2 * QDM2 compatible decoder
3 * Copyright (c) 2003 Ewald Snel
4 * Copyright (c) 2005 Benjamin Larsson
5 * Copyright (c) 2005 Alex Beregszaszi
6 * Copyright (c) 2005 Roberto Togni
8 * This file is part of Libav.
10 * Libav is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Lesser General Public
12 * License as published by the Free Software Foundation; either
13 * version 2.1 of the License, or (at your option) any later version.
15 * Libav is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Lesser General Public License for more details.
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with Libav; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
30 * The decoder is not perfect yet, there are still some distortions
31 * especially on files encoded with 16 or 8 subbands.
38 #define BITSTREAM_READER_LE
39 #include "libavutil/channel_layout.h"
45 #include "mpegaudiodsp.h"
46 #include "mpegaudio.h"
49 #include "qdm2_tablegen.h"
55 #define QDM2_LIST_ADD(list, size, packet) \
58 list[size - 1].next = &list[size]; \
60 list[size].packet = packet; \
61 list[size].next = NULL; \
65 // Result is 8, 16 or 30
66 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
68 #define FIX_NOISE_IDX(noise_idx) \
69 if ((noise_idx) >= 3840) \
70 (noise_idx) -= 3840; \
72 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
74 #define SAMPLES_NEEDED \
75 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
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);
80 #define QDM2_MAX_FRAME_SIZE 512
82 typedef int8_t sb_int8_array
[2][30][64];
88 int type
; ///< subpacket type
89 unsigned int size
; ///< subpacket size
90 const uint8_t *data
; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
94 * A node in the subpacket list
96 typedef struct QDM2SubPNode
{
97 QDM2SubPacket
*packet
; ///< packet
98 struct QDM2SubPNode
*next
; ///< pointer to next packet in the list, NULL if leaf node
108 QDM2Complex
*complex;
126 DECLARE_ALIGNED(32, QDM2Complex
, complex)[MPA_MAX_CHANNELS
][256];
130 * QDM2 decoder context
135 /// Parameters from codec header, do not change during playback
136 int nb_channels
; ///< number of channels
137 int channels
; ///< number of channels
138 int group_size
; ///< size of frame group (16 frames per group)
139 int fft_size
; ///< size of FFT, in complex numbers
140 int checksum_size
; ///< size of data block, used also for checksum
142 /// Parameters built from header parameters, do not change during playback
143 int group_order
; ///< order of frame group
144 int fft_order
; ///< order of FFT (actually fftorder+1)
145 int frame_size
; ///< size of data frame
147 int sub_sampling
; ///< subsampling: 0=25%, 1=50%, 2=100% */
148 int coeff_per_sb_select
; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
149 int cm_table_select
; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
151 /// Packets and packet lists
152 QDM2SubPacket sub_packets
[16]; ///< the packets themselves
153 QDM2SubPNode sub_packet_list_A
[16]; ///< list of all packets
154 QDM2SubPNode sub_packet_list_B
[16]; ///< FFT packets B are on list
155 int sub_packets_B
; ///< number of packets on 'B' list
156 QDM2SubPNode sub_packet_list_C
[16]; ///< packets with errors?
157 QDM2SubPNode sub_packet_list_D
[16]; ///< DCT packets
160 FFTTone fft_tones
[1000];
163 FFTCoefficient fft_coefs
[1000];
165 int fft_coefs_min_index
[5];
166 int fft_coefs_max_index
[5];
167 int fft_level_exp
[6];
168 RDFTContext rdft_ctx
;
172 const uint8_t *compressed_data
;
174 float output_buffer
[QDM2_MAX_FRAME_SIZE
* 2];
177 MPADSPContext mpadsp
;
178 DECLARE_ALIGNED(32, float, synth_buf
)[MPA_MAX_CHANNELS
][512*2];
179 int synth_buf_offset
[MPA_MAX_CHANNELS
];
180 DECLARE_ALIGNED(32, float, sb_samples
)[MPA_MAX_CHANNELS
][128][SBLIMIT
];
181 DECLARE_ALIGNED(32, float, samples
)[MPA_MAX_CHANNELS
* MPA_FRAME_SIZE
];
183 /// Mixed temporary data used in decoding
184 float tone_level
[MPA_MAX_CHANNELS
][30][64];
185 int8_t coding_method
[MPA_MAX_CHANNELS
][30][64];
186 int8_t quantized_coeffs
[MPA_MAX_CHANNELS
][10][8];
187 int8_t tone_level_idx_base
[MPA_MAX_CHANNELS
][30][8];
188 int8_t tone_level_idx_hi1
[MPA_MAX_CHANNELS
][3][8][8];
189 int8_t tone_level_idx_mid
[MPA_MAX_CHANNELS
][26][8];
190 int8_t tone_level_idx_hi2
[MPA_MAX_CHANNELS
][26];
191 int8_t tone_level_idx
[MPA_MAX_CHANNELS
][30][64];
192 int8_t tone_level_idx_temp
[MPA_MAX_CHANNELS
][30][64];
195 int has_errors
; ///< packet has errors
196 int superblocktype_2_3
; ///< select fft tables and some algorithm based on superblock type
197 int do_synth_filter
; ///< used to perform or skip synthesis filter
200 int noise_idx
; ///< index for dithering noise table
204 static VLC vlc_tab_level
;
205 static VLC vlc_tab_diff
;
206 static VLC vlc_tab_run
;
207 static VLC fft_level_exp_alt_vlc
;
208 static VLC fft_level_exp_vlc
