2 * QDM2 compatible decoder
3 * Copyright (c) 2003 Ewald Snel
4 * Copyright (c) 2005 Benjamin Larsson
5 * Copyright (c) 2005 Alex Beregszaszi
6 * Copyright (c) 2005 Roberto Togni
8 * This file is part of FFmpeg.
10 * FFmpeg is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Lesser General Public
12 * License as published by the Free Software Foundation; either
13 * version 2.1 of the License, or (at your option) any later version.
15 * FFmpeg is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Lesser General Public License for more details.
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with FFmpeg; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
30 * The decoder is not perfect yet, there are still some distortions
31 * especially on files encoded with 16 or 8 subbands.
38 #define BITSTREAM_READER_LE
39 #include "libavutil/channel_layout.h"
44 #include "mpegaudiodsp.h"
45 #include "mpegaudio.h"
48 #include "qdm2_tablegen.h"
54 #define QDM2_LIST_ADD(list, size, packet) \
57 list[size - 1].next = &list[size]; \
59 list[size].packet = packet; \
60 list[size].next = NULL; \
64 // Result is 8, 16 or 30
65 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
67 #define FIX_NOISE_IDX(noise_idx) \
68 if ((noise_idx) >= 3840) \
69 (noise_idx) -= 3840; \
71 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
73 #define SAMPLES_NEEDED \
74 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
76 #define SAMPLES_NEEDED_2(why) \
77 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
79 #define QDM2_MAX_FRAME_SIZE 512
81 typedef int8_t sb_int8_array[2][30][64];
87 int type; ///< subpacket type
88 unsigned int size; ///< subpacket size
89 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
93 * A node in the subpacket list
95 typedef struct QDM2SubPNode {
96 QDM2SubPacket *packet; ///< packet
97 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
107 QDM2Complex *complex;
125 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
129 * QDM2 decoder context
132 /// Parameters from codec header, do not change during playback
133 int nb_channels; ///< number of channels
134 int channels; ///< number of channels
135 int group_size; ///< size of frame group (16 frames per group)
136 int fft_size; ///< size of FFT, in complex numbers
137 int checksum_size; ///< size of data block, used also for checksum
139 /// Parameters built from header parameters, do not change during playback
140 int group_order; ///< order of frame group
141 int fft_order; ///< order of FFT (actually fftorder+1)
142 int frame_size; ///< size of data frame
144 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
145 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
146 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
148 /// Packets and packet lists
149 QDM2SubPacket sub_packets[16]; ///< the packets themselves
150 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
151 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
152 int sub_packets_B; ///< number of packets on 'B' list
153 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
154 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
157 FFTTone fft_tones[1000];
160 FFTCoefficient fft_coefs[1000];
162 int fft_coefs_min_index[5];
163 int fft_coefs_max_index[5];
164 int fft_level_exp[6];
165 RDFTContext rdft_ctx;
169 const uint8_t *compressed_data;
171 float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
174 MPADSPContext mpadsp;
175 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
176 int synth_buf_offset[MPA_MAX_CHANNELS];
177 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
178 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
180 /// Mixed temporary data used in decoding
181 float tone_level[MPA_MAX_CHANNELS][30][64];
182 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
183 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
184 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
185 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
186 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
187 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
188 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
189 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
192 int has_errors; ///< packet has errors
193 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
194 int do_synth_filter; ///< used to perform or skip synthesis filter
197 int noise_idx; ///< index for dithering noise table
201 static VLC vlc_tab_level;
202 static VLC vlc_tab_diff;
203 static VLC vlc_tab_run;
204 static VLC fft_level_exp_alt_vlc;
205 static VLC fft_level_exp_vlc;
206 static VLC fft_stereo_exp_vlc;
207 static VLC fft_stereo_phase_vlc;
208 static VLC vlc_tab_tone_level_idx_hi1;
209 static VLC vlc_tab_tone_level_idx_mid;
210 static VLC vlc_tab_tone_level_idx_hi2;
211 static VLC vlc_tab_type30;
212 static VLC vlc_tab_type34;
213 static VLC vlc_tab_fft_tone_offset[5];
215 static const uint16_t qdm2_vlc_offs[] = {
216 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
219 static av_cold void qdm2_init_vlc(void)
221 static int vlcs_initialized = 0;
222 static VLC_TYPE qdm2_table[3838][2];
224 if (!vlcs_initialized) {
226 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
227 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
228 init_vlc (&vlc_tab_level, 8, 24,
229 vlc_tab_level_huffbits, 1, 1,
230 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
232 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
233 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
234 init_vlc (&vlc_tab_diff, 8, 37,
235 vlc_tab_diff_huffbits, 1, 1,
236 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
238 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
239 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
240 init_vlc (&vlc_tab_run, 5, 6,
241 vlc_tab_run_huffbits, 1, 1,
242 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
244 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
245 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
246 init_vlc (&fft_level_exp_alt_vlc, 8, 28,
247 fft_level_exp_alt_huffbits, 1, 1,
248 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
251 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
252 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
253 init_vlc (&fft_level_exp_vlc, 8, 20,
254 fft_level_exp_huffbits, 1, 1,
