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
43 #include "bytestream.h"
45 #include "mpegaudio.h"
46 #include "mpegaudiodsp.h"
49 #include "qdm2_tablegen.h"
51 #define QDM2_LIST_ADD(list, size, packet) \
54 list[size - 1].next = &list[size]; \
56 list[size].packet = packet; \
57 list[size].next = NULL; \
61 // Result is 8, 16 or 30
62 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
64 #define FIX_NOISE_IDX(noise_idx) \
65 if ((noise_idx) >= 3840) \
66 (noise_idx) -= 3840; \
68 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
70 #define SAMPLES_NEEDED \
71 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
73 #define SAMPLES_NEEDED_2(why) \
74 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
76 #define QDM2_MAX_FRAME_SIZE 512
78 typedef int8_t sb_int8_array[2][30][64];
83 typedef struct QDM2SubPacket {
84 int type; ///< subpacket type
85 unsigned int size; ///< subpacket size
86 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
90 * A node in the subpacket list
92 typedef struct QDM2SubPNode {
93 QDM2SubPacket *packet; ///< packet
94 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
97 typedef struct QDM2Complex {
102 typedef struct FFTTone {
104 QDM2Complex *complex;
113 typedef struct FFTCoefficient {
121 typedef struct QDM2FFT {
122 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
126 * QDM2 decoder context
128 typedef struct QDM2Context {
129 /// Parameters from codec header, do not change during playback
130 int nb_channels; ///< number of channels
131 int channels; ///< number of channels
132 int group_size; ///< size of frame group (16 frames per group)
133 int fft_size; ///< size of FFT, in complex numbers
134 int checksum_size; ///< size of data block, used also for checksum
136 /// Parameters built from header parameters, do not change during playback
137 int group_order; ///< order of frame group
138 int fft_order; ///< order of FFT (actually fftorder+1)
139 int frame_size; ///< size of data frame
141 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
142 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
143 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
145 /// Packets and packet lists
146 QDM2SubPacket sub_packets[16]; ///< the packets themselves
147 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
148 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
149 int sub_packets_B; ///< number of packets on 'B' list
150 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
151 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
154 FFTTone fft_tones[1000];
157 FFTCoefficient fft_coefs[1000];
159 int fft_coefs_min_index[5];
160 int fft_coefs_max_index[5];
161 int fft_level_exp[6];
162 RDFTContext rdft_ctx;
166 const uint8_t *compressed_data;
168 float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
171 MPADSPContext mpadsp;
172 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
173 int synth_buf_offset[MPA_MAX_CHANNELS];
174 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
175 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
177 /// Mixed temporary data used in decoding
178 float tone_level[MPA_MAX_CHANNELS][30][64];
179 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
180 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
181 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
182 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
183 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
184 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
185 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
186 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
189 int has_errors; ///< packet has errors
190 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
191 int do_synth_filter; ///< used to perform or skip synthesis filter
194 int noise_idx; ///< index for dithering noise table
197 static const int switchtable[23] = {
198 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
201 static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
205 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
207 /* stage-2, 3 bits exponent escape sequence */
209 value = get_bits(gb, get_bits(gb, 3) + 1);
211 /* stage-3, optional */
216 av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
220 tmp= vlc_stage3_values[value];
222 if ((value & ~3) > 0)
223 tmp += get_bits(gb, (value >> 2));
230 static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
232 int value = qdm2_get_vlc(gb, vlc, 0, depth);
234 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
240 * @param data pointer to data to be checksummed
241 * @param length data length
242 * @param value checksum value
244 * @return 0 if checksum is OK
246 static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
250 for (i = 0; i < length; i++)
253 return (uint16_t)(value & 0xffff);
257 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
259 * @param gb bitreader context
260 * @param sub_packet packet under analysis
262 static void qdm2_decode_sub_packet_header(GetBitContext *gb,
263 QDM2SubPacket *sub_packet)
265 sub_packet->type = get_bits(gb, 8);
267 if (sub_packet->type == 0) {
268 sub_packet->size = 0;
269 sub_packet->data = NULL;
271 sub_packet->size = get_bits(gb, 8);
273 if (sub_packet->type & 0x80) {
274 sub_packet->size <<= 8;
275 sub_packet->size |= get_bits(gb, 8);
276 sub_packet->type &= 0x7f;
279 if (sub_packet->type == 0x7f)
280 sub_packet->type |= (get_bits(gb, 8) << 8);
282 // FIXME: this depends on bitreader-internal data
283 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
286 av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
287 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
291 * Return node pointer to first packet of requested type in list.
