed = 25;
257 ldw = i;
258 for (j = 0; j < 3 ;j++) {
259 random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
260 ldw = (uint32_t)ldw % (uint32_t)random_seed;
261 tmp64_1 = (random_seed * 0x66666667);
262 hdw = (uint32_t)(tmp64_1 >> 33);
263 random_seed = hdw + (ldw >> 31);
264 }
265 }
266 }
267
268
269 static av_cold void init_noise_samples(void) {
270 int i;
271 int random_seed = 0;
272 float delta = 1.0 / 16384.0;
273 for (i = 0; i < 128;i++) {
274 random_seed = random_seed * 214013 + 2531011;
275 noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
276 }
277 }
278
279 static const uint16_t qdm2_vlc_offs[] = {
280 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
281 };
282
283 static av_cold void qdm2_init_vlc(void)
284 {
285 static int vlcs_initialized = 0;
286 static VLC_TYPE qdm2_table[3838][2];
287
288 if (!vlcs_initialized) {
289
290 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
291 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
292 init_vlc (&vlc_tab_level, 8, 24,
293 vlc_tab_level_huffbits, 1, 1,
294 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
295
296 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
297 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
298 init_vlc (&vlc_tab_diff, 8, 37,
299 vlc_tab_diff_huffbits, 1, 1,
300 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
301
302 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
303 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
304 init_vlc (&vlc_tab_run, 5, 6,
305 vlc_tab_run_huffbits, 1, 1,
306 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
307
308 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
309 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
310 init_vlc (&fft_level_exp_alt_vlc, 8, 28,
311 fft_level_exp_alt_huffbits, 1, 1,
312 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
313
314
315 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
316 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
317 init_vlc (&fft_level_exp_vlc, 8, 20,
318 fft_level_exp_huffbits, 1, 1,
319 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
320
321 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
322 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
323 init_vlc (&fft_stereo_exp_vlc, 6, 7,
324 fft_stereo_exp_huffbits, 1, 1,
325 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
326
327 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
328 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
329 init_vlc (&fft_stereo_phase_vlc, 6, 9,
330 fft_stereo_phase_huffbits, 1, 1,
331 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
332
333 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
334 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
335 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
336 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
337 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
338
339 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
340 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
341 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
342 vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
343 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
344
345 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
346 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
347 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
348 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
349 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
350
351 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
352 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
353 init_vlc (&vlc_tab_type30, 6, 9,
354 vlc_tab_type30_huffbits, 1, 1,
355 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
356
357 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
358 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
359 init_vlc (&vlc_tab_type34, 5, 10,
360 vlc_tab_type34_huffbits, 1, 1,
361 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
362
363 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
364 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
365 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
366 vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
367 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
368
369 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
370 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
371 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
372 vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
373 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
374
375 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
376 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
377 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
378 vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
379 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
380
381 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
382 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
383 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
384 vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
385 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
386
387 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
388 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
389 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
390 vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
391 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
392
393 vlcs_initialized=1;
394 }
395 }
396
397
398 /* for floating point to fixed point conversion */
399 static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
400
401
402 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
403 {
404 int value;
405
406 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
407
408 /* stage-2, 3 bits exponent escape sequence */
409 if (value-- == 0)
410 value = get_bits (gb, get_bits (gb, 3) + 1);
411
412 /* stage-3, optional */
413 if (flag) {
414 int tmp = vlc_stage3_values[value];
415
416 if ((value & ~3) > 0)
417 tmp += get_bits (gb, (value >> 2));
418 value = tmp;
419 }
420
421 return value;
422 }
423
424
425 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
426 {
427 int value = qdm2_get_vlc (gb, vlc, 0, depth);
428
429 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
430 }
431
432
433 /**
434 * QDM2 checksum
435 *
436 * @param data pointer to data to be checksum'ed
437 * @param length data length
438 * @param value checksum value
439 *
440 * @return 0 if checksum is OK
441 */
442 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
443 int i;
444
445 for (i=0; i < length; i++)
446 value -= data[i];
447
448 return (uint16_t)(value & 0xffff);
449 }
450
451
452 /**
453 * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
454 *
455 * @param gb bitreader context
456 * @param sub_packet packet under analysis
457 */
458 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
459 {
460 sub_packet->type = get_bits (gb, 8);
461
462 if (sub_packet->type == 0) {
463 sub_packet->size = 0;
464 sub_packet->data = NULL;
465 } else {
466 sub_packet->size = get_bits (gb, 8);
467
468 if (sub_packet->type & 0x80) {
469 sub_packet->size <<= 8;
470 sub_packet->size |= get_bits (gb, 8);
471 sub_packet->type &= 0x7f;
472 }
473
474 if (sub_packet->type == 0x7f)
475 sub_packet->type |= (get_bits (gb, 8) << 8);
476
477 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
478 }
479
480 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
481 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
482 }
483
484
485 /**
486 * Return node pointer to first packet of requested type in list.
487 *
488 * @param list list of subpackets to be scanned
489 * @param type type of searched subpacket
490 * @return node pointer for subpacket if found, else NULL
491 */
492 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
493 {
494 while (list != NULL && list->packet != NULL) {
495 if (list->packet->type == type)
496 return list;
497 list = list->next;
498 }
499 return NULL;
500 }
501
502
503 /**
504 * Replaces 8 elements with their average value.
505 * Called by qdm2_decode_superblock before starting subblock decoding.
506 *
507 * @param q context
508 */
509 static void average_quantized_coeffs (QDM2Context *q)
510 {
511 int i, j, n, ch, sum;
512
513 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
514
515 for (ch = 0; ch < q->nb_channels; ch++)
516 for (i = 0; i < n; i++) {
517 sum = 0;
518
519 for (j = 0; j < 8; j++)
520 sum += q->quantized_coeffs[ch][i][j];
521
522 sum /= 8;
523 if (sum > 0)
524 sum--;
525
526 for (j=0; j < 8; j++)
527 q->quantized_coeffs[ch][i][j] = sum;
528 }
529 }
530
531
532 /**
533 * Build subband samples with noise weighted by q->tone_level.
