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 library is free software; you can redistribute it and/or
9
* modify it under the terms of the GNU Lesser General Public
10
* License as published by the Free Software Foundation; either
11
* version 2 of the License, or (at your option) any later version.
13
* This library is distributed in the hope that it will be useful,
14
* but WITHOUT ANY WARRANTY; without even the implied warranty of
15
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16
* Lesser General Public License for more details.
18
* You should have received a copy of the GNU Lesser General Public
19
* License along with this library; if not, write to the Free Software
20
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
27
* @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
28
* The decoder is not perfect yet, there are still some distortions
29
* especially on files encoded with 16 or 8 subbands.
36
#define ALT_BITSTREAM_READER_LE
38
#include "bitstream.h"
41
#ifdef CONFIG_MPEGAUDIO_HP
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#define USE_HIGHPRECISION
45
#include "mpegaudio.h"
53
#define SOFTCLIP_THRESHOLD 27600
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#define HARDCLIP_THRESHOLD 35716
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#define QDM2_LIST_ADD(list, size, packet) \
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list[size - 1].next = &list[size]; \
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list[size].packet = packet; \
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list[size].next = NULL; \
67
// Result is 8, 16 or 30
68
#define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
70
#define FIX_NOISE_IDX(noise_idx) \
71
if ((noise_idx) >= 3840) \
72
(noise_idx) -= 3840; \
74
#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
76
#define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
78
#define SAMPLES_NEEDED \
79
av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
81
#define SAMPLES_NEEDED_2(why) \
82
av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
85
typedef int8_t sb_int8_array[2][30][64];
91
int type; ///< subpacket type
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unsigned int size; ///< subpacket size
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const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
97
* A node in the subpacket list
99
typedef struct _QDM2SubPNode {
100
QDM2SubPacket *packet; ///< packet
101
struct _QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
130
QDM2Complex complex[256 + 1] __attribute__((aligned(16)));
131
float samples_im[MPA_MAX_CHANNELS][256];
132
float samples_re[MPA_MAX_CHANNELS][256];
136
* QDM2 decoder context
139
/// Parameters from codec header, do not change during playback
140
int nb_channels; ///< number of channels
141
int channels; ///< number of channels
142
int group_size; ///< size of frame group (16 frames per group)
143
int fft_size; ///< size of FFT, in complex numbers
144
int checksum_size; ///< size of data block, used also for checksum
146
/// Parameters built from header parameters, do not change during playback
147
int group_order; ///< order of frame group
148
int fft_order; ///< order of FFT (actually fftorder+1)
149
int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
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int frame_size; ///< size of data frame
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int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
153
int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
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int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
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/// Packets and packet lists
157
QDM2SubPacket sub_packets[16]; ///< the packets themselves
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QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
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QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
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int sub_packets_B; ///< number of packets on 'B' list
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QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
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QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
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FFTTone fft_tones[1000];
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FFTCoefficient fft_coefs[1000];
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int fft_coefs_min_index[5];
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int fft_coefs_max_index[5];
172
int fft_level_exp[6];
174
FFTComplex exptab[128];
178
uint8_t *compressed_data;
180
float output_buffer[1024];
183
MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16)));
184
int synth_buf_offset[MPA_MAX_CHANNELS];
185
int32_t sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT] __attribute__((aligned(16)));
187
/// Mixed temporary data used in decoding
188
float tone_level[MPA_MAX_CHANNELS][30][64];
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int8_t coding_method[MPA_MAX_CHANNELS][30][64];
190
int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
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int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
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int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
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int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
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int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
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int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
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int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
199
int has_errors; ///< packet has errors
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int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
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int do_synth_filter; ///< used to perform or skip synthesis filter
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int noise_idx; ///< index for dithering noise table
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static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
210
static VLC vlc_tab_level;
211
static VLC vlc_tab_diff;
212
static VLC vlc_tab_run;
213
static VLC fft_level_exp_alt_vlc;
214
static VLC fft_level_exp_vlc;
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static VLC fft_stereo_exp_vlc;
216
static VLC fft_stereo_phase_vlc;
217
static VLC vlc_tab_tone_level_idx_hi1;
218
static VLC vlc_tab_tone_level_idx_mid;
219
static VLC vlc_tab_tone_level_idx_hi2;
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static VLC vlc_tab_type30;
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static VLC vlc_tab_type34;
222
static VLC vlc_tab_fft_tone_offset[5];
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static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
225
static float noise_table[4096];
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static uint8_t random_dequant_index[256][5];
227
static uint8_t random_dequant_type24[128][3];
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static float noise_samples[128];
230
static MPA_INT mpa_window[512] __attribute__((aligned(16)));
233
static void softclip_table_init(void) {
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double dfl = SOFTCLIP_THRESHOLD - 32767;
236
float delta = 1.0 / -dfl;
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for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
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softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
242
// random generated table
243
static void rnd_table_init(void) {
247
uint64_t random_seed = 0;
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float delta = 1.0 / 16384.0;
249
for(i = 0; i < 4096 ;i++) {
250
random_seed = random_seed * 214013 + 2531011;
