2 * QDM2 compatible decoder
3 * Copyright (c) 2003 Ewald Snel
4 * Copyright (c) 2005 Benjamin Larsson
5 * Copyright (c) 2005 Alex Beregszaszi
6 * Copyright (c) 2005 Roberto Togni
8 * This file is part of FFmpeg.
10 * FFmpeg is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Lesser General Public
12 * License as published by the Free Software Foundation; either
13 * version 2.1 of the License, or (at your option) any later version.
15 * FFmpeg is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Lesser General Public License for more details.
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with FFmpeg; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
30 * The decoder is not perfect yet, there are still some distortions
31 * especially on files encoded with 16 or 8 subbands.
38 #define BITSTREAM_READER_LE
39 #include "libavutil/channel_layout.h"
44 #include "mpegaudiodsp.h"
45 #include "mpegaudio.h"
48 #include "qdm2_tablegen.h"
54 #define QDM2_LIST_ADD(list, size, packet) \
57 list[size - 1].next = &list[size]; \
59 list[size].packet = packet; \
60 list[size].next = NULL; \
64 // Result is 8, 16 or 30
65 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
67 #define FIX_NOISE_IDX(noise_idx) \
68 if ((noise_idx) >= 3840) \
69 (noise_idx) -= 3840; \
71 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
73 #define SAMPLES_NEEDED \
74 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
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);
79 #define QDM2_MAX_FRAME_SIZE 512
81 typedef int8_t sb_int8_array[2][30][64];
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)
93 * A node in the subpacket list
95 typedef struct QDM2SubPNode {
96 QDM2SubPacket *packet; ///< packet
97 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
107 QDM2Complex *complex;
125 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
129 * QDM2 decoder context
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
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
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)
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
157 FFTTone fft_tones[1000];
160 FFTCoefficient fft_coefs[1000];
162 int fft_coefs_min_index[5];
163 int fft_coefs_max_index[5];
164 int fft_level_exp[6];
165 RDFTContext rdft_ctx;
169 const uint8_t *compressed_data;
171 float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
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];
180 /// Mixed temporary data used in decoding
181 float tone_level[MPA_MAX_CHANNELS][30][64];
182 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
183 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
184 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
185 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
186 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
187 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
188 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
189 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
192 int has_errors; ///< packet has errors
193 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
194 int do_synth_filter; ///< used to perform or skip synthesis filter
197 int noise_idx; ///< index for dithering noise table
201 static VLC vlc_tab_level;
202 static VLC vlc_tab_diff;
203 static VLC vlc_tab_run;
204 static VLC fft_level_exp_alt_vlc;
205 static VLC fft_level_exp_vlc;
206 static VLC fft_stereo_exp_vlc;
207 static VLC fft_stereo_phase_vlc;
208 static VLC vlc_tab_tone_level_idx_hi1;
209 static VLC vlc_tab_tone_level_idx_mid;
210 static VLC vlc_tab_tone_level_idx_hi2;
211 static VLC vlc_tab_type30;
212 static VLC vlc_tab_type34;
213 static VLC vlc_tab_fft_tone_offset[5];
215 static 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,
219 static 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
223 static av_cold void qdm2_init_vlc(void)
225 static VLC_TYPE qdm2_table[3838][2];
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);
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);
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);
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] -
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);
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);
263 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
264 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] -
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);
271 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
272 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] -
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);
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] -
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);
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] -
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);
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] -
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);
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);
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);
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] -
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);
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] -
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);
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] -
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);
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] -
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);
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] -
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);
366 static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth)
370 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
372 /* stage-2, 3 bits exponent escape sequence */
374 value = get_bits(gb, get_bits(gb, 3) + 1);
376 /* stage-3, optional */
381 av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
385 tmp= vlc_stage3_values[value];
387 if ((value & ~3) > 0)
388 tmp += get_bits(gb, (value >> 2));
395 static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth)
397 int value = qdm2_get_vlc(gb, vlc, 0, depth);
399 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
405 * @param data pointer to data to be checksum'ed
406 * @param length data length
407 * @param value checksum value
409 * @return 0 if checksum is OK
411 static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
415 for (i = 0; i < length; i++)
418 return (uint16_t)(value & 0xffff);
422 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
424 * @param gb bitreader context
425 * @param sub_packet packet under analysis
427 static void qdm2_decode_sub_packet_header(GetBitContext *gb,
428 QDM2SubPacket *sub_packet)
430 sub_packet->type = get_bits(gb, 8);
432 if (sub_packet->type == 0) {
433 sub_packet->size = 0;
434 sub_packet->data = NULL;
436 sub_packet->size = get_bits(gb, 8);
438 if (sub_packet->type & 0x80) {
439 sub_packet->size <<= 8;
440 sub_packet->size |= get_bits(gb, 8);
441 sub_packet->type &= 0x7f;
444 if (sub_packet->type == 0x7f)
445 sub_packet->type |= (get_bits(gb, 8) << 8);
447 // FIXME: this depends on bitreader-internal data
448 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
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);
456 * Return node pointer to first packet of requested type in list.
