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 Libav.
10 * Libav 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 * Libav 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 Libav; 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 ALT_BITSTREAM_READER_LE
43 #include "mpegaudiodsp.h"
44 #include "mpegaudio.h"
47 #include "qdm2_tablegen.h"
53 #define QDM2_LIST_ADD(list, size, packet) \
56 list[size - 1].next = &list[size]; \
58 list[size].packet = packet; \
59 list[size].next = NULL; \
63 // Result is 8, 16 or 30
64 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
66 #define FIX_NOISE_IDX(noise_idx) \
67 if ((noise_idx) >= 3840) \
68 (noise_idx) -= 3840; \
70 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
72 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
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);
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 fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
143 int frame_size; ///< size of data frame
145 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
146 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
147 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
149 /// Packets and packet lists
150 QDM2SubPacket sub_packets[16]; ///< the packets themselves
151 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
152 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
153 int sub_packets_B; ///< number of packets on 'B' list
154 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
155 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
158 FFTTone fft_tones[1000];
161 FFTCoefficient fft_coefs[1000];
163 int fft_coefs_min_index[5];
164 int fft_coefs_max_index[5];
165 int fft_level_exp[6];
166 RDFTContext rdft_ctx;
170 const uint8_t *compressed_data;
172 float output_buffer[1024];
175 MPADSPContext mpadsp;
176 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
177 int synth_buf_offset[MPA_MAX_CHANNELS];
178 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
179 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
181 /// Mixed temporary data used in decoding
182 float tone_level[MPA_MAX_CHANNELS][30][64];
183 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
184 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
185 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
186 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
187 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
188 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
189 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
190 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
193 int has_errors; ///< packet has errors
194 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
195 int do_synth_filter; ///< used to perform or skip synthesis filter
198 int noise_idx; ///< index for dithering noise table
202 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
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 */
348 int tmp = vlc_stage3_values[value];
350 if ((value & ~3) > 0)
351 tmp += get_bits (gb, (value >> 2));
359 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
361 int value = qdm2_get_vlc (gb, vlc, 0, depth);
363 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
370 * @param data pointer to data to be checksum'ed
371 * @param length data length
372 * @param value checksum value
374 * @return 0 if checksum is OK
376 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
379 for (i=0; i < length; i++)
382 return (uint16_t)(value & 0xffff);
387 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
389 * @param gb bitreader context
390 * @param sub_packet packet under analysis
392 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
394 sub_packet->type = get_bits (gb, 8);
396 if (sub_packet->type == 0) {
397 sub_packet->size = 0;
398 sub_packet->data = NULL;
400 sub_packet->size = get_bits (gb, 8);
402 if (sub_packet->type & 0x80) {
403 sub_packet->size <<= 8;
404 sub_packet->size |= get_bits (gb, 8);
405 sub_packet->type &= 0x7f;
408 if (sub_packet->type == 0x7f)
409 sub_packet->type |= (get_bits (gb, 8) << 8);
411 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
414 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
415 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
420 * Return node pointer to first packet of requested type in list.
422 * @param list list of subpackets to be scanned
423 * @param type type of searched subpacket
424 * @return node pointer for subpacket if found, else NULL
426 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
428 while (list != NULL && list->packet != NULL) {
429 if (list->packet->type == type)
438 * Replace 8 elements with their average value.
439 * Called by qdm2_decode_superblock before starting subblock decoding.
443 static void average_quantized_coeffs (QDM2Context *q)
445 int i, j, n, ch, sum;
447 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
449 for (ch = 0; ch < q->nb_channels; ch++)
450 for (i = 0; i < n; i++) {
453 for (j = 0; j < 8; j++)
454 sum += q->quantized_coeffs[ch][i][j];
460 for (j=0; j < 8; j++)
461 q->quantized_coeffs[ch][i][j] = sum;
467 * Build subband samples with noise weighted by q->tone_level.
468 * Called by synthfilt_build_sb_samples.
