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
43 #include "mpegaudiodsp.h"
44 #include "mpegaudio.h"
47 #include "qdm2_tablegen.h"
53 #define QDM2_LIST_ADD(list, size, packet) \
56 list[size - 1].next = &list[size]; \
58 list[size].packet = packet; \
59 list[size].next = NULL; \
63 // Result is 8, 16 or 30
64 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
66 #define FIX_NOISE_IDX(noise_idx) \
67 if ((noise_idx) >= 3840) \
68 (noise_idx) -= 3840; \
70 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
72 #define SAMPLES_NEEDED \
73 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
75 #define SAMPLES_NEEDED_2(why) \
76 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
78 #define QDM2_MAX_FRAME_SIZE 512
80 typedef int8_t sb_int8_array[2][30][64];
86 int type; ///< subpacket type
87 unsigned int size; ///< subpacket size
88 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
92 * A node in the subpacket list
94 typedef struct QDM2SubPNode {
95 QDM2SubPacket *packet; ///< packet
96 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
106 QDM2Complex *complex;
124 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
128 * QDM2 decoder context
133 /// Parameters from codec header, do not change during playback
134 int nb_channels; ///< number of channels
135 int channels; ///< number of channels
136 int group_size; ///< size of frame group (16 frames per group)
137 int fft_size; ///< size of FFT, in complex numbers
138 int checksum_size; ///< size of data block, used also for checksum
140 /// Parameters built from header parameters, do not change during playback
141 int group_order; ///< order of frame group
142 int fft_order; ///< order of FFT (actually fftorder+1)
143 int frame_size; ///< size of data frame
145 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
146 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
147 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
149 /// Packets and packet lists
150 QDM2SubPacket sub_packets[16]; ///< the packets themselves
151 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
152 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
153 int sub_packets_B; ///< number of packets on 'B' list
154 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
155 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
158 FFTTone fft_tones[1000];
161 FFTCoefficient fft_coefs[1000];
163 int fft_coefs_min_index[5];
164 int fft_coefs_max_index[5];
165 int fft_level_exp[6];
166 RDFTContext rdft_ctx;
170 const uint8_t *compressed_data;
172 float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
175 MPADSPContext mpadsp;
176 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
177 int synth_buf_offset[MPA_MAX_CHANNELS];
178 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
179 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
181 /// Mixed temporary data used in decoding
182 float tone_level[MPA_MAX_CHANNELS][30][64];
183 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
184 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
185 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
186 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
187 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
188 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
189 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
190 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
193 int has_errors; ///< packet has errors
194 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
195 int do_synth_filter; ///< used to perform or skip synthesis filter
198 int noise_idx; ///< index for dithering noise table
202 static VLC vlc_tab_level;
203 static VLC vlc_tab_diff;
204 static VLC vlc_tab_run;
205 static VLC fft_level_exp_alt_vlc;
206 static VLC fft_level_exp_vlc;
207 static VLC fft_stereo_exp_vlc;
208 static VLC fft_stereo_phase_vlc;
209 static VLC vlc_tab_tone_level_idx_hi1;
210 static VLC vlc_tab_tone_level_idx_mid;
211 static VLC vlc_tab_tone_level_idx_hi2;
212 static VLC vlc_tab_type30;
213 static VLC vlc_tab_type34;
214 static VLC vlc_tab_fft_tone_offset[5];
216 static const uint16_t qdm2_vlc_offs[] = {
217 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
220 static av_cold void qdm2_init_vlc(void)
222 static int vlcs_initialized = 0;
223 static VLC_TYPE qdm2_table[3838][2];
225 if (!vlcs_initialized) {
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, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
233 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
234 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
235 init_vlc (&vlc_tab_diff, 8, 37,
236 vlc_tab_diff_huffbits, 1, 1,
237 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
239 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
240 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
241 init_vlc (&vlc_tab_run, 5, 6,
242 vlc_tab_run_huffbits, 1, 1,
243 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
245 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
246 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
247 init_vlc (&fft_level_exp_alt_vlc, 8, 28,
248 fft_level_exp_alt_huffbits, 1, 1,
249 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
252 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
253 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
254 init_vlc (&fft_level_exp_vlc, 8, 20,
255 fft_level_exp_huffbits, 1, 1,
256 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
258 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
259 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
260 init_vlc (&fft_stereo_exp_vlc, 6, 7,
261 fft_stereo_exp_huffbits, 1, 1,
262 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
264 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
265 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
266 init_vlc (&fft_stereo_phase_vlc, 6, 9,
267 fft_stereo_phase_huffbits, 1, 1,
268 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
270 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
271 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
272 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
273 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
274 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
276 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
277 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
278 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
279 vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
280 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
282 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
283 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
