ffmpeg / libavcodec / qdm2.c @ 9106a698
History | View | Annotate | Download (66.1 KB)
| 1 |
/*
|
|---|---|
| 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
|
| 7 |
*
|
| 8 |
* This file is part of FFmpeg.
|
| 9 |
*
|
| 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.
|
| 14 |
*
|
| 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.
|
| 19 |
*
|
| 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
|
| 23 |
*/
|
| 24 |
|
| 25 |
/**
|
| 26 |
* @file libavcodec/qdm2.c
|
| 27 |
* QDM2 decoder
|
| 28 |
* @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
|
| 29 |
* The decoder is not perfect yet, there are still some distortions
|
| 30 |
* especially on files encoded with 16 or 8 subbands.
|
| 31 |
*/
|
| 32 |
|
| 33 |
#include <math.h> |
| 34 |
#include <stddef.h> |
| 35 |
#include <stdio.h> |
| 36 |
|
| 37 |
#define ALT_BITSTREAM_READER_LE
|
| 38 |
#include "avcodec.h" |
| 39 |
#include "get_bits.h" |
| 40 |
#include "dsputil.h" |
| 41 |
#include "mpegaudio.h" |
| 42 |
|
| 43 |
#include "qdm2data.h" |
| 44 |
|
| 45 |
#undef NDEBUG
|
| 46 |
#include <assert.h> |
| 47 |
|
| 48 |
|
| 49 |
#define SOFTCLIP_THRESHOLD 27600 |
| 50 |
#define HARDCLIP_THRESHOLD 35716 |
| 51 |
|
| 52 |
|
| 53 |
#define QDM2_LIST_ADD(list, size, packet) \
|
| 54 |
do { \
|
| 55 |
if (size > 0) { \ |
| 56 |
list[size - 1].next = &list[size]; \
|
| 57 |
} \ |
| 58 |
list[size].packet = packet; \ |
| 59 |
list[size].next = NULL; \
|
| 60 |
size++; \ |
| 61 |
} while(0) |
| 62 |
|
| 63 |
// Result is 8, 16 or 30
|
| 64 |
#define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling)) |
| 65 |
|
| 66 |
#define FIX_NOISE_IDX(noise_idx) \
|
| 67 |
if ((noise_idx) >= 3840) \ |
| 68 |
(noise_idx) -= 3840; \
|
| 69 |
|
| 70 |
#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
|
| 71 |
|
| 72 |
#define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
|
| 73 |
|
| 74 |
#define SAMPLES_NEEDED \
|
| 75 |
av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); |
| 76 |
|
| 77 |
#define SAMPLES_NEEDED_2(why) \
|
| 78 |
av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); |
| 79 |
|
| 80 |
|
| 81 |
typedef int8_t sb_int8_array[2][30][64]; |
| 82 |
|
| 83 |
/**
|
| 84 |
* Subpacket
|
| 85 |
*/
|
| 86 |
typedef struct { |
| 87 |
int type; ///< subpacket type |
| 88 |
unsigned int size; ///< subpacket size |
| 89 |
const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) |
| 90 |
} QDM2SubPacket; |
| 91 |
|
| 92 |
/**
|
| 93 |
* A node in the subpacket list
|
| 94 |
*/
|
| 95 |
typedef struct QDM2SubPNode { |
| 96 |
QDM2SubPacket *packet; ///< packet
|
| 97 |
struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node |
| 98 |
} QDM2SubPNode; |
| 99 |
|
| 100 |
typedef struct { |
| 101 |
float re;
|
| 102 |
float im;
|
| 103 |
} QDM2Complex; |
| 104 |
|
| 105 |
typedef struct { |
| 106 |
float level;
|
| 107 |
QDM2Complex *complex;
|
| 108 |
const float *table; |
| 109 |
int phase;
|
| 110 |
int phase_shift;
|
| 111 |
int duration;
|
| 112 |
short time_index;
|
| 113 |
short cutoff;
|
| 114 |
} FFTTone; |
| 115 |
|
| 116 |
typedef struct { |
| 117 |
int16_t sub_packet; |
| 118 |
uint8_t channel; |
| 119 |
int16_t offset; |
| 120 |
int16_t exp; |
| 121 |
uint8_t phase; |
| 122 |
} FFTCoefficient; |
| 123 |
|
| 124 |
typedef struct { |
| 125 |
DECLARE_ALIGNED_16(QDM2Complex, complex[MPA_MAX_CHANNELS][256]); |
| 126 |
} QDM2FFT; |
| 127 |
|
| 128 |
/**
|
| 129 |
* QDM2 decoder context
|
| 130 |
*/
|
| 131 |
typedef struct { |
| 132 |
/// Parameters from codec header, do not change during playback
|
| 133 |
int nb_channels; ///< number of channels |
| 134 |
int channels; ///< number of channels |
| 135 |
int group_size; ///< size of frame group (16 frames per group) |
| 136 |
int fft_size; ///< size of FFT, in complex numbers |
| 137 |
int checksum_size; ///< size of data block, used also for checksum |
| 138 |
|
| 139 |
/// Parameters built from header parameters, do not change during playback
|
| 140 |
int group_order; ///< order of frame group |
| 141 |
int fft_order; ///< order of FFT (actually fftorder+1) |
| 142 |
int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im) |
| 143 |
int frame_size; ///< size of data frame |
| 144 |
int frequency_range;
|
| 145 |
int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ |
| 146 |
int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 |
| 147 |
int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) |
| 148 |
|
| 149 |
/// Packets and packet lists
|
| 150 |
QDM2SubPacket sub_packets[16]; ///< the packets themselves |
| 151 |
QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets |
| 152 |
QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list |
| 153 |
int sub_packets_B; ///< number of packets on 'B' list |
| 154 |
QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? |
| 155 |
QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets |
| 156 |
|
| 157 |
/// FFT and tones
|
| 158 |
FFTTone fft_tones[1000];
|
| 159 |
int fft_tone_start;
|
| 160 |
int fft_tone_end;
|
| 161 |
FFTCoefficient fft_coefs[1000];
|
| 162 |
int fft_coefs_index;
|
| 163 |
int fft_coefs_min_index[5]; |
| 164 |
int fft_coefs_max_index[5]; |
| 165 |
int fft_level_exp[6]; |
| 166 |
RDFTContext rdft_ctx; |
| 167 |
QDM2FFT fft; |
| 168 |
|
| 169 |
/// I/O data
|
| 170 |
const uint8_t *compressed_data;
|
| 171 |
int compressed_size;
|
| 172 |
float output_buffer[1024]; |
| 173 |
|
| 174 |
/// Synthesis filter
|
| 175 |
DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]); |
| 176 |
int synth_buf_offset[MPA_MAX_CHANNELS];
|
| 177 |
DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]);
|
| 178 |
|
| 179 |
/// Mixed temporary data used in decoding
|
| 180 |
float tone_level[MPA_MAX_CHANNELS][30][64]; |
| 181 |
int8_t coding_method[MPA_MAX_CHANNELS][30][64]; |
| 182 |
int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; |
| 183 |
int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; |
| 184 |
int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; |
| 185 |
int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; |
| 186 |
int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
|
| 187 |
int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; |
| 188 |
int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; |
| 189 |
|
| 190 |
// Flags
|
| 191 |
int has_errors; ///< packet has errors |
| 192 |
int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type |
| 193 |
int do_synth_filter; ///< used to perform or skip synthesis filter |
| 194 |
|
| 195 |
int sub_packet;
|
| 196 |
int noise_idx; ///< index for dithering noise table |
| 197 |
} QDM2Context; |
| 198 |
|
| 199 |
|
| 200 |
static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
|
| 201 |
|
| 202 |
static VLC vlc_tab_level;
|
| 203 |
static VLC vlc_tab_diff;
|
| 204 |
static VLC vlc_tab_run;
|
| 205 |
static VLC fft_level_exp_alt_vlc;
|
| 206 |
static VLC fft_level_exp_vlc;
|
| 207 |
static VLC fft_stereo_exp_vlc;
|
| 208 |
static VLC fft_stereo_phase_vlc;
|
| 209 |
static VLC vlc_tab_tone_level_idx_hi1;
|
| 210 |
static VLC vlc_tab_tone_level_idx_mid;
|
| 211 |
static VLC vlc_tab_tone_level_idx_hi2;
|
| 212 |
static VLC vlc_tab_type30;
|
| 213 |
static VLC vlc_tab_type34;
|
| 214 |
static VLC vlc_tab_fft_tone_offset[5]; |
| 215 |
|
| 216 |
static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1]; |
| 217 |
static float noise_table[4096]; |
| 218 |
static uint8_t random_dequant_index[256][5]; |
| 219 |
static uint8_t random_dequant_type24[128][3]; |
| 220 |
static float noise_samples[128]; |
| 221 |
|
| 222 |
static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]); |
| 223 |
|
| 224 |
|
| 225 |
static av_cold void softclip_table_init(void) { |
| 226 |
int i;
|
| 227 |
double dfl = SOFTCLIP_THRESHOLD - 32767; |
| 228 |
float delta = 1.0 / -dfl; |
| 229 |
for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++) |
| 230 |
softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF); |
| 231 |
} |
| 232 |
|
| 233 |
|
| 234 |
// random generated table
|
| 235 |
static av_cold void rnd_table_init(void) { |
| 236 |
int i,j;
|
| 237 |
uint32_t ldw,hdw; |
| 238 |
uint64_t tmp64_1; |
| 239 |
uint64_t random_seed = 0;
|
| 240 |
float delta = 1.0 / 16384.0; |
| 241 |
for(i = 0; i < 4096 ;i++) { |
| 242 |
random_seed = random_seed * 214013 + 2531011; |
| 243 |
noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3; |
| 244 |
} |
| 245 |
|
| 246 |
for (i = 0; i < 256 ;i++) { |
| 247 |
random_seed = 81;
|
| 248 |
ldw = i; |
| 249 |
for (j = 0; j < 5 ;j++) { |
| 250 |
random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
|
| 251 |
ldw = (uint32_t)ldw % (uint32_t)random_seed; |
| 252 |
tmp64_1 = (random_seed * 0x55555556);
|
| 253 |
hdw = (uint32_t)(tmp64_1 >> 32);
|
| 254 |