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
462 static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
465 while (list != NULL && list->packet != NULL) {
466 if (list->packet->type == type)
474 * Replace 8 elements with their average value.
475 * Called by qdm2_decode_superblock before starting subblock decoding.
479 static void average_quantized_coeffs(QDM2Context *q)
481 int i, j, n, ch, sum;
483 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
485 for (ch = 0; ch < q->nb_channels; ch++)
486 for (i = 0; i < n; i++) {
489 for (j = 0; j < 8; j++)
490 sum += q->quantized_coeffs[ch][i][j];
496 for (j = 0; j < 8; j++)
497 q->quantized_coeffs[ch][i][j] = sum;
502 * Build subband samples with noise weighted by q->tone_level.
503 * Called by synthfilt_build_sb_samples.
506 * @param sb subband index
508 static void build_sb_samples_from_noise(QDM2Context *q, int sb)
512 FIX_NOISE_IDX(q->noise_idx);
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];
528 * Called while processing data from subpackets 11 and 12.
529 * Used after making changes to coding_method array.
531 * @param sb subband index
532 * @param channels number of channels
533 * @param coding_method q->coding_method[0][0][0]
535 static int fix_coding_method_array(int sb, int channels,
536 sb_int8_array coding_method)
542 for (ch = 0; ch < channels; ch++) {
543 for (j = 0; j < 64; ) {
544 if (coding_method[ch][sb][j] < 8)
546 if ((coding_method[ch][sb][j] - 8) > 22) {
550 switch (switchtable[coding_method[ch][sb][j] - 8]) {
574 for (k = 0; k < run; k++) {
576 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) {
579 //not debugged, almost never used
580 memset(&coding_method[ch][sb][j + k], case_val,
582 memset(&coding_method[ch][sb][j + k], case_val,
595 * Related to synthesis filter
596 * Called by process_subpacket_10
599 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
601 static void fill_tone_level_array(QDM2Context *q, int flag)
603 int i, sb, ch, sb_used;
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];
613 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
616 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
619 sb_used = QDM2_SB_USED(q->sub_sampling);
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;
629 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
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;
645 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
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;
658 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
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;
667 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
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.