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noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
254
for (i = 0; i < 256 ;i++) {
257
for (j = 0; j < 5 ;j++) {
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random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
259
ldw = (uint32_t)ldw % (uint32_t)random_seed;
260
tmp64_1 = (random_seed * 0x55555556);
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hdw = (uint32_t)(tmp64_1 >> 32);
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random_seed = (uint64_t)(hdw + (ldw >> 31));
265
for (i = 0; i < 128 ;i++) {
268
for (j = 0; j < 3 ;j++) {
269
random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
270
ldw = (uint32_t)ldw % (uint32_t)random_seed;
271
tmp64_1 = (random_seed * 0x66666667);
272
hdw = (uint32_t)(tmp64_1 >> 33);
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random_seed = hdw + (ldw >> 31);
279
static void init_noise_samples(void) {
282
float delta = 1.0 / 16384.0;
283
for (i = 0; i < 128;i++) {
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random_seed = random_seed * 214013 + 2531011;
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noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
290
static void qdm2_init_vlc(void)
292
init_vlc (&vlc_tab_level, 8, 24,
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vlc_tab_level_huffbits, 1, 1,
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vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
296
init_vlc (&vlc_tab_diff, 8, 37,
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vlc_tab_diff_huffbits, 1, 1,
298
vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
300
init_vlc (&vlc_tab_run, 5, 6,
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vlc_tab_run_huffbits, 1, 1,
302
vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
304
init_vlc (&fft_level_exp_alt_vlc, 8, 28,
305
fft_level_exp_alt_huffbits, 1, 1,
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fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
308
init_vlc (&fft_level_exp_vlc, 8, 20,
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fft_level_exp_huffbits, 1, 1,
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fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
312
init_vlc (&fft_stereo_exp_vlc, 6, 7,
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fft_stereo_exp_huffbits, 1, 1,
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fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
316
init_vlc (&fft_stereo_phase_vlc, 6, 9,
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fft_stereo_phase_huffbits, 1, 1,
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fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
320
init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
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vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
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vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
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vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
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vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
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vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
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vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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init_vlc (&vlc_tab_type30, 6, 9,
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vlc_tab_type30_huffbits, 1, 1,
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vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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init_vlc (&vlc_tab_type34, 5, 10,
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vlc_tab_type34_huffbits, 1, 1,
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vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
340
init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
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vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
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vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
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vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
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vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
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vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
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vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
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vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
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vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
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vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
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vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
362
/* for floating point to fixed point conversion */
363
static float f2i_scale = (float) (1 << (FRAC_BITS - 15));
366
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
370
value = get_vlc2(gb, vlc->table, vlc->bits, depth);
372
/* stage-2, 3 bits exponent escape sequence */
374
value = get_bits (gb, get_bits (gb, 3) + 1);
376
/* stage-3, optional */
378
int tmp = vlc_stage3_values[value];
380
if ((value & ~3) > 0)
381
tmp += get_bits (gb, (value >> 2));
389
static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
391
int value = qdm2_get_vlc (gb, vlc, 0, depth);
393
return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
400
* @param data pointer to data to be checksum'ed
401
* @param length data length
402
* @param value checksum value
404
* @return 0 if checksum is OK
406
static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) {
409
for (i=0; i < length; i++)
412
return (uint16_t)(value & 0xffff);
417
* Fills a QDM2SubPacket structure with packet type, size, and data pointer.
419
* @param gb bitreader context
420
* @param sub_packet packet under analysis
422
static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
424
sub_packet->type = get_bits (gb, 8);
426
if (sub_packet->type == 0) {
427
sub_packet->size = 0;
428
sub_packet->data = NULL;
430
sub_packet->size = get_bits (gb, 8);
432
if (sub_packet->type & 0x80) {
433
sub_packet->size <<= 8;
434
sub_packet->size |= get_bits (gb, 8);
435
sub_packet->type &= 0x7f;
438
if (sub_packet->type == 0x7f)
439
sub_packet->type |= (get_bits (gb, 8) << 8);
441
sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
444
av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
445
sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
450
* Return node pointer to first packet of requested type in list.
452
* @param list list of subpackets to be scanned
453
* @param type type of searched subpacket
454
* @return node pointer for subpacket if found, else NULL
456
static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
458
while (list != NULL && list->packet != NULL) {
459
if (list->packet->type == type)
468
* Replaces 8 elements with their average value.
469
* Called by qdm2_decode_superblock before starting subblock decoding.
473
static void average_quantized_coeffs (QDM2Context *q)
475
int i, j, n, ch, sum;
477
n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
479
for (ch = 0; ch < q->nb_channels; ch++)
480
for (i = 0; i < n; i++) {
483
for (j = 0; j < 8; j++)
484
sum += q->quantized_coeffs[ch][i][j];
490
for (j=0; j < 8; j++)
491
q->quantized_coeffs[ch][i][j] = sum;
497
* Build subband samples with noise weighted by q->tone_level.
498
* Called by synthfilt_build_sb_samples.
501
* @param sb subband index
503
static void build_sb_samples_from_noise (QDM2Context *q, int sb)
507
FIX_NOISE_IDX(q->noise_idx);
512
for (ch = 0; ch < q->nb_channels; ch++)
513
for (j = 0; j < 64; j++) {
514
q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
515
q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
521
* Called while processing data from subpackets 11 and 12.
522
* Used after making changes to coding_method array.
