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"
45 #include "mpegaudiodsp.h"
46 #include "mpegaudio.h"
49 #include "qdm2_tablegen.h"
55 #define QDM2_LIST_ADD(list, size, packet) \
58 list[size - 1].next = &list[size]; \
60 list[size].packet = packet; \
61 list[size].next = NULL; \
65 // Result is 8, 16 or 30
66 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
68 #define FIX_NOISE_IDX(noise_idx) \
69 if ((noise_idx) >= 3840) \
70 (noise_idx) -= 3840; \
72 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
74 #define SAMPLES_NEEDED \
75 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
77 #define SAMPLES_NEEDED_2(why) \
78 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
80 #define QDM2_MAX_FRAME_SIZE 512
82 typedef int8_t sb_int8_array[2][30][64];
88 int type; ///< subpacket type
89 unsigned int size; ///< subpacket size
90 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
94 * A node in the subpacket list
96 typedef struct QDM2SubPNode {
97 QDM2SubPacket *packet; ///< packet
98 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
108 QDM2Complex *complex;
126 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
130 * QDM2 decoder context
135 /// Parameters from codec header, do not change during playback
136 int nb_channels; ///< number of channels
137 int channels; ///< number of channels
138 int group_size; ///< size of frame group (16 frames per group)
139 int fft_size; ///< size of FFT, in complex numbers
140 int checksum_size; ///< size of data block, used also for checksum
142 /// Parameters built from header parameters, do not change during playback
143 int group_order; ///< order of frame group
144 int fft_order; ///< order of FFT (actually fftorder+1)
145 int frame_size; ///< size of data frame
147 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
148 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
149 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
151 /// Packets and packet lists
152 QDM2SubPacket sub_packets[16]; ///< the packets themselves
153 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
154 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
155 int sub_packets_B; ///< number of packets on 'B' list
156 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
157 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
160 FFTTone fft_tones[1000];
163 FFTCoefficient fft_coefs[1000];
165 int fft_coefs_min_index[5];
166 int fft_coefs_max_index[5];
167 int fft_level_exp[6];
168 RDFTContext rdft_ctx;
172 const uint8_t *compressed_data;
174 float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
177 MPADSPContext mpadsp;
178 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
179 int synth_buf_offset[MPA_MAX_CHANNELS];
180 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
181 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
183 /// Mixed temporary data used in decoding
184 float tone_level[MPA_MAX_CHANNELS][30][64];
185 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
186 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
187 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
188 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
189 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
190 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
191 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
192 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
195 int has_errors; ///< packet has errors
196 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
197 int do_synth_filter; ///< used to perform or skip synthesis filter
200 int noise_idx; ///< index for dithering noise table
204 static VLC vlc_tab_level;
205 static VLC vlc_tab_diff;
206 static VLC vlc_tab_run;
207 static VLC fft_level_exp_alt_vlc;
208 static VLC fft_level_exp_vlc;
209 static VLC fft_stereo_exp_vlc;
210 static VLC fft_stereo_phase_vlc;
211 static VLC vlc_tab_tone_level_idx_hi1;
212 static VLC vlc_tab_tone_level_idx_mid;
213 static VLC vlc_tab_tone_level_idx_hi2;
214 static VLC vlc_tab_type30;
215 static VLC vlc_tab_type34;
216 static VLC vlc_tab_fft_tone_offset[5];
218 static const uint16_t qdm2_vlc_offs[] = {
219 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
222 static av_cold void qdm2_init_vlc(void)
224 static int vlcs_initialized = 0;
225 static VLC_TYPE qdm2_table[3838][2];
227 if (!vlcs_initialized) {
229 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
230 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
231 init_vlc (&vlc_tab_level, 8, 24,
232 vlc_tab_level_huffbits, 1, 1,
233 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
235 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
236 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
237 init_vlc (&vlc_tab_diff, 8, 37,
238 vlc_tab_diff_huffbits, 1, 1,
239 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
241 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
242 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
243 init_vlc (&vlc_tab_run, 5, 6,
244 vlc_tab_run_huffbits, 1, 1,
245 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
247 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
248 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
249 init_vlc (&fft_level_exp_alt_vlc, 8, 28,
250 fft_level_exp_alt_huffbits, 1, 1,
251 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
254 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
255 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
256 init_vlc (&fft_level_exp_vlc, 8, 20,
257 fft_level_exp_huffbits, 1, 1,
258 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
260 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
261 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
262 init_vlc (&fft_stereo_exp_vlc, 6, 7,
263 fft_stereo_exp_huffbits, 1, 1,
264 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
266 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
267 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
