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 void 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) > 22) {
548 switch (switchtable[coding_method[ch][sb][j] - 8]) {
572 for (k = 0; k < run; k++) {
574 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) {
577 //not debugged, almost never used
578 memset(&coding_method[ch][sb][j + k], case_val,
580 memset(&coding_method[ch][sb][j + k], case_val,
592 * Related to synthesis filter
593 * Called by process_subpacket_10
596 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
598 static void fill_tone_level_array(QDM2Context *q, int flag)
600 int i, sb, ch, sb_used;
603 for (ch = 0; ch < q->nb_channels; ch++)
604 for (sb = 0; sb < 30; sb++)
605 for (i = 0; i < 8; i++) {
606 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
607 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
608 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
610 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
613 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
616 sb_used = QDM2_SB_USED(q->sub_sampling);
618 if ((q->superblocktype_2_3 != 0) && !flag) {
619 for (sb = 0; sb < sb_used; sb++)
620 for (ch = 0; ch < q->nb_channels; ch++)
621 for (i = 0; i < 64; i++) {
622 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
623 if (q->tone_level_idx[ch][sb][i] < 0)
624 q->tone_level[ch][sb][i] = 0;
626 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
629 tab = q->superblocktype_2_3 ? 0 : 1;
630 for (sb = 0; sb < sb_used; sb++) {
631 if ((sb >= 4) && (sb <= 23)) {
632 for (ch = 0; ch < q->nb_channels; ch++)
633 for (i = 0; i < 64; i++) {
634 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
635 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
636 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
637 q->tone_level_idx_hi2[ch][sb - 4];
638 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
639 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
640 q->tone_level[ch][sb][i] = 0;
642 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
646 for (ch = 0; ch < q->nb_channels; ch++)
647 for (i = 0; i < 64; i++) {
648 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
649 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
650 q->tone_level_idx_hi2[ch][sb - 4];
651 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
652 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
653 q->tone_level[ch][sb][i] = 0;
655 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
658 for (ch = 0; ch < q->nb_channels; ch++)
659 for (i = 0; i < 64; i++) {
660 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
661 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
662 q->tone_level[ch][sb][i] = 0;
664 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
673 * Related to synthesis filter
674 * Called by process_subpacket_11
675 * c is built with data from subpacket 11
676 * Most of this function is used only if superblock_type_2_3 == 0,
677 * never seen it in samples.
679 * @param tone_level_idx
680 * @param tone_level_idx_temp
681 * @param coding_method q->coding_method[0][0][0]
682 * @param nb_channels number of channels
683 * @param c coming from subpacket 11, passed as 8*c
684 * @param superblocktype_2_3 flag based on superblock packet type
685 * @param cm_table_select q->cm_table_select
687 static void fill_coding_method_array(sb_int8_array tone_level_idx,
688 sb_int8_array tone_level_idx_temp,
689 sb_int8_array coding_method,
691 int c, int superblocktype_2_3,
695 int tmp, acc, esp_40, comp;
696 int add1, add2, add3, add4;
699 if (!superblocktype_2_3) {
700 /* This case is untested, no samples available */
701 avpriv_request_sample(NULL, "!superblocktype_2_3");
703 for (ch = 0; ch < nb_channels; ch++)
704 for (sb = 0; sb < 30; sb++) {
705 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
706 add1 = tone_level_idx[ch][sb][j] - 10;
709 add2 = add3 = add4 = 0;
711 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
716 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
721 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
725 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
728 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
730 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
733 for (ch = 0; ch < nb_channels; ch++)
734 for (sb = 0; sb < 30; sb++)
735 for (j = 0; j < 64; j++)
736 acc += tone_level_idx_temp[ch][sb][j];
738 multres = 0x66666667LL * (acc * 10);
739 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
740 for (ch = 0; ch < nb_channels; ch++)
741 for (sb = 0; sb < 30; sb++)
742 for (j = 0; j < 64; j++) {
743 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
746 comp /= 256; // signed shift
774 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
776 for (sb = 0; sb < 30; sb++)
777 fix_coding_method_array(sb, nb_channels, coding_method);
778 for (ch = 0; ch < nb_channels; ch++)
779 for (sb = 0; sb < 30; sb++)
780 for (j = 0; j < 64; j++)