293 * @param list list of subpackets to be scanned
294 * @param type type of searched subpacket
295 * @return node pointer for subpacket if found, else NULL
297 static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
300 while (list && list->packet) {
301 if (list->packet->type == type)
309 * Replace 8 elements with their average value.
310 * Called by qdm2_decode_superblock before starting subblock decoding.
314 static void average_quantized_coeffs(QDM2Context *q)
316 int i, j, n, ch, sum;
318 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
320 for (ch = 0; ch < q->nb_channels; ch++)
321 for (i = 0; i < n; i++) {
324 for (j = 0; j < 8; j++)
325 sum += q->quantized_coeffs[ch][i][j];
331 for (j = 0; j < 8; j++)
332 q->quantized_coeffs[ch][i][j] = sum;
337 * Build subband samples with noise weighted by q->tone_level.
338 * Called by synthfilt_build_sb_samples.
341 * @param sb subband index
343 static void build_sb_samples_from_noise(QDM2Context *q, int sb)
347 FIX_NOISE_IDX(q->noise_idx);
352 for (ch = 0; ch < q->nb_channels; ch++) {
353 for (j = 0; j < 64; j++) {
354 q->sb_samples[ch][j * 2][sb] =
355 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
356 q->sb_samples[ch][j * 2 + 1][sb] =
357 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
363 * Called while processing data from subpackets 11 and 12.
364 * Used after making changes to coding_method array.
366 * @param sb subband index
367 * @param channels number of channels
368 * @param coding_method q->coding_method[0][0][0]
370 static int fix_coding_method_array(int sb, int channels,
371 sb_int8_array coding_method)
377 for (ch = 0; ch < channels; ch++) {
378 for (j = 0; j < 64; ) {
379 if (coding_method[ch][sb][j] < 8)
381 if ((coding_method[ch][sb][j] - 8) > 22) {
385 switch (switchtable[coding_method[ch][sb][j] - 8]) {
409 for (k = 0; k < run; k++) {
411 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) {
414 //not debugged, almost never used
415 memset(&coding_method[ch][sb][j + k], case_val,
417 memset(&coding_method[ch][sb][j + k], case_val,
430 * Related to synthesis filter
431 * Called by process_subpacket_10
434 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
436 static void fill_tone_level_array(QDM2Context *q, int flag)
438 int i, sb, ch, sb_used;
441 for (ch = 0; ch < q->nb_channels; ch++)
442 for (sb = 0; sb < 30; sb++)
443 for (i = 0; i < 8; i++) {
444 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
445 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
446 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
448 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
451 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
454 sb_used = QDM2_SB_USED(q->sub_sampling);
456 if ((q->superblocktype_2_3 != 0) && !flag) {
457 for (sb = 0; sb < sb_used; sb++)
458 for (ch = 0; ch < q->nb_channels; ch++)
459 for (i = 0; i < 64; i++) {
460 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
461 if (q->tone_level_idx[ch][sb][i] < 0)
462 q->tone_level[ch][sb][i] = 0;
464 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
467 tab = q->superblocktype_2_3 ? 0 : 1;
468 for (sb = 0; sb < sb_used; sb++) {
469 if ((sb >= 4) && (sb <= 23)) {
470 for (ch = 0; ch < q->nb_channels; ch++)
471 for (i = 0; i < 64; i++) {
472 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
473 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
474 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
475 q->tone_level_idx_hi2[ch][sb - 4];
476 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
477 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
478 q->tone_level[ch][sb][i] = 0;
480 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
484 for (ch = 0; ch < q->nb_channels; ch++)
485 for (i = 0; i < 64; i++) {
486 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
487 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
488 q->tone_level_idx_hi2[ch][sb - 4];
489 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
490 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
491 q->tone_level[ch][sb][i] = 0;
493 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
496 for (ch = 0; ch < q->nb_channels; ch++)
497 for (i = 0; i < 64; i++) {
498 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
499 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
500 q->tone_level[ch][sb][i] = 0;
502 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
511 * Related to synthesis filter
512 * Called by process_subpacket_11
513 * c is built with data from subpacket 11
514 * Most of this function is used only if superblock_type_2_3 == 0,
515 * never seen it in samples.