268 init_vlc (&fft_stereo_phase_vlc, 6, 9,
269 fft_stereo_phase_huffbits, 1, 1,
270 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
272 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
273 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
274 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
275 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
276 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
278 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
279 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
280 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
281 vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
282 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
284 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
285 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
286 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
287 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
288 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
290 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
291 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
292 init_vlc (&vlc_tab_type30, 6, 9,
293 vlc_tab_type30_huffbits, 1, 1,
294 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
296 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
297 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
298 init_vlc (&vlc_tab_type34, 5, 10,
299 vlc_tab_type34_huffbits, 1, 1,
300 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
302 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
303 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
304 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
305 vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
306 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
308 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
309 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
310 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
311 vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
312 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
314 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
315 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
316 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
317 vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
318 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
320 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
321 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
322 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
323 vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
324 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
326 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
327 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
328 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
329 vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
330 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
336 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
340 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
342 /* stage-2, 3 bits exponent escape sequence */
344 value = get_bits (gb, get_bits (gb, 3) + 1);
346 /* stage-3, optional */
351 av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
355 tmp= vlc_stage3_values[value];
357 if ((value & ~3) > 0)
358 tmp += get_bits (gb, (value >> 2));
366 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
368 int value = qdm2_get_vlc (gb, vlc, 0, depth);
370 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
377 * @param data pointer to data to be checksum'ed
378 * @param length data length
379 * @param value checksum value
381 * @return 0 if checksum is OK
383 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
386 for (i=0; i < length; i++)
389 return (uint16_t)(value & 0xffff);
394 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
396 * @param gb bitreader context
397 * @param sub_packet packet under analysis
399 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
401 sub_packet->type = get_bits (gb, 8);
403 if (sub_packet->type == 0) {
404 sub_packet->size = 0;
405 sub_packet->data = NULL;
407 sub_packet->size = get_bits (gb, 8);
409 if (sub_packet->type & 0x80) {
410 sub_packet->size <<= 8;
411 sub_packet->size |= get_bits (gb, 8);
412 sub_packet->type &= 0x7f;
415 if (sub_packet->type == 0x7f)
416 sub_packet->type |= (get_bits (gb, 8) << 8);
418 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
421 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
422 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
427 * Return node pointer to first packet of requested type in list.
429 * @param list list of subpackets to be scanned
430 * @param type type of searched subpacket
431 * @return node pointer for subpacket if found, else NULL
433 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
435 while (list != NULL && list->packet != NULL) {
436 if (list->packet->type == type)
445 * Replace 8 elements with their average value.
446 * Called by qdm2_decode_superblock before starting subblock decoding.
450 static void average_quantized_coeffs (QDM2Context *q)
452 int i, j, n, ch, sum;
454 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
456 for (ch = 0; ch < q->nb_channels; ch++)
457 for (i = 0; i < n; i++) {
460 for (j = 0; j < 8; j++)
461 sum += q->quantized_coeffs[ch][i][j];
467 for (j=0; j < 8; j++)
468 q->quantized_coeffs[ch][i][j] = sum;
474 * Build subband samples with noise weighted by q->tone_level.
475 * Called by synthfilt_build_sb_samples.
478 * @param sb subband index
480 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
484 FIX_NOISE_IDX(q->noise_idx);
489 for (ch = 0; ch < q->nb_channels; ch++)
490 for (j = 0; j < 64; j++) {
491 q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
492 q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
498 * Called while processing data from subpackets 11 and 12.
499 * Used after making changes to coding_method array.