534 * Called by synthfilt_build_sb_samples.
535 *
536 * @param q context
537 * @param sb subband index
538 */
539 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
540 {
541 int ch, j;
542
543 FIX_NOISE_IDX(q->noise_idx);
544
545 if (!q->nb_channels)
546 return;
547
548 for (ch = 0; ch < q->nb_channels; ch++)
549 for (j = 0; j < 64; j++) {
550 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);
551 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);
552 }
553 }
554
555
556 /**
557 * Called while processing data from subpackets 11 and 12.
558 * Used after making changes to coding_method array.
559 *
560 * @param sb subband index
561 * @param channels number of channels
562 * @param coding_method q->coding_method[0][0][0]
563 */
564 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
565 {
566 int j,k;
567 int ch;
568 int run, case_val;
569 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};
570
571 for (ch = 0; ch < channels; ch++) {
572 for (j = 0; j < 64; ) {
573 if((coding_method[ch][sb][j] - 8) > 22) {
574 run = 1;
575 case_val = 8;
576 } else {
577 switch (switchtable[coding_method[ch][sb][j]-8]) {
578 case 0: run = 10; case_val = 10; break;
579 case 1: run = 1; case_val = 16; break;
580 case 2: run = 5; case_val = 24; break;
581 case 3: run = 3; case_val = 30; break;
582 case 4: run = 1; case_val = 30; break;
583 case 5: run = 1; case_val = 8; break;
584 default: run = 1; case_val = 8; break;
585 }
586 }
587 for (k = 0; k < run; k++)
588 if (j + k < 128)
589 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
590 if (k > 0) {
591 SAMPLES_NEEDED
592 //not debugged, almost never used
593 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
594 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
595 }
596 j += run;
597 }
598 }
599 }
600
601
602 /**
603 * Related to synthesis filter
604 * Called by process_subpacket_10
605 *
606 * @param q context
607 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
608 */
609 static void fill_tone_level_array (QDM2Context *q, int flag)
610 {
611 int i, sb, ch, sb_used;
612 int tmp, tab;
613
614 // This should never happen
615 if (q->nb_channels <= 0)
616 return;
617
618 for (ch = 0; ch < q->nb_channels; ch++)
619 for (sb = 0; sb < 30; sb++)
620 for (i = 0; i < 8; i++) {
621 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
622 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
623 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
624 else
625 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
626 if(tmp < 0)
627 tmp += 0xff;
628 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
629 }
630
631 sb_used = QDM2_SB_USED(q->sub_sampling);
632
633 if ((q->superblocktype_2_3 != 0) && !flag) {
634 for (sb = 0; sb < sb_used; sb++)
635 for (ch = 0; ch < q->nb_channels; ch++)
636 for (i = 0; i < 64; i++) {
637 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
638 if (q->tone_level_idx[ch][sb][i] < 0)
639 q->tone_level[ch][sb][i] = 0;
640 else
641 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
642 }
643 } else {
644 tab = q->superblocktype_2_3 ? 0 : 1;
645 for (sb = 0; sb < sb_used; sb++) {
646 if ((sb >= 4) && (sb <= 23)) {
647 for (ch = 0; ch < q->nb_channels; ch++)
648 for (i = 0; i < 64; i++) {
649 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
650 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
651 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
652 q->tone_level_idx_hi2[ch][sb - 4];
653 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
654 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
655 q->tone_level[ch][sb][i] = 0;
656 else
657 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
658 }
659 } else {
660 if (sb > 4) {
661 for (ch = 0; ch < q->nb_channels; ch++)
662 for (i = 0; i < 64; i++) {
663 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
664 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
665 q->tone_level_idx_hi2[ch][sb - 4];
666 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
667 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
668 q->tone_level[ch][sb][i] = 0;
669 else
670 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
671 }
672 } else {
673 for (ch = 0; ch < q->nb_channels; ch++)
674 for (i = 0; i < 64; i++) {
675 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
676 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
677 q->tone_level[ch][sb][i] = 0;
678 else
679 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
680 }
681 }
682 }
683 }
684 }
685
686 return;
687 }
688
689
690 /**
691 * Related to synthesis filter
692 * Called by process_subpacket_11
693 * c is built with data from subpacket 11
694 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
695 *
696 * @param tone_level_idx
697 * @param tone_level_idx_temp
698 * @param coding_method q->coding_method[0][0][0]
699 * @param nb_channels number of channels
700 * @param c coming from subpacket 11, passed as 8*c
701 * @param superblocktype_2_3 flag based on superblock packet type
702 * @param cm_table_select q->cm_table_select
703 */
704 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
705 sb_int8_array coding_method, int nb_channels,
706 int c, int superblocktype_2_3, int cm_table_select)
707 {
708 int ch, sb, j;
709 int tmp, acc, esp_40, comp;
710 int add1, add2, add3, add4;
711 int64_t multres;
712
713 // This should never happen
714 if (nb_channels <= 0)
715 return;
716
717 if (!superblocktype_2_3) {
718 /* This case is untested, no samples available */
719 SAMPLES_NEEDED
720 for (ch = 0; ch < nb_channels; ch++)
721 for (sb = 0; sb < 30; sb++) {
722 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
723 add1 = tone_level_idx[ch][sb][j] - 10;
724 if (add1 < 0)
725 add1 = 0;
726 add2 = add3 = add4 = 0;
727 if (sb > 1) {
728 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
729 if (add2 < 0)
730 add2 = 0;
731 }
732 if (sb > 0) {
733 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
734 if (add3 < 0)
735 add3 = 0;
736 }
737 if (sb < 29) {
738 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
739 if (add4 < 0)
740 add4 = 0;
741 }
742 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
743 if (tmp < 0)
744 tmp = 0;
745 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
746 }
747 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
748 }
749 acc = 0;
750 for (ch = 0; ch < nb_channels; ch++)
751 for (sb = 0; sb < 30; sb++)
752 for (j = 0; j < 64; j++)