782 if (coding_method[ch][sb][j] < 10)
783 coding_method[ch][sb][j] = 10;
786 if (coding_method[ch][sb][j] < 16)
787 coding_method[ch][sb][j] = 16;
789 if (coding_method[ch][sb][j] < 30)
790 coding_method[ch][sb][j] = 30;
793 } else { // superblocktype_2_3 != 0
794 for (ch = 0; ch < nb_channels; ch++)
795 for (sb = 0; sb < 30; sb++)
796 for (j = 0; j < 64; j++)
797 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
803 * Called by process_subpacket_11 to process more data from subpacket 11
805 * Called by process_subpacket_12 to process data from subpacket 12 with
809 * @param gb bitreader context
810 * @param length packet length in bits
811 * @param sb_min lower subband processed (sb_min included)
812 * @param sb_max higher subband processed (sb_max excluded)
814 static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
815 int length, int sb_min, int sb_max)
817 int sb, j, k, n, ch, run, channels;
818 int joined_stereo, zero_encoding, chs;
820 float type34_div = 0;
821 float type34_predictor;
822 float samples[10], sign_bits[16];
825 // If no data use noise
826 for (sb=sb_min; sb < sb_max; sb++)
827 build_sb_samples_from_noise (q, sb);
832 for (sb = sb_min; sb < sb_max; sb++) {
833 channels = q->nb_channels;
835 if (q->nb_channels <= 1 || sb < 12)
840 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0;
843 if (get_bits_left(gb) >= 16)
844 for (j = 0; j < 16; j++)
845 sign_bits[j] = get_bits1 (gb);
847 if (q->coding_method[0][sb][0] <= 0) {
848 av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
849 return AVERROR_INVALIDDATA;
852 for (j = 0; j < 64; j++)
853 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
854 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
856 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
860 for (ch = 0; ch < channels; ch++) {
861 FIX_NOISE_IDX(q->noise_idx);
862 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
863 type34_predictor = 0.0;
866 for (j = 0; j < 128; ) {
867 switch (q->coding_method[ch][sb][j / 2]) {
869 if (get_bits_left(gb) >= 10) {
871 for (k = 0; k < 5; k++) {
872 if ((j + 2 * k) >= 128)
874 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
879 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
880 return AVERROR_INVALIDDATA;
883 for (k = 0; k < 5; k++)
884 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
886 for (k = 0; k < 5; k++)
887 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
889 for (k = 0; k < 10; k++)
890 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
896 if (get_bits_left(gb) >= 1) {
901 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
904 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
910 if (get_bits_left(gb) >= 10) {
912 for (k = 0; k < 5; k++) {
915 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
918 n = get_bits (gb, 8);
920 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
921 return AVERROR_INVALIDDATA;
924 for (k = 0; k < 5; k++)
925 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
928 for (k = 0; k < 5; k++)
929 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
935 if (get_bits_left(gb) >= 7) {
938 av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
939 return AVERROR_INVALIDDATA;
942 for (k = 0; k < 3; k++)
943 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
945 for (k = 0; k < 3; k++)
946 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
952 if (get_bits_left(gb) >= 4) {
953 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
954 if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
955 av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
956 return AVERROR_INVALIDDATA;
958 samples[0] = type30_dequant[index];
960 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
966 if (get_bits_left(gb) >= 7) {
968 type34_div = (float)(1 << get_bits(gb, 2));
969 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
970 type34_predictor = samples[0];
973 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
974 if (index >= FF_ARRAY_ELEMS(type34_delta)) {
975 av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
976 return AVERROR_INVALIDDATA;
978 samples[0] = type34_delta[index] / type34_div + type34_predictor;
979 type34_predictor = samples[0];
982 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
988 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
994 float tmp[10][MPA_MAX_CHANNELS];
995 for (k = 0; k < run; k++) {
996 tmp[k][0] = samples[k];
998 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
1000 for (chs = 0; chs < q->nb_channels; chs++)
1001 for (k = 0; k < run; k++)
1003 q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
1005 for (k = 0; k < run; k++)
1007 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
1018 * Init the first element of a channel in quantized_coeffs with data
1019 * from packet 10 (quantized_coeffs[ch][0]).
1020 * This is similar to process_subpacket_9, but for a single channel
1021 * and for element [0]
1022 * same VLC tables as process_subpacket_9 are used.