501 * @param sb subband index
502 * @param channels number of channels
503 * @param coding_method q->coding_method[0][0][0]
505 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
510 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};
512 for (ch = 0; ch < channels; ch++) {
513 for (j = 0; j < 64; ) {
514 if((coding_method[ch][sb][j] - 8) > 22) {
518 switch (switchtable[coding_method[ch][sb][j]-8]) {
519 case 0: run = 10; case_val = 10; break;
520 case 1: run = 1; case_val = 16; break;
521 case 2: run = 5; case_val = 24; break;
522 case 3: run = 3; case_val = 30; break;
523 case 4: run = 1; case_val = 30; break;
524 case 5: run = 1; case_val = 8; break;
525 default: run = 1; case_val = 8; break;
528 for (k = 0; k < run; k++)
530 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
533 //not debugged, almost never used
534 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
535 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
544 * Related to synthesis filter
545 * Called by process_subpacket_10
548 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
550 static void fill_tone_level_array (QDM2Context *q, int flag)
552 int i, sb, ch, sb_used;
555 for (ch = 0; ch < q->nb_channels; ch++)
556 for (sb = 0; sb < 30; sb++)
557 for (i = 0; i < 8; i++) {
558 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
559 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
560 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
562 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
565 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
568 sb_used = QDM2_SB_USED(q->sub_sampling);
570 if ((q->superblocktype_2_3 != 0) && !flag) {
571 for (sb = 0; sb < sb_used; sb++)
572 for (ch = 0; ch < q->nb_channels; ch++)
573 for (i = 0; i < 64; i++) {
574 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
575 if (q->tone_level_idx[ch][sb][i] < 0)
576 q->tone_level[ch][sb][i] = 0;
578 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
581 tab = q->superblocktype_2_3 ? 0 : 1;
582 for (sb = 0; sb < sb_used; sb++) {
583 if ((sb >= 4) && (sb <= 23)) {
584 for (ch = 0; ch < q->nb_channels; ch++)
585 for (i = 0; i < 64; i++) {
586 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
587 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
588 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
589 q->tone_level_idx_hi2[ch][sb - 4];
590 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
591 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
592 q->tone_level[ch][sb][i] = 0;
594 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
598 for (ch = 0; ch < q->nb_channels; ch++)
599 for (i = 0; i < 64; i++) {
600 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
601 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
602 q->tone_level_idx_hi2[ch][sb - 4];
603 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
604 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
605 q->tone_level[ch][sb][i] = 0;
607 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
610 for (ch = 0; ch < q->nb_channels; ch++)
611 for (i = 0; i < 64; i++) {
612 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
613 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
614 q->tone_level[ch][sb][i] = 0;
616 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
628 * Related to synthesis filter
629 * Called by process_subpacket_11
630 * c is built with data from subpacket 11
631 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
633 * @param tone_level_idx
634 * @param tone_level_idx_temp
635 * @param coding_method q->coding_method[0][0][0]
636 * @param nb_channels number of channels
637 * @param c coming from subpacket 11, passed as 8*c
638 * @param superblocktype_2_3 flag based on superblock packet type
639 * @param cm_table_select q->cm_table_select
641 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
642 sb_int8_array coding_method, int nb_channels,
643 int c, int superblocktype_2_3, int cm_table_select)
646 int tmp, acc, esp_40, comp;
647 int add1, add2, add3, add4;
650 if (!superblocktype_2_3) {
651 /* This case is untested, no samples available */
653 for (ch = 0; ch < nb_channels; ch++)
654 for (sb = 0; sb < 30; sb++) {
655 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
656 add1 = tone_level_idx[ch][sb][j] - 10;
659 add2 = add3 = add4 = 0;
661 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
666 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
671 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
675 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
678 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
680 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
683 for (ch = 0; ch < nb_channels; ch++)
684 for (sb = 0; sb < 30; sb++)
685 for (j = 0; j < 64; j++)
686 acc += tone_level_idx_temp[ch][sb][j];
688 multres = 0x66666667 * (acc * 10);
689 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
690 for (ch = 0; ch < nb_channels; ch++)
691 for (sb = 0; sb < 30; sb++)
692 for (j = 0; j < 64; j++) {
693 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
696 comp /= 256; // signed shift
724 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
726 for (sb = 0; sb < 30; sb++)
727 fix_coding_method_array(sb, nb_channels, coding_method);
728 for (ch = 0; ch < nb_channels; ch++)
729 for (sb = 0; sb < 30; sb++)
730 for (j = 0; j < 64; j++)
732 if (coding_method[ch][sb][j] < 10)
733 coding_method[ch][sb][j] = 10;
736 if (coding_method[ch][sb][j] < 16)
737 coding_method[ch][sb][j] = 16;
739 if (coding_method[ch][sb][j] < 30)
740 coding_method[ch][sb][j] = 30;
743 } else { // superblocktype_2_3 != 0