;
209 static VLC fft_stereo_exp_vlc
;
210 static VLC fft_stereo_phase_vlc
;
211 static VLC vlc_tab_tone_level_idx_hi1
;
212 static VLC vlc_tab_tone_level_idx_mid
;
213 static VLC vlc_tab_tone_level_idx_hi2
;
214 static VLC vlc_tab_type30
;
215 static VLC vlc_tab_type34
;
216 static VLC vlc_tab_fft_tone_offset
[5];
218 static const uint16_t qdm2_vlc_offs
[] = {
219 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
222 static av_cold
void qdm2_init_vlc(void)
224 static int vlcs_initialized
= 0;
225 static VLC_TYPE qdm2_table
[3838][2];
227 if (!vlcs_initialized
) {
229 vlc_tab_level
.table
= &qdm2_table
[qdm2_vlc_offs
[0]];
230 vlc_tab_level
.table_allocated
= qdm2_vlc_offs
[1] - qdm2_vlc_offs
[0];
231 init_vlc (&vlc_tab_level
, 8, 24,
232 vlc_tab_level_huffbits
, 1, 1,
233 vlc_tab_level_huffcodes
, 2, 2, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
235 vlc_tab_diff
.table
= &qdm2_table
[qdm2_vlc_offs
[1]];
236 vlc_tab_diff
.table_allocated
= qdm2_vlc_offs
[2] - qdm2_vlc_offs
[1];
237 init_vlc (&vlc_tab_diff
, 8, 37,
238 vlc_tab_diff_huffbits
, 1, 1,
239 vlc_tab_diff_huffcodes
, 2, 2, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
241 vlc_tab_run
.table
= &qdm2_table
[qdm2_vlc_offs
[2]];
242 vlc_tab_run
.table_allocated
= qdm2_vlc_offs
[3] - qdm2_vlc_offs
[2];
243 init_vlc (&vlc_tab_run
, 5, 6,
244 vlc_tab_run_huffbits
, 1, 1,
245 vlc_tab_run_huffcodes
, 1, 1, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
247 fft_level_exp_alt_vlc
.table
= &qdm2_table
[qdm2_vlc_offs
[3]];
248 fft_level_exp_alt_vlc
.table_allocated
= qdm2_vlc_offs
[4] - qdm2_vlc_offs
[3];
249 init_vlc (&fft_level_exp_alt_vlc
, 8, 28,
250 fft_level_exp_alt_huffbits
, 1, 1,
251 fft_level_exp_alt_huffcodes
, 2, 2, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
254 fft_level_exp_vlc
.table
= &qdm2_table
[qdm2_vlc_offs
[4]];
255 fft_level_exp_vlc
.table_allocated
= qdm2_vlc_offs
[5] - qdm2_vlc_offs
[4];
256 init_vlc (&fft_level_exp_vlc
, 8, 20,
257 fft_level_exp_huffbits
, 1, 1,
258 fft_level_exp_huffcodes
, 2, 2, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
260 fft_stereo_exp_vlc
.table
= &qdm2_table
[qdm2_vlc_offs
[5]];
261 fft_stereo_exp_vlc
.table_allocated
= qdm2_vlc_offs
[6] - qdm2_vlc_offs
[5];
262 init_vlc (&fft_stereo_exp_vlc
, 6, 7,
263 fft_stereo_exp_huffbits
, 1, 1,
264 fft_stereo_exp_huffcodes
, 1, 1, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
266 fft_stereo_phase_vlc
.table
= &qdm2_table
[qdm2_vlc_offs
[6]];
267 fft_stereo_phase_vlc
.table_allocated
= qdm2_vlc_offs
[7] - qdm2_vlc_offs
[6];
268 init_vlc (&fft_stereo_phase_vlc
, 6, 9,
269 fft_stereo_phase_huffbits
, 1, 1,
270 fft_stereo_phase_huffcodes
, 1, 1, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
272 vlc_tab_tone_level_idx_hi1
.table
= &qdm2_table
[qdm2_vlc_offs
[7]];
273 vlc_tab_tone_level_idx_hi1
.table_allocated
= qdm2_vlc_offs
[8] - qdm2_vlc_offs
[7];
274 init_vlc (&vlc_tab_tone_level_idx_hi1
, 8, 20,
275 vlc_tab_tone_level_idx_hi1_huffbits
, 1, 1,
276 vlc_tab_tone_level_idx_hi1_huffcodes
, 2, 2, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
278 vlc_tab_tone_level_idx_mid
.table
= &qdm2_table
[qdm2_vlc_offs
[8]];
279 vlc_tab_tone_level_idx_mid
.table_allocated
= qdm2_vlc_offs
[9] - qdm2_vlc_offs
[8];
280 init_vlc (&vlc_tab_tone_level_idx_mid
, 8, 24,
281 vlc_tab_tone_level_idx_mid_huffbits
, 1, 1,
282 vlc_tab_tone_level_idx_mid_huffcodes
, 2, 2, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
284 vlc_tab_tone_level_idx_hi2
.table
= &qdm2_table
[qdm2_vlc_offs
[9]];
285 vlc_tab_tone_level_idx_hi2
.table_allocated
= qdm2_vlc_offs
[10] - qdm2_vlc_offs
[9];
286 init_vlc (&vlc_tab_tone_level_idx_hi2
, 8, 24,
287 vlc_tab_tone_level_idx_hi2_huffbits
, 1, 1,
288 vlc_tab_tone_level_idx_hi2_huffcodes
, 2, 2, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
290 vlc_tab_type30
.table
= &qdm2_table
[qdm2_vlc_offs
[10]];
291 vlc_tab_type30
.table_allocated
= qdm2_vlc_offs
[11] - qdm2_vlc_offs
[10];
292 init_vlc (&vlc_tab_type30
, 6, 9,
293 vlc_tab_type30_huffbits
, 1, 1,
294 vlc_tab_type30_huffcodes
, 1, 1, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
296 vlc_tab_type34
.table
= &qdm2_table
[qdm2_vlc_offs
[11]];
297 vlc_tab_type34
.table_allocated
= qdm2_vlc_offs
[12] - qdm2_vlc_offs
[11];
298 init_vlc (&vlc_tab_type34
, 5, 10,
299 vlc_tab_type34_huffbits
, 1, 1,
300 vlc_tab_type34_huffcodes
, 1, 1, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
302 vlc_tab_fft_tone_offset
[0].table
= &qdm2_table
[qdm2_vlc_offs
[12]];
303 vlc_tab_fft_tone_offset
[0].table_allocated
= qdm2_vlc_offs
[13] - qdm2_vlc_offs
[12];
304 init_vlc (&vlc_tab_fft_tone_offset
[0], 8, 23,
305 vlc_tab_fft_tone_offset_0_huffbits
, 1, 1,
306 vlc_tab_fft_tone_offset_0_huffcodes
, 2, 2, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
308 vlc_tab_fft_tone_offset
