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 #include "libavutil/channel_layout.h"
40 #define BITSTREAM_READER_LE
44 #include "mpegaudio.h"
45 #include "mpegaudiodsp.h"
48 #include "qdm2_tablegen.h"
50 #define QDM2_LIST_ADD(list, size, packet) \
53 list[size - 1].next = &list[size]; \
55 list[size].packet = packet; \
56 list[size].next = NULL; \
60 // Result is 8, 16 or 30
61 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
63 #define FIX_NOISE_IDX(noise_idx) \
64 if ((noise_idx) >= 3840) \
65 (noise_idx) -= 3840; \
67 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
69 #define SAMPLES_NEEDED \
70 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
72 #define SAMPLES_NEEDED_2(why) \
73 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
75 #define QDM2_MAX_FRAME_SIZE 512
77 typedef int8_t sb_int8_array[2][30][64];
82 typedef struct QDM2SubPacket {
83 int type; ///< subpacket type
84 unsigned int size; ///< subpacket size
85 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
89 * A node in the subpacket list
91 typedef struct QDM2SubPNode {
92 QDM2SubPacket *packet; ///< packet
93 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
96 typedef struct QDM2Complex {
101 typedef struct FFTTone {
103 QDM2Complex *complex;
112 typedef struct FFTCoefficient {
120 typedef struct QDM2FFT {
121 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
125 * QDM2 decoder context
127 typedef struct QDM2Context {
128 /// Parameters from codec header, do not change during playback
129 int nb_channels; ///< number of channels
130 int channels; ///< number of channels
131 int group_size; ///< size of frame group (16 frames per group)
132 int fft_size; ///< size of FFT, in complex numbers
133 int checksum_size; ///< size of data block, used also for checksum
135 /// Parameters built from header parameters, do not change during playback
136 int group_order; ///< order of frame group
137 int fft_order; ///< order of FFT (actually fftorder+1)
138 int frame_size; ///< size of data frame
140 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
141 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
142 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
144 /// Packets and packet lists
145 QDM2SubPacket sub_packets[16]; ///< the packets themselves
146 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
147 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
148 int sub_packets_B; ///< number of packets on 'B' list
149 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
150 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
153 FFTTone fft_tones[1000];
156 FFTCoefficient fft_coefs[1000];
158 int fft_coefs_min_index[5];
159 int fft_coefs_max_index[5];
160 int fft_level_exp[6];
161 RDFTContext rdft_ctx;
165 const uint8_t *compressed_data;
167 float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
170 MPADSPContext mpadsp;
171 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
172 int synth_buf_offset[MPA_MAX_CHANNELS];
173 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
174 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
176 /// Mixed temporary data used in decoding
177 float tone_level[MPA_MAX_CHANNELS][30][64];
178 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
179 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
180 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
181 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
182 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
183 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
184 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
185 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
188 int has_errors; ///< packet has errors
189 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
190 int do_synth_filter; ///< used to perform or skip synthesis filter
193 int noise_idx; ///< index for dithering noise table
196 static const int switchtable[23] = {
197 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
200 static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
204 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
206 /* stage-2, 3 bits exponent escape sequence */
208 value = get_bits(gb, get_bits(gb, 3) + 1);
210 /* stage-3, optional */
215 av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
219 tmp= vlc_stage3_values[value];
221 if ((value & ~3) > 0)
222 tmp += get_bits(gb, (value >> 2));
229 static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
231 int value = qdm2_get_vlc(gb, vlc, 0, depth);
233 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
239 * @param data pointer to data to be checksummed
240 * @param length data length
241 * @param value checksum value
243 * @return 0 if checksum is OK
245 static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
249 for (i = 0; i < length; i++)
252 return (uint16_t)(value & 0xffff);
256 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
258 * @param gb bitreader context
259 * @param sub_packet packet under analysis
261 static void qdm2_decode_sub_packet_header(GetBitContext *gb,
262 QDM2SubPacket *sub_packet)
264 sub_packet->type = get_bits(gb, 8);
266 if (sub_packet->type == 0) {
267 sub_packet->size = 0;
268 sub_packet->data = NULL;
270 sub_packet->size = get_bits(gb, 8);
272 if (sub_packet->type & 0x80) {
273 sub_packet->size <<= 8;
274 sub_packet->size |= get_bits(gb, 8);
275 sub_packet->type &= 0x7f;
278 if (sub_packet->type == 0x7f)
279 sub_packet->type |= (get_bits(gb, 8) << 8);
281 // FIXME: this depends on bitreader-internal data
282 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
285 av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
286 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
290 * Return node pointer to first packet of requested type in list.