284 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
285 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
286 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
288 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
289 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
290 init_vlc (&vlc_tab_type30, 6, 9,
291 vlc_tab_type30_huffbits, 1, 1,
292 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
294 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
295 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
296 init_vlc (&vlc_tab_type34, 5, 10,
297 vlc_tab_type34_huffbits, 1, 1,
298 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
300 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
301 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
302 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
303 vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
304 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
306 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
307 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
308 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
309 vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
310 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
312 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
313 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
314 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
315 vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
316 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
318 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
319 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
320 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
321 vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
322 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
324 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
325 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
326 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
327 vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
328 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
334 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
338 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
340 /* stage-2, 3 bits exponent escape sequence */
342 value = get_bits (gb, get_bits (gb, 3) + 1);
344 /* stage-3, optional */
349 av_log(0, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
353 tmp= vlc_stage3_values[value];
355 if ((value & ~3) > 0)
356 tmp += get_bits (gb, (value >> 2));
364 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
366 int value = qdm2_get_vlc (gb, vlc, 0, depth);
368 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
375 * @param data pointer to data to be checksum'ed
376 * @param length data length
377 * @param value checksum value
379 * @return 0 if checksum is OK
381 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
384 for (i=0; i < length; i++)
387 return (uint16_t)(value & 0xffff);
392 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
394 * @param gb bitreader context
395 * @param sub_packet packet under analysis
397 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
399 sub_packet->type = get_bits (gb, 8);
401 if (sub_packet->type == 0) {
402 sub_packet->size = 0;
403 sub_packet->data = NULL;
405 sub_packet->size = get_bits (gb, 8);
407 if (sub_packet->type & 0x80) {
408 sub_packet->size <<= 8;
409 sub_packet->size |= get_bits (gb, 8);
410 sub_packet->type &= 0x7f;
413 if (sub_packet->type == 0x7f)
414 sub_packet->type |= (get_bits (gb, 8) << 8);
416 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
419 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
420 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
425 * Return node pointer to first packet of requested type in list.
427 * @param list list of subpackets to be scanned
428 * @param type type of searched subpacket
429 * @return node pointer for subpacket if found, else NULL
431 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
433 while (list != NULL && list->packet != NULL) {
434 if (list->packet->type == type)
443 * Replace 8 elements with their average value.
444 * Called by qdm2_decode_superblock before starting subblock decoding.
448 static void average_quantized_coeffs (QDM2Context *q)
450 int i, j, n, ch, sum;
452 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
454 for (ch = 0; ch < q->nb_channels; ch++)
455 for (i = 0; i < n; i++) {
458 for (j = 0; j < 8; j++)
459 sum += q->quantized_coeffs[ch][i][j];
465 for (j=0; j < 8; j++)
466 q->quantized_coeffs[ch][i][j] = sum;
472 * Build subband samples with noise weighted by q->tone_level.
473 * Called by synthfilt_build_sb_samples.
476 * @param sb subband index
478 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
482 FIX_NOISE_IDX(q->noise_idx);
487 for (ch = 0; ch < q->nb_channels; ch++)
488 for (j = 0; j < 64; j++) {
489 q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
490 q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
496 * Called while processing data from subpackets 11 and 12.
497 * Used after making changes to coding_method array.
499 * @param sb subband index
500 * @param channels number of channels
501 * @param coding_method q->coding_method[0][0][0]
503 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
508 static const int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
510 for (ch = 0; ch < channels; ch++) {
511 for (j = 0; j < 64; ) {
512 if((coding_method[ch][sb][j] - 8) > 22) {
516 switch (switchtable[coding_method[ch][sb][j]-8]) {
517 case 0: run = 10; case_val = 10; break;
518 case 1: run = 1; case_val = 16; break;
519 case 2: run = 5; case_val = 24; break;
520 case 3: run = 3; case_val = 30; break;
521 case 4: run = 1; case_val = 30; break;
522 case 5: run = 1; case_val = 8; break;
523 default: run = 1; case_val = 8; break;
526 for (k = 0; k < run; k++)
528 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
531 //not debugged, almost never used
532 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
533 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
542 * Related to synthesis filter
543 * Called by process_subpacket_10
546 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
548 static void fill_tone_level_array (QDM2Context *q, int flag)
550 int i, sb, ch, sb_used;
553 // This should never happen
554 if (q->nb_channels <= 0)
557 for (ch = 0; ch < q->nb_channels; ch++)
558 for (sb = 0; sb < 30; sb++)
559 for (i = 0; i < 8; i++) {
560 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
561 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
562 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