255 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
257 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
258 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
259 init_vlc (&fft_stereo_exp_vlc, 6, 7,
260 fft_stereo_exp_huffbits, 1, 1,
261 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
263 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
264 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
265 init_vlc (&fft_stereo_phase_vlc, 6, 9,
266 fft_stereo_phase_huffbits, 1, 1,
267 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
269 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
270 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
271 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
272 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
273 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
275 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
276 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
277 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
278 vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
279 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
281 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
282 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
283 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
284 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
285 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
287 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
288 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
289 init_vlc (&vlc_tab_type30, 6, 9,
290 vlc_tab_type30_huffbits, 1, 1,
291 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
293 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
294 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
295 init_vlc (&vlc_tab_type34, 5, 10,
296 vlc_tab_type34_huffbits, 1, 1,
297 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
299 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
300 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
301 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
302 vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
303 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
305 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
306 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
307 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
308 vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
309 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
311 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
312 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
313 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
314 vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
315 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
317 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
318 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
319 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
320 vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
321 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
323 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
324 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
325 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
326 vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
327 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
333 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
337 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
339 /* stage-2, 3 bits exponent escape sequence */
341 value = get_bits (gb, get_bits (gb, 3) + 1);
343 /* stage-3, optional */
348 av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
352 tmp= vlc_stage3_values[value];
354 if ((value & ~3) > 0)
355 tmp += get_bits (gb, (value >> 2));
363 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
365 int value = qdm2_get_vlc (gb, vlc, 0, depth);
367 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
374 * @param data pointer to data to be checksum'ed
375 * @param length data length
376 * @param value checksum value
378 * @return 0 if checksum is OK
380 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
383 for (i=0; i < length; i++)
386 return (uint16_t)(value & 0xffff);
391 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
393 * @param gb bitreader context
394 * @param sub_packet packet under analysis
396 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
398 sub_packet->type = get_bits (gb, 8);
400 if (sub_packet->type == 0) {
401 sub_packet->size = 0;
402 sub_packet->data = NULL;
404 sub_packet->size = get_bits (gb, 8);
406 if (sub_packet->type & 0x80) {
407 sub_packet->size <<= 8;
408 sub_packet->size |= get_bits (gb, 8);
409 sub_packet->type &= 0x7f;
412 if (sub_packet->type == 0x7f)
413 sub_packet->type |= (get_bits (gb, 8) << 8);
415 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
418 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
419 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
424 * Return node pointer to first packet of requested type in list.
426 * @param list list of subpackets to be scanned
427 * @param type type of searched subpacket
428 * @return node pointer for subpacket if found, else NULL
430 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
432 while (list != NULL && list->packet != NULL) {
433 if (list->packet->type == type)
442 * Replace 8 elements with their average value.
443 * Called by qdm2_decode_superblock before starting subblock decoding.
447 static void average_quantized_coeffs (QDM2Context *q)
449 int i, j, n, ch, sum;
451 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
453 for (ch = 0; ch < q->nb_channels; ch++)
454 for (i = 0; i < n; i++) {
457 for (j = 0; j < 8; j++)
458 sum += q->quantized_coeffs[ch][i][j];
464 for (j=0; j < 8; j++)
465 q->quantized_coeffs[ch][i][j] = sum;
471 * Build subband samples with noise weighted by q->tone_level.
472 * Called by synthfilt_build_sb_samples.
475 * @param sb subband index
477 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
481 FIX_NOISE_IDX(q->noise_idx);
486 for (ch = 0; ch < q->nb_channels; ch++)
487 for (j = 0; j < 64; j++) {
488 q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
489 q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
495 * Called while processing data from subpackets 11 and 12.
496 * Used after making changes to coding_method array.