random_seed = (uint64_t)(hdw + (ldw >> 31));
|
| 255 |
} |
| 256 |
} |
| 257 |
for (i = 0; i < 128 ;i++) { |
| 258 |
random_seed = 25;
|
| 259 |
ldw = i; |
| 260 |
for (j = 0; j < 3 ;j++) { |
| 261 |
random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
|
| 262 |
ldw = (uint32_t)ldw % (uint32_t)random_seed; |
| 263 |
tmp64_1 = (random_seed * 0x66666667);
|
| 264 |
hdw = (uint32_t)(tmp64_1 >> 33);
|
| 265 |
random_seed = hdw + (ldw >> 31);
|
| 266 |
} |
| 267 |
} |
| 268 |
} |
| 269 |
|
| 270 |
|
| 271 |
static av_cold void init_noise_samples(void) { |
| 272 |
int i;
|
| 273 |
int random_seed = 0; |
| 274 |
float delta = 1.0 / 16384.0; |
| 275 |
for (i = 0; i < 128;i++) { |
| 276 |
random_seed = random_seed * 214013 + 2531011; |
| 277 |
noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0); |
| 278 |
} |
| 279 |
} |
| 280 |
|
| 281 |
|
| 282 |
static av_cold void qdm2_init_vlc(void) |
| 283 |
{
|
| 284 |
init_vlc (&vlc_tab_level, 8, 24, |
| 285 |
vlc_tab_level_huffbits, 1, 1, |
| 286 |
vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 287 |
|
| 288 |
init_vlc (&vlc_tab_diff, 8, 37, |
| 289 |
vlc_tab_diff_huffbits, 1, 1, |
| 290 |
vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 291 |
|
| 292 |
init_vlc (&vlc_tab_run, 5, 6, |
| 293 |
vlc_tab_run_huffbits, 1, 1, |
| 294 |
vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 295 |
|
| 296 |
init_vlc (&fft_level_exp_alt_vlc, 8, 28, |
| 297 |
fft_level_exp_alt_huffbits, 1, 1, |
| 298 |
fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 299 |
|
| 300 |
init_vlc (&fft_level_exp_vlc, 8, 20, |
| 301 |
fft_level_exp_huffbits, 1, 1, |
| 302 |
fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 303 |
|
| 304 |
init_vlc (&fft_stereo_exp_vlc, 6, 7, |
| 305 |
fft_stereo_exp_huffbits, 1, 1, |
| 306 |
fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 307 |
|
| 308 |
init_vlc (&fft_stereo_phase_vlc, 6, 9, |
| 309 |
fft_stereo_phase_huffbits, 1, 1, |
| 310 |
fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 311 |
|
| 312 |
init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, |
| 313 |
vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, |
| 314 |
vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 315 |
|
| 316 |
init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, |
| 317 |
vlc_tab_tone_level_idx_mid_huffbits, 1, 1, |
| 318 |
vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 319 |
|
| 320 |
init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, |
| 321 |
vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, |
| 322 |
vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 323 |
|
| 324 |
init_vlc (&vlc_tab_type30, 6, 9, |
| 325 |
vlc_tab_type30_huffbits, 1, 1, |
| 326 |
vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 327 |
|
| 328 |
init_vlc (&vlc_tab_type34, 5, 10, |
| 329 |
vlc_tab_type34_huffbits, 1, 1, |
| 330 |
vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 331 |
|
| 332 |
init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, |
| 333 |
vlc_tab_fft_tone_offset_0_huffbits, 1, 1, |
| 334 |
vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 335 |
|
| 336 |
init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, |
| 337 |
vlc_tab_fft_tone_offset_1_huffbits, 1, 1, |
| 338 |
vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 339 |
|
| 340 |
init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, |
| 341 |
vlc_tab_fft_tone_offset_2_huffbits, 1, 1, |
| 342 |
vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 343 |
|
| 344 |
init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, |
| 345 |
vlc_tab_fft_tone_offset_3_huffbits, 1, 1, |
| 346 |
vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 347 |
|
| 348 |
init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, |
| 349 |
vlc_tab_fft_tone_offset_4_huffbits, 1, 1, |
| 350 |
vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); |
| 351 |
} |
| 352 |
|
| 353 |
|
| 354 |
/* for floating point to fixed point conversion */
|
| 355 |
static const float f2i_scale = (float) (1 << (FRAC_BITS - 15)); |
| 356 |
|
| 357 |
|
| 358 |
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) |
| 359 |
{
|
| 360 |
int value;
|
| 361 |
|
| 362 |
value = get_vlc2(gb, vlc->table, vlc->bits, depth); |
| 363 |
|
| 364 |
/* stage-2, 3 bits exponent escape sequence */
|
| 365 |
if (value-- == 0) |
| 366 |
value = get_bits (gb, get_bits (gb, 3) + 1); |
| 367 |
|
| 368 |
/* stage-3, optional */
|
| 369 |
if (flag) {
|
| 370 |
int tmp = vlc_stage3_values[value];
|
| 371 |
|
| 372 |
if ((value & ~3) > 0) |
| 373 |
tmp += get_bits (gb, (value >> 2));
|
| 374 |
value = tmp; |
| 375 |
} |
| 376 |
|
| 377 |
return value;
|
| 378 |
} |
| 379 |
|
| 380 |
|
| 381 |
static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) |
| 382 |
{
|
| 383 |
int value = qdm2_get_vlc (gb, vlc, 0, depth); |
| 384 |
|
| 385 |
return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); |
| 386 |
} |
| 387 |
|
| 388 |
|
| 389 |
/**
|
| 390 |
* QDM2 checksum
|
| 391 |
*
|
| 392 |
* @param data pointer to data to be checksum'ed
|
| 393 |
* @param length data length
|
| 394 |
* @param value checksum value
|
| 395 |
*
|
| 396 |
* @return 0 if checksum is OK
|
| 397 |
*/
|
| 398 |
static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) { |
| 399 |
int i;
|
| 400 |
|
| 401 |
for (i=0; i < length; i++) |
| 402 |
value -= data[i]; |
| 403 |
|
| 404 |
return (uint16_t)(value & 0xffff); |
| 405 |
} |
| 406 |
|
| 407 |
|
| 408 |
/**
|
| 409 |
* Fills a QDM2SubPacket structure with packet type, size, and data pointer.
|
| 410 |
*
|
| 411 |
* @param gb bitreader context
|
| 412 |
* @param sub_packet packet under analysis
|
| 413 |
*/
|
| 414 |
static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) |
| 415 |
{
|
| 416 |
sub_packet->type = get_bits (gb, 8);
|
| 417 |
|
| 418 |
if (sub_packet->type == 0) { |
| 419 |
sub_packet->size = 0;
|
| 420 |
sub_packet->data = NULL;
|
| 421 |
} else {
|
| 422 |
sub_packet->size = get_bits (gb, 8);
|
| 423 |
|
| 424 |
if (sub_packet->type & 0x80) { |
| 425 |
sub_packet->size <<= 8;
|
| 426 |
sub_packet->size |= get_bits (gb, 8);
|
| 427 |
sub_packet->type &= 0x7f;
|
| 428 |
} |
| 429 |
|
| 430 |
if (sub_packet->type == 0x7f) |
| 431 |
sub_packet->type |= (get_bits (gb, 8) << 8); |
| 432 |
|
| 433 |
sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data |
| 434 |
} |
| 435 |
|
| 436 |
av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", |
| 437 |
sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
|
| 438 |
} |
| 439 |
|
| 440 |
|
| 441 |
/**
|
| 442 |
* Return node pointer to first packet of requested type in list.
|
| 443 |
*
|
| 444 |
* @param list list of subpackets to be scanned
|
| 445 |
* @param type type of searched subpacket
|
| 446 |
* @return node pointer for subpacket if found, else NULL
|
| 447 |
*/
|
| 448 |
static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) |
| 449 |
{
|
| 450 |
while (list != NULL && list->packet != NULL) { |
| 451 |
if (list->packet->type == type)
|
| 452 |
return list;
|
| 453 |
list = list->next; |
| 454 |
} |
| 455 |
return NULL; |
| 456 |
} |
| 457 |
|
| 458 |
|
| 459 |
/**
|
| 460 |
* Replaces 8 elements with their average value.
|
| 461 |
* Called by qdm2_decode_superblock before starting subblock decoding.
|
| 462 |
*
|
| 463 |
* @param q context
|
| 464 |
*/
|
| 465 |
static void average_quantized_coeffs (QDM2Context *q) |
| 466 |
{
|
| 467 |
int i, j, n, ch, sum;
|
| 468 |
|
| 469 |
n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; |
| 470 |
|
| 471 |
for (ch = 0; ch < q->nb_channels; ch++) |
| 472 |
for (i = 0; i < n; i++) { |
| 473 |
sum = 0;
|
| 474 |
|
| 475 |
for (j = 0; j < 8; j++) |
| 476 |
sum += q->quantized_coeffs[ch][i][j]; |
| 477 |
|
| 478 |
sum /= 8;
|
| 479 |
if (sum > 0) |
| 480 |
sum--; |
| 481 |
|
| 482 |
for (j=0; j < 8; j++) |
| 483 |
q->quantized_coeffs[ch][i][j] = sum; |
| 484 |
} |
| 485 |
} |
| 486 |
|
| 487 |
|
| 488 |
/**
|
| 489 |
* Build subband samples with noise weighted by q->tone_level.
|
| 490 |
* Called by synthfilt_build_sb_samples.
|
| 491 |
*
|
| 492 |
* @param q context
|
| 493 |
* @param sb subband index
|
| 494 |
*/
|
| 495 |
static void build_sb_samples_from_noise (QDM2Context *q, int sb) |
| 496 |
{
|
| 497 |
int ch, j;
|
| 498 |
|
| 499 |
FIX_NOISE_IDX(q->noise_idx); |
| 500 |
|
| 501 |
if (!q->nb_channels)
|
| 502 |
return;
|
| 503 |
|
| 504 |
for (ch = 0; ch < q->nb_channels; ch++) |
| 505 |
for (j = 0; j < 64; j++) { |
| 506 |
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); |
| 507 |
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); |
| 508 |
} |
| 509 |
} |
| 510 |
|
| 511 |
|
| 512 |
/**
|
| 513 |
* Called while processing data from subpackets 11 and 12.
|
| 514 |
* Used after making changes to coding_method array.