524
* @param sb subband index
525
* @param channels number of channels
526
* @param coding_method q->coding_method[0][0][0]
528
static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
533
int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
535
for (ch = 0; ch < channels; ch++) {
536
for (j = 0; j < 64; ) {
537
if((coding_method[ch][sb][j] - 8) > 22) {
541
switch (switchtable[coding_method[ch][sb][j]-8]) {
542
case 0: run = 10; case_val = 10; break;
543
case 1: run = 1; case_val = 16; break;
544
case 2: run = 5; case_val = 24; break;
545
case 3: run = 3; case_val = 30; break;
546
case 4: run = 1; case_val = 30; break;
547
case 5: run = 1; case_val = 8; break;
548
default: run = 1; case_val = 8; break;
551
for (k = 0; k < run; k++)
553
if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
556
//not debugged, almost never used
557
memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
558
memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
567
* Related to synthesis filter
568
* Called by process_subpacket_10
571
* @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
573
static void fill_tone_level_array (QDM2Context *q, int flag)
575
int i, sb, ch, sb_used;
578
// This should never happen
579
if (q->nb_channels <= 0)
582
for (ch = 0; ch < q->nb_channels; ch++)
583
for (sb = 0; sb < 30; sb++)
584
for (i = 0; i < 8; i++) {
585
if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
586
tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
587
q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
589
tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
592
q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
595
sb_used = QDM2_SB_USED(q->sub_sampling);
597
if ((q->superblocktype_2_3 != 0) && !flag) {
598
for (sb = 0; sb < sb_used; sb++)
599
for (ch = 0; ch < q->nb_channels; ch++)
600
for (i = 0; i < 64; i++) {
601
q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
602
if (q->tone_level_idx[ch][sb][i] < 0)
603
q->tone_level[ch][sb][i] = 0;
605
q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
608
tab = q->superblocktype_2_3 ? 0 : 1;
609
for (sb = 0; sb < sb_used; sb++) {
610
if ((sb >= 4) && (sb <= 23)) {
611
for (ch = 0; ch < q->nb_channels; ch++)
612
for (i = 0; i < 64; i++) {
613
tmp = q->tone_level_idx_base[ch][sb][i / 8] -
614
q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
615
q->tone_level_idx_mid[ch][sb - 4][i / 8] -
616
q->tone_level_idx_hi2[ch][sb - 4];
617
q->tone_level_idx[ch][sb][i] = tmp & 0xff;
618
if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
619
q->tone_level[ch][sb][i] = 0;
621
q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
625
for (ch = 0; ch < q->nb_channels; ch++)
626
for (i = 0; i < 64; i++) {
627
tmp = q->tone_level_idx_base[ch][sb][i / 8] -
628
q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
629
q->tone_level_idx_hi2[ch][sb - 4];
630
q->tone_level_idx[ch][sb][i] = tmp & 0xff;
631
if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
632
q->tone_level[ch][sb][i] = 0;
634
q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
637
for (ch = 0; ch < q->nb_channels; ch++)
638
for (i = 0; i < 64; i++) {
639
tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
640
if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
641
q->tone_level[ch][sb][i] = 0;
643
q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
655
* Related to synthesis filter
656
* Called by process_subpacket_11
657
* c is built with data from subpacket 11
658
* Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
660
* @param tone_level_idx
661
* @param tone_level_idx_temp
662
* @param coding_method q->coding_method[0][0][0]
663
* @param nb_channels number of channels
664
* @param c coming from subpacket 11, passed as 8*c
665
* @param superblocktype_2_3 flag based on superblock packet type
666
* @param cm_table_select q->cm_table_select
668
static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
669
sb_int8_array coding_method, int nb_channels,
670
int c, int superblocktype_2_3, int cm_table_select)
673
int tmp, acc, esp_40, comp;
674
int add1, add2, add3, add4;
677
// This should never happen
678
if (nb_channels <= 0)
681
if (!superblocktype_2_3) {
682
/* This case is untested, no samples available */
684
for (ch = 0; ch < nb_channels; ch++)
685
for (sb = 0; sb < 30; sb++) {
686
for (j = 1; j < 64; j++) {
687
add1 = tone_level_idx[ch][sb][j] - 10;
690
add2 = add3 = add4 = 0;
692
add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
697
add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
702
add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
706
tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
709
tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
711
tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
714
for (ch = 0; ch < nb_channels; ch++)
715
for (sb = 0; sb < 30; sb++)
716
for (j = 0; j < 64; j++)
717
acc += tone_level_idx_temp[ch][sb][j];
719
tmp = c * 256 / (acc & 0xffff);
720
multres = 0x66666667 * (acc * 10);
721
esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
722
for (ch = 0; ch < nb_channels; ch++)
723
for (sb = 0; sb < 30; sb++)
724
for (j = 0; j < 64; j++) {
725
comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
728
comp /= 256; // signed shift
756
coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
758
for (sb = 0; sb < 30; sb++)
759
fix_coding_method_array(sb, nb_channels, coding_method);
760
for (ch = 0; ch < nb_channels; ch++)
761
for (sb = 0; sb < 30; sb++)
762
for (j = 0; j < 64; j++)
764
if (coding_method[ch][sb][j] < 10)
765
coding_method[ch][sb][j] = 10;
768
if (coding_method[ch][sb][j] < 16)
769
coding_method[ch][sb][j] = 16;
771
if (coding_method[ch][sb][j] < 30)
772
coding_method[ch][sb][j] = 30;
775
} else { // superblocktype_2_3 != 0
776
for (ch = 0; ch < nb_channels; ch++)
777
for (sb = 0; sb < 30; sb++)
778
for (j = 0; j < 64; j++)
779
coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
788
* Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
789
* Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
792
* @param gb bitreader context
793
* @param length packet length in bits
794
* @param sb_min lower subband processed (sb_min included)
795
* @param sb_max higher subband processed (sb_max excluded)
797
static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
799
int sb, j, k, n, ch, run, channels;
800
int joined_stereo, zero_encoding, chs;
802
float type34_div = 0;
803
float type34_predictor;
804
float samples[10], sign_bits[16];
807
// If no data use noise
808
for (sb=sb_min; sb < sb_max; sb++)
809
build_sb_samples_from_noise (q, sb);
814
for (sb = sb_min; sb < sb_max; sb++) {
815
FIX_NOISE_IDX(q->noise_idx);
817
channels = q->nb_channels;
819
if (q->nb_channels <= 1 || sb < 12)
824
joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
827
if (BITS_LEFT(length,gb) >= 16)
828
for (j = 0; j < 16; j++)
829
sign_bits[j] = get_bits1 (gb);
831
for (j = 0; j < 64; j++)
832
if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
833
q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
835
fix_coding_method_array(sb, q->nb_channels, q->coding_method);
839
for (ch = 0; ch < channels; ch++) {
840
zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
841
type34_predictor = 0.0;
844
for (j = 0; j < 128; ) {
845
switch (q->coding_method[ch][sb][j / 2]) {
847
if (BITS_LEFT(length,gb) >= 10) {
849
for (k = 0; k < 5; k++) {
850
if ((j + 2 * k) >= 128)
852
samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
856
for (k = 0; k < 5; k++)
857
samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
859
for (k = 0; k < 5; k++)
860
samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
862
for (k = 0; k < 10; k++)
863
samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
869
if (BITS_LEFT(length,gb) >= 1) {
874
f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
877
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
883
if (BITS_LEFT(length,gb) >= 10) {
885
for (k = 0; k < 5; k++) {
888
samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
891
n = get_bits (gb, 8);
892
for (k = 0; k < 5; k++)
893
samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
896
for (k = 0; k < 5; k++)
897
samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
903
if (BITS_LEFT(length,gb) >= 7) {
905
for (k = 0; k < 3; k++)
906
samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
908
for (k = 0; k < 3; k++)
909
samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
915
if (BITS_LEFT(length,gb) >= 4)
916
samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
918
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
924
if (BITS_LEFT(length,gb) >= 7) {
926
type34_div = (float)(1 << get_bits(gb, 2));
927
samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
928
type34_predictor = samples[0];
931
samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
932
type34_predictor = samples[0];
935
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
941
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
947
float tmp[10][MPA_MAX_CHANNELS];
949
for (k = 0; k < run; k++) {
950
tmp[k][0] = samples[k];
951
tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
953
for (chs = 0; chs < q->nb_channels; chs++)
954
for (k = 0; k < run; k++)
956
q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
958
for (k = 0; k < run; k++)
960
q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
971
* Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
972
* This is similar to process_subpacket_9, but for a single channel and for element [0]
973
* same VLC tables as process_subpacket_9 are used.
976
* @param quantized_coeffs pointer to quantized_coeffs[ch][0]
977
* @param gb bitreader context
978
* @param length packet length in bits
980
static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
982
int i, k, run, level, diff;
984
if (BITS_LEFT(length,gb) < 16)
986
level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
988
quantized_coeffs[0] = level;
990
for (i = 0; i < 7; ) {
991
if (BITS_LEFT(length,gb) < 16)
993
run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
995
if (BITS_LEFT(length,gb) < 16)
997
diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
999
for (k = 1; k <= run; k++)
1000
quantized_coeffs[i + k] = (level + ((k * diff) / run));
1009
* Related to synthesis filter, process data from packet 10
1010
* Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1011
* Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1014
* @param gb bitreader context
1015
* @param length packet length in bits
1017
static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1019
int sb, j, k, n, ch;
1021
for (ch = 0; ch < q->nb_channels; ch++) {
1022
init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1024
if (BITS_LEFT(length,gb) < 16) {
1025
memset(q->quantized_coeffs[ch][0], 0, 8);
1030
n = q->sub_sampling + 1;
1032
for (sb = 0; sb < n; sb++)
1033
for (ch = 0; ch < q->nb_channels; ch++)
1034
for (j = 0; j < 8; j++) {
1035
if (BITS_LEFT(length,gb) < 1)
1037
if (get_bits1(gb)) {
1038
for (k=0; k < 8; k++) {
1039
if (BITS_LEFT(length,gb) < 16)
1041
q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1044
for (k=0; k < 8; k++)
1045
q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1049
n = QDM2_SB_USED(q->sub_sampling) - 4;
1051
for (sb = 0; sb < n; sb++)
1052
for (ch = 0; ch < q->nb_channels; ch++) {
1053
if (BITS_LEFT(length,gb) < 16)
1055
q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1057
q->tone_level_idx_hi2[ch][sb] -= 16;
1059
for (j = 0; j < 8; j++)
1060
q->tone_level_idx_mid[ch][sb][j] = -16;
1063
n = QDM2_SB_USED(q->sub_sampling) - 5;
1065
for (sb = 0; sb < n; sb++)
1066
for (ch = 0; ch < q->nb_channels; ch++)
1067
for (j = 0; j < 8; j++) {
1068
if (BITS_LEFT(length,gb) < 16)
1070
q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1075
* Process subpacket 9, init quantized_coeffs with data from it
1078
