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"
47 #include "qdm2_tablegen.h"
49 #define QDM2_LIST_ADD(list, size, packet) \
52 list[size - 1].next = &list[size]; \
54 list[size].packet = packet; \
55 list[size].next = NULL; \
59 // Result is 8, 16 or 30
60 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
62 #define FIX_NOISE_IDX(noise_idx) \
63 if ((noise_idx) >= 3840) \
64 (noise_idx) -= 3840; \
66 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
68 #define SAMPLES_NEEDED \
69 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
71 #define SAMPLES_NEEDED_2(why) \
72 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
74 #define QDM2_MAX_FRAME_SIZE 512
76 typedef int8_t sb_int8_array[2][30][64];
81 typedef struct QDM2SubPacket {
82 int type; ///< subpacket type
83 unsigned int size; ///< subpacket size
84 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
88 * A node in the subpacket list
90 typedef struct QDM2SubPNode {
91 QDM2SubPacket *packet; ///< packet
92 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
95 typedef struct QDM2Complex {
100 typedef struct FFTTone {
102 QDM2Complex *complex;
111 typedef struct FFTCoefficient {
119 typedef struct QDM2FFT {
120 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
124 * QDM2 decoder context
126 typedef struct QDM2Context {
127 /// Parameters from codec header, do not change during playback
128 int nb_channels; ///< number of channels
129 int channels; ///< number of channels
130 int group_size; ///< size of frame group (16 frames per group)
131 int fft_size; ///< size of FFT, in complex numbers
132 int checksum_size; ///< size of data block, used also for checksum
134 /// Parameters built from header parameters, do not change during playback
135 int group_order; ///< order of frame group
136 int fft_order; ///< order of FFT (actually fftorder+1)
137 int frame_size; ///< size of data frame
139 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
140 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
141 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
143 /// Packets and packet lists
144 QDM2SubPacket sub_packets[16]; ///< the packets themselves
145 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
146 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
147 int sub_packets_B; ///< number of packets on 'B' list
148 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
149 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
152 FFTTone fft_tones[1000];
155 FFTCoefficient fft_coefs[1000];
157 int fft_coefs_min_index[5];
158 int fft_coefs_max_index[5];
159 int fft_level_exp[6];
160 RDFTContext rdft_ctx;
164 const uint8_t *compressed_data;
166 float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
169 MPADSPContext mpadsp;
170 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
171 int synth_buf_offset[MPA_MAX_CHANNELS];
172 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
173 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
175 /// Mixed temporary data used in decoding
176 float tone_level[MPA_MAX_CHANNELS][30][64];
177 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
178 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
179 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
180 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
181 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
182 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
183 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
184 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
187 int has_errors; ///< packet has errors
188 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
189 int do_synth_filter; ///< used to perform or skip synthesis filter
192 int noise_idx; ///< index for dithering noise table
195 static const int switchtable[23] = {
196 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
199 static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
203 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
205 /* stage-2, 3 bits exponent escape sequence */
207 value = get_bits(gb, get_bits(gb, 3) + 1);
209 /* stage-3, optional */
214 av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
218 tmp= vlc_stage3_values[value];
220 if ((value & ~3) > 0)
221 tmp += get_bits(gb, (value >> 2));
228 static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
230 int value = qdm2_get_vlc(gb, vlc, 0, depth);
232 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
238 * @param data pointer to data to be checksum'ed
239 * @param length data length
240 * @param value checksum value
242 * @return 0 if checksum is OK
244 static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
248 for (i = 0; i < length; i++)
251 return (uint16_t)(value & 0xffff);
255 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
257 * @param gb bitreader context
258 * @param sub_packet packet under analysis
260 static void qdm2_decode_sub_packet_header(GetBitContext *gb,
261 QDM2SubPacket *sub_packet)
263 sub_packet->type = get_bits(gb, 8);
265 if (sub_packet->type == 0) {
266 sub_packet->size = 0;
267 sub_packet->data = NULL;
269 sub_packet->size = get_bits(gb, 8);
271 if (sub_packet->type & 0x80) {
272 sub_packet->size <<= 8;
273 sub_packet->size |= get_bits(gb, 8);
274 sub_packet->type &= 0x7f;
277 if (sub_packet->type == 0x7f)
278 sub_packet->type |= (get_bits(gb, 8) << 8);
280 // FIXME: this depends on bitreader-internal data
281 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
284 av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
285 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
289 * Return node pointer to first packet of requested type in list.
291 * @param list list of subpackets to be scanned
292 * @param type type of searched subpacket
293 * @return node pointer for subpacket if found, else NULL
295 static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
298 while (list && list->packet) {
299 if (list->packet->type == type)
307 * Replace 8 elements with their average value.