|
| 515 |
*
|
| 516 |
* @param sb subband index
|
| 517 |
* @param channels number of channels
|
| 518 |
* @param coding_method q->coding_method[0][0][0]
|
| 519 |
*/
|
| 520 |
static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) |
| 521 |
{
|
| 522 |
int j,k;
|
| 523 |
int ch;
|
| 524 |
int run, case_val;
|
| 525 |
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}; |
| 526 |
|
| 527 |
for (ch = 0; ch < channels; ch++) { |
| 528 |
for (j = 0; j < 64; ) { |
| 529 |
if((coding_method[ch][sb][j] - 8) > 22) { |
| 530 |
run = 1;
|
| 531 |
case_val = 8;
|
| 532 |
} else {
|
| 533 |
switch (switchtable[coding_method[ch][sb][j]-8]) { |
| 534 |
case 0: run = 10; case_val = 10; break; |
| 535 |
case 1: run = 1; case_val = 16; break; |
| 536 |
case 2: run = 5; case_val = 24; break; |
| 537 |
case 3: run = 3; case_val = 30; break; |
| 538 |
case 4: run = 1; case_val = 30; break; |
| 539 |
case 5: run = 1; case_val = 8; break; |
| 540 |
default: run = 1; case_val = 8; break; |
| 541 |
} |
| 542 |
} |
| 543 |
for (k = 0; k < run; k++) |
| 544 |
if (j + k < 128) |
| 545 |
if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) |
| 546 |
if (k > 0) { |
| 547 |
SAMPLES_NEEDED |
| 548 |
//not debugged, almost never used
|
| 549 |
memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
|
| 550 |
memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); |
| 551 |
} |
| 552 |
j += run; |
| 553 |
} |
| 554 |
} |
| 555 |
} |
| 556 |
|
| 557 |
|
| 558 |
/**
|
| 559 |
* Related to synthesis filter
|
| 560 |
* Called by process_subpacket_10
|
| 561 |
*
|
| 562 |
* @param q context
|
| 563 |
* @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
|
| 564 |
*/
|
| 565 |
static void fill_tone_level_array (QDM2Context *q, int flag) |
| 566 |
{
|
| 567 |
int i, sb, ch, sb_used;
|
| 568 |
int tmp, tab;
|
| 569 |
|
| 570 |
// This should never happen
|
| 571 |
if (q->nb_channels <= 0) |
| 572 |
return;
|
| 573 |
|
| 574 |
for (ch = 0; ch < q->nb_channels; ch++) |
| 575 |
for (sb = 0; sb < 30; sb++) |
| 576 |
for (i = 0; i < 8; i++) { |
| 577 |
if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) |
| 578 |
tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ |
| 579 |
q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; |
| 580 |
else
|
| 581 |
tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; |
| 582 |
if(tmp < 0) |
| 583 |
tmp += 0xff;
|
| 584 |
q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; |
| 585 |
} |
| 586 |
|
| 587 |
sb_used = QDM2_SB_USED(q->sub_sampling); |
| 588 |
|
| 589 |
if ((q->superblocktype_2_3 != 0) && !flag) { |
| 590 |
for (sb = 0; sb < sb_used; sb++) |
| 591 |
for (ch = 0; ch < q->nb_channels; ch++) |
| 592 |
for (i = 0; i < 64; i++) { |
| 593 |
q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
|
| 594 |
if (q->tone_level_idx[ch][sb][i] < 0) |
| 595 |
q->tone_level[ch][sb][i] = 0;
|
| 596 |
else
|
| 597 |
q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; |
| 598 |
} |
| 599 |
} else {
|
| 600 |
tab = q->superblocktype_2_3 ? 0 : 1; |
| 601 |
for (sb = 0; sb < sb_used; sb++) { |
| 602 |
if ((sb >= 4) && (sb <= 23)) { |
| 603 |
for (ch = 0; ch < q->nb_channels; ch++) |
| 604 |
for (i = 0; i < 64; i++) { |
| 605 |
tmp = q->tone_level_idx_base[ch][sb][i / 8] -
|
| 606 |
q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - |
| 607 |
q->tone_level_idx_mid[ch][sb - 4][i / 8] - |
| 608 |
q->tone_level_idx_hi2[ch][sb - 4];
|
| 609 |
q->tone_level_idx[ch][sb][i] = tmp & 0xff;
|
| 610 |
if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) |
| 611 |
q->tone_level[ch][sb][i] = 0;
|
| 612 |
else
|
| 613 |
q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
|
| 614 |
} |
| 615 |
} else {
|
| 616 |
if (sb > 4) { |
| 617 |
for (ch = 0; ch < q->nb_channels; ch++) |
| 618 |
for (i = 0; i < 64; i++) { |
| 619 |
tmp = q->tone_level_idx_base[ch][sb][i / 8] -
|
| 620 |
q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - |
| 621 |
q->tone_level_idx_hi2[ch][sb - 4];
|
| 622 |
q->tone_level_idx[ch][sb][i] = tmp & 0xff;
|
| 623 |
if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) |
| 624 |
q->tone_level[ch][sb][i] = 0;
|
| 625 |
else
|
| 626 |
q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
|
| 627 |
} |
| 628 |
} else {
|
| 629 |
for (ch = 0; ch < q->nb_channels; ch++) |
| 630 |
for (i = 0; i < 64; i++) { |
| 631 |
tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
|
| 632 |
if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) |
| 633 |
q->tone_level[ch][sb][i] = 0;
|
| 634 |
else
|
| 635 |
q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
|
| 636 |
} |
| 637 |
} |
| 638 |
} |
| 639 |
} |
| 640 |
} |
| 641 |
|
| 642 |
return;
|
| 643 |
} |
| 644 |
|
| 645 |
|
| 646 |
/**
|
| 647 |
* Related to synthesis filter
|
| 648 |
* Called by process_subpacket_11
|
| 649 |
* c is built with data from subpacket 11
|
| 650 |
* Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
|
| 651 |
*
|
| 652 |
* @param tone_level_idx
|
| 653 |
* @param tone_level_idx_temp
|
| 654 |
* @param coding_method q->coding_method[0][0][0]
|
| 655 |
* @param nb_channels number of channels
|
| 656 |
* @param c coming from subpacket 11, passed as 8*c
|
| 657 |
* @param superblocktype_2_3 flag based on superblock packet type
|
| 658 |
* @param cm_table_select q->cm_table_select
|
| 659 |
*/
|
| 660 |
static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, |
| 661 |
sb_int8_array coding_method, int nb_channels,
|
| 662 |
int c, int superblocktype_2_3, int cm_table_select) |
| 663 |
{
|
| 664 |
int ch, sb, j;
|
| 665 |
int tmp, acc, esp_40, comp;
|
| 666 |
int add1, add2, add3, add4;
|
| 667 |
int64_t multres; |
| 668 |
|
| 669 |
// This should never happen
|
| 670 |
if (nb_channels <= 0) |
| 671 |
return;
|
| 672 |
|
| 673 |
if (!superblocktype_2_3) {
|
| 674 |
/* This case is untested, no samples available */
|
| 675 |
SAMPLES_NEEDED |
| 676 |
for (ch = 0; ch < nb_channels; ch++) |
| 677 |
for (sb = 0; sb < 30; sb++) { |
| 678 |
for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer |
| 679 |
add1 = tone_level_idx[ch][sb][j] - 10;
|
| 680 |
if (add1 < 0) |
| 681 |
add1 = 0;
|
| 682 |
add2 = add3 = add4 = 0;
|
| 683 |
if (sb > 1) { |
| 684 |
add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; |
| 685 |
if (add2 < 0) |
| 686 |
add2 = 0;
|
| 687 |
} |
| 688 |
if (sb > 0) { |
| 689 |
add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; |
| 690 |
if (add3 < 0) |
| 691 |
add3 = 0;
|
| 692 |
} |
| 693 |
if (sb < 29) { |
| 694 |
add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; |
| 695 |
if (add4 < 0) |
| 696 |
add4 = 0;
|
| 697 |
} |
| 698 |
tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; |
| 699 |
if (tmp < 0) |
| 700 |
tmp = 0;
|
| 701 |
tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; |
| 702 |
} |
| 703 |
tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; |
| 704 |
} |
| 705 |
acc = 0;
|
| 706 |
for (ch = 0; ch < nb_channels; ch++) |
| 707 |
for (sb = 0; sb < 30; sb++) |
| 708 |
for (j = 0; j < 64; j++) |
| 709 |
acc += tone_level_idx_temp[ch][sb][j]; |
| 710 |
if (acc)
|
| 711 |
tmp = c * 256 / (acc & 0xffff); |
| 712 |
multres = 0x66666667 * (acc * 10); |
| 713 |
esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); |
| 714 |
for (ch = 0; ch < nb_channels; ch++) |
| 715 |
for (sb = 0; sb < 30; sb++) |
| 716 |
for (j = 0; j < 64; j++) { |
| 717 |
comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
|
| 718 |
if (comp < 0) |
| 719 |
comp += 0xff;
|
| 720 |
comp /= 256; // signed shift |
| 721 |
switch(sb) {
|
| 722 |
case 0: |
| 723 |
if (comp < 30) |
| 724 |
comp = 30;
|
| 725 |
comp += 15;
|
| 726 |
break;
|
| 727 |
case 1: |
| 728 |
if (comp < 24) |
| 729 |
comp = 24;
|
| 730 |
comp += 10;
|
| 731 |
break;
|
| 732 |
case 2: |
| 733 |
case 3: |
| 734 |
case 4: |
| 735 |
if (comp < 16) |
| 736 |
comp = 16;
|
| 737 |
} |
| 738 |
if (comp <= 5) |
| 739 |
tmp = 0;
|
| 740 |
else if (comp <= 10) |
| 741 |
tmp = 10;
|
| 742 |
else if (comp <= 16) |
| 743 |
tmp = 16;
|
| 744 |
else if (comp <= 24) |
| 745 |
tmp = -1;
|
| 746 |
else
|
| 747 |
tmp = 0;
|
| 748 |
coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; |
| 749 |
} |
| 750 |
for (sb = 0; sb < 30; sb++) |
| 751 |
fix_coding_method_array(sb, nb_channels, coding_method); |
| 752 |
for (ch = 0; ch < nb_channels; ch++) |
| 753 |
for (sb = 0; sb < 30; sb++) |
| 754 |
for (j = 0; j < 64; j++) |
| 755 |
if (sb >= 10) { |
| 756 |
if (coding_method[ch][sb][j] < 10) |
| 757 |
coding_method[ch][sb][j] = 10;
|
| 758 |
} else {
|
| 759 |
if (sb >= 2) { |
| 760 |
if (coding_method[ch][sb][j] < 16) |
| 761 |
coding_method[ch][sb][j] = 16;
|
| 762 |
} else {
|
| 763 |
if (coding_method[ch][sb][j] < 30) |