1024 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
1025 * @param gb bitreader context
1027 static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
1030 int i, k, run, level, diff;
1032 if (get_bits_left(gb) < 16)
1034 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
1036 quantized_coeffs[0] = level;
1038 for (i = 0; i < 7; ) {
1039 if (get_bits_left(gb) < 16)
1041 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
1046 if (get_bits_left(gb) < 16)
1048 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
1050 for (k = 1; k <= run; k++)
1051 quantized_coeffs[i + k] = (level + ((k * diff) / run));
1060 * Related to synthesis filter, process data from packet 10
1061 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1062 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
1063 * data from packet 10
1066 * @param gb bitreader context
1068 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
1070 int sb, j, k, n, ch;
1072 for (ch = 0; ch < q->nb_channels; ch++) {
1073 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
1075 if (get_bits_left(gb) < 16) {
1076 memset(q->quantized_coeffs[ch][0], 0, 8);
1081 n = q->sub_sampling + 1;
1083 for (sb = 0; sb < n; sb++)
1084 for (ch = 0; ch < q->nb_channels; ch++)
1085 for (j = 0; j < 8; j++) {
1086 if (get_bits_left(gb) < 1)
1088 if (get_bits1(gb)) {
1089 for (k=0; k < 8; k++) {
1090 if (get_bits_left(gb) < 16)
1092 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1095 for (k=0; k < 8; k++)
1096 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1100 n = QDM2_SB_USED(q->sub_sampling) - 4;
1102 for (sb = 0; sb < n; sb++)
1103 for (ch = 0; ch < q->nb_channels; ch++) {
1104 if (get_bits_left(gb) < 16)
1106 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1108 q->tone_level_idx_hi2[ch][sb] -= 16;
1110 for (j = 0; j < 8; j++)
1111 q->tone_level_idx_mid[ch][sb][j] = -16;
1114 n = QDM2_SB_USED(q->sub_sampling) - 5;
1116 for (sb = 0; sb < n; sb++)
1117 for (ch = 0; ch < q->nb_channels; ch++)
1118 for (j = 0; j < 8; j++) {
1119 if (get_bits_left(gb) < 16)
1121 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1126 * Process subpacket 9, init quantized_coeffs with data from it
1129 * @param node pointer to node with packet
1131 static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
1134 int i, j, k, n, ch, run, level, diff;
1136 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1138 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
1140 for (i = 1; i < n; i++)
1141 for (ch = 0; ch < q->nb_channels; ch++) {
1142 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1143 q->quantized_coeffs[ch][i][0] = level;
1145 for (j = 0; j < (8 - 1); ) {
1146 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1147 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1152 for (k = 1; k <= run; k++)
1153 q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
1160 for (ch = 0; ch < q->nb_channels; ch++)
1161 for (i = 0; i < 8; i++)
1162 q->quantized_coeffs[ch][0][i] = 0;
1168 * Process subpacket 10 if not null, else
1171 * @param node pointer to node with packet
1173 static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
1178 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1179 init_tone_level_dequantization(q, &gb);
1180 fill_tone_level_array(q, 1);
1182 fill_tone_level_array(q, 0);
1187 * Process subpacket 11
1190 * @param node pointer to node with packet
1192 static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
1198 length = node->packet->size * 8;
1199 init_get_bits(&gb, node->packet->data, length);
1203 int c = get_bits(&gb, 13);
1206 fill_coding_method_array(q->tone_level_idx,
1207 q->tone_level_idx_temp, q->coding_method,
1208 q->nb_channels, 8 * c,
1209 q->superblocktype_2_3, q->cm_table_select);
1212 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1216 * Process subpacket 12
1219 * @param node pointer to node with packet
1221 static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
1227 length = node->packet->size * 8;
1228 init_get_bits(&gb, node->packet->data, length);
1231 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1235 * Process new subpackets for synthesis filter
1238 * @param list list with synthesis filter packets (list D)
1240 static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
1242 QDM2SubPNode *nodes[4];
1244 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1245 if (nodes[0] != NULL)
1246 process_subpacket_9(q, nodes[0]);
1248 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1249 if (nodes[1] != NULL)
1250 process_subpacket_10(q, nodes[1]);
1252 process_subpacket_10(q, NULL);
1254 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1255 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1256 process_subpacket_11(q, nodes[2]);
1258 process_subpacket_11(q, NULL);
1260 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1261 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1262 process_subpacket_12(q, nodes[3]);
1264 process_subpacket_12(q, NULL);
1268 * Decode superblock, fill packet lists.