292 * @param list list of subpackets to be scanned
293 * @param type type of searched subpacket
294 * @return node pointer for subpacket if found, else NULL
296 static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
299 while (list && list->packet) {
300 if (list->packet->type == type)
308 * Replace 8 elements with their average value.
309 * Called by qdm2_decode_superblock before starting subblock decoding.
313 static void average_quantized_coeffs(QDM2Context *q)
315 int i, j, n, ch, sum;
317 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
319 for (ch = 0; ch < q->nb_channels; ch++)
320 for (i = 0; i < n; i++) {
323 for (j = 0; j < 8; j++)
324 sum += q->quantized_coeffs[ch][i][j];
330 for (j = 0; j < 8; j++)
331 q->quantized_coeffs[ch][i][j] = sum;
336 * Build subband samples with noise weighted by q->tone_level.
337 * Called by synthfilt_build_sb_samples.
340 * @param sb subband index
342 static void build_sb_samples_from_noise(QDM2Context *q, int sb)
346 FIX_NOISE_IDX(q->noise_idx);
351 for (ch = 0; ch < q->nb_channels; ch++) {
352 for (j = 0; j < 64; j++) {
353 q->sb_samples[ch][j * 2][sb] =
354 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
355 q->sb_samples[ch][j * 2 + 1][sb] =
356 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
362 * Called while processing data from subpackets 11 and 12.
363 * Used after making changes to coding_method array.
365 * @param sb subband index
366 * @param channels number of channels
367 * @param coding_method q->coding_method[0][0][0]
369 static int fix_coding_method_array(int sb, int channels,
370 sb_int8_array coding_method)
376 for (ch = 0; ch < channels; ch++) {
377 for (j = 0; j < 64; ) {
378 if (coding_method[ch][sb][j] < 8)
380 if ((coding_method[ch][sb][j] - 8) > 22) {
384 switch (switchtable[coding_method[ch][sb][j] - 8]) {
408 for (k = 0; k < run; k++) {
410 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) {
413 //not debugged, almost never used
414 memset(&coding_method[ch][sb][j + k], case_val,
416 memset(&coding_method[ch][sb][j + k], case_val,
429 * Related to synthesis filter
430 * Called by process_subpacket_10
433 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
435 static void fill_tone_level_array(QDM2Context *q, int flag)
437 int i, sb, ch, sb_used;
440 for (ch = 0; ch < q->nb_channels; ch++)
441 for (sb = 0; sb < 30; sb++)
442 for (i = 0; i < 8; i++) {
443 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
444 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
445 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
447 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
450 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
453 sb_used = QDM2_SB_USED(q->sub_sampling);
455 if ((q->superblocktype_2_3 != 0) && !flag) {
456 for (sb = 0; sb < sb_used; sb++)
457 for (ch = 0; ch < q->nb_channels; ch++)
458 for (i = 0; i < 64; i++) {
459 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
460 if (q->tone_level_idx[ch][sb][i] < 0)
461 q->tone_level[ch][sb][i] = 0;
463 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
466 tab = q->superblocktype_2_3 ? 0 : 1;
467 for (sb = 0; sb < sb_used; sb++) {
468 if ((sb >= 4) && (sb <= 23)) {
469 for (ch = 0; ch < q->nb_channels; ch++)
470 for (i = 0; i < 64; i++) {
471 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
472 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
473 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
474 q->tone_level_idx_hi2[ch][sb - 4];
475 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
476 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
477 q->tone_level[ch][sb][i] = 0;
479 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
483 for (ch = 0; ch < q->nb_channels; ch++)
484 for (i = 0; i < 64; i++) {
485 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
486 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
487 q->tone_level_idx_hi2[ch][sb - 4];
488 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
489 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
490 q->tone_level[ch][sb][i] = 0;
492 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
495 for (ch = 0; ch < q->nb_channels; ch++)
496 for (i = 0; i < 64; i++) {
497 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
498 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
499 q->tone_level[ch][sb][i] = 0;
501 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
510 * Related to synthesis filter
511 * Called by process_subpacket_11
512 * c is built with data from subpacket 11
513 * Most of this function is used only if superblock_type_2_3 == 0,
514 * never seen it in samples.