498 * @param sb subband index
499 * @param channels number of channels
500 * @param coding_method q->coding_method[0][0][0]
502 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
507 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};
509 for (ch = 0; ch < channels; ch++) {
510 for (j = 0; j < 64; ) {
511 if((coding_method[ch][sb][j] - 8) > 22) {
515 switch (switchtable[coding_method[ch][sb][j]-8]) {
516 case 0: run = 10; case_val = 10; break;
517 case 1: run = 1; case_val = 16; break;
518 case 2: run = 5; case_val = 24; break;
519 case 3: run = 3; case_val = 30; break;
520 case 4: run = 1; case_val = 30; break;
521 case 5: run = 1; case_val = 8; break;
522 default: run = 1; case_val = 8; break;
525 for (k = 0; k < run; k++)
527 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
530 //not debugged, almost never used
531 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
532 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
541 * Related to synthesis filter
542 * Called by process_subpacket_10
545 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
547 static void fill_tone_level_array (QDM2Context *q, int flag)
549 int i, sb, ch, sb_used;
552 for (ch = 0; ch < q->nb_channels; ch++)
553 for (sb = 0; sb < 30; sb++)
554 for (i = 0; i < 8; i++) {
555 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
556 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
557 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
559 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
562 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
565 sb_used = QDM2_SB_USED(q->sub_sampling);
567 if ((q->superblocktype_2_3 != 0) && !flag) {
568 for (sb = 0; sb < sb_used; sb++)
569 for (ch = 0; ch < q->nb_channels; ch++)
570 for (i = 0; i < 64; i++) {
571 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
572 if (q->tone_level_idx[ch][sb][i] < 0)
573 q->tone_level[ch][sb][i] = 0;
575 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
578 tab = q->superblocktype_2_3 ? 0 : 1;
579 for (sb = 0; sb < sb_used; sb++) {
580 if ((sb >= 4) && (sb <= 23)) {
581 for (ch = 0; ch < q->nb_channels; ch++)
582 for (i = 0; i < 64; i++) {
583 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
584 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
585 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
586 q->tone_level_idx_hi2[ch][sb - 4];
587 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
588 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
589 q->tone_level[ch][sb][i] = 0;
591 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
595 for (ch = 0; ch < q->nb_channels; ch++)
596 for (i = 0; i < 64; i++) {
597 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
598 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
599 q->tone_level_idx_hi2[ch][sb - 4];
600 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
601 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
602 q->tone_level[ch][sb][i] = 0;
604 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
607 for (ch = 0; ch < q->nb_channels; ch++)
608 for (i = 0; i < 64; i++) {
609 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
610 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
611 q->tone_level[ch][sb][i] = 0;
613 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
625 * Related to synthesis filter
626 * Called by process_subpacket_11
627 * c is built with data from subpacket 11
628 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
630 * @param tone_level_idx
631 * @param tone_level_idx_temp
632 * @param coding_method q->coding_method[0][0][0]
633 * @param nb_channels number of channels
634 * @param c coming from subpacket 11, passed as 8*c
635 * @param superblocktype_2_3 flag based on superblock packet type
636 * @param cm_table_select q->cm_table_select
638 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
639 sb_int8_array coding_method, int nb_channels,
640 int c, int superblocktype_2_3, int cm_table_select)
643 int tmp, acc, esp_40, comp;
644 int add1, add2, add3, add4;
647 if (!superblocktype_2_3) {
648 /* This case is untested, no samples available */
649 avpriv_request_sample(NULL, "!superblocktype_2_3");
651 for (ch = 0; ch < nb_channels; ch++)
652 for (sb = 0; sb < 30; sb++) {
653 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
654 add1 = tone_level_idx[ch][sb][j] - 10;
657 add2 = add3 = add4 = 0;
659 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
664 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
669 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
673 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
676 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
678 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
681 for (ch = 0; ch < nb_channels; ch++)
682 for (sb = 0; sb < 30; sb++)
683 for (j = 0; j < 64; j++)
684 acc += tone_level_idx_temp[ch][sb][j];
686 multres = 0x66666667LL * (acc * 10);
687 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
688 for (ch = 0; ch < nb_channels; ch++)
689 for (sb = 0; sb < 30; sb++)
690 for (j = 0; j < 64; j++) {
691 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
694 comp /= 256; // signed shift
722 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
724 for (sb = 0; sb < 30; sb++)
725 fix_coding_method_array(sb, nb_channels, coding_method);
726 for (ch = 0; ch < nb_channels; ch++)
727 for (sb = 0; sb < 30; sb++)
728 for (j = 0; j < 64; j++)
730 if (coding_method[ch][sb][j] < 10)
731 coding_method[ch][sb][j] = 10;
734 if (coding_method[ch][sb][j] < 16)
735 coding_method[ch][sb][j] = 16;
737 if (coding_method[ch][sb][j] < 30)
738 coding_method[ch][sb][j] = 30;
741 } else { // superblocktype_2_3 != 0