564 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
567 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
570 sb_used = QDM2_SB_USED(q->sub_sampling);
572 if ((q->superblocktype_2_3 != 0) && !flag) {
573 for (sb = 0; sb < sb_used; sb++)
574 for (ch = 0; ch < q->nb_channels; ch++)
575 for (i = 0; i < 64; i++) {
576 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
577 if (q->tone_level_idx[ch][sb][i] < 0)
578 q->tone_level[ch][sb][i] = 0;
580 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
583 tab = q->superblocktype_2_3 ? 0 : 1;
584 for (sb = 0; sb < sb_used; sb++) {
585 if ((sb >= 4) && (sb <= 23)) {
586 for (ch = 0; ch < q->nb_channels; ch++)
587 for (i = 0; i < 64; i++) {
588 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
589 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
590 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
591 q->tone_level_idx_hi2[ch][sb - 4];
592 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
593 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
594 q->tone_level[ch][sb][i] = 0;
596 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
600 for (ch = 0; ch < q->nb_channels; ch++)
601 for (i = 0; i < 64; i++) {
602 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
603 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
604 q->tone_level_idx_hi2[ch][sb - 4];
605 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
606 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
607 q->tone_level[ch][sb][i] = 0;
609 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
612 for (ch = 0; ch < q->nb_channels; ch++)
613 for (i = 0; i < 64; i++) {
614 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
615 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
616 q->tone_level[ch][sb][i] = 0;
618 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
630 * Related to synthesis filter
631 * Called by process_subpacket_11
632 * c is built with data from subpacket 11
633 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
635 * @param tone_level_idx
636 * @param tone_level_idx_temp
637 * @param coding_method q->coding_method[0][0][0]
638 * @param nb_channels number of channels
639 * @param c coming from subpacket 11, passed as 8*c
640 * @param superblocktype_2_3 flag based on superblock packet type
641 * @param cm_table_select q->cm_table_select
643 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
644 sb_int8_array coding_method, int nb_channels,
645 int c, int superblocktype_2_3, int cm_table_select)
648 int tmp, acc, esp_40, comp;
649 int add1, add2, add3, add4;
652 // This should never happen
653 if (nb_channels <= 0)
656 if (!superblocktype_2_3) {
657 /* This case is untested, no samples available */
659 for (ch = 0; ch < nb_channels; ch++)
660 for (sb = 0; sb < 30; sb++) {
661 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
662 add1 = tone_level_idx[ch][sb][j] - 10;
665 add2 = add3 = add4 = 0;
667 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
672 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
677 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
681 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
684 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
686 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
689 for (ch = 0; ch < nb_channels; ch++)
690 for (sb = 0; sb < 30; sb++)
691 for (j = 0; j < 64; j++)
692 acc += tone_level_idx_temp[ch][sb][j];
694 multres = 0x66666667 * (acc * 10);
695 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
696 for (ch = 0; ch < nb_channels; ch++)
697 for (sb = 0; sb < 30; sb++)
698 for (j = 0; j < 64; j++) {
699 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
702 comp /= 256; // signed shift
730 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
732 for (sb = 0; sb < 30; sb++)
733 fix_coding_method_array(sb, nb_channels, coding_method);
734 for (ch = 0; ch < nb_channels; ch++)
735 for (sb = 0; sb < 30; sb++)
736 for (j = 0; j < 64; j++)
738 if (coding_method[ch][sb][j] < 10)
739 coding_method[ch][sb][j] = 10;
742 if (coding_method[ch][sb][j] < 16)
743 coding_method[ch][sb][j] = 16;
745 if (coding_method[ch][sb][j] < 30)
746 coding_method[ch][sb][j] = 30;
749 } else { // superblocktype_2_3 != 0
750 for (ch = 0; ch < nb_channels; ch++)
751 for (sb = 0; sb < 30; sb++)
752 for (j = 0; j < 64; j++)
753 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
762 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
763 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
766 * @param gb bitreader context
767 * @param length packet length in bits
768 * @param sb_min lower subband processed (sb_min included)
769 * @param sb_max higher subband processed (sb_max excluded)
771 static int synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
773 int sb, j, k, n, ch, run, channels;
774 int joined_stereo, zero_encoding, chs;
776 float type34_div = 0;
777 float type34_predictor;
778 float samples[10], sign_bits[16];
781 // If no data use noise
782 for (sb=sb_min; sb < sb_max; sb++)
783 build_sb_samples_from_noise (q, sb);
788 for (sb = sb_min; sb < sb_max; sb++) {
789 FIX_NOISE_IDX(q->noise_idx);
791 channels = q->nb_channels;
793 if (q->nb_channels <= 1 || sb < 12)
798 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0;
801 if (get_bits_left(gb) >= 16)
802 for (j = 0; j < 16; j++)
803 sign_bits[j] = get_bits1 (gb);
805 for (j = 0; j < 64; j++)
806 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
807 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
809 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
813 for (ch = 0; ch < channels; ch++) {
814 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
815 type34_predictor = 0.0;
818 for (j = 0; j < 128; ) {
819 switch (q->coding_method[ch][sb][j / 2]) {
821 if (get_bits_left(gb) >= 10) {
823 for (k = 0; k < 5; k++) {
824 if ((j + 2 * k) >= 128)
826 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
830 for (k = 0; k < 5; k++)
831 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
833 for (k = 0; k < 5; k++)
834 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
836 for (k = 0; k < 10; k++)
837 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
843 if (get_bits_left(gb) >= 1) {
848 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
851 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
857 if (get_bits_left(gb) >= 10) {
859 for (k = 0; k < 5; k++) {
862 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
865 n = get_bits (gb, 8);
866 for (k = 0; k < 5; k++)
867 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
870 for (k = 0; k < 5; k++)
871 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