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
690 static 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,
694 int c, int superblocktype_2_3,
698 int tmp, acc, esp_40, comp;
699 int add1, add2, add3, add4;
702 if (!superblocktype_2_3) {
703 /* This case is untested, no samples available */
704 avpriv_request_sample(NULL, "!superblocktype_2_3");
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;
712 add2 = add3 = add4 = 0;
714 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
719 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
724 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
728 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
731 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
733 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
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];
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;
749 comp /= 256; // signed shift
777 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
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++)
785 if (coding_method[ch][sb][j] < 10)
786 coding_method[ch][sb][j] = 10;
789 if (coding_method[ch][sb][j] < 16)
790 coding_method[ch][sb][j] = 16;
792 if (coding_method[ch][sb][j] < 30)
793 coding_method[ch][sb][j] = 30;
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];
806 * Called by process_subpacket_11 to process more data from subpacket 11
808 * Called by process_subpacket_12 to process data from subpacket 12 with
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)
817 static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
818 int length, int sb_min, int sb_max)
820 int sb, j, k, n, ch, run, channels;
821 int joined_stereo, zero_encoding;
823 float type34_div = 0;
824 float type34_predictor;
825 float samples[10], sign_bits[16];
828 // If no data use noise
829 for (sb=sb_min; sb < sb_max; sb++)
830 build_sb_samples_from_noise (q, sb);
835 for (sb = sb_min; sb < sb_max; sb++) {
836 channels = q->nb_channels;
838 if (q->nb_channels <= 1 || sb < 12)
843 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0;
846 if (get_bits_left(gb) >= 16)
847 for (j = 0; j < 16; j++)
848 sign_bits[j] = get_bits1 (gb);
850 for (j = 0; j < 64; j++)
851 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
852 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
854 if (fix_coding_method_array(sb, q->nb_channels,
856 av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
857 build_sb_samples_from_noise(q, sb);
863 for (ch = 0; ch < channels; ch++) {
864 FIX_NOISE_IDX(q->noise_idx);
865 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
866 type34_predictor = 0.0;
869 for (j = 0; j < 128; ) {
870 switch (q->coding_method[ch][sb][j / 2]) {
872 if (get_bits_left(gb) >= 10) {
874 for (k = 0; k < 5; k++) {
875 if ((j + 2 * k) >= 128)
877 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
882 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
883 return AVERROR_INVALIDDATA;
886 for (k = 0; k < 5; k++)
887 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
889 for (k = 0; k < 5; k++)
890 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
892 for (k = 0; k < 10; k++)
893 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
899 if (get_bits_left(gb) >= 1) {
904 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
907 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
913 if (get_bits_left(gb) >= 10) {
915 for (k = 0; k < 5; k++) {
918 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
921 n = get_bits (gb, 8);
923 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
924 return AVERROR_INVALIDDATA;
927 for (k = 0; k < 5; k++)
928 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
931 for (k = 0; k < 5; k++)
932 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
938 if (get_bits_left(gb) >= 7) {
941 av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
942 return AVERROR_INVALIDDATA;
945 for (k = 0; k < 3; k++)
946 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
948 for (k = 0; k < 3; k++)
949 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
955 if (get_bits_left(gb) >= 4) {
956 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
957 if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
958 av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
959 return AVERROR_INVALIDDATA;
961 samples[0] = type30_dequant[index];
963 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
969 if (get_bits_left(gb) >= 7) {
971 type34_div = (float)(1 << get_bits(gb, 2));
972 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
973 type34_predictor = samples[0];
976 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
977 if (index >= FF_ARRAY_ELEMS(type34_delta)) {
978 av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
979 return AVERROR_INVALIDDATA;
981 samples[0] = type34_delta[index] / type34_div + type34_predictor;
982 type34_predictor = samples[0];
985 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
991 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
997 for (k = 0; k < run && j + k < 128; k++) {
998 q->sb_samples[0][j + k][sb] =
999 q->tone_level[0][sb][(j + k) / 2] * samples[k];
1000 if (q->nb_channels == 2) {
1001 if (sign_bits[(j + k) / 8])
1002 q->sb_samples[1][j + k][sb] =
1003 q->tone_level[1][sb][(j + k) / 2] * -samples[k];
1005 q->sb_samples[1][j + k][sb] =
1006 q->tone_level[1][sb][(j + k) / 2] * samples[k];
1010 for (k = 0; k < run; k++)
1012 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
1023 * Init the first element of a channel in quantized_coeffs with data
1024 * from packet 10 (quantized_coeffs[ch][0]).
1025 * This is similar to process_subpacket_9, but for a single channel
1026 * and for element [0]
1027 * same VLC tables as process_subpacket_9 are used.
