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Imported Debian version 2.4.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / qdm2.c
CommitLineData
2ba45a60
DM		 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
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 "internal.h"
43#include "rdft.h"
44#include "mpegaudiodsp.h"
45#include "mpegaudio.h"
46
47#include "qdm2data.h"
48#include "qdm2_tablegen.h"
49
50#undef NDEBUG
51#include <assert.h>
52
53
54#define QDM2_LIST_ADD(list, size, packet) \
55do { \
56 if (size > 0) { \
57 list[size - 1].next = &list[size]; \
58 } \
59 list[size].packet = packet; \
60 list[size].next = NULL; \
61 size++; \
62} while(0)
63
64// Result is 8, 16 or 30
65#define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
66
67#define FIX_NOISE_IDX(noise_idx) \
68 if ((noise_idx) >= 3840) \
69 (noise_idx) -= 3840; \
70
71#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
72
73#define SAMPLES_NEEDED \
74 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
75
76#define SAMPLES_NEEDED_2(why) \
77 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
78
79#define QDM2_MAX_FRAME_SIZE 512
80
81typedef int8_t sb_int8_array[2][30][64];
82
83/**
84 * Subpacket
85 */
86typedef 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 */
95typedef struct QDM2SubPNode {
96 QDM2SubPacket *packet; ///< packet
97 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
98} QDM2SubPNode;
99
100typedef struct {
101 float re;
102 float im;
103} QDM2Complex;
104
105typedef 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
116typedef 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
124typedef struct {
125 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
126} QDM2FFT;
127
128/**
129 * QDM2 decoder context
130 */
131typedef 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 frame_size; ///< size of data frame
143 int frequency_range;
144 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
145 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
146 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
147
148 /// Packets and packet lists
149 QDM2SubPacket sub_packets[16]; ///< the packets themselves
150 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
151 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
152 int sub_packets_B; ///< number of packets on 'B' list
153 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
154 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
155
156 /// FFT and tones
157 FFTTone fft_tones[1000];
158 int fft_tone_start;
159 int fft_tone_end;
160 FFTCoefficient fft_coefs[1000];
161 int fft_coefs_index;
162 int fft_coefs_min_index[5];
163 int fft_coefs_max_index[5];
164 int fft_level_exp[6];
165 RDFTContext rdft_ctx;
166 QDM2FFT fft;
167
168 /// I/O data
169 const uint8_t *compressed_data;
170 int compressed_size;
171 float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
172
173 /// Synthesis filter
174 MPADSPContext mpadsp;
175 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
176 int synth_buf_offset[MPA_MAX_CHANNELS];
177 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
178 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
179
180 /// Mixed temporary data used in decoding
181 float tone_level[MPA_MAX_CHANNELS][30][64];
182 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
183 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
184 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
185 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
186 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
187 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
188 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
189 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
190
191 // Flags
192 int has_errors; ///< packet has errors
193 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
194 int do_synth_filter; ///< used to perform or skip synthesis filter
195
196 int sub_packet;
197 int noise_idx; ///< index for dithering noise table
198} QDM2Context;
199
200
201static VLC vlc_tab_level;
202static VLC vlc_tab_diff;
203static VLC vlc_tab_run;
204static VLC fft_level_exp_alt_vlc;
205static VLC fft_level_exp_vlc;
206static VLC fft_stereo_exp_vlc;
207static VLC fft_stereo_phase_vlc;
208static VLC vlc_tab_tone_level_idx_hi1;
209static VLC vlc_tab_tone_level_idx_mid;
210static VLC vlc_tab_tone_level_idx_hi2;
211static VLC vlc_tab_type30;
212static VLC vlc_tab_type34;
213static VLC vlc_tab_fft_tone_offset[5];
214
215static const uint16_t qdm2_vlc_offs[] = {
216 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
217};
218
219static const int switchtable[23] = {
220 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
221};
222
223static av_cold void qdm2_init_vlc(void)
224{
225 static VLC_TYPE qdm2_table[3838][2];
226
227 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
228 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
229 init_vlc(&vlc_tab_level, 8, 24,
230 vlc_tab_level_huffbits, 1, 1,
231 vlc_tab_level_huffcodes, 2, 2,
232 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
233
234 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
235 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
236 init_vlc(&vlc_tab_diff, 8, 37,
237 vlc_tab_diff_huffbits, 1, 1,
238 vlc_tab_diff_huffcodes, 2, 2,
239 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,
246 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
247
248 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
249 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] -
250 qdm2_vlc_offs[3];
251 init_vlc(&fft_level_exp_alt_vlc, 8, 28,
252 fft_level_exp_alt_huffbits, 1, 1,
253 fft_level_exp_alt_huffcodes, 2, 2,
254 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
255
256 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
257 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
258 init_vlc(&fft_level_exp_vlc, 8, 20,
259 fft_level_exp_huffbits, 1, 1,
260 fft_level_exp_huffcodes, 2, 2,
261 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
262
263 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
264 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] -
265 qdm2_vlc_offs[5];
266 init_vlc(&fft_stereo_exp_vlc, 6, 7,
267 fft_stereo_exp_huffbits, 1, 1,
268 fft_stereo_exp_huffcodes, 1, 1,
269 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
270
271 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
272 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] -
273 qdm2_vlc_offs[6];
274 init_vlc(&fft_stereo_phase_vlc, 6, 9,
275 fft_stereo_phase_huffbits, 1, 1,
276 fft_stereo_phase_huffcodes, 1, 1,
277 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
278
279 vlc_tab_tone_level_idx_hi1.table =
280 &qdm2_table[qdm2_vlc_offs[7]];
281 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] -
282 qdm2_vlc_offs[7];
283 init_vlc(&vlc_tab_tone_level_idx_hi1, 8, 20,
284 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
285 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2,
286 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
287
288 vlc_tab_tone_level_idx_mid.table =
289 &qdm2_table[qdm2_vlc_offs[8]];