| 764 |
coding_method[ch][sb][j] = 30;
|
| 765 |
} |
| 766 |
} |
| 767 |
} else { // superblocktype_2_3 != 0 |
| 768 |
for (ch = 0; ch < nb_channels; ch++) |
| 769 |
for (sb = 0; sb < 30; sb++) |
| 770 |
for (j = 0; j < 64; j++) |
| 771 |
coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; |
| 772 |
} |
| 773 |
|
| 774 |
return;
|
| 775 |
} |
| 776 |
|
| 777 |
|
| 778 |
/**
|
| 779 |
*
|
| 780 |
* Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
|
| 781 |
* Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
|
| 782 |
*
|
| 783 |
* @param q context
|
| 784 |
* @param gb bitreader context
|
| 785 |
* @param length packet length in bits
|
| 786 |
* @param sb_min lower subband processed (sb_min included)
|
| 787 |
* @param sb_max higher subband processed (sb_max excluded)
|
| 788 |
*/
|
| 789 |
static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) |
| 790 |
{
|
| 791 |
int sb, j, k, n, ch, run, channels;
|
| 792 |
int joined_stereo, zero_encoding, chs;
|
| 793 |
int type34_first;
|
| 794 |
float type34_div = 0; |
| 795 |
float type34_predictor;
|
| 796 |
float samples[10], sign_bits[16]; |
| 797 |
|
| 798 |
if (length == 0) { |
| 799 |
// If no data use noise
|
| 800 |
for (sb=sb_min; sb < sb_max; sb++)
|
| 801 |
build_sb_samples_from_noise (q, sb); |
| 802 |
|
| 803 |
return;
|
| 804 |
} |
| 805 |
|
| 806 |
for (sb = sb_min; sb < sb_max; sb++) {
|
| 807 |
FIX_NOISE_IDX(q->noise_idx); |
| 808 |
|
| 809 |
channels = q->nb_channels; |
| 810 |
|
| 811 |
if (q->nb_channels <= 1 || sb < 12) |
| 812 |
joined_stereo = 0;
|
| 813 |
else if (sb >= 24) |
| 814 |
joined_stereo = 1;
|
| 815 |
else
|
| 816 |
joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0; |
| 817 |
|
| 818 |
if (joined_stereo) {
|
| 819 |
if (BITS_LEFT(length,gb) >= 16) |
| 820 |
for (j = 0; j < 16; j++) |
| 821 |
sign_bits[j] = get_bits1 (gb); |
| 822 |
|
| 823 |
for (j = 0; j < 64; j++) |
| 824 |
if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) |
| 825 |
q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; |
| 826 |
|
| 827 |
fix_coding_method_array(sb, q->nb_channels, q->coding_method); |
| 828 |
channels = 1;
|
| 829 |
} |
| 830 |
|
| 831 |
for (ch = 0; ch < channels; ch++) { |
| 832 |
zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; |
| 833 |
type34_predictor = 0.0; |
| 834 |
type34_first = 1;
|
| 835 |
|
| 836 |
for (j = 0; j < 128; ) { |
| 837 |
switch (q->coding_method[ch][sb][j / 2]) { |
| 838 |
case 8: |
| 839 |
if (BITS_LEFT(length,gb) >= 10) { |
| 840 |
if (zero_encoding) {
|
| 841 |
for (k = 0; k < 5; k++) { |
| 842 |
if ((j + 2 * k) >= 128) |
| 843 |
break;
|
| 844 |
samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; |
| 845 |
} |
| 846 |
} else {
|
| 847 |
n = get_bits(gb, 8);
|
| 848 |
for (k = 0; k < 5; k++) |
| 849 |
samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
|
| 850 |
} |
| 851 |
for (k = 0; k < 5; k++) |
| 852 |
samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); |
| 853 |
} else {
|
| 854 |
for (k = 0; k < 10; k++) |
| 855 |
samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); |
| 856 |
} |
| 857 |
run = 10;
|
| 858 |
break;
|
| 859 |
|
| 860 |
case 10: |
| 861 |
if (BITS_LEFT(length,gb) >= 1) { |
| 862 |
float f = 0.81; |
| 863 |
|
| 864 |
if (get_bits1(gb))
|
| 865 |
f = -f; |
| 866 |
f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; |
| 867 |
samples[0] = f;
|
| 868 |
} else {
|
| 869 |
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
|
| 870 |
} |
| 871 |
run = 1;
|
| 872 |
break;
|
| 873 |
|
| 874 |
case 16: |
| 875 |
if (BITS_LEFT(length,gb) >= 10) { |
| 876 |
if (zero_encoding) {
|
| 877 |
for (k = 0; k < 5; k++) { |
| 878 |
if ((j + k) >= 128) |
| 879 |
break;
|
| 880 |
samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; |
| 881 |
} |
| 882 |
} else {
|
| 883 |
n = get_bits (gb, 8);
|
| 884 |
for (k = 0; k < 5; k++) |
| 885 |
samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; |
| 886 |
} |
| 887 |
} else {
|
| 888 |
for (k = 0; k < 5; k++) |
| 889 |
samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); |
| 890 |
} |
| 891 |
run = 5;
|
| 892 |
break;
|
| 893 |
|
| 894 |
case 24: |
| 895 |
if (BITS_LEFT(length,gb) >= 7) { |
| 896 |
n = get_bits(gb, 7);
|
| 897 |
for (k = 0; k < 3; k++) |
| 898 |
samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; |
| 899 |
} else {
|
| 900 |
for (k = 0; k < 3; k++) |
| 901 |
samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); |
| 902 |
} |
| 903 |
run = 3;
|
| 904 |
break;
|
| 905 |
|
| 906 |
case 30: |
| 907 |
if (BITS_LEFT(length,gb) >= 4) |
| 908 |
samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)]; |
| 909 |
else
|
| 910 |
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
|
| 911 |
|
| 912 |
run = 1;
|
| 913 |
break;
|
| 914 |
|
| 915 |
case 34: |
| 916 |
if (BITS_LEFT(length,gb) >= 7) { |
| 917 |
if (type34_first) {
|
| 918 |
type34_div = (float)(1 << get_bits(gb, 2)); |
| 919 |
samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; |
| 920 |
type34_predictor = samples[0];
|
| 921 |
type34_first = 0;
|
| 922 |
} else {
|
| 923 |
samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor; |
| 924 |
type34_predictor = samples[0];
|
| 925 |
} |
| 926 |
} else {
|
| 927 |
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
|
| 928 |
} |
| 929 |
run = 1;
|
| 930 |
break;
|
| 931 |
|
| 932 |
default:
|
| 933 |
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
|
| 934 |
run = 1;
|
| 935 |
break;
|
| 936 |
} |
| 937 |
|
| 938 |
if (joined_stereo) {
|
| 939 |
float tmp[10][MPA_MAX_CHANNELS]; |
| 940 |
|
| 941 |
for (k = 0; k < run; k++) { |
| 942 |
tmp[k][0] = samples[k];
|
| 943 |
tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; |
| 944 |
} |
| 945 |
for (chs = 0; chs < q->nb_channels; chs++) |
| 946 |
for (k = 0; k < run; k++) |
| 947 |
if ((j + k) < 128) |
| 948 |
q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5); |
| 949 |
} else {
|
| 950 |
for (k = 0; k < run; k++) |
| 951 |
if ((j + k) < 128) |
| 952 |
q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5); |
| 953 |
} |
| 954 |
|
| 955 |
j += run; |
| 956 |
} // j loop
|
| 957 |
} // channel loop
|
| 958 |
} // subband loop
|
| 959 |
} |
| 960 |
|
| 961 |
|
| 962 |
/**
|
| 963 |
* Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
|
| 964 |
* This is similar to process_subpacket_9, but for a single channel and for element [0]
|
| 965 |
* same VLC tables as process_subpacket_9 are used.
|
| 966 |
*
|
| 967 |
* @param q context
|
| 968 |
* @param quantized_coeffs pointer to quantized_coeffs[ch][0]
|
| 969 |
* @param gb bitreader context
|
| 970 |
* @param length packet length in bits
|
| 971 |
*/
|
| 972 |
static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) |
| 973 |
{
|
| 974 |
int i, k, run, level, diff;
|
| 975 |
|
| 976 |
if (BITS_LEFT(length,gb) < 16) |
| 977 |
return;
|
| 978 |
level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); |
| 979 |
|
| 980 |
quantized_coeffs[0] = level;
|
| 981 |
|
| 982 |
for (i = 0; i < 7; ) { |
| 983 |
if (BITS_LEFT(length,gb) < 16) |
| 984 |
break;
|
| 985 |
run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; |
| 986 |
|
| 987 |
if (BITS_LEFT(length,gb) < 16) |
| 988 |
break;
|
| 989 |
diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
|
| 990 |
|
| 991 |
for (k = 1; k <= run; k++) |
| 992 |
quantized_coeffs[i + k] = (level + ((k * diff) / run)); |
| 993 |
|
| 994 |
level += diff; |
| 995 |
i += run; |
| 996 |
} |
| 997 |
} |
| 998 |
|
| 999 |
|
| 1000 |
/**
|
| 1001 |
* Related to synthesis filter, process data from packet 10
|
| 1002 |
* Init part of quantized_coeffs via function init_quantized_coeffs_elem0
|
| 1003 |
* Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
|
| 1004 |
*
|
| 1005 |
* @param q context
|
| 1006 |
* @param gb bitreader context
|
| 1007 |
* @param length packet length in bits
|
| 1008 |
*/
|
| 1009 |
static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) |
| 1010 |
{
|
| 1011 |
int sb, j, k, n, ch;
|
| 1012 |
|
| 1013 |
for (ch = 0; ch < q->nb_channels; ch++) { |
| 1014 |
init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
|
| 1015 |
|
| 1016 |
if (BITS_LEFT(length,gb) < 16) { |
| 1017 |
memset(q->quantized_coeffs[ch][0], 0, 8); |
| 1018 |
break;
|
| 1019 |
} |
| 1020 |
} |
| 1021 |
|
| 1022 |
n = q->sub_sampling + 1;
|
| 1023 |
|
| 1024 |
for (sb = 0; sb < n; sb++) |
| 1025 |
for (ch = 0; ch < q->nb_channels; ch++) |
| 1026 |
for (j = 0; j < 8; j++) { |
| 1027 |
if (BITS_LEFT(length,gb) < 1) |
| 1028 |
break;
|
| 1029 |
if (get_bits1(gb)) {
|
| 1030 |
for (k=0; k < 8; k++) { |
| 1031 |
if (BITS_LEFT(length,gb) < 16) |
| 1032 |
break;
|
| 1033 |
q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); |