517 * @param tone_level_idx
518 * @param tone_level_idx_temp
519 * @param coding_method q->coding_method[0][0][0]
520 * @param nb_channels number of channels
521 * @param c coming from subpacket 11, passed as 8*c
522 * @param superblocktype_2_3 flag based on superblock packet type
523 * @param cm_table_select q->cm_table_select
525 static void fill_coding_method_array(sb_int8_array tone_level_idx,
526 sb_int8_array tone_level_idx_temp,
527 sb_int8_array coding_method,
529 int c, int superblocktype_2_3,
533 int tmp, acc, esp_40, comp;
534 int add1, add2, add3, add4;
537 if (!superblocktype_2_3) {
538 /* This case is untested, no samples available */
539 avpriv_request_sample(NULL, "!superblocktype_2_3");
541 for (ch = 0; ch < nb_channels; ch++) {
542 for (sb = 0; sb < 30; sb++) {
543 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
544 add1 = tone_level_idx[ch][sb][j] - 10;
547 add2 = add3 = add4 = 0;
549 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
554 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
559 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
563 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
566 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
568 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
572 for (ch = 0; ch < nb_channels; ch++)
573 for (sb = 0; sb < 30; sb++)
574 for (j = 0; j < 64; j++)
575 acc += tone_level_idx_temp[ch][sb][j];
577 multres = 0x66666667LL * (acc * 10);
578 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
579 for (ch = 0; ch < nb_channels; ch++)
580 for (sb = 0; sb < 30; sb++)
581 for (j = 0; j < 64; j++) {
582 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
585 comp /= 256; // signed shift
613 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
615 for (sb = 0; sb < 30; sb++)
616 fix_coding_method_array(sb, nb_channels, coding_method);
617 for (ch = 0; ch < nb_channels; ch++)
618 for (sb = 0; sb < 30; sb++)
619 for (j = 0; j < 64; j++)
621 if (coding_method[ch][sb][j] < 10)
622 coding_method[ch][sb][j] = 10;
625 if (coding_method[ch][sb][j] < 16)
626 coding_method[ch][sb][j] = 16;
628 if (coding_method[ch][sb][j] < 30)
629 coding_method[ch][sb][j] = 30;
632 } else { // superblocktype_2_3 != 0
633 for (ch = 0; ch < nb_channels; ch++)
634 for (sb = 0; sb < 30; sb++)
635 for (j = 0; j < 64; j++)
636 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
641 * Called by process_subpacket_11 to process more data from subpacket 11
643 * Called by process_subpacket_12 to process data from subpacket 12 with
647 * @param gb bitreader context
648 * @param length packet length in bits
649 * @param sb_min lower subband processed (sb_min included)
650 * @param sb_max higher subband processed (sb_max excluded)
652 static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
653 int length, int sb_min, int sb_max)
655 int sb, j, k, n, ch, run, channels;
656 int joined_stereo, zero_encoding;
658 float type34_div = 0;
659 float type34_predictor;
661 int sign_bits[16] = {0};
664 // If no data use noise
665 for (sb=sb_min; sb < sb_max; sb++)
666 build_sb_samples_from_noise(q, sb);
671 for (sb = sb_min; sb < sb_max; sb++) {
672 channels = q->nb_channels;
674 if (q->nb_channels <= 1 || sb < 12)
679 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
682 if (get_bits_left(gb) >= 16)
683 for (j = 0; j < 16; j++)
684 sign_bits[j] = get_bits1(gb);
686 for (j = 0; j < 64; j++)
687 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
688 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
690 if (fix_coding_method_array(sb, q->nb_channels,
692 av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
693 build_sb_samples_from_noise(q, sb);
699 for (ch = 0; ch < channels; ch++) {
700 FIX_NOISE_IDX(q->noise_idx);
701 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
702 type34_predictor = 0.0;
705 for (j = 0; j < 128; ) {
706 switch (q->coding_method[ch][sb][j / 2]) {
708 if (get_bits_left(gb) >= 10) {
710 for (k = 0; k < 5; k++) {
711 if ((j + 2 * k) >= 128)
713 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
718 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
719 return AVERROR_INVALIDDATA;
722 for (k = 0; k < 5; k++)
723 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
725 for (k = 0; k < 5; k++)
726 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
728 for (k = 0; k < 10; k++)
729 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
735 if (get_bits_left(gb) >= 1) {
740 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
743 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
749 if (get_bits_left(gb) >= 10) {
751 for (k = 0; k < 5; k++) {
754 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
757 n = get_bits (gb, 8);
759 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
760 return AVERROR_INVALIDDATA;
763 for (k = 0; k < 5; k++)
764 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
767 for (k = 0; k < 5; k++)
768 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
774 if (get_bits_left(gb) >= 7) {
777 av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
778 return AVERROR_INVALIDDATA;
781 for (k = 0; k < 3; k++)
782 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
784 for (k = 0; k < 3; k++)
785 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
791 if (get_bits_left(gb) >= 4) {
792 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
793 if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
794 av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
795 return AVERROR_INVALIDDATA;
797 samples[0] = type30_dequant[index];
799 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
805 if (get_bits_left(gb) >= 7) {
807 type34_div = (float)(1 << get_bits(gb, 2));
808 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
809 type34_predictor = samples[0];
812 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
813 if (index >= FF_ARRAY_ELEMS(type34_delta)) {
814 av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
815 return AVERROR_INVALIDDATA;
817 samples[0] = type34_delta[index] / type34_div + type34_predictor;
818 type34_predictor = samples[0];
821 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
827 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
833 for (k = 0; k < run && j + k < 128; k++) {
834 q->sb_samples[0][j + k][sb] =
835 q->tone_level[0][sb][(j + k) / 2] * samples[k];
836 if (q->nb_channels == 2) {
837 if (sign_bits[(j + k) / 8])
838 q->sb_samples[1][j + k][sb] =
839 q->tone_level[1][sb][(j + k) / 2] * -samples[k];
841 q->sb_samples[1][j + k][sb] =
842 q->tone_level[1][sb][(j + k) / 2] * samples[k];
846 for (k = 0; k < run; k++)
848 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
859 * Init the first element of a channel in quantized_coeffs with data
860 * from packet 10 (quantized_coeffs[ch][0]).
