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asfdec: also read Metadata Library Object
[FFMpeg-mirror/mplayer-patches.git] / libavcodec / qdm2.c
blob269a051f69e0f2c0567129c4c1afacf15860db05
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 Libav.
9 *
10 * Libav 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 * Libav 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 Libav; 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
27 * QDM2 decoder
28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
29 *
30 * The decoder is not perfect yet, there are still some distortions
31 * especially on files encoded with 16 or 8 subbands.
32 */
33
34 #include <math.h>
35 #include <stddef.h>
36 #include <stdio.h>
37
38 #define BITSTREAM_READER_LE
39 #include "libavutil/channel_layout.h"
40 #include "avcodec.h"
41 #include "get_bits.h"
42 #include "dsputil.h"
43 #include "internal.h"
44 #include "rdft.h"
45 #include "mpegaudiodsp.h"
46 #include "mpegaudio.h"
47
48 #include "qdm2data.h"
49 #include "qdm2_tablegen.h"
50
51 #undef NDEBUG
52 #include <assert.h>
53
54
55 #define QDM2_LIST_ADD(list, size, packet) \
56 do { \
57 if (size > 0) { \
58 list[size - 1].next = &list[size]; \
59 } \
60 list[size].packet = packet; \
61 list[size].next = NULL; \
62 size++; \
63 } while(0)
64
65 // Result is 8, 16 or 30
66 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
67
68 #define FIX_NOISE_IDX(noise_idx) \
69 if ((noise_idx) >= 3840) \
70 (noise_idx) -= 3840; \
71
72 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
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 #define QDM2_MAX_FRAME_SIZE 512
81
82 typedef int8_t sb_int8_array[2][30][64];
83
84 /**
85 * Subpacket
86 */
87 typedef struct {
88 int type; ///< subpacket type
89 unsigned int size; ///< subpacket size
90 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
91 } QDM2SubPacket;
92
93 /**
94 * A node in the subpacket list
95 */
96 typedef struct QDM2SubPNode {
97 QDM2SubPacket *packet; ///< packet
98 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
99 } QDM2SubPNode;
100
101 typedef struct {
102 float re;
103 float im;
104 } QDM2Complex;
105
106 typedef struct {
107 float level;
108 QDM2Complex *complex;
109 const float *table;
110 int phase;
111 int phase_shift;
112 int duration;
113 short time_index;
114 short cutoff;
115 } FFTTone;
116
117 typedef struct {
118 int16_t sub_packet;
119 uint8_t channel;
120 int16_t offset;
121 int16_t exp;
122 uint8_t phase;
123 } FFTCoefficient;
124
125 typedef struct {
126 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
127 } QDM2FFT;
128
129 /**
130 * QDM2 decoder context
131 */
132 typedef struct {
133 AVFrame frame;
134
135 /// Parameters from codec header, do not change during playback
136 int nb_channels; ///< number of channels
137 int channels; ///< number of channels
138 int group_size; ///< size of frame group (16 frames per group)
139 int fft_size; ///< size of FFT, in complex numbers
140 int checksum_size; ///< size of data block, used also for checksum
141
142 /// Parameters built from header parameters, do not change during playback
143 int group_order; ///< order of frame group
144 int fft_order; ///< order of FFT (actually fftorder+1)
145 int frame_size; ///< size of data frame
146 int frequency_range;
147 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
148 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
149 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
150
151 /// Packets and packet lists
152 QDM2SubPacket sub_packets[16]; ///< the packets themselves
153 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
154 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
155 int sub_packets_B; ///< number of packets on 'B' list
156 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
157 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
158
159 /// FFT and tones
160 FFTTone fft_tones[1000];
161 int fft_tone_start;
162 int fft_tone_end;
163 FFTCoefficient fft_coefs[1000];
164 int fft_coefs_index;
165 int fft_coefs_min_index[5];
166 int fft_coefs_max_index[5];
167 int fft_level_exp[6];
168 RDFTContext rdft_ctx;
169 QDM2FFT fft;
170
171 /// I/O data
172 const uint8_t *compressed_data;
173 int compressed_size;
174 float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
175
176 /// Synthesis filter
177 MPADSPContext mpadsp;
178 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
179 int synth_buf_offset[MPA_MAX_CHANNELS];
180 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
181 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
182
183 /// Mixed temporary data used in decoding
184 float tone_level[MPA_MAX_CHANNELS][30][64];
185 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
186 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
187 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
188 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
189 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
190 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
191 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
192 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
193
194 // Flags
195 int has_errors; ///< packet has errors
196 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
197 int do_synth_filter; ///< used to perform or skip synthesis filter
198
199 int sub_packet;
200 int noise_idx; ///< index for dithering noise table
201 } QDM2Context;
202
203
204 static VLC vlc_tab_level;
205 static VLC vlc_tab_diff;
206 static VLC vlc_tab_run;
207 static VLC fft_level_exp_alt_vlc;
208 static VLC fft_level_exp_vlc;
209 static VLC fft_stereo_exp_vlc;
210 static VLC fft_stereo_phase_vlc;
211 static VLC vlc_tab_tone_level_idx_hi1;
212 static VLC vlc_tab_tone_level_idx_mid;
213 static VLC vlc_tab_tone_level_idx_hi2;
214 static VLC vlc_tab_type30;
215 static VLC vlc_tab_type34;
216 static VLC vlc_tab_fft_tone_offset[5];
217
218 static const uint16_t qdm2_vlc_offs[] = {
219 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
220 };
221
222 static av_cold void qdm2_init_vlc(void)
223 {
224 static int vlcs_initialized = 0;
225 static VLC_TYPE qdm2_table[3838][2];
226
227 if (!vlcs_initialized) {
228
229 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
230 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
231 init_vlc (&vlc_tab_level, 8, 24,
232 vlc_tab_level_huffbits, 1, 1,
233 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
234
235 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
236 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
237 init_vlc (&vlc_tab_diff, 8, 37,
238 vlc_tab_diff_huffbits, 1, 1,
239 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
240
241 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
242 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
243 init_vlc (&vlc_tab_run, 5, 6,
244 vlc_tab_run_huffbits, 1, 1,
245 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
246
247 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
248 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
249 init_vlc (&fft_level_exp_alt_vlc, 8, 28,
250 fft_level_exp_alt_huffbits, 1, 1,
251 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
252
253
254 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