290 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] -
291 qdm2_vlc_offs[8];
292 init_vlc(&vlc_tab_tone_level_idx_mid, 8, 24,
293 vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
294 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2,
295 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
296
297 vlc_tab_tone_level_idx_hi2.table =
298 &qdm2_table[qdm2_vlc_offs[9]];
299 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] -
300 qdm2_vlc_offs[9];
301 init_vlc(&vlc_tab_tone_level_idx_hi2, 8, 24,
302 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
303 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2,
304 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
305
306 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
307 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
308 init_vlc(&vlc_tab_type30, 6, 9,
309 vlc_tab_type30_huffbits, 1, 1,
310 vlc_tab_type30_huffcodes, 1, 1,
311 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
312
313 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
314 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
315 init_vlc(&vlc_tab_type34, 5, 10,
316 vlc_tab_type34_huffbits, 1, 1,
317 vlc_tab_type34_huffcodes, 1, 1,
318 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
319
320 vlc_tab_fft_tone_offset[0].table =
321 &qdm2_table[qdm2_vlc_offs[12]];
322 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] -
323 qdm2_vlc_offs[12];
324 init_vlc(&vlc_tab_fft_tone_offset[0], 8, 23,
325 vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
326 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2,
327 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
328
329 vlc_tab_fft_tone_offset[1].table =
330 &qdm2_table[qdm2_vlc_offs[13]];
331 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] -
332 qdm2_vlc_offs[13];
333 init_vlc(&vlc_tab_fft_tone_offset[1], 8, 28,
334 vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
335 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2,
336 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
337
338 vlc_tab_fft_tone_offset[2].table =
339 &qdm2_table[qdm2_vlc_offs[14]];
340 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] -
341 qdm2_vlc_offs[14];
342 init_vlc(&vlc_tab_fft_tone_offset[2], 8, 32,
343 vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
344 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2,
345 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
346
347 vlc_tab_fft_tone_offset[3].table =
348 &qdm2_table[qdm2_vlc_offs[15]];
349 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] -
350 qdm2_vlc_offs[15];
351 init_vlc(&vlc_tab_fft_tone_offset[3], 8, 35,
352 vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
353 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2,
354 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
355
356 vlc_tab_fft_tone_offset[4].table =
357 &qdm2_table[qdm2_vlc_offs[16]];
358 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] -
359 qdm2_vlc_offs[16];
360 init_vlc(&vlc_tab_fft_tone_offset[4], 8, 38,
361 vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
362 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2,
363 INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
364}
365
366static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth)
367{
368 int value;
369
370 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
371
372 /* stage-2, 3 bits exponent escape sequence */
373 if (value-- == 0)
374 value = get_bits(gb, get_bits(gb, 3) + 1);
375
376 /* stage-3, optional */
377 if (flag) {
378 int tmp;
379
380 if (value >= 60) {
381 av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
382 return 0;
383 }
384
385 tmp= vlc_stage3_values[value];
386
387 if ((value & ~3) > 0)
388 tmp += get_bits(gb, (value >> 2));
389 value = tmp;
390 }
391
392 return value;
393}
394
395static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth)
396{
397 int value = qdm2_get_vlc(gb, vlc, 0, depth);
398
399 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
400}
401
402/**
403 * QDM2 checksum
404 *
405 * @param data pointer to data to be checksum'ed
406 * @param length data length
407 * @param value checksum value
408 *
409 * @return 0 if checksum is OK
410 */
411static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
412{
413 int i;
414
415 for (i = 0; i < length; i++)
416 value -= data[i];
417
418 return (uint16_t)(value & 0xffff);
419}
420
421/**
422 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
423 *
424 * @param gb bitreader context
425 * @param sub_packet packet under analysis
426 */
427static void qdm2_decode_sub_packet_header(GetBitContext *gb,
428 QDM2SubPacket *sub_packet)
429{
430 sub_packet->type = get_bits(gb, 8);
431
432 if (sub_packet->type == 0) {
433 sub_packet->size = 0;
434 sub_packet->data = NULL;
435 } else {
436 sub_packet->size = get_bits(gb, 8);
437
438 if (sub_packet->type & 0x80) {
439 sub_packet->size <<= 8;
440 sub_packet->size |= get_bits(gb, 8);
441 sub_packet->type &= 0x7f;
442 }
443
444 if (sub_packet->type == 0x7f)
445 sub_packet->type |= (get_bits(gb, 8) << 8);
446
447 // FIXME: this depends on bitreader-internal data
448 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
449 }
450
451 av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
452 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
453}
454
455/**
456 * Return node pointer to first packet of requested type in list.
457 *
458 * @param list list of subpackets to be scanned
459 * @param type type of searched subpacket
460 * @return node pointer for subpacket if found, else NULL
461 */
462static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
463 int type)
464{
465 while (list && list->packet) {
466 if (list->packet->type == type)
467 return list;
468 list = list->next;
469 }
470 return NULL;
471}
472
473/**
474 * Replace 8 elements with their average value.
475 * Called by qdm2_decode_superblock before starting subblock decoding.
476 *
477 * @param q context
478 */
479static void average_quantized_coeffs(QDM2Context *q)
480{
481 int i, j, n, ch, sum;
482
483 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
484
485 for (ch = 0; ch < q->nb_channels; ch++)
486 for (i = 0; i < n; i++) {
487 sum = 0;
488
489 for (j = 0; j < 8; j++)
490 sum += q->quantized_coeffs[ch][i][j];
491
492 sum /= 8;
493 if (sum > 0)
494 sum--;
495
496 for (j = 0; j < 8; j++)
497 q->quantized_coeffs[ch][i][j] = sum;
498 }
499}
500
501/**
502 * Build subband samples with noise weighted by q->tone_level.
503 * Called by synthfilt_build_sb_samples.
504 *
505 * @param q context
506 * @param sb subband index
507 */
508static void build_sb_samples_from_noise(QDM2Context *q, int sb)
509{
510 int ch, j;
511
512 FIX_NOISE_IDX(q->noise_idx);
513
514 if (!q->nb_channels)
515 return;
516
517 for (ch = 0; ch < q->nb_channels; ch++) {
518 for (j = 0; j < 64; j++) {
519 q->sb_samples[ch][j * 2][sb] =
520 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
521 q->sb_samples[ch][j * 2 + 1][sb] =
522 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
523 }
524 }
525}
526
527/**
528 * Called while processing data from subpackets 11 and 12.
529 * Used after making changes to coding_method array.