744 for (ch = 0; ch < nb_channels; ch++)
745 for (sb = 0; sb < 30; sb++)
746 for (j = 0; j < 64; j++)
747 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
756 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
757 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
760 * @param gb bitreader context
761 * @param length packet length in bits
762 * @param sb_min lower subband processed (sb_min included)
763 * @param sb_max higher subband processed (sb_max excluded)
765 static int synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
767 int sb, j, k, n, ch, run, channels;
768 int joined_stereo, zero_encoding, chs;
770 float type34_div = 0;
771 float type34_predictor;
772 float samples[10], sign_bits[16];
775 // If no data use noise
776 for (sb=sb_min; sb < sb_max; sb++)
777 build_sb_samples_from_noise (q, sb);
782 for (sb = sb_min; sb < sb_max; sb++) {
783 FIX_NOISE_IDX(q->noise_idx);
785 channels = q->nb_channels;
787 if (q->nb_channels <= 1 || sb < 12)
792 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0;
795 if (get_bits_left(gb) >= 16)
796 for (j = 0; j < 16; j++)
797 sign_bits[j] = get_bits1 (gb);
799 if (q->coding_method[0][sb][0] <= 0) {
800 av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
801 return AVERROR_INVALIDDATA;
804 for (j = 0; j < 64; j++)
805 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
806 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
808 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
812 for (ch = 0; ch < channels; ch++) {
813 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
814 type34_predictor = 0.0;
817 for (j = 0; j < 128; ) {
818 switch (q->coding_method[ch][sb][j / 2]) {
820 if (get_bits_left(gb) >= 10) {
822 for (k = 0; k < 5; k++) {
823 if ((j + 2 * k) >= 128)
825 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
829 for (k = 0; k < 5; k++)
830 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
832 for (k = 0; k < 5; k++)
833 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
835 for (k = 0; k < 10; k++)
836 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
842 if (get_bits_left(gb) >= 1) {
847 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
850 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
856 if (get_bits_left(gb) >= 10) {
858 for (k = 0; k < 5; k++) {
861 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
864 n = get_bits (gb, 8);
865 for (k = 0; k < 5; k++)
866 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
869 for (k = 0; k < 5; k++)
870 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
876 if (get_bits_left(gb) >= 7) {
878 for (k = 0; k < 3; k++)
879 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
881 for (k = 0; k < 3; k++)
882 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
888 if (get_bits_left(gb) >= 4) {
889 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
890 if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
891 av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
892 return AVERROR_INVALIDDATA;
894 samples[0] = type30_dequant[index];
896 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
902 if (get_bits_left(gb) >= 7) {
904 type34_div = (float)(1 << get_bits(gb, 2));
905 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
906 type34_predictor = samples[0];
909 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
910 if (index >= FF_ARRAY_ELEMS(type34_delta)) {
911 av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
912 return AVERROR_INVALIDDATA;
914 samples[0] = type34_delta[index] / type34_div + type34_predictor;
915 type34_predictor = samples[0];
918 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
924 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
930 float tmp[10][MPA_MAX_CHANNELS];
931 for (k = 0; k < run; k++) {
932 tmp[k][0] = samples[k];
934 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
936 for (chs = 0; chs < q->nb_channels; chs++)
937 for (k = 0; k < run; k++)
939 q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
941 for (k = 0; k < run; k++)
943 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
955 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
956 * This is similar to process_subpacket_9, but for a single channel and for element [0]
957 * same VLC tables as process_subpacket_9 are used.
959 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
960 * @param gb bitreader context
962 static int init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb)
964 int i, k, run, level, diff;
966 if (get_bits_left(gb) < 16)
968 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
970 quantized_coeffs[0] = level;
972 for (i = 0; i < 7; ) {
973 if (get_bits_left(gb) < 16)
975 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
980 if (get_bits_left(gb) < 16)
982 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
984 for (k = 1; k <= run; k++)
985 quantized_coeffs[i + k] = (level + ((k * diff) / run));
995 * Related to synthesis filter, process data from packet 10
996 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
997 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1000 * @param gb bitreader context
1002 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
1004 int sb, j, k, n, ch;
1006 for (ch = 0; ch < q->nb_channels; ch++) {
1007 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
1009 if (get_bits_left(gb) < 16) {
1010 memset(q->quantized_coeffs[ch][0], 0, 8);
1015 n = q->sub_sampling + 1;
1017 for (sb = 0; sb < n; sb++)
1018 for (ch = 0; ch < q->nb_channels; ch++)
1019 for (j = 0; j < 8; j++) {
1020 if (get_bits_left(gb) < 1)
1022 if (get_bits1(gb)) {
1023 for (k=0; k < 8; k++) {
1024 if (get_bits_left(gb) < 16)