255 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
256 init_vlc (&fft_level_exp_vlc, 8, 20,
257 fft_level_exp_huffbits, 1, 1,
258 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
259
260 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
261 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
262 init_vlc (&fft_stereo_exp_vlc, 6, 7,
263 fft_stereo_exp_huffbits, 1, 1,
264 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
265
266 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
267 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
268 init_vlc (&fft_stereo_phase_vlc, 6, 9,
269 fft_stereo_phase_huffbits, 1, 1,
270 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
271
272 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
273 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
274 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
275 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
276 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
277
278 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
279 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
280 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
281 vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
282 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
283
284 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
285 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
286 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
287 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
288 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
289
290 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
291 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
292 init_vlc (&vlc_tab_type30, 6, 9,
293 vlc_tab_type30_huffbits, 1, 1,
294 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
295
296 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
297 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
298 init_vlc (&vlc_tab_type34, 5, 10,
299 vlc_tab_type34_huffbits, 1, 1,
300 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
301
302 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
303 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
304 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
305 vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
306 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
307
308 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
309 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
310 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
311 vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
312 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
313
314 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
315 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
316 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
317 vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
318 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
319
320 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
321 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
322 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
323 vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
324 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
325
326 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
327 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
328 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
329 vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
330 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
331
332 vlcs_initialized=1;
333 }
334 }
335
336 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
337 {
338 int value;
339
340 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
341
342 /* stage-2, 3 bits exponent escape sequence */
343 if (value-- == 0)
344 value = get_bits (gb, get_bits (gb, 3) + 1);
345
346 /* stage-3, optional */
347 if (flag) {
348 int tmp = vlc_stage3_values[value];
349
350 if ((value & ~3) > 0)
351 tmp += get_bits (gb, (value >> 2));
352 value = tmp;
353 }
354
355 return value;
356 }
357
358
359 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
360 {
361 int value = qdm2_get_vlc (gb, vlc, 0, depth);
362
363 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
364 }
365
366
367 /**
368 * QDM2 checksum
369 *
370 * @param data pointer to data to be checksum'ed
371 * @param length data length
372 * @param value checksum value
373 *
374 * @return 0 if checksum is OK
375 */
376 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
377 int i;
378
379 for (i=0; i < length; i++)
380 value -= data[i];
381
382 return (uint16_t)(value & 0xffff);
383 }
384
385
386 /**
387 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
388 *
389 * @param gb bitreader context
390 * @param sub_packet packet under analysis
391 */
392 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
393 {
394 sub_packet->type = get_bits (gb, 8);
395
396 if (sub_packet->type == 0) {
397 sub_packet->size = 0;
398 sub_packet->data = NULL;
399 } else {
400 sub_packet->size = get_bits (gb, 8);
401
402 if (sub_packet->type & 0x80) {
403 sub_packet->size <<= 8;
404 sub_packet->size |= get_bits (gb, 8);
405 sub_packet->type &= 0x7f;
406 }
407
408 if (sub_packet->type == 0x7f)
409 sub_packet->type |= (get_bits (gb, 8) << 8);
410
411 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
412 }
413
414 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
415 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
416 }
417
418
419 /**
420 * Return node pointer to first packet of requested type in list.
421 *
422 * @param list list of subpackets to be scanned
423 * @param type type of searched subpacket
424 * @return node pointer for subpacket if found, else NULL
425 */
426 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
427 {
428 while (list != NULL && list->packet != NULL) {
429 if (list->packet->type == type)
430 return list;
431 list = list->next;
432 }
433 return NULL;
434 }
435
436
437 /**
438 * Replace 8 elements with their average value.
439 * Called by qdm2_decode_superblock before starting subblock decoding.
440 *
441 * @param q context
442 */
443 static void average_quantized_coeffs (QDM2Context *q)
444 {
445 int i, j, n, ch, sum;
446
447 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
448
449 for (ch = 0; ch < q->nb_channels; ch++)
450 for (i = 0; i < n; i++) {
451 sum = 0;
452
453 for (j = 0; j < 8; j++)
454 sum += q->quantized_coeffs[ch][i][j];
455
456 sum /= 8;
457 if (sum > 0)
458 sum--;
459
460 for (j=0; j < 8; j++)
461 q->quantized_coeffs[ch][i][j] = sum;
462 }
463 }
464
465
466 /**
467 * Build subband samples with noise weighted by q->tone_level.
468 * Called by synthfilt_build_sb_samples.
469 *
470 * @param q context
471 * @param sb subband index
472 */
473 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
474 {
475 int ch, j;
476
477 FIX_NOISE_IDX(q->noise_idx);
478
479 if (!q->nb_channels)
480 return;
481
482 for (ch = 0; ch < q->nb_channels; ch++)
483 for (j = 0; j < 64; j++) {
484 q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
485 q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
486 }
487 }
488
489
490 /**
491 * Called while processing data from subpackets 11 and 12.
492 * Used after making changes to coding_method array.