[1].table
= &qdm2_table
[qdm2_vlc_offs
[13]];
309 vlc_tab_fft_tone_offset
[1].table_allocated
= qdm2_vlc_offs
[14] - qdm2_vlc_offs
[13];
310 init_vlc (&vlc_tab_fft_tone_offset
[1], 8, 28,
311 vlc_tab_fft_tone_offset_1_huffbits
, 1, 1,
312 vlc_tab_fft_tone_offset_1_huffcodes
, 2, 2, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
314 vlc_tab_fft_tone_offset
[2].table
= &qdm2_table
[qdm2_vlc_offs
[14]];
315 vlc_tab_fft_tone_offset
[2].table_allocated
= qdm2_vlc_offs
[15] - qdm2_vlc_offs
[14];
316 init_vlc (&vlc_tab_fft_tone_offset
[2], 8, 32,
317 vlc_tab_fft_tone_offset_2_huffbits
, 1, 1,
318 vlc_tab_fft_tone_offset_2_huffcodes
, 2, 2, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
320 vlc_tab_fft_tone_offset
[3].table
= &qdm2_table
[qdm2_vlc_offs
[15]];
321 vlc_tab_fft_tone_offset
[3].table_allocated
= qdm2_vlc_offs
[16] - qdm2_vlc_offs
[15];
322 init_vlc (&vlc_tab_fft_tone_offset
[3], 8, 35,
323 vlc_tab_fft_tone_offset_3_huffbits
, 1, 1,
324 vlc_tab_fft_tone_offset_3_huffcodes
, 2, 2, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
326 vlc_tab_fft_tone_offset
[4].table
= &qdm2_table
[qdm2_vlc_offs
[16]];
327 vlc_tab_fft_tone_offset
[4].table_allocated
= qdm2_vlc_offs
[17] - qdm2_vlc_offs
[16];
328 init_vlc (&vlc_tab_fft_tone_offset
[4], 8, 38,
329 vlc_tab_fft_tone_offset_4_huffbits
, 1, 1,
330 vlc_tab_fft_tone_offset_4_huffcodes
, 2, 2, INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
336 static int qdm2_get_vlc (GetBitContext
*gb
, VLC
*vlc
, int flag
, int depth
)
340 value
= get_vlc2(gb
, vlc
->table
, vlc
->bits
, depth
);
342 /* stage-2, 3 bits exponent escape sequence */
344 value
= get_bits (gb
, get_bits (gb
, 3) + 1);
346 /* stage-3, optional */
348 int tmp
= vlc_stage3_values
[value
];
350 if ((value
& ~3) > 0)
351 tmp
+= get_bits (gb
, (value
>> 2));
359 static int qdm2_get_se_vlc (VLC
*vlc
, GetBitContext
*gb
, int depth
)
361 int value
= qdm2_get_vlc (gb
, vlc
, 0, depth
);
363 return (value
& 1) ? ((value
+ 1) >> 1) : -(value
>> 1);
370 * @param data pointer to data to be checksum'ed
371 * @param length data length
372 * @param value checksum value
374 * @return 0 if checksum is OK
376 static uint16_t qdm2_packet_checksum (const uint8_t *data
, int length
, int value
) {
379 for (i
=0; i
< length
; i
++)
382 return (uint16_t)(value
& 0xffff);
387 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
389 * @param gb bitreader context
390 * @param sub_packet packet under analysis
392 static void qdm2_decode_sub_packet_header (GetBitContext
*gb
, QDM2SubPacket
*sub_packet
)
394 sub_packet
->type
= get_bits (gb
, 8);
396 if (sub_packet
->type
== 0) {
397 sub_packet
->size
= 0;
398 sub_packet
->data
= NULL
;
400 sub_packet
->size
= get_bits (gb
, 8);
402 if (sub_packet
->type
& 0x80) {
403 sub_packet
->size
<<= 8;
404 sub_packet
->size
|= get_bits (gb
, 8);
405 sub_packet
->type
&= 0x7f;
408 if (sub_packet
->type
== 0x7f)
409 sub_packet
->type
|= (get_bits (gb
, 8) << 8);
411 sub_packet
->data
= &gb
->buffer
[get_bits_count(gb
) / 8]; // FIXME: this depends on bitreader internal data
414 av_log(NULL
,AV_LOG_DEBUG
,"Subpacket: type=%d size=%d start_offs=%x\n",
415 sub_packet
->type
, sub_packet
->size
, get_bits_count(gb
) / 8);
420 * Return node pointer to first packet of requested type in list.
422 * @param list list of subpackets to be scanned
423 * @param type type of searched subpacket
424 * @return node pointer for subpacket if found, else NULL
426 static QDM2SubPNode
* qdm2_search_subpacket_type_in_list (QDM2SubPNode
*list
, int type
)
428 while (list
!= NULL
&& list
->packet
!= NULL
) {
429 if (list
->packet
->type
== type
)
438 * Replace 8 elements with their average value.
439 * Called by qdm2_decode_superblock before starting subblock decoding.
443 static void average_quantized_coeffs (QDM2Context
*q
)
445 int i
, j
, n
, ch
, sum
;
447 n
= coeff_per_sb_for_avg
[q
->coeff_per_sb_select
][QDM2_SB_USED(q
->sub_sampling
) - 1] + 1;
449 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
450 for (i
= 0; i
< n
; i
++) {
453 for (j
= 0; j
< 8; j
++)
454 sum
+= q
->quantized_coeffs
[ch
][i
][j
];
460 for (j
=0; j
< 8; j
++)
461 q
->quantized_coeffs
[ch
][i
][j
] = sum
;
467 * Build subband samples with noise weighted by q->tone_level.
468 * Called by synthfilt_build_sb_samples.
471 * @param sb subband index
473 static void build_sb_samples_from_noise (QDM2Context
*q
, int sb
)
477 FIX_NOISE_IDX(q
->noise_idx
);
482 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
483 for (j
= 0; j
< 64; j
++) {
484 q
->sb_samples
[ch
][j
* 2][sb
] = SB_DITHERING_NOISE(sb
,q
->noise_idx
) * q
->tone_level
[ch
][sb
][j
];
485 q
->sb_samples
[ch
][j
* 2 + 1][sb
] = SB_DITHERING_NOISE(sb
,q
->noise_idx
) * q
->tone_level
[ch
][sb
][j
];
491 * Called while processing data from subpackets 11 and 12.