| 1034 |
} |
| 1035 |
} else {
|
| 1036 |
for (k=0; k < 8; k++) |
| 1037 |
q->tone_level_idx_hi1[ch][sb][j][k] = 0;
|
| 1038 |
} |
| 1039 |
} |
| 1040 |
|
| 1041 |
n = QDM2_SB_USED(q->sub_sampling) - 4;
|
| 1042 |
|
| 1043 |
for (sb = 0; sb < n; sb++) |
| 1044 |
for (ch = 0; ch < q->nb_channels; ch++) { |
| 1045 |
if (BITS_LEFT(length,gb) < 16) |
| 1046 |
break;
|
| 1047 |
q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); |
| 1048 |
if (sb > 19) |
| 1049 |
q->tone_level_idx_hi2[ch][sb] -= 16;
|
| 1050 |
else
|
| 1051 |
for (j = 0; j < 8; j++) |
| 1052 |
q->tone_level_idx_mid[ch][sb][j] = -16;
|
| 1053 |
} |
| 1054 |
|
| 1055 |
n = QDM2_SB_USED(q->sub_sampling) - 5;
|
| 1056 |
|
| 1057 |
for (sb = 0; sb < n; sb++) |
| 1058 |
for (ch = 0; ch < q->nb_channels; ch++) |
| 1059 |
for (j = 0; j < 8; j++) { |
| 1060 |
if (BITS_LEFT(length,gb) < 16) |
| 1061 |
break;
|
| 1062 |
q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; |
| 1063 |
} |
| 1064 |
} |
| 1065 |
|
| 1066 |
/**
|
| 1067 |
* Process subpacket 9, init quantized_coeffs with data from it
|
| 1068 |
*
|
| 1069 |
* @param q context
|
| 1070 |
* @param node pointer to node with packet
|
| 1071 |
*/
|
| 1072 |
static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) |
| 1073 |
{
|
| 1074 |
GetBitContext gb; |
| 1075 |
int i, j, k, n, ch, run, level, diff;
|
| 1076 |
|
| 1077 |
init_get_bits(&gb, node->packet->data, node->packet->size*8);
|
| 1078 |
|
| 1079 |
n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function |
| 1080 |
|
| 1081 |
for (i = 1; i < n; i++) |
| 1082 |
for (ch=0; ch < q->nb_channels; ch++) { |
| 1083 |
level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); |
| 1084 |
q->quantized_coeffs[ch][i][0] = level;
|
| 1085 |
|
| 1086 |
for (j = 0; j < (8 - 1); ) { |
| 1087 |
run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; |
| 1088 |
diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
|
| 1089 |
|
| 1090 |
for (k = 1; k <= run; k++) |
| 1091 |
q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); |
| 1092 |
|
| 1093 |
level += diff; |
| 1094 |
j += run; |
| 1095 |
} |
| 1096 |
} |
| 1097 |
|
| 1098 |
for (ch = 0; ch < q->nb_channels; ch++) |
| 1099 |
for (i = 0; i < 8; i++) |
| 1100 |
q->quantized_coeffs[ch][0][i] = 0; |
| 1101 |
} |
| 1102 |
|
| 1103 |
|
| 1104 |
/**
|
| 1105 |
* Process subpacket 10 if not null, else
|
| 1106 |
*
|
| 1107 |
* @param q context
|
| 1108 |
* @param node pointer to node with packet
|
| 1109 |
* @param length packet length in bits
|
| 1110 |
*/
|
| 1111 |
static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) |
| 1112 |
{
|
| 1113 |
GetBitContext gb; |
| 1114 |
|
| 1115 |
init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
| 1116 |
|
| 1117 |
if (length != 0) { |
| 1118 |
init_tone_level_dequantization(q, &gb, length); |
| 1119 |
fill_tone_level_array(q, 1);
|
| 1120 |
} else {
|
| 1121 |
fill_tone_level_array(q, 0);
|
| 1122 |
} |
| 1123 |
} |
| 1124 |
|
| 1125 |
|
| 1126 |
/**
|
| 1127 |
* Process subpacket 11
|
| 1128 |
*
|
| 1129 |
* @param q context
|
| 1130 |
* @param node pointer to node with packet
|
| 1131 |
* @param length packet length in bit
|
| 1132 |
*/
|
| 1133 |
static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length) |
| 1134 |
{
|
| 1135 |
GetBitContext gb; |
| 1136 |
|
| 1137 |
init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
| 1138 |
if (length >= 32) { |
| 1139 |
int c = get_bits (&gb, 13); |
| 1140 |
|
| 1141 |
if (c > 3) |
| 1142 |
fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, |
| 1143 |
q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
|
| 1144 |
} |
| 1145 |
|
| 1146 |
synthfilt_build_sb_samples(q, &gb, length, 0, 8); |
| 1147 |
} |
| 1148 |
|
| 1149 |
|
| 1150 |
/**
|
| 1151 |
* Process subpacket 12
|
| 1152 |
*
|
| 1153 |
* @param q context
|
| 1154 |
* @param node pointer to node with packet
|
| 1155 |
* @param length packet length in bits
|
| 1156 |
*/
|
| 1157 |
static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) |
| 1158 |
{
|
| 1159 |
GetBitContext gb; |
| 1160 |
|
| 1161 |
init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
| 1162 |
synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
|
| 1163 |
} |
| 1164 |
|
| 1165 |
/*
|
| 1166 |
* Process new subpackets for synthesis filter
|
| 1167 |
*
|
| 1168 |
* @param q context
|
| 1169 |
* @param list list with synthesis filter packets (list D)
|
| 1170 |
*/
|
| 1171 |
static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) |
| 1172 |
{
|
| 1173 |
QDM2SubPNode *nodes[4];
|
| 1174 |
|
| 1175 |
nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); |
| 1176 |
if (nodes[0] != NULL) |
| 1177 |
process_subpacket_9(q, nodes[0]);
|
| 1178 |
|
| 1179 |
nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); |
| 1180 |
if (nodes[1] != NULL) |
| 1181 |
process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3); |
| 1182 |
else
|
| 1183 |
process_subpacket_10(q, NULL, 0); |
| 1184 |
|
| 1185 |
nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); |
| 1186 |
if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) |
| 1187 |
process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3)); |
| 1188 |
else
|
| 1189 |
process_subpacket_11(q, NULL, 0); |
| 1190 |
|
| 1191 |
nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); |
| 1192 |
if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) |
| 1193 |
process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3)); |
| 1194 |
else
|
| 1195 |
process_subpacket_12(q, NULL, 0); |
| 1196 |
} |
| 1197 |
|
| 1198 |
|
| 1199 |
/*
|
| 1200 |
* Decode superblock, fill packet lists.
|
| 1201 |
*
|
| 1202 |
* @param q context
|
| 1203 |
*/
|
| 1204 |
static void qdm2_decode_super_block (QDM2Context *q) |
| 1205 |
{
|
| 1206 |
GetBitContext gb; |
| 1207 |
QDM2SubPacket header, *packet; |
| 1208 |
int i, packet_bytes, sub_packet_size, sub_packets_D;
|
| 1209 |
unsigned int next_index = 0; |
| 1210 |
|
| 1211 |
memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); |
| 1212 |
memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); |
| 1213 |
memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); |
| 1214 |
|
| 1215 |
q->sub_packets_B = 0;
|
| 1216 |
sub_packets_D = 0;
|
| 1217 |
|
| 1218 |
average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
|
| 1219 |
|
| 1220 |
init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
|
| 1221 |
qdm2_decode_sub_packet_header(&gb, &header); |
| 1222 |
|
| 1223 |
if (header.type < 2 || header.type >= 8) { |
| 1224 |
q->has_errors = 1;
|
| 1225 |
av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); |
| 1226 |
return;
|
| 1227 |
} |
| 1228 |
|
| 1229 |
q->superblocktype_2_3 = (header.type == 2 || header.type == 3); |
| 1230 |
packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
|
| 1231 |
|
| 1232 |
init_get_bits(&gb, header.data, header.size*8);
|
| 1233 |
|
| 1234 |
if (header.type == 2 || header.type == 4 || header.type == 5) { |
| 1235 |
int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8); |
| 1236 |
|
| 1237 |
csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); |
| 1238 |
|
| 1239 |
if (csum != 0) { |
| 1240 |
q->has_errors = 1;
|
| 1241 |
av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); |
| 1242 |
return;
|
| 1243 |
} |
| 1244 |
} |
| 1245 |
|
| 1246 |
q->sub_packet_list_B[0].packet = NULL; |
| 1247 |
q->sub_packet_list_D[0].packet = NULL; |
| 1248 |
|
| 1249 |
for (i = 0; i < 6; i++) |
| 1250 |
if (--q->fft_level_exp[i] < 0) |
| 1251 |
q->fft_level_exp[i] = 0;
|
| 1252 |
|
| 1253 |
for (i = 0; packet_bytes > 0; i++) { |
| 1254 |
int j;
|
| 1255 |
|
| 1256 |
q->sub_packet_list_A[i].next = NULL;
|
| 1257 |
|
| 1258 |
if (i > 0) { |
| 1259 |
q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
|
| 1260 |
|
| 1261 |
/* seek to next block */
|
| 1262 |
init_get_bits(&gb, header.data, header.size*8);
|
| 1263 |
skip_bits(&gb, next_index*8);
|
| 1264 |
|
| 1265 |
if (next_index >= header.size)
|
| 1266 |
break;
|
| 1267 |
} |
| 1268 |
|
| 1269 |
/* decode subpacket */
|
| 1270 |
packet = &q->sub_packets[i]; |
| 1271 |
qdm2_decode_sub_packet_header(&gb, packet); |
| 1272 |
next_index = packet->size + get_bits_count(&gb) / 8;
|
| 1273 |
sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; |
| 1274 |
|
| 1275 |
if (packet->type == 0) |
| 1276 |
break;
|
| 1277 |
|
| 1278 |
if (sub_packet_size > packet_bytes) {
|
| 1279 |
if (packet->type != 10 && packet->type != 11 && packet->type != 12) |
| 1280 |
break;
|
| 1281 |
packet->size += packet_bytes - sub_packet_size; |
| 1282 |
} |
| 1283 |
|
| 1284 |
packet_bytes -= sub_packet_size; |
| 1285 |
|
| 1286 |
/* add subpacket to 'all subpackets' list */
|