742 for (ch = 0; ch < nb_channels; ch++)
743 for (sb = 0; sb < 30; sb++)
744 for (j = 0; j < 64; j++)
745 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
754 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
755 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
758 * @param gb bitreader context
759 * @param length packet length in bits
760 * @param sb_min lower subband processed (sb_min included)
761 * @param sb_max higher subband processed (sb_max excluded)
763 static int synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
765 int sb, j, k, n, ch, run, channels;
766 int joined_stereo, zero_encoding, chs;
768 float type34_div = 0;
769 float type34_predictor;
770 float samples[10], sign_bits[16];
773 // If no data use noise
774 for (sb=sb_min; sb < sb_max; sb++)
775 build_sb_samples_from_noise (q, sb);
780 for (sb = sb_min; sb < sb_max; sb++) {
781 FIX_NOISE_IDX(q->noise_idx);
783 channels = q->nb_channels;
785 if (q->nb_channels <= 1 || sb < 12)
790 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0;
793 if (get_bits_left(gb) >= 16)
794 for (j = 0; j < 16; j++)
795 sign_bits[j] = get_bits1 (gb);
797 if (q->coding_method[0][sb][0] <= 0) {
798 av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
799 return AVERROR_INVALIDDATA;
802 for (j = 0; j < 64; j++)
803 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
804 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
806 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
810 for (ch = 0; ch < channels; ch++) {
811 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
812 type34_predictor = 0.0;
815 for (j = 0; j < 128; ) {
816 switch (q->coding_method[ch][sb][j / 2]) {
818 if (get_bits_left(gb) >= 10) {
820 for (k = 0; k < 5; k++) {
821 if ((j + 2 * k) >= 128)
823 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
828 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
829 return AVERROR_INVALIDDATA;
832 for (k = 0; k < 5; k++)
833 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
835 for (k = 0; k < 5; k++)
836 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
838 for (k = 0; k < 10; k++)
839 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
845 if (get_bits_left(gb) >= 1) {
850 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
853 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
859 if (get_bits_left(gb) >= 10) {
861 for (k = 0; k < 5; k++) {
864 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
867 n = get_bits (gb, 8);
869 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
870 return AVERROR_INVALIDDATA;
873 for (k = 0; k < 5; k++)
874 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
877 for (k = 0; k < 5; k++)
878 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
884 if (get_bits_left(gb) >= 7) {
887 av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
888 return AVERROR_INVALIDDATA;
891 for (k = 0; k < 3; k++)
892 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
894 for (k = 0; k < 3; k++)
895 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
901 if (get_bits_left(gb) >= 4) {
902 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
903 if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
904 av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
905 return AVERROR_INVALIDDATA;
907 samples[0] = type30_dequant[index];
909 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
915 if (get_bits_left(gb) >= 7) {
917 type34_div = (float)(1 << get_bits(gb, 2));
918 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
919 type34_predictor = samples[0];
922 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
923 if (index >= FF_ARRAY_ELEMS(type34_delta)) {
924 av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
925 return AVERROR_INVALIDDATA;
927 samples[0] = type34_delta[index] / type34_div + type34_predictor;
928 type34_predictor = samples[0];
931 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
937 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
943 float tmp[10][MPA_MAX_CHANNELS];
944 for (k = 0; k < run; k++) {
945 tmp[k][0] = samples[k];
947 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
949 for (chs = 0; chs < q->nb_channels; chs++)
950 for (k = 0; k < run; k++)
952 q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
954 for (k = 0; k < run; k++)
956 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
968 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
969 * This is similar to process_subpacket_9, but for a single channel and for element [0]
970 * same VLC tables as process_subpacket_9 are used.
972 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
973 * @param gb bitreader context
975 static int init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb)
977 int i, k, run, level, diff;
979 if (get_bits_left(gb) < 16)
981 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
983 quantized_coeffs[0] = level;
985 for (i = 0; i < 7; ) {
986 if (get_bits_left(gb) < 16)
988 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
993 if (get_bits_left(gb) < 16)
995 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
997 for (k = 1; k <= run; k++)
998 quantized_coeffs[i + k] = (level + ((k * diff) / run));
1008 * Related to synthesis filter, process data from packet 10
1009 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1010 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1013 * @param gb bitreader context
1015 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
1017 int sb, j, k, n, ch;
1019 for (ch = 0; ch < q->nb_channels; ch++) {
1020 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
1022 if (get_bits_left(gb) < 16) {
1023 memset(q->quantized_coeffs[ch][0], 0, 8);
1028 n = q->sub_sampling + 1;