861 * This is similar to process_subpacket_9, but for a single channel
862 * and for element [0]
863 * same VLC tables as process_subpacket_9 are used.
865 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
866 * @param gb bitreader context
868 static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
871 int i, k, run, level, diff;
873 if (get_bits_left(gb) < 16)
875 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
877 quantized_coeffs[0] = level;
879 for (i = 0; i < 7; ) {
880 if (get_bits_left(gb) < 16)
882 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
887 if (get_bits_left(gb) < 16)
889 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
891 for (k = 1; k <= run; k++)
892 quantized_coeffs[i + k] = (level + ((k * diff) / run));
901 * Related to synthesis filter, process data from packet 10
902 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
903 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
904 * data from packet 10
907 * @param gb bitreader context
909 static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
913 for (ch = 0; ch < q->nb_channels; ch++) {
914 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
916 if (get_bits_left(gb) < 16) {
917 memset(q->quantized_coeffs[ch][0], 0, 8);
922 n = q->sub_sampling + 1;
924 for (sb = 0; sb < n; sb++)
925 for (ch = 0; ch < q->nb_channels; ch++)
926 for (j = 0; j < 8; j++) {
927 if (get_bits_left(gb) < 1)
930 for (k=0; k < 8; k++) {
931 if (get_bits_left(gb) < 16)
933 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
936 for (k=0; k < 8; k++)
937 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
941 n = QDM2_SB_USED(q->sub_sampling) - 4;
943 for (sb = 0; sb < n; sb++)
944 for (ch = 0; ch < q->nb_channels; ch++) {
945 if (get_bits_left(gb) < 16)
947 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
949 q->tone_level_idx_hi2[ch][sb] -= 16;
951 for (j = 0; j < 8; j++)
952 q->tone_level_idx_mid[ch][sb][j] = -16;
955 n = QDM2_SB_USED(q->sub_sampling) - 5;
957 for (sb = 0; sb < n; sb++)
958 for (ch = 0; ch < q->nb_channels; ch++)
959 for (j = 0; j < 8; j++) {
960 if (get_bits_left(gb) < 16)
962 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
967 * Process subpacket 9, init quantized_coeffs with data from it
970 * @param node pointer to node with packet
972 static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
975 int i, j, k, n, ch, run, level, diff;
977 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
979 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
981 for (i = 1; i < n; i++)
982 for (ch = 0; ch < q->nb_channels; ch++) {
983 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
984 q->quantized_coeffs[ch][i][0] = level;
986 for (j = 0; j < (8 - 1); ) {
987 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
988 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
993 for (k = 1; k <= run; k++)
994 q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
1001 for (ch = 0; ch < q->nb_channels; ch++)
1002 for (i = 0; i < 8; i++)
1003 q->quantized_coeffs[ch][0][i] = 0;
1009 * Process subpacket 10 if not null, else
1012 * @param node pointer to node with packet
1014 static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
1019 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1020 init_tone_level_dequantization(q, &gb);
1021 fill_tone_level_array(q, 1);
1023 fill_tone_level_array(q, 0);
1028 * Process subpacket 11
1031 * @param node pointer to node with packet
1033 static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
1039 length = node->packet->size * 8;
1040 init_get_bits(&gb, node->packet->data, length);
1044 int c = get_bits(&gb, 13);
1047 fill_coding_method_array(q->tone_level_idx,
1048 q->tone_level_idx_temp, q->coding_method,
1049 q->nb_channels, 8 * c,
1050 q->superblocktype_2_3, q->cm_table_select);
1053 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1057 * Process subpacket 12
1060 * @param node pointer to node with packet
1062 static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
1068 length = node->packet->size * 8;
1069 init_get_bits(&gb, node->packet->data, length);
1072 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1076 * Process new subpackets for synthesis filter
1079 * @param list list with synthesis filter packets (list D)
1081 static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
1083 QDM2SubPNode *nodes[4];
1085 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1087 process_subpacket_9(q, nodes[0]);
1089 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1091 process_subpacket_10(q, nodes[1]);
1093 process_subpacket_10(q, NULL);
1095 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1096 if (nodes[0] && nodes[1] && nodes[2])
1097 process_subpacket_11(q, nodes[2]);
1099 process_subpacket_11(q, NULL);
1101 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1102 if (nodes[0] && nodes[1] && nodes[3])
1103 process_subpacket_12(q, nodes[3]);
1105 process_subpacket_12(q, NULL);
1109 * Decode superblock, fill packet lists.