| 1287 |
q->sub_packet_list_A[i].packet = packet; |
| 1288 |
|
| 1289 |
/* add subpacket to related list */
|
| 1290 |
if (packet->type == 8) { |
| 1291 |
SAMPLES_NEEDED_2("packet type 8");
|
| 1292 |
return;
|
| 1293 |
} else if (packet->type >= 9 && packet->type <= 12) { |
| 1294 |
/* packets for MPEG Audio like Synthesis Filter */
|
| 1295 |
QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); |
| 1296 |
} else if (packet->type == 13) { |
| 1297 |
for (j = 0; j < 6; j++) |
| 1298 |
q->fft_level_exp[j] = get_bits(&gb, 6);
|
| 1299 |
} else if (packet->type == 14) { |
| 1300 |
for (j = 0; j < 6; j++) |
| 1301 |
q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); |
| 1302 |
} else if (packet->type == 15) { |
| 1303 |
SAMPLES_NEEDED_2("packet type 15")
|
| 1304 |
return;
|
| 1305 |
} else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { |
| 1306 |
/* packets for FFT */
|
| 1307 |
QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); |
| 1308 |
} |
| 1309 |
} // Packet bytes loop
|
| 1310 |
|
| 1311 |
/* **************************************************************** */
|
| 1312 |
if (q->sub_packet_list_D[0].packet != NULL) { |
| 1313 |
process_synthesis_subpackets(q, q->sub_packet_list_D); |
| 1314 |
q->do_synth_filter = 1;
|
| 1315 |
} else if (q->do_synth_filter) { |
| 1316 |
process_subpacket_10(q, NULL, 0); |
| 1317 |
process_subpacket_11(q, NULL, 0); |
| 1318 |
process_subpacket_12(q, NULL, 0); |
| 1319 |
} |
| 1320 |
/* **************************************************************** */
|
| 1321 |
} |
| 1322 |
|
| 1323 |
|
| 1324 |
static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, |
| 1325 |
int offset, int duration, int channel, |
| 1326 |
int exp, int phase) |
| 1327 |
{
|
| 1328 |
if (q->fft_coefs_min_index[duration] < 0) |
| 1329 |
q->fft_coefs_min_index[duration] = q->fft_coefs_index; |
| 1330 |
|
| 1331 |
q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); |
| 1332 |
q->fft_coefs[q->fft_coefs_index].channel = channel; |
| 1333 |
q->fft_coefs[q->fft_coefs_index].offset = offset; |
| 1334 |
q->fft_coefs[q->fft_coefs_index].exp = exp; |
| 1335 |
q->fft_coefs[q->fft_coefs_index].phase = phase; |
| 1336 |
q->fft_coefs_index++; |
| 1337 |
} |
| 1338 |
|
| 1339 |
|
| 1340 |
static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) |
| 1341 |
{
|
| 1342 |
int channel, stereo, phase, exp;
|
| 1343 |
int local_int_4, local_int_8, stereo_phase, local_int_10;
|
| 1344 |
int local_int_14, stereo_exp, local_int_20, local_int_28;
|
| 1345 |
int n, offset;
|
| 1346 |
|
| 1347 |
local_int_4 = 0;
|
| 1348 |
local_int_28 = 0;
|
| 1349 |
local_int_20 = 2;
|
| 1350 |
local_int_8 = (4 - duration);
|
| 1351 |
local_int_10 = 1 << (q->group_order - duration - 1); |
| 1352 |
offset = 1;
|
| 1353 |
|
| 1354 |
while (1) { |
| 1355 |
if (q->superblocktype_2_3) {
|
| 1356 |
while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { |
| 1357 |
offset = 1;
|
| 1358 |
if (n == 0) { |
| 1359 |
local_int_4 += local_int_10; |
| 1360 |
local_int_28 += (1 << local_int_8);
|
| 1361 |
} else {
|
| 1362 |
local_int_4 += 8*local_int_10;
|
| 1363 |
local_int_28 += (8 << local_int_8);
|
| 1364 |
} |
| 1365 |
} |
| 1366 |
offset += (n - 2);
|
| 1367 |
} else {
|
| 1368 |
offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); |
| 1369 |
while (offset >= (local_int_10 - 1)) { |
| 1370 |
offset += (1 - (local_int_10 - 1)); |
| 1371 |
local_int_4 += local_int_10; |
| 1372 |
local_int_28 += (1 << local_int_8);
|
| 1373 |
} |
| 1374 |
} |
| 1375 |
|
| 1376 |
if (local_int_4 >= q->group_size)
|
| 1377 |
return;
|
| 1378 |
|
| 1379 |
local_int_14 = (offset >> local_int_8); |
| 1380 |
|
| 1381 |
if (q->nb_channels > 1) { |
| 1382 |
channel = get_bits1(gb); |
| 1383 |
stereo = get_bits1(gb); |
| 1384 |
} else {
|
| 1385 |
channel = 0;
|
| 1386 |
stereo = 0;
|
| 1387 |
} |
| 1388 |
|
| 1389 |
exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); |
| 1390 |
exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; |
| 1391 |
exp = (exp < 0) ? 0 : exp; |
| 1392 |
|
| 1393 |
phase = get_bits(gb, 3);
|
| 1394 |
stereo_exp = 0;
|
| 1395 |
stereo_phase = 0;
|
| 1396 |
|
| 1397 |
if (stereo) {
|
| 1398 |
stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); |
| 1399 |
stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); |
| 1400 |
if (stereo_phase < 0) |
| 1401 |
stereo_phase += 8;
|
| 1402 |
} |
| 1403 |
|
| 1404 |
if (q->frequency_range > (local_int_14 + 1)) { |
| 1405 |
int sub_packet = (local_int_20 + local_int_28);
|
| 1406 |
|
| 1407 |
qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); |
| 1408 |
if (stereo)
|
| 1409 |
qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
|
| 1410 |
} |
| 1411 |
|
| 1412 |
offset++; |
| 1413 |
} |
| 1414 |
} |
| 1415 |
|
| 1416 |
|
| 1417 |
static void qdm2_decode_fft_packets (QDM2Context *q) |
| 1418 |
{
|
| 1419 |
int i, j, min, max, value, type, unknown_flag;
|
| 1420 |
GetBitContext gb; |
| 1421 |
|
| 1422 |
if (q->sub_packet_list_B[0].packet == NULL) |
| 1423 |
return;
|
| 1424 |
|
| 1425 |
/* reset minimum indexes for FFT coefficients */
|
| 1426 |
q->fft_coefs_index = 0;
|
| 1427 |
for (i=0; i < 5; i++) |
| 1428 |
q->fft_coefs_min_index[i] = -1;
|
| 1429 |
|
| 1430 |
/* process subpackets ordered by type, largest type first */
|
| 1431 |
for (i = 0, max = 256; i < q->sub_packets_B; i++) { |
| 1432 |
QDM2SubPacket *packet= NULL;
|
| 1433 |
|
| 1434 |
/* find subpacket with largest type less than max */
|
| 1435 |
for (j = 0, min = 0; j < q->sub_packets_B; j++) { |
| 1436 |
value = q->sub_packet_list_B[j].packet->type; |
| 1437 |
if (value > min && value < max) {
|
| 1438 |
min = value; |
| 1439 |
packet = q->sub_packet_list_B[j].packet; |
| 1440 |
} |
| 1441 |
} |
| 1442 |
|
| 1443 |
max = min; |
| 1444 |
|
| 1445 |
/* check for errors (?) */
|
| 1446 |
if (!packet)
|
| 1447 |
return;
|
| 1448 |
|
| 1449 |
if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) |
| 1450 |
return;
|
| 1451 |
|
| 1452 |
/* decode FFT tones */
|
| 1453 |
init_get_bits (&gb, packet->data, packet->size*8);
|
| 1454 |
|
| 1455 |
if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) |
| 1456 |
unknown_flag = 1;
|
| 1457 |
else
|
| 1458 |
unknown_flag = 0;
|
| 1459 |
|
| 1460 |
type = packet->type; |
| 1461 |
|
| 1462 |
if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { |
| 1463 |
int duration = q->sub_sampling + 5 - (type & 15); |
| 1464 |
|
| 1465 |
if (duration >= 0 && duration < 4) |
| 1466 |
qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); |
| 1467 |
} else if (type == 31) { |
| 1468 |
for (j=0; j < 4; j++) |
| 1469 |
qdm2_fft_decode_tones(q, j, &gb, unknown_flag); |
| 1470 |
} else if (type == 46) { |
| 1471 |
for (j=0; j < 6; j++) |
| 1472 |
q->fft_level_exp[j] = get_bits(&gb, 6);
|
| 1473 |
for (j=0; j < 4; j++) |
| 1474 |
qdm2_fft_decode_tones(q, j, &gb, unknown_flag); |
| 1475 |
} |
| 1476 |
} // Loop on B packets
|
| 1477 |
|
| 1478 |
/* calculate maximum indexes for FFT coefficients */
|
| 1479 |
for (i = 0, j = -1; i < 5; i++) |
| 1480 |
if (q->fft_coefs_min_index[i] >= 0) { |
| 1481 |
if (j >= 0) |
| 1482 |
q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; |
| 1483 |
j = i; |
| 1484 |
} |
| 1485 |
if (j >= 0) |
| 1486 |
q->fft_coefs_max_index[j] = q->fft_coefs_index; |
| 1487 |
} |
| 1488 |
|
| 1489 |
|
| 1490 |
static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) |
| 1491 |
{
|
| 1492 |
float level, f[6]; |
| 1493 |
int i;
|
| 1494 |
QDM2Complex c; |
| 1495 |
const double iscale = 2.0*M_PI / 512.0; |
| 1496 |
|
| 1497 |
tone->phase += tone->phase_shift; |
| 1498 |
|
| 1499 |
/* calculate current level (maximum amplitude) of tone */
|
| 1500 |
level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; |
| 1501 |
c.im = level * sin(tone->phase*iscale); |
| 1502 |
c.re = level * cos(tone->phase*iscale); |
| 1503 |
|
| 1504 |
/* generate FFT coefficients for tone */
|
| 1505 |
if (tone->duration >= 3 || tone->cutoff >= 3) { |
| 1506 |
tone->complex[0].im += c.im; |
| 1507 |
tone->complex[0].re += c.re; |
| 1508 |
tone->complex[1].im -= c.im; |
| 1509 |
tone->complex[1].re -= c.re; |
| 1510 |
} else {
|
| 1511 |
f[1] = -tone->table[4]; |
| 1512 |
f[0] = tone->table[3] - tone->table[0]; |
| 1513 |
f[2] = 1.0 - tone->table[2] - tone->table[3]; |
| 1514 |
f[3] = tone->table[1] + tone->table[4] - 1.0; |
| 1515 |
f[4] = tone->table[0] - tone->table[1]; |
| 1516 |
f[5] = tone->table[2]; |
| 1517 |
for (i = 0; i < 2; i++) { |
| 1518 |
tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
|
| 1519 |
tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
|
| 1520 |
} |
| 1521 |
for (i = 0; i < 4; i++) { |
| 1522 |
tone->complex[i].re += c.re * f[i+2]; |