1030 for (sb = 0; sb < n; sb++)
1031 for (ch = 0; ch < q->nb_channels; ch++)
1032 for (j = 0; j < 8; j++) {
1033 if (get_bits_left(gb) < 1)
1035 if (get_bits1(gb)) {
1036 for (k=0; k < 8; k++) {
1037 if (get_bits_left(gb) < 16)
1039 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1042 for (k=0; k < 8; k++)
1043 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1047 n = QDM2_SB_USED(q->sub_sampling) - 4;
1049 for (sb = 0; sb < n; sb++)
1050 for (ch = 0; ch < q->nb_channels; ch++) {
1051 if (get_bits_left(gb) < 16)
1053 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1055 q->tone_level_idx_hi2[ch][sb] -= 16;
1057 for (j = 0; j < 8; j++)
1058 q->tone_level_idx_mid[ch][sb][j] = -16;
1061 n = QDM2_SB_USED(q->sub_sampling) - 5;
1063 for (sb = 0; sb < n; sb++)
1064 for (ch = 0; ch < q->nb_channels; ch++)
1065 for (j = 0; j < 8; j++) {
1066 if (get_bits_left(gb) < 16)
1068 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1073 * Process subpacket 9, init quantized_coeffs with data from it
1076 * @param node pointer to node with packet
1078 static int process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1081 int i, j, k, n, ch, run, level, diff;
1083 init_get_bits(&gb, node->packet->data, node->packet->size*8);
1085 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1087 for (i = 1; i < n; i++)
1088 for (ch=0; ch < q->nb_channels; ch++) {
1089 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1090 q->quantized_coeffs[ch][i][0] = level;
1092 for (j = 0; j < (8 - 1); ) {
1093 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1094 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1099 for (k = 1; k <= run; k++)
1100 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1107 for (ch = 0; ch < q->nb_channels; ch++)
1108 for (i = 0; i < 8; i++)
1109 q->quantized_coeffs[ch][0][i] = 0;
1116 * Process subpacket 10 if not null, else
1119 * @param node pointer to node with packet
1121 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node)
1126 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1127 init_tone_level_dequantization(q, &gb);
1128 fill_tone_level_array(q, 1);
1130 fill_tone_level_array(q, 0);
1136 * Process subpacket 11
1139 * @param node pointer to node with packet
1141 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node)
1147 length = node->packet->size * 8;
1148 init_get_bits(&gb, node->packet->data, length);
1152 int c = get_bits (&gb, 13);
1155 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1156 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1159 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1164 * Process subpacket 12
1167 * @param node pointer to node with packet
1169 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node)
1175 length = node->packet->size * 8;
1176 init_get_bits(&gb, node->packet->data, length);
1179 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1183 * Process new subpackets for synthesis filter
1186 * @param list list with synthesis filter packets (list D)
1188 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1190 QDM2SubPNode *nodes[4];
1192 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1193 if (nodes[0] != NULL)
1194 process_subpacket_9(q, nodes[0]);
1196 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1197 if (nodes[1] != NULL)
1198 process_subpacket_10(q, nodes[1]);
1200 process_subpacket_10(q, NULL);
1202 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1203 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1204 process_subpacket_11(q, nodes[2]);
1206 process_subpacket_11(q, NULL);
1208 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1209 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1210 process_subpacket_12(q, nodes[3]);
1212 process_subpacket_12(q, NULL);
1217 * Decode superblock, fill packet lists.
1221 static void qdm2_decode_super_block (QDM2Context *q)
1224 QDM2SubPacket header, *packet;
1225 int i, packet_bytes, sub_packet_size, sub_packets_D;
1226 unsigned int next_index = 0;
1228 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1229 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1230 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1232 q->sub_packets_B = 0;
1235 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1237 init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1238 qdm2_decode_sub_packet_header(&gb, &header);
1240 if (header.type < 2 || header.type >= 8) {
1242 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1246 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1247 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1249 init_get_bits(&gb, header.data, header.size*8);
1251 if (header.type == 2 || header.type == 4 || header.type == 5) {
1252 int csum = 257 * get_bits(&gb, 8);
1253 csum += 2 * get_bits(&gb, 8);
1255 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1259 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1264 q->sub_packet_list_B[0].packet = NULL;
1265 q->sub_packet_list_D[0].packet = NULL;
1267 for (i = 0; i < 6; i++)
1268 if (--q->fft_level_exp[i] < 0)
1269 q->fft_level_exp[i] = 0;
1271 for (i = 0; packet_bytes > 0; i++) {
1274 if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1275 SAMPLES_NEEDED_2("too many packet bytes");
1279 q->sub_packet_list_A[i].next = NULL;
1282 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1284 /* seek to next block */
1285 init_get_bits(&gb, header.data, header.size*8);
1286 skip_bits(&gb, next_index*8);
1288 if (next_index >= header.size)
1292 /* decode subpacket */
1293 packet = &q->sub_packets[i];
1294 qdm2_decode_sub_packet_header(&gb, packet);
1295 next_index = packet->size + get_bits_count(&gb) / 8;