877 if (get_bits_left(gb) >= 7) {
879 for (k = 0; k < 3; k++)
880 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
882 for (k = 0; k < 3; k++)
883 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
889 if (get_bits_left(gb) >= 4) {
890 unsigned v = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
891 if (v >= FF_ARRAY_ELEMS(type30_dequant))
892 return AVERROR_INVALIDDATA;
893 samples[0] = type30_dequant[v];
895 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
901 if (get_bits_left(gb) >= 7) {
903 type34_div = (float)(1 << get_bits(gb, 2));
904 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
905 type34_predictor = samples[0];
908 unsigned v = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
909 if (v >= FF_ARRAY_ELEMS(type34_delta))
910 return AVERROR_INVALIDDATA;
911 samples[0] = type34_delta[v] / type34_div + type34_predictor;
912 type34_predictor = samples[0];
915 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
921 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
927 float tmp[10][MPA_MAX_CHANNELS];
929 for (k = 0; k < run; k++) {
930 tmp[k][0] = samples[k];
931 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
933 for (chs = 0; chs < q->nb_channels; chs++)
934 for (k = 0; k < run; k++)
936 q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
938 for (k = 0; k < run; k++)
940 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
952 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
953 * This is similar to process_subpacket_9, but for a single channel and for element [0]
954 * same VLC tables as process_subpacket_9 are used.
956 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
957 * @param gb bitreader context
959 static int init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb)
961 int i, k, run, level, diff;
963 if (get_bits_left(gb) < 16)
965 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
967 quantized_coeffs[0] = level;
969 for (i = 0; i < 7; ) {
970 if (get_bits_left(gb) < 16)
972 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
977 if (get_bits_left(gb) < 16)
979 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
981 for (k = 1; k <= run; k++)
982 quantized_coeffs[i + k] = (level + ((k * diff) / run));
992 * Related to synthesis filter, process data from packet 10
993 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
994 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
997 * @param gb bitreader context
999 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
1001 int sb, j, k, n, ch;
1003 for (ch = 0; ch < q->nb_channels; ch++) {
1004 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
1006 if (get_bits_left(gb) < 16) {
1007 memset(q->quantized_coeffs[ch][0], 0, 8);
1012 n = q->sub_sampling + 1;
1014 for (sb = 0; sb < n; sb++)
1015 for (ch = 0; ch < q->nb_channels; ch++)
1016 for (j = 0; j < 8; j++) {
1017 if (get_bits_left(gb) < 1)
1019 if (get_bits1(gb)) {
1020 for (k=0; k < 8; k++) {
1021 if (get_bits_left(gb) < 16)
1023 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1026 for (k=0; k < 8; k++)
1027 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1031 n = QDM2_SB_USED(q->sub_sampling) - 4;
1033 for (sb = 0; sb < n; sb++)
1034 for (ch = 0; ch < q->nb_channels; ch++) {
1035 if (get_bits_left(gb) < 16)
1037 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1039 q->tone_level_idx_hi2[ch][sb] -= 16;
1041 for (j = 0; j < 8; j++)
1042 q->tone_level_idx_mid[ch][sb][j] = -16;
1045 n = QDM2_SB_USED(q->sub_sampling) - 5;
1047 for (sb = 0; sb < n; sb++)
1048 for (ch = 0; ch < q->nb_channels; ch++)
1049 for (j = 0; j < 8; j++) {
1050 if (get_bits_left(gb) < 16)
1052 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1057 * Process subpacket 9, init quantized_coeffs with data from it
1060 * @param node pointer to node with packet
1062 static int process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1065 int i, j, k, n, ch, run, level, diff;
1067 init_get_bits(&gb, node->packet->data, node->packet->size*8);
1069 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1071 for (i = 1; i < n; i++)
1072 for (ch=0; ch < q->nb_channels; ch++) {
1073 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1074 q->quantized_coeffs[ch][i][0] = level;
1076 for (j = 0; j < (8 - 1); ) {
1077 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1078 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1083 for (k = 1; k <= run; k++)
1084 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1091 for (ch = 0; ch < q->nb_channels; ch++)
1092 for (i = 0; i < 8; i++)
1093 q->quantized_coeffs[ch][0][i] = 0;
1100 * Process subpacket 10 if not null, else
1103 * @param node pointer to node with packet
1104 * @param length packet length in bits
1106 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node)
1111 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1112 init_tone_level_dequantization(q, &gb);
1113 fill_tone_level_array(q, 1);
1115 fill_tone_level_array(q, 0);
1121 * Process subpacket 11
1124 * @param node pointer to node with packet
1126 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node)
1132 length = node->packet->size * 8;
1133 init_get_bits(&gb, node->packet->data, length);
1137 int c = get_bits (&gb, 13);
1140 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1141 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1144 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1149 * Process subpacket 12
1152 * @param node pointer to node with packet
1153 * @param length packet length in bits
1155 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node)
1161 length = node->packet->size * 8;
1162 init_get_bits(&gb, node->packet->data, length);
1165 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1169 * Process new subpackets for synthesis filter
1172 * @param list list with synthesis filter packets (list D)
1174 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1176 QDM2SubPNode *nodes[4];
1178 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1179 if (nodes[0] != NULL)
1180 process_subpacket_9(q, nodes[0]);
1182 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1183 if (nodes[1] != NULL)
1184 process_subpacket_10(q, nodes[1]);
1186 process_subpacket_10(q, NULL);
1188 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1189 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1190 process_subpacket_11(q, nodes[2]);
1192 process_subpacket_11(q, NULL);
1194 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1195 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1196 process_subpacket_12(q, nodes[3]);
1198 process_subpacket_12(q, NULL);
1203 * Decode superblock, fill packet lists.