493 *
494 * @param sb subband index
495 * @param channels number of channels
496 * @param coding_method q->coding_method[0][0][0]
497 */
498 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
499 {
500 int j,k;
501 int ch;
502 int run, case_val;
503 static const 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};
504
505 for (ch = 0; ch < channels; ch++) {
506 for (j = 0; j < 64; ) {
507 if((coding_method[ch][sb][j] - 8) > 22) {
508 run = 1;
509 case_val = 8;
510 } else {
511 switch (switchtable[coding_method[ch][sb][j]-8]) {
512 case 0: run = 10; case_val = 10; break;
513 case 1: run = 1; case_val = 16; break;
514 case 2: run = 5; case_val = 24; break;
515 case 3: run = 3; case_val = 30; break;
516 case 4: run = 1; case_val = 30; break;
517 case 5: run = 1; case_val = 8; break;
518 default: run = 1; case_val = 8; break;
519 }
520 }
521 for (k = 0; k < run; k++)
522 if (j + k < 128)
523 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
524 if (k > 0) {
525 SAMPLES_NEEDED
526 //not debugged, almost never used
527 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
528 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
529 }
530 j += run;
531 }
532 }
533 }
534
535
536 /**
537 * Related to synthesis filter
538 * Called by process_subpacket_10
539 *
540 * @param q context
541 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
542 */
543 static void fill_tone_level_array (QDM2Context *q, int flag)
544 {
545 int i, sb, ch, sb_used;
546 int tmp, tab;
547
548 for (ch = 0; ch < q->nb_channels; ch++)
549 for (sb = 0; sb < 30; sb++)
550 for (i = 0; i < 8; i++) {
551 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
552 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
553 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
554 else
555 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
556 if(tmp < 0)
557 tmp += 0xff;
558 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
559 }
560
561 sb_used = QDM2_SB_USED(q->sub_sampling);
562
563 if ((q->superblocktype_2_3 != 0) && !flag) {
564 for (sb = 0; sb < sb_used; sb++)
565 for (ch = 0; ch < q->nb_channels; ch++)
566 for (i = 0; i < 64; i++) {
567 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
568 if (q->tone_level_idx[ch][sb][i] < 0)
569 q->tone_level[ch][sb][i] = 0;
570 else
571 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
572 }
573 } else {
574 tab = q->superblocktype_2_3 ? 0 : 1;
575 for (sb = 0; sb < sb_used; sb++) {
576 if ((sb >= 4) && (sb <= 23)) {
577 for (ch = 0; ch < q->nb_channels; ch++)
578 for (i = 0; i < 64; i++) {
579 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
580 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
581 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
582 q->tone_level_idx_hi2[ch][sb - 4];
583 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
584 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
585 q->tone_level[ch][sb][i] = 0;
586 else
587 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
588 }
589 } else {
590 if (sb > 4) {
591 for (ch = 0; ch < q->nb_channels; ch++)
592 for (i = 0; i < 64; i++) {
593 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
594 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
595 q->tone_level_idx_hi2[ch][sb - 4];
596 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
597 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
598 q->tone_level[ch][sb][i] = 0;
599 else
600 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
601 }
602 } else {
603 for (ch = 0; ch < q->nb_channels; ch++)
604 for (i = 0; i < 64; i++) {
605 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
606 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
607 q->tone_level[ch][sb][i] = 0;
608 else
609 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
610 }
611 }
612 }
613 }
614 }
615
616 return;
617 }
618
619
620 /**
621 * Related to synthesis filter
622 * Called by process_subpacket_11
623 * c is built with data from subpacket 11
624 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
625 *
626 * @param tone_level_idx
627 * @param tone_level_idx_temp
628 * @param coding_method q->coding_method[0][0][0]
629 * @param nb_channels number of channels
630 * @param c coming from subpacket 11, passed as 8*c
631 * @param superblocktype_2_3 flag based on superblock packet type
632 * @param cm_table_select q->cm_table_select
633 */
634 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
635 sb_int8_array coding_method, int nb_channels,
636 int c, int superblocktype_2_3, int cm_table_select)
637 {
638 int ch, sb, j;
639 int tmp, acc, esp_40, comp;
640 int add1, add2, add3, add4;
641 int64_t multres;
642
643 if (!superblocktype_2_3) {
644 /* This case is untested, no samples available */
645 SAMPLES_NEEDED
646 for (ch = 0; ch < nb_channels; ch++)
647 for (sb = 0; sb < 30; sb++) {
648 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
649 add1 = tone_level_idx[ch][sb][j] - 10;
650 if (add1 < 0)
651 add1 = 0;
652 add2 = add3 = add4 = 0;
653 if (sb > 1) {
654 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
655 if (add2 < 0)
656 add2 = 0;
657 }
658 if (sb > 0) {
659 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
660 if (add3 < 0)
661 add3 = 0;
662 }
663 if (sb < 29) {
664 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
665 if (add4 < 0)
666 add4 = 0;
667 }
668 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
669 if (tmp < 0)
670 tmp = 0;
671 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
672 }
673 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
674 }
675 acc = 0;
676 for (ch = 0; ch < nb_channels; ch++)
677 for (sb = 0; sb < 30; sb++)
678 for (j = 0; j < 64; j++)
679 acc += tone_level_idx_temp[ch][sb][j];
680
681 multres = 0x66666667 * (acc * 10);
682 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
683 for (ch = 0; ch < nb_channels; ch++)
684 for (sb = 0; sb < 30; sb++)
685 for (j = 0; j < 64; j++) {
686 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
687 if (comp < 0)
688 comp += 0xff;
689 comp /= 256; // signed shift
690 switch(sb) {
691 case 0:
692 if (comp < 30)
693 comp = 30;
694 comp += 15;
695 break;
696 case 1:
697 if (comp < 24)
698 comp = 24;
699 comp += 10;
700 break;
701 case 2:
702 case 3:
703 case 4:
704 if (comp < 16)
705 comp = 16;
706 }
707 if (comp <= 5)
708 tmp = 0;
709 else if (comp <= 10)
710 tmp = 10;
711 else if (comp <= 16)
712 tmp = 16;
713 else if (comp <= 24)
714 tmp = -1;
715 else
716 tmp = 0;
717 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
718 }
719 for (sb = 0; sb < 30; sb++)
720 fix_coding_method_array(sb, nb_channels, coding_method);
721 for (ch = 0; ch < nb_channels; ch++)
722 for (sb = 0; sb < 30; sb++)
723 for (j = 0; j < 64; j++)
724 if (sb >= 10) {
725 if (coding_method[ch][sb][j] < 10)
726 coding_method[ch][sb][j] = 10;