1296 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1298 if (packet->type == 0)
1301 if (sub_packet_size > packet_bytes) {
1302 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1304 packet->size += packet_bytes - sub_packet_size;
1307 packet_bytes -= sub_packet_size;
1309 /* add subpacket to 'all subpackets' list */
1310 q->sub_packet_list_A[i].packet = packet;
1312 /* add subpacket to related list */
1313 if (packet->type == 8) {
1314 SAMPLES_NEEDED_2("packet type 8");
1316 } else if (packet->type >= 9 && packet->type <= 12) {
1317 /* packets for MPEG Audio like Synthesis Filter */
1318 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1319 } else if (packet->type == 13) {
1320 for (j = 0; j < 6; j++)
1321 q->fft_level_exp[j] = get_bits(&gb, 6);
1322 } else if (packet->type == 14) {
1323 for (j = 0; j < 6; j++)
1324 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1325 } else if (packet->type == 15) {
1326 SAMPLES_NEEDED_2("packet type 15")
1328 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1329 /* packets for FFT */
1330 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1332 } // Packet bytes loop
1334 /* **************************************************************** */
1335 if (q->sub_packet_list_D[0].packet != NULL) {
1336 process_synthesis_subpackets(q, q->sub_packet_list_D);
1337 q->do_synth_filter = 1;
1338 } else if (q->do_synth_filter) {
1339 process_subpacket_10(q, NULL);
1340 process_subpacket_11(q, NULL);
1341 process_subpacket_12(q, NULL);
1343 /* **************************************************************** */
1347 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1348 int offset, int duration, int channel,
1351 if (q->fft_coefs_min_index[duration] < 0)
1352 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1354 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1355 q->fft_coefs[q->fft_coefs_index].channel = channel;
1356 q->fft_coefs[q->fft_coefs_index].offset = offset;
1357 q->fft_coefs[q->fft_coefs_index].exp = exp;
1358 q->fft_coefs[q->fft_coefs_index].phase = phase;
1359 q->fft_coefs_index++;
1363 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1365 int channel, stereo, phase, exp;
1366 int local_int_4, local_int_8, stereo_phase, local_int_10;
1367 int local_int_14, stereo_exp, local_int_20, local_int_28;
1373 local_int_8 = (4 - duration);
1374 local_int_10 = 1 << (q->group_order - duration - 1);
1377 while (get_bits_left(gb)>0) {
1378 if (q->superblocktype_2_3) {
1379 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1380 if (get_bits_left(gb)<0) {
1381 if(local_int_4 < q->group_size)
1382 av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1387 local_int_4 += local_int_10;
1388 local_int_28 += (1 << local_int_8);
1390 local_int_4 += 8*local_int_10;
1391 local_int_28 += (8 << local_int_8);
1396 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1397 while (offset >= (local_int_10 - 1)) {
1398 offset += (1 - (local_int_10 - 1));
1399 local_int_4 += local_int_10;
1400 local_int_28 += (1 << local_int_8);
1404 if (local_int_4 >= q->group_size)
1407 local_int_14 = (offset >> local_int_8);
1408 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1411 if (q->nb_channels > 1) {
1412 channel = get_bits1(gb);
1413 stereo = get_bits1(gb);
1419 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1420 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1421 exp = (exp < 0) ? 0 : exp;
1423 phase = get_bits(gb, 3);
1428 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1429 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1430 if (stereo_phase < 0)
1434 if (q->frequency_range > (local_int_14 + 1)) {
1435 int sub_packet = (local_int_20 + local_int_28);
1437 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1439 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1447 static void qdm2_decode_fft_packets (QDM2Context *q)
1449 int i, j, min, max, value, type, unknown_flag;
1452 if (q->sub_packet_list_B[0].packet == NULL)
1455 /* reset minimum indexes for FFT coefficients */
1456 q->fft_coefs_index = 0;
1457 for (i=0; i < 5; i++)
1458 q->fft_coefs_min_index[i] = -1;
1460 /* process subpackets ordered by type, largest type first */
1461 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1462 QDM2SubPacket *packet= NULL;
1464 /* find subpacket with largest type less than max */
1465 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1466 value = q->sub_packet_list_B[j].packet->type;
1467 if (value > min && value < max) {
1469 packet = q->sub_packet_list_B[j].packet;
1475 /* check for errors (?) */
1479 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1482 /* decode FFT tones */
1483 init_get_bits (&gb, packet->data, packet->size*8);
1485 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1490 type = packet->type;
1492 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1493 int duration = q->sub_sampling + 5 - (type & 15);
1495 if (duration >= 0 && duration < 4)
1496 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1497 } else if (type == 31) {
1498 for (j=0; j < 4; j++)
1499 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1500 } else if (type == 46) {
1501 for (j=0; j < 6; j++)
1502 q->fft_level_exp[j] = get_bits(&gb, 6);
1503 for (j=0; j < 4; j++)
1504 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1506 } // Loop on B packets
1508 /* calculate maximum indexes for FFT coefficients */
1509 for (i = 0, j = -1; i < 5; i++)
1510 if (q->fft_coefs_min_index[i] >= 0) {
1512 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1516 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1520 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1525 const double iscale = 2.0*M_PI / 512.0;
1527 tone->phase += tone->phase_shift;