1207 static void qdm2_decode_super_block (QDM2Context *q)
1210 QDM2SubPacket header, *packet;
1211 int i, packet_bytes, sub_packet_size, sub_packets_D;
1212 unsigned int next_index = 0;
1214 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1215 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1216 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1218 q->sub_packets_B = 0;
1221 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1223 init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1224 qdm2_decode_sub_packet_header(&gb, &header);
1226 if (header.type < 2 || header.type >= 8) {
1228 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1232 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1233 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1235 init_get_bits(&gb, header.data, header.size*8);
1237 if (header.type == 2 || header.type == 4 || header.type == 5) {
1238 int csum = 257 * get_bits(&gb, 8);
1239 csum += 2 * get_bits(&gb, 8);
1241 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1245 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1250 q->sub_packet_list_B[0].packet = NULL;
1251 q->sub_packet_list_D[0].packet = NULL;
1253 for (i = 0; i < 6; i++)
1254 if (--q->fft_level_exp[i] < 0)
1255 q->fft_level_exp[i] = 0;
1257 for (i = 0; packet_bytes > 0; i++) {
1260 q->sub_packet_list_A[i].next = NULL;
1263 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1265 /* seek to next block */
1266 init_get_bits(&gb, header.data, header.size*8);
1267 skip_bits(&gb, next_index*8);
1269 if (next_index >= header.size)
1273 /* decode subpacket */
1274 packet = &q->sub_packets[i];
1275 qdm2_decode_sub_packet_header(&gb, packet);
1276 next_index = packet->size + get_bits_count(&gb) / 8;
1277 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1279 if (packet->type == 0)
1282 if (sub_packet_size > packet_bytes) {
1283 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1285 packet->size += packet_bytes - sub_packet_size;
1288 packet_bytes -= sub_packet_size;
1290 /* add subpacket to 'all subpackets' list */
1291 q->sub_packet_list_A[i].packet = packet;
1293 /* add subpacket to related list */
1294 if (packet->type == 8) {
1295 SAMPLES_NEEDED_2("packet type 8");
1297 } else if (packet->type >= 9 && packet->type <= 12) {
1298 /* packets for MPEG Audio like Synthesis Filter */
1299 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1300 } else if (packet->type == 13) {
1301 for (j = 0; j < 6; j++)
1302 q->fft_level_exp[j] = get_bits(&gb, 6);
1303 } else if (packet->type == 14) {
1304 for (j = 0; j < 6; j++)
1305 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1306 } else if (packet->type == 15) {
1307 SAMPLES_NEEDED_2("packet type 15")
1309 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1310 /* packets for FFT */
1311 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1313 } // Packet bytes loop
1315 /* **************************************************************** */
1316 if (q->sub_packet_list_D[0].packet != NULL) {
1317 process_synthesis_subpackets(q, q->sub_packet_list_D);
1318 q->do_synth_filter = 1;
1319 } else if (q->do_synth_filter) {
1320 process_subpacket_10(q, NULL);
1321 process_subpacket_11(q, NULL);
1322 process_subpacket_12(q, NULL);
1324 /* **************************************************************** */
1328 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1329 int offset, int duration, int channel,
1332 if (q->fft_coefs_min_index[duration] < 0)
1333 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1335 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1336 q->fft_coefs[q->fft_coefs_index].channel = channel;
1337 q->fft_coefs[q->fft_coefs_index].offset = offset;
1338 q->fft_coefs[q->fft_coefs_index].exp = exp;
1339 q->fft_coefs[q->fft_coefs_index].phase = phase;
1340 q->fft_coefs_index++;
1344 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1346 int channel, stereo, phase, exp;
1347 int local_int_4, local_int_8, stereo_phase, local_int_10;
1348 int local_int_14, stereo_exp, local_int_20, local_int_28;
1354 local_int_8 = (4 - duration);
1355 local_int_10 = 1 << (q->group_order - duration - 1);
1358 while (get_bits_left(gb)>0) {
1359 if (q->superblocktype_2_3) {
1360 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1361 if (get_bits_left(gb)<0) {
1362 av_log(0, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1367 local_int_4 += local_int_10;
1368 local_int_28 += (1 << local_int_8);
1370 local_int_4 += 8*local_int_10;
1371 local_int_28 += (8 << local_int_8);
1376 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1377 while (offset >= (local_int_10 - 1)) {
1378 offset += (1 - (local_int_10 - 1));
1379 local_int_4 += local_int_10;
1380 local_int_28 += (1 << local_int_8);
1384 if (local_int_4 >= q->group_size)
1387 local_int_14 = (offset >> local_int_8);
1388 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1391 if (q->nb_channels > 1) {
1392 channel = get_bits1(gb);
1393 stereo = get_bits1(gb);