727 } else {
728 if (sb >= 2) {
729 if (coding_method[ch][sb][j] < 16)
730 coding_method[ch][sb][j] = 16;
731 } else {
732 if (coding_method[ch][sb][j] < 30)
733 coding_method[ch][sb][j] = 30;
734 }
735 }
736 } else { // superblocktype_2_3 != 0
737 for (ch = 0; ch < nb_channels; ch++)
738 for (sb = 0; sb < 30; sb++)
739 for (j = 0; j < 64; j++)
740 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
741 }
742
743 return;
744 }
745
746
747 /**
748 *
749 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
750 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
751 *
752 * @param q context
753 * @param gb bitreader context
754 * @param length packet length in bits
755 * @param sb_min lower subband processed (sb_min included)
756 * @param sb_max higher subband processed (sb_max excluded)
757 */
758 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
759 {
760 int sb, j, k, n, ch, run, channels;
761 int joined_stereo, zero_encoding, chs;
762 int type34_first;
763 float type34_div = 0;
764 float type34_predictor;
765 float samples[10], sign_bits[16];
766
767 if (length == 0) {
768 // If no data use noise
769 for (sb=sb_min; sb < sb_max; sb++)
770 build_sb_samples_from_noise (q, sb);
771
772 return;
773 }
774
775 for (sb = sb_min; sb < sb_max; sb++) {
776 FIX_NOISE_IDX(q->noise_idx);
777
778 channels = q->nb_channels;
779
780 if (q->nb_channels <= 1 || sb < 12)
781 joined_stereo = 0;
782 else if (sb >= 24)
783 joined_stereo = 1;
784 else
785 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0;
786
787 if (joined_stereo) {
788 if (get_bits_left(gb) >= 16)
789 for (j = 0; j < 16; j++)
790 sign_bits[j] = get_bits1 (gb);
791
792 for (j = 0; j < 64; j++)
793 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
794 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
795
796 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
797 channels = 1;
798 }
799
800 for (ch = 0; ch < channels; ch++) {
801 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
802 type34_predictor = 0.0;
803 type34_first = 1;
804
805 for (j = 0; j < 128; ) {
806 switch (q->coding_method[ch][sb][j / 2]) {
807 case 8:
808 if (get_bits_left(gb) >= 10) {
809 if (zero_encoding) {
810 for (k = 0; k < 5; k++) {
811 if ((j + 2 * k) >= 128)
812 break;
813 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
814 }
815 } else {
816 n = get_bits(gb, 8);
817 for (k = 0; k < 5; k++)
818 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
819 }
820 for (k = 0; k < 5; k++)
821 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
822 } else {
823 for (k = 0; k < 10; k++)
824 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
825 }
826 run = 10;
827 break;
828
829 case 10:
830 if (get_bits_left(gb) >= 1) {
831 float f = 0.81;
832
833 if (get_bits1(gb))
834 f = -f;
835 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
836 samples[0] = f;
837 } else {
838 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
839 }
840 run = 1;
841 break;
842
843 case 16:
844 if (get_bits_left(gb) >= 10) {
845 if (zero_encoding) {
846 for (k = 0; k < 5; k++) {
847 if ((j + k) >= 128)
848 break;
849 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
850 }
851 } else {
852 n = get_bits (gb, 8);
853 for (k = 0; k < 5; k++)
854 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
855 }
856 } else {
857 for (k = 0; k < 5; k++)
858 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
859 }
860 run = 5;
861 break;
862
863 case 24:
864 if (get_bits_left(gb) >= 7) {
865 n = get_bits(gb, 7);
866 for (k = 0; k < 3; k++)
867 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
868 } else {
869 for (k = 0; k < 3; k++)
870 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
871 }
872 run = 3;
873 break;
874
875 case 30:
876 if (get_bits_left(gb) >= 4) {
877 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
878 if (index < FF_ARRAY_ELEMS(type30_dequant)) {
879 samples[0] = type30_dequant[index];
880 } else
881 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
882 } else
883 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
884
885 run = 1;
886 break;
887
888 case 34:
889 if (get_bits_left(gb) >= 7) {
890 if (type34_first) {
891 type34_div = (float)(1 << get_bits(gb, 2));
892 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
893 type34_predictor = samples[0];
894 type34_first = 0;
895 } else {
896 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
897 if (index < FF_ARRAY_ELEMS(type34_delta)) {
898 samples[0] = type34_delta[index] / type34_div + type34_predictor;
899 type34_predictor = samples[0];
900 } else
901 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
902 }
903 } else {
904 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
905 }
906 run = 1;
907 break;
908
909 default:
910 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
911 run = 1;
912 break;
913 }
914
915 if (joined_stereo) {
916 float tmp[10][MPA_MAX_CHANNELS];
917
918 for (k = 0; k < run; k++) {
919 tmp[k][0] = samples[k];
920 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
921 }
922 for (chs = 0; chs < q->nb_channels; chs++)
923 for (k = 0; k < run; k++)
924 if ((j + k) < 128)
925 q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
926 } else {
927 for (k = 0; k < run; k++)
928 if ((j + k) < 128)
929 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
930 }
931
932 j += run;
933 } // j loop
934 } // channel loop
935 } // subband loop
936 }
937
938
939 /**
940 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
941 * This is similar to process_subpacket_9, but for a single channel and for element [0]
942 * same VLC tables as process_subpacket_9 are used.
943 *
944 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
945 * @param gb bitreader context
946 */
947 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb)
948 {
949 int i, k, run, level, diff;
950
951 if (get_bits_left(gb) < 16)
952 return;
953 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
954
955 quantized_coeffs[0] = level;
956
957 for (i = 0; i < 7; ) {
958 if (get_bits_left(gb) < 16)
959 break;
960 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
961
962 if (get_bits_left(gb) < 16)
963 break;
964 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
965
966 for (k = 1; k <= run; k++)
967 quantized_coeffs[i + k] = (level + ((k * diff) / run));
968
969 level += diff;
970 i += run;
971 }
972 }
973
974
975 /**
976 * Related to synthesis filter, process data from packet 10
977 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
978 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
979 *
980 * @param q context
981 * @param gb bitreader context
982 */
983 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
984 {
985 int sb, j, k, n, ch;
986
987 for (ch = 0; ch < q->nb_channels; ch++) {
988 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