753 acc += tone_level_idx_temp[ch][sb][j];
754
755 multres = 0x66666667 * (acc * 10);
756 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
757 for (ch = 0; ch < nb_channels; ch++)
758 for (sb = 0; sb < 30; sb++)
759 for (j = 0; j < 64; j++) {
760 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
761 if (comp < 0)
762 comp += 0xff;
763 comp /= 256; // signed shift
764 switch(sb) {
765 case 0:
766 if (comp < 30)
767 comp = 30;
768 comp += 15;
769 break;
770 case 1:
771 if (comp < 24)
772 comp = 24;
773 comp += 10;
774 break;
775 case 2:
776 case 3:
777 case 4:
778 if (comp < 16)
779 comp = 16;
780 }
781 if (comp <= 5)
782 tmp = 0;
783 else if (comp <= 10)
784 tmp = 10;
785 else if (comp <= 16)
786 tmp = 16;
787 else if (comp <= 24)
788 tmp = -1;
789 else
790 tmp = 0;
791 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
792 }
793 for (sb = 0; sb < 30; sb++)
794 fix_coding_method_array(sb, nb_channels, coding_method);
795 for (ch = 0; ch < nb_channels; ch++)
796 for (sb = 0; sb < 30; sb++)
797 for (j = 0; j < 64; j++)
798 if (sb >= 10) {
799 if (coding_method[ch][sb][j] < 10)
800 coding_method[ch][sb][j] = 10;
801 } else {
802 if (sb >= 2) {
803 if (coding_method[ch][sb][j] < 16)
804 coding_method[ch][sb][j] = 16;
805 } else {
806 if (coding_method[ch][sb][j] < 30)
807 coding_method[ch][sb][j] = 30;
808 }
809 }
810 } else { // superblocktype_2_3 != 0
811 for (ch = 0; ch < nb_channels; ch++)
812 for (sb = 0; sb < 30; sb++)
813 for (j = 0; j < 64; j++)
814 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
815 }
816
817 return;
818 }
819
820
821 /**
822 *
823 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
824 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
825 *
826 * @param q context
827 * @param gb bitreader context
828 * @param length packet length in bits
829 * @param sb_min lower subband processed (sb_min included)
830 * @param sb_max higher subband processed (sb_max excluded)
831 */
832 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
833 {
834 int sb, j, k, n, ch, run, channels;
835 int joined_stereo, zero_encoding, chs;
836 int type34_first;
837 float type34_div = 0;
838 float type34_predictor;
839 float samples[10], sign_bits[16];
840
841 if (length == 0) {
842 // If no data use noise
843 for (sb=sb_min; sb < sb_max; sb++)
844 build_sb_samples_from_noise (q, sb);
845
846 return;
847 }
848
849 for (sb = sb_min; sb < sb_max; sb++) {
850 FIX_NOISE_IDX(q->noise_idx);
851
852 channels = q->nb_channels;
853
854 if (q->nb_channels <= 1 || sb < 12)
855 joined_stereo = 0;
856 else if (sb >= 24)
857 joined_stereo = 1;
858 else
859 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
860
861 if (joined_stereo) {
862 if (BITS_LEFT(length,gb) >= 16)
863 for (j = 0; j < 16; j++)
864 sign_bits[j] = get_bits1 (gb);
865
866 for (j = 0; j < 64; j++)
867 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
868 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
869
870 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
871 channels = 1;
872 }
873
874 for (ch = 0; ch < channels; ch++) {
875 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
876 type34_predictor = 0.0;
877 type34_first = 1;
878
879 for (j = 0; j < 128; ) {
880 switch (q->coding_method[ch][sb][j / 2]) {
881 case 8:
882 if (BITS_LEFT(length,gb) >= 10) {
883 if (zero_encoding) {
884 for (k = 0; k < 5; k++) {
885 if ((j + 2 * k) >= 128)
886 break;
887 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
888 }
889 } else {
890 n = get_bits(gb, 8);
891 for (k = 0; k < 5; k++)
892 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
893 }
894 for (k = 0; k < 5; k++)
895 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
896 } else {
897 for (k = 0; k < 10; k++)
898 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
899 }
900 run = 10;
901 break;
902
903 case 10:
904 if (BITS_LEFT(length,gb) >= 1) {
905 float f = 0.81;
906
907 if (get_bits1(gb))
908 f = -f;
909 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
910 samples[0] = f;
911 } else {
912 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
913 }
914 run = 1;
915 break;
916
917 case 16:
918 if (BITS_LEFT(length,gb) >= 10) {
919 if (zero_encoding) {
920 for (k = 0; k < 5; k++) {
921 if ((j + k) >= 128)
922 break;
923 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
924 }
925 } else {
926 n = get_bits (gb, 8);
927 for (k = 0; k < 5; k++)
928 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
929 }
930 } else {
931 for (k = 0; k < 5; k++)
932 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
933 }
934 run = 5;
935 break;
936
937 case 24:
938 if (BITS_LEFT(length,gb) >= 7) {
939 n = get_bits(gb, 7);
940 for (k = 0; k < 3; k++)
941 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
942 } else {
943 for (k = 0; k < 3; k++)
944 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
945 }
946 run = 3;
947 break;
948
949 case 30:
950 if (BITS_LEFT(length,gb) >= 4)
951 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
952 else
953 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
954
955 run = 1;
956 break;
957
958 case 34:
959 if (BITS_LEFT(length,gb) >= 7) {
960 if (type34_first) {
961 type34_div = (float)(1 << get_bits(gb, 2));
962 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
963 type34_predictor = samples[0];
964 type34_first = 0;
965 } else {
966 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
967 type34_predictor = samples[0];
968 }
969 } else {
970 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
971 }
972 run = 1;
973 break;
974
975 default:
976 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
977 run = 1;
978 break;
979 }
980
981 if (joined_stereo) {
982 float tmp[10][MPA_MAX_CHANNELS];
983
984 for (k = 0; k < run; k++) {
985 tmp[k][0] = samples[k];
986 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
987 }
988 for (chs = 0; chs < q->nb_channels; chs++)
989 for (k = 0; k < run; k++)
990 if ((j + k) < 128)
991 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
992 } else {
993 for (k = 0; k < run; k++)
994 if ((j + k) < 128)
995 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
996 }
997
998 j += run;
999 } // j loop
1000 } // channel loop
1001 } // subband loop
1002 }
1003
1004
1005 /**
1006 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
1007 * This is similar to process_subpacket_9, but for a single channel and for element [0]
1008 * same VLC tables as process_subpacket_9 are used.