516 * @param tone_level_idx
517 * @param tone_level_idx_temp
518 * @param coding_method q->coding_method[0][0][0]
519 * @param nb_channels number of channels
520 * @param c coming from subpacket 11, passed as 8*c
521 * @param superblocktype_2_3 flag based on superblock packet type
522 * @param cm_table_select q->cm_table_select
524 static void fill_coding_method_array(sb_int8_array tone_level_idx,
525 sb_int8_array tone_level_idx_temp,
526 sb_int8_array coding_method,
528 int c, int superblocktype_2_3,
532 int tmp, acc, esp_40, comp;
533 int add1, add2, add3, add4;
536 if (!superblocktype_2_3) {
537 /* This case is untested, no samples available */
538 avpriv_request_sample(NULL, "!superblocktype_2_3");
540 for (ch = 0; ch < nb_channels; ch++) {
541 for (sb = 0; sb < 30; sb++) {
542 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
543 add1 = tone_level_idx[ch][sb][j] - 10;
546 add2 = add3 = add4 = 0;
548 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
553 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
558 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
562 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
565 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
567 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
571 for (ch = 0; ch < nb_channels; ch++)
572 for (sb = 0; sb < 30; sb++)
573 for (j = 0; j < 64; j++)
574 acc += tone_level_idx_temp[ch][sb][j];
576 multres = 0x66666667LL * (acc * 10);
577 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
578 for (ch = 0; ch < nb_channels; ch++)
579 for (sb = 0; sb < 30; sb++)
580 for (j = 0; j < 64; j++) {
581 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
584 comp /= 256; // signed shift
612 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
614 for (sb = 0; sb < 30; sb++)
615 fix_coding_method_array(sb, nb_channels, coding_method);
616 for (ch = 0; ch < nb_channels; ch++)
617 for (sb = 0; sb < 30; sb++)
618 for (j = 0; j < 64; j++)
620 if (coding_method[ch][sb][j] < 10)
621 coding_method[ch][sb][j] = 10;
624 if (coding_method[ch][sb][j] < 16)
625 coding_method[ch][sb][j] = 16;
627 if (coding_method[ch][sb][j] < 30)
628 coding_method[ch][sb][j] = 30;
631 } else { // superblocktype_2_3 != 0
632 for (ch = 0; ch < nb_channels; ch++)
633 for (sb = 0; sb < 30; sb++)
634 for (j = 0; j < 64; j++)
635 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
640 * Called by process_subpacket_11 to process more data from subpacket 11
642 * Called by process_subpacket_12 to process data from subpacket 12 with
646 * @param gb bitreader context
647 * @param length packet length in bits
648 * @param sb_min lower subband processed (sb_min included)
649 * @param sb_max higher subband processed (sb_max excluded)
651 static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
652 int length, int sb_min, int sb_max)
654 int sb, j, k, n, ch, run, channels;
655 int joined_stereo, zero_encoding;
657 float type34_div = 0;
658 float type34_predictor;
660 int sign_bits[16] = {0};
663 // If no data use noise
664 for (sb=sb_min; sb < sb_max; sb++)
665 build_sb_samples_from_noise(q, sb);
670 for (sb = sb_min; sb < sb_max; sb++) {
671 channels = q->nb_channels;
673 if (q->nb_channels <= 1 || sb < 12)
678 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
681 if (get_bits_left(gb) >= 16)
682 for (j = 0; j < 16; j++)
683 sign_bits[j] = get_bits1(gb);
685 for (j = 0; j < 64; j++)
686 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
687 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
689 if (fix_coding_method_array(sb, q->nb_channels,
691 av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
692 build_sb_samples_from_noise(q, sb);
698 for (ch = 0; ch < channels; ch++) {
699 FIX_NOISE_IDX(q->noise_idx);
700 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
701 type34_predictor = 0.0;
704 for (j = 0; j < 128; ) {
705 switch (q->coding_method[ch][sb][j / 2]) {
707 if (get_bits_left(gb) >= 10) {
709 for (k = 0; k < 5; k++) {
710 if ((j + 2 * k) >= 128)
712 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
717 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
718 return AVERROR_INVALIDDATA;
721 for (k = 0; k < 5; k++)
722 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
724 for (k = 0; k < 5; k++)
725 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
727 for (k = 0; k < 10; k++)
728 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
734 if (get_bits_left(gb) >= 1) {
739 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
742 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
748 if (get_bits_left(gb) >= 10) {
750 for (k = 0; k < 5; k++) {
753 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
756 n = get_bits (gb, 8);
758 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
759 return AVERROR_INVALIDDATA;
762 for (k = 0; k < 5; k++)
763 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
766 for (k = 0; k < 5; k++)
767 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
773 if (get_bits_left(gb) >= 7) {
776 av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
777 return AVERROR_INVALIDDATA;
780 for (k = 0; k < 3; k++)
781 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
783 for (k = 0; k < 3; k++)
784 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
790 if (get_bits_left(gb) >= 4) {
791 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
792 if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
793 av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
794 return AVERROR_INVALIDDATA;
796 samples[0] = type30_dequant[index];
798 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
804 if (get_bits_left(gb) >= 7) {
806 type34_div = (float)(1 << get_bits(gb, 2));
807 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
808 type34_predictor = samples[0];
811 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
812 if (index >= FF_ARRAY_ELEMS(type34_delta)) {
813 av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
814 return AVERROR_INVALIDDATA;
816 samples[0] = type34_delta[index] / type34_div + type34_predictor;
817 type34_predictor = samples[0];
820 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
826 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
832 for (k = 0; k < run && j + k < 128; k++) {
833 q->sb_samples[0][j + k][sb] =
834 q->tone_level[0][sb][(j + k) / 2] * samples[k];
835 if (q->nb_channels == 2) {
836 if (sign_bits[(j + k) / 8])
837 q->sb_samples[1][j + k][sb] =
838 q->tone_level[1][sb][(j + k) / 2] * -samples[k];
840 q->sb_samples[1][j + k][sb] =
841 q->tone_level[1][sb][(j + k) / 2] * samples[k];
845 for (k = 0; k < run; k++)
847 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
858 * Init the first element of a channel in quantized_coeffs with data
859 * from packet 10 (quantized_coeffs[ch][0]).