471 * @param sb subband index
473 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
477 FIX_NOISE_IDX(q->noise_idx);
482 for (ch = 0; ch < q->nb_channels; ch++)
483 for (j = 0; j < 64; j++) {
484 q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
485 q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
491 * Called while processing data from subpackets 11 and 12.
492 * Used after making changes to coding_method array.
494 * @param sb subband index
495 * @param channels number of channels
496 * @param coding_method q->coding_method[0][0][0]
498 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
503 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};
505 for (ch = 0; ch < channels; ch++) {
506 for (j = 0; j < 64; ) {
507 if((coding_method[ch][sb][j] - 8) > 22) {
511 switch (switchtable[coding_method[ch][sb][j]-8]) {
512 case 0: run = 10; case_val = 10; break;
513 case 1: run = 1; case_val = 16; break;
514 case 2: run = 5; case_val = 24; break;
515 case 3: run = 3; case_val = 30; break;
516 case 4: run = 1; case_val = 30; break;
517 case 5: run = 1; case_val = 8; break;
518 default: run = 1; case_val = 8; break;
521 for (k = 0; k < run; k++)
523 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
526 //not debugged, almost never used
527 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
528 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
537 * Related to synthesis filter
538 * Called by process_subpacket_10
541 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
543 static void fill_tone_level_array (QDM2Context *q, int flag)
545 int i, sb, ch, sb_used;
548 // This should never happen
549 if (q->nb_channels <= 0)
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 // This should never happen
648 if (nb_channels <= 0)
651 if (!superblocktype_2_3) {
652 /* This case is untested, no samples available */
654 for (ch = 0; ch < nb_channels; ch++)
655 for (sb = 0; sb < 30; sb++) {
656 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
657 add1 = tone_level_idx[ch][sb][j] - 10;
660 add2 = add3 = add4 = 0;
662 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
667 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
672 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
676 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
679 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
681 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
684 for (ch = 0; ch < nb_channels; ch++)
685 for (sb = 0; sb < 30; sb++)
686 for (j = 0; j < 64; j++)
687 acc += tone_level_idx_temp[ch][sb][j];
689 multres = 0x66666667 * (acc * 10);
690 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
691 for (ch = 0; ch < nb_channels; ch++)
692 for (sb = 0; sb < 30; sb++)
693 for (j = 0; j < 64; j++) {
694 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
697 comp /= 256; // signed shift
725 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
727 for (sb = 0; sb < 30; sb++)
728 fix_coding_method_array(sb, nb_channels, coding_method);
729 for (ch = 0; ch < nb_channels; ch++)
730 for (sb = 0; sb < 30; sb++)
731 for (j = 0; j < 64; j++)
733 if (coding_method[ch][sb][j] < 10)
734 coding_method[ch][sb][j] = 10;
737 if (coding_method[ch][sb][j] < 16)
738 coding_method[ch][sb][j] = 16;
740 if (coding_method[ch][sb][j] < 30)
741 coding_method[ch][sb][j] = 30;
744 } else { // superblocktype_2_3 != 0
745 for (ch = 0; ch < nb_channels; ch++)
746 for (sb = 0; sb < 30; sb++)
747 for (j = 0; j < 64; j++)
748 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
757 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
758 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
761 * @param gb bitreader context
762 * @param length packet length in bits
763 * @param sb_min lower subband processed (sb_min included)
764 * @param sb_max higher subband processed (sb_max excluded)
766 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
768 int sb, j, k, n, ch, run, channels;
769 int joined_stereo, zero_encoding, chs;
771 float type34_div = 0;
772 float type34_predictor;
773 float samples[10], sign_bits[16];
776 // If no data use noise
777 for (sb=sb_min; sb < sb_max; sb++)
778 build_sb_samples_from_noise (q, sb);
783 for (sb = sb_min; sb < sb_max; sb++) {
784 FIX_NOISE_IDX(q->noise_idx);
786 channels = q->nb_channels;
788 if (q->nb_channels <= 1 || sb < 12)
793 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
796 if (BITS_LEFT(length,gb) >= 16)
797 for (j = 0; j < 16; j++)
798 sign_bits[j] = get_bits1 (gb);
800 for (j = 0; j < 64; j++)