| 1523 |
tone->complex[i].im += c.im * f[i+2]; |
| 1524 |
} |
| 1525 |
} |
| 1526 |
|
| 1527 |
/* copy the tone if it has not yet died out */
|
| 1528 |
if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { |
| 1529 |
memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
|
| 1530 |
q->fft_tone_end = (q->fft_tone_end + 1) % 1000; |
| 1531 |
} |
| 1532 |
} |
| 1533 |
|
| 1534 |
|
| 1535 |
static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) |
| 1536 |
{
|
| 1537 |
int i, j, ch;
|
| 1538 |
const double iscale = 0.25 * M_PI; |
| 1539 |
|
| 1540 |
for (ch = 0; ch < q->channels; ch++) { |
| 1541 |
memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); |
| 1542 |
} |
| 1543 |
|
| 1544 |
|
| 1545 |
/* apply FFT tones with duration 4 (1 FFT period) */
|
| 1546 |
if (q->fft_coefs_min_index[4] >= 0) |
| 1547 |
for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { |
| 1548 |
float level;
|
| 1549 |
QDM2Complex c; |
| 1550 |
|
| 1551 |
if (q->fft_coefs[i].sub_packet != sub_packet)
|
| 1552 |
break;
|
| 1553 |
|
| 1554 |
ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; |
| 1555 |
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]; |
| 1556 |
|
| 1557 |
c.re = level * cos(q->fft_coefs[i].phase * iscale); |
| 1558 |
c.im = level * sin(q->fft_coefs[i].phase * iscale); |
| 1559 |
q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; |
| 1560 |
q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; |
| 1561 |
q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; |
| 1562 |
q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; |
| 1563 |
} |
| 1564 |
|
| 1565 |
/* generate existing FFT tones */
|
| 1566 |
for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
|
| 1567 |
qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); |
| 1568 |
q->fft_tone_start = (q->fft_tone_start + 1) % 1000; |
| 1569 |
} |
| 1570 |
|
| 1571 |
/* create and generate new FFT tones with duration 0 (long) to 3 (short) */
|
| 1572 |
for (i = 0; i < 4; i++) |
| 1573 |
if (q->fft_coefs_min_index[i] >= 0) { |
| 1574 |
for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
|
| 1575 |
int offset, four_i;
|
| 1576 |
FFTTone tone; |
| 1577 |
|
| 1578 |
if (q->fft_coefs[j].sub_packet != sub_packet)
|
| 1579 |
break;
|
| 1580 |
|
| 1581 |
four_i = (4 - i);
|
| 1582 |
offset = q->fft_coefs[j].offset >> four_i; |
| 1583 |
ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; |
| 1584 |
|
| 1585 |
if (offset < q->frequency_range) {
|
| 1586 |
if (offset < 2) |
| 1587 |
tone.cutoff = offset; |
| 1588 |
else
|
| 1589 |
tone.cutoff = (offset >= 60) ? 3 : 2; |
| 1590 |
|
| 1591 |
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]; |
| 1592 |
tone.complex = &q->fft.complex[ch][offset]; |
| 1593 |
tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; |
| 1594 |
tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; |
| 1595 |
tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); |
| 1596 |
tone.duration = i; |
| 1597 |
tone.time_index = 0;
|
| 1598 |
|
| 1599 |
qdm2_fft_generate_tone(q, &tone); |
| 1600 |
} |
| 1601 |
} |
| 1602 |
q->fft_coefs_min_index[i] = j; |
| 1603 |
} |
| 1604 |
} |
| 1605 |
|
| 1606 |
|
| 1607 |
static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) |
| 1608 |
{
|
| 1609 |
const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; |
| 1610 |
int i;
|
| 1611 |
q->fft.complex[channel][0].re *= 2.0f; |
| 1612 |
q->fft.complex[channel][0].im = 0.0f; |
| 1613 |
ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
|
| 1614 |
/* add samples to output buffer */
|
| 1615 |
for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) |
| 1616 |
q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; |
| 1617 |
} |
| 1618 |
|
| 1619 |
|
| 1620 |
/**
|
| 1621 |
* @param q context
|
| 1622 |
* @param index subpacket number
|
| 1623 |
*/
|
| 1624 |
static void qdm2_synthesis_filter (QDM2Context *q, int index) |
| 1625 |
{
|
| 1626 |
OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; |
| 1627 |
int i, k, ch, sb_used, sub_sampling, dither_state = 0; |
| 1628 |
|
| 1629 |
/* copy sb_samples */
|
| 1630 |
sb_used = QDM2_SB_USED(q->sub_sampling); |
| 1631 |
|
| 1632 |
for (ch = 0; ch < q->channels; ch++) |
| 1633 |
for (i = 0; i < 8; i++) |
| 1634 |
for (k=sb_used; k < SBLIMIT; k++)
|
| 1635 |
q->sb_samples[ch][(8 * index) + i][k] = 0; |
| 1636 |
|
| 1637 |
for (ch = 0; ch < q->nb_channels; ch++) { |
| 1638 |
OUT_INT *samples_ptr = samples + ch; |
| 1639 |
|
| 1640 |
for (i = 0; i < 8; i++) { |
| 1641 |
ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]), |
| 1642 |
mpa_window, &dither_state, |
| 1643 |
samples_ptr, q->nb_channels, |
| 1644 |
q->sb_samples[ch][(8 * index) + i]);
|
| 1645 |
samples_ptr += 32 * q->nb_channels;
|
| 1646 |
} |
| 1647 |
} |
| 1648 |
|
| 1649 |
/* add samples to output buffer */
|
| 1650 |
sub_sampling = (4 >> q->sub_sampling);
|
| 1651 |
|
| 1652 |
for (ch = 0; ch < q->channels; ch++) |
| 1653 |
for (i = 0; i < q->frame_size; i++) |
| 1654 |
q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16)); |
| 1655 |
} |
| 1656 |
|
| 1657 |
|
| 1658 |
/**
|
| 1659 |
* Init static data (does not depend on specific file)
|
| 1660 |
*
|
| 1661 |
* @param q context
|
| 1662 |
*/
|
| 1663 |
static av_cold void qdm2_init(QDM2Context *q) { |
| 1664 |
static int initialized = 0; |
| 1665 |
|
| 1666 |
if (initialized != 0) |
| 1667 |
return;
|
| 1668 |
initialized = 1;
|
| 1669 |
|
| 1670 |
qdm2_init_vlc(); |
| 1671 |
ff_mpa_synth_init(mpa_window); |
| 1672 |
softclip_table_init(); |
| 1673 |
rnd_table_init(); |
| 1674 |
init_noise_samples(); |
| 1675 |
|
| 1676 |
av_log(NULL, AV_LOG_DEBUG, "init done\n"); |
| 1677 |
} |
| 1678 |
|
| 1679 |
|
| 1680 |
#if 0
|
| 1681 |
static void dump_context(QDM2Context *q)
|
| 1682 |
{
|
| 1683 |
int i;
|
| 1684 |
#define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
|
| 1685 |
PRINT("compressed_data",q->compressed_data);
|
| 1686 |
PRINT("compressed_size",q->compressed_size);
|
| 1687 |
PRINT("frame_size",q->frame_size);
|
| 1688 |
PRINT("checksum_size",q->checksum_size);
|
| 1689 |
PRINT("channels",q->channels);
|
| 1690 |
PRINT("nb_channels",q->nb_channels);
|
| 1691 |
PRINT("fft_frame_size",q->fft_frame_size);
|
| 1692 |
PRINT("fft_size",q->fft_size);
|
| 1693 |
PRINT("sub_sampling",q->sub_sampling);
|
| 1694 |
PRINT("fft_order",q->fft_order);
|
| 1695 |
PRINT("group_order",q->group_order);
|
| 1696 |
PRINT("group_size",q->group_size);
|
| 1697 |
PRINT("sub_packet",q->sub_packet);
|
| 1698 |
PRINT("frequency_range",q->frequency_range);
|
| 1699 |
PRINT("has_errors",q->has_errors);
|
| 1700 |
PRINT("fft_tone_end",q->fft_tone_end);
|
| 1701 |
PRINT("fft_tone_start",q->fft_tone_start);
|
| 1702 |
PRINT("fft_coefs_index",q->fft_coefs_index);
|
| 1703 |
PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
|
| 1704 |
PRINT("cm_table_select",q->cm_table_select);
|
| 1705 |
PRINT("noise_idx",q->noise_idx);
|
| 1706 |
|
| 1707 |
for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
|
| 1708 |
{
|
| 1709 |
FFTTone *t = &q->fft_tones[i];
|
| 1710 |
|
| 1711 |
av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
|
| 1712 |
av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
|
| 1713 |
// PRINT(" level", t->level);
|
| 1714 |
PRINT(" phase", t->phase);
|
| 1715 |
PRINT(" phase_shift", t->phase_shift);
|
| 1716 |
PRINT(" duration", t->duration);
|
| 1717 |
PRINT(" samples_im", t->samples_im);
|
| 1718 |
PRINT(" samples_re", t->samples_re);
|
| 1719 |
PRINT(" table", t->table);
|
| 1720 |
}
|
| 1721 |
|
| 1722 |
}
|
| 1723 |
#endif
|
| 1724 |
|
| 1725 |
|
| 1726 |
/**
|
| 1727 |
* Init parameters from codec extradata
|
| 1728 |
*/
|
| 1729 |
static av_cold int qdm2_decode_init(AVCodecContext *avctx) |
| 1730 |
{
|
| 1731 |
QDM2Context *s = avctx->priv_data; |
| 1732 |
uint8_t *extradata; |
| 1733 |
int extradata_size;
|
| 1734 |
int tmp_val, tmp, size;
|
| 1735 |
|
| 1736 |
/* extradata parsing
|
| 1737 |
|
| 1738 |
Structure:
|
| 1739 |
wave {
|
| 1740 |
frma (QDM2)
|
| 1741 |
QDCA
|
| 1742 |
QDCP
|
| 1743 |
}
|
| 1744 |
|
| 1745 |
32 size (including this field)
|
| 1746 |
32 tag (=frma)
|
| 1747 |
32 type (=QDM2 or QDMC)
|
| 1748 |
|
| 1749 |
32 size (including this field, in bytes)
|
| 1750 |
32 tag (=QDCA) // maybe mandatory parameters
|
| 1751 |
32 unknown (=1)
|
| 1752 |
32 channels (=2)
|
| 1753 |
32 samplerate (=44100)
|
| 1754 |
32 bitrate (=96000)
|
| 1755 |
32 block size (=4096)
|
| 1756 |
32 frame size (=256) (for one channel)
|
| 1757 |
32 packet size (=1300)
|
| 1758 |
|
| 1759 |
32 size (including this field, in bytes)
|
| 1760 |
32 tag (=QDCP) // maybe some tuneable parameters
|
| 1761 |
32 float1 (=1.0)
|
| 1762 |
32 zero ?