1029 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
1030 * @param gb bitreader context
1032 static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
1035 int i, k, run, level, diff;
1037 if (get_bits_left(gb) < 16)
1039 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
1041 quantized_coeffs[0] = level;
1043 for (i = 0; i < 7; ) {
1044 if (get_bits_left(gb) < 16)
1046 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
1051 if (get_bits_left(gb) < 16)
1053 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
1055 for (k = 1; k <= run; k++)
1056 quantized_coeffs[i + k] = (level + ((k * diff) / run));
1065 * Related to synthesis filter, process data from packet 10
1066 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1067 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
1068 * data from packet 10
1071 * @param gb bitreader context
1073 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
1075 int sb, j, k, n, ch;
1077 for (ch = 0; ch < q->nb_channels; ch++) {
1078 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
1080 if (get_bits_left(gb) < 16) {
1081 memset(q->quantized_coeffs[ch][0], 0, 8);
1086 n = q->sub_sampling + 1;
1088 for (sb = 0; sb < n; sb++)
1089 for (ch = 0; ch < q->nb_channels; ch++)
1090 for (j = 0; j < 8; j++) {
1091 if (get_bits_left(gb) < 1)
1093 if (get_bits1(gb)) {
1094 for (k=0; k < 8; k++) {
1095 if (get_bits_left(gb) < 16)
1097 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1100 for (k=0; k < 8; k++)
1101 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1105 n = QDM2_SB_USED(q->sub_sampling) - 4;
1107 for (sb = 0; sb < n; sb++)
1108 for (ch = 0; ch < q->nb_channels; ch++) {
1109 if (get_bits_left(gb) < 16)
1111 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1113 q->tone_level_idx_hi2[ch][sb] -= 16;
1115 for (j = 0; j < 8; j++)
1116 q->tone_level_idx_mid[ch][sb][j] = -16;
1119 n = QDM2_SB_USED(q->sub_sampling) - 5;
1121 for (sb = 0; sb < n; sb++)
1122 for (ch = 0; ch < q->nb_channels; ch++)
1123 for (j = 0; j < 8; j++) {
1124 if (get_bits_left(gb) < 16)
1126 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1131 * Process subpacket 9, init quantized_coeffs with data from it
1134 * @param node pointer to node with packet
1136 static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
1139 int i, j, k, n, ch, run, level, diff;
1141 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1143 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
1145 for (i = 1; i < n; i++)
1146 for (ch = 0; ch < q->nb_channels; ch++) {
1147 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1148 q->quantized_coeffs[ch][i][0] = level;
1150 for (j = 0; j < (8 - 1); ) {
1151 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1152 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1157 for (k = 1; k <= run; k++)
1158 q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
1165 for (ch = 0; ch < q->nb_channels; ch++)
1166 for (i = 0; i < 8; i++)
1167 q->quantized_coeffs[ch][0][i] = 0;
1173 * Process subpacket 10 if not null, else
1176 * @param node pointer to node with packet
1178 static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
1183 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1184 init_tone_level_dequantization(q, &gb);
1185 fill_tone_level_array(q, 1);
1187 fill_tone_level_array(q, 0);
1192 * Process subpacket 11
1195 * @param node pointer to node with packet
1197 static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
1203 length = node->packet->size * 8;
1204 init_get_bits(&gb, node->packet->data, length);
1208 int c = get_bits(&gb, 13);
1211 fill_coding_method_array(q->tone_level_idx,
1212 q->tone_level_idx_temp, q->coding_method,
1213 q->nb_channels, 8 * c,
1214 q->superblocktype_2_3, q->cm_table_select);
1217 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1221 * Process subpacket 12
1224 * @param node pointer to node with packet
1226 static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
1232 length = node->packet->size * 8;
1233 init_get_bits(&gb, node->packet->data, length);
1236 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1240 * Process new subpackets for synthesis filter
1243 * @param list list with synthesis filter packets (list D)
1245 static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
1247 QDM2SubPNode *nodes[4];
1249 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1250 if (nodes[0] != NULL)
1251 process_subpacket_9(q, nodes[0]);
1253 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1254 if (nodes[1] != NULL)
1255 process_subpacket_10(q, nodes[1]);
1257 process_subpacket_10(q, NULL);
1259 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1260 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1261 process_subpacket_11(q, nodes[2]);
1263 process_subpacket_11(q, NULL);
1265 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1266 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1267 process_subpacket_12(q, nodes[3]);
1269 process_subpacket_12(q, NULL);
1273 * Decode superblock, fill packet lists.