1529 /* calculate current level (maximum amplitude) of tone */
1530 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1531 c.im = level * sin(tone->phase*iscale);
1532 c.re = level * cos(tone->phase*iscale);
1534 /* generate FFT coefficients for tone */
1535 if (tone->duration >= 3 || tone->cutoff >= 3) {
1536 tone->complex[0].im += c.im;
1537 tone->complex[0].re += c.re;
1538 tone->complex[1].im -= c.im;
1539 tone->complex[1].re -= c.re;
1541 f[1] = -tone->table[4];
1542 f[0] = tone->table[3] - tone->table[0];
1543 f[2] = 1.0 - tone->table[2] - tone->table[3];
1544 f[3] = tone->table[1] + tone->table[4] - 1.0;
1545 f[4] = tone->table[0] - tone->table[1];
1546 f[5] = tone->table[2];
1547 for (i = 0; i < 2; i++) {
1548 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1549 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1551 for (i = 0; i < 4; i++) {
1552 tone->complex[i].re += c.re * f[i+2];
1553 tone->complex[i].im += c.im * f[i+2];
1557 /* copy the tone if it has not yet died out */
1558 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1559 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1560 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1565 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1568 const double iscale = 0.25 * M_PI;
1570 for (ch = 0; ch < q->channels; ch++) {
1571 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1575 /* apply FFT tones with duration 4 (1 FFT period) */
1576 if (q->fft_coefs_min_index[4] >= 0)
1577 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1581 if (q->fft_coefs[i].sub_packet != sub_packet)
1584 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1585 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];
1587 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1588 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1589 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1590 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1591 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1592 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1595 /* generate existing FFT tones */
1596 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1597 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1598 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1601 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1602 for (i = 0; i < 4; i++)
1603 if (q->fft_coefs_min_index[i] >= 0) {
1604 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1608 if (q->fft_coefs[j].sub_packet != sub_packet)
1612 offset = q->fft_coefs[j].offset >> four_i;
1613 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1615 if (offset < q->frequency_range) {
1617 tone.cutoff = offset;
1619 tone.cutoff = (offset >= 60) ? 3 : 2;
1621 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];
1622 tone.complex = &q->fft.complex[ch][offset];
1623 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1624 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1625 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1627 tone.time_index = 0;
1629 qdm2_fft_generate_tone(q, &tone);
1632 q->fft_coefs_min_index[i] = j;
1637 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1639 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1640 float *out = q->output_buffer + channel;
1642 q->fft.complex[channel][0].re *= 2.0f;
1643 q->fft.complex[channel][0].im = 0.0f;
1644 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1645 /* add samples to output buffer */
1646 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1647 out[0] += q->fft.complex[channel][i].re * gain;
1648 out[q->channels] += q->fft.complex[channel][i].im * gain;
1649 out += 2 * q->channels;
1656 * @param index subpacket number
1658 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1660 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1662 /* copy sb_samples */
1663 sb_used = QDM2_SB_USED(q->sub_sampling);
1665 for (ch = 0; ch < q->channels; ch++)
1666 for (i = 0; i < 8; i++)
1667 for (k=sb_used; k < SBLIMIT; k++)
1668 q->sb_samples[ch][(8 * index) + i][k] = 0;
1670 for (ch = 0; ch < q->nb_channels; ch++) {
1671 float *samples_ptr = q->samples + ch;
1673 for (i = 0; i < 8; i++) {
1674 ff_mpa_synth_filter_float(&q->mpadsp,
1675 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1676 ff_mpa_synth_window_float, &dither_state,
1677 samples_ptr, q->nb_channels,
1678 q->sb_samples[ch][(8 * index) + i]);
1679 samples_ptr += 32 * q->nb_channels;
1683 /* add samples to output buffer */
1684 sub_sampling = (4 >> q->sub_sampling);
1686 for (ch = 0; ch < q->channels; ch++)
1687 for (i = 0; i < q->frame_size; i++)
1688 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1693 * Init static data (does not depend on specific file)
1697 static av_cold void qdm2_init(QDM2Context *q) {
1698 static int initialized = 0;
1700 if (initialized != 0)
1705 ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1706 softclip_table_init();
1708 init_noise_samples();
1710 av_log(NULL, AV_LOG_DEBUG, "init done\n");
1715 * Init parameters from codec extradata
1717 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1719 QDM2Context *s = avctx->priv_data;
1722 int tmp_val, tmp, size;
1724 /* extradata parsing
1733 32 size (including this field)
1735 32 type (=QDM2 or QDMC)
1737 32 size (including this field, in bytes)
1738 32 tag (=QDCA) // maybe mandatory parameters
1741 32 samplerate (=44100)
1743 32 block size (=4096)
1744 32 frame size (=256) (for one channel)
1745 32 packet size (=1300)
1747 32 size (including this field, in bytes)
1748 32 tag (=QDCP) // maybe some tuneable parameters
1758 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1759 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1763 extradata = avctx->extradata;