308 * Called by qdm2_decode_superblock before starting subblock decoding.
312 static void average_quantized_coeffs(QDM2Context *q)
314 int i, j, n, ch, sum;
316 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
318 for (ch = 0; ch < q->nb_channels; ch++)
319 for (i = 0; i < n; i++) {
322 for (j = 0; j < 8; j++)
323 sum += q->quantized_coeffs[ch][i][j];
329 for (j = 0; j < 8; j++)
330 q->quantized_coeffs[ch][i][j] = sum;
335 * Build subband samples with noise weighted by q->tone_level.
336 * Called by synthfilt_build_sb_samples.
339 * @param sb subband index
341 static void build_sb_samples_from_noise(QDM2Context *q, int sb)
345 FIX_NOISE_IDX(q->noise_idx);
350 for (ch = 0; ch < q->nb_channels; ch++) {
351 for (j = 0; j < 64; j++) {
352 q->sb_samples[ch][j * 2][sb] =
353 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
354 q->sb_samples[ch][j * 2 + 1][sb] =
355 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
361 * Called while processing data from subpackets 11 and 12.
362 * Used after making changes to coding_method array.
364 * @param sb subband index
365 * @param channels number of channels
366 * @param coding_method q->coding_method[0][0][0]
368 static int fix_coding_method_array(int sb, int channels,
369 sb_int8_array coding_method)
375 for (ch = 0; ch < channels; ch++) {
376 for (j = 0; j < 64; ) {
377 if (coding_method[ch][sb][j] < 8)
379 if ((coding_method[ch][sb][j] - 8) > 22) {
383 switch (switchtable[coding_method[ch][sb][j] - 8]) {
407 for (k = 0; k < run; k++) {
409 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) {
412 //not debugged, almost never used
413 memset(&coding_method[ch][sb][j + k], case_val,
415 memset(&coding_method[ch][sb][j + k], case_val,
428 * Related to synthesis filter
429 * Called by process_subpacket_10
432 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
434 static void fill_tone_level_array(QDM2Context *q, int flag)
436 int i, sb, ch, sb_used;
439 for (ch = 0; ch < q->nb_channels; ch++)
440 for (sb = 0; sb < 30; sb++)
441 for (i = 0; i < 8; i++) {
442 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
443 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
444 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
446 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
449 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
452 sb_used = QDM2_SB_USED(q->sub_sampling);
454 if ((q->superblocktype_2_3 != 0) && !flag) {
455 for (sb = 0; sb < sb_used; sb++)
456 for (ch = 0; ch < q->nb_channels; ch++)
457 for (i = 0; i < 64; i++) {
458 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
459 if (q->tone_level_idx[ch][sb][i] < 0)
460 q->tone_level[ch][sb][i] = 0;
462 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
465 tab = q->superblocktype_2_3 ? 0 : 1;
466 for (sb = 0; sb < sb_used; sb++) {
467 if ((sb >= 4) && (sb <= 23)) {
468 for (ch = 0; ch < q->nb_channels; ch++)
469 for (i = 0; i < 64; i++) {
470 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
471 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
472 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
473 q->tone_level_idx_hi2[ch][sb - 4];
474 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
475 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
476 q->tone_level[ch][sb][i] = 0;
478 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
482 for (ch = 0; ch < q->nb_channels; ch++)
483 for (i = 0; i < 64; i++) {
484 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
485 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
486 q->tone_level_idx_hi2[ch][sb - 4];
487 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
488 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
489 q->tone_level[ch][sb][i] = 0;
491 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
494 for (ch = 0; ch < q->nb_channels; ch++)
495 for (i = 0; i < 64; i++) {
496 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
497 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
498 q->tone_level[ch][sb][i] = 0;
500 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
509 * Related to synthesis filter
510 * Called by process_subpacket_11
511 * c is built with data from subpacket 11
512 * Most of this function is used only if superblock_type_2_3 == 0,
513 * never seen it in samples.