* @param node pointer to node with packet
1080
static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1083
int i, j, k, n, ch, run, level, diff;
1085
init_get_bits(&gb, node->packet->data, node->packet->size*8);
1087
n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1089
for (i = 1; i < n; i++)
1090
for (ch=0; ch < q->nb_channels; ch++) {
1091
level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1092
q->quantized_coeffs[ch][i][0] = level;
1094
for (j = 0; j < (8 - 1); ) {
1095
run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1096
diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1098
for (k = 1; k <= run; k++)
1099
q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1106
for (ch = 0; ch < q->nb_channels; ch++)
1107
for (i = 0; i < 8; i++)
1108
q->quantized_coeffs[ch][0][i] = 0;
1113
* Process subpacket 10 if not null, else
1116
* @param node pointer to node with packet
1117
* @param length packet length in bits
1119
static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1123
init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1126
init_tone_level_dequantization(q, &gb, length);
1127
fill_tone_level_array(q, 1);
1129
fill_tone_level_array(q, 0);
1135
* Process subpacket 11
1138
* @param node pointer to node with packet
1139
* @param length packet length in bit
1141
static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1145
init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1147
int c = get_bits (&gb, 13);
1150
fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1151
q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1154
synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1159
* Process subpacket 12
1162
* @param node pointer to node with packet
1163
* @param length packet length in bits
1165
static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1169
init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1170
synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1174
* Process new subpackets for synthesis filter
1177
* @param list list with synthesis filter packets (list D)
1179
static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1181
QDM2SubPNode *nodes[4];
1183
nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1184
if (nodes[0] != NULL)
1185
process_subpacket_9(q, nodes[0]);
1187
nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1188
if (nodes[1] != NULL)
1189
process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1191
process_subpacket_10(q, NULL, 0);
1193
nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1194
if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1195
process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1197
process_subpacket_11(q, NULL, 0);
1199
nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1200
if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1201
process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1203
process_subpacket_12(q, NULL, 0);
1208
* Decode superblock, fill packet lists.
1212
static void qdm2_decode_super_block (QDM2Context *q)
1215
QDM2SubPacket header, *packet;
1216
int i, packet_bytes, sub_packet_size, sub_packets_D;
1217
unsigned int next_index = 0;
1219
memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1220
memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1221
memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1223
q->sub_packets_B = 0;
1226
average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1228
init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1229
qdm2_decode_sub_packet_header(&gb, &header);
1231
if (header.type < 2 || header.type >= 8) {
1233
av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1237
q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1238
packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1240
init_get_bits(&gb, header.data, header.size*8);
1242
if (header.type == 2 || header.type == 4 || header.type == 5) {
1243
int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1245
csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1249
av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1254
q->sub_packet_list_B[0].packet = NULL;
1255
q->sub_packet_list_D[0].packet = NULL;
1257
for (i = 0; i < 6; i++)
1258
if (--q->fft_level_exp[i] < 0)
1259
q->fft_level_exp[i] = 0;
1261
for (i = 0; packet_bytes > 0; i++) {
1264
q->sub_packet_list_A[i].next = NULL;
1267
q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1269
/* seek to next block */
1270
init_get_bits(&gb, header.data, header.size*8);
1271
skip_bits(&gb, next_index*8);
1273
if (next_index >= header.size)
1277
/* decode subpacket */
1278
packet = &q->sub_packets[i];
1279
qdm2_decode_sub_packet_header(&gb, packet);
1280
next_index = packet->size + get_bits_count(&gb) / 8;
1281
sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1283
if (packet->type == 0)
1286
if (sub_packet_size > packet_bytes) {
1287
if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1289
packet->size += packet_bytes - sub_packet_size;
1292
packet_bytes -= sub_packet_size;
1294
/* add subpacket to 'all subpackets' list */
1295
q->sub_packet_list_A[i].packet = packet;
1297
/* add subpacket to related list */
1298
if (packet->type == 8) {
1299
SAMPLES_NEEDED_2("packet type 8");
1301
} else if (packet->type >= 9 && packet->type <= 12) {
1302
/* packets for MPEG Audio like Synthesis Filter */
1303
QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1304
} else if (packet->type == 13) {
1305
for (j = 0; j < 6; j++)
1306
q->fft_level_exp[j] = get_bits(&gb, 6);
1307
} else if (packet->type == 14) {
1308
for (j = 0; j < 6; j++)
1309
q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1310
} else if (packet->type == 15) {
1311
SAMPLES_NEEDED_2("packet type 15")
1313
} else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1314
/* packets for FFT */
1315
QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1317
} // Packet bytes loop
1319
/* **************************************************************** */