989
990 if (get_bits_left(gb) < 16) {
991 memset(q->quantized_coeffs[ch][0], 0, 8);
992 break;
993 }
994 }
995
996 n = q->sub_sampling + 1;
997
998 for (sb = 0; sb < n; sb++)
999 for (ch = 0; ch < q->nb_channels; ch++)
1000 for (j = 0; j < 8; j++) {
1001 if (get_bits_left(gb) < 1)
1002 break;
1003 if (get_bits1(gb)) {
1004 for (k=0; k < 8; k++) {
1005 if (get_bits_left(gb) < 16)
1006 break;
1007 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1008 }
1009 } else {
1010 for (k=0; k < 8; k++)
1011 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1012 }
1013 }
1014
1015 n = QDM2_SB_USED(q->sub_sampling) - 4;
1016
1017 for (sb = 0; sb < n; sb++)
1018 for (ch = 0; ch < q->nb_channels; ch++) {
1019 if (get_bits_left(gb) < 16)
1020 break;
1021 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1022 if (sb > 19)
1023 q->tone_level_idx_hi2[ch][sb] -= 16;
1024 else
1025 for (j = 0; j < 8; j++)
1026 q->tone_level_idx_mid[ch][sb][j] = -16;
1027 }
1028
1029 n = QDM2_SB_USED(q->sub_sampling) - 5;
1030
1031 for (sb = 0; sb < n; sb++)
1032 for (ch = 0; ch < q->nb_channels; ch++)
1033 for (j = 0; j < 8; j++) {
1034 if (get_bits_left(gb) < 16)
1035 break;
1036 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1037 }
1038 }
1039
1040 /**
1041 * Process subpacket 9, init quantized_coeffs with data from it
1042 *
1043 * @param q context
1044 * @param node pointer to node with packet
1045 */
1046 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1047 {
1048 GetBitContext gb;
1049 int i, j, k, n, ch, run, level, diff;
1050
1051 init_get_bits(&gb, node->packet->data, node->packet->size*8);
1052
1053 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1054
1055 for (i = 1; i < n; i++)
1056 for (ch=0; ch < q->nb_channels; ch++) {
1057 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1058 q->quantized_coeffs[ch][i][0] = level;
1059
1060 for (j = 0; j < (8 - 1); ) {
1061 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1062 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1063
1064 for (k = 1; k <= run; k++)
1065 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1066
1067 level += diff;
1068 j += run;
1069 }
1070 }
1071
1072 for (ch = 0; ch < q->nb_channels; ch++)
1073 for (i = 0; i < 8; i++)
1074 q->quantized_coeffs[ch][0][i] = 0;
1075 }
1076
1077
1078 /**
1079 * Process subpacket 10 if not null, else
1080 *
1081 * @param q context
1082 * @param node pointer to node with packet
1083 */
1084 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node)
1085 {
1086 GetBitContext gb;
1087
1088 if (node) {
1089 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1090 init_tone_level_dequantization(q, &gb);
1091 fill_tone_level_array(q, 1);
1092 } else {
1093 fill_tone_level_array(q, 0);
1094 }
1095 }
1096
1097
1098 /**
1099 * Process subpacket 11
1100 *
1101 * @param q context
1102 * @param node pointer to node with packet
1103 */
1104 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node)
1105 {
1106 GetBitContext gb;
1107 int length = 0;
1108
1109 if (node) {
1110 length = node->packet->size * 8;
1111 init_get_bits(&gb, node->packet->data, length);
1112 }
1113
1114 if (length >= 32) {
1115 int c = get_bits (&gb, 13);
1116
1117 if (c > 3)
1118 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1119 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1120 }
1121
1122 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1123 }
1124
1125
1126 /**
1127 * Process subpacket 12
1128 *
1129 * @param q context
1130 * @param node pointer to node with packet
1131 */
1132 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node)
1133 {
1134 GetBitContext gb;
1135 int length = 0;
1136
1137 if (node) {
1138 length = node->packet->size * 8;
1139 init_get_bits(&gb, node->packet->data, length);
1140 }
1141
1142 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1143 }
1144
1145 /*
1146 * Process new subpackets for synthesis filter
1147 *
1148 * @param q context
1149 * @param list list with synthesis filter packets (list D)
1150 */
1151 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1152 {
1153 QDM2SubPNode *nodes[4];
1154
1155 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1156 if (nodes[0] != NULL)
1157 process_subpacket_9(q, nodes[0]);
1158
1159 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1160 if (nodes[1] != NULL)
1161 process_subpacket_10(q, nodes[1]);
1162 else
1163 process_subpacket_10(q, NULL);
1164
1165 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1166 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1167 process_subpacket_11(q, nodes[2]);
1168 else
1169 process_subpacket_11(q, NULL);
1170
1171 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1172 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1173 process_subpacket_12(q, nodes[3]);
1174 else
1175 process_subpacket_12(q, NULL);
1176 }
1177
1178
1179 /*
1180 * Decode superblock, fill packet lists.
1181 *
1182 * @param q context
1183 */
1184 static void qdm2_decode_super_block (QDM2Context *q)
1185 {
1186 GetBitContext gb;
1187 QDM2SubPacket header, *packet;
1188 int i, packet_bytes, sub_packet_size, sub_packets_D;
1189 unsigned int next_index = 0;
1190
1191 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1192 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1193 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1194
1195 q->sub_packets_B = 0;
1196 sub_packets_D = 0;
1197
1198 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1199
1200 init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1201 qdm2_decode_sub_packet_header(&gb, &header);
1202
1203 if (header.type < 2 || header.type >= 8) {
1204 q->has_errors = 1;
1205 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1206 return;
1207 }
1208
1209 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1210 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1211
1212 init_get_bits(&gb, header.data, header.size*8);
1213
1214 if (header.type == 2 || header.type == 4 || header.type == 5) {
1215 int csum = 257 * get_bits(&gb, 8);
1216 csum += 2 * get_bits(&gb, 8);
1217
1218 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1219
1220 if (csum != 0) {
1221 q->has_errors = 1;
1222 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1223 return;
1224 }
1225 }
1226
1227 q->sub_packet_list_B[0].packet = NULL;
1228 q->sub_packet_list_D[0].packet = NULL;
1229
1230 for (i = 0; i < 6; i++)
1231 if (--q->fft_level_exp[i] < 0)
1232 q->fft_level_exp[i] = 0;
1233
1234 for (i = 0; packet_bytes > 0; i++) {
1235 int j;
1236
1237 q->sub_packet_list_A[i].next = NULL;
1238
1239 if (i > 0) {
1240 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1241
1242 /* seek to next block */
1243 init_get_bits(&gb, header.data, header.size*8);
1244 skip_bits(&gb, next_index*8);
1245