1272 static void qdm2_decode_super_block(QDM2Context *q)
1275 QDM2SubPacket header, *packet;
1276 int i, packet_bytes, sub_packet_size, sub_packets_D;
1277 unsigned int next_index = 0;
1279 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1280 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1281 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1283 q->sub_packets_B = 0;
1286 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1288 init_get_bits(&gb, q->compressed_data, q->compressed_size * 8);
1289 qdm2_decode_sub_packet_header(&gb, &header);
1291 if (header.type < 2 || header.type >= 8) {
1293 av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1297 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1298 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1300 init_get_bits(&gb, header.data, header.size * 8);
1302 if (header.type == 2 || header.type == 4 || header.type == 5) {
1303 int csum = 257 * get_bits(&gb, 8);
1304 csum += 2 * get_bits(&gb, 8);
1306 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1310 av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1315 q->sub_packet_list_B[0].packet = NULL;
1316 q->sub_packet_list_D[0].packet = NULL;
1318 for (i = 0; i < 6; i++)
1319 if (--q->fft_level_exp[i] < 0)
1320 q->fft_level_exp[i] = 0;
1322 for (i = 0; packet_bytes > 0; i++) {
1325 if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1326 SAMPLES_NEEDED_2("too many packet bytes");
1330 q->sub_packet_list_A[i].next = NULL;
1333 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1335 /* seek to next block */
1336 init_get_bits(&gb, header.data, header.size * 8);
1337 skip_bits(&gb, next_index * 8);
1339 if (next_index >= header.size)
1343 /* decode subpacket */
1344 packet = &q->sub_packets[i];
1345 qdm2_decode_sub_packet_header(&gb, packet);
1346 next_index = packet->size + get_bits_count(&gb) / 8;
1347 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1349 if (packet->type == 0)
1352 if (sub_packet_size > packet_bytes) {
1353 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1355 packet->size += packet_bytes - sub_packet_size;
1358 packet_bytes -= sub_packet_size;
1360 /* add subpacket to 'all subpackets' list */
1361 q->sub_packet_list_A[i].packet = packet;
1363 /* add subpacket to related list */
1364 if (packet->type == 8) {
1365 SAMPLES_NEEDED_2("packet type 8");
1367 } else if (packet->type >= 9 && packet->type <= 12) {
1368 /* packets for MPEG Audio like Synthesis Filter */
1369 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1370 } else if (packet->type == 13) {
1371 for (j = 0; j < 6; j++)
1372 q->fft_level_exp[j] = get_bits(&gb, 6);
1373 } else if (packet->type == 14) {
1374 for (j = 0; j < 6; j++)
1375 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1376 } else if (packet->type == 15) {
1377 SAMPLES_NEEDED_2("packet type 15")
1379 } else if (packet->type >= 16 && packet->type < 48 &&
1380 !fft_subpackets[packet->type - 16]) {
1381 /* packets for FFT */
1382 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1384 } // Packet bytes loop
1386 if (q->sub_packet_list_D[0].packet != NULL) {
1387 process_synthesis_subpackets(q, q->sub_packet_list_D);
1388 q->do_synth_filter = 1;
1389 } else if (q->do_synth_filter) {
1390 process_subpacket_10(q, NULL);
1391 process_subpacket_11(q, NULL);
1392 process_subpacket_12(q, NULL);
1396 static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1397 int offset, int duration, int channel,
1400 if (q->fft_coefs_min_index[duration] < 0)
1401 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1403 q->fft_coefs[q->fft_coefs_index].sub_packet =
1404 ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1405 q->fft_coefs[q->fft_coefs_index].channel = channel;
1406 q->fft_coefs[q->fft_coefs_index].offset = offset;
1407 q->fft_coefs[q->fft_coefs_index].exp = exp;
1408 q->fft_coefs[q->fft_coefs_index].phase = phase;
1409 q->fft_coefs_index++;
1412 static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
1413 GetBitContext *gb, int b)
1415 int channel, stereo, phase, exp;
1416 int local_int_4, local_int_8, stereo_phase, local_int_10;
1417 int local_int_14, stereo_exp, local_int_20, local_int_28;
1423 local_int_8 = (4 - duration);
1424 local_int_10 = 1 << (q->group_order - duration - 1);
1427 while (get_bits_left(gb)>0) {
1428 if (q->superblocktype_2_3) {
1429 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1430 if (get_bits_left(gb)<0) {
1431 if(local_int_4 < q->group_size)
1432 av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1437 local_int_4 += local_int_10;
1438 local_int_28 += (1 << local_int_8);
1440 local_int_4 += 8 * local_int_10;
1441 local_int_28 += (8 << local_int_8);
1446 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1447 while (offset >= (local_int_10 - 1)) {