801 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
802 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
804 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
808 for (ch = 0; ch < channels; ch++) {
809 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
810 type34_predictor = 0.0;
813 for (j = 0; j < 128; ) {
814 switch (q->coding_method[ch][sb][j / 2]) {
816 if (BITS_LEFT(length,gb) >= 10) {
818 for (k = 0; k < 5; k++) {
819 if ((j + 2 * k) >= 128)
821 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
825 for (k = 0; k < 5; k++)
826 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
828 for (k = 0; k < 5; k++)
829 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
831 for (k = 0; k < 10; k++)
832 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
838 if (BITS_LEFT(length,gb) >= 1) {
843 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
846 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
852 if (BITS_LEFT(length,gb) >= 10) {
854 for (k = 0; k < 5; k++) {
857 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
860 n = get_bits (gb, 8);
861 for (k = 0; k < 5; k++)
862 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
865 for (k = 0; k < 5; k++)
866 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
872 if (BITS_LEFT(length,gb) >= 7) {
874 for (k = 0; k < 3; k++)
875 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
877 for (k = 0; k < 3; k++)
878 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
884 if (BITS_LEFT(length,gb) >= 4)
885 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
887 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
893 if (BITS_LEFT(length,gb) >= 7) {
895 type34_div = (float)(1 << get_bits(gb, 2));
896 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
897 type34_predictor = samples[0];
900 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
901 type34_predictor = samples[0];
904 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
910 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
916 float tmp[10][MPA_MAX_CHANNELS];
918 for (k = 0; k < run; k++) {
919 tmp[k][0] = samples[k];
920 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
922 for (chs = 0; chs < q->nb_channels; chs++)
923 for (k = 0; k < run; k++)
925 q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
927 for (k = 0; k < run; k++)
929 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
940 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
941 * This is similar to process_subpacket_9, but for a single channel and for element [0]
942 * same VLC tables as process_subpacket_9 are used.
944 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
945 * @param gb bitreader context
946 * @param length packet length in bits
948 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
950 int i, k, run, level, diff;
952 if (BITS_LEFT(length,gb) < 16)
954 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
956 quantized_coeffs[0] = level;
958 for (i = 0; i < 7; ) {
959 if (BITS_LEFT(length,gb) < 16)
961 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
963 if (BITS_LEFT(length,gb) < 16)
965 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
967 for (k = 1; k <= run; k++)
968 quantized_coeffs[i + k] = (level + ((k * diff) / run));
977 * Related to synthesis filter, process data from packet 10
978 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
979 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
982 * @param gb bitreader context
983 * @param length packet length in bits
985 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
989 for (ch = 0; ch < q->nb_channels; ch++) {
990 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
992 if (BITS_LEFT(length,gb) < 16) {
993 memset(q->quantized_coeffs[ch][0], 0, 8);
998 n = q->sub_sampling + 1;
1000 for (sb = 0; sb < n; sb++)
1001 for (ch = 0; ch < q->nb_channels; ch++)
1002 for (j = 0; j < 8; j++) {
1003 if (BITS_LEFT(length,gb) < 1)
1005 if (get_bits1(gb)) {
1006 for (k=0; k < 8; k++) {
1007 if (BITS_LEFT(length,gb) < 16)
1009 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1012 for (k=0; k < 8; k++)
1013 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1017 n = QDM2_SB_USED(q->sub_sampling) - 4;
1019 for (sb = 0; sb < n; sb++)
1020 for (ch = 0; ch < q->nb_channels; ch++) {
1021 if (BITS_LEFT(length,gb) < 16)
1023 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1025 q->tone_level_idx_hi2[ch][sb] -= 16;
1027 for (j = 0; j < 8; j++)
1028 q->tone_level_idx_mid[ch][sb][j] = -16;
1031 n = QDM2_SB_USED(q->sub_sampling) - 5;