530 *
531 * @param sb subband index
532 * @param channels number of channels
533 * @param coding_method q->coding_method[0][0][0]
534 */
535static int fix_coding_method_array(int sb, int channels,
536 sb_int8_array coding_method)
537{
538 int j, k;
539 int ch;
540 int run, case_val;
541
542 for (ch = 0; ch < channels; ch++) {
543 for (j = 0; j < 64; ) {
544 if (coding_method[ch][sb][j] < 8)
545 return -1;
546 if ((coding_method[ch][sb][j] - 8) > 22) {
547 run = 1;
548 case_val = 8;
549 } else {
550 switch (switchtable[coding_method[ch][sb][j] - 8]) {
551 case 0: run = 10;
552 case_val = 10;
553 break;
554 case 1: run = 1;
555 case_val = 16;
556 break;
557 case 2: run = 5;
558 case_val = 24;
559 break;
560 case 3: run = 3;
561 case_val = 30;
562 break;
563 case 4: run = 1;
564 case_val = 30;
565 break;
566 case 5: run = 1;
567 case_val = 8;
568 break;
569 default: run = 1;
570 case_val = 8;
571 break;
572 }
573 }
574 for (k = 0; k < run; k++) {
575 if (j + k < 128) {
576 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) {
577 if (k > 0) {
578 SAMPLES_NEEDED
579 //not debugged, almost never used
580 memset(&coding_method[ch][sb][j + k], case_val,
581 k *sizeof(int8_t));
582 memset(&coding_method[ch][sb][j + k], case_val,
583 3 * sizeof(int8_t));
584 }
585 }
586 }
587 }
588 j += run;
589 }
590 }
591 return 0;
592}
593
594/**
595 * Related to synthesis filter
596 * Called by process_subpacket_10
597 *
598 * @param q context
599 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
600 */
601static void fill_tone_level_array(QDM2Context *q, int flag)
602{
603 int i, sb, ch, sb_used;
604 int tmp, tab;
605
606 for (ch = 0; ch < q->nb_channels; ch++)
607 for (sb = 0; sb < 30; sb++)
608 for (i = 0; i < 8; i++) {
609 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
610 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
611 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
612 else
613 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
614 if(tmp < 0)
615 tmp += 0xff;
616 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
617 }
618
619 sb_used = QDM2_SB_USED(q->sub_sampling);
620
621 if ((q->superblocktype_2_3 != 0) && !flag) {
622 for (sb = 0; sb < sb_used; sb++)
623 for (ch = 0; ch < q->nb_channels; ch++)
624 for (i = 0; i < 64; i++) {
625 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
626 if (q->tone_level_idx[ch][sb][i] < 0)
627 q->tone_level[ch][sb][i] = 0;
628 else
629 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
630 }
631 } else {
632 tab = q->superblocktype_2_3 ? 0 : 1;
633 for (sb = 0; sb < sb_used; sb++) {
634 if ((sb >= 4) && (sb <= 23)) {
635 for (ch = 0; ch < q->nb_channels; ch++)
636 for (i = 0; i < 64; i++) {
637 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
638 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
639 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
640 q->tone_level_idx_hi2[ch][sb - 4];
641 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
642 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
643 q->tone_level[ch][sb][i] = 0;
644 else
645 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
646 }
647 } else {
648 if (sb > 4) {
649 for (ch = 0; ch < q->nb_channels; ch++)
650 for (i = 0; i < 64; i++) {
651 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
652 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
653 q->tone_level_idx_hi2[ch][sb - 4];
654 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
655 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
656 q->tone_level[ch][sb][i] = 0;
657 else
658 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
659 }
660 } else {
661 for (ch = 0; ch < q->nb_channels; ch++)
662 for (i = 0; i < 64; i++) {
663 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
664 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
665 q->tone_level[ch][sb][i] = 0;
666 else
667 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
668 }
669 }
670 }
671 }
672 }
673}
674
675/**
676 * Related to synthesis filter
677 * Called by process_subpacket_11
678 * c is built with data from subpacket 11
679 * Most of this function is used only if superblock_type_2_3 == 0,
680 * never seen it in samples.
681 *
682 * @param tone_level_idx
683 * @param tone_level_idx_temp
684 * @param coding_method q->coding_method[0][0][0]
685 * @param nb_channels number of channels
686 * @param c coming from subpacket 11, passed as 8*c
687 * @param superblocktype_2_3 flag based on superblock packet type
688 * @param cm_table_select q->cm_table_select
689 */
690static void fill_coding_method_array(sb_int8_array tone_level_idx,
691 sb_int8_array tone_level_idx_temp,
692 sb_int8_array coding_method,
693 int nb_channels,
694 int c, int superblocktype_2_3,
695 int cm_table_select)
696{
697 int ch, sb, j;
698 int tmp, acc, esp_40, comp;
699 int add1, add2, add3, add4;
700 int64_t multres;
701
702 if (!superblocktype_2_3) {
703 /* This case is untested, no samples available */
704 avpriv_request_sample(NULL, "!superblocktype_2_3");
705 return;
706 for (ch = 0; ch < nb_channels; ch++)
707 for (sb = 0; sb < 30; sb++) {
708 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
709 add1 = tone_level_idx[ch][sb][j] - 10;
710 if (add1 < 0)
711 add1 = 0;
712 add2 = add3 = add4 = 0;
713 if (sb > 1) {
714 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
715 if (add2 < 0)
716 add2 = 0;
717 }
718 if (sb > 0) {
719 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
720 if (add3 < 0)
721 add3 = 0;
722 }
723 if (sb < 29) {
724 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
725 if (add4 < 0)
726 add4 = 0;
727 }
728 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
729 if (tmp < 0)
730 tmp = 0;
731 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
732 }
733 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
734 }
735 acc = 0;
736 for (ch = 0; ch < nb_channels; ch++)
737 for (sb = 0; sb < 30; sb++)
738 for (j = 0; j < 64; j++)
739 acc += tone_level_idx_temp[ch][sb][j];
740
741 multres = 0x66666667LL * (acc * 10);
742 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
743 for (ch = 0; ch < nb_channels; ch++)
744 for (sb = 0; sb < 30; sb++)
745 for (j = 0; j < 64; j++) {
746 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
747 if (comp < 0)
748 comp += 0xff;
749 comp /= 256; // signed shift
750 switch(sb) {
751 case 0:
752 if (comp < 30)
753 comp = 30;
754 comp += 15;
755 break;
756 case 1:
757 if (comp < 24)
758 comp = 24;
759 comp += 10;
760 break;
761 case 2:
762 case 3:
763 case 4:
764 if (comp < 16)
765 comp = 16;
766 }
767 if (comp <= 5)
768 tmp = 0;
769 else if (comp <= 10)
770 tmp = 10;
771 else if (comp <= 16)
772 tmp = 16;
773 else if (comp <= 24)
774 tmp = -1;
775 else
776 tmp = 0;
777 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
778 }
779 for (sb = 0; sb < 30; sb++)
780 fix_coding_method_array(sb, nb_channels, coding_method);
781 for (ch = 0; ch < nb_channels; ch++)
782 for (sb = 0; sb < 30; sb++)
783 for (j = 0; j < 64; j++)
784 if (sb >= 10) {
785 if (coding_method[ch][sb][j] < 10)
786 coding_method[ch][sb][j] = 10;
787 } else {
788 if (sb >= 2) {
789 if (coding_method[ch][sb][j] < 16)
790 coding_method[ch][sb][j] = 16;
791 } else {
792 if (coding_method[ch][sb][j] < 30)
793 coding_method[ch][sb][j] = 30;
794 }
795 }
796 } else { // superblocktype_2_3 != 0
797 for (ch = 0; ch < nb_channels; ch++)
798 for (sb = 0; sb < 30; sb++)
799 for (j = 0; j < 64; j++)
800 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
801 }
802}
803
804/**
805 *
806 * Called by process_subpacket_11 to process more data from subpacket 11
807 * with sb 0-8.
808 * Called by process_subpacket_12 to process data from subpacket 12 with
809 * sb 8-sb_used.