1246 if (next_index >= header.size)
1247 break;
1248 }
1249
1250 /* decode subpacket */
1251 packet = &q->sub_packets[i];
1252 qdm2_decode_sub_packet_header(&gb, packet);
1253 next_index = packet->size + get_bits_count(&gb) / 8;
1254 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1255
1256 if (packet->type == 0)
1257 break;
1258
1259 if (sub_packet_size > packet_bytes) {
1260 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1261 break;
1262 packet->size += packet_bytes - sub_packet_size;
1263 }
1264
1265 packet_bytes -= sub_packet_size;
1266
1267 /* add subpacket to 'all subpackets' list */
1268 q->sub_packet_list_A[i].packet = packet;
1269
1270 /* add subpacket to related list */
1271 if (packet->type == 8) {
1272 SAMPLES_NEEDED_2("packet type 8");
1273 return;
1274 } else if (packet->type >= 9 && packet->type <= 12) {
1275 /* packets for MPEG Audio like Synthesis Filter */
1276 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1277 } else if (packet->type == 13) {
1278 for (j = 0; j < 6; j++)
1279 q->fft_level_exp[j] = get_bits(&gb, 6);
1280 } else if (packet->type == 14) {
1281 for (j = 0; j < 6; j++)
1282 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1283 } else if (packet->type == 15) {
1284 SAMPLES_NEEDED_2("packet type 15")
1285 return;
1286 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1287 /* packets for FFT */
1288 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1289 }
1290 } // Packet bytes loop
1291
1292 /* **************************************************************** */
1293 if (q->sub_packet_list_D[0].packet != NULL) {
1294 process_synthesis_subpackets(q, q->sub_packet_list_D);
1295 q->do_synth_filter = 1;
1296 } else if (q->do_synth_filter) {
1297 process_subpacket_10(q, NULL);
1298 process_subpacket_11(q, NULL);
1299 process_subpacket_12(q, NULL);
1300 }
1301 /* **************************************************************** */
1302 }
1303
1304
1305 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1306 int offset, int duration, int channel,
1307 int exp, int phase)
1308 {
1309 if (q->fft_coefs_min_index[duration] < 0)
1310 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1311
1312 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1313 q->fft_coefs[q->fft_coefs_index].channel = channel;
1314 q->fft_coefs[q->fft_coefs_index].offset = offset;
1315 q->fft_coefs[q->fft_coefs_index].exp = exp;
1316 q->fft_coefs[q->fft_coefs_index].phase = phase;
1317 q->fft_coefs_index++;
1318 }
1319
1320
1321 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1322 {
1323 int channel, stereo, phase, exp;
1324 int local_int_4, local_int_8, stereo_phase, local_int_10;
1325 int local_int_14, stereo_exp, local_int_20, local_int_28;
1326 int n, offset;
1327
1328 local_int_4 = 0;
1329 local_int_28 = 0;
1330 local_int_20 = 2;
1331 local_int_8 = (4 - duration);
1332 local_int_10 = 1 << (q->group_order - duration - 1);
1333 offset = 1;
1334
1335 while (1) {
1336 if (q->superblocktype_2_3) {
1337 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1338 offset = 1;
1339 if (n == 0) {
1340 local_int_4 += local_int_10;
1341 local_int_28 += (1 << local_int_8);
1342 } else {
1343 local_int_4 += 8*local_int_10;
1344 local_int_28 += (8 << local_int_8);
1345 }
1346 }
1347 offset += (n - 2);
1348 } else {
1349 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1350 while (offset >= (local_int_10 - 1)) {
1351 offset += (1 - (local_int_10 - 1));
1352 local_int_4 += local_int_10;
1353 local_int_28 += (1 << local_int_8);
1354 }
1355 }
1356
1357 if (local_int_4 >= q->group_size)
1358 return;
1359
1360 local_int_14 = (offset >> local_int_8);
1361 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1362 return;
1363
1364 if (q->nb_channels > 1) {
1365 channel = get_bits1(gb);
1366 stereo = get_bits1(gb);
1367 } else {
1368 channel = 0;
1369 stereo = 0;
1370 }
1371
1372 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1373 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1374 exp = (exp < 0) ? 0 : exp;
1375
1376 phase = get_bits(gb, 3);
1377 stereo_exp = 0;
1378 stereo_phase = 0;
1379
1380 if (stereo) {
1381 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1382 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1383 if (stereo_phase < 0)
1384 stereo_phase += 8;
1385 }
1386
1387 if (q->frequency_range > (local_int_14 + 1)) {
1388 int sub_packet = (local_int_20 + local_int_28);
1389
1390 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1391 if (stereo)
1392 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1393 }
1394
1395 offset++;
1396 }
1397 }
1398
1399
1400 static void qdm2_decode_fft_packets (QDM2Context *q)
1401 {
1402 int i, j, min, max, value, type, unknown_flag;
1403 GetBitContext gb;
1404
1405 if (q->sub_packet_list_B[0].packet == NULL)
1406 return;
1407
1408 /* reset minimum indexes for FFT coefficients */
1409 q->fft_coefs_index = 0;
1410 for (i=0; i < 5; i++)
1411 q->fft_coefs_min_index[i] = -1;
1412
1413 /* process subpackets ordered by type, largest type first */
1414 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1415 QDM2SubPacket *packet= NULL;
1416
1417 /* find subpacket with largest type less than max */
1418 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1419 value = q->sub_packet_list_B[j].packet->type;
1420 if (value > min && value < max) {
1421 min = value;
1422 packet = q->sub_packet_list_B[j].packet;
1423 }
1424 }
1425
1426 max = min;
1427
1428 /* check for errors (?) */
1429 if (!packet)
1430 return;
1431
1432 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1433 return;
1434
1435 /* decode FFT tones */
1436 init_get_bits (&gb, packet->data, packet->size*8);
1437
1438 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1439 unknown_flag = 1;
1440 else
1441 unknown_flag = 0;
1442
1443 type = packet->type;
1444
1445 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1446 int duration = q->sub_sampling + 5 - (type & 15);
1447
1448 if (duration >= 0 && duration < 4)
1449 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1450 } else if (type == 31) {
1451 for (j=0; j < 4; j++)
1452 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1453 } else if (type == 46) {
1454 for (j=0; j < 6; j++)
1455 q->fft_level_exp[j] = get_bits(&gb, 6);
1456 for (j=0; j < 4; j++)
1457 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1458 }
1459 } // Loop on B packets
1460
1461 /* calculate maximum indexes for FFT coefficients */
1462 for (i = 0, j = -1; i < 5; i++)
1463 if (q->fft_coefs_min_index[i] >= 0) {
1464 if (j >= 0)
1465 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1466 j = i;
1467 }
1468 if (j >= 0)
1469 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1470 }
1471
1472
1473 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1474 {
1475 float level, f[6];
1476 int i;
1477 QDM2Complex c;