1113 static void qdm2_decode_super_block(QDM2Context *q)
1116 QDM2SubPacket header, *packet;
1117 int i, packet_bytes, sub_packet_size, sub_packets_D;
1118 unsigned int next_index = 0;
1120 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1121 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1122 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1124 q->sub_packets_B = 0;
1127 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1129 init_get_bits(&gb, q->compressed_data, q->compressed_size * 8);
1130 qdm2_decode_sub_packet_header(&gb, &header);
1132 if (header.type < 2 || header.type >= 8) {
1134 av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1138 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1139 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1141 init_get_bits(&gb, header.data, header.size * 8);
1143 if (header.type == 2 || header.type == 4 || header.type == 5) {
1144 int csum = 257 * get_bits(&gb, 8);
1145 csum += 2 * get_bits(&gb, 8);
1147 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1151 av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1156 q->sub_packet_list_B[0].packet = NULL;
1157 q->sub_packet_list_D[0].packet = NULL;
1159 for (i = 0; i < 6; i++)
1160 if (--q->fft_level_exp[i] < 0)
1161 q->fft_level_exp[i] = 0;
1163 for (i = 0; packet_bytes > 0; i++) {
1166 if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1167 SAMPLES_NEEDED_2("too many packet bytes");
1171 q->sub_packet_list_A[i].next = NULL;
1174 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1176 /* seek to next block */
1177 init_get_bits(&gb, header.data, header.size * 8);
1178 skip_bits(&gb, next_index * 8);
1180 if (next_index >= header.size)
1184 /* decode subpacket */
1185 packet = &q->sub_packets[i];
1186 qdm2_decode_sub_packet_header(&gb, packet);
1187 next_index = packet->size + get_bits_count(&gb) / 8;
1188 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1190 if (packet->type == 0)
1193 if (sub_packet_size > packet_bytes) {
1194 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1196 packet->size += packet_bytes - sub_packet_size;
1199 packet_bytes -= sub_packet_size;
1201 /* add subpacket to 'all subpackets' list */
1202 q->sub_packet_list_A[i].packet = packet;
1204 /* add subpacket to related list */
1205 if (packet->type == 8) {
1206 SAMPLES_NEEDED_2("packet type 8");
1208 } else if (packet->type >= 9 && packet->type <= 12) {
1209 /* packets for MPEG Audio like Synthesis Filter */
1210 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1211 } else if (packet->type == 13) {
1212 for (j = 0; j < 6; j++)
1213 q->fft_level_exp[j] = get_bits(&gb, 6);
1214 } else if (packet->type == 14) {
1215 for (j = 0; j < 6; j++)
1216 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1217 } else if (packet->type == 15) {
1218 SAMPLES_NEEDED_2("packet type 15")
1220 } else if (packet->type >= 16 && packet->type < 48 &&
1221 !fft_subpackets[packet->type - 16]) {
1222 /* packets for FFT */
1223 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1225 } // Packet bytes loop
1227 if (q->sub_packet_list_D[0].packet) {
1228 process_synthesis_subpackets(q, q->sub_packet_list_D);
1229 q->do_synth_filter = 1;
1230 } else if (q->do_synth_filter) {
1231 process_subpacket_10(q, NULL);
1232 process_subpacket_11(q, NULL);
1233 process_subpacket_12(q, NULL);
1237 static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1238 int offset, int duration, int channel,
1241 if (q->fft_coefs_min_index[duration] < 0)
1242 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1244 q->fft_coefs[q->fft_coefs_index].sub_packet =
1245 ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1246 q->fft_coefs[q->fft_coefs_index].channel = channel;
1247 q->fft_coefs[q->fft_coefs_index].offset = offset;
1248 q->fft_coefs[q->fft_coefs_index].exp = exp;
1249 q->fft_coefs[q->fft_coefs_index].phase = phase;
1250 q->fft_coefs_index++;
1253 static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
1254 GetBitContext *gb, int b)
1256 int channel, stereo, phase, exp;
1257 int local_int_4, local_int_8, stereo_phase, local_int_10;
1258 int local_int_14, stereo_exp, local_int_20, local_int_28;
1264 local_int_8 = (4 - duration);
1265 local_int_10 = 1 << (q->group_order - duration - 1);
1268 while (get_bits_left(gb)>0) {
1269 if (q->superblocktype_2_3) {
1270 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1271 if (get_bits_left(gb)<0) {
1272 if(local_int_4 < q->group_size)
1273 av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1278 local_int_4 += local_int_10;
1279 local_int_28 += (1 << local_int_8);
1281 local_int_4 += 8 * local_int_10;
1282 local_int_28 += (8 << local_int_8);