810 *
811 * @param q context
812 * @param gb bitreader context
813 * @param length packet length in bits
814 * @param sb_min lower subband processed (sb_min included)
815 * @param sb_max higher subband processed (sb_max excluded)
816 */
817static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
818 int length, int sb_min, int sb_max)
819{
820 int sb, j, k, n, ch, run, channels;
821 int joined_stereo, zero_encoding;
822 int type34_first;
823 float type34_div = 0;
824 float type34_predictor;
825 float samples[10];
826 int sign_bits[16] = {0};
827
828 if (length == 0) {
829 // If no data use noise
830 for (sb=sb_min; sb < sb_max; sb++)
831 build_sb_samples_from_noise(q, sb);
832
833 return 0;
834 }
835
836 for (sb = sb_min; sb < sb_max; sb++) {
837 channels = q->nb_channels;
838
839 if (q->nb_channels <= 1 || sb < 12)
840 joined_stereo = 0;
841 else if (sb >= 24)
842 joined_stereo = 1;
843 else
844 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
845
846 if (joined_stereo) {
847 if (get_bits_left(gb) >= 16)
848 for (j = 0; j < 16; j++)
849 sign_bits[j] = get_bits1(gb);
850
851 for (j = 0; j < 64; j++)
852 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
853 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
854
855 if (fix_coding_method_array(sb, q->nb_channels,
856 q->coding_method)) {
857 av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
858 build_sb_samples_from_noise(q, sb);
859 continue;
860 }
861 channels = 1;
862 }
863
864 for (ch = 0; ch < channels; ch++) {
865 FIX_NOISE_IDX(q->noise_idx);
866 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
867 type34_predictor = 0.0;
868 type34_first = 1;
869
870 for (j = 0; j < 128; ) {
871 switch (q->coding_method[ch][sb][j / 2]) {
872 case 8:
873 if (get_bits_left(gb) >= 10) {
874 if (zero_encoding) {
875 for (k = 0; k < 5; k++) {
876 if ((j + 2 * k) >= 128)
877 break;
878 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
879 }
880 } else {
881 n = get_bits(gb, 8);
882 if (n >= 243) {
883 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
884 return AVERROR_INVALIDDATA;
885 }
886
887 for (k = 0; k < 5; k++)
888 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
889 }
890 for (k = 0; k < 5; k++)
891 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
892 } else {
893 for (k = 0; k < 10; k++)
894 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
895 }
896 run = 10;
897 break;
898
899 case 10:
900 if (get_bits_left(gb) >= 1) {
901 float f = 0.81;
902
903 if (get_bits1(gb))
904 f = -f;
905 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
906 samples[0] = f;
907 } else {
908 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
909 }
910 run = 1;
911 break;
912
913 case 16:
914 if (get_bits_left(gb) >= 10) {
915 if (zero_encoding) {
916 for (k = 0; k < 5; k++) {
917 if ((j + k) >= 128)
918 break;
919 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
920 }
921 } else {
922 n = get_bits (gb, 8);
923 if (n >= 243) {
924 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
925 return AVERROR_INVALIDDATA;
926 }
927
928 for (k = 0; k < 5; k++)
929 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
930 }
931 } else {
932 for (k = 0; k < 5; k++)
933 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
934 }
935 run = 5;
936 break;
937
938 case 24:
939 if (get_bits_left(gb) >= 7) {
940 n = get_bits(gb, 7);
941 if (n >= 125) {
942 av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
943 return AVERROR_INVALIDDATA;
944 }
945
946 for (k = 0; k < 3; k++)
947 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
948 } else {
949 for (k = 0; k < 3; k++)
950 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
951 }
952 run = 3;
953 break;
954
955 case 30:
956 if (get_bits_left(gb) >= 4) {
957 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
958 if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
959 av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
960 return AVERROR_INVALIDDATA;
961 }
962 samples[0] = type30_dequant[index];
963 } else
964 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
965
966 run = 1;
967 break;
968
969 case 34:
970 if (get_bits_left(gb) >= 7) {
971 if (type34_first) {
972 type34_div = (float)(1 << get_bits(gb, 2));
973 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
974 type34_predictor = samples[0];
975 type34_first = 0;
976 } else {
977 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
978 if (index >= FF_ARRAY_ELEMS(type34_delta)) {
979 av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
980 return AVERROR_INVALIDDATA;
981 }
982 samples[0] = type34_delta[index] / type34_div + type34_predictor;
983 type34_predictor = samples[0];
984 }
985 } else {
986 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
987 }
988 run = 1;
989 break;
990
991 default:
992 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
993 run = 1;
994 break;
995 }
996
997 if (joined_stereo) {
998 for (k = 0; k < run && j + k < 128; k++) {
999 q->sb_samples[0][j + k][sb] =
1000 q->tone_level[0][sb][(j + k) / 2] * samples[k];
1001 if (q->nb_channels == 2) {
1002 if (sign_bits[(j + k) / 8])
1003 q->sb_samples[1][j + k][sb] =
1004 q->tone_level[1][sb][(j + k) / 2] * -samples[k];
1005 else
1006 q->sb_samples[1][j + k][sb] =
1007 q->tone_level[1][sb][(j + k) / 2] * samples[k];
1008 }
1009 }
1010 } else {
1011 for (k = 0; k < run; k++)
1012 if ((j + k) < 128)
1013 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
1014 }
1015
1016 j += run;
1017 } // j loop
1018 } // channel loop
1019 } // subband loop
1020 return 0;
1021}
1022
1023/**
1024 * Init the first element of a channel in quantized_coeffs with data
1025 * from packet 10 (quantized_coeffs[ch][0]).
1026 * This is similar to process_subpacket_9, but for a single channel
1027 * and for element [0]
1028 * same VLC tables as process_subpacket_9 are used.