1448 offset += (1 - (local_int_10 - 1));
1449 local_int_4 += local_int_10;
1450 local_int_28 += (1 << local_int_8);
1454 if (local_int_4 >= q->group_size)
1457 local_int_14 = (offset >> local_int_8);
1458 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1461 if (q->nb_channels > 1) {
1462 channel = get_bits1(gb);
1463 stereo = get_bits1(gb);
1469 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1470 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1471 exp = (exp < 0) ? 0 : exp;
1473 phase = get_bits(gb, 3);
1478 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1479 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1480 if (stereo_phase < 0)
1484 if (q->frequency_range > (local_int_14 + 1)) {
1485 int sub_packet = (local_int_20 + local_int_28);
1487 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1488 channel, exp, phase);
1490 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1492 stereo_exp, stereo_phase);
1498 static void qdm2_decode_fft_packets(QDM2Context *q)
1500 int i, j, min, max, value, type, unknown_flag;
1503 if (q->sub_packet_list_B[0].packet == NULL)
1506 /* reset minimum indexes for FFT coefficients */
1507 q->fft_coefs_index = 0;
1508 for (i = 0; i < 5; i++)
1509 q->fft_coefs_min_index[i] = -1;
1511 /* process subpackets ordered by type, largest type first */
1512 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1513 QDM2SubPacket *packet = NULL;
1515 /* find subpacket with largest type less than max */
1516 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1517 value = q->sub_packet_list_B[j].packet->type;
1518 if (value > min && value < max) {
1520 packet = q->sub_packet_list_B[j].packet;
1526 /* check for errors (?) */
1531 (packet->type < 16 || packet->type >= 48 ||
1532 fft_subpackets[packet->type - 16]))
1535 /* decode FFT tones */
1536 init_get_bits(&gb, packet->data, packet->size * 8);
1538 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1543 type = packet->type;
1545 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1546 int duration = q->sub_sampling + 5 - (type & 15);
1548 if (duration >= 0 && duration < 4)
1549 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1550 } else if (type == 31) {
1551 for (j = 0; j < 4; j++)
1552 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1553 } else if (type == 46) {
1554 for (j = 0; j < 6; j++)
1555 q->fft_level_exp[j] = get_bits(&gb, 6);
1556 for (j = 0; j < 4; j++)
1557 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1559 } // Loop on B packets
1561 /* calculate maximum indexes for FFT coefficients */
1562 for (i = 0, j = -1; i < 5; i++)
1563 if (q->fft_coefs_min_index[i] >= 0) {
1565 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1569 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1572 static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
1577 const double iscale = 2.0 * M_PI / 512.0;
1579 tone->phase += tone->phase_shift;
1581 /* calculate current level (maximum amplitude) of tone */
1582 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1583 c.im = level * sin(tone->phase * iscale);
1584 c.re = level * cos(tone->phase * iscale);
1586 /* generate FFT coefficients for tone */
1587 if (tone->duration >= 3 || tone->cutoff >= 3) {
1588 tone->complex[0].im += c.im;
1589 tone->complex[0].re += c.re;
1590 tone->complex[1].im -= c.im;
1591 tone->complex[1].re -= c.re;
1593 f[1] = -tone->table[4];
1594 f[0] = tone->table[3] - tone->table[0];
1595 f[2] = 1.0 - tone->table[2] - tone->table[3];
1596 f[3] = tone->table[1] + tone->table[4] - 1.0;
1597 f[4] = tone->table[0] - tone->table[1];
1598 f[5] = tone->table[2];
1599 for (i = 0; i < 2; i++) {
1600 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1602 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1603 c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1605 for (i = 0; i < 4; i++) {
1606 tone->complex[i].re += c.re * f[i + 2];
1607 tone->complex[i].im += c.im * f[i + 2];
1611 /* copy the tone if it has not yet died out */
1612 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1613 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1614 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1618 static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1621 const double iscale = 0.25 * M_PI;
1623 for (ch = 0; ch < q->channels; ch++) {
1624 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1628 /* apply FFT tones with duration 4 (1 FFT period) */
1629 if (q->fft_coefs_min_index[4] >= 0)
1630 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1634 if (q->fft_coefs[i].sub_packet != sub_packet)
1637 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1638 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];