1399 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1400 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1401 exp = (exp < 0) ? 0 : exp;
1403 phase = get_bits(gb, 3);
1408 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1409 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1410 if (stereo_phase < 0)
1414 if (q->frequency_range > (local_int_14 + 1)) {
1415 int sub_packet = (local_int_20 + local_int_28);
1417 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1419 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1427 static void qdm2_decode_fft_packets (QDM2Context *q)
1429 int i, j, min, max, value, type, unknown_flag;
1432 if (q->sub_packet_list_B[0].packet == NULL)
1435 /* reset minimum indexes for FFT coefficients */
1436 q->fft_coefs_index = 0;
1437 for (i=0; i < 5; i++)
1438 q->fft_coefs_min_index[i] = -1;
1440 /* process subpackets ordered by type, largest type first */
1441 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1442 QDM2SubPacket *packet= NULL;
1444 /* find subpacket with largest type less than max */
1445 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1446 value = q->sub_packet_list_B[j].packet->type;
1447 if (value > min && value < max) {
1449 packet = q->sub_packet_list_B[j].packet;
1455 /* check for errors (?) */
1459 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1462 /* decode FFT tones */
1463 init_get_bits (&gb, packet->data, packet->size*8);
1465 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1470 type = packet->type;
1472 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1473 int duration = q->sub_sampling + 5 - (type & 15);
1475 if (duration >= 0 && duration < 4)
1476 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1477 } else if (type == 31) {
1478 for (j=0; j < 4; j++)
1479 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1480 } else if (type == 46) {
1481 for (j=0; j < 6; j++)
1482 q->fft_level_exp[j] = get_bits(&gb, 6);
1483 for (j=0; j < 4; j++)
1484 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1486 } // Loop on B packets
1488 /* calculate maximum indexes for FFT coefficients */
1489 for (i = 0, j = -1; i < 5; i++)
1490 if (q->fft_coefs_min_index[i] >= 0) {
1492 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1496 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1500 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1505 const double iscale = 2.0*M_PI / 512.0;
1507 tone->phase += tone->phase_shift;
1509 /* calculate current level (maximum amplitude) of tone */
1510 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1511 c.im = level * sin(tone->phase*iscale);
1512 c.re = level * cos(tone->phase*iscale);
1514 /* generate FFT coefficients for tone */
1515 if (tone->duration >= 3 || tone->cutoff >= 3) {
1516 tone->complex[0].im += c.im;
1517 tone->complex[0].re += c.re;
1518 tone->complex[1].im -= c.im;
1519 tone->complex[1].re -= c.re;
1521 f[1] = -tone->table[4];
1522 f[0] = tone->table[3] - tone->table[0];
1523 f[2] = 1.0 - tone->table[2] - tone->table[3];
1524 f[3] = tone->table[1] + tone->table[4] - 1.0;
1525 f[4] = tone->table[0] - tone->table[1];
1526 f[5] = tone->table[2];
1527 for (i = 0; i < 2; i++) {
1528 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1529 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1531 for (i = 0; i < 4; i++) {
1532 tone->complex[i].re += c.re * f[i+2];
1533 tone->complex[i].im += c.im * f[i+2];
1537 /* copy the tone if it has not yet died out */
1538 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1539 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1540 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1545 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1548 const double iscale = 0.25 * M_PI;
1550 for (ch = 0; ch < q->channels; ch++) {
1551 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1555 /* apply FFT tones with duration 4 (1 FFT period) */
1556 if (q->fft_coefs_min_index[4] >= 0)
1557 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1561 if (q->fft_coefs[i].sub_packet != sub_packet)
1564 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1565 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];
1567 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1568 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1569 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1570 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1571 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1572 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1575 /* generate existing FFT tones */
1576 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1577 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1578 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1581 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1582 for (i = 0; i < 4; i++)
1583 if (q->fft_coefs_min_index[i] >= 0) {
1584 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1588 if (q->fft_coefs[j].sub_packet != sub_packet)
1592 offset = q->fft_coefs[j].offset >> four_i;
1593 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1595 if (offset < q->frequency_range) {