492 * Used after making changes to coding_method array.
494 * @param sb subband index
495 * @param channels number of channels
496 * @param coding_method q->coding_method[0][0][0]
498 static void fix_coding_method_array (int sb
, int channels
, sb_int8_array coding_method
)
503 static const 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};
505 for (ch
= 0; ch
< channels
; ch
++) {
506 for (j
= 0; j
< 64; ) {
507 if((coding_method
[ch
][sb
][j
] - 8) > 22) {
511 switch (switchtable
[coding_method
[ch
][sb
][j
]-8]) {
512 case 0: run
= 10; case_val
= 10; break;
513 case 1: run
= 1; case_val
= 16; break;
514 case 2: run
= 5; case_val
= 24; break;
515 case 3: run
= 3; case_val
= 30; break;
516 case 4: run
= 1; case_val
= 30; break;
517 case 5: run
= 1; case_val
= 8; break;
518 default: run
= 1; case_val
= 8; break;
521 for (k
= 0; k
< run
; k
++)
523 if (coding_method
[ch
][sb
+ (j
+ k
) / 64][(j
+ k
) % 64] > coding_method
[ch
][sb
][j
])
526 //not debugged, almost never used
527 memset(&coding_method
[ch
][sb
][j
+ k
], case_val
, k
* sizeof(int8_t));
528 memset(&coding_method
[ch
][sb
][j
+ k
], case_val
, 3 * sizeof(int8_t));
537 * Related to synthesis filter
538 * Called by process_subpacket_10
541 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
543 static void fill_tone_level_array (QDM2Context
*q
, int flag
)
545 int i
, sb
, ch
, sb_used
;
548 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
549 for (sb
= 0; sb
< 30; sb
++)
550 for (i
= 0; i
< 8; i
++) {
551 if ((tab
=coeff_per_sb_for_dequant
[q
->coeff_per_sb_select
][sb
]) < (last_coeff
[q
->coeff_per_sb_select
] - 1))
552 tmp
= q
->quantized_coeffs
[ch
][tab
+ 1][i
] * dequant_table
[q
->coeff_per_sb_select
][tab
+ 1][sb
]+
553 q
->quantized_coeffs
[ch
][tab
][i
] * dequant_table
[q
->coeff_per_sb_select
][tab
][sb
];
555 tmp
= q
->quantized_coeffs
[ch
][tab
][i
] * dequant_table
[q
->coeff_per_sb_select
][tab
][sb
];
558 q
->tone_level_idx_base
[ch
][sb
][i
] = (tmp
/ 256) & 0xff;
561 sb_used
= QDM2_SB_USED(q
->sub_sampling
);
563 if ((q
->superblocktype_2_3
!= 0) && !flag
) {
564 for (sb
= 0; sb
< sb_used
; sb
++)
565 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
566 for (i
= 0; i
< 64; i
++) {
567 q
->tone_level_idx
[ch
][sb
][i
] = q
->tone_level_idx_base
[ch
][sb
][i
/ 8];
568 if (q
->tone_level_idx
[ch
][sb
][i
] < 0)
569 q
->tone_level
[ch
][sb
][i
] = 0;
571 q
->tone_level
[ch
][sb
][i
] = fft_tone_level_table
[0][q
->tone_level_idx
[ch
][sb
][i
] & 0x3f];
574 tab
= q
->superblocktype_2_3
? 0 : 1;
575 for (sb
= 0; sb
< sb_used
; sb
++) {
576 if ((sb
>= 4) && (sb
<= 23)) {
577 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
578 for (i
= 0; i
< 64; i
++) {
579 tmp
= q
->tone_level_idx_base
[ch
][sb
][i
/ 8] -
580 q
->tone_level_idx_hi1
[ch
][sb
/ 8][i
/ 8][i
% 8] -
581 q
->tone_level_idx_mid
[ch
][sb
- 4][i
/ 8] -
582 q
->tone_level_idx_hi2
[ch
][sb
- 4];
583 q
->tone_level_idx
[ch
][sb
][i
] = tmp
& 0xff;
584 if ((tmp
< 0) || (!q
->superblocktype_2_3
&& !tmp
))
585 q
->tone_level
[ch
][sb
][i
] = 0;
587 q
->tone_level
[ch
][sb
][i
] = fft_tone_level_table
[tab
][tmp
& 0x3f];
591 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
592 for (i
= 0; i
< 64; i
++) {
593 tmp
= q
->tone_level_idx_base
[ch
][sb
][i
/ 8] -
594 q
->tone_level_idx_hi1
[ch
][2][i
/ 8][i
% 8] -
595 q
->tone_level_idx_hi2
[ch
][sb
- 4];
596 q
->tone_level_idx
[ch
][sb
][i
] = tmp
& 0xff;
597 if ((tmp
< 0) || (!q
->superblocktype_2_3
&& !tmp
))
598 q
->tone_level
[ch
][sb
][i
] = 0;
600 q
->tone_level
[ch
][sb
][i
] = fft_tone_level_table
[tab
][tmp
& 0x3f];
603 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
604 for (i
= 0; i
< 64; i
++) {
605 tmp
= q
->tone_level_idx
[ch
][sb
][i
] = q
->tone_level_idx_base
[ch
][sb
][i
/ 8];
606 if ((tmp
< 0) || (!q
->superblocktype_2_3
&& !tmp
))
607 q
->tone_level
[ch
][sb
][i
] = 0;
609 q
->tone_level
[ch
][sb
][i
] = fft_tone_level_table
[tab
][tmp
& 0x3f];
621 * Related to synthesis filter
622 * Called by process_subpacket_11
623 * c is built with data from subpacket 11
624 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
626 * @param tone_level_idx
627 * @param tone_level_idx_temp
628 * @param coding_method q->coding_method[0][0][0]
629 * @param nb_channels number of channels
630 * @param c coming from subpacket 11, passed as 8*c
631 * @param superblocktype_2_3 flag based on superblock packet type
632 * @param cm_table_select q->cm_table_select
634 static void fill_coding_method_array (sb_int8_array tone_level_idx