1287 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1288 while (offset >= (local_int_10 - 1)) {
1289 offset += (1 - (local_int_10 - 1));
1290 local_int_4 += local_int_10;
1291 local_int_28 += (1 << local_int_8);
1295 if (local_int_4 >= q->group_size)
1298 local_int_14 = (offset >> local_int_8);
1299 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1302 if (q->nb_channels > 1) {
1303 channel = get_bits1(gb);
1304 stereo = get_bits1(gb);
1310 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1311 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1312 exp = (exp < 0) ? 0 : exp;
1314 phase = get_bits(gb, 3);
1319 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1320 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1321 if (stereo_phase < 0)
1325 if (q->frequency_range > (local_int_14 + 1)) {
1326 int sub_packet = (local_int_20 + local_int_28);
1328 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1329 channel, exp, phase);
1331 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1333 stereo_exp, stereo_phase);
1339 static void qdm2_decode_fft_packets(QDM2Context *q)
1341 int i, j, min, max, value, type, unknown_flag;
1344 if (!q->sub_packet_list_B[0].packet)
1347 /* reset minimum indexes for FFT coefficients */
1348 q->fft_coefs_index = 0;
1349 for (i = 0; i < 5; i++)
1350 q->fft_coefs_min_index[i] = -1;
1352 /* process subpackets ordered by type, largest type first */
1353 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1354 QDM2SubPacket *packet = NULL;
1356 /* find subpacket with largest type less than max */
1357 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1358 value = q->sub_packet_list_B[j].packet->type;
1359 if (value > min && value < max) {
1361 packet = q->sub_packet_list_B[j].packet;
1367 /* check for errors (?) */
1372 (packet->type < 16 || packet->type >= 48 ||
1373 fft_subpackets[packet->type - 16]))
1376 /* decode FFT tones */
1377 init_get_bits(&gb, packet->data, packet->size * 8);
1379 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1384 type = packet->type;
1386 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1387 int duration = q->sub_sampling + 5 - (type & 15);
1389 if (duration >= 0 && duration < 4)
1390 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1391 } else if (type == 31) {
1392 for (j = 0; j < 4; j++)
1393 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1394 } else if (type == 46) {
1395 for (j = 0; j < 6; j++)
1396 q->fft_level_exp[j] = get_bits(&gb, 6);
1397 for (j = 0; j < 4; j++)
1398 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1400 } // Loop on B packets
1402 /* calculate maximum indexes for FFT coefficients */
1403 for (i = 0, j = -1; i < 5; i++)
1404 if (q->fft_coefs_min_index[i] >= 0) {
1406 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1410 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1413 static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
1418 const double iscale = 2.0 * M_PI / 512.0;
1420 tone->phase += tone->phase_shift;
1422 /* calculate current level (maximum amplitude) of tone */
1423 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1424 c.im = level * sin(tone->phase * iscale);
1425 c.re = level * cos(tone->phase * iscale);
1427 /* generate FFT coefficients for tone */
1428 if (tone->duration >= 3 || tone->cutoff >= 3) {
1429 tone->complex[0].im += c.im;
1430 tone->complex[0].re += c.re;
1431 tone->complex[1].im -= c.im;
1432 tone->complex[1].re -= c.re;
1434 f[1] = -tone->table[4];
1435 f[0] = tone->table[3] - tone->table[0];
1436 f[2] = 1.0 - tone->table[2] - tone->table[3];
1437 f[3] = tone->table[1] + tone->table[4] - 1.0;
1438 f[4] = tone->table[0] - tone->table[1];
1439 f[5] = tone->table[2];
1440 for (i = 0; i < 2; i++) {
1441 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1443 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1444 c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1446 for (i = 0; i < 4; i++) {
1447 tone->complex[i].re += c.re * f[i + 2];
1448 tone->complex[i].im += c.im * f[i + 2];
1452 /* copy the tone if it has not yet died out */
1453 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1454 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1455 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1459 static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1462 const double iscale = 0.25 * M_PI;
1464 for (ch = 0; ch < q->channels; ch++) {
1465 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1469 /* apply FFT tones with duration 4 (1 FFT period) */
1470 if (q->fft_coefs_min_index[4] >= 0)
1471 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1475 if (q->fft_coefs[i].sub_packet != sub_packet)
1478 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1479 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];