1033 for (sb = 0; sb < n; sb++)
1034 for (ch = 0; ch < q->nb_channels; ch++)
1035 for (j = 0; j < 8; j++) {
1036 if (BITS_LEFT(length,gb) < 16)
1038 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1043 * Process subpacket 9, init quantized_coeffs with data from it
1046 * @param node pointer to node with packet
1048 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1051 int i, j, k, n, ch, run, level, diff;
1053 init_get_bits(&gb, node->packet->data, node->packet->size*8);
1055 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1057 for (i = 1; i < n; i++)
1058 for (ch=0; ch < q->nb_channels; ch++) {
1059 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1060 q->quantized_coeffs[ch][i][0] = level;
1062 for (j = 0; j < (8 - 1); ) {
1063 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1064 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1066 for (k = 1; k <= run; k++)
1067 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1074 for (ch = 0; ch < q->nb_channels; ch++)
1075 for (i = 0; i < 8; i++)
1076 q->quantized_coeffs[ch][0][i] = 0;
1081 * Process subpacket 10 if not null, else
1084 * @param node pointer to node with packet
1085 * @param length packet length in bits
1087 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1091 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1094 init_tone_level_dequantization(q, &gb, length);
1095 fill_tone_level_array(q, 1);
1097 fill_tone_level_array(q, 0);
1103 * Process subpacket 11
1106 * @param node pointer to node with packet
1107 * @param length packet length in bit
1109 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1113 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1115 int c = get_bits (&gb, 13);
1118 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1119 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1122 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1127 * Process subpacket 12
1130 * @param node pointer to node with packet
1131 * @param length packet length in bits
1133 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1137 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1138 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1142 * Process new subpackets for synthesis filter
1145 * @param list list with synthesis filter packets (list D)
1147 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1149 QDM2SubPNode *nodes[4];
1151 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1152 if (nodes[0] != NULL)
1153 process_subpacket_9(q, nodes[0]);
1155 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1156 if (nodes[1] != NULL)
1157 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1159 process_subpacket_10(q, NULL, 0);
1161 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1162 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1163 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1165 process_subpacket_11(q, NULL, 0);
1167 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1168 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1169 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1171 process_subpacket_12(q, NULL, 0);
1176 * Decode superblock, fill packet lists.
1180 static void qdm2_decode_super_block (QDM2Context *q)
1183 QDM2SubPacket header, *packet;
1184 int i, packet_bytes, sub_packet_size, sub_packets_D;
1185 unsigned int next_index = 0;
1187 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1188 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1189 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1191 q->sub_packets_B = 0;
1194 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1196 init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1197 qdm2_decode_sub_packet_header(&gb, &header);
1199 if (header.type < 2 || header.type >= 8) {
1201 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1205 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1206 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1208 init_get_bits(&gb, header.data, header.size*8);
1210 if (header.type == 2 || header.type == 4 || header.type == 5) {
1211 int csum = 257 * get_bits(&gb, 8);
1212 csum += 2 * get_bits(&gb, 8);
1214 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1218 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1223 q->sub_packet_list_B[0].packet = NULL;
1224 q->sub_packet_list_D[0].packet = NULL;
1226 for (i = 0; i < 6; i++)
1227 if (--q->fft_level_exp[i] < 0)
1228 q->fft_level_exp[i] = 0;
1230 for (i = 0; packet_bytes > 0; i++) {