860 * This is similar to process_subpacket_9, but for a single channel
861 * and for element [0]
862 * same VLC tables as process_subpacket_9 are used.
864 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
865 * @param gb bitreader context
867 static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
870 int i, k, run, level, diff;
872 if (get_bits_left(gb) < 16)
874 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
876 quantized_coeffs[0] = level;
878 for (i = 0; i < 7; ) {
879 if (get_bits_left(gb) < 16)
881 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
886 if (get_bits_left(gb) < 16)
888 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
890 for (k = 1; k <= run; k++)
891 quantized_coeffs[i + k] = (level + ((k * diff) / run));
900 * Related to synthesis filter, process data from packet 10
901 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
902 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
903 * data from packet 10
906 * @param gb bitreader context
908 static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
912 for (ch = 0; ch < q->nb_channels; ch++) {
913 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
915 if (get_bits_left(gb) < 16) {
916 memset(q->quantized_coeffs[ch][0], 0, 8);
921 n = q->sub_sampling + 1;
923 for (sb = 0; sb < n; sb++)
924 for (ch = 0; ch < q->nb_channels; ch++)
925 for (j = 0; j < 8; j++) {
926 if (get_bits_left(gb) < 1)
929 for (k=0; k < 8; k++) {
930 if (get_bits_left(gb) < 16)
932 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
935 for (k=0; k < 8; k++)
936 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
940 n = QDM2_SB_USED(q->sub_sampling) - 4;
942 for (sb = 0; sb < n; sb++)
943 for (ch = 0; ch < q->nb_channels; ch++) {
944 if (get_bits_left(gb) < 16)
946 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
948 q->tone_level_idx_hi2[ch][sb] -= 16;
950 for (j = 0; j < 8; j++)
951 q->tone_level_idx_mid[ch][sb][j] = -16;
954 n = QDM2_SB_USED(q->sub_sampling) - 5;
956 for (sb = 0; sb < n; sb++)
957 for (ch = 0; ch < q->nb_channels; ch++)
958 for (j = 0; j < 8; j++) {
959 if (get_bits_left(gb) < 16)
961 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
966 * Process subpacket 9, init quantized_coeffs with data from it
969 * @param node pointer to node with packet
971 static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
974 int i, j, k, n, ch, run, level, diff;
976 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
978 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
980 for (i = 1; i < n; i++)
981 for (ch = 0; ch < q->nb_channels; ch++) {
982 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
983 q->quantized_coeffs[ch][i][0] = level;
985 for (j = 0; j < (8 - 1); ) {
986 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
987 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
992 for (k = 1; k <= run; k++)
993 q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
1000 for (ch = 0; ch < q->nb_channels; ch++)
1001 for (i = 0; i < 8; i++)
1002 q->quantized_coeffs[ch][0][i] = 0;
1008 * Process subpacket 10 if not null, else
1011 * @param node pointer to node with packet
1013 static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
1018 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1019 init_tone_level_dequantization(q, &gb);
1020 fill_tone_level_array(q, 1);
1022 fill_tone_level_array(q, 0);
1027 * Process subpacket 11
1030 * @param node pointer to node with packet
1032 static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
1038 length = node->packet->size * 8;
1039 init_get_bits(&gb, node->packet->data, length);
1043 int c = get_bits(&gb, 13);
1046 fill_coding_method_array(q->tone_level_idx,
1047 q->tone_level_idx_temp, q->coding_method,
1048 q->nb_channels, 8 * c,
1049 q->superblocktype_2_3, q->cm_table_select);
1052 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1056 * Process subpacket 12
1059 * @param node pointer to node with packet
1061 static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
1067 length = node->packet->size * 8;
1068 init_get_bits(&gb, node->packet->data, length);
1071 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1075 * Process new subpackets for synthesis filter
1078 * @param list list with synthesis filter packets (list D)
1080 static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
1082 QDM2SubPNode *nodes[4];
1084 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1086 process_subpacket_9(q, nodes[0]);
1088 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1090 process_subpacket_10(q, nodes[1]);
1092 process_subpacket_10(q, NULL);
1094 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1095 if (nodes[0] && nodes[1] && nodes[2])
1096 process_subpacket_11(q, nodes[2]);
1098 process_subpacket_11(q, NULL);
1100 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1101 if (nodes[0] && nodes[1] && nodes[3])
1102 process_subpacket_12(q, nodes[3]);
1104 process_subpacket_12(q, NULL);
1108 * Decode superblock, fill packet lists.