1277 static void qdm2_decode_super_block(QDM2Context *q)
1280 QDM2SubPacket header, *packet;
1281 int i, packet_bytes, sub_packet_size, sub_packets_D;
1282 unsigned int next_index = 0;
1284 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1285 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1286 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1288 q->sub_packets_B = 0;
1291 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1293 init_get_bits(&gb, q->compressed_data, q->compressed_size * 8);
1294 qdm2_decode_sub_packet_header(&gb, &header);
1296 if (header.type < 2 || header.type >= 8) {
1298 av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1302 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1303 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1305 init_get_bits(&gb, header.data, header.size * 8);
1307 if (header.type == 2 || header.type == 4 || header.type == 5) {
1308 int csum = 257 * get_bits(&gb, 8);
1309 csum += 2 * get_bits(&gb, 8);
1311 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1315 av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1320 q->sub_packet_list_B[0].packet = NULL;
1321 q->sub_packet_list_D[0].packet = NULL;
1323 for (i = 0; i < 6; i++)
1324 if (--q->fft_level_exp[i] < 0)
1325 q->fft_level_exp[i] = 0;
1327 for (i = 0; packet_bytes > 0; i++) {
1330 if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1331 SAMPLES_NEEDED_2("too many packet bytes");
1335 q->sub_packet_list_A[i].next = NULL;
1338 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1340 /* seek to next block */
1341 init_get_bits(&gb, header.data, header.size * 8);
1342 skip_bits(&gb, next_index * 8);
1344 if (next_index >= header.size)
1348 /* decode subpacket */
1349 packet = &q->sub_packets[i];
1350 qdm2_decode_sub_packet_header(&gb, packet);
1351 next_index = packet->size + get_bits_count(&gb) / 8;
1352 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1354 if (packet->type == 0)
1357 if (sub_packet_size > packet_bytes) {
1358 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1360 packet->size += packet_bytes - sub_packet_size;
1363 packet_bytes -= sub_packet_size;
1365 /* add subpacket to 'all subpackets' list */
1366 q->sub_packet_list_A[i].packet = packet;
1368 /* add subpacket to related list */
1369 if (packet->type == 8) {
1370 SAMPLES_NEEDED_2("packet type 8");
1372 } else if (packet->type >= 9 && packet->type <= 12) {
1373 /* packets for MPEG Audio like Synthesis Filter */
1374 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1375 } else if (packet->type == 13) {
1376 for (j = 0; j < 6; j++)
1377 q->fft_level_exp[j] = get_bits(&gb, 6);
1378 } else if (packet->type == 14) {
1379 for (j = 0; j < 6; j++)
1380 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1381 } else if (packet->type == 15) {
1382 SAMPLES_NEEDED_2("packet type 15")
1384 } else if (packet->type >= 16 && packet->type < 48 &&
1385 !fft_subpackets[packet->type - 16]) {
1386 /* packets for FFT */
1387 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1389 } // Packet bytes loop
1391 if (q->sub_packet_list_D[0].packet != NULL) {
1392 process_synthesis_subpackets(q, q->sub_packet_list_D);
1393 q->do_synth_filter = 1;
1394 } else if (q->do_synth_filter) {
1395 process_subpacket_10(q, NULL);
1396 process_subpacket_11(q, NULL);
1397 process_subpacket_12(q, NULL);
1401 static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1402 int offset, int duration, int channel,
1405 if (q->fft_coefs_min_index[duration] < 0)
1406 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1408 q->fft_coefs[q->fft_coefs_index].sub_packet =
1409 ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1410 q->fft_coefs[q->fft_coefs_index].channel = channel;
1411 q->fft_coefs[q->fft_coefs_index].offset = offset;
1412 q->fft_coefs[q->fft_coefs_index].exp = exp;
1413 q->fft_coefs[q->fft_coefs_index].phase = phase;
1414 q->fft_coefs_index++;
1417 static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
1418 GetBitContext *gb, int b)
1420 int channel, stereo, phase, exp;
1421 int local_int_4, local_int_8, stereo_phase, local_int_10;
1422 int local_int_14, stereo_exp, local_int_20, local_int_28;
1428 local_int_8 = (4 - duration);
1429 local_int_10 = 1 << (q->group_order - duration - 1);
1432 while (get_bits_left(gb)>0) {
1433 if (q->superblocktype_2_3) {
1434 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1435 if (get_bits_left(gb)<0) {
1436 if(local_int_4 < q->group_size)
1437 av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1442 local_int_4 += local_int_10;
1443 local_int_28 += (1 << local_int_8);
1445 local_int_4 += 8 * local_int_10;
1446 local_int_28 += (8 << local_int_8);
1451 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1452 while (offset >= (local_int_10 - 1)) {