1640 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1641 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1642 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1643 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1644 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1645 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1648 /* generate existing FFT tones */
1649 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1650 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1651 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1654 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1655 for (i = 0; i < 4; i++)
1656 if (q->fft_coefs_min_index[i] >= 0) {
1657 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1661 if (q->fft_coefs[j].sub_packet != sub_packet)
1665 offset = q->fft_coefs[j].offset >> four_i;
1666 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1668 if (offset < q->frequency_range) {
1670 tone.cutoff = offset;
1672 tone.cutoff = (offset >= 60) ? 3 : 2;
1674 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];
1675 tone.complex = &q->fft.complex[ch][offset];
1676 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1677 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1678 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1680 tone.time_index = 0;
1682 qdm2_fft_generate_tone(q, &tone);
1685 q->fft_coefs_min_index[i] = j;
1689 static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1691 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1692 float *out = q->output_buffer + channel;
1694 q->fft.complex[channel][0].re *= 2.0f;
1695 q->fft.complex[channel][0].im = 0.0f;
1696 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1697 /* add samples to output buffer */
1698 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1699 out[0] += q->fft.complex[channel][i].re * gain;
1700 out[q->channels] += q->fft.complex[channel][i].im * gain;
1701 out += 2 * q->channels;
1707 * @param index subpacket number
1709 static void qdm2_synthesis_filter(QDM2Context *q, int index)
1711 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1713 /* copy sb_samples */
1714 sb_used = QDM2_SB_USED(q->sub_sampling);
1716 for (ch = 0; ch < q->channels; ch++)
1717 for (i = 0; i < 8; i++)
1718 for (k = sb_used; k < SBLIMIT; k++)
1719 q->sb_samples[ch][(8 * index) + i][k] = 0;
1721 for (ch = 0; ch < q->nb_channels; ch++) {
1722 float *samples_ptr = q->samples + ch;
1724 for (i = 0; i < 8; i++) {
1725 ff_mpa_synth_filter_float(&q->mpadsp,
1726 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1727 ff_mpa_synth_window_float, &dither_state,
1728 samples_ptr, q->nb_channels,
1729 q->sb_samples[ch][(8 * index) + i]);
1730 samples_ptr += 32 * q->nb_channels;
1734 /* add samples to output buffer */
1735 sub_sampling = (4 >> q->sub_sampling);
1737 for (ch = 0; ch < q->channels; ch++)
1738 for (i = 0; i < q->frame_size; i++)
1739 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1743 * Init static data (does not depend on specific file)
1747 static av_cold void qdm2_init_static_data(AVCodec *codec) {
1749 ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1750 softclip_table_init();
1752 init_noise_samples();
1756 * Init parameters from codec extradata
1758 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1760 QDM2Context *s = avctx->priv_data;
1763 int tmp_val, tmp, size;
1765 /* extradata parsing
1774 32 size (including this field)
1776 32 type (=QDM2 or QDMC)
1778 32 size (including this field, in bytes)
1779 32 tag (=QDCA) // maybe mandatory parameters
1782 32 samplerate (=44100)
1784 32 block size (=4096)
1785 32 frame size (=256) (for one channel)
1786 32 packet size (=1300)
1788 32 size (including this field, in bytes)
1789 32 tag (=QDCP) // maybe some tuneable parameters
1799 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1800 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1804 extradata = avctx->extradata;
1805 extradata_size = avctx->extradata_size;
1807 while (extradata_size > 7) {
1808 if (!memcmp(extradata, "frmaQDM", 7))
1814 if (extradata_size < 12) {
1815 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1820 if (memcmp(extradata, "frmaQDM", 7)) {
1821 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1825 if (extradata[7] == 'C') {
1827 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1832 extradata_size -= 8;
1834 size = AV_RB32(extradata);
1836 if(size > extradata_size){
1837 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1838 extradata_size, size);
1843 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1844 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1845 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1851 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1853 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1854 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1855 return AVERROR_INVALIDDATA;