1764 extradata_size = avctx->extradata_size;
1766 while (extradata_size > 7) {
1767 if (!memcmp(extradata, "frmaQDM", 7))
1773 if (extradata_size < 12) {
1774 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1779 if (memcmp(extradata, "frmaQDM", 7)) {
1780 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1784 if (extradata[7] == 'C') {
1786 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1791 extradata_size -= 8;
1793 size = AV_RB32(extradata);
1795 if(size > extradata_size){
1796 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1797 extradata_size, size);
1802 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1803 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1804 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1810 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1812 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1813 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1814 return AVERROR_INVALIDDATA;
1816 avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1819 avctx->sample_rate = AV_RB32(extradata);
1822 avctx->bit_rate = AV_RB32(extradata);
1825 s->group_size = AV_RB32(extradata);
1828 s->fft_size = AV_RB32(extradata);
1831 s->checksum_size = AV_RB32(extradata);
1832 if (s->checksum_size >= 1U << 28) {
1833 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1834 return AVERROR_INVALIDDATA;
1837 s->fft_order = av_log2(s->fft_size) + 1;
1839 // something like max decodable tones
1840 s->group_order = av_log2(s->group_size) + 1;
1841 s->frame_size = s->group_size / 16; // 16 iterations per super block
1843 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1844 return AVERROR_INVALIDDATA;
1846 s->sub_sampling = s->fft_order - 7;
1847 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1849 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1850 case 0: tmp = 40; break;
1851 case 1: tmp = 48; break;
1852 case 2: tmp = 56; break;
1853 case 3: tmp = 72; break;
1854 case 4: tmp = 80; break;
1855 case 5: tmp = 100;break;
1856 default: tmp=s->sub_sampling; break;
1859 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1860 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1861 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1862 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1863 s->cm_table_select = tmp_val;
1865 if (avctx->bit_rate <= 8000)
1866 s->coeff_per_sb_select = 0;
1867 else if (avctx->bit_rate < 16000)
1868 s->coeff_per_sb_select = 1;
1870 s->coeff_per_sb_select = 2;
1872 // Fail on unknown fft order
1873 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1874 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1877 if (s->fft_size != (1 << (s->fft_order - 1))) {
1878 av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1879 return AVERROR_INVALIDDATA;
1882 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1883 ff_mpadsp_init(&s->mpadsp);
1887 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1893 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1895 QDM2Context *s = avctx->priv_data;
1897 ff_rdft_end(&s->rdft_ctx);
1903 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1906 const int frame_size = (q->frame_size * q->channels);
1908 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1911 /* select input buffer */
1912 q->compressed_data = in;
1913 q->compressed_size = q->checksum_size;
1915 /* copy old block, clear new block of output samples */
1916 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1917 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1919 /* decode block of QDM2 compressed data */
1920 if (q->sub_packet == 0) {
1921 q->has_errors = 0; // zero it for a new super block
1922 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1923 qdm2_decode_super_block(q);
1926 /* parse subpackets */
1927 if (!q->has_errors) {
1928 if (q->sub_packet == 2)
1929 qdm2_decode_fft_packets(q);
1931 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1934 /* sound synthesis stage 1 (FFT) */
1935 for (ch = 0; ch < q->channels; ch++) {
1936 qdm2_calculate_fft(q, ch, q->sub_packet);
1938 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1939 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1944 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1945 if (!q->has_errors && q->do_synth_filter)
1946 qdm2_synthesis_filter(q, q->sub_packet);
1948 q->sub_packet = (q->sub_packet + 1) % 16;
1950 /* clip and convert output float[] to 16bit signed samples */
1951 for (i = 0; i < frame_size; i++) {
1952 int value = (int)q->output_buffer[i];
1954 if (value > SOFTCLIP_THRESHOLD)
1955 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1956 else if (value < -SOFTCLIP_THRESHOLD)
1957 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1966 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1967 int *got_frame_ptr, AVPacket *avpkt)
1969 AVFrame *frame = data;
1970 const uint8_t *buf = avpkt->data;
1971 int buf_size = avpkt->size;
1972 QDM2Context *s = avctx->priv_data;
1978 if(buf_size < s->checksum_size)
1981 /* get output buffer */
1982 frame->nb_samples = 16 * s->frame_size;
1983 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1985 out = (int16_t *)frame->data[0];
1987 for (i = 0; i < 16; i++) {
1988 if (qdm2_decode(s, buf, out) < 0)
1990 out += s->channels * s->frame_size;
1995 return s->checksum_size;
1998 AVCodec ff_qdm2_decoder =
2001 .type = AVMEDIA_TYPE_AUDIO,
2002 .id = AV_CODEC_ID_QDM2,
2003 .priv_data_size = sizeof(QDM2Context),
2004 .init = qdm2_decode_init,
2005 .close = qdm2_decode_close,
2006 .decode = qdm2_decode_frame,
2007 .capabilities = CODEC_CAP_DR1,
2008 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),