1597 tone.cutoff = offset;
1599 tone.cutoff = (offset >= 60) ? 3 : 2;
1601 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];
1602 tone.complex = &q->fft.complex[ch][offset];
1603 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1604 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1605 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1607 tone.time_index = 0;
1609 qdm2_fft_generate_tone(q, &tone);
1612 q->fft_coefs_min_index[i] = j;
1617 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1619 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1620 float *out = q->output_buffer + channel;
1622 q->fft.complex[channel][0].re *= 2.0f;
1623 q->fft.complex[channel][0].im = 0.0f;
1624 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1625 /* add samples to output buffer */
1626 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1627 out[0] += q->fft.complex[channel][i].re * gain;
1628 out[q->channels] += q->fft.complex[channel][i].im * gain;
1629 out += 2 * q->channels;
1636 * @param index subpacket number
1638 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1640 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1642 /* copy sb_samples */
1643 sb_used = QDM2_SB_USED(q->sub_sampling);
1645 for (ch = 0; ch < q->channels; ch++)
1646 for (i = 0; i < 8; i++)
1647 for (k=sb_used; k < SBLIMIT; k++)
1648 q->sb_samples[ch][(8 * index) + i][k] = 0;
1650 for (ch = 0; ch < q->nb_channels; ch++) {
1651 float *samples_ptr = q->samples + ch;
1653 for (i = 0; i < 8; i++) {
1654 ff_mpa_synth_filter_float(&q->mpadsp,
1655 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1656 ff_mpa_synth_window_float, &dither_state,
1657 samples_ptr, q->nb_channels,
1658 q->sb_samples[ch][(8 * index) + i]);
1659 samples_ptr += 32 * q->nb_channels;
1663 /* add samples to output buffer */
1664 sub_sampling = (4 >> q->sub_sampling);
1666 for (ch = 0; ch < q->channels; ch++)
1667 for (i = 0; i < q->frame_size; i++)
1668 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1673 * Init static data (does not depend on specific file)
1677 static av_cold void qdm2_init(QDM2Context *q) {
1678 static int initialized = 0;
1680 if (initialized != 0)
1685 ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1686 softclip_table_init();
1688 init_noise_samples();
1690 av_log(NULL, AV_LOG_DEBUG, "init done\n");
1695 static void dump_context(QDM2Context *q)
1698 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1699 PRINT("compressed_data",q->compressed_data);
1700 PRINT("compressed_size",q->compressed_size);
1701 PRINT("frame_size",q->frame_size);
1702 PRINT("checksum_size",q->checksum_size);
1703 PRINT("channels",q->channels);
1704 PRINT("nb_channels",q->nb_channels);
1705 PRINT("fft_size",q->fft_size);
1706 PRINT("sub_sampling",q->sub_sampling);
1707 PRINT("fft_order",q->fft_order);
1708 PRINT("group_order",q->group_order);
1709 PRINT("group_size",q->group_size);
1710 PRINT("sub_packet",q->sub_packet);
1711 PRINT("frequency_range",q->frequency_range);
1712 PRINT("has_errors",q->has_errors);
1713 PRINT("fft_tone_end",q->fft_tone_end);
1714 PRINT("fft_tone_start",q->fft_tone_start);
1715 PRINT("fft_coefs_index",q->fft_coefs_index);
1716 PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1717 PRINT("cm_table_select",q->cm_table_select);
1718 PRINT("noise_idx",q->noise_idx);
1720 for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1722 FFTTone *t = &q->fft_tones[i];
1724 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1725 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1726 // PRINT(" level", t->level);
1727 PRINT(" phase", t->phase);
1728 PRINT(" phase_shift", t->phase_shift);
1729 PRINT(" duration", t->duration);
1730 PRINT(" samples_im", t->samples_im);
1731 PRINT(" samples_re", t->samples_re);
1732 PRINT(" table", t->table);
1740 * Init parameters from codec extradata
1742 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1744 QDM2Context *s = avctx->priv_data;
1747 int tmp_val, tmp, size;
1749 /* extradata parsing
1758 32 size (including this field)
1760 32 type (=QDM2 or QDMC)
1762 32 size (including this field, in bytes)
1763 32 tag (=QDCA) // maybe mandatory parameters
1766 32 samplerate (=44100)
1768 32 block size (=4096)
1769 32 frame size (=256) (for one channel)
1770 32 packet size (=1300)
1772 32 size (including this field, in bytes)
1773 32 tag (=QDCP) // maybe some tuneable parameters
1783 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1784 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1788 extradata = avctx->extradata;
1789 extradata_size = avctx->extradata_size;
1791 while (extradata_size > 7) {
1792 if (!memcmp(extradata, "frmaQDM", 7))
1798 if (extradata_size < 12) {
1799 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1804 if (memcmp(extradata, "frmaQDM", 7)) {
1805 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1809 if (extradata[7] == 'C') {
1811 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1816 extradata_size -= 8;
1818 size = AV_RB32(extradata);
1820 if(size > extradata_size){