1478 const double iscale = 2.0*M_PI / 512.0;
1479
1480 tone->phase += tone->phase_shift;
1481
1482 /* calculate current level (maximum amplitude) of tone */
1483 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1484 c.im = level * sin(tone->phase*iscale);
1485 c.re = level * cos(tone->phase*iscale);
1486
1487 /* generate FFT coefficients for tone */
1488 if (tone->duration >= 3 || tone->cutoff >= 3) {
1489 tone->complex[0].im += c.im;
1490 tone->complex[0].re += c.re;
1491 tone->complex[1].im -= c.im;
1492 tone->complex[1].re -= c.re;
1493 } else {
1494 f[1] = -tone->table[4];
1495 f[0] = tone->table[3] - tone->table[0];
1496 f[2] = 1.0 - tone->table[2] - tone->table[3];
1497 f[3] = tone->table[1] + tone->table[4] - 1.0;
1498 f[4] = tone->table[0] - tone->table[1];
1499 f[5] = tone->table[2];
1500 for (i = 0; i < 2; i++) {
1501 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1502 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1503 }
1504 for (i = 0; i < 4; i++) {
1505 tone->complex[i].re += c.re * f[i+2];
1506 tone->complex[i].im += c.im * f[i+2];
1507 }
1508 }
1509
1510 /* copy the tone if it has not yet died out */
1511 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1512 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1513 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1514 }
1515 }
1516
1517
1518 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1519 {
1520 int i, j, ch;
1521 const double iscale = 0.25 * M_PI;
1522
1523 for (ch = 0; ch < q->channels; ch++) {
1524 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1525 }
1526
1527
1528 /* apply FFT tones with duration 4 (1 FFT period) */
1529 if (q->fft_coefs_min_index[4] >= 0)
1530 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1531 float level;
1532 QDM2Complex c;
1533
1534 if (q->fft_coefs[i].sub_packet != sub_packet)
1535 break;
1536
1537 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1538 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];
1539
1540 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1541 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1542 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1543 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1544 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1545 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1546 }
1547
1548 /* generate existing FFT tones */
1549 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1550 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1551 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1552 }
1553
1554 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1555 for (i = 0; i < 4; i++)
1556 if (q->fft_coefs_min_index[i] >= 0) {
1557 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1558 int offset, four_i;
1559 FFTTone tone;
1560
1561 if (q->fft_coefs[j].sub_packet != sub_packet)
1562 break;
1563
1564 four_i = (4 - i);
1565 offset = q->fft_coefs[j].offset >> four_i;
1566 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1567
1568 if (offset < q->frequency_range) {
1569 if (offset < 2)
1570 tone.cutoff = offset;
1571 else
1572 tone.cutoff = (offset >= 60) ? 3 : 2;
1573
1574 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];
1575 tone.complex = &q->fft.complex[ch][offset];
1576 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1577 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1578 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1579 tone.duration = i;
1580 tone.time_index = 0;
1581
1582 qdm2_fft_generate_tone(q, &tone);
1583 }
1584 }
1585 q->fft_coefs_min_index[i] = j;
1586 }
1587 }
1588
1589
1590 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1591 {
1592 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1593 float *out = q->output_buffer + channel;
1594 int i;
1595 q->fft.complex[channel][0].re *= 2.0f;
1596 q->fft.complex[channel][0].im = 0.0f;
1597 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1598 /* add samples to output buffer */
1599 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1600 out[0] += q->fft.complex[channel][i].re * gain;
1601 out[q->channels] += q->fft.complex[channel][i].im * gain;
1602 out += 2 * q->channels;
1603 }
1604 }
1605
1606
1607 /**
1608 * @param q context
1609 * @param index subpacket number
1610 */
1611 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1612 {
1613 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1614
1615 /* copy sb_samples */
1616 sb_used = QDM2_SB_USED(q->sub_sampling);
1617
1618 for (ch = 0; ch < q->channels; ch++)
1619 for (i = 0; i < 8; i++)
1620 for (k=sb_used; k < SBLIMIT; k++)
1621 q->sb_samples[ch][(8 * index) + i][k] = 0;
1622
1623 for (ch = 0; ch < q->nb_channels; ch++) {
1624 float *samples_ptr = q->samples + ch;
1625
1626 for (i = 0; i < 8; i++) {
1627 ff_mpa_synth_filter_float(&q->mpadsp,
1628 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1629 ff_mpa_synth_window_float, &dither_state,
1630 samples_ptr, q->nb_channels,
1631 q->sb_samples[ch][(8 * index) + i]);
1632 samples_ptr += 32 * q->nb_channels;
1633 }
1634 }
1635
1636 /* add samples to output buffer */
1637 sub_sampling = (4 >> q->sub_sampling);
1638
1639 for (ch = 0; ch < q->channels; ch++)
1640 for (i = 0; i < q->frame_size; i++)
1641 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1642 }
1643
1644
1645 /**
1646 * Init static data (does not depend on specific file)
1647 *
1648 * @param q context
1649 */
1650 static av_cold void qdm2_init(QDM2Context *q) {
1651 static int initialized = 0;
1652
1653 if (initialized != 0)
1654 return;
1655 initialized = 1;
1656
1657 qdm2_init_vlc();
1658 ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1659 softclip_table_init();
1660 rnd_table_init();
1661 init_noise_samples();
1662
1663 av_log(NULL, AV_LOG_DEBUG, "init done\n");
1664 }
1665
1666
1667 /**
1668 * Init parameters from codec extradata
1669 */
1670 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1671 {
1672 QDM2Context *s = avctx->priv_data;
1673 uint8_t *extradata;
1674 int extradata_size;
1675 int tmp_val, tmp, size;
1676
1677 /* extradata parsing
1678
1679 Structure:
1680 wave {
1681 frma (QDM2)
1682 QDCA
1683 QDCP
1684 }
1685
1686 32 size (including this field)
1687 32 tag (=frma)
1688 32 type (=QDM2 or QDMC)
1689
1690 32 size (including this field, in bytes)
1691 32 tag (=QDCA) // maybe mandatory parameters
1692 32 unknown (=1)
1693 32 channels (=2)
1694 32 samplerate (=44100)
1695 32 bitrate (=96000)
1696 32 block size (=4096)
1697 32 frame size (=256) (for one channel)
1698 32 packet size (=1300)
1699
1700 32 size (including this field, in bytes)
1701 32 tag (=QDCP) // maybe some tuneable parameters
1702 32 float1 (=1.0)
1703 32 zero ?
1704 32 float2 (=1.0)
1705 32 float3 (=1.0)
1706 32 unknown (27)
1707 32 unknown (8)
1708 32 zero ?