1320
if (q->sub_packet_list_D[0].packet != NULL) {
1321
process_synthesis_subpackets(q, q->sub_packet_list_D);
1322
q->do_synth_filter = 1;
1323
} else if (q->do_synth_filter) {
1324
process_subpacket_10(q, NULL, 0);
1325
process_subpacket_11(q, NULL, 0);
1326
process_subpacket_12(q, NULL, 0);
1328
/* **************************************************************** */
1332
static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1333
int offset, int duration, int channel,
1336
if (q->fft_coefs_min_index[duration] < 0)
1337
q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1339
q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1340
q->fft_coefs[q->fft_coefs_index].channel = channel;
1341
q->fft_coefs[q->fft_coefs_index].offset = offset;
1342
q->fft_coefs[q->fft_coefs_index].exp = exp;
1343
q->fft_coefs[q->fft_coefs_index].phase = phase;
1344
q->fft_coefs_index++;
1348
static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1350
int channel, stereo, phase, exp;
1351
int local_int_4, local_int_8, stereo_phase, local_int_10;
1352
int local_int_14, stereo_exp, local_int_20, local_int_28;
1358
local_int_8 = (4 - duration);
1359
local_int_10 = 1 << (q->group_order - duration - 1);
1363
if (q->superblocktype_2_3) {
1364
while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1367
local_int_4 += local_int_10;
1368
local_int_28 += (1 << local_int_8);
1370
local_int_4 += 8*local_int_10;
1371
local_int_28 += (8 << local_int_8);
1376
offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1377
while (offset >= (local_int_10 - 1)) {
1378
offset += (1 - (local_int_10 - 1));
1379
local_int_4 += local_int_10;
1380
local_int_28 += (1 << local_int_8);
1384
if (local_int_4 >= q->group_size)
1387
local_int_14 = (offset >> local_int_8);
1389
if (q->nb_channels > 1) {
1390
channel = get_bits1(gb);
1391
stereo = get_bits1(gb);
1397
exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1398
exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1399
exp = (exp < 0) ? 0 : exp;
1401
phase = get_bits(gb, 3);
1406
stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1407
stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1408
if (stereo_phase < 0)
1412
if (q->frequency_range > (local_int_14 + 1)) {
1413
int sub_packet = (local_int_20 + local_int_28);
1415
qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1417
qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1425
static void qdm2_decode_fft_packets (QDM2Context *q)
1427
int i, j, min, max, value, type, unknown_flag;
1430
if (q->sub_packet_list_B[0].packet == NULL)
1433
/* reset minimum indices for FFT coefficients */
1434
q->fft_coefs_index = 0;
1435
for (i=0; i < 5; i++)
1436
q->fft_coefs_min_index[i] = -1;
1438
/* process subpackets ordered by type, largest type first */
1439
for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1440
QDM2SubPacket *packet;
1442
/* find subpacket with largest type less than max */
1443
for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) {
1444
value = q->sub_packet_list_B[j].packet->type;
1445
if (value > min && value < max) {
1447
packet = q->sub_packet_list_B[j].packet;
1453
/* check for errors (?) */
1454
if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1457
/* decode FFT tones */
1458
init_get_bits (&gb, packet->data, packet->size*8);
1460
if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1465
type = packet->type;
1467
if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1468
int duration = q->sub_sampling + 5 - (type & 15);
1470
if (duration >= 0 && duration < 4)
1471
qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1472
} else if (type == 31) {
1473
for (j=0; j < 4; j++)
1474
qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1475
} else if (type == 46) {
1476
for (j=0; j < 6; j++)
1477
q->fft_level_exp[j] = get_bits(&gb, 6);
1478
for (j=0; j < 4; j++)
1479
qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1481
} // Loop on B packets
1483
/* calculate maximum indices for FFT coefficients */
1484
for (i = 0, j = -1; i < 5; i++)
1485
if (q->fft_coefs_min_index[i] >= 0) {
1487
q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1491
q->fft_coefs_max_index[j] = q->fft_coefs_index;
1495
static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1500
const double iscale = 2.0*M_PI / 512.0;
1502
tone->phase += tone->phase_shift;
1504
/* calculate current level (maximum amplitude) of tone */
1505
level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1506
c.im = level * sin(tone->phase*iscale);
1507
c.re = level * cos(tone->phase*iscale);
1509
/* generate FFT coefficients for tone */
1510
if (tone->duration >= 3 || tone->cutoff >= 3) {
1511
tone->samples_im[0] += c.im;
1512
tone->samples_re[0] += c.re;
1513
tone->samples_im[1] -= c.im;
1514
tone->samples_re[1] -= c.re;
1516
f[1] = -tone->table[4];
1517
f[0] = tone->table[3] - tone->table[0];
1518
f[2] = 1.0 - tone->table[2] - tone->table[3];
1519
f[3] = tone->table[1] + tone->table[4] - 1.0;
1520
f[4] = tone->table[0] - tone->table[1];
1521
f[5] = tone->table[2];
1522
for (i = 0; i < 2; i++) {
1523
tone->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i];
1524
tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1526
for (i = 0; i < 4; i++) {
1527
tone->samples_re[i] += c.re * f[i+2];
1528
tone->samples_im[i] += c.im * f[i+2];
1532
/* copy the tone if it has not yet died out */
1533
if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1534
memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1535
q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1540
static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1543
const double iscale = 0.25 * M_PI;
1545
for (ch = 0; ch < q->channels; ch++) {
1546
memset(q->fft.samples_im[ch], 0, q->fft_size * sizeof(float));
1547
memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float));
1551
/* apply FFT tones with duration 4 (1 FFT period) */
1552
if (q->fft_coefs_min_index[4] >= 0)