1009 *
1010 * @param q context
1011 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
1012 * @param gb bitreader context
1013 * @param length packet length in bits
1014 */
1015 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
1016 {
1017 int i, k, run, level, diff;
1018
1019 if (BITS_LEFT(length,gb) < 16)
1020 return;
1021 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
1022
1023 quantized_coeffs[0] = level;
1024
1025 for (i = 0; i < 7; ) {
1026 if (BITS_LEFT(length,gb) < 16)
1027 break;
1028 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
1029
1030 if (BITS_LEFT(length,gb) < 16)
1031 break;
1032 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
1033
1034 for (k = 1; k <= run; k++)
1035 quantized_coeffs[i + k] = (level + ((k * diff) / run));
1036
1037 level += diff;
1038 i += run;
1039 }
1040 }
1041
1042
1043 /**
1044 * Related to synthesis filter, process data from packet 10
1045 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1046 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1047 *
1048 * @param q context
1049 * @param gb bitreader context
1050 * @param length packet length in bits
1051 */
1052 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1053 {
1054 int sb, j, k, n, ch;
1055
1056 for (ch = 0; ch < q->nb_channels; ch++) {
1057 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1058
1059 if (BITS_LEFT(length,gb) < 16) {
1060 memset(q->quantized_coeffs[ch][0], 0, 8);
1061 break;
1062 }
1063 }
1064
1065 n = q->sub_sampling + 1;
1066
1067 for (sb = 0; sb < n; sb++)
1068 for (ch = 0; ch < q->nb_channels; ch++)
1069 for (j = 0; j < 8; j++) {
1070 if (BITS_LEFT(length,gb) < 1)
1071 break;
1072 if (get_bits1(gb)) {
1073 for (k=0; k < 8; k++) {
1074 if (BITS_LEFT(length,gb) < 16)
1075 break;
1076 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1077 }
1078 } else {
1079 for (k=0; k < 8; k++)
1080 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1081 }
1082 }
1083
1084 n = QDM2_SB_USED(q->sub_sampling) - 4;
1085
1086 for (sb = 0; sb < n; sb++)
1087 for (ch = 0; ch < q->nb_channels; ch++) {
1088 if (BITS_LEFT(length,gb) < 16)
1089 break;
1090 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1091 if (sb > 19)
1092 q->tone_level_idx_hi2[ch][sb] -= 16;
1093 else
1094 for (j = 0; j < 8; j++)
1095 q->tone_level_idx_mid[ch][sb][j] = -16;
1096 }
1097
1098 n = QDM2_SB_USED(q->sub_sampling) - 5;
1099
1100 for (sb = 0; sb < n; sb++)
1101 for (ch = 0; ch < q->nb_channels; ch++)
1102 for (j = 0; j < 8; j++) {
1103 if (BITS_LEFT(length,gb) < 16)
1104 break;
1105 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1106 }
1107 }
1108
1109 /**
1110 * Process subpacket 9, init quantized_coeffs with data from it
1111 *
1112 * @param q context
1113 * @param node pointer to node with packet
1114 */
1115 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1116 {
1117 GetBitContext gb;
1118 int i, j, k, n, ch, run, level, diff;
1119
1120 init_get_bits(&gb, node->packet->data, node->packet->size*8);
1121
1122 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1123
1124 for (i = 1; i < n; i++)
1125 for (ch=0; ch < q->nb_channels; ch++) {
1126 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1127 q->quantized_coeffs[ch][i][0] = level;
1128
1129 for (j = 0; j < (8 - 1); ) {
1130 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1131 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1132
1133 for (k = 1; k <= run; k++)
1134 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1135
1136 level += diff;
1137 j += run;
1138 }
1139 }
1140
1141 for (ch = 0; ch < q->nb_channels; ch++)
1142 for (i = 0; i < 8; i++)
1143 q->quantized_coeffs[ch][0][i] = 0;
1144 }
1145
1146
1147 /**
1148 * Process subpacket 10 if not null, else
1149 *
1150 * @param q context
1151 * @param node pointer to node with packet
1152 * @param length packet length in bits
1153 */
1154 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1155 {
1156 GetBitContext gb;
1157
1158 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1159
1160 if (length != 0) {
1161 init_tone_level_dequantization(q, &gb, length);
1162 fill_tone_level_array(q, 1);
1163 } else {
1164 fill_tone_level_array(q, 0);
1165 }
1166 }
1167
1168
1169 /**
1170 * Process subpacket 11
1171 *
1172 * @param q context
1173 * @param node pointer to node with packet
1174 * @param length packet length in bit
1175 */
1176 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1177 {
1178 GetBitContext gb;
1179
1180 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1181 if (length >= 32) {
1182 int c = get_bits (&gb, 13);
1183
1184 if (c > 3)
1185 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1186 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1187 }
1188
1189 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1190 }
1191
1192
1193 /**
1194 * Process subpacket 12
1195 *
1196 * @param q context
1197 * @param node pointer to node with packet
1198 * @param length packet length in bits
1199 */
1200 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1201 {
1202 GetBitContext gb;
1203
1204 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1205 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1206 }
1207
1208 /*
1209 * Process new subpackets for synthesis filter
1210 *
1211 * @param q context
1212 * @param list list with synthesis filter packets (list D)
1213 */
1214 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1215 {
1216 QDM2SubPNode *nodes[4];
1217
1218 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1219 if (nodes[0] != NULL)
1220 process_subpacket_9(q, nodes[0]);
1221
1222 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1223 if (nodes[1] != NULL)
1224 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1225 else
1226 process_subpacket_10(q, NULL, 0);
1227
1228 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1229 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1230 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1231 else
1232 process_subpacket_11(q, NULL, 0);
1233
1234 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1235 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1236 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1237 else
1238 process_subpacket_12(q, NULL, 0);
1239 }
1240
1241
1242 /*
1243 * Decode superblock, fill packet lists.