515 * @param tone_level_idx
516 * @param tone_level_idx_temp
517 * @param coding_method q->coding_method[0][0][0]
518 * @param nb_channels number of channels
519 * @param c coming from subpacket 11, passed as 8*c
520 * @param superblocktype_2_3 flag based on superblock packet type
521 * @param cm_table_select q->cm_table_select
523 static void fill_coding_method_array(sb_int8_array tone_level_idx,
524 sb_int8_array tone_level_idx_temp,
525 sb_int8_array coding_method,
527 int c, int superblocktype_2_3,
531 int tmp, acc, esp_40, comp;
532 int add1, add2, add3, add4;
535 if (!superblocktype_2_3) {
536 /* This case is untested, no samples available */
537 avpriv_request_sample(NULL, "!superblocktype_2_3");
539 for (ch = 0; ch < nb_channels; ch++)
540 for (sb = 0; sb < 30; sb++) {
541 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
542 add1 = tone_level_idx[ch][sb][j] - 10;
545 add2 = add3 = add4 = 0;
547 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
552 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
557 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
561 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
564 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
566 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
569 for (ch = 0; ch < nb_channels; ch++)
570 for (sb = 0; sb < 30; sb++)
571 for (j = 0; j < 64; j++)
572 acc += tone_level_idx_temp[ch][sb][j];
574 multres = 0x66666667LL * (acc * 10);
575 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
576 for (ch = 0; ch < nb_channels; ch++)
577 for (sb = 0; sb < 30; sb++)
578 for (j = 0; j < 64; j++) {
579 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
582 comp /= 256; // signed shift
610 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
612 for (sb = 0; sb < 30; sb++)
613 fix_coding_method_array(sb, nb_channels, coding_method);
614 for (ch = 0; ch < nb_channels; ch++)
615 for (sb = 0; sb < 30; sb++)
616 for (j = 0; j < 64; j++)
618 if (coding_method[ch][sb][j] < 10)
619 coding_method[ch][sb][j] = 10;
622 if (coding_method[ch][sb][j] < 16)
623 coding_method[ch][sb][j] = 16;
625 if (coding_method[ch][sb][j] < 30)
626 coding_method[ch][sb][j] = 30;
629 } else { // superblocktype_2_3 != 0
630 for (ch = 0; ch < nb_channels; ch++)
631 for (sb = 0; sb < 30; sb++)
632 for (j = 0; j < 64; j++)
633 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
639 * Called by process_subpacket_11 to process more data from subpacket 11
641 * Called by process_subpacket_12 to process data from subpacket 12 with
645 * @param gb bitreader context
646 * @param length packet length in bits
647 * @param sb_min lower subband processed (sb_min included)
648 * @param sb_max higher subband processed (sb_max excluded)
650 static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
651 int length, int sb_min, int sb_max)
653 int sb, j, k, n, ch, run, channels;
654 int joined_stereo, zero_encoding;
656 float type34_div = 0;
657 float type34_predictor;
659 int sign_bits[16] = {0};
662 // If no data use noise
663 for (sb=sb_min; sb < sb_max; sb++)
664 build_sb_samples_from_noise(q, sb);
669 for (sb = sb_min; sb < sb_max; sb++) {
670 channels = q->nb_channels;
672 if (q->nb_channels <= 1 || sb < 12)
677 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
680 if (get_bits_left(gb) >= 16)
681 for (j = 0; j < 16; j++)
682 sign_bits[j] = get_bits1(gb);
684 for (j = 0; j < 64; j++)
685 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
686 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
688 if (fix_coding_method_array(sb, q->nb_channels,
690 av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
691 build_sb_samples_from_noise(q, sb);
697 for (ch = 0; ch < channels; ch++) {
698 FIX_NOISE_IDX(q->noise_idx);
699 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
700 type34_predictor = 0.0;
703 for (j = 0; j < 128; ) {
704 switch (q->coding_method[ch][sb][j / 2]) {
706 if (get_bits_left(gb) >= 10) {
708 for (k = 0; k < 5; k++) {
709 if ((j + 2 * k) >= 128)
711 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
716 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
717 return AVERROR_INVALIDDATA;
720 for (k = 0; k < 5; k++)
721 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
723 for (k = 0; k < 5; k++)
724 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
726 for (k = 0; k < 10; k++)
727 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
733 if (get_bits_left(gb) >= 1) {
738 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
741 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
747 if (get_bits_left(gb) >= 10) {
749 for (k = 0; k < 5; k++) {
752 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
755 n = get_bits (gb, 8);
757 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
758 return AVERROR_INVALIDDATA;
761 for (k = 0; k < 5; k++)
762 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
765 for (k = 0; k < 5; k++)
766 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
772 if (get_bits_left(gb) >= 7) {
775 av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
776 return AVERROR_INVALIDDATA;
779 for (k = 0; k < 3; k++)
780 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
782 for (k = 0; k < 3; k++)
783 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
789 if (get_bits_left(gb) >= 4) {
790 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
791 if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
792 av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
793 return AVERROR_INVALIDDATA;
795 samples[0] = type30_dequant[index];
797 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
803 if (get_bits_left(gb) >= 7) {
805 type34_div = (float)(1 << get_bits(gb, 2));
806 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
807 type34_predictor = samples[0];
810 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
811 if (index >= FF_ARRAY_ELEMS(type34_delta)) {
812 av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
813 return AVERROR_INVALIDDATA;
815 samples[0] = type34_delta[index] / type34_div + type34_predictor;
816 type34_predictor = samples[0];
819 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
825 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
831 for (k = 0; k < run && j + k < 128; k++) {
832 q->sb_samples[0][j + k][sb] =
833 q->tone_level[0][sb][(j + k) / 2] * samples[k];
834 if (q->nb_channels == 2) {
835 if (sign_bits[(j + k) / 8])
836 q->sb_samples[1][j + k][sb] =
837 q->tone_level[1][sb][(j + k) / 2] * -samples[k];
839 q->sb_samples[1][j + k][sb] =
840 q->tone_level[1][sb][(j + k) / 2] * samples[k];
844 for (k = 0; k < run; k++)
846 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
857 * Init the first element of a channel in quantized_coeffs with data
858 * from packet 10 (quantized_coeffs[ch][0]).