1029 *
1030 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
1031 * @param gb bitreader context
1032 */
1033static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
1034 GetBitContext *gb)
1035{
1036 int i, k, run, level, diff;
1037
1038 if (get_bits_left(gb) < 16)
1039 return -1;
1040 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
1041
1042 quantized_coeffs[0] = level;
1043
1044 for (i = 0; i < 7; ) {
1045 if (get_bits_left(gb) < 16)
1046 return -1;
1047 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
1048
1049 if (i + run >= 8)
1050 return -1;
1051
1052 if (get_bits_left(gb) < 16)
1053 return -1;
1054 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
1055
1056 for (k = 1; k <= run; k++)
1057 quantized_coeffs[i + k] = (level + ((k * diff) / run));
1058
1059 level += diff;
1060 i += run;
1061 }
1062 return 0;
1063}
1064
1065/**
1066 * Related to synthesis filter, process data from packet 10
1067 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1068 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
1069 * data from packet 10
1070 *
1071 * @param q context
1072 * @param gb bitreader context
1073 */
1074static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
1075{
1076 int sb, j, k, n, ch;
1077
1078 for (ch = 0; ch < q->nb_channels; ch++) {
1079 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
1080
1081 if (get_bits_left(gb) < 16) {
1082 memset(q->quantized_coeffs[ch][0], 0, 8);
1083 break;
1084 }
1085 }
1086
1087 n = q->sub_sampling + 1;
1088
1089 for (sb = 0; sb < n; sb++)
1090 for (ch = 0; ch < q->nb_channels; ch++)
1091 for (j = 0; j < 8; j++) {
1092 if (get_bits_left(gb) < 1)
1093 break;
1094 if (get_bits1(gb)) {
1095 for (k=0; k < 8; k++) {
1096 if (get_bits_left(gb) < 16)
1097 break;
1098 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1099 }
1100 } else {
1101 for (k=0; k < 8; k++)
1102 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1103 }
1104 }
1105
1106 n = QDM2_SB_USED(q->sub_sampling) - 4;
1107
1108 for (sb = 0; sb < n; sb++)
1109 for (ch = 0; ch < q->nb_channels; ch++) {
1110 if (get_bits_left(gb) < 16)
1111 break;
1112 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1113 if (sb > 19)
1114 q->tone_level_idx_hi2[ch][sb] -= 16;
1115 else
1116 for (j = 0; j < 8; j++)
1117 q->tone_level_idx_mid[ch][sb][j] = -16;
1118 }
1119
1120 n = QDM2_SB_USED(q->sub_sampling) - 5;
1121
1122 for (sb = 0; sb < n; sb++)
1123 for (ch = 0; ch < q->nb_channels; ch++)
1124 for (j = 0; j < 8; j++) {
1125 if (get_bits_left(gb) < 16)
1126 break;
1127 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1128 }
1129}
1130
1131/**
1132 * Process subpacket 9, init quantized_coeffs with data from it
1133 *
1134 * @param q context
1135 * @param node pointer to node with packet
1136 */
1137static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
1138{
1139 GetBitContext gb;
1140 int i, j, k, n, ch, run, level, diff;
1141
1142 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1143
1144 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
1145
1146 for (i = 1; i < n; i++)
1147 for (ch = 0; ch < q->nb_channels; ch++) {
1148 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1149 q->quantized_coeffs[ch][i][0] = level;
1150
1151 for (j = 0; j < (8 - 1); ) {
1152 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1153 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1154
1155 if (j + run >= 8)
1156 return -1;
1157
1158 for (k = 1; k <= run; k++)
1159 q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
1160
1161 level += diff;
1162 j += run;
1163 }
1164 }
1165
1166 for (ch = 0; ch < q->nb_channels; ch++)
1167 for (i = 0; i < 8; i++)
1168 q->quantized_coeffs[ch][0][i] = 0;
1169
1170 return 0;
1171}
1172
1173/**
1174 * Process subpacket 10 if not null, else
1175 *
1176 * @param q context
1177 * @param node pointer to node with packet
1178 */
1179static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
1180{
1181 GetBitContext gb;
1182
1183 if (node) {
1184 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1185 init_tone_level_dequantization(q, &gb);
1186 fill_tone_level_array(q, 1);
1187 } else {
1188 fill_tone_level_array(q, 0);
1189 }
1190}
1191
1192/**
1193 * Process subpacket 11
1194 *
1195 * @param q context
1196 * @param node pointer to node with packet
1197 */
1198static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
1199{
1200 GetBitContext gb;
1201 int length = 0;
1202
1203 if (node) {
1204 length = node->packet->size * 8;
1205 init_get_bits(&gb, node->packet->data, length);
1206 }
1207
1208 if (length >= 32) {
1209 int c = get_bits(&gb, 13);
1210
1211 if (c > 3)
1212 fill_coding_method_array(q->tone_level_idx,
1213 q->tone_level_idx_temp, q->coding_method,
1214 q->nb_channels, 8 * c,
1215 q->superblocktype_2_3, q->cm_table_select);
1216 }
1217
1218 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1219}
1220
1221/**
1222 * Process subpacket 12
1223 *
1224 * @param q context
1225 * @param node pointer to node with packet
1226 */
1227static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
1228{
1229 GetBitContext gb;
1230 int length = 0;
1231
1232 if (node) {
1233 length = node->packet->size * 8;
1234 init_get_bits(&gb, node->packet->data, length);
1235 }
1236
1237 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1238}
1239
1240/**
1241 * Process new subpackets for synthesis filter
1242 *
1243 * @param q context
1244 * @param list list with synthesis filter packets (list D)
1245 */
1246static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
1247{
1248 QDM2SubPNode *nodes[4];
1249
1250 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1251 if (nodes[0])
1252 process_subpacket_9(q, nodes[0]);
1253
1254 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1255 if (nodes[1])
1256 process_subpacket_10(q, nodes[1]);
1257 else
1258 process_subpacket_10(q, NULL);
1259
1260 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1261 if (nodes[0] && nodes[1] && nodes[2])
1262 process_subpacket_11(q, nodes[2]);
1263 else
1264 process_subpacket_11(q, NULL);
1265
1266 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1267 if (nodes[0] && nodes[1] && nodes[3])
1268 process_subpacket_12(q, nodes[3]);
1269 else
1270 process_subpacket_12(q, NULL);
1271}
1272
1273/**
1274 * Decode superblock, fill packet lists.