1244 *
1245 * @param q context
1246 */
1247 static void qdm2_decode_super_block (QDM2Context *q)
1248 {
1249 GetBitContext gb;
1250 QDM2SubPacket header, *packet;
1251 int i, packet_bytes, sub_packet_size, sub_packets_D;
1252 unsigned int next_index = 0;
1253
1254 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1255 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1256 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1257
1258 q->sub_packets_B = 0;
1259 sub_packets_D = 0;
1260
1261 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1262
1263 init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1264 qdm2_decode_sub_packet_header(&gb, &header);
1265
1266 if (header.type < 2 || header.type >= 8) {
1267 q->has_errors = 1;
1268 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1269 return;
1270 }
1271
1272 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1273 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1274
1275 init_get_bits(&gb, header.data, header.size*8);
1276
1277 if (header.type == 2 || header.type == 4 || header.type == 5) {
1278 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1279
1280 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1281
1282 if (csum != 0) {
1283 q->has_errors = 1;
1284 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1285 return;
1286 }
1287 }
1288
1289 q->sub_packet_list_B[0].packet = NULL;
1290 q->sub_packet_list_D[0].packet = NULL;
1291
1292 for (i = 0; i < 6; i++)
1293 if (--q->fft_level_exp[i] < 0)
1294 q->fft_level_exp[i] = 0;
1295
1296 for (i = 0; packet_bytes > 0; i++) {
1297 int j;
1298
1299 q->sub_packet_list_A[i].next = NULL;
1300
1301 if (i > 0) {
1302 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1303
1304 /* seek to next block */
1305 init_get_bits(&gb, header.data, header.size*8);
1306 skip_bits(&gb, next_index*8);
1307
1308 if (next_index >= header.size)
1309 break;
1310 }
1311
1312 /* decode subpacket */
1313 packet = &q->sub_packets[i];
1314 qdm2_decode_sub_packet_header(&gb, packet);
1315 next_index = packet->size + get_bits_count(&gb) / 8;
1316 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1317
1318 if (packet->type == 0)
1319 break;
1320
1321 if (sub_packet_size > packet_bytes) {
1322 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1323 break;
1324 packet->size += packet_bytes - sub_packet_size;
1325 }
1326
1327 packet_bytes -= sub_packet_size;
1328
1329 /* add subpacket to 'all subpackets' list */
1330 q->sub_packet_list_A[i].packet = packet;
1331
1332 /* add subpacket to related list */
1333 if (packet->type == 8) {
1334 SAMPLES_NEEDED_2("packet type 8");
1335 return;
1336 } else if (packet->type >= 9 && packet->type <= 12) {
1337 /* packets for MPEG Audio like Synthesis Filter */
1338 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1339 } else if (packet->type == 13) {
1340 for (j = 0; j < 6; j++)
1341 q->fft_level_exp[j] = get_bits(&gb, 6);
1342 } else if (packet->type == 14) {
1343 for (j = 0; j < 6; j++)
1344 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1345 } else if (packet->type == 15) {
1346 SAMPLES_NEEDED_2("packet type 15")
1347 return;
1348 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1349 /* packets for FFT */
1350 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1351 }
1352 } // Packet bytes loop
1353
1354 /* **************************************************************** */
1355 if (q->sub_packet_list_D[0].packet != NULL) {
1356 process_synthesis_subpackets(q, q->sub_packet_list_D);
1357 q->do_synth_filter = 1;
1358 } else if (q->do_synth_filter) {
1359 process_subpacket_10(q, NULL, 0);
1360 process_subpacket_11(q, NULL, 0);
1361 process_subpacket_12(q, NULL, 0);
1362 }
1363 /* **************************************************************** */
1364 }
1365
1366
1367 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1368 int offset, int duration, int channel,
1369 int exp, int phase)
1370 {
1371 if (q->fft_coefs_min_index[duration] < 0)
1372 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1373
1374 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1375 q->fft_coefs[q->fft_coefs_index].channel = channel;
1376 q->fft_coefs[q->fft_coefs_index].offset = offset;
1377 q->fft_coefs[q->fft_coefs_index].exp = exp;
1378 q->fft_coefs[q->fft_coefs_index].phase = phase;
1379 q->fft_coefs_index++;
1380 }
1381
1382
1383 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1384 {
1385 int channel, stereo, phase, exp;
1386 int local_int_4, local_int_8, stereo_phase, local_int_10;
1387 int local_int_14, stereo_exp, local_int_20, local_int_28;
1388 int n, offset;
1389
1390 local_int_4 = 0;
1391 local_int_28 = 0;
1392 local_int_20 = 2;
1393 local_int_8 = (4 - duration);
1394 local_int_10 = 1 << (q->group_order - duration - 1);
1395 offset = 1;
1396
1397 while (1) {
1398 if (q->superblocktype_2_3) {
1399 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1400 offset = 1;
1401 if (n == 0) {
1402 local_int_4 += local_int_10;
1403 local_int_28 += (1 << local_int_8);
1404 } else {
1405 local_int_4 += 8*local_int_10;
1406 local_int_28 += (8 << local_int_8);
1407 }
1408 }
1409 offset += (n - 2);
1410 } else {
1411 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1412 while (offset >= (local_int_10 - 1)) {
1413 offset += (1 - (local_int_10 - 1));
1414 local_int_4 += local_int_10;
1415 local_int_28 += (1 << local_int_8);
1416 }
1417 }
1418
1419 if (local_int_4 >= q->group_size)
1420 return;
1421
1422 local_int_14 = (offset >> local_int_8);
1423
1424 if (q->nb_channels > 1) {
1425 channel = get_bits1(gb);
1426 stereo = get_bits1(gb);
1427 } else {
1428 channel = 0;