1112 static void qdm2_decode_super_block(QDM2Context *q)
1115 QDM2SubPacket header, *packet;
1116 int i, packet_bytes, sub_packet_size, sub_packets_D;
1117 unsigned int next_index = 0;
1119 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1120 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1121 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1123 q->sub_packets_B = 0;
1126 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1128 init_get_bits(&gb, q->compressed_data, q->compressed_size * 8);
1129 qdm2_decode_sub_packet_header(&gb, &header);
1131 if (header.type < 2 || header.type >= 8) {
1133 av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1137 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1138 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1140 init_get_bits(&gb, header.data, header.size * 8);
1142 if (header.type == 2 || header.type == 4 || header.type == 5) {
1143 int csum = 257 * get_bits(&gb, 8);
1144 csum += 2 * get_bits(&gb, 8);
1146 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1150 av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1155 q->sub_packet_list_B[0].packet = NULL;
1156 q->sub_packet_list_D[0].packet = NULL;
1158 for (i = 0; i < 6; i++)
1159 if (--q->fft_level_exp[i] < 0)
1160 q->fft_level_exp[i] = 0;
1162 for (i = 0; packet_bytes > 0; i++) {
1165 if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1166 SAMPLES_NEEDED_2("too many packet bytes");
1170 q->sub_packet_list_A[i].next = NULL;
1173 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1175 /* seek to next block */
1176 init_get_bits(&gb, header.data, header.size * 8);
1177 skip_bits(&gb, next_index * 8);
1179 if (next_index >= header.size)
1183 /* decode subpacket */
1184 packet = &q->sub_packets[i];
1185 qdm2_decode_sub_packet_header(&gb, packet);
1186 next_index = packet->size + get_bits_count(&gb) / 8;
1187 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1189 if (packet->type == 0)
1192 if (sub_packet_size > packet_bytes) {
1193 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1195 packet->size += packet_bytes - sub_packet_size;
1198 packet_bytes -= sub_packet_size;
1200 /* add subpacket to 'all subpackets' list */
1201 q->sub_packet_list_A[i].packet = packet;
1203 /* add subpacket to related list */
1204 if (packet->type == 8) {
1205 SAMPLES_NEEDED_2("packet type 8");
1207 } else if (packet->type >= 9 && packet->type <= 12) {
1208 /* packets for MPEG Audio like Synthesis Filter */
1209 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1210 } else if (packet->type == 13) {
1211 for (j = 0; j < 6; j++)
1212 q->fft_level_exp[j] = get_bits(&gb, 6);
1213 } else if (packet->type == 14) {
1214 for (j = 0; j < 6; j++)
1215 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1216 } else if (packet->type == 15) {
1217 SAMPLES_NEEDED_2("packet type 15")
1219 } else if (packet->type >= 16 && packet->type < 48 &&
1220 !fft_subpackets[packet->type - 16]) {
1221 /* packets for FFT */
1222 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1224 } // Packet bytes loop
1226 if (q->sub_packet_list_D[0].packet) {
1227 process_synthesis_subpackets(q, q->sub_packet_list_D);
1228 q->do_synth_filter = 1;
1229 } else if (q->do_synth_filter) {
1230 process_subpacket_10(q, NULL);
1231 process_subpacket_11(q, NULL);
1232 process_subpacket_12(q, NULL);
1236 static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1237 int offset, int duration, int channel,
1240 if (q->fft_coefs_min_index[duration] < 0)
1241 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1243 q->fft_coefs[q->fft_coefs_index].sub_packet =
1244 ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1245 q->fft_coefs[q->fft_coefs_index].channel = channel;
1246 q->fft_coefs[q->fft_coefs_index].offset = offset;
1247 q->fft_coefs[q->fft_coefs_index].exp = exp;
1248 q->fft_coefs[q->fft_coefs_index].phase = phase;
1249 q->fft_coefs_index++;
1252 static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
1253 GetBitContext *gb, int b)
1255 int channel, stereo, phase, exp;
1256 int local_int_4, local_int_8, stereo_phase, local_int_10;
1257 int local_int_14, stereo_exp, local_int_20, local_int_28;
1263 local_int_8 = (4 - duration);
1264 local_int_10 = 1 << (q->group_order - duration - 1);
1267 while (get_bits_left(gb)>0) {
1268 if (q->superblocktype_2_3) {
1269 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1270 if (get_bits_left(gb)<0) {
1271 if(local_int_4 < q->group_size)
1272 av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1277 local_int_4 += local_int_10;
1278 local_int_28 += (1 << local_int_8);
1280 local_int_4 += 8 * local_int_10;
1281 local_int_28 += (8 << local_int_8);
1286 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1287 while (offset >= (local_int_10 - 1)) {
1288 offset += (1 - (local_int_10 - 1));
1289 local_int_4 += local_int_10;