1233 q->sub_packet_list_A[i].next = NULL;
1236 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1238 /* seek to next block */
1239 init_get_bits(&gb, header.data, header.size*8);
1240 skip_bits(&gb, next_index*8);
1242 if (next_index >= header.size)
1246 /* decode subpacket */
1247 packet = &q->sub_packets[i];
1248 qdm2_decode_sub_packet_header(&gb, packet);
1249 next_index = packet->size + get_bits_count(&gb) / 8;
1250 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1252 if (packet->type == 0)
1255 if (sub_packet_size > packet_bytes) {
1256 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1258 packet->size += packet_bytes - sub_packet_size;
1261 packet_bytes -= sub_packet_size;
1263 /* add subpacket to 'all subpackets' list */
1264 q->sub_packet_list_A[i].packet = packet;
1266 /* add subpacket to related list */
1267 if (packet->type == 8) {
1268 SAMPLES_NEEDED_2("packet type 8");
1270 } else if (packet->type >= 9 && packet->type <= 12) {
1271 /* packets for MPEG Audio like Synthesis Filter */
1272 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1273 } else if (packet->type == 13) {
1274 for (j = 0; j < 6; j++)
1275 q->fft_level_exp[j] = get_bits(&gb, 6);
1276 } else if (packet->type == 14) {
1277 for (j = 0; j < 6; j++)
1278 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1279 } else if (packet->type == 15) {
1280 SAMPLES_NEEDED_2("packet type 15")
1282 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1283 /* packets for FFT */
1284 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1286 } // Packet bytes loop
1288 /* **************************************************************** */
1289 if (q->sub_packet_list_D[0].packet != NULL) {
1290 process_synthesis_subpackets(q, q->sub_packet_list_D);
1291 q->do_synth_filter = 1;
1292 } else if (q->do_synth_filter) {
1293 process_subpacket_10(q, NULL, 0);
1294 process_subpacket_11(q, NULL, 0);
1295 process_subpacket_12(q, NULL, 0);
1297 /* **************************************************************** */
1301 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1302 int offset, int duration, int channel,
1305 if (q->fft_coefs_min_index[duration] < 0)
1306 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1308 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1309 q->fft_coefs[q->fft_coefs_index].channel = channel;
1310 q->fft_coefs[q->fft_coefs_index].offset = offset;
1311 q->fft_coefs[q->fft_coefs_index].exp = exp;
1312 q->fft_coefs[q->fft_coefs_index].phase = phase;
1313 q->fft_coefs_index++;
1317 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1319 int channel, stereo, phase, exp;
1320 int local_int_4, local_int_8, stereo_phase, local_int_10;
1321 int local_int_14, stereo_exp, local_int_20, local_int_28;
1327 local_int_8 = (4 - duration);
1328 local_int_10 = 1 << (q->group_order - duration - 1);
1332 if (q->superblocktype_2_3) {
1333 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1336 local_int_4 += local_int_10;
1337 local_int_28 += (1 << local_int_8);
1339 local_int_4 += 8*local_int_10;
1340 local_int_28 += (8 << local_int_8);
1345 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1346 while (offset >= (local_int_10 - 1)) {
1347 offset += (1 - (local_int_10 - 1));
1348 local_int_4 += local_int_10;
1349 local_int_28 += (1 << local_int_8);
1353 if (local_int_4 >= q->group_size)
1356 local_int_14 = (offset >> local_int_8);
1358 if (q->nb_channels > 1) {
1359 channel = get_bits1(gb);
1360 stereo = get_bits1(gb);
1366 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1367 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1368 exp = (exp < 0) ? 0 : exp;
1370 phase = get_bits(gb, 3);
1375 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1376 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1377 if (stereo_phase < 0)
1381 if (q->frequency_range > (local_int_14 + 1)) {
1382 int sub_packet = (local_int_20 + local_int_28);
1384 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1386 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1394 static void qdm2_decode_fft_packets (QDM2Context *q)
1396 int i, j, min, max, value, type, unknown_flag;
1399 if (q->sub_packet_list_B[0].packet == NULL)
1402 /* reset minimum indexes for FFT coefficients */
1403 q->fft_coefs_index = 0;
1404 for (i=0; i < 5; i++)
1405 q->fft_coefs_min_index[i] = -1;
1407 /* process subpackets ordered by type, largest type first */
1408 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1409 QDM2SubPacket *packet= NULL;
1411 /* find subpacket with largest type less than max */