859 * This is similar to process_subpacket_9, but for a single channel
860 * and for element [0]
861 * same VLC tables as process_subpacket_9 are used.
863 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
864 * @param gb bitreader context
866 static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
869 int i, k, run, level, diff;
871 if (get_bits_left(gb) < 16)
873 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
875 quantized_coeffs[0] = level;
877 for (i = 0; i < 7; ) {
878 if (get_bits_left(gb) < 16)
880 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
885 if (get_bits_left(gb) < 16)
887 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
889 for (k = 1; k <= run; k++)
890 quantized_coeffs[i + k] = (level + ((k * diff) / run));
899 * Related to synthesis filter, process data from packet 10
900 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
901 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
902 * data from packet 10
905 * @param gb bitreader context
907 static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
911 for (ch = 0; ch < q->nb_channels; ch++) {
912 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
914 if (get_bits_left(gb) < 16) {
915 memset(q->quantized_coeffs[ch][0], 0, 8);
920 n = q->sub_sampling + 1;
922 for (sb = 0; sb < n; sb++)
923 for (ch = 0; ch < q->nb_channels; ch++)
924 for (j = 0; j < 8; j++) {
925 if (get_bits_left(gb) < 1)
928 for (k=0; k < 8; k++) {
929 if (get_bits_left(gb) < 16)
931 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
934 for (k=0; k < 8; k++)
935 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
939 n = QDM2_SB_USED(q->sub_sampling) - 4;
941 for (sb = 0; sb < n; sb++)
942 for (ch = 0; ch < q->nb_channels; ch++) {
943 if (get_bits_left(gb) < 16)
945 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
947 q->tone_level_idx_hi2[ch][sb] -= 16;
949 for (j = 0; j < 8; j++)
950 q->tone_level_idx_mid[ch][sb][j] = -16;
953 n = QDM2_SB_USED(q->sub_sampling) - 5;
955 for (sb = 0; sb < n; sb++)
956 for (ch = 0; ch < q->nb_channels; ch++)
957 for (j = 0; j < 8; j++) {
958 if (get_bits_left(gb) < 16)
960 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
965 * Process subpacket 9, init quantized_coeffs with data from it
968 * @param node pointer to node with packet
970 static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
973 int i, j, k, n, ch, run, level, diff;
975 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
977 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
979 for (i = 1; i < n; i++)
980 for (ch = 0; ch < q->nb_channels; ch++) {
981 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
982 q->quantized_coeffs[ch][i][0] = level;
984 for (j = 0; j < (8 - 1); ) {
985 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
986 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
991 for (k = 1; k <= run; k++)
992 q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
999 for (ch = 0; ch < q->nb_channels; ch++)
1000 for (i = 0; i < 8; i++)
1001 q->quantized_coeffs[ch][0][i] = 0;
1007 * Process subpacket 10 if not null, else
1010 * @param node pointer to node with packet
1012 static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
1017 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1018 init_tone_level_dequantization(q, &gb);
1019 fill_tone_level_array(q, 1);
1021 fill_tone_level_array(q, 0);
1026 * Process subpacket 11
1029 * @param node pointer to node with packet
1031 static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
1037 length = node->packet->size * 8;
1038 init_get_bits(&gb, node->packet->data, length);
1042 int c = get_bits(&gb, 13);
1045 fill_coding_method_array(q->tone_level_idx,
1046 q->tone_level_idx_temp, q->coding_method,
1047 q->nb_channels, 8 * c,
1048 q->superblocktype_2_3, q->cm_table_select);
1051 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1055 * Process subpacket 12
1058 * @param node pointer to node with packet
1060 static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
1066 length = node->packet->size * 8;
1067 init_get_bits(&gb, node->packet->data, length);
1070 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1074 * Process new subpackets for synthesis filter
1077 * @param list list with synthesis filter packets (list D)
1079 static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
1081 QDM2SubPNode *nodes[4];
1083 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1085 process_subpacket_9(q, nodes[0]);
1087 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1089 process_subpacket_10(q, nodes[1]);
1091 process_subpacket_10(q, NULL);
1093 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1094 if (nodes[0] && nodes[1] && nodes[2])
1095 process_subpacket_11(q, nodes[2]);
1097 process_subpacket_11(q, NULL);
1099 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1100 if (nodes[0] && nodes[1] && nodes[3])
1101 process_subpacket_12(q, nodes[3]);
1103 process_subpacket_12(q, NULL);
1107 * Decode superblock, fill packet lists.