1553
for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1557
if (q->fft_coefs[i].sub_packet != sub_packet)
1560
ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1561
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];
1563
c.re = level * cos(q->fft_coefs[i].phase * iscale);
1564
c.im = level * sin(q->fft_coefs[i].phase * iscale);
1565
q->fft.samples_re[ch][q->fft_coefs[i].offset + 0] += c.re;
1566
q->fft.samples_im[ch][q->fft_coefs[i].offset + 0] += c.im;
1567
q->fft.samples_re[ch][q->fft_coefs[i].offset + 1] -= c.re;
1568
q->fft.samples_im[ch][q->fft_coefs[i].offset + 1] -= c.im;
1571
/* generate existing FFT tones */
1572
for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1573
qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1574
q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1577
/* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1578
for (i = 0; i < 4; i++)
1579
if (q->fft_coefs_min_index[i] >= 0) {
1580
for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1584
if (q->fft_coefs[j].sub_packet != sub_packet)
1588
offset = q->fft_coefs[j].offset >> four_i;
1589
ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1591
if (offset < q->frequency_range) {
1593
tone.cutoff = offset;
1595
tone.cutoff = (offset >= 60) ? 3 : 2;
1597
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];
1598
tone.samples_im = &q->fft.samples_im[ch][offset];
1599
tone.samples_re = &q->fft.samples_re[ch][offset];
1600
tone.table = (float*)fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1601
tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1602
tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1604
tone.time_index = 0;
1606
qdm2_fft_generate_tone(q, &tone);
1609
q->fft_coefs_min_index[i] = j;
1614
static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1616
const int n = 1 << (q->fft_order - 1);
1617
const int n2 = n >> 1;
1618
const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f;
1619
float c, s, f0, f1, f2, f3;
1622
/* prerotation (or something like that) */
1623
for (i=1; i < n2; i++) {
1625
c = q->exptab[i].re;
1626
s = -q->exptab[i].im;
1627
f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain;
1628
f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain;
1629
f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain;
1630
f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain;
1631
q->fft.complex[i].re = s * f0 - c * f1 + f2;
1632
q->fft.complex[i].im = c * f0 + s * f1 + f3;
1633
q->fft.complex[j].re = -s * f0 + c * f1 + f2;
1634
q->fft.complex[j].im = c * f0 + s * f1 - f3;
1637
q->fft.complex[ 0].re = q->fft.samples_re[channel][ 0] * gain * 2.0;
1638
q->fft.complex[ 0].im = q->fft.samples_re[channel][ 0] * gain * 2.0;
1639
q->fft.complex[n2].re = q->fft.samples_re[channel][n2] * gain * 2.0;
1640
q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0;
1642
ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex);
1643
ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex);
1644
/* add samples to output buffer */
1645
for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1646
q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex)[i];
1652
* @param index subpacket number
1654
static void qdm2_synthesis_filter (QDM2Context *q, int index)
1656
OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
1657
int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1659
/* copy sb_samples */
1660
sb_used = QDM2_SB_USED(q->sub_sampling);
1662
for (ch = 0; ch < q->channels; ch++)
1663
for (i = 0; i < 8; i++)
1664
for (k=sb_used; k < SBLIMIT; k++)
1665
q->sb_samples[ch][(8 * index) + i][k] = 0;
1667
for (ch = 0; ch < q->nb_channels; ch++) {
1668
OUT_INT *samples_ptr = samples + ch;
1670
for (i = 0; i < 8; i++) {
1671
ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1672
mpa_window, &dither_state,
1673
samples_ptr, q->nb_channels,
1674
q->sb_samples[ch][(8 * index) + i]);
1675
samples_ptr += 32 * q->nb_channels;
1679
/* add samples to output buffer */
1680
sub_sampling = (4 >> q->sub_sampling);
1682
for (ch = 0; ch < q->channels; ch++)
1683
for (i = 0; i < q->frame_size; i++)
1684
q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
1689
* Init static data (does not depend on specific file)
1693
static void qdm2_init(QDM2Context *q) {
1694
static int inited = 0;
1701
ff_mpa_synth_init(mpa_window);
1702
softclip_table_init();
1704
init_noise_samples();
1706
av_log(NULL, AV_LOG_DEBUG, "init done\n");
1711
static void dump_context(QDM2Context *q)
1714
#define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1715
PRINT("compressed_data",q->compressed_data);
1716
PRINT("compressed_size",q->compressed_size);
1717
PRINT("frame_size",q->frame_size);
1718
PRINT("checksum_size",q->checksum_size);
1719
PRINT("channels",q->channels);
1720
PRINT("nb_channels",q->nb_channels);
1721
PRINT("fft_frame_size",q->fft_frame_size);
1722
PRINT("fft_size",q->fft_size);
1723
PRINT("sub_sampling",q->sub_sampling);
1724
PRINT("fft_order",q->fft_order);
1725
PRINT("group_order",q->group_order);
1726
PRINT("group_size",q->group_size);
1727
PRINT("sub_packet",q->sub_packet);
1728
PRINT("frequency_range",q->frequency_range);
1729
PRINT("has_errors",q->has_errors);
1730
PRINT("fft_tone_end",q->fft_tone_end);
1731
PRINT("fft_tone_start",q->fft_tone_start);
1732
PRINT("fft_coefs_index",q->fft_coefs_index);
1733
PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1734
PRINT("cm_table_select",q->cm_table_select);
1735
PRINT("noise_idx",q->noise_idx);
1737
for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1739
FFTTone *t = &q->fft_tones[i];
1741
av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1742
av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1743
// PRINT(" level", t->level);
1744
PRINT(" phase", t->phase);
1745
PRINT(" phase_shift", t->phase_shift);
1746
PRINT(" duration", t->duration);
1747
PRINT(" samples_im", t->samples_im);
1748
PRINT(" samples_re", t->samples_re);
1749