1453 offset += (1 - (local_int_10 - 1));
1454 local_int_4 += local_int_10;
1455 local_int_28 += (1 << local_int_8);
1459 if (local_int_4 >= q->group_size)
1462 local_int_14 = (offset >> local_int_8);
1463 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1466 if (q->nb_channels > 1) {
1467 channel = get_bits1(gb);
1468 stereo = get_bits1(gb);
1474 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1475 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1476 exp = (exp < 0) ? 0 : exp;
1478 phase = get_bits(gb, 3);
1483 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1484 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1485 if (stereo_phase < 0)
1489 if (q->frequency_range > (local_int_14 + 1)) {
1490 int sub_packet = (local_int_20 + local_int_28);
1492 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1493 channel, exp, phase);
1495 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1497 stereo_exp, stereo_phase);
1503 static void qdm2_decode_fft_packets(QDM2Context *q)
1505 int i, j, min, max, value, type, unknown_flag;
1508 if (q->sub_packet_list_B[0].packet == NULL)
1511 /* reset minimum indexes for FFT coefficients */
1512 q->fft_coefs_index = 0;
1513 for (i = 0; i < 5; i++)
1514 q->fft_coefs_min_index[i] = -1;
1516 /* process subpackets ordered by type, largest type first */
1517 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1518 QDM2SubPacket *packet = NULL;
1520 /* find subpacket with largest type less than max */
1521 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1522 value = q->sub_packet_list_B[j].packet->type;
1523 if (value > min && value < max) {
1525 packet = q->sub_packet_list_B[j].packet;
1531 /* check for errors (?) */
1536 (packet->type < 16 || packet->type >= 48 ||
1537 fft_subpackets[packet->type - 16]))
1540 /* decode FFT tones */
1541 init_get_bits(&gb, packet->data, packet->size * 8);
1543 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1548 type = packet->type;
1550 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1551 int duration = q->sub_sampling + 5 - (type & 15);
1553 if (duration >= 0 && duration < 4)
1554 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1555 } else if (type == 31) {
1556 for (j = 0; j < 4; j++)
1557 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1558 } else if (type == 46) {
1559 for (j = 0; j < 6; j++)
1560 q->fft_level_exp[j] = get_bits(&gb, 6);
1561 for (j = 0; j < 4; j++)
1562 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1564 } // Loop on B packets
1566 /* calculate maximum indexes for FFT coefficients */
1567 for (i = 0, j = -1; i < 5; i++)
1568 if (q->fft_coefs_min_index[i] >= 0) {
1570 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1574 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1577 static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
1582 const double iscale = 2.0 * M_PI / 512.0;
1584 tone->phase += tone->phase_shift;
1586 /* calculate current level (maximum amplitude) of tone */
1587 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1588 c.im = level * sin(tone->phase * iscale);
1589 c.re = level * cos(tone->phase * iscale);
1591 /* generate FFT coefficients for tone */
1592 if (tone->duration >= 3 || tone->cutoff >= 3) {
1593 tone->complex[0].im += c.im;
1594 tone->complex[0].re += c.re;
1595 tone->complex[1].im -= c.im;
1596 tone->complex[1].re -= c.re;
1598 f[1] = -tone->table[4];
1599 f[0] = tone->table[3] - tone->table[0];
1600 f[2] = 1.0 - tone->table[2] - tone->table[3];
1601 f[3] = tone->table[1] + tone->table[4] - 1.0;
1602 f[4] = tone->table[0] - tone->table[1];
1603 f[5] = tone->table[2];
1604 for (i = 0; i < 2; i++) {
1605 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1607 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1608 c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1610 for (i = 0; i < 4; i++) {
1611 tone->complex[i].re += c.re * f[i + 2];
1612 tone->complex[i].im += c.im * f[i + 2];
1616 /* copy the tone if it has not yet died out */
1617 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1618 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1619 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1623 static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1626 const double iscale = 0.25 * M_PI;
1628 for (ch = 0; ch < q->channels; ch++) {
1629 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1633 /* apply FFT tones with duration 4 (1 FFT period) */
1634 if (q->fft_coefs_min_index[4] >= 0)
1635 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1639 if (q->fft_coefs[i].sub_packet != sub_packet)