1290 local_int_28 += (1 << local_int_8);
1294 if (local_int_4 >= q->group_size)
1297 local_int_14 = (offset >> local_int_8);
1298 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1301 if (q->nb_channels > 1) {
1302 channel = get_bits1(gb);
1303 stereo = get_bits1(gb);
1309 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1310 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1311 exp = (exp < 0) ? 0 : exp;
1313 phase = get_bits(gb, 3);
1318 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1319 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1320 if (stereo_phase < 0)
1324 if (q->frequency_range > (local_int_14 + 1)) {
1325 int sub_packet = (local_int_20 + local_int_28);
1327 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1328 channel, exp, phase);
1330 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1332 stereo_exp, stereo_phase);
1338 static void qdm2_decode_fft_packets(QDM2Context *q)
1340 int i, j, min, max, value, type, unknown_flag;
1343 if (!q->sub_packet_list_B[0].packet)
1346 /* reset minimum indexes for FFT coefficients */
1347 q->fft_coefs_index = 0;
1348 for (i = 0; i < 5; i++)
1349 q->fft_coefs_min_index[i] = -1;
1351 /* process subpackets ordered by type, largest type first */
1352 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1353 QDM2SubPacket *packet = NULL;
1355 /* find subpacket with largest type less than max */
1356 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1357 value = q->sub_packet_list_B[j].packet->type;
1358 if (value > min && value < max) {
1360 packet = q->sub_packet_list_B[j].packet;
1366 /* check for errors (?) */
1371 (packet->type < 16 || packet->type >= 48 ||
1372 fft_subpackets[packet->type - 16]))
1375 /* decode FFT tones */
1376 init_get_bits(&gb, packet->data, packet->size * 8);
1378 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1383 type = packet->type;
1385 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1386 int duration = q->sub_sampling + 5 - (type & 15);
1388 if (duration >= 0 && duration < 4)
1389 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1390 } else if (type == 31) {
1391 for (j = 0; j < 4; j++)
1392 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1393 } else if (type == 46) {
1394 for (j = 0; j < 6; j++)
1395 q->fft_level_exp[j] = get_bits(&gb, 6);
1396 for (j = 0; j < 4; j++)
1397 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1399 } // Loop on B packets
1401 /* calculate maximum indexes for FFT coefficients */
1402 for (i = 0, j = -1; i < 5; i++)
1403 if (q->fft_coefs_min_index[i] >= 0) {
1405 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1409 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1412 static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
1417 const double iscale = 2.0 * M_PI / 512.0;
1419 tone->phase += tone->phase_shift;
1421 /* calculate current level (maximum amplitude) of tone */
1422 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1423 c.im = level * sin(tone->phase * iscale);
1424 c.re = level * cos(tone->phase * iscale);
1426 /* generate FFT coefficients for tone */
1427 if (tone->duration >= 3 || tone->cutoff >= 3) {
1428 tone->complex[0].im += c.im;
1429 tone->complex[0].re += c.re;
1430 tone->complex[1].im -= c.im;
1431 tone->complex[1].re -= c.re;
1433 f[1] = -tone->table[4];
1434 f[0] = tone->table[3] - tone->table[0];
1435 f[2] = 1.0 - tone->table[2] - tone->table[3];
1436 f[3] = tone->table[1] + tone->table[4] - 1.0;
1437 f[4] = tone->table[0] - tone->table[1];
1438 f[5] = tone->table[2];
1439 for (i = 0; i < 2; i++) {
1440 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1442 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1443 c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1445 for (i = 0; i < 4; i++) {
1446 tone->complex[i].re += c.re * f[i + 2];
1447 tone->complex[i].im += c.im * f[i + 2];
1451 /* copy the tone if it has not yet died out */
1452 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1453 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1454 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1458 static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1461 const double iscale = 0.25 * M_PI;
1463 for (ch = 0; ch < q->channels; ch++) {
1464 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1468 /* apply FFT tones with duration 4 (1 FFT period) */
1469 if (q->fft_coefs_min_index[4] >= 0)
1470 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1474 if (q->fft_coefs[i].sub_packet != sub_packet)
1477 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1478 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];
1480 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1481 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1482 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1483 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1484 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1485 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1488 /* generate existing FFT tones */