1429 stereo = 0;
1430 }
1431
1432 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1433 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1434 exp = (exp < 0) ? 0 : exp;
1435
1436 phase = get_bits(gb, 3);
1437 stereo_exp = 0;
1438 stereo_phase = 0;
1439
1440 if (stereo) {
1441 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1442 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1443 if (stereo_phase < 0)
1444 stereo_phase += 8;
1445 }
1446
1447 if (q->frequency_range > (local_int_14 + 1)) {
1448 int sub_packet = (local_int_20 + local_int_28);
1449
1450 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1451 if (stereo)
1452 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1453 }
1454
1455 offset++;
1456 }
1457 }
1458
1459
1460 static void qdm2_decode_fft_packets (QDM2Context *q)
1461 {
1462 int i, j, min, max, value, type, unknown_flag;
1463 GetBitContext gb;
1464
1465 if (q->sub_packet_list_B[0].packet == NULL)
1466 return;
1467
1468 /* reset minimum indexes for FFT coefficients */
1469 q->fft_coefs_index = 0;
1470 for (i=0; i < 5; i++)
1471 q->fft_coefs_min_index[i] = -1;
1472
1473 /* process subpackets ordered by type, largest type first */
1474 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1475 QDM2SubPacket *packet= NULL;
1476
1477 /* find subpacket with largest type less than max */
1478 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1479 value = q->sub_packet_list_B[j].packet->type;
1480 if (value > min && value < max) {
1481 min = value;
1482 packet = q->sub_packet_list_B[j].packet;
1483 }
1484 }
1485
1486 max = min;
1487
1488 /* check for errors (?) */
1489 if (!packet)
1490 return;
1491
1492 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1493 return;
1494
1495 /* decode FFT tones */
1496 init_get_bits (&gb, packet->data, packet->size*8);
1497
1498 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1499 unknown_flag = 1;
1500 else
1501 unknown_flag = 0;
1502
1503 type = packet->type;
1504
1505 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1506 int duration = q->sub_sampling + 5 - (type & 15);
1507
1508 if (duration >= 0 && duration < 4)
1509 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1510 } else if (type == 31) {
1511 for (j=0; j < 4; j++)
1512 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1513 } else if (type == 46) {
1514 for (j=0; j < 6; j++)
1515 q->fft_level_exp[j] = get_bits(&gb, 6);
1516 for (j=0; j < 4; j++)
1517 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1518 }
1519 } // Loop on B packets
1520
1521 /* calculate maximum indexes for FFT coefficients */
1522 for (i = 0, j = -1; i < 5; i++)
1523 if (q->fft_coefs_min_index[i] >= 0) {
1524 if (j >= 0)
1525 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1526 j = i;
1527 }
1528 if (j >= 0)
1529 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1530 }
1531
1532
1533 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1534 {
1535 float level, f[6];
1536 int i;
1537 QDM2Complex c;
1538 const double iscale = 2.0*M_PI / 512.0;
1539
1540 tone->phase += tone->phase_shift;
1541
1542 /* calculate current level (maximum amplitude) of tone */
1543 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1544 c.im = level * sin(tone->phase*iscale);
1545 c.re = level * cos(tone->phase*iscale);
1546
1547 /* generate FFT coefficients for tone */
1548 if (tone->duration >= 3 || tone->cutoff >= 3) {
1549 tone->complex[0].im += c.im;
1550 tone->complex[0].re += c.re;
1551 tone->complex[1].im -= c.im;
1552 tone->complex[1].re -= c.re;
1553 } else {
1554 f[1] = -tone->table[4];
1555 f[0] = tone->table[3] - tone->table[0];
1556 f[2] = 1.0 - tone->table[2] - tone->table[3];
1557 f[3] = tone->table[1] + tone->table[4] - 1.0;
1558 f[4] = tone->table[0] - tone->table[1];
1559 f[5] = tone->table[2];
1560 for (i = 0; i < 2; i++) {
1561 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1562 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1563 }
1564 for (i = 0; i < 4; i++) {
1565 tone->complex[i].re += c.re * f[i+2];
1566 tone->complex[i].im += c.im * f[i+2];
1567 }
1568 }
1569
1570 /* copy the tone if it has not yet died out */
1571 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1572 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1573 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1574 }
1575 }
1576
1577
1578 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1579 {
1580 int i, j, ch;
1581 const double iscale = 0.25 * M_PI;
1582
1583 for (ch = 0; ch < q->channels; ch++) {
1584 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1585 }
1586
1587
1588 /* apply FFT tones with duration 4 (1 FFT period) */
1589 if (q->fft_coefs_min_index[4] >= 0)
1590 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1591 float level;
1592 QDM2Complex c;
1593
1594 if (q->fft_coefs[i].sub_packet != sub_packet)
1595 break;
1596
1597 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1598 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];
1599
1600 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1601 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1602 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1603 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1604 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1605 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1606 }
1607
1608 /* generate existing FFT tones */
1609 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1610 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1611 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1612 }
1613
1614 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1615 for (i = 0; i < 4; i++)