1111 static void qdm2_decode_super_block(QDM2Context *q)
1114 QDM2SubPacket header, *packet;
1115 int i, packet_bytes, sub_packet_size, sub_packets_D;
1116 unsigned int next_index = 0;
1118 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1119 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1120 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1122 q->sub_packets_B = 0;
1125 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1127 init_get_bits(&gb, q->compressed_data, q->compressed_size * 8);
1128 qdm2_decode_sub_packet_header(&gb, &header);
1130 if (header.type < 2 || header.type >= 8) {
1132 av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1136 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1137 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1139 init_get_bits(&gb, header.data, header.size * 8);
1141 if (header.type == 2 || header.type == 4 || header.type == 5) {
1142 int csum = 257 * get_bits(&gb, 8);
1143 csum += 2 * get_bits(&gb, 8);
1145 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1149 av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1154 q->sub_packet_list_B[0].packet = NULL;
1155 q->sub_packet_list_D[0].packet = NULL;
1157 for (i = 0; i < 6; i++)
1158 if (--q->fft_level_exp[i] < 0)
1159 q->fft_level_exp[i] = 0;
1161 for (i = 0; packet_bytes > 0; i++) {
1164 if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1165 SAMPLES_NEEDED_2("too many packet bytes");
1169 q->sub_packet_list_A[i].next = NULL;
1172 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1174 /* seek to next block */
1175 init_get_bits(&gb, header.data, header.size * 8);
1176 skip_bits(&gb, next_index * 8);
1178 if (next_index >= header.size)
1182 /* decode subpacket */
1183 packet = &q->sub_packets[i];
1184 qdm2_decode_sub_packet_header(&gb, packet);
1185 next_index = packet->size + get_bits_count(&gb) / 8;
1186 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1188 if (packet->type == 0)
1191 if (sub_packet_size > packet_bytes) {
1192 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1194 packet->size += packet_bytes - sub_packet_size;
1197 packet_bytes -= sub_packet_size;
1199 /* add subpacket to 'all subpackets' list */
1200 q->sub_packet_list_A[i].packet = packet;
1202 /* add subpacket to related list */
1203 if (packet->type == 8) {
1204 SAMPLES_NEEDED_2("packet type 8");
1206 } else if (packet->type >= 9 && packet->type <= 12) {
1207 /* packets for MPEG Audio like Synthesis Filter */
1208 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1209 } else if (packet->type == 13) {
1210 for (j = 0; j < 6; j++)
1211 q->fft_level_exp[j] = get_bits(&gb, 6);
1212 } else if (packet->type == 14) {
1213 for (j = 0; j < 6; j++)
1214 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1215 } else if (packet->type == 15) {
1216 SAMPLES_NEEDED_2("packet type 15")
1218 } else if (packet->type >= 16 && packet->type < 48 &&
1219 !fft_subpackets[packet->type - 16]) {
1220 /* packets for FFT */
1221 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1223 } // Packet bytes loop
1225 if (q->sub_packet_list_D[0].packet) {
1226 process_synthesis_subpackets(q, q->sub_packet_list_D);
1227 q->do_synth_filter = 1;
1228 } else if (q->do_synth_filter) {
1229 process_subpacket_10(q, NULL);
1230 process_subpacket_11(q, NULL);
1231 process_subpacket_12(q, NULL);
1235 static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1236 int offset, int duration, int channel,
1239 if (q->fft_coefs_min_index[duration] < 0)
1240 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1242 q->fft_coefs[q->fft_coefs_index].sub_packet =
1243 ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1244 q->fft_coefs[q->fft_coefs_index].channel = channel;
1245 q->fft_coefs[q->fft_coefs_index].offset = offset;
1246 q->fft_coefs[q->fft_coefs_index].exp = exp;
1247 q->fft_coefs[q->fft_coefs_index].phase = phase;
1248 q->fft_coefs_index++;
1251 static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
1252 GetBitContext *gb, int b)
1254 int channel, stereo, phase, exp;
1255 int local_int_4, local_int_8, stereo_phase, local_int_10;
1256 int local_int_14, stereo_exp, local_int_20, local_int_28;
1262 local_int_8 = (4 - duration);
1263 local_int_10 = 1 << (q->group_order - duration - 1);
1266 while (get_bits_left(gb)>0) {
1267 if (q->superblocktype_2_3) {
1268 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1269 if (get_bits_left(gb)<0) {
1270 if(local_int_4 < q->group_size)
1271 av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1276 local_int_4 += local_int_10;
1277 local_int_28 += (1 << local_int_8);
1279 local_int_4 += 8 * local_int_10;
1280 local_int_28 += (8 << local_int_8);
1285 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1286 while (offset >= (local_int_10 - 1)) {
1287 offset += (1 - (local_int_10 - 1));
1288 local_int_4 += local_int_10;