1489 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1490 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1491 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1494 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1495 for (i = 0; i < 4; i++)
1496 if (q->fft_coefs_min_index[i] >= 0) {
1497 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1501 if (q->fft_coefs[j].sub_packet != sub_packet)
1505 offset = q->fft_coefs[j].offset >> four_i;
1506 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1508 if (offset < q->frequency_range) {
1510 tone.cutoff = offset;
1512 tone.cutoff = (offset >= 60) ? 3 : 2;
1514 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];
1515 tone.complex = &q->fft.complex[ch][offset];
1516 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1517 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1518 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1520 tone.time_index = 0;
1522 qdm2_fft_generate_tone(q, &tone);
1525 q->fft_coefs_min_index[i] = j;
1529 static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1531 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1532 float *out = q->output_buffer + channel;
1534 q->fft.complex[channel][0].re *= 2.0f;
1535 q->fft.complex[channel][0].im = 0.0f;
1536 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1537 /* add samples to output buffer */
1538 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1539 out[0] += q->fft.complex[channel][i].re * gain;
1540 out[q->channels] += q->fft.complex[channel][i].im * gain;
1541 out += 2 * q->channels;
1547 * @param index subpacket number
1549 static void qdm2_synthesis_filter(QDM2Context *q, int index)
1551 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1553 /* copy sb_samples */
1554 sb_used = QDM2_SB_USED(q->sub_sampling);
1556 for (ch = 0; ch < q->channels; ch++)
1557 for (i = 0; i < 8; i++)
1558 for (k = sb_used; k < SBLIMIT; k++)
1559 q->sb_samples[ch][(8 * index) + i][k] = 0;
1561 for (ch = 0; ch < q->nb_channels; ch++) {
1562 float *samples_ptr = q->samples + ch;
1564 for (i = 0; i < 8; i++) {
1565 ff_mpa_synth_filter_float(&q->mpadsp,
1566 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1567 ff_mpa_synth_window_float, &dither_state,
1568 samples_ptr, q->nb_channels,
1569 q->sb_samples[ch][(8 * index) + i]);
1570 samples_ptr += 32 * q->nb_channels;
1574 /* add samples to output buffer */
1575 sub_sampling = (4 >> q->sub_sampling);
1577 for (ch = 0; ch < q->channels; ch++)
1578 for (i = 0; i < q->frame_size; i++)
1579 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1583 * Init static data (does not depend on specific file)
1587 static av_cold void qdm2_init_static_data(void) {
1594 ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1595 softclip_table_init();
1597 init_noise_samples();
1603 * Init parameters from codec extradata
1605 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1607 QDM2Context *s = avctx->priv_data;
1610 int tmp_val, tmp, size;
1612 qdm2_init_static_data();
1614 /* extradata parsing
1623 32 size (including this field)
1625 32 type (=QDM2 or QDMC)
1627 32 size (including this field, in bytes)
1628 32 tag (=QDCA) // maybe mandatory parameters
1631 32 samplerate (=44100)
1633 32 block size (=4096)
1634 32 frame size (=256) (for one channel)
1635 32 packet size (=1300)
1637 32 size (including this field, in bytes)
1638 32 tag (=QDCP) // maybe some tuneable parameters
1648 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1649 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1650 return AVERROR_INVALIDDATA;
1653 extradata = avctx->extradata;
1654 extradata_size = avctx->extradata_size;
1656 while (extradata_size > 7) {
1657 if (!memcmp(extradata, "frmaQDM", 7))
1663 if (extradata_size < 12) {
1664 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1666 return AVERROR_INVALIDDATA;
1669 if (memcmp(extradata, "frmaQDM", 7)) {
1670 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1671 return AVERROR_INVALIDDATA;
1674 if (extradata[7] == 'C') {
1676 avpriv_report_missing_feature(avctx, "QDMC version 1");
1677 return AVERROR_PATCHWELCOME;
1681 extradata_size -= 8;
1683 size = AV_RB32(extradata);
1685 if(size > extradata_size){
1686 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1687 extradata_size, size);
1688 return AVERROR_INVALIDDATA;
1692 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1693 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1694 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1695 return AVERROR_INVALIDDATA;
1700 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1702 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1703 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1704 return AVERROR_INVALIDDATA;