1616 if (q->fft_coefs_min_index[i] >= 0) {
1617 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1618 int offset, four_i;
1619 FFTTone tone;
1620
1621 if (q->fft_coefs[j].sub_packet != sub_packet)
1622 break;
1623
1624 four_i = (4 - i);
1625 offset = q->fft_coefs[j].offset >> four_i;
1626 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1627
1628 if (offset < q->frequency_range) {
1629 if (offset < 2)
1630 tone.cutoff = offset;
1631 else
1632 tone.cutoff = (offset >= 60) ? 3 : 2;
1633
1634 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];
1635 tone.complex = &q->fft.complex[ch][offset];
1636 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1637 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1638 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1639 tone.duration = i;
1640 tone.time_index = 0;
1641
1642 qdm2_fft_generate_tone(q, &tone);
1643 }
1644 }
1645 q->fft_coefs_min_index[i] = j;
1646 }
1647 }
1648
1649
1650 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1651 {
1652 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1653 int i;
1654 q->fft.complex[channel][0].re *= 2.0f;
1655 q->fft.complex[channel][0].im = 0.0f;
1656 ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1657 /* add samples to output buffer */
1658 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1659 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
1660 }
1661
1662
1663 /**
1664 * @param q context
1665 * @param index subpacket number
1666 */
1667 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1668 {
1669 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
1670 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1671
1672 /* copy sb_samples */
1673 sb_used = QDM2_SB_USED(q->sub_sampling);
1674
1675 for (ch = 0; ch < q->channels; ch++)
1676 for (i = 0; i < 8; i++)
1677 for (k=sb_used; k < SBLIMIT; k++)
1678 q->sb_samples[ch][(8 * index) + i][k] = 0;
1679
1680 for (ch = 0; ch < q->nb_channels; ch++) {
1681 OUT_INT *samples_ptr = samples + ch;
1682
1683 for (i = 0; i < 8; i++) {
1684 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1685 ff_mpa_synth_window, &dither_state,
1686 samples_ptr, q->nb_channels,
1687 q->sb_samples[ch][(8 * index) + i]);
1688 samples_ptr += 32 * q->nb_channels;
1689 }
1690 }
1691
1692 /* add samples to output buffer */
1693 sub_sampling = (4 >> q->sub_sampling);
1694
1695 for (ch = 0; ch < q->channels; ch++)
1696 for (i = 0; i < q->frame_size; i++)
1697 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
1698 }
1699
1700
1701 /**
1702 * Init static data (does not depend on specific file)
1703 *
1704 * @param q context
1705 */
1706 static av_cold void qdm2_init(QDM2Context *q) {
1707 static int initialized = 0;
1708
1709 if (initialized != 0)
1710 return;
1711 initialized = 1;
1712
1713 qdm2_init_vlc();
1714 ff_mpa_synth_init(ff_mpa_synth_window);
1715 softclip_table_init();
1716 rnd_table_init();
1717 init_noise_samples();
1718
1719 av_log(NULL, AV_LOG_DEBUG, "init done\n");
1720 }
1721
1722
1723 #if 0
1724 static void dump_context(QDM2Context *q)
1725 {
1726 int i;
1727 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1728 PRINT("compressed_data",q->compressed_data);
1729 PRINT("compressed_size",q->compressed_size);
1730 PRINT("frame_size",q->frame_size);
1731 PRINT("checksum_size",q->checksum_size);
1732 PRINT("channels",q->channels);
1733 PRINT("nb_channels",q->nb_channels);
1734 PRINT("fft_frame_size",q->fft_frame_size);
1735 PRINT("fft_size",q->fft_size);
1736 PRINT("sub_sampling",q->sub_sampling);
1737 PRINT("fft_order",q->fft_order);
1738 PRINT("group_order",q->group_order);
1739 PRINT("group_size",q->group_size);
1740 PRINT("sub_packet",q->sub_packet);
1741 PRINT("frequency_range",q->frequency_range);
1742 PRINT("has_errors",q->has_errors);
1743 PRINT("fft_tone_end",q->fft_tone_end);
1744 PRINT("fft_tone_start",q->fft_tone_start);
1745 PRINT("fft_coefs_index",q->fft_coefs_index);
1746 PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1747 PRINT("cm_table_select",q->cm_table_select);
1748 PRINT("noise_idx",q->noise_idx);
1749
1750 for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1751 {
1752 FFTTone *t = &q->fft_tones[i];
1753
1754 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1755 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1756 // PRINT(" level", t->level);
1757 PRINT(" phase", t->phase);
1758 PRINT(" phase_shift", t->phase_shift);
1759 PRINT(" duration", t->duration);
1760 PRINT(" samples_im", t->samples_im);
1761 PRINT(" samples_re", t->samples_re);
1762 PRINT(" table", t->table);
1763 }
1764
1765 }
1766 #endif
1767
1768
1769 /**
1770 * Init parameters from codec extradata
1771 */
1772 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1773 {
1774 QDM2Context *s = avctx->priv_data;
1775 uint8_t *extradata;
1776 int extradata_size;
1777 int tmp_val, tmp, size;
1778
1779 /* extradata parsing
1780
1781 Structure:
1782 wave {
1783 frma (QDM2)
1784 QDCA
1785 QDCP
1786 }
1787
1788 32 size (including this field)
1789 32 tag (=frma)
1790 32 type (=QDM2 or QDMC)
1791
1792 32 size (including this field, in bytes)
1793 32 tag (=QDCA) // maybe mandatory parameters
1794 32 unknown (=1)
1795 32 channels (=2)
1796 32 samplerate (=44100)
1797 32 bitrate (=96000)
1798 32 block size (=4096)
1799 32 frame size (=256) (for one channel)
1800 32 packet size (=1300)
1801
1802 32 size (including this field, in bytes)
1803 32 tag (=QDCP) // maybe some tuneable parameters
1804 32 float1 (=1.0)
1805 32 zero ?
1806 32 float2 (=1.0)
1807 32 float3 (=1.0)
1808 32 unknown (27)
1809 32 unknown (8)
1810 32 zero ?