1481 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1482 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1483 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1484 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1485 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1486 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1489 /* generate existing FFT tones */
1490 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1491 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1492 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1495 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1496 for (i = 0; i < 4; i++)
1497 if (q->fft_coefs_min_index[i] >= 0) {
1498 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1502 if (q->fft_coefs[j].sub_packet != sub_packet)
1506 offset = q->fft_coefs[j].offset >> four_i;
1507 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1509 if (offset < q->frequency_range) {
1511 tone.cutoff = offset;
1513 tone.cutoff = (offset >= 60) ? 3 : 2;
1515 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];
1516 tone.complex = &q->fft.complex[ch][offset];
1517 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1518 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1519 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1521 tone.time_index = 0;
1523 qdm2_fft_generate_tone(q, &tone);
1526 q->fft_coefs_min_index[i] = j;
1530 static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1532 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1533 float *out = q->output_buffer + channel;
1535 q->fft.complex[channel][0].re *= 2.0f;
1536 q->fft.complex[channel][0].im = 0.0f;
1537 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1538 /* add samples to output buffer */
1539 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1540 out[0] += q->fft.complex[channel][i].re * gain;
1541 out[q->channels] += q->fft.complex[channel][i].im * gain;
1542 out += 2 * q->channels;
1548 * @param index subpacket number
1550 static void qdm2_synthesis_filter(QDM2Context *q, int index)
1552 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1554 /* copy sb_samples */
1555 sb_used = QDM2_SB_USED(q->sub_sampling);
1557 for (ch = 0; ch < q->channels; ch++)
1558 for (i = 0; i < 8; i++)
1559 for (k = sb_used; k < SBLIMIT; k++)
1560 q->sb_samples[ch][(8 * index) + i][k] = 0;
1562 for (ch = 0; ch < q->nb_channels; ch++) {
1563 float *samples_ptr = q->samples + ch;
1565 for (i = 0; i < 8; i++) {
1566 ff_mpa_synth_filter_float(&q->mpadsp,
1567 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1568 ff_mpa_synth_window_float, &dither_state,
1569 samples_ptr, q->nb_channels,
1570 q->sb_samples[ch][(8 * index) + i]);
1571 samples_ptr += 32 * q->nb_channels;
1575 /* add samples to output buffer */
1576 sub_sampling = (4 >> q->sub_sampling);
1578 for (ch = 0; ch < q->channels; ch++)
1579 for (i = 0; i < q->frame_size; i++)
1580 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1584 * Init static data (does not depend on specific file)
1588 static av_cold void qdm2_init_static_data(void) {
1595 ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1596 softclip_table_init();
1598 init_noise_samples();
1604 * Init parameters from codec extradata
1606 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1608 QDM2Context *s = avctx->priv_data;
1609 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 bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1655 while (bytestream2_get_bytes_left(&gb) > 8) {
1656 if (bytestream2_peek_be64(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |
1657 (uint64_t)MKBETAG('Q','D','M','2')))
1659 bytestream2_skip(&gb, 1);
1662 if (bytestream2_get_bytes_left(&gb) < 12) {
1663 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1664 bytestream2_get_bytes_left(&gb));
1665 return AVERROR_INVALIDDATA;
1668 bytestream2_skip(&gb, 8);
1669 size = bytestream2_get_be32(&gb);
1671 if (size > bytestream2_get_bytes_left(&gb)) {
1672 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1673 bytestream2_get_bytes_left(&gb), size);
1674 return AVERROR_INVALIDDATA;
1677 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1678 if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) {
1679 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1680 return AVERROR_INVALIDDATA;
1683 bytestream2_skip(&gb, 4);
1685 avctx->channels = s->nb_channels = s->channels = bytestream2_get_be32(&gb);
1686 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1687 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1688 return AVERROR_INVALIDDATA;
1690 avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1693 avctx->sample_rate = bytestream2_get_be32(&gb);