1412 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1413 value = q->sub_packet_list_B[j].packet->type;
1414 if (value > min && value < max) {
1416 packet = q->sub_packet_list_B[j].packet;
1422 /* check for errors (?) */
1426 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1429 /* decode FFT tones */
1430 init_get_bits (&gb, packet->data, packet->size*8);
1432 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1437 type = packet->type;
1439 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1440 int duration = q->sub_sampling + 5 - (type & 15);
1442 if (duration >= 0 && duration < 4)
1443 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1444 } else if (type == 31) {
1445 for (j=0; j < 4; j++)
1446 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1447 } else if (type == 46) {
1448 for (j=0; j < 6; j++)
1449 q->fft_level_exp[j] = get_bits(&gb, 6);
1450 for (j=0; j < 4; j++)
1451 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1453 } // Loop on B packets
1455 /* calculate maximum indexes for FFT coefficients */
1456 for (i = 0, j = -1; i < 5; i++)
1457 if (q->fft_coefs_min_index[i] >= 0) {
1459 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1463 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1467 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1472 const double iscale = 2.0*M_PI / 512.0;
1474 tone->phase += tone->phase_shift;
1476 /* calculate current level (maximum amplitude) of tone */
1477 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1478 c.im = level * sin(tone->phase*iscale);
1479 c.re = level * cos(tone->phase*iscale);
1481 /* generate FFT coefficients for tone */
1482 if (tone->duration >= 3 || tone->cutoff >= 3) {
1483 tone->complex[0].im += c.im;
1484 tone->complex[0].re += c.re;
1485 tone->complex[1].im -= c.im;
1486 tone->complex[1].re -= c.re;
1488 f[1] = -tone->table[4];
1489 f[0] = tone->table[3] - tone->table[0];
1490 f[2] = 1.0 - tone->table[2] - tone->table[3];
1491 f[3] = tone->table[1] + tone->table[4] - 1.0;
1492 f[4] = tone->table[0] - tone->table[1];
1493 f[5] = tone->table[2];
1494 for (i = 0; i < 2; i++) {
1495 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1496 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1498 for (i = 0; i < 4; i++) {
1499 tone->complex[i].re += c.re * f[i+2];
1500 tone->complex[i].im += c.im * f[i+2];
1504 /* copy the tone if it has not yet died out */
1505 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1506 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1507 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1512 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1515 const double iscale = 0.25 * M_PI;
1517 for (ch = 0; ch < q->channels; ch++) {
1518 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1522 /* apply FFT tones with duration 4 (1 FFT period) */
1523 if (q->fft_coefs_min_index[4] >= 0)
1524 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1528 if (q->fft_coefs[i].sub_packet != sub_packet)
1531 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1532 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];
1534 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1535 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1536 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1537 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1538 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1539 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1542 /* generate existing FFT tones */
1543 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1544 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1545 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1548 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1549 for (i = 0; i < 4; i++)
1550 if (q->fft_coefs_min_index[i] >= 0) {
1551 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1555 if (q->fft_coefs[j].sub_packet != sub_packet)
1559 offset = q->fft_coefs[j].offset >> four_i;
1560 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1562 if (offset < q->frequency_range) {
1564 tone.cutoff = offset;
1566 tone.cutoff = (offset >= 60) ? 3 : 2;
1568 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];
1569 tone.complex = &q->fft.complex[ch][offset];
1570 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1571 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1572 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1574 tone.time_index = 0;
1576 qdm2_fft_generate_tone(q, &tone);
1579 q->fft_coefs_min_index[i] = j;