1642 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1643 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 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1646 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1647 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1648 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1649 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1650 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1653 /* generate existing FFT tones */
1654 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1655 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1656 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1659 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1660 for (i = 0; i < 4; i++)
1661 if (q->fft_coefs_min_index[i] >= 0) {
1662 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1666 if (q->fft_coefs[j].sub_packet != sub_packet)
1670 offset = q->fft_coefs[j].offset >> four_i;
1671 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1673 if (offset < q->frequency_range) {
1675 tone.cutoff = offset;
1677 tone.cutoff = (offset >= 60) ? 3 : 2;
1679 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];
1680 tone.complex = &q->fft.complex[ch][offset];
1681 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1682 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1683 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1685 tone.time_index = 0;
1687 qdm2_fft_generate_tone(q, &tone);
1690 q->fft_coefs_min_index[i] = j;
1694 static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1696 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1697 float *out = q->output_buffer + channel;
1699 q->fft.complex[channel][0].re *= 2.0f;
1700 q->fft.complex[channel][0].im = 0.0f;
1701 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1702 /* add samples to output buffer */
1703 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1704 out[0] += q->fft.complex[channel][i].re * gain;
1705 out[q->channels] += q->fft.complex[channel][i].im * gain;
1706 out += 2 * q->channels;
1712 * @param index subpacket number
1714 static void qdm2_synthesis_filter(QDM2Context *q, int index)
1716 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1718 /* copy sb_samples */
1719 sb_used = QDM2_SB_USED(q->sub_sampling);
1721 for (ch = 0; ch < q->channels; ch++)
1722 for (i = 0; i < 8; i++)
1723 for (k = sb_used; k < SBLIMIT; k++)
1724 q->sb_samples[ch][(8 * index) + i][k] = 0;
1726 for (ch = 0; ch < q->nb_channels; ch++) {
1727 float *samples_ptr = q->samples + ch;
1729 for (i = 0; i < 8; i++) {
1730 ff_mpa_synth_filter_float(&q->mpadsp,
1731 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1732 ff_mpa_synth_window_float, &dither_state,
1733 samples_ptr, q->nb_channels,
1734 q->sb_samples[ch][(8 * index) + i]);
1735 samples_ptr += 32 * q->nb_channels;
1739 /* add samples to output buffer */
1740 sub_sampling = (4 >> q->sub_sampling);
1742 for (ch = 0; ch < q->channels; ch++)
1743 for (i = 0; i < q->frame_size; i++)
1744 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1748 * Init static data (does not depend on specific file)
1752 static av_cold void qdm2_init_static_data(AVCodec *codec) {
1754 ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1755 softclip_table_init();
1757 init_noise_samples();
1761 * Init parameters from codec extradata
1763 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1765 QDM2Context *s = avctx->priv_data;
1768 int tmp_val, tmp, size;
1770 /* extradata parsing
1779 32 size (including this field)
1781 32 type (=QDM2 or QDMC)
1783 32 size (including this field, in bytes)
1784 32 tag (=QDCA) // maybe mandatory parameters
1787 32 samplerate (=44100)
1789 32 block size (=4096)
1790 32 frame size (=256) (for one channel)
1791 32 packet size (=1300)
1793 32 size (including this field, in bytes)
1794 32 tag (=QDCP) // maybe some tuneable parameters
1804 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1805 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1809 extradata = avctx->extradata;
1810 extradata_size = avctx->extradata_size;
1812 while (extradata_size > 7) {
1813 if (!memcmp(extradata, "frmaQDM", 7))
1819 if (extradata_size < 12) {
1820 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1825 if (memcmp(extradata, "frmaQDM", 7)) {
1826 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1830 if (extradata[7] == 'C') {
1832 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1837 extradata_size -= 8;
1839 size = AV_RB32(extradata);
1841 if(size > extradata_size){
1842 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1843 extradata_size, size);
1848 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1849 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1850 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1856 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1858 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1859 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1860 return AVERROR_INVALIDDATA;