1857 avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1860 avctx->sample_rate = AV_RB32(extradata);
1863 avctx->bit_rate = AV_RB32(extradata);
1866 s->group_size = AV_RB32(extradata);
1869 s->fft_size = AV_RB32(extradata);
1872 s->checksum_size = AV_RB32(extradata);
1873 if (s->checksum_size >= 1U << 28) {
1874 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1875 return AVERROR_INVALIDDATA;
1878 s->fft_order = av_log2(s->fft_size) + 1;
1880 // something like max decodable tones
1881 s->group_order = av_log2(s->group_size) + 1;
1882 s->frame_size = s->group_size / 16; // 16 iterations per super block
1884 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1885 return AVERROR_INVALIDDATA;
1887 s->sub_sampling = s->fft_order - 7;
1888 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1890 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1891 case 0: tmp = 40; break;
1892 case 1: tmp = 48; break;
1893 case 2: tmp = 56; break;
1894 case 3: tmp = 72; break;
1895 case 4: tmp = 80; break;
1896 case 5: tmp = 100;break;
1897 default: tmp=s->sub_sampling; break;
1900 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1901 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1902 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1903 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1904 s->cm_table_select = tmp_val;
1906 if (avctx->bit_rate <= 8000)
1907 s->coeff_per_sb_select = 0;
1908 else if (avctx->bit_rate < 16000)
1909 s->coeff_per_sb_select = 1;
1911 s->coeff_per_sb_select = 2;
1913 // Fail on unknown fft order
1914 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1915 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1918 if (s->fft_size != (1 << (s->fft_order - 1))) {
1919 av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1920 return AVERROR_INVALIDDATA;
1923 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1924 ff_mpadsp_init(&s->mpadsp);
1926 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1931 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1933 QDM2Context *s = avctx->priv_data;
1935 ff_rdft_end(&s->rdft_ctx);
1940 static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1943 const int frame_size = (q->frame_size * q->channels);
1945 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1948 /* select input buffer */
1949 q->compressed_data = in;
1950 q->compressed_size = q->checksum_size;
1952 /* copy old block, clear new block of output samples */
1953 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1954 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1956 /* decode block of QDM2 compressed data */
1957 if (q->sub_packet == 0) {
1958 q->has_errors = 0; // zero it for a new super block
1959 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1960 qdm2_decode_super_block(q);
1963 /* parse subpackets */
1964 if (!q->has_errors) {
1965 if (q->sub_packet == 2)
1966 qdm2_decode_fft_packets(q);
1968 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1971 /* sound synthesis stage 1 (FFT) */
1972 for (ch = 0; ch < q->channels; ch++) {
1973 qdm2_calculate_fft(q, ch, q->sub_packet);
1975 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1976 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1981 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1982 if (!q->has_errors && q->do_synth_filter)
1983 qdm2_synthesis_filter(q, q->sub_packet);
1985 q->sub_packet = (q->sub_packet + 1) % 16;
1987 /* clip and convert output float[] to 16bit signed samples */
1988 for (i = 0; i < frame_size; i++) {
1989 int value = (int)q->output_buffer[i];
1991 if (value > SOFTCLIP_THRESHOLD)
1992 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1993 else if (value < -SOFTCLIP_THRESHOLD)
1994 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
2002 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
2003 int *got_frame_ptr, AVPacket *avpkt)
2005 AVFrame *frame = data;
2006 const uint8_t *buf = avpkt->data;
2007 int buf_size = avpkt->size;
2008 QDM2Context *s = avctx->priv_data;
2014 if(buf_size < s->checksum_size)
2017 /* get output buffer */
2018 frame->nb_samples = 16 * s->frame_size;
2019 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
2021 out = (int16_t *)frame->data[0];
2023 for (i = 0; i < 16; i++) {
2024 if (qdm2_decode(s, buf, out) < 0)
2026 out += s->channels * s->frame_size;
2031 return s->checksum_size;
2034 AVCodec ff_qdm2_decoder = {
2036 .type = AVMEDIA_TYPE_AUDIO,
2037 .id = AV_CODEC_ID_QDM2,
2038 .priv_data_size = sizeof(QDM2Context),
2039 .init = qdm2_decode_init,
2040 .init_static_data = qdm2_init_static_data,
2041 .close = qdm2_decode_close,
2042 .decode = qdm2_decode_frame,
2043 .capabilities = CODEC_CAP_DR1,
2044 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),