1275 *
1276 * @param q context
1277 */
1278static void qdm2_decode_super_block(QDM2Context *q)
1279{
1280 GetBitContext gb;
1281 QDM2SubPacket header, *packet;
1282 int i, packet_bytes, sub_packet_size, sub_packets_D;
1283 unsigned int next_index = 0;
1284
1285 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1286 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1287 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1288
1289 q->sub_packets_B = 0;
1290 sub_packets_D = 0;
1291
1292 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1293
1294 init_get_bits(&gb, q->compressed_data, q->compressed_size * 8);
1295 qdm2_decode_sub_packet_header(&gb, &header);
1296
1297 if (header.type < 2 || header.type >= 8) {
1298 q->has_errors = 1;
1299 av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1300 return;
1301 }
1302
1303 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1304 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1305
1306 init_get_bits(&gb, header.data, header.size * 8);
1307
1308 if (header.type == 2 || header.type == 4 || header.type == 5) {
1309 int csum = 257 * get_bits(&gb, 8);
1310 csum += 2 * get_bits(&gb, 8);
1311
1312 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1313
1314 if (csum != 0) {
1315 q->has_errors = 1;
1316 av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1317 return;
1318 }
1319 }
1320
1321 q->sub_packet_list_B[0].packet = NULL;
1322 q->sub_packet_list_D[0].packet = NULL;
1323
1324 for (i = 0; i < 6; i++)
1325 if (--q->fft_level_exp[i] < 0)
1326 q->fft_level_exp[i] = 0;
1327
1328 for (i = 0; packet_bytes > 0; i++) {
1329 int j;
1330
1331 if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1332 SAMPLES_NEEDED_2("too many packet bytes");
1333 return;
1334 }
1335
1336 q->sub_packet_list_A[i].next = NULL;
1337
1338 if (i > 0) {
1339 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1340
1341 /* seek to next block */
1342 init_get_bits(&gb, header.data, header.size * 8);
1343 skip_bits(&gb, next_index * 8);
1344
1345 if (next_index >= header.size)
1346 break;
1347 }
1348
1349 /* decode subpacket */
1350 packet = &q->sub_packets[i];
1351 qdm2_decode_sub_packet_header(&gb, packet);
1352 next_index = packet->size + get_bits_count(&gb) / 8;
1353 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1354
1355 if (packet->type == 0)
1356 break;
1357
1358 if (sub_packet_size > packet_bytes) {
1359 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1360 break;
1361 packet->size += packet_bytes - sub_packet_size;
1362 }
1363
1364 packet_bytes -= sub_packet_size;
1365
1366 /* add subpacket to 'all subpackets' list */
1367 q->sub_packet_list_A[i].packet = packet;
1368
1369 /* add subpacket to related list */
1370 if (packet->type == 8) {
1371 SAMPLES_NEEDED_2("packet type 8");
1372 return;
1373 } else if (packet->type >= 9 && packet->type <= 12) {
1374 /* packets for MPEG Audio like Synthesis Filter */
1375 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1376 } else if (packet->type == 13) {
1377 for (j = 0; j < 6; j++)
1378 q->fft_level_exp[j] = get_bits(&gb, 6);
1379 } else if (packet->type == 14) {
1380 for (j = 0; j < 6; j++)
1381 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1382 } else if (packet->type == 15) {
1383 SAMPLES_NEEDED_2("packet type 15")
1384 return;
1385 } else if (packet->type >= 16 && packet->type < 48 &&
1386 !fft_subpackets[packet->type - 16]) {
1387 /* packets for FFT */
1388 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1389 }
1390 } // Packet bytes loop
1391
1392 if (q->sub_packet_list_D[0].packet) {
1393 process_synthesis_subpackets(q, q->sub_packet_list_D);
1394 q->do_synth_filter = 1;
1395 } else if (q->do_synth_filter) {
1396 process_subpacket_10(q, NULL);
1397 process_subpacket_11(q, NULL);
1398 process_subpacket_12(q, NULL);
1399 }
1400}
1401
1402static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1403 int offset, int duration, int channel,
1404 int exp, int phase)
1405{
1406 if (q->fft_coefs_min_index[duration] < 0)
1407 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1408
1409 q->fft_coefs[q->fft_coefs_index].sub_packet =
1410 ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1411 q->fft_coefs[q->fft_coefs_index].channel = channel;
1412 q->fft_coefs[q->fft_coefs_index].offset = offset;
1413 q->fft_coefs[q->fft_coefs_index].exp = exp;
1414 q->fft_coefs[q->fft_coefs_index].phase = phase;
1415 q->fft_coefs_index++;
1416}
1417
1418static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
1419 GetBitContext *gb, int b)
1420{
1421 int channel, stereo, phase, exp;
1422 int local_int_4, local_int_8, stereo_phase, local_int_10;
1423 int local_int_14, stereo_exp, local_int_20, local_int_28;
1424 int n, offset;
1425
1426 local_int_4 = 0;
1427 local_int_28 = 0;
1428 local_int_20 = 2;
1429 local_int_8 = (4 - duration);
1430 local_int_10 = 1 << (q->group_order - duration - 1);
1431 offset = 1;
1432
1433 while (get_bits_left(gb)>0) {
1434 if (q->superblocktype_2_3) {
1435 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1436 if (get_bits_left(gb)<0) {
1437 if(local_int_4 < q->group_size)
1438 av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1439 return;
1440 }
1441 offset = 1;
1442 if (n == 0) {
1443 local_int_4 += local_int_10;
1444 local_int_28 += (1 << local_int_8);
1445 } else {
1446 local_int_4 += 8 * local_int_10;
1447 local_int_28 += (8 << local_int_8);
1448 }
1449 }
1450 offset += (n - 2);
1451 } else {
1452 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1453 while (offset >= (local_int_10 - 1)) {
1454 offset += (1 - (local_int_10 - 1));
1455 local_int_4 += local_int_10;
1456 local_int_28 += (1 << local_int_8);
1457 }
1458 }
1459
1460 if (local_int_4 >= q->group_size)
1461 return;
1462
1463 local_int_14 = (offset >> local_int_8);
1464 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1465 return;
1466
1467 if (q->nb_channels > 1) {
1468 channel = get_bits1(gb);
1469 stereo = get_bits1(gb);
1470 } else {
1471 channel = 0;
1472 stereo = 0;
1473 }
1474
1475 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1476 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1477 exp = (exp < 0) ? 0 : exp;
1478
1479 phase = get_bits(gb, 3);
1480 stereo_exp = 0;
1481 stereo_phase = 0;
1482
1483 if (stereo) {
1484 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1485 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1486 if (stereo_phase < 0)
1487 stereo_phase += 8;
1488 }
1489
1490 if (q->frequency_range > (local_int_14 + 1)) {
1491 int sub_packet = (local_int_20 + local_int_28);
1492
1493 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1494 channel, exp, phase);
1495 if (stereo)
1496 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1497 1 - channel,
1498 stereo_exp, stereo_phase);
1499 }
1500 offset++;
1501 }
1502}
1503
1504static void qdm2_decode_fft_packets(QDM2Context *q)
1505{
1506 int i, j, min, max, value, type, unknown_flag;
1507 GetBitContext gb;
1508
1509 if (!q->sub_packet_list_B[0].packet)
1510 return;
1511
1512 /* reset minimum indexes for FFT coefficients */
1513 q->fft_coefs_index = 0;
1514 for (i = 0; i < 5; i++)
1515 q->fft_coefs_min_index[i] = -1;
1516
1517 /* process subpackets ordered by type, largest type first */
1518 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1519 QDM2SubPacket *packet = NULL;
1520
1521 /* find subpacket with largest type less than max */
1522 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1523 value = q->sub_packet_list_B[j].packet->type;
1524 if (value > min && value < max) {
1525 min = value;
1526 packet = q->sub_packet_list_B[j].packet;
1527 }
1528 }
1529
1530 max = min;
1531
1532 /* check for errors (?) */
1533 if (!packet)
1534 return;
1535
1536 if (i == 0 &&
1537 (packet->type < 16 || packet->type >= 48 ||
1538 fft_subpackets[packet->type - 16]))
1539 return;
1540
1541 /* decode FFT tones */
1542 init_get_bits(&gb, packet->data, packet->size * 8);