1709 */
1710
1711 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1712 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1713 return -1;
1714 }
1715
1716 extradata = avctx->extradata;
1717 extradata_size = avctx->extradata_size;
1718
1719 while (extradata_size > 7) {
1720 if (!memcmp(extradata, "frmaQDM", 7))
1721 break;
1722 extradata++;
1723 extradata_size--;
1724 }
1725
1726 if (extradata_size < 12) {
1727 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1728 extradata_size);
1729 return -1;
1730 }
1731
1732 if (memcmp(extradata, "frmaQDM", 7)) {
1733 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1734 return -1;
1735 }
1736
1737 if (extradata[7] == 'C') {
1738 // s->is_qdmc = 1;
1739 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1740 return -1;
1741 }
1742
1743 extradata += 8;
1744 extradata_size -= 8;
1745
1746 size = AV_RB32(extradata);
1747
1748 if(size > extradata_size){
1749 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1750 extradata_size, size);
1751 return -1;
1752 }
1753
1754 extradata += 4;
1755 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1756 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1757 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1758 return -1;
1759 }
1760
1761 extradata += 8;
1762
1763 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1764 extradata += 4;
1765 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS)
1766 return AVERROR_INVALIDDATA;
1767 avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1768 AV_CH_LAYOUT_MONO;
1769
1770 avctx->sample_rate = AV_RB32(extradata);
1771 extradata += 4;
1772
1773 avctx->bit_rate = AV_RB32(extradata);
1774 extradata += 4;
1775
1776 s->group_size = AV_RB32(extradata);
1777 extradata += 4;
1778
1779 s->fft_size = AV_RB32(extradata);
1780 extradata += 4;
1781
1782 s->checksum_size = AV_RB32(extradata);
1783 if (s->checksum_size >= 1U << 28) {
1784 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1785 return AVERROR_INVALIDDATA;
1786 }
1787
1788 s->fft_order = av_log2(s->fft_size) + 1;
1789
1790 // something like max decodable tones
1791 s->group_order = av_log2(s->group_size) + 1;
1792 s->frame_size = s->group_size / 16; // 16 iterations per super block
1793 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1794 return AVERROR_INVALIDDATA;
1795
1796 s->sub_sampling = s->fft_order - 7;
1797 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1798
1799 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1800 case 0: tmp = 40; break;
1801 case 1: tmp = 48; break;
1802 case 2: tmp = 56; break;
1803 case 3: tmp = 72; break;
1804 case 4: tmp = 80; break;
1805 case 5: tmp = 100;break;
1806 default: tmp=s->sub_sampling; break;
1807 }
1808 tmp_val = 0;
1809 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1810 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1811 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1812 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1813 s->cm_table_select = tmp_val;
1814
1815 if (s->sub_sampling == 0)
1816 tmp = 7999;
1817 else
1818 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1819 /*
1820 0: 7999 -> 0
1821 1: 20000 -> 2
1822 2: 28000 -> 2
1823 */
1824 if (tmp < 8000)
1825 s->coeff_per_sb_select = 0;
1826 else if (tmp <= 16000)
1827 s->coeff_per_sb_select = 1;
1828 else
1829 s->coeff_per_sb_select = 2;
1830
1831 // Fail on unknown fft order
1832 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1833 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1834 return -1;
1835 }
1836
1837 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1838 ff_mpadsp_init(&s->mpadsp);
1839
1840 qdm2_init(s);
1841
1842 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1843
1844 avcodec_get_frame_defaults(&s->frame);
1845 avctx->coded_frame = &s->frame;
1846
1847 return 0;
1848 }
1849
1850
1851 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1852 {
1853 QDM2Context *s = avctx->priv_data;
1854
1855 ff_rdft_end(&s->rdft_ctx);
1856
1857 return 0;
1858 }
1859
1860
1861 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1862 {
1863 int ch, i;
1864 const int frame_size = (q->frame_size * q->channels);
1865
1866 /* select input buffer */
1867 q->compressed_data = in;
1868 q->compressed_size = q->checksum_size;
1869
1870 /* copy old block, clear new block of output samples */
1871 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1872 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1873
1874 /* decode block of QDM2 compressed data */
1875 if (q->sub_packet == 0) {
1876 q->has_errors = 0; // zero it for a new super block
1877 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1878 qdm2_decode_super_block(q);
1879 }
1880
1881 /* parse subpackets */
1882 if (!q->has_errors) {
1883 if (q->sub_packet == 2)
1884 qdm2_decode_fft_packets(q);
1885
1886 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1887 }
1888
1889 /* sound synthesis stage 1 (FFT) */
1890 for (ch = 0; ch < q->channels; ch++) {
1891 qdm2_calculate_fft(q, ch, q->sub_packet);
1892
1893 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1894 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1895 return -1;
1896 }
1897 }
1898
1899 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1900 if (!q->has_errors && q->do_synth_filter)
1901 qdm2_synthesis_filter(q, q->sub_packet);
1902
1903 q->sub_packet = (q->sub_packet + 1) % 16;
1904
1905 /* clip and convert output float[] to 16bit signed samples */
1906 for (i = 0; i < frame_size; i++) {
1907 int value = (int)q->output_buffer[i];
1908
1909 if (value > SOFTCLIP_THRESHOLD)
1910 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1911 else if (value < -SOFTCLIP_THRESHOLD)
1912 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1913
1914 out[i] = value;
1915 }
1916
1917 return 0;
1918 }
1919
1920
1921 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1922 int *got_frame_ptr, AVPacket *avpkt)
1923 {
1924 const uint8_t *buf = avpkt->data;
1925 int buf_size = avpkt->size;
1926 QDM2Context *s = avctx->priv_data;
1927 int16_t *out;
1928 int i, ret;
1929
1930 if(!buf)
1931 return 0;
1932 if(buf_size < s->checksum_size)
1933 return -1;
1934
1935 /* get output buffer */
1936 s->frame.nb_samples = 16 * s->frame_size;
1937 if ((ret = ff_get_buffer(avctx, &s->frame)) < 0) {
1938 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1939 return ret;
1940 }
1941 out = (int16_t *)s->frame.data[0];
1942
1943 for (i = 0; i < 16; i++) {
1944 if (qdm2_decode(s, buf, out) < 0)
1945 return -1;
1946 out += s->channels * s->frame_size;
1947 }
1948
1949 *got_frame_ptr = 1;
1950 *(AVFrame *)data = s->frame;
1951
1952 return s->checksum_size;
1953 }
1954
1955 AVCodec ff_qdm2_decoder =
1956 {
1957 .name = "qdm2",
1958 .type = AVMEDIA_TYPE_AUDIO,
1959 .id = AV_CODEC_ID_QDM2,
1960 .priv_data_size = sizeof(QDM2Context),
1961 .init = qdm2_decode_init,
1962 .close = qdm2_decode_close,
1963 .decode = qdm2_decode_frame,
1964 .capabilities = CODEC_CAP_DR1,
1965 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
1966 };
A couple of patches to make building with MPlayer easier