1289 local_int_28 += (1 << local_int_8);
1293 if (local_int_4 >= q->group_size)
1296 local_int_14 = (offset >> local_int_8);
1297 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1300 if (q->nb_channels > 1) {
1301 channel = get_bits1(gb);
1302 stereo = get_bits1(gb);
1308 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1309 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1310 exp = (exp < 0) ? 0 : exp;
1312 phase = get_bits(gb, 3);
1317 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1318 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1319 if (stereo_phase < 0)
1323 if (q->frequency_range > (local_int_14 + 1)) {
1324 int sub_packet = (local_int_20 + local_int_28);
1326 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1327 channel, exp, phase);
1329 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1331 stereo_exp, stereo_phase);
1337 static void qdm2_decode_fft_packets(QDM2Context *q)
1339 int i, j, min, max, value, type, unknown_flag;
1342 if (!q->sub_packet_list_B[0].packet)
1345 /* reset minimum indexes for FFT coefficients */
1346 q->fft_coefs_index = 0;
1347 for (i = 0; i < 5; i++)
1348 q->fft_coefs_min_index[i] = -1;
1350 /* process subpackets ordered by type, largest type first */
1351 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1352 QDM2SubPacket *packet = NULL;
1354 /* find subpacket with largest type less than max */
1355 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1356 value = q->sub_packet_list_B[j].packet->type;
1357 if (value > min && value < max) {
1359 packet = q->sub_packet_list_B[j].packet;
1365 /* check for errors (?) */
1370 (packet->type < 16 || packet->type >= 48 ||
1371 fft_subpackets[packet->type - 16]))
1374 /* decode FFT tones */
1375 init_get_bits(&gb, packet->data, packet->size * 8);
1377 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1382 type = packet->type;
1384 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1385 int duration = q->sub_sampling + 5 - (type & 15);
1387 if (duration >= 0 && duration < 4)
1388 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1389 } else if (type == 31) {
1390 for (j = 0; j < 4; j++)
1391 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1392 } else if (type == 46) {
1393 for (j = 0; j < 6; j++)
1394 q->fft_level_exp[j] = get_bits(&gb, 6);
1395 for (j = 0; j < 4; j++)
1396 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1398 } // Loop on B packets
1400 /* calculate maximum indexes for FFT coefficients */
1401 for (i = 0, j = -1; i < 5; i++)
1402 if (q->fft_coefs_min_index[i] >= 0) {
1404 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1408 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1411 static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
1416 const double iscale = 2.0 * M_PI / 512.0;
1418 tone->phase += tone->phase_shift;
1420 /* calculate current level (maximum amplitude) of tone */
1421 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1422 c.im = level * sin(tone->phase * iscale);
1423 c.re = level * cos(tone->phase * iscale);
1425 /* generate FFT coefficients for tone */
1426 if (tone->duration >= 3 || tone->cutoff >= 3) {
1427 tone->complex[0].im += c.im;
1428 tone->complex[0].re += c.re;
1429 tone->complex[1].im -= c.im;
1430 tone->complex[1].re -= c.re;
1432 f[1] = -tone->table[4];
1433 f[0] = tone->table[3] - tone->table[0];
1434 f[2] = 1.0 - tone->table[2] - tone->table[3];
1435 f[3] = tone->table[1] + tone->table[4] - 1.0;
1436 f[4] = tone->table[0] - tone->table[1];
1437 f[5] = tone->table[2];
1438 for (i = 0; i < 2; i++) {
1439 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1441 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1442 c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1444 for (i = 0; i < 4; i++) {
1445 tone->complex[i].re += c.re * f[i + 2];
1446 tone->complex[i].im += c.im * f[i + 2];
1450 /* copy the tone if it has not yet died out */
1451 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1452 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1453 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1457 static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1460 const double iscale = 0.25 * M_PI;
1462 for (ch = 0; ch < q->channels; ch++) {
1463 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1467 /* apply FFT tones with duration 4 (1 FFT period) */
1468 if (q->fft_coefs_min_index[4] >= 0)
1469 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1473 if (q->fft_coefs[i].sub_packet != sub_packet)
1476 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1477 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];
1479 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1480 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1481 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1482 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1483 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1484 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1487 /* generate existing FFT tones */