1821 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1822 extradata_size, size);
1827 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1828 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1829 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1835 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1837 if (s->channels > MPA_MAX_CHANNELS)
1838 return AVERROR_INVALIDDATA;
1840 avctx->sample_rate = AV_RB32(extradata);
1843 avctx->bit_rate = AV_RB32(extradata);
1846 s->group_size = AV_RB32(extradata);
1849 s->fft_size = AV_RB32(extradata);
1852 s->checksum_size = AV_RB32(extradata);
1853 if (s->checksum_size >= 1U << 28) {
1854 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1855 return AVERROR_INVALIDDATA;
1858 s->fft_order = av_log2(s->fft_size) + 1;
1860 // something like max decodable tones
1861 s->group_order = av_log2(s->group_size) + 1;
1862 s->frame_size = s->group_size / 16; // 16 iterations per super block
1864 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1865 return AVERROR_INVALIDDATA;
1867 s->sub_sampling = s->fft_order - 7;
1868 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1870 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1871 case 0: tmp = 40; break;
1872 case 1: tmp = 48; break;
1873 case 2: tmp = 56; break;
1874 case 3: tmp = 72; break;
1875 case 4: tmp = 80; break;
1876 case 5: tmp = 100;break;
1877 default: tmp=s->sub_sampling; break;
1880 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1881 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1882 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1883 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1884 s->cm_table_select = tmp_val;
1886 if (s->sub_sampling == 0)
1889 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1896 s->coeff_per_sb_select = 0;
1897 else if (tmp <= 16000)
1898 s->coeff_per_sb_select = 1;
1900 s->coeff_per_sb_select = 2;
1902 // Fail on unknown fft order
1903 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1904 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1908 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1909 ff_mpadsp_init(&s->mpadsp);
1913 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1915 avcodec_get_frame_defaults(&s->frame);
1916 avctx->coded_frame = &s->frame;
1923 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1925 QDM2Context *s = avctx->priv_data;
1927 ff_rdft_end(&s->rdft_ctx);
1933 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1936 const int frame_size = (q->frame_size * q->channels);
1938 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1941 /* select input buffer */
1942 q->compressed_data = in;
1943 q->compressed_size = q->checksum_size;
1947 /* copy old block, clear new block of output samples */
1948 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1949 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1951 /* decode block of QDM2 compressed data */
1952 if (q->sub_packet == 0) {
1953 q->has_errors = 0; // zero it for a new super block
1954 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1955 qdm2_decode_super_block(q);
1958 /* parse subpackets */
1959 if (!q->has_errors) {
1960 if (q->sub_packet == 2)
1961 qdm2_decode_fft_packets(q);
1963 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1966 /* sound synthesis stage 1 (FFT) */
1967 for (ch = 0; ch < q->channels; ch++) {
1968 qdm2_calculate_fft(q, ch, q->sub_packet);
1970 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1971 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1976 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1977 if (!q->has_errors && q->do_synth_filter)
1978 qdm2_synthesis_filter(q, q->sub_packet);
1980 q->sub_packet = (q->sub_packet + 1) % 16;
1982 /* clip and convert output float[] to 16bit signed samples */
1983 for (i = 0; i < frame_size; i++) {
1984 int value = (int)q->output_buffer[i];
1986 if (value > SOFTCLIP_THRESHOLD)
1987 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1988 else if (value < -SOFTCLIP_THRESHOLD)
1989 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1998 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1999 int *got_frame_ptr, AVPacket *avpkt)
2001 const uint8_t *buf = avpkt->data;
2002 int buf_size = avpkt->size;
2003 QDM2Context *s = avctx->priv_data;
2009 if(buf_size < s->checksum_size)
2012 /* get output buffer */
2013 s->frame.nb_samples = 16 * s->frame_size;
2014 if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
2015 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
2018 out = (int16_t *)s->frame.data[0];
2020 for (i = 0; i < 16; i++) {
2021 if (qdm2_decode(s, buf, out) < 0)
2023 out += s->channels * s->frame_size;
2027 *(AVFrame *)data = s->frame;
2029 return s->checksum_size;
2032 AVCodec ff_qdm2_decoder =
2035 .type = AVMEDIA_TYPE_AUDIO,
2036 .id = CODEC_ID_QDM2,
2037 .priv_data_size = sizeof(QDM2Context),
2038 .init = qdm2_decode_init,
2039 .close = qdm2_decode_close,
2040 .decode = qdm2_decode_frame,
2041 .capabilities = CODEC_CAP_DR1,
2042 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),