1584 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1586 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1588 q->fft.complex[channel][0].re *= 2.0f;
1589 q->fft.complex[channel][0].im = 0.0f;
1590 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1591 /* add samples to output buffer */
1592 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1593 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
1599 * @param index subpacket number
1601 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1603 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1605 /* copy sb_samples */
1606 sb_used = QDM2_SB_USED(q->sub_sampling);
1608 for (ch = 0; ch < q->channels; ch++)
1609 for (i = 0; i < 8; i++)
1610 for (k=sb_used; k < SBLIMIT; k++)
1611 q->sb_samples[ch][(8 * index) + i][k] = 0;
1613 for (ch = 0; ch < q->nb_channels; ch++) {
1614 float *samples_ptr = q->samples + ch;
1616 for (i = 0; i < 8; i++) {
1617 ff_mpa_synth_filter_float(&q->mpadsp,
1618 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1619 ff_mpa_synth_window_float, &dither_state,
1620 samples_ptr, q->nb_channels,
1621 q->sb_samples[ch][(8 * index) + i]);
1622 samples_ptr += 32 * q->nb_channels;
1626 /* add samples to output buffer */
1627 sub_sampling = (4 >> q->sub_sampling);
1629 for (ch = 0; ch < q->channels; ch++)
1630 for (i = 0; i < q->frame_size; i++)
1631 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1636 * Init static data (does not depend on specific file)
1640 static av_cold void qdm2_init(QDM2Context *q) {
1641 static int initialized = 0;
1643 if (initialized != 0)
1648 ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1649 softclip_table_init();
1651 init_noise_samples();
1653 av_log(NULL, AV_LOG_DEBUG, "init done\n");
1658 static void dump_context(QDM2Context *q)
1661 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1662 PRINT("compressed_data",q->compressed_data);
1663 PRINT("compressed_size",q->compressed_size);
1664 PRINT("frame_size",q->frame_size);
1665 PRINT("checksum_size",q->checksum_size);
1666 PRINT("channels",q->channels);
1667 PRINT("nb_channels",q->nb_channels);
1668 PRINT("fft_frame_size",q->fft_frame_size);
1669 PRINT("fft_size",q->fft_size);
1670 PRINT("sub_sampling",q->sub_sampling);
1671 PRINT("fft_order",q->fft_order);
1672 PRINT("group_order",q->group_order);
1673 PRINT("group_size",q->group_size);
1674 PRINT("sub_packet",q->sub_packet);
1675 PRINT("frequency_range",q->frequency_range);
1676 PRINT("has_errors",q->has_errors);
1677 PRINT("fft_tone_end",q->fft_tone_end);
1678 PRINT("fft_tone_start",q->fft_tone_start);
1679 PRINT("fft_coefs_index",q->fft_coefs_index);
1680 PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1681 PRINT("cm_table_select",q->cm_table_select);
1682 PRINT("noise_idx",q->noise_idx);
1684 for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1686 FFTTone *t = &q->fft_tones[i];
1688 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1689 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1690 // PRINT(" level", t->level);
1691 PRINT(" phase", t->phase);
1692 PRINT(" phase_shift", t->phase_shift);
1693 PRINT(" duration", t->duration);
1694 PRINT(" samples_im", t->samples_im);
1695 PRINT(" samples_re", t->samples_re);
1696 PRINT(" table", t->table);
1704 * Init parameters from codec extradata
1706 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1708 QDM2Context *s = avctx->priv_data;
1711 int tmp_val, tmp, size;
1713 /* extradata parsing
1722 32 size (including this field)
1724 32 type (=QDM2 or QDMC)
1726 32 size (including this field, in bytes)
1727 32 tag (=QDCA) // maybe mandatory parameters
1730 32 samplerate (=44100)
1732 32 block size (=4096)
1733 32 frame size (=256) (for one channel)
1734 32 packet size (=1300)
1736 32 size (including this field, in bytes)
1737 32 tag (=QDCP) // maybe some tuneable parameters
1747 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1748 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1752 extradata = avctx->extradata;
1753 extradata_size = avctx->extradata_size;
1755 while (extradata_size > 7) {
1756 if (!memcmp(extradata, "frmaQDM", 7))
1762 if (extradata_size < 12) {
1763 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1768 if (memcmp(extradata, "frmaQDM", 7)) {
1769 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1773 if (extradata[7] == 'C') {
1775 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1780 extradata_size -= 8;
1782 size = AV_RB32(extradata);
1784 if(size > extradata_size){