|
| 1763 |
32 float2 (=1.0)
|
| 1764 |
32 float3 (=1.0)
|
| 1765 |
32 unknown (27)
|
| 1766 |
32 unknown (8)
|
| 1767 |
32 zero ?
|
| 1768 |
*/
|
| 1769 |
|
| 1770 |
if (!avctx->extradata || (avctx->extradata_size < 48)) { |
| 1771 |
av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
|
| 1772 |
return -1; |
| 1773 |
} |
| 1774 |
|
| 1775 |
extradata = avctx->extradata; |
| 1776 |
extradata_size = avctx->extradata_size; |
| 1777 |
|
| 1778 |
while (extradata_size > 7) { |
| 1779 |
if (!memcmp(extradata, "frmaQDM", 7)) |
| 1780 |
break;
|
| 1781 |
extradata++; |
| 1782 |
extradata_size--; |
| 1783 |
} |
| 1784 |
|
| 1785 |
if (extradata_size < 12) { |
| 1786 |
av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
|
| 1787 |
extradata_size); |
| 1788 |
return -1; |
| 1789 |
} |
| 1790 |
|
| 1791 |
if (memcmp(extradata, "frmaQDM", 7)) { |
| 1792 |
av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
|
| 1793 |
return -1; |
| 1794 |
} |
| 1795 |
|
| 1796 |
if (extradata[7] == 'C') { |
| 1797 |
// s->is_qdmc = 1;
|
| 1798 |
av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
|
| 1799 |
return -1; |
| 1800 |
} |
| 1801 |
|
| 1802 |
extradata += 8;
|
| 1803 |
extradata_size -= 8;
|
| 1804 |
|
| 1805 |
size = AV_RB32(extradata); |
| 1806 |
|
| 1807 |
if(size > extradata_size){
|
| 1808 |
av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
|
| 1809 |
extradata_size, size); |
| 1810 |
return -1; |
| 1811 |
} |
| 1812 |
|
| 1813 |
extradata += 4;
|
| 1814 |
av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
|
| 1815 |
if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { |
| 1816 |
av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
|
| 1817 |
return -1; |
| 1818 |
} |
| 1819 |
|
| 1820 |
extradata += 8;
|
| 1821 |
|
| 1822 |
avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); |
| 1823 |
extradata += 4;
|
| 1824 |
|
| 1825 |
avctx->sample_rate = AV_RB32(extradata); |
| 1826 |
extradata += 4;
|
| 1827 |
|
| 1828 |
avctx->bit_rate = AV_RB32(extradata); |
| 1829 |
extradata += 4;
|
| 1830 |
|
| 1831 |
s->group_size = AV_RB32(extradata); |
| 1832 |
extradata += 4;
|
| 1833 |
|
| 1834 |
s->fft_size = AV_RB32(extradata); |
| 1835 |
extradata += 4;
|
| 1836 |
|
| 1837 |
s->checksum_size = AV_RB32(extradata); |
| 1838 |
extradata += 4;
|
| 1839 |
|
| 1840 |
s->fft_order = av_log2(s->fft_size) + 1;
|
| 1841 |
s->fft_frame_size = 2 * s->fft_size; // complex has two floats |
| 1842 |
|
| 1843 |
// something like max decodable tones
|
| 1844 |
s->group_order = av_log2(s->group_size) + 1;
|
| 1845 |
s->frame_size = s->group_size / 16; // 16 iterations per super block |
| 1846 |
|
| 1847 |
s->sub_sampling = s->fft_order - 7;
|
| 1848 |
s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); |
| 1849 |
|
| 1850 |
switch ((s->sub_sampling * 2 + s->channels - 1)) { |
| 1851 |
case 0: tmp = 40; break; |
| 1852 |
case 1: tmp = 48; break; |
| 1853 |
case 2: tmp = 56; break; |
| 1854 |
case 3: tmp = 72; break; |
| 1855 |
case 4: tmp = 80; break; |
| 1856 |
case 5: tmp = 100;break; |
| 1857 |
default: tmp=s->sub_sampling; break; |
| 1858 |
} |
| 1859 |
tmp_val = 0;
|
| 1860 |
if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; |
| 1861 |
if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; |
| 1862 |
if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; |
| 1863 |
if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; |
| 1864 |
s->cm_table_select = tmp_val; |
| 1865 |
|
| 1866 |
if (s->sub_sampling == 0) |
| 1867 |
tmp = 7999;
|
| 1868 |
else
|
| 1869 |
tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; |
| 1870 |
/*
|
| 1871 |
0: 7999 -> 0
|
| 1872 |
1: 20000 -> 2
|
| 1873 |
2: 28000 -> 2
|
| 1874 |
*/
|
| 1875 |
if (tmp < 8000) |
| 1876 |
s->coeff_per_sb_select = 0;
|
| 1877 |
else if (tmp <= 16000) |
| 1878 |
s->coeff_per_sb_select = 1;
|
| 1879 |
else
|
| 1880 |
s->coeff_per_sb_select = 2;
|
| 1881 |
|
| 1882 |
// Fail on unknown fft order
|
| 1883 |
if ((s->fft_order < 7) || (s->fft_order > 9)) { |
| 1884 |
av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
|
| 1885 |
return -1; |
| 1886 |
} |
| 1887 |
|
| 1888 |
ff_rdft_init(&s->rdft_ctx, s->fft_order, IRDFT); |
| 1889 |
|
| 1890 |
qdm2_init(s); |
| 1891 |
|
| 1892 |
avctx->sample_fmt = SAMPLE_FMT_S16; |
| 1893 |
|
| 1894 |
// dump_context(s);
|
| 1895 |
return 0; |
| 1896 |
} |
| 1897 |
|
| 1898 |
|
| 1899 |
static av_cold int qdm2_decode_close(AVCodecContext *avctx) |
| 1900 |
{
|
| 1901 |
QDM2Context *s = avctx->priv_data; |
| 1902 |
|
| 1903 |
ff_rdft_end(&s->rdft_ctx); |
| 1904 |
|
| 1905 |
return 0; |
| 1906 |
} |
| 1907 |
|
| 1908 |
|
| 1909 |
static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) |
| 1910 |
{
|
| 1911 |
int ch, i;
|
| 1912 |
const int frame_size = (q->frame_size * q->channels); |
| 1913 |
|
| 1914 |
/* select input buffer */
|
| 1915 |
q->compressed_data = in; |
| 1916 |
q->compressed_size = q->checksum_size; |
| 1917 |
|
| 1918 |
// dump_context(q);
|
| 1919 |
|
| 1920 |
/* copy old block, clear new block of output samples */
|
| 1921 |
memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); |
| 1922 |
memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); |
| 1923 |
|
| 1924 |
/* decode block of QDM2 compressed data */
|
| 1925 |
if (q->sub_packet == 0) { |
| 1926 |
q->has_errors = 0; // zero it for a new super block |
| 1927 |
av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); |
| 1928 |
qdm2_decode_super_block(q); |
| 1929 |
} |
| 1930 |
|
| 1931 |
/* parse subpackets */
|
| 1932 |
if (!q->has_errors) {
|
| 1933 |
if (q->sub_packet == 2) |
| 1934 |
qdm2_decode_fft_packets(q); |
| 1935 |
|
| 1936 |
qdm2_fft_tone_synthesizer(q, q->sub_packet); |
| 1937 |
} |
| 1938 |
|
| 1939 |
/* sound synthesis stage 1 (FFT) */
|
| 1940 |
for (ch = 0; ch < q->channels; ch++) { |
| 1941 |
qdm2_calculate_fft(q, ch, q->sub_packet); |
| 1942 |
|
| 1943 |
if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { |
| 1944 |
SAMPLES_NEEDED_2("has errors, and C list is not empty")
|
| 1945 |
return;
|
| 1946 |
} |
| 1947 |
} |
| 1948 |
|
| 1949 |
/* sound synthesis stage 2 (MPEG audio like synthesis filter) */
|
| 1950 |
if (!q->has_errors && q->do_synth_filter)
|
| 1951 |
qdm2_synthesis_filter(q, q->sub_packet); |
| 1952 |
|
| 1953 |
q->sub_packet = (q->sub_packet + 1) % 16; |
| 1954 |
|
| 1955 |
/* clip and convert output float[] to 16bit signed samples */
|
| 1956 |
for (i = 0; i < frame_size; i++) { |
| 1957 |
int value = (int)q->output_buffer[i]; |
| 1958 |
|
| 1959 |
if (value > SOFTCLIP_THRESHOLD)
|
| 1960 |
value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
|
| 1961 |
else if (value < -SOFTCLIP_THRESHOLD) |
| 1962 |
value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
|
| 1963 |
|
| 1964 |
out[i] = value; |
| 1965 |
} |
| 1966 |
} |
| 1967 |
|
| 1968 |
|
| 1969 |
static int qdm2_decode_frame(AVCodecContext *avctx, |
| 1970 |
void *data, int *data_size, |
| 1971 |
AVPacket *avpkt) |
| 1972 |
{
|
| 1973 |
const uint8_t *buf = avpkt->data;
|
| 1974 |
int buf_size = avpkt->size;
|
| 1975 |
QDM2Context *s = avctx->priv_data; |
| 1976 |
|
| 1977 |
if(!buf)
|
| 1978 |
return 0; |
| 1979 |
if(buf_size < s->checksum_size)
|
| 1980 |
return -1; |
| 1981 |
|
| 1982 |
*data_size = s->channels * s->frame_size * sizeof(int16_t);
|
| 1983 |
|
| 1984 |
av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
|
| 1985 |
buf_size, buf, s->checksum_size, data, *data_size); |
| 1986 |
|
| 1987 |
qdm2_decode(s, buf, data); |
| 1988 |
|
| 1989 |
// reading only when next superblock found
|
| 1990 |
if (s->sub_packet == 0) { |
| 1991 |
return s->checksum_size;
|
| 1992 |
} |
| 1993 |
|
| 1994 |
return 0; |
| 1995 |
} |
| 1996 |
|
| 1997 |
AVCodec qdm2_decoder = |
| 1998 |
{
|
| 1999 |
.name = "qdm2",
|
| 2000 |
.type = CODEC_TYPE_AUDIO, |
| 2001 |
.id = CODEC_ID_QDM2, |
| 2002 |
.priv_data_size = sizeof(QDM2Context),
|
| 2003 |
.init = qdm2_decode_init, |
| 2004 |
.close = qdm2_decode_close, |
| 2005 |
.decode = qdm2_decode_frame, |
| 2006 |
.long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
|
| 2007 |
}; |