, sb_int8_array tone_level_idx_temp
,
635 sb_int8_array coding_method
, int nb_channels
,
636 int c
, int superblocktype_2_3
, int cm_table_select
)
639 int tmp
, acc
, esp_40
, comp
;
640 int add1
, add2
, add3
, add4
;
643 if (!superblocktype_2_3
) {
644 /* This case is untested, no samples available */
646 for (ch
= 0; ch
< nb_channels
; ch
++)
647 for (sb
= 0; sb
< 30; sb
++) {
648 for (j
= 1; j
< 63; j
++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
649 add1
= tone_level_idx
[ch
][sb
][j
] - 10;
652 add2
= add3
= add4
= 0;
654 add2
= tone_level_idx
[ch
][sb
- 2][j
] + tone_level_idx_offset_table
[sb
][0] - 6;
659 add3
= tone_level_idx
[ch
][sb
- 1][j
] + tone_level_idx_offset_table
[sb
][1] - 6;
664 add4
= tone_level_idx
[ch
][sb
+ 1][j
] + tone_level_idx_offset_table
[sb
][3] - 6;
668 tmp
= tone_level_idx
[ch
][sb
][j
+ 1] * 2 - add4
- add3
- add2
- add1
;
671 tone_level_idx_temp
[ch
][sb
][j
+ 1] = tmp
& 0xff;
673 tone_level_idx_temp
[ch
][sb
][0] = tone_level_idx_temp
[ch
][sb
][1];
676 for (ch
= 0; ch
< nb_channels
; ch
++)
677 for (sb
= 0; sb
< 30; sb
++)
678 for (j
= 0; j
< 64; j
++)
679 acc
+= tone_level_idx_temp
[ch
][sb
][j
];
681 multres
= 0x66666667 * (acc
* 10);
682 esp_40
= (multres
>> 32) / 8 + ((multres
& 0xffffffff) >> 31);
683 for (ch
= 0; ch
< nb_channels
; ch
++)
684 for (sb
= 0; sb
< 30; sb
++)
685 for (j
= 0; j
< 64; j
++) {
686 comp
= tone_level_idx_temp
[ch
][sb
][j
]* esp_40
* 10;
689 comp
/= 256; // signed shift
717 coding_method
[ch
][sb
][j
] = ((tmp
& 0xfffa) + 30 )& 0xff;
719 for (sb
= 0; sb
< 30; sb
++)
720 fix_coding_method_array(sb
, nb_channels
, coding_method
);
721 for (ch
= 0; ch
< nb_channels
; ch
++)
722 for (sb
= 0; sb
< 30; sb
++)
723 for (j
= 0; j
< 64; j
++)
725 if (coding_method
[ch
][sb
][j
] < 10)
726 coding_method
[ch
][sb
][j
] = 10;
729 if (coding_method
[ch
][sb
][j
] < 16)
730 coding_method
[ch
][sb
][j
] = 16;
732 if (coding_method
[ch
][sb
][j
] < 30)
733 coding_method
[ch
][sb
][j
] = 30;
736 } else { // superblocktype_2_3 != 0
737 for (ch
= 0; ch
< nb_channels
; ch
++)
738 for (sb
= 0; sb
< 30; sb
++)
739 for (j
= 0; j
< 64; j
++)
740 coding_method
[ch
][sb
][j
] = coding_method_table
[cm_table_select
][sb
];
749 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
750 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
753 * @param gb bitreader context
754 * @param length packet length in bits
755 * @param sb_min lower subband processed (sb_min included)
756 * @param sb_max higher subband processed (sb_max excluded)
758 static void synthfilt_build_sb_samples (QDM2Context
*q
, GetBitContext
*gb
, int length
, int sb_min
, int sb_max
)
760 int sb
, j
, k
, n
, ch
, run
, channels
;
761 int joined_stereo
, zero_encoding
, chs
;
763 float type34_div
= 0;
764 float type34_predictor
;
765 float samples
[10], sign_bits
[16];
768 // If no data use noise
769 for (sb
=sb_min
; sb
< sb_max
; sb
++)
770 build_sb_samples_from_noise (q
, sb
);
775 for (sb
= sb_min
; sb
< sb_max
; sb
++) {
776 FIX_NOISE_IDX(q
->noise_idx
);
778 channels
= q
->nb_channels
;
780 if (q
->nb_channels
<= 1 || sb
< 12)
785 joined_stereo
= (get_bits_left(gb
) >= 1) ? get_bits1 (gb
) : 0;
788 if (get_bits_left(gb
) >= 16)
789 for (j
= 0; j
< 16; j
++)
790 sign_bits
[j
] = get_bits1 (gb
);
792 for (j
= 0; j
< 64; j
++)
793 if (q
->coding_method
[1][sb
][j
] > q
->coding_method
[0][sb
][j
])
794 q
->coding_method
[0][sb
][j
] = q
->coding_method
[1][sb
][j
];
796 fix_coding_method_array(sb
, q
->nb_channels
, q
->coding_method
);
800 for (ch
= 0; ch
< channels
; ch
++) {
801 zero_encoding
= (get_bits_left(gb
) >= 1) ? get_bits1(gb
) : 0;
802 type34_predictor
= 0.0;
805 for (j
= 0; j
< 128; ) {
806 switch (q
->coding_method
[ch
][sb
][j
/ 2]) {
808 if (get_bits_left(gb
) >= 10) {
810 for (k
= 0; k
< 5; k
++) {
811 if ((j
+ 2 * k
) >= 128)
813 samples
[2 * k
] = get_bits1(gb
) ? dequant_1bit
[joined_stereo
][2 * get_bits1(gb
)] : 0;
817 for (k
= 0; k
< 5; k
++)
818 samples
[2 * k
] = dequant_1bit
[joined_stereo
][random_dequant_index
[n
][k
]];
820 for (k
= 0; k
< 5; k
++)
821 samples
[2 * k
+ 1] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
823 for (k
= 0; k
< 10; k
++)
824 samples
[k
] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
830 if (get_bits_left(gb
) >= 1) {
835 f
-= noise_samples
[((sb
+ 1) * (j
+5 * ch
+ 1)) & 127] * 9.0 / 40.0;
838 samples
[0] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
844 if (get_bits_left(gb
) >= 10) {
846 for (k
= 0; k
< 5; k
++) {
849 samples
[k
] = (get_bits1(gb