1706 avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1709 avctx->sample_rate = AV_RB32(extradata);
1712 avctx->bit_rate = AV_RB32(extradata);
1715 s->group_size = AV_RB32(extradata);
1718 s->fft_size = AV_RB32(extradata);
1721 s->checksum_size = AV_RB32(extradata);
1722 if (s->checksum_size >= 1U << 28) {
1723 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1724 return AVERROR_INVALIDDATA;
1727 s->fft_order = av_log2(s->fft_size) + 1;
1729 // something like max decodable tones
1730 s->group_order = av_log2(s->group_size) + 1;
1731 s->frame_size = s->group_size / 16; // 16 iterations per super block
1733 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1734 return AVERROR_INVALIDDATA;
1736 s->sub_sampling = s->fft_order - 7;
1737 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1739 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1740 case 0: tmp = 40; break;
1741 case 1: tmp = 48; break;
1742 case 2: tmp = 56; break;
1743 case 3: tmp = 72; break;
1744 case 4: tmp = 80; break;
1745 case 5: tmp = 100;break;
1746 default: tmp=s->sub_sampling; break;
1749 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1750 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1751 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1752 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1753 s->cm_table_select = tmp_val;
1755 if (avctx->bit_rate <= 8000)
1756 s->coeff_per_sb_select = 0;
1757 else if (avctx->bit_rate < 16000)
1758 s->coeff_per_sb_select = 1;
1760 s->coeff_per_sb_select = 2;
1762 // Fail on unknown fft order
1763 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1764 avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
1765 return AVERROR_PATCHWELCOME;
1767 if (s->fft_size != (1 << (s->fft_order - 1))) {
1768 av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1769 return AVERROR_INVALIDDATA;
1772 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1773 ff_mpadsp_init(&s->mpadsp);
1775 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1780 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1782 QDM2Context *s = avctx->priv_data;
1784 ff_rdft_end(&s->rdft_ctx);
1789 static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1792 const int frame_size = (q->frame_size * q->channels);
1794 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1797 /* select input buffer */
1798 q->compressed_data = in;
1799 q->compressed_size = q->checksum_size;
1801 /* copy old block, clear new block of output samples */
1802 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1803 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1805 /* decode block of QDM2 compressed data */
1806 if (q->sub_packet == 0) {
1807 q->has_errors = 0; // zero it for a new super block
1808 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1809 qdm2_decode_super_block(q);
1812 /* parse subpackets */
1813 if (!q->has_errors) {
1814 if (q->sub_packet == 2)
1815 qdm2_decode_fft_packets(q);
1817 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1820 /* sound synthesis stage 1 (FFT) */
1821 for (ch = 0; ch < q->channels; ch++) {
1822 qdm2_calculate_fft(q, ch, q->sub_packet);
1824 if (!q->has_errors && q->sub_packet_list_C[0].packet) {
1825 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1830 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1831 if (!q->has_errors && q->do_synth_filter)
1832 qdm2_synthesis_filter(q, q->sub_packet);
1834 q->sub_packet = (q->sub_packet + 1) % 16;
1836 /* clip and convert output float[] to 16-bit signed samples */
1837 for (i = 0; i < frame_size; i++) {
1838 int value = (int)q->output_buffer[i];
1840 if (value > SOFTCLIP_THRESHOLD)
1841 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1842 else if (value < -SOFTCLIP_THRESHOLD)
1843 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1851 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1852 int *got_frame_ptr, AVPacket *avpkt)
1854 AVFrame *frame = data;
1855 const uint8_t *buf = avpkt->data;
1856 int buf_size = avpkt->size;
1857 QDM2Context *s = avctx->priv_data;
1863 if(buf_size < s->checksum_size)
1866 /* get output buffer */
1867 frame->nb_samples = 16 * s->frame_size;
1868 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1870 out = (int16_t *)frame->data[0];
1872 for (i = 0; i < 16; i++) {
1873 if ((ret = qdm2_decode(s, buf, out)) < 0)
1875 out += s->channels * s->frame_size;
1880 return s->checksum_size;
1883 AVCodec ff_qdm2_decoder = {
1885 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
1886 .type = AVMEDIA_TYPE_AUDIO,
1887 .id = AV_CODEC_ID_QDM2,
1888 .priv_data_size = sizeof(QDM2Context),
1889 .init = qdm2_decode_init,
1890 .close = qdm2_decode_close,
1891 .decode = qdm2_decode_frame,
1892 .capabilities = AV_CODEC_CAP_DR1,