1026 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1029 for (k=0; k < 8; k++)
1030 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1034 n = QDM2_SB_USED(q->sub_sampling) - 4;
1036 for (sb = 0; sb < n; sb++)
1037 for (ch = 0; ch < q->nb_channels; ch++) {
1038 if (get_bits_left(gb) < 16)
1040 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1042 q->tone_level_idx_hi2[ch][sb] -= 16;
1044 for (j = 0; j < 8; j++)
1045 q->tone_level_idx_mid[ch][sb][j] = -16;
1048 n = QDM2_SB_USED(q->sub_sampling) - 5;
1050 for (sb = 0; sb < n; sb++)
1051 for (ch = 0; ch < q->nb_channels; ch++)
1052 for (j = 0; j < 8; j++) {
1053 if (get_bits_left(gb) < 16)
1055 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1060 * Process subpacket 9, init quantized_coeffs with data from it
1063 * @param node pointer to node with packet
1065 static int process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1068 int i, j, k, n, ch, run, level, diff;
1070 init_get_bits(&gb, node->packet->data, node->packet->size*8);
1072 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1074 for (i = 1; i < n; i++)
1075 for (ch=0; ch < q->nb_channels; ch++) {
1076 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1077 q->quantized_coeffs[ch][i][0] = level;
1079 for (j = 0; j < (8 - 1); ) {
1080 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1081 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1086 for (k = 1; k <= run; k++)
1087 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1094 for (ch = 0; ch < q->nb_channels; ch++)
1095 for (i = 0; i < 8; i++)
1096 q->quantized_coeffs[ch][0][i] = 0;
1103 * Process subpacket 10 if not null, else
1106 * @param node pointer to node with packet
1108 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node)
1113 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1114 init_tone_level_dequantization(q, &gb);
1115 fill_tone_level_array(q, 1);
1117 fill_tone_level_array(q, 0);
1123 * Process subpacket 11
1126 * @param node pointer to node with packet
1128 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node)
1134 length = node->packet->size * 8;
1135 init_get_bits(&gb, node->packet->data, length);
1139 int c = get_bits (&gb, 13);
1142 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1143 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1146 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1151 * Process subpacket 12
1154 * @param node pointer to node with packet
1156 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node)
1162 length = node->packet->size * 8;
1163 init_get_bits(&gb, node->packet->data, length);
1166 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1170 * Process new subpackets for synthesis filter
1173 * @param list list with synthesis filter packets (list D)
1175 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1177 QDM2SubPNode *nodes[4];
1179 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1180 if (nodes[0] != NULL)
1181 process_subpacket_9(q, nodes[0]);
1183 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1184 if (nodes[1] != NULL)
1185 process_subpacket_10(q, nodes[1]);
1187 process_subpacket_10(q, NULL);
1189 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1190 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1191 process_subpacket_11(q, nodes[2]);
1193 process_subpacket_11(q, NULL);
1195 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1196 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1197 process_subpacket_12(q, nodes[3]);
1199 process_subpacket_12(q, NULL);
1204 * Decode superblock, fill packet lists.
1208 static void qdm2_decode_super_block (QDM2Context *q)
1211 QDM2SubPacket header, *packet;
1212 int i, packet_bytes, sub_packet_size, sub_packets_D;
1213 unsigned int next_index = 0;
1215 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1216 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1217 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1219 q->sub_packets_B = 0;
1222 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1224 init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1225 qdm2_decode_sub_packet_header(&gb, &header);
1227 if (header.type < 2 || header.type >= 8) {
1229 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1233 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1234 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1236 init_get_bits(&gb, header.data, header.size*8);
1238 if (header.type == 2 || header.type == 4 || header.type == 5) {
1239 int csum = 257 * get_bits(&gb, 8);
1240 csum += 2 * get_bits(&gb, 8);
1242 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1246 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1251 q->sub_packet_list_B[0].packet = NULL;
1252 q->sub_packet_list_D[0].packet = NULL;
1254 for (i = 0; i < 6; i++)
1255 if (--q->fft_level_exp[i] < 0)
1256 q->fft_level_exp[i] = 0;
1258 for (i = 0; packet_bytes > 0; i++) {
1261 if (i>=FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1262 SAMPLES_NEEDED_2("too many packet bytes");
1266 q->sub_packet_list_A[i].next = NULL;
1269 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1271 /* seek to next block */
1272 init_get_bits(&gb, header.data, header.size*8);
1273 skip_bits(&gb, next_index*8);
1275 if (next_index >= header.size)
1279 /* decode subpacket */
1280 packet = &q->sub_packets[i];
1281 qdm2_decode_sub_packet_header(&gb, packet);
1282 next_index = packet->size + get_bits_count(&gb) / 8;
1283 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1285 if (packet->type == 0)
1288 if (sub_packet_size > packet_bytes) {
1289 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1291 packet->size += packet_bytes - sub_packet_size;