1811 */
1812
1813 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1814 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1815 return -1;
1816 }
1817
1818 extradata = avctx->extradata;
1819 extradata_size = avctx->extradata_size;
1820
1821 while (extradata_size > 7) {
1822 if (!memcmp(extradata, "frmaQDM", 7))
1823 break;
1824 extradata++;
1825 extradata_size--;
1826 }
1827
1828 if (extradata_size < 12) {
1829 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1830 extradata_size);
1831 return -1;
1832 }
1833
1834 if (memcmp(extradata, "frmaQDM", 7)) {
1835 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1836 return -1;
1837 }
1838
1839 if (extradata[7] == 'C') {
1840 // s->is_qdmc = 1;
1841 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1842 return -1;
1843 }
1844
1845 extradata += 8;
1846 extradata_size -= 8;
1847
1848 size = AV_RB32(extradata);
1849
1850 if(size > extradata_size){
1851 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1852 extradata_size, size);
1853 return -1;
1854 }
1855
1856 extradata += 4;
1857 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1858 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1859 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1860 return -1;
1861 }
1862
1863 extradata += 8;
1864
1865 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1866 extradata += 4;
1867
1868 avctx->sample_rate = AV_RB32(extradata);
1869 extradata += 4;
1870
1871 avctx->bit_rate = AV_RB32(extradata);
1872 extradata += 4;
1873
1874 s->group_size = AV_RB32(extradata);
1875 extradata += 4;
1876
1877 s->fft_size = AV_RB32(extradata);
1878 extradata += 4;
1879
1880 s->checksum_size = AV_RB32(extradata);
1881
1882 s->fft_order = av_log2(s->fft_size) + 1;
1883 s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1884
1885 // something like max decodable tones
1886 s->group_order = av_log2(s->group_size) + 1;
1887 s->frame_size = s->group_size / 16; // 16 iterations per super block
1888
1889 s->sub_sampling = s->fft_order - 7;
1890 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1891
1892 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1893 case 0: tmp = 40; break;
1894 case 1: tmp = 48; break;
1895 case 2: tmp = 56; break;
1896 case 3: tmp = 72; break;
1897 case 4: tmp = 80; break;
1898 case 5: tmp = 100;break;
1899 default: tmp=s->sub_sampling; break;
1900 }
1901 tmp_val = 0;
1902 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1903 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1904 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1905 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1906 s->cm_table_select = tmp_val;
1907
1908 if (s->sub_sampling == 0)
1909 tmp = 7999;
1910 else
1911 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1912 /*
1913 0: 7999 -> 0
1914 1: 20000 -> 2
1915 2: 28000 -> 2
1916 */
1917 if (tmp < 8000)
1918 s->coeff_per_sb_select = 0;
1919 else if (tmp <= 16000)
1920 s->coeff_per_sb_select = 1;
1921 else
1922 s->coeff_per_sb_select = 2;
1923
1924 // Fail on unknown fft order
1925 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1926 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1927 return -1;
1928 }
1929
1930 ff_rdft_init(&s->rdft_ctx, s->fft_order, IRDFT);
1931
1932 qdm2_init(s);
1933
1934 avctx->sample_fmt = SAMPLE_FMT_S16;
1935
1936 // dump_context(s);
1937 return 0;
1938 }
1939
1940
1941 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1942 {
1943 QDM2Context *s = avctx->priv_data;
1944
1945 ff_rdft_end(&s->rdft_ctx);
1946
1947 return 0;
1948 }
1949
1950
1951 static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1952 {
1953 int ch, i;
1954 const int frame_size = (q->frame_size * q->channels);
1955
1956 /* select input buffer */
1957 q->compressed_data = in;
1958 q->compressed_size = q->checksum_size;
1959
1960 // dump_context(q);
1961
1962 /* copy old block, clear new block of output samples */
1963 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1964 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1965
1966 /* decode block of QDM2 compressed data */
1967 if (q->sub_packet == 0) {
1968 q->has_errors = 0; // zero it for a new super block
1969 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1970 qdm2_decode_super_block(q);
1971 }
1972
1973 /* parse subpackets */
1974 if (!q->has_errors) {
1975 if (q->sub_packet == 2)
1976 qdm2_decode_fft_packets(q);
1977
1978 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1979 }
1980
1981 /* sound synthesis stage 1 (FFT) */
1982 for (ch = 0; ch < q->channels; ch++) {
1983 qdm2_calculate_fft(q, ch, q->sub_packet);
1984
1985 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1986 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1987 return;
1988 }
1989 }
1990
1991 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1992 if (!q->has_errors && q->do_synth_filter)
1993 qdm2_synthesis_filter(q, q->sub_packet);
1994
1995 q->sub_packet = (q->sub_packet + 1) % 16;
1996
1997 /* clip and convert output float[] to 16bit signed samples */
1998 for (i = 0; i < frame_size; i++) {
1999 int value = (int)q->output_buffer[i];
2000
2001 if (value > SOFTCLIP_THRESHOLD)
2002 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
2003 else if (value < -SOFTCLIP_THRESHOLD)
2004 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
2005
2006 out[i] = value;
2007 }
2008 }
2009
2010
2011 static int qdm2_decode_frame(AVCodecContext *avctx,
2012 void *data, int *data_size,
2013 AVPacket *avpkt)
2014 {
2015 const uint8_t *buf = avpkt->data;
2016 int buf_size = avpkt->size;
2017 QDM2Context *s = avctx->priv_data;
2018
2019 if(!buf)
2020 return 0;
2021 if(buf_size < s->checksum_size)
2022 return -1;
2023
2024 *data_size = s->channels * s->frame_size * sizeof(int16_t);
2025
2026 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
2027 buf_size, buf, s->checksum_size, data, *data_size);
2028
2029 qdm2_decode(s, buf, data);
2030
2031 // reading only when next superblock found
2032 if (s->sub_packet == 0) {
2033 return s->checksum_size;
2034 }
2035
2036 return 0;
2037 }
2038
2039 AVCodec qdm2_decoder =
2040 {
2041 .name = "qdm2",
2042 .type = CODEC_TYPE_AUDIO,
2043 .id = CODEC_ID_QDM2,
2044 .priv_data_size = sizeof(QDM2Context),
2045 .init = qdm2_decode_init,
2046 .close = qdm2_decode_close,
2047 .decode = qdm2_decode_frame,
2048 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
2049 };
lucabe's unfinished and experimental patches for ffmpeg