1488 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1489 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1490 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1493 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1494 for (i = 0; i < 4; i++)
1495 if (q->fft_coefs_min_index[i] >= 0) {
1496 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1500 if (q->fft_coefs[j].sub_packet != sub_packet)
1504 offset = q->fft_coefs[j].offset >> four_i;
1505 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1507 if (offset < q->frequency_range) {
1509 tone.cutoff = offset;
1511 tone.cutoff = (offset >= 60) ? 3 : 2;
1513 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];
1514 tone.complex = &q->fft.complex[ch][offset];
1515 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1516 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1517 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1519 tone.time_index = 0;
1521 qdm2_fft_generate_tone(q, &tone);
1524 q->fft_coefs_min_index[i] = j;
1528 static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1530 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1531 float *out = q->output_buffer + channel;
1533 q->fft.complex[channel][0].re *= 2.0f;
1534 q->fft.complex[channel][0].im = 0.0f;
1535 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1536 /* add samples to output buffer */
1537 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1538 out[0] += q->fft.complex[channel][i].re * gain;
1539 out[q->channels] += q->fft.complex[channel][i].im * gain;
1540 out += 2 * q->channels;
1546 * @param index subpacket number
1548 static void qdm2_synthesis_filter(QDM2Context *q, int index)
1550 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1552 /* copy sb_samples */
1553 sb_used = QDM2_SB_USED(q->sub_sampling);
1555 for (ch = 0; ch < q->channels; ch++)
1556 for (i = 0; i < 8; i++)
1557 for (k = sb_used; k < SBLIMIT; k++)
1558 q->sb_samples[ch][(8 * index) + i][k] = 0;
1560 for (ch = 0; ch < q->nb_channels; ch++) {
1561 float *samples_ptr = q->samples + ch;
1563 for (i = 0; i < 8; i++) {
1564 ff_mpa_synth_filter_float(&q->mpadsp,
1565 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1566 ff_mpa_synth_window_float, &dither_state,
1567 samples_ptr, q->nb_channels,
1568 q->sb_samples[ch][(8 * index) + i]);
1569 samples_ptr += 32 * q->nb_channels;
1573 /* add samples to output buffer */
1574 sub_sampling = (4 >> q->sub_sampling);
1576 for (ch = 0; ch < q->channels; ch++)
1577 for (i = 0; i < q->frame_size; i++)
1578 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1582 * Init static data (does not depend on specific file)
1586 static av_cold void qdm2_init_static_data(void) {
1593 ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1594 softclip_table_init();
1596 init_noise_samples();
1602 * Init parameters from codec extradata
1604 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1606 QDM2Context *s = avctx->priv_data;
1609 int tmp_val, tmp, size;
1611 qdm2_init_static_data();
1613 /* extradata parsing
1622 32 size (including this field)
1624 32 type (=QDM2 or QDMC)
1626 32 size (including this field, in bytes)
1627 32 tag (=QDCA) // maybe mandatory parameters
1630 32 samplerate (=44100)
1632 32 block size (=4096)
1633 32 frame size (=256) (for one channel)
1634 32 packet size (=1300)
1636 32 size (including this field, in bytes)
1637 32 tag (=QDCP) // maybe some tuneable parameters
1647 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1648 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1649 return AVERROR_INVALIDDATA;
1652 extradata = avctx->extradata;
1653 extradata_size = avctx->extradata_size;
1655 while (extradata_size > 7) {
1656 if (!memcmp(extradata, "frmaQDM", 7))
1662 if (extradata_size < 12) {
1663 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1665 return AVERROR_INVALIDDATA;
1668 if (memcmp(extradata, "frmaQDM", 7)) {
1669 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1670 return AVERROR_INVALIDDATA;
1673 if (extradata[7] == 'C') {
1675 avpriv_report_missing_feature(avctx, "QDMC version 1");
1676 return AVERROR_PATCHWELCOME;
1680 extradata_size -= 8;
1682 size = AV_RB32(extradata);
1684 if(size > extradata_size){
1685 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1686 extradata_size, size);
1687 return AVERROR_INVALIDDATA;
1691 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1692 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1693 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1694 return AVERROR_INVALIDDATA;
1699 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1701 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1702 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1703 return AVERROR_INVALIDDATA;