1543
1544 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1545 unknown_flag = 1;
1546 else
1547 unknown_flag = 0;
1548
1549 type = packet->type;
1550
1551 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1552 int duration = q->sub_sampling + 5 - (type & 15);
1553
1554 if (duration >= 0 && duration < 4)
1555 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1556 } else if (type == 31) {
1557 for (j = 0; j < 4; j++)
1558 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1559 } else if (type == 46) {
1560 for (j = 0; j < 6; j++)
1561 q->fft_level_exp[j] = get_bits(&gb, 6);
1562 for (j = 0; j < 4; j++)
1563 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1564 }
1565 } // Loop on B packets
1566
1567 /* calculate maximum indexes for FFT coefficients */
1568 for (i = 0, j = -1; i < 5; i++)
1569 if (q->fft_coefs_min_index[i] >= 0) {
1570 if (j >= 0)
1571 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1572 j = i;
1573 }
1574 if (j >= 0)
1575 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1576}
1577
1578static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
1579{
1580 float level, f[6];
1581 int i;
1582 QDM2Complex c;
1583 const double iscale = 2.0 * M_PI / 512.0;
1584
1585 tone->phase += tone->phase_shift;
1586
1587 /* calculate current level (maximum amplitude) of tone */
1588 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1589 c.im = level * sin(tone->phase * iscale);
1590 c.re = level * cos(tone->phase * iscale);
1591
1592 /* generate FFT coefficients for tone */
1593 if (tone->duration >= 3 || tone->cutoff >= 3) {
1594 tone->complex[0].im += c.im;
1595 tone->complex[0].re += c.re;
1596 tone->complex[1].im -= c.im;
1597 tone->complex[1].re -= c.re;
1598 } else {
1599 f[1] = -tone->table[4];
1600 f[0] = tone->table[3] - tone->table[0];
1601 f[2] = 1.0 - tone->table[2] - tone->table[3];
1602 f[3] = tone->table[1] + tone->table[4] - 1.0;
1603 f[4] = tone->table[0] - tone->table[1];
1604 f[5] = tone->table[2];
1605 for (i = 0; i < 2; i++) {
1606 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1607 c.re * f[i];
1608 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1609 c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1610 }
1611 for (i = 0; i < 4; i++) {
1612 tone->complex[i].re += c.re * f[i + 2];
1613 tone->complex[i].im += c.im * f[i + 2];
1614 }
1615 }
1616
1617 /* copy the tone if it has not yet died out */
1618 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1619 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1620 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1621 }
1622}
1623
1624static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1625{
1626 int i, j, ch;
1627 const double iscale = 0.25 * M_PI;
1628
1629 for (ch = 0; ch < q->channels; ch++) {
1630 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1631 }
1632
1633
1634 /* apply FFT tones with duration 4 (1 FFT period) */
1635 if (q->fft_coefs_min_index[4] >= 0)
1636 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1637 float level;
1638 QDM2Complex c;
1639
1640 if (q->fft_coefs[i].sub_packet != sub_packet)
1641 break;
1642
1643 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1644 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];
1645
1646 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1647 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1648 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1649 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1650 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1651 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1652 }
1653
1654 /* generate existing FFT tones */
1655 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1656 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1657 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1658 }
1659
1660 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1661 for (i = 0; i < 4; i++)
1662 if (q->fft_coefs_min_index[i] >= 0) {
1663 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1664 int offset, four_i;
1665 FFTTone tone;
1666
1667 if (q->fft_coefs[j].sub_packet != sub_packet)
1668 break;
1669
1670 four_i = (4 - i);
1671 offset = q->fft_coefs[j].offset >> four_i;
1672 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1673
1674 if (offset < q->frequency_range) {
1675 if (offset < 2)
1676 tone.cutoff = offset;
1677 else
1678 tone.cutoff = (offset >= 60) ? 3 : 2;
1679
1680 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];
1681 tone.complex = &q->fft.complex[ch][offset];
1682 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1683 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1684 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1685 tone.duration = i;
1686 tone.time_index = 0;
1687
1688 qdm2_fft_generate_tone(q, &tone);
1689 }
1690 }
1691 q->fft_coefs_min_index[i] = j;
1692 }
1693}
1694
1695static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1696{
1697 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1698 float *out = q->output_buffer + channel;
1699 int i;
1700 q->fft.complex[channel][0].re *= 2.0f;
1701 q->fft.complex[channel][0].im = 0.0f;
1702 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1703 /* add samples to output buffer */
1704 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1705 out[0] += q->fft.complex[channel][i].re * gain;
1706 out[q->channels] += q->fft.complex[channel][i].im * gain;
1707 out += 2 * q->channels;
1708 }
1709}
1710
1711/**
1712 * @param q context
1713 * @param index subpacket number
1714 */
1715static void qdm2_synthesis_filter(QDM2Context *q, int index)
1716{
1717 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1718
1719 /* copy sb_samples */
1720 sb_used = QDM2_SB_USED(q->sub_sampling);
1721
1722 for (ch = 0; ch < q->channels; ch++)
1723 for (i = 0; i < 8; i++)
1724 for (k = sb_used; k < SBLIMIT; k++)
1725 q->sb_samples[ch][(8 * index) + i][k] = 0;
1726
1727 for (ch = 0; ch < q->nb_channels; ch++) {
1728 float *samples_ptr = q->samples + ch;
1729
1730 for (i = 0; i < 8; i++) {
1731 ff_mpa_synth_filter_float(&q->mpadsp,
1732 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1733 ff_mpa_synth_window_float, &dither_state,
1734 samples_ptr, q->nb_channels,
1735 q->sb_samples[ch][(8 * index) + i]);
1736 samples_ptr += 32 * q->nb_channels;
1737 }
1738 }
1739
1740 /* add samples to output buffer */
1741 sub_sampling = (4 >> q->sub_sampling);
1742
1743 for (ch = 0; ch < q->channels; ch++)
1744 for (i = 0; i < q->frame_size; i++)
1745 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1746}
1747
1748/**
1749 * Init static data (does not depend on specific file)
1750 *
1751 * @param q context
1752 */
1753static av_cold void qdm2_init_static_data(void) {
1754 static int done;
1755
1756 if(done)
1757 return;
1758
1759 qdm2_init_vlc();
1760 ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1761 softclip_table_init();
1762 rnd_table_init();
1763 init_noise_samples();
1764
1765 done = 1;
1766}
1767
1768/**
1769 * Init parameters from codec extradata
1770 */
1771static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1772{
1773 QDM2Context *s = avctx->priv_data;
1774 uint8_t *extradata;
1775 int extradata_size;
1776 int tmp_val, tmp, size;
1777
1778 qdm2_init_static_data();
1779
1780 /* extradata parsing
1781
1782 Structure:
1783 wave {
1784 frma (QDM2)
1785 QDCA
1786 QDCP
1787 }
1788
1789 32 size (including this field)
1790 32 tag (=frma)
1791 32 type (=QDM2 or QDMC)
1792
1793 32 size (including this field, in bytes)
1794 32 tag (=QDCA) // maybe mandatory parameters
1795 32 unknown (=1)
1796 32 channels (=2)
1797 32 samplerate (=44100)
1798 32 bitrate (=96000)
1799 32 block size (=4096)
1800 32 frame size (=256) (for one channel)
1801 32 packet size (=1300)
1802
1803 32 size (including this field, in bytes)
1804 32 tag (=QDCP) // maybe some tuneable parameters
1805 32 float1 (=1.0)
1806 32 zero ?