PRINT(" table", t->table);
1757
* Init parameters from codec extradata
1759
static int qdm2_decode_init(AVCodecContext *avctx)
1761
QDM2Context *s = avctx->priv_data;
1764
int tmp_val, tmp, size;
1768
/* extradata parsing
1777
32 size (including this field)
1779
32 type (=QDM2 or QDMC)
1781
32 size (including this field, in bytes)
1782
32 tag (=QDCA) // maybe mandatory parameters
1785
32 samplerate (=44100)
1787
32 block size (=4096)
1788
32 frame size (=256) (for one channel)
1789
32 packet size (=1300)
1791
32 size (including this field, in bytes)
1792
32 tag (=QDCP) // maybe some tuneable parameters
1802
if (!avctx->extradata || (avctx->extradata_size < 48)) {
1803
av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1807
extradata = avctx->extradata;
1808
extradata_size = avctx->extradata_size;
1810
while (extradata_size > 7) {
1811
if (!memcmp(extradata, "frmaQDM", 7))
1817
if (extradata_size < 12) {
1818
av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1823
if (memcmp(extradata, "frmaQDM", 7)) {
1824
av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1828
if (extradata[7] == 'C') {
1830
av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1835
extradata_size -= 8;
1837
size = BE_32(extradata);
1839
if(size > extradata_size){
1840
av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1841
extradata_size, size);
1846
av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1847
if (BE_32(extradata) != MKBETAG('Q','D','C','A')) {
1848
av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1854
avctx->channels = s->nb_channels = s->channels = BE_32(extradata);
1857
avctx->sample_rate = BE_32(extradata);
1860
avctx->bit_rate = BE_32(extradata);
1863
s->group_size = BE_32(extradata);
1866
s->fft_size = BE_32(extradata);
1869
s->checksum_size = BE_32(extradata);
1872
s->fft_order = av_log2(s->fft_size) + 1;
1873
s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1875
// something like max decodable tones
1876
s->group_order = av_log2(s->group_size) + 1;
1877
s->frame_size = s->group_size / 16; // 16 iterations per super block
1879
s->sub_sampling = s->fft_order - 7;
1880
s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1882
switch ((s->sub_sampling * 2 + s->channels - 1)) {
1883
case 0: tmp = 40; break;
1884
case 1: tmp = 48; break;
1885
case 2: tmp = 56; break;
1886
case 3: tmp = 72; break;
1887
case 4: tmp = 80; break;
1888
case 5: tmp = 100;break;
1889
default: tmp=s->sub_sampling; break;
1892
if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1893
if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1894
if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1895
if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1896
s->cm_table_select = tmp_val;
1898
if (s->sub_sampling == 0)
1901
tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1908
s->coeff_per_sb_select = 0;
1909
else if (tmp <= 16000)
1910
s->coeff_per_sb_select = 1;
1912
s->coeff_per_sb_select = 2;
1914
// Fail on unknown fft order, if it's > 9 it can overflow s->exptab[]
1915
if ((s->fft_order < 7) || (s->fft_order > 9)) {
1916
av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1920
ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1);
1922
for (i = 1; i < (1 << (s->fft_order - 2)); i++) {
1923
alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1));
1924
s->exptab[i].re = cos(alpha);
1925
s->exptab[i].im = sin(alpha);
1935
static int qdm2_decode_close(AVCodecContext *avctx)
1937
QDM2Context *s = avctx->priv_data;
1939
ff_fft_end(&s->fft_ctx);
1945
static void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out)
1948
const int frame_size = (q->frame_size * q->channels);
1950
/* select input buffer */
1951
q->compressed_data = in;
1952
q->compressed_size = q->checksum_size;
1956
/* copy old block, clear new block of output samples */
1957
memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1958
memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1960
/* decode block of QDM2 compressed data */
1961
if (q->sub_packet == 0) {
1962
q->has_errors = 0; // zero it for a new super block
1963
av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1964
qdm2_decode_super_block(q);
1967
/* parse subpackets */
1968
if (!q->has_errors) {
1969
if (q->sub_packet == 2)
1970
qdm2_decode_fft_packets(q);
1972
qdm2_fft_tone_synthesizer(q, q->sub_packet);
1975
/* sound synthesis stage 1 (FFT) */
1976
for (ch = 0; ch < q->channels; ch++) {
1977
qdm2_calculate_fft(q, ch, q->sub_packet);
1979
if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1980
SAMPLES_NEEDED_2("has errors, and C list is not empty")
1985
/* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1986
if (!q->has_errors && q->do_synth_filter)
1987
qdm2_synthesis_filter(q, q->sub_packet);
1989
q->sub_packet = (q->sub_packet + 1) % 16;
1991
/* clip and convert output float[] to 16bit signed samples */
1992
for (i = 0; i < frame_size; i++) {
1993
int value = (int)q->output_buffer[i];
1995
if (value > SOFTCLIP_THRESHOLD)
1996
value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1997
else if (value < -SOFTCLIP_THRESHOLD)
1998
value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
2005
static int qdm2_decode_frame(AVCodecContext *avctx,
2006
void *data, int *data_size,
2007
uint8_t *buf, int buf_size)
2009
QDM2Context *s = avctx->priv_data;
2013
if(buf_size < s->checksum_size)
2016
*data_size = s->channels * s->frame_size * sizeof(int16_t);
2018
av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
2019
buf_size, buf, s->checksum_size, data, *data_size);
2021
qdm2_decode(s, buf, data);
2023
// reading only when next superblock found
2024
if (s->sub_packet == 0) {
2025
return s->checksum_size;
2031
AVCodec qdm2_decoder =
2034
.type = CODEC_TYPE_AUDIO,
2035
.id = CODEC_ID_QDM2,
2036
.priv_data_size = sizeof(QDM2Context),
2037
.init = qdm2_decode_init,
2038
.close = qdm2_decode_close,
2039
.decode = qdm2_decode_frame,
added
removed
avidemux/ADM_lavcodec/qdm2.c