1294 packet_bytes -= sub_packet_size;
1296 /* add subpacket to 'all subpackets' list */
1297 q->sub_packet_list_A[i].packet = packet;
1299 /* add subpacket to related list */
1300 if (packet->type == 8) {
1301 SAMPLES_NEEDED_2("packet type 8");
1303 } else if (packet->type >= 9 && packet->type <= 12) {
1304 /* packets for MPEG Audio like Synthesis Filter */
1305 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1306 } else if (packet->type == 13) {
1307 for (j = 0; j < 6; j++)
1308 q->fft_level_exp[j] = get_bits(&gb, 6);
1309 } else if (packet->type == 14) {
1310 for (j = 0; j < 6; j++)
1311 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1312 } else if (packet->type == 15) {
1313 SAMPLES_NEEDED_2("packet type 15")
1315 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1316 /* packets for FFT */
1317 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1319 } // Packet bytes loop
1321 /* **************************************************************** */
1322 if (q->sub_packet_list_D[0].packet != NULL) {
1323 process_synthesis_subpackets(q, q->sub_packet_list_D);
1324 q->do_synth_filter = 1;
1325 } else if (q->do_synth_filter) {
1326 process_subpacket_10(q, NULL);
1327 process_subpacket_11(q, NULL);
1328 process_subpacket_12(q, NULL);
1330 /* **************************************************************** */
1334 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1335 int offset, int duration, int channel,
1338 if (q->fft_coefs_min_index[duration] < 0)
1339 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1341 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1342 q->fft_coefs[q->fft_coefs_index].channel = channel;
1343 q->fft_coefs[q->fft_coefs_index].offset = offset;
1344 q->fft_coefs[q->fft_coefs_index].exp = exp;
1345 q->fft_coefs[q->fft_coefs_index].phase = phase;
1346 q->fft_coefs_index++;
1350 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1352 int channel, stereo, phase, exp;
1353 int local_int_4, local_int_8, stereo_phase, local_int_10;
1354 int local_int_14, stereo_exp, local_int_20, local_int_28;
1360 local_int_8 = (4 - duration);
1361 local_int_10 = 1 << (q->group_order - duration - 1);
1364 while (get_bits_left(gb)>0) {
1365 if (q->superblocktype_2_3) {
1366 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1367 if (get_bits_left(gb)<0) {
1368 if(local_int_4 < q->group_size)
1369 av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1374 local_int_4 += local_int_10;
1375 local_int_28 += (1 << local_int_8);
1377 local_int_4 += 8*local_int_10;
1378 local_int_28 += (8 << local_int_8);
1383 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1384 while (offset >= (local_int_10 - 1)) {
1385 offset += (1 - (local_int_10 - 1));
1386 local_int_4 += local_int_10;
1387 local_int_28 += (1 << local_int_8);
1391 if (local_int_4 >= q->group_size)
1394 local_int_14 = (offset >> local_int_8);
1395 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1398 if (q->nb_channels > 1) {
1399 channel = get_bits1(gb);
1400 stereo = get_bits1(gb);
1406 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1407 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1408 exp = (exp < 0) ? 0 : exp;
1410 phase = get_bits(gb, 3);
1415 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1416 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1417 if (stereo_phase < 0)
1421 if (q->frequency_range > (local_int_14 + 1)) {
1422 int sub_packet = (local_int_20 + local_int_28);
1424 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1426 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1434 static void qdm2_decode_fft_packets (QDM2Context *q)
1436 int i, j, min, max, value, type, unknown_flag;
1439 if (q->sub_packet_list_B[0].packet == NULL)
1442 /* reset minimum indexes for FFT coefficients */
1443 q->fft_coefs_index = 0;
1444 for (i=0; i < 5; i++)
1445 q->fft_coefs_min_index[i] = -1;
1447 /* process subpackets ordered by type, largest type first */
1448 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1449 QDM2SubPacket *packet= NULL;
1451 /* find subpacket with largest type less than max */
1452 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1453 value = q->sub_packet_list_B[j].packet->type;
1454 if (value > min && value < max) {
1456 packet = q->sub_packet_list_B[j].packet;
1462 /* check for errors (?) */
1466 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1469 /* decode FFT tones */
1470 init_get_bits (&gb, packet->data, packet->size*8);
1472 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1477 type = packet->type;
1479 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1480 int duration = q->sub_sampling + 5 - (type & 15);
1482 if (duration >= 0 && duration < 4)
1483 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1484 } else if (type == 31) {
1485 for (j=0; j < 4; j++)
1486 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1487 } else if (type == 46) {
1488 for (j=0; j < 6; j++)
1489 q->fft_level_exp[j] = get_bits(&gb, 6);
1490 for (j=0; j < 4; j++)
1491 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1493 } // Loop on B packets
1495 /* calculate maximum indexes for FFT coefficients */
1496 for (i = 0, j = -1; i < 5; i++)
1497 if (q->fft_coefs_min_index[i] >= 0) {
1499 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1503 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1507 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1512 const double iscale = 2.0*M_PI / 512.0;
1514 tone->phase += tone->phase_shift;
1516 /* calculate current level (maximum amplitude) of tone */
1517 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1518 c.im = level * sin(tone->phase*iscale);