1807 32 float2 (=1.0)
1808 32 float3 (=1.0)
1809 32 unknown (27)
1810 32 unknown (8)
1811 32 zero ?
1812 */
1813
1814 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1815 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1816 return -1;
1817 }
1818
1819 extradata = avctx->extradata;
1820 extradata_size = avctx->extradata_size;
1821
1822 while (extradata_size > 7) {
1823 if (!memcmp(extradata, "frmaQDM", 7))
1824 break;
1825 extradata++;
1826 extradata_size--;
1827 }
1828
1829 if (extradata_size < 12) {
1830 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1831 extradata_size);
1832 return -1;
1833 }
1834
1835 if (memcmp(extradata, "frmaQDM", 7)) {
1836 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1837 return -1;
1838 }
1839
1840 if (extradata[7] == 'C') {
1841// s->is_qdmc = 1;
1842 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1843 return -1;
1844 }
1845
1846 extradata += 8;
1847 extradata_size -= 8;
1848
1849 size = AV_RB32(extradata);
1850
1851 if(size > extradata_size){
1852 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1853 extradata_size, size);
1854 return -1;
1855 }
1856
1857 extradata += 4;
1858 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1859 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1860 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1861 return -1;
1862 }
1863
1864 extradata += 8;
1865
1866 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1867 extradata += 4;
1868 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1869 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1870 return AVERROR_INVALIDDATA;
1871 }
1872 avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1873 AV_CH_LAYOUT_MONO;
1874
1875 avctx->sample_rate = AV_RB32(extradata);
1876 extradata += 4;
1877
1878 avctx->bit_rate = AV_RB32(extradata);
1879 extradata += 4;
1880
1881 s->group_size = AV_RB32(extradata);
1882 extradata += 4;
1883
1884 s->fft_size = AV_RB32(extradata);
1885 extradata += 4;
1886
1887 s->checksum_size = AV_RB32(extradata);
1888 if (s->checksum_size >= 1U << 28) {
1889 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1890 return AVERROR_INVALIDDATA;
1891 }
1892
1893 s->fft_order = av_log2(s->fft_size) + 1;
1894
1895 // something like max decodable tones
1896 s->group_order = av_log2(s->group_size) + 1;
1897 s->frame_size = s->group_size / 16; // 16 iterations per super block
1898
1899 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1900 return AVERROR_INVALIDDATA;
1901
1902 s->sub_sampling = s->fft_order - 7;
1903 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1904
1905 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1906 case 0: tmp = 40; break;
1907 case 1: tmp = 48; break;
1908 case 2: tmp = 56; break;
1909 case 3: tmp = 72; break;
1910 case 4: tmp = 80; break;
1911 case 5: tmp = 100;break;
1912 default: tmp=s->sub_sampling; break;
1913 }
1914 tmp_val = 0;
1915 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1916 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1917 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1918 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1919 s->cm_table_select = tmp_val;
1920
1921 if (avctx->bit_rate <= 8000)
1922 s->coeff_per_sb_select = 0;
1923 else if (avctx->bit_rate < 16000)
1924 s->coeff_per_sb_select = 1;
1925 else
1926 s->coeff_per_sb_select = 2;
1927
1928 // Fail on unknown fft order
1929 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1930 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1931 return -1;
1932 }
1933 if (s->fft_size != (1 << (s->fft_order - 1))) {
1934 av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1935 return AVERROR_INVALIDDATA;
1936 }
1937
1938 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1939 ff_mpadsp_init(&s->mpadsp);
1940
1941 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1942
1943 return 0;
1944}
1945
1946static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1947{
1948 QDM2Context *s = avctx->priv_data;
1949
1950 ff_rdft_end(&s->rdft_ctx);
1951
1952 return 0;
1953}
1954
1955static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1956{
1957 int ch, i;
1958 const int frame_size = (q->frame_size * q->channels);
1959
1960 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1961 return -1;
1962
1963 /* select input buffer */
1964 q->compressed_data = in;
1965 q->compressed_size = q->checksum_size;
1966
1967 /* copy old block, clear new block of output samples */
1968 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1969 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1970
1971 /* decode block of QDM2 compressed data */
1972 if (q->sub_packet == 0) {
1973 q->has_errors = 0; // zero it for a new super block
1974 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1975 qdm2_decode_super_block(q);
1976 }
1977
1978 /* parse subpackets */
1979 if (!q->has_errors) {
1980 if (q->sub_packet == 2)
1981 qdm2_decode_fft_packets(q);
1982
1983 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1984 }
1985
1986 /* sound synthesis stage 1 (FFT) */
1987 for (ch = 0; ch < q->channels; ch++) {
1988 qdm2_calculate_fft(q, ch, q->sub_packet);
1989
1990 if (!q->has_errors && q->sub_packet_list_C[0].packet) {
1991 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1992 return -1;
1993 }
1994 }
1995
1996 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1997 if (!q->has_errors && q->do_synth_filter)
1998 qdm2_synthesis_filter(q, q->sub_packet);
1999
2000 q->sub_packet = (q->sub_packet + 1) % 16;
2001
2002 /* clip and convert output float[] to 16bit signed samples */
2003 for (i = 0; i < frame_size; i++) {
2004 int value = (int)q->output_buffer[i];
2005
2006 if (value > SOFTCLIP_THRESHOLD)
2007 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
2008 else if (value < -SOFTCLIP_THRESHOLD)
2009 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
2010
2011 out[i] = value;
2012 }
2013
2014 return 0;
2015}
2016
2017static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
2018 int *got_frame_ptr, AVPacket *avpkt)
2019{
2020 AVFrame *frame = data;
2021 const uint8_t *buf = avpkt->data;
2022 int buf_size = avpkt->size;
2023 QDM2Context *s = avctx->priv_data;
2024 int16_t *out;
2025 int i, ret;
2026
2027 if(!buf)
2028 return 0;
2029 if(buf_size < s->checksum_size)
2030 return -1;
2031
2032 /* get output buffer */
2033 frame->nb_samples = 16 * s->frame_size;
2034 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
2035 return ret;
2036 out = (int16_t *)frame->data[0];
2037
2038 for (i = 0; i < 16; i++) {
2039 if (qdm2_decode(s, buf, out) < 0)
2040 return -1;
2041 out += s->channels * s->frame_size;
2042 }
2043
2044 *got_frame_ptr = 1;
2045
2046 return s->checksum_size;
2047}
2048
2049AVCodec ff_qdm2_decoder = {
2050 .name = "qdm2",
2051 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
2052 .type = AVMEDIA_TYPE_AUDIO,
2053 .id = AV_CODEC_ID_QDM2,
2054 .priv_data_size = sizeof(QDM2Context),
2055 .init = qdm2_decode_init,
2056 .close = qdm2_decode_close,
2057 .decode = qdm2_decode_frame,
2058 .capabilities = CODEC_CAP_DR1,
2059};
ffmpeg Piment Noir Debian package repository