) == 0) ? 0 : dequant_1bit
[joined_stereo
][2 * get_bits1(gb
)];
852 n
= get_bits (gb
, 8);
853 for (k
= 0; k
< 5; k
++)
854 samples
[k
] = dequant_1bit
[joined_stereo
][random_dequant_index
[n
][k
]];
857 for (k
= 0; k
< 5; k
++)
858 samples
[k
] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
864 if (get_bits_left(gb
) >= 7) {
866 for (k
= 0; k
< 3; k
++)
867 samples
[k
] = (random_dequant_type24
[n
][k
] - 2.0) * 0.5;
869 for (k
= 0; k
< 3; k
++)
870 samples
[k
] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
876 if (get_bits_left(gb
) >= 4) {
877 unsigned index
= qdm2_get_vlc(gb
, &vlc_tab_type30
, 0, 1);
878 if (index
< FF_ARRAY_ELEMS(type30_dequant
)) {
879 samples
[0] = type30_dequant
[index
];
881 samples
[0] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
883 samples
[0] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
889 if (get_bits_left(gb
) >= 7) {
891 type34_div
= (float)(1 << get_bits(gb
, 2));
892 samples
[0] = ((float)get_bits(gb
, 5) - 16.0) / 15.0;
893 type34_predictor
= samples
[0];
896 unsigned index
= qdm2_get_vlc(gb
, &vlc_tab_type34
, 0, 1);
897 if (index
< FF_ARRAY_ELEMS(type34_delta
)) {
898 samples
[0] = type34_delta
[index
] / type34_div
+ type34_predictor
;
899 type34_predictor
= samples
[0];
901 samples
[0] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
904 samples
[0] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
910 samples
[0] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
916 float tmp
[10][MPA_MAX_CHANNELS
];
918 for (k
= 0; k
< run
; k
++) {
919 tmp
[k
][0] = samples
[k
];
920 tmp
[k
][1] = (sign_bits
[(j
+ k
) / 8]) ? -samples
[k
] : samples
[k
];
922 for (chs
= 0; chs
< q
->nb_channels
; chs
++)
923 for (k
= 0; k
< run
; k
++)
925 q
->sb_samples
[chs
][j
+ k
][sb
] = q
->tone_level
[chs
][sb
][((j
+ k
)/2)] * tmp
[k
][chs
];
927 for (k
= 0; k
< run
; k
++)
929 q
->sb_samples
[ch
][j
+ k
][sb
] = q
->tone_level
[ch
][sb
][(j
+ k
)/2] * samples
[k
];
940 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
941 * This is similar to process_subpacket_9, but for a single channel and for element [0]
942 * same VLC tables as process_subpacket_9 are used.
944 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
945 * @param gb bitreader context
947 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs
, GetBitContext
*gb
)
949 int i
, k
, run
, level
, diff
;
951 if (get_bits_left(gb
) < 16)
953 level
= qdm2_get_vlc(gb
, &vlc_tab_level
, 0, 2);
955 quantized_coeffs
[0] = level
;
957 for (i
= 0; i
< 7; ) {
958 if (get_bits_left(gb
) < 16)
960 run
= qdm2_get_vlc(gb
, &vlc_tab_run
, 0, 1) + 1;
962 if (get_bits_left(gb
) < 16)
964 diff
= qdm2_get_se_vlc(&vlc_tab_diff
, gb
, 2);
966 for (k
= 1; k
<= run
; k
++)
967 quantized_coeffs
[i
+ k
] = (level
+ ((k
* diff
) / run
));
976 * Related to synthesis filter, process data from packet 10
977 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
978 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
981 * @param gb bitreader context
983 static void init_tone_level_dequantization (QDM2Context
*q
, GetBitContext
*gb
)
987 for (ch
= 0; ch
< q
->nb_channels
; ch
++) {
988 init_quantized_coeffs_elem0(q
->quantized_coeffs
[ch
][0], gb
);
990 if (get_bits_left(gb
) < 16) {
991 memset(q
->quantized_coeffs
[ch
][0], 0, 8);
996 n
= q
->sub_sampling
+ 1;
998 for (sb
= 0; sb
< n
; sb
++)
999 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
1000 for (j
= 0; j
< 8; j
++) {
1001 if (get_bits_left(gb
) < 1)
1003 if (get_bits1(gb
)) {
1004 for (k
=0; k
< 8; k
++) {
1005 if (get_bits_left(gb
) < 16)
1007 q
->tone_level_idx_hi1
[ch
][sb
][j
][k
] = qdm2_get_vlc(gb
, &vlc_tab_tone_level_idx_hi1
, 0, 2);
1010 for (k
=0; k
< 8; k
++)
1011 q
->tone_level_idx_hi1
[ch
][sb
][j
][k
] = 0;
1015 n
= QDM2_SB_USED(q
->sub_sampling
) - 4;
1017 for (sb
= 0; sb
< n
; sb
++)
1018 for (ch
= 0; ch
< q
->nb_channels
; ch
++) {
1019 if (get_bits_left(gb
) < 16)
1021 q
->tone_level_idx_hi2
[ch
][sb
] = qdm2_get_vlc(gb
, &vlc_tab_tone_level_idx_hi2
, 0, 2);
1023 q
->tone_level_idx_hi2
[ch
][sb
] -= 16;
1025 for (j
= 0; j
< 8; j
++)
1026 q
->tone_level_idx_mid
[ch
][sb
][j
] = -16;
1029 n
= QDM2_SB_USED(q
->sub_sampling
) - 5;
1031 for (sb
= 0; sb
< n
; sb
++)
1032 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
1033 for (j
= 0; j
< 8; j
++) {
1034 if (get_bits_left(gb
) < 16)
1036 q
->tone_level_idx_mid
[ch
][sb
][j
] = qdm2_get_vlc(gb
, &vlc_tab_tone_level_idx_mid
, 0, 2) - 32;
1041 * Process subpacket 9, init quantized_coeffs with data from it