1519 c.re = level * cos(tone->phase*iscale);
1521 /* generate FFT coefficients for tone */
1522 if (tone->duration >= 3 || tone->cutoff >= 3) {
1523 tone->complex[0].im += c.im;
1524 tone->complex[0].re += c.re;
1525 tone->complex[1].im -= c.im;
1526 tone->complex[1].re -= c.re;
1528 f[1] = -tone->table[4];
1529 f[0] = tone->table[3] - tone->table[0];
1530 f[2] = 1.0 - tone->table[2] - tone->table[3];
1531 f[3] = tone->table[1] + tone->table[4] - 1.0;
1532 f[4] = tone->table[0] - tone->table[1];
1533 f[5] = tone->table[2];
1534 for (i = 0; i < 2; i++) {
1535 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1536 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1538 for (i = 0; i < 4; i++) {
1539 tone->complex[i].re += c.re * f[i+2];
1540 tone->complex[i].im += c.im * f[i+2];
1544 /* copy the tone if it has not yet died out */
1545 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1546 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1547 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1552 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1555 const double iscale = 0.25 * M_PI;
1557 for (ch = 0; ch < q->channels; ch++) {
1558 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1562 /* apply FFT tones with duration 4 (1 FFT period) */
1563 if (q->fft_coefs_min_index[4] >= 0)
1564 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1568 if (q->fft_coefs[i].sub_packet != sub_packet)
1571 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1572 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];
1574 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1575 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1576 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1577 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1578 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1579 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1582 /* generate existing FFT tones */
1583 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1584 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1585 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1588 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1589 for (i = 0; i < 4; i++)
1590 if (q->fft_coefs_min_index[i] >= 0) {
1591 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1595 if (q->fft_coefs[j].sub_packet != sub_packet)
1599 offset = q->fft_coefs[j].offset >> four_i;
1600 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1602 if (offset < q->frequency_range) {
1604 tone.cutoff = offset;
1606 tone.cutoff = (offset >= 60) ? 3 : 2;
1608 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];
1609 tone.complex = &q->fft.complex[ch][offset];
1610 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1611 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1612 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1614 tone.time_index = 0;
1616 qdm2_fft_generate_tone(q, &tone);
1619 q->fft_coefs_min_index[i] = j;
1624 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1626 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1627 float *out = q->output_buffer + channel;
1629 q->fft.complex[channel][0].re *= 2.0f;
1630 q->fft.complex[channel][0].im = 0.0f;
1631 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1632 /* add samples to output buffer */
1633 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1634 out[0] += q->fft.complex[channel][i].re * gain;
1635 out[q->channels] += q->fft.complex[channel][i].im * gain;
1636 out += 2 * q->channels;
1643 * @param index subpacket number
1645 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1647 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1649 /* copy sb_samples */
1650 sb_used = QDM2_SB_USED(q->sub_sampling);
1652 for (ch = 0; ch < q->channels; ch++)
1653 for (i = 0; i < 8; i++)
1654 for (k=sb_used; k < SBLIMIT; k++)
1655 q->sb_samples[ch][(8 * index) + i][k] = 0;
1657 for (ch = 0; ch < q->nb_channels; ch++) {
1658 float *samples_ptr = q->samples + ch;
1660 for (i = 0; i < 8; i++) {
1661 ff_mpa_synth_filter_float(&q->mpadsp,
1662 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1663 ff_mpa_synth_window_float, &dither_state,
1664 samples_ptr, q->nb_channels,
1665 q->sb_samples[ch][(8 * index) + i]);
1666 samples_ptr += 32 * q->nb_channels;
1670 /* add samples to output buffer */
1671 sub_sampling = (4 >> q->sub_sampling);
1673 for (ch = 0; ch < q->channels; ch++)
1674 for (i = 0; i < q->frame_size; i++)
1675 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1680 * Init static data (does not depend on specific file)
1684 static av_cold void qdm2_init(QDM2Context *q) {
1685 static int initialized = 0;
1687 if (initialized != 0)
1692 ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1693 softclip_table_init();
1695 init_noise_samples();
1697 av_log(NULL, AV_LOG_DEBUG, "init done\n");
1702 * Init parameters from codec extradata
1704 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1706 QDM2Context *s = avctx->priv_data;
1709 int tmp_val, tmp, size;
1711 /* extradata parsing
1720 32 size (including this field)
1722 32 type (=QDM2 or QDMC)
1724 32 size (including this field, in bytes)
1725 32 tag (=QDCA) // maybe mandatory parameters
1728 32 samplerate (=44100)
1730 32 block size (=4096)
1731 32 frame size (=256) (for one channel)
1732 32 packet size (=1300)
1734 32 size (including this field, in bytes)
1735 32 tag (=QDCP) // maybe some tuneable parameters
1745 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1746 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1750 extradata = avctx->extradata;
1751 extradata_size = avctx->extradata_size;