1862 avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1865 avctx->sample_rate = AV_RB32(extradata);
1868 avctx->bit_rate = AV_RB32(extradata);
1871 s->group_size = AV_RB32(extradata);
1874 s->fft_size = AV_RB32(extradata);
1877 s->checksum_size = AV_RB32(extradata);
1878 if (s->checksum_size >= 1U << 28) {
1879 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1880 return AVERROR_INVALIDDATA;
1883 s->fft_order = av_log2(s->fft_size) + 1;
1885 // something like max decodable tones
1886 s->group_order = av_log2(s->group_size) + 1;
1887 s->frame_size = s->group_size / 16; // 16 iterations per super block
1889 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1890 return AVERROR_INVALIDDATA;
1892 s->sub_sampling = s->fft_order - 7;
1893 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1895 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1896 case 0: tmp = 40; break;
1897 case 1: tmp = 48; break;
1898 case 2: tmp = 56; break;
1899 case 3: tmp = 72; break;
1900 case 4: tmp = 80; break;
1901 case 5: tmp = 100;break;
1902 default: tmp=s->sub_sampling; break;
1905 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1906 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1907 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1908 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1909 s->cm_table_select = tmp_val;
1911 if (avctx->bit_rate <= 8000)
1912 s->coeff_per_sb_select = 0;
1913 else if (avctx->bit_rate < 16000)
1914 s->coeff_per_sb_select = 1;
1916 s->coeff_per_sb_select = 2;
1918 // Fail on unknown fft order
1919 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1920 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1923 if (s->fft_size != (1 << (s->fft_order - 1))) {
1924 av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1925 return AVERROR_INVALIDDATA;
1928 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1929 ff_mpadsp_init(&s->mpadsp);
1931 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1936 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1938 QDM2Context *s = avctx->priv_data;
1940 ff_rdft_end(&s->rdft_ctx);
1945 static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1948 const int frame_size = (q->frame_size * q->channels);
1950 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1953 /* select input buffer */
1954 q->compressed_data = in;
1955 q->compressed_size = q->checksum_size;
1957 /* copy old block, clear new block of output samples */
1958 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1959 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1961 /* decode block of QDM2 compressed data */
1962 if (q->sub_packet == 0) {
1963 q->has_errors = 0; // zero it for a new super block
1964 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1965 qdm2_decode_super_block(q);
1968 /* parse subpackets */
1969 if (!q->has_errors) {
1970 if (q->sub_packet == 2)
1971 qdm2_decode_fft_packets(q);
1973 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1976 /* sound synthesis stage 1 (FFT) */
1977 for (ch = 0; ch < q->channels; ch++) {
1978 qdm2_calculate_fft(q, ch, q->sub_packet);
1980 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1981 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1986 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1987 if (!q->has_errors && q->do_synth_filter)
1988 qdm2_synthesis_filter(q, q->sub_packet);
1990 q->sub_packet = (q->sub_packet + 1) % 16;
1992 /* clip and convert output float[] to 16bit signed samples */
1993 for (i = 0; i < frame_size; i++) {
1994 int value = (int)q->output_buffer[i];
1996 if (value > SOFTCLIP_THRESHOLD)
1997 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1998 else if (value < -SOFTCLIP_THRESHOLD)
1999 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
2007 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
2008 int *got_frame_ptr, AVPacket *avpkt)
2010 AVFrame *frame = data;
2011 const uint8_t *buf = avpkt->data;
2012 int buf_size = avpkt->size;
2013 QDM2Context *s = avctx->priv_data;
2019 if(buf_size < s->checksum_size)
2022 /* get output buffer */
2023 frame->nb_samples = 16 * s->frame_size;
2024 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
2026 out = (int16_t *)frame->data[0];
2028 for (i = 0; i < 16; i++) {
2029 if (qdm2_decode(s, buf, out) < 0)
2031 out += s->channels * s->frame_size;
2036 return s->checksum_size;
2039 AVCodec ff_qdm2_decoder = {
2041 .type = AVMEDIA_TYPE_AUDIO,
2042 .id = AV_CODEC_ID_QDM2,
2043 .priv_data_size = sizeof(QDM2Context),
2044 .init = qdm2_decode_init,
2045 .init_static_data = qdm2_init_static_data,
2046 .close = qdm2_decode_close,
2047 .decode = qdm2_decode_frame,
2048 .capabilities = CODEC_CAP_DR1,
2049 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),