1044 * @param node pointer to node with packet
1046 static void process_subpacket_9 (QDM2Context
*q
, QDM2SubPNode
*node
)
1049 int i
, j
, k
, n
, ch
, run
, level
, diff
;
1051 init_get_bits(&gb
, node
->packet
->data
, node
->packet
->size
*8);
1053 n
= coeff_per_sb_for_avg
[q
->coeff_per_sb_select
][QDM2_SB_USED(q
->sub_sampling
) - 1] + 1; // same as averagesomething function
1055 for (i
= 1; i
< n
; i
++)
1056 for (ch
=0; ch
< q
->nb_channels
; ch
++) {
1057 level
= qdm2_get_vlc(&gb
, &vlc_tab_level
, 0, 2);
1058 q
->quantized_coeffs
[ch
][i
][0] = level
;
1060 for (j
= 0; j
< (8 - 1); ) {
1061 run
= qdm2_get_vlc(&gb
, &vlc_tab_run
, 0, 1) + 1;
1062 diff
= qdm2_get_se_vlc(&vlc_tab_diff
, &gb
, 2);
1064 for (k
= 1; k
<= run
; k
++)
1065 q
->quantized_coeffs
[ch
][i
][j
+ k
] = (level
+ ((k
*diff
) / run
));
1072 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
1073 for (i
= 0; i
< 8; i
++)
1074 q
->quantized_coeffs
[ch
][0][i
] = 0;
1079 * Process subpacket 10 if not null, else
1082 * @param node pointer to node with packet
1084 static void process_subpacket_10 (QDM2Context
*q
, QDM2SubPNode
*node
)
1089 init_get_bits(&gb
, node
->packet
->data
, node
->packet
->size
* 8);
1090 init_tone_level_dequantization(q
, &gb
);
1091 fill_tone_level_array(q
, 1);
1093 fill_tone_level_array(q
, 0);
1099 * Process subpacket 11
1102 * @param node pointer to node with packet
1104 static void process_subpacket_11 (QDM2Context
*q
, QDM2SubPNode
*node
)
1110 length
= node
->packet
->size
* 8;
1111 init_get_bits(&gb
, node
->packet
->data
, length
);
1115 int c
= get_bits (&gb
, 13);
1118 fill_coding_method_array (q
->tone_level_idx
, q
->tone_level_idx_temp
, q
->coding_method
,
1119 q
->nb_channels
, 8*c
, q
->superblocktype_2_3
, q
->cm_table_select
);
1122 synthfilt_build_sb_samples(q
, &gb
, length
, 0, 8);
1127 * Process subpacket 12
1130 * @param node pointer to node with packet
1132 static void process_subpacket_12 (QDM2Context
*q
, QDM2SubPNode
*node
)
1138 length
= node
->packet
->size
* 8;
1139 init_get_bits(&gb
, node
->packet
->data
, length
);
1142 synthfilt_build_sb_samples(q
, &gb
, length
, 8, QDM2_SB_USED(q
->sub_sampling
));
1146 * Process new subpackets for synthesis filter
1149 * @param list list with synthesis filter packets (list D)
1151 static void process_synthesis_subpackets (QDM2Context
*q
, QDM2SubPNode
*list
)
1153 QDM2SubPNode
*nodes
[4];
1155 nodes
[0] = qdm2_search_subpacket_type_in_list(list
, 9);
1156 if (nodes
[0] != NULL
)
1157 process_subpacket_9(q
, nodes
[0]);
1159 nodes
[1] = qdm2_search_subpacket_type_in_list(list
, 10);
1160 if (nodes
[1] != NULL
)
1161 process_subpacket_10(q
, nodes
[1]);
1163 process_subpacket_10(q
, NULL
);
1165 nodes
[2] = qdm2_search_subpacket_type_in_list(list
, 11);
1166 if (nodes
[0] != NULL
&& nodes
[1] != NULL
&& nodes
[2] != NULL
)
1167 process_subpacket_11(q
, nodes
[2]);
1169 process_subpacket_11(q
, NULL
);
1171 nodes
[3] = qdm2_search_subpacket_type_in_list(list
, 12);
1172 if (nodes
[0] != NULL
&& nodes
[1] != NULL
&& nodes
[3] != NULL
)
1173 process_subpacket_12(q
, nodes
[3]);
1175 process_subpacket_12(q
, NULL
);
1180 * Decode superblock, fill packet lists.
1184 static void qdm2_decode_super_block (QDM2Context
*q
)
1187 QDM2SubPacket header
, *packet
;
1188 int i
, packet_bytes
, sub_packet_size
, sub_packets_D
;
1189 unsigned int next_index
= 0;
1191 memset(q
->tone_level_idx_hi1
, 0, sizeof(q
->tone_level_idx_hi1
));
1192 memset(q
->tone_level_idx_mid
, 0, sizeof(q
->tone_level_idx_mid
));
1193 memset(q
->tone_level_idx_hi2
, 0, sizeof(q
->tone_level_idx_hi2
));
1195 q
->sub_packets_B
= 0;
1198 average_quantized_coeffs(q
); // average elements in quantized_coeffs[max_ch][10][8]
1200 init_get_bits(&gb
, q
->compressed_data
, q
->compressed_size
*8);
1201 qdm2_decode_sub_packet_header(&gb
, &header
);
1203 if (header
.type
< 2 || header
.type
>= 8) {
1205 av_log(NULL
,AV_LOG_ERROR
,"bad superblock type\n");
1209 q
->superblocktype_2_3
= (header
.type
== 2 || header
.type
== 3);
1210 packet_bytes
= (q
->compressed_size
- get_bits_count(&gb
) / 8);
1212 init_get_bits(&gb
, header
.data
, header
.size
*8);
1214 if (header
.type
== 2 || header
.type
== 4 || header
.type
== 5) {
1215 int csum
= 257 * get_bits(&gb
, 8);
1216 csum
+= 2 * get_bits(&gb
, 8);
1218 csum
= qdm2_packet_checksum(q
->compressed_data
, q
->checksum_size
, csum
);
1222 av_log(NULL
,AV_LOG_ERROR
,"bad packet checksum\n");
1227 q
->sub_packet_list_B
[0].packet
= NULL
;
1228 q
->sub_packet_list_D