1753 while (extradata_size > 7) {
1754 if (!memcmp(extradata, "frmaQDM", 7))
1760 if (extradata_size < 12) {
1761 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1766 if (memcmp(extradata, "frmaQDM", 7)) {
1767 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1771 if (extradata[7] == 'C') {
1773 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1778 extradata_size -= 8;
1780 size = AV_RB32(extradata);
1782 if(size > extradata_size){
1783 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1784 extradata_size, size);
1789 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1790 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1791 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1797 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1799 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1800 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1801 return AVERROR_INVALIDDATA;
1803 avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1806 avctx->sample_rate = AV_RB32(extradata);
1809 avctx->bit_rate = AV_RB32(extradata);
1812 s->group_size = AV_RB32(extradata);
1815 s->fft_size = AV_RB32(extradata);
1818 s->checksum_size = AV_RB32(extradata);
1819 if (s->checksum_size >= 1U << 28) {
1820 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1821 return AVERROR_INVALIDDATA;
1824 s->fft_order = av_log2(s->fft_size) + 1;
1826 // something like max decodable tones
1827 s->group_order = av_log2(s->group_size) + 1;
1828 s->frame_size = s->group_size / 16; // 16 iterations per super block
1830 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1831 return AVERROR_INVALIDDATA;
1833 s->sub_sampling = s->fft_order - 7;
1834 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1836 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1837 case 0: tmp = 40; break;
1838 case 1: tmp = 48; break;
1839 case 2: tmp = 56; break;
1840 case 3: tmp = 72; break;
1841 case 4: tmp = 80; break;
1842 case 5: tmp = 100;break;
1843 default: tmp=s->sub_sampling; break;
1846 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1847 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1848 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1849 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1850 s->cm_table_select = tmp_val;
1852 if (s->sub_sampling == 0)
1855 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1862 s->coeff_per_sb_select = 0;
1863 else if (tmp <= 16000)
1864 s->coeff_per_sb_select = 1;
1866 s->coeff_per_sb_select = 2;
1868 // Fail on unknown fft order
1869 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1870 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1874 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1875 ff_mpadsp_init(&s->mpadsp);
1879 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1881 avcodec_get_frame_defaults(&s->frame);
1882 avctx->coded_frame = &s->frame;
1888 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1890 QDM2Context *s = avctx->priv_data;
1892 ff_rdft_end(&s->rdft_ctx);
1898 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1901 const int frame_size = (q->frame_size * q->channels);
1903 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1906 /* select input buffer */
1907 q->compressed_data = in;
1908 q->compressed_size = q->checksum_size;
1910 /* copy old block, clear new block of output samples */
1911 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1912 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1914 /* decode block of QDM2 compressed data */
1915 if (q->sub_packet == 0) {
1916 q->has_errors = 0; // zero it for a new super block
1917 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1918 qdm2_decode_super_block(q);
1921 /* parse subpackets */
1922 if (!q->has_errors) {
1923 if (q->sub_packet == 2)
1924 qdm2_decode_fft_packets(q);
1926 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1929 /* sound synthesis stage 1 (FFT) */
1930 for (ch = 0; ch < q->channels; ch++) {
1931 qdm2_calculate_fft(q, ch, q->sub_packet);
1933 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1934 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1939 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1940 if (!q->has_errors && q->do_synth_filter)
1941 qdm2_synthesis_filter(q, q->sub_packet);
1943 q->sub_packet = (q->sub_packet + 1) % 16;
1945 /* clip and convert output float[] to 16bit signed samples */
1946 for (i = 0; i < frame_size; i++) {
1947 int value = (int)q->output_buffer[i];
1949 if (value > SOFTCLIP_THRESHOLD)
1950 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1951 else if (value < -SOFTCLIP_THRESHOLD)
1952 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1961 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1962 int *got_frame_ptr, AVPacket *avpkt)
1964 const uint8_t *buf = avpkt->data;
1965 int buf_size = avpkt->size;
1966 QDM2Context *s = avctx->priv_data;
1972 if(buf_size < s->checksum_size)
1975 /* get output buffer */
1976 s->frame.nb_samples = 16 * s->frame_size;
1977 if ((ret = ff_get_buffer(avctx, &s->frame)) < 0) {
1978 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1981 out = (int16_t *)s->frame.data[0];
1983 for (i = 0; i < 16; i++) {
1984 if (qdm2_decode(s, buf, out) < 0)
1986 out += s->channels * s->frame_size;
1990 *(AVFrame *)data = s->frame;
1992 return s->checksum_size;
1995 AVCodec ff_qdm2_decoder =
1998 .type = AVMEDIA_TYPE_AUDIO,
1999 .id = AV_CODEC_ID_QDM2,
2000 .priv_data_size = sizeof(QDM2Context),
2001 .init = qdm2_decode_init,
2002 .close = qdm2_decode_close,
2003 .decode = qdm2_decode_frame,
2004 .capabilities = CODEC_CAP_DR1,
2005 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),