1705 avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1708 avctx->sample_rate = AV_RB32(extradata);
1711 avctx->bit_rate = AV_RB32(extradata);
1714 s->group_size = AV_RB32(extradata);
1717 s->fft_size = AV_RB32(extradata);
1720 s->checksum_size = AV_RB32(extradata);
1721 if (s->checksum_size >= 1U << 28) {
1722 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1723 return AVERROR_INVALIDDATA;
1726 s->fft_order = av_log2(s->fft_size) + 1;
1728 // something like max decodable tones
1729 s->group_order = av_log2(s->group_size) + 1;
1730 s->frame_size = s->group_size / 16; // 16 iterations per super block
1732 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1733 return AVERROR_INVALIDDATA;
1735 s->sub_sampling = s->fft_order - 7;
1736 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1738 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1739 case 0: tmp = 40; break;
1740 case 1: tmp = 48; break;
1741 case 2: tmp = 56; break;
1742 case 3: tmp = 72; break;
1743 case 4: tmp = 80; break;
1744 case 5: tmp = 100;break;
1745 default: tmp=s->sub_sampling; break;
1748 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1749 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1750 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1751 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1752 s->cm_table_select = tmp_val;
1754 if (avctx->bit_rate <= 8000)
1755 s->coeff_per_sb_select = 0;
1756 else if (avctx->bit_rate < 16000)
1757 s->coeff_per_sb_select = 1;
1759 s->coeff_per_sb_select = 2;
1761 // Fail on unknown fft order
1762 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1763 avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
1764 return AVERROR_PATCHWELCOME;
1766 if (s->fft_size != (1 << (s->fft_order - 1))) {
1767 av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1768 return AVERROR_INVALIDDATA;
1771 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1772 ff_mpadsp_init(&s->mpadsp);
1774 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1779 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1781 QDM2Context *s = avctx->priv_data;
1783 ff_rdft_end(&s->rdft_ctx);
1788 static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1791 const int frame_size = (q->frame_size * q->channels);
1793 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1796 /* select input buffer */
1797 q->compressed_data = in;
1798 q->compressed_size = q->checksum_size;
1800 /* copy old block, clear new block of output samples */
1801 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1802 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1804 /* decode block of QDM2 compressed data */
1805 if (q->sub_packet == 0) {
1806 q->has_errors = 0; // zero it for a new super block
1807 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1808 qdm2_decode_super_block(q);
1811 /* parse subpackets */
1812 if (!q->has_errors) {
1813 if (q->sub_packet == 2)
1814 qdm2_decode_fft_packets(q);
1816 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1819 /* sound synthesis stage 1 (FFT) */
1820 for (ch = 0; ch < q->channels; ch++) {
1821 qdm2_calculate_fft(q, ch, q->sub_packet);
1823 if (!q->has_errors && q->sub_packet_list_C[0].packet) {
1824 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1829 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1830 if (!q->has_errors && q->do_synth_filter)
1831 qdm2_synthesis_filter(q, q->sub_packet);
1833 q->sub_packet = (q->sub_packet + 1) % 16;
1835 /* clip and convert output float[] to 16bit signed samples */
1836 for (i = 0; i < frame_size; i++) {
1837 int value = (int)q->output_buffer[i];
1839 if (value > SOFTCLIP_THRESHOLD)
1840 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1841 else if (value < -SOFTCLIP_THRESHOLD)
1842 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1850 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1851 int *got_frame_ptr, AVPacket *avpkt)
1853 AVFrame *frame = data;
1854 const uint8_t *buf = avpkt->data;
1855 int buf_size = avpkt->size;
1856 QDM2Context *s = avctx->priv_data;
1862 if(buf_size < s->checksum_size)
1865 /* get output buffer */
1866 frame->nb_samples = 16 * s->frame_size;
1867 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1869 out = (int16_t *)frame->data[0];
1871 for (i = 0; i < 16; i++) {
1872 if ((ret = qdm2_decode(s, buf, out)) < 0)
1874 out += s->channels * s->frame_size;
1879 return s->checksum_size;
1882 AVCodec ff_qdm2_decoder = {
1884 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
1885 .type = AVMEDIA_TYPE_AUDIO,
1886 .id = AV_CODEC_ID_QDM2,
1887 .priv_data_size = sizeof(QDM2Context),
1888 .init = qdm2_decode_init,
1889 .close = qdm2_decode_close,
1890 .decode = qdm2_decode_frame,
1891 .capabilities = CODEC_CAP_DR1,