1785 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1786 extradata_size, size);
1791 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1792 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1793 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1799 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1802 avctx->sample_rate = AV_RB32(extradata);
1805 avctx->bit_rate = AV_RB32(extradata);
1808 s->group_size = AV_RB32(extradata);
1811 s->fft_size = AV_RB32(extradata);
1814 s->checksum_size = AV_RB32(extradata);
1816 s->fft_order = av_log2(s->fft_size) + 1;
1817 s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1819 // something like max decodable tones
1820 s->group_order = av_log2(s->group_size) + 1;
1821 s->frame_size = s->group_size / 16; // 16 iterations per super block
1823 s->sub_sampling = s->fft_order - 7;
1824 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1826 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1827 case 0: tmp = 40; break;
1828 case 1: tmp = 48; break;
1829 case 2: tmp = 56; break;
1830 case 3: tmp = 72; break;
1831 case 4: tmp = 80; break;
1832 case 5: tmp = 100;break;
1833 default: tmp=s->sub_sampling; break;
1836 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1837 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1838 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1839 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1840 s->cm_table_select = tmp_val;
1842 if (s->sub_sampling == 0)
1845 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1852 s->coeff_per_sb_select = 0;
1853 else if (tmp <= 16000)
1854 s->coeff_per_sb_select = 1;
1856 s->coeff_per_sb_select = 2;
1858 // Fail on unknown fft order
1859 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1860 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1864 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1865 ff_mpadsp_init(&s->mpadsp);
1869 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1876 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1878 QDM2Context *s = avctx->priv_data;
1880 ff_rdft_end(&s->rdft_ctx);
1886 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1889 const int frame_size = (q->frame_size * q->channels);
1891 /* select input buffer */
1892 q->compressed_data = in;
1893 q->compressed_size = q->checksum_size;
1897 /* copy old block, clear new block of output samples */
1898 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1899 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1901 /* decode block of QDM2 compressed data */
1902 if (q->sub_packet == 0) {
1903 q->has_errors = 0; // zero it for a new super block
1904 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1905 qdm2_decode_super_block(q);
1908 /* parse subpackets */
1909 if (!q->has_errors) {
1910 if (q->sub_packet == 2)
1911 qdm2_decode_fft_packets(q);
1913 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1916 /* sound synthesis stage 1 (FFT) */
1917 for (ch = 0; ch < q->channels; ch++) {
1918 qdm2_calculate_fft(q, ch, q->sub_packet);
1920 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1921 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1926 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1927 if (!q->has_errors && q->do_synth_filter)
1928 qdm2_synthesis_filter(q, q->sub_packet);
1930 q->sub_packet = (q->sub_packet + 1) % 16;
1932 /* clip and convert output float[] to 16bit signed samples */
1933 for (i = 0; i < frame_size; i++) {
1934 int value = (int)q->output_buffer[i];
1936 if (value > SOFTCLIP_THRESHOLD)
1937 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1938 else if (value < -SOFTCLIP_THRESHOLD)
1939 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1948 static int qdm2_decode_frame(AVCodecContext *avctx,
1949 void *data, int *data_size,
1952 const uint8_t *buf = avpkt->data;
1953 int buf_size = avpkt->size;
1954 QDM2Context *s = avctx->priv_data;
1955 int16_t *out = data;
1960 if(buf_size < s->checksum_size)
1963 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
1964 buf_size, buf, s->checksum_size, data, *data_size);
1966 for (i = 0; i < 16; i++) {
1967 if (qdm2_decode(s, buf, out) < 0)
1969 out += s->channels * s->frame_size;
1972 *data_size = (uint8_t*)out - (uint8_t*)data;
1974 return s->checksum_size;
1977 AVCodec ff_qdm2_decoder =
1980 .type = AVMEDIA_TYPE_AUDIO,
1981 .id = CODEC_ID_QDM2,
1982 .priv_data_size = sizeof(QDM2Context),
1983 .init = qdm2_decode_init,
1984 .close = qdm2_decode_close,
1985 .decode = qdm2_decode_frame,
1986 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),