1694 avctx->bit_rate = bytestream2_get_be32(&gb);
1695 s->group_size = bytestream2_get_be32(&gb);
1696 s->fft_size = bytestream2_get_be32(&gb);
1697 s->checksum_size = bytestream2_get_be32(&gb);
1698 if (s->checksum_size >= 1U << 28) {
1699 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1700 return AVERROR_INVALIDDATA;
1703 s->fft_order = av_log2(s->fft_size) + 1;
1705 // something like max decodable tones
1706 s->group_order = av_log2(s->group_size) + 1;
1707 s->frame_size = s->group_size / 16; // 16 iterations per super block
1709 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1710 return AVERROR_INVALIDDATA;
1712 s->sub_sampling = s->fft_order - 7;
1713 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1715 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1716 case 0: tmp = 40; break;
1717 case 1: tmp = 48; break;
1718 case 2: tmp = 56; break;
1719 case 3: tmp = 72; break;
1720 case 4: tmp = 80; break;
1721 case 5: tmp = 100;break;
1722 default: tmp=s->sub_sampling; break;
1725 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1726 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1727 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1728 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1729 s->cm_table_select = tmp_val;
1731 if (avctx->bit_rate <= 8000)
1732 s->coeff_per_sb_select = 0;
1733 else if (avctx->bit_rate < 16000)
1734 s->coeff_per_sb_select = 1;
1736 s->coeff_per_sb_select = 2;
1738 // Fail on unknown fft order
1739 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1740 avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
1741 return AVERROR_PATCHWELCOME;
1743 if (s->fft_size != (1 << (s->fft_order - 1))) {
1744 av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1745 return AVERROR_INVALIDDATA;
1748 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1749 ff_mpadsp_init(&s->mpadsp);
1751 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1756 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1758 QDM2Context *s = avctx->priv_data;
1760 ff_rdft_end(&s->rdft_ctx);
1765 static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1768 const int frame_size = (q->frame_size * q->channels);
1770 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1773 /* select input buffer */
1774 q->compressed_data = in;
1775 q->compressed_size = q->checksum_size;
1777 /* copy old block, clear new block of output samples */
1778 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1779 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1781 /* decode block of QDM2 compressed data */
1782 if (q->sub_packet == 0) {
1783 q->has_errors = 0; // zero it for a new super block
1784 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1785 qdm2_decode_super_block(q);
1788 /* parse subpackets */
1789 if (!q->has_errors) {
1790 if (q->sub_packet == 2)
1791 qdm2_decode_fft_packets(q);
1793 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1796 /* sound synthesis stage 1 (FFT) */
1797 for (ch = 0; ch < q->channels; ch++) {
1798 qdm2_calculate_fft(q, ch, q->sub_packet);
1800 if (!q->has_errors && q->sub_packet_list_C[0].packet) {
1801 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1806 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1807 if (!q->has_errors && q->do_synth_filter)
1808 qdm2_synthesis_filter(q, q->sub_packet);
1810 q->sub_packet = (q->sub_packet + 1) % 16;
1812 /* clip and convert output float[] to 16-bit signed samples */
1813 for (i = 0; i < frame_size; i++) {
1814 int value = (int)q->output_buffer[i];
1816 if (value > SOFTCLIP_THRESHOLD)
1817 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1818 else if (value < -SOFTCLIP_THRESHOLD)
1819 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1827 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1828 int *got_frame_ptr, AVPacket *avpkt)
1830 AVFrame *frame = data;
1831 const uint8_t *buf = avpkt->data;
1832 int buf_size = avpkt->size;
1833 QDM2Context *s = avctx->priv_data;
1839 if(buf_size < s->checksum_size)
1842 /* get output buffer */
1843 frame->nb_samples = 16 * s->frame_size;
1844 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1846 out = (int16_t *)frame->data[0];
1848 for (i = 0; i < 16; i++) {
1849 if ((ret = qdm2_decode(s, buf, out)) < 0)
1851 out += s->channels * s->frame_size;
1856 return s->checksum_size;
1859 AVCodec ff_qdm2_decoder = {
1861 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
1862 .type = AVMEDIA_TYPE_AUDIO,
1863 .id = AV_CODEC_ID_QDM2,
1864 .priv_data_size = sizeof(QDM2Context),
1865 .init = qdm2_decode_init,
1866 .close = qdm2_decode_close,
1867 .decode = qdm2_decode_frame,
1868 .capabilities = AV_CODEC_CAP_DR1,