[0].packet
= NULL
;
1230 for (i
= 0; i
< 6; i
++)
1231 if (--q
->fft_level_exp
[i
] < 0)
1232 q
->fft_level_exp
[i
] = 0;
1234 for (i
= 0; packet_bytes
> 0; i
++) {
1237 q
->sub_packet_list_A
[i
].next
= NULL
;
1240 q
->sub_packet_list_A
[i
- 1].next
= &q
->sub_packet_list_A
[i
];
1242 /* seek to next block */
1243 init_get_bits(&gb
, header
.data
, header
.size
*8);
1244 skip_bits(&gb
, next_index
*8);
1246 if (next_index
>= header
.size
)
1250 /* decode subpacket */
1251 packet
= &q
->sub_packets
[i
];
1252 qdm2_decode_sub_packet_header(&gb
, packet
);
1253 next_index
= packet
->size
+ get_bits_count(&gb
) / 8;
1254 sub_packet_size
= ((packet
->size
> 0xff) ? 1 : 0) + packet
->size
+ 2;
1256 if (packet
->type
== 0)
1259 if (sub_packet_size
> packet_bytes
) {
1260 if (packet
->type
!= 10 && packet
->type
!= 11 && packet
->type
!= 12)
1262 packet
->size
+= packet_bytes
- sub_packet_size
;
1265 packet_bytes
-= sub_packet_size
;
1267 /* add subpacket to 'all subpackets' list */
1268 q
->sub_packet_list_A
[i
].packet
= packet
;
1270 /* add subpacket to related list */
1271 if (packet
->type
== 8) {
1272 SAMPLES_NEEDED_2("packet type 8");
1274 } else if (packet
->type
>= 9 && packet
->type
<= 12) {
1275 /* packets for MPEG Audio like Synthesis Filter */
1276 QDM2_LIST_ADD(q
->sub_packet_list_D
, sub_packets_D
, packet
);
1277 } else if (packet
->type
== 13) {
1278 for (j
= 0; j
< 6; j
++)
1279 q
->fft_level_exp
[j
] = get_bits(&gb
, 6);
1280 } else if (packet
->type
== 14) {
1281 for (j
= 0; j
< 6; j
++)
1282 q
->fft_level_exp
[j
] = qdm2_get_vlc(&gb
, &fft_level_exp_vlc
, 0, 2);
1283 } else if (packet
->type
== 15) {
1284 SAMPLES_NEEDED_2("packet type 15")
1286 } else if (packet
->type
>= 16 && packet
->type
< 48 && !fft_subpackets
[packet
->type
- 16]) {
1287 /* packets for FFT */
1288 QDM2_LIST_ADD(q
->sub_packet_list_B
, q
->sub_packets_B
, packet
);
1290 } // Packet bytes loop
1292 /* **************************************************************** */
1293 if (q
->sub_packet_list_D
[0].packet
!= NULL
) {
1294 process_synthesis_subpackets(q
, q
->sub_packet_list_D
);
1295 q
->do_synth_filter
= 1;
1296 } else if (q
->do_synth_filter
) {
1297 process_subpacket_10(q
, NULL
);
1298 process_subpacket_11(q
, NULL
);
1299 process_subpacket_12(q
, NULL
);
1301 /* **************************************************************** */
1305 static void qdm2_fft_init_coefficient (QDM2Context
*q
, int sub_packet
,
1306 int offset
, int duration
, int channel
,
1309 if (q
->fft_coefs_min_index
[duration
] < 0)
1310 q
->fft_coefs_min_index
[duration
] = q
->fft_coefs_index
;
1312 q
->fft_coefs
[q
->fft_coefs_index
].sub_packet
= ((sub_packet
>= 16) ? (sub_packet
- 16) : sub_packet
);
1313 q
->fft_coefs
[q
->fft_coefs_index
].channel
= channel
;
1314 q
->fft_coefs
[q
->fft_coefs_index
].offset
= offset
;
1315 q
->fft_coefs
[q
->fft_coefs_index
].exp
= exp
;
1316 q
->fft_coefs
[q
->fft_coefs_index
].phase
= phase
;
1317 q
->fft_coefs_index
++;
1321 static void qdm2_fft_decode_tones (QDM2Context
*q
, int duration
, GetBitContext
*gb
, int b
)
1323 int channel
, stereo
, phase
, exp
;
1324 int local_int_4
, local_int_8
, stereo_phase
, local_int_10
;
1325 int local_int_14
, stereo_exp
, local_int_20
, local_int_28
;
1331 local_int_8
= (4 - duration
);
1332 local_int_10
= 1 << (q
->group_order
- duration
- 1);
1336 if (q
->superblocktype_2_3
) {
1337 while ((n
= qdm2_get_vlc(gb
, &vlc_tab_fft_tone_offset
[local_int_8
], 1, 2)) < 2) {
1340 local_int_4
+= local_int_10
;
1341 local_int_28
+= (1 << local_int_8
);
1343 local_int_4
+= 8*local_int_10
;
1344 local_int_28
+= (8 << local_int_8
);
1349 offset
+= qdm2_get_vlc(gb
, &vlc_tab_fft_tone_offset
[local_int_8
], 1, 2);
1350 while (offset
>= (local_int_10
- 1)) {
1351 offset
+= (1 - (local_int_10
- 1));
1352 local_int_4
+= local_int_10
;
1353 local_int_28
+= (1 << local_int_8
);
1357 if (local_int_4
>= q
->group_size
)
1360 local_int_14
= (offset
>> local_int_8
);
1361 if (local_int_14
>= FF_ARRAY_ELEMS(fft_level_index_table
))
1364 if (q
->nb_channels
> 1) {
1365 channel
= get_bits1(gb
);
1366 stereo
= get_bits1(gb
);
1372 exp
= qdm2_get_vlc(gb
, (b
? &fft_level_exp_vlc
: &fft_level_exp_alt_vlc
), 0, 2);
1373 exp
+= q
->fft_level_exp
[fft_level_index_table
[local_int_14
]];
1374 exp
= (exp
< 0) ? 0 : exp
;
1376 phase
= get_bits(gb
, 3);
1381 stereo_exp
= (exp
- qdm2_get_vlc(gb
, &fft_stereo_exp_vlc
, 0, 1));
1382 stereo_phase
= (phase
- qdm2_get_vlc(gb
, &fft_stereo_phase_vlc
, 0, 1));
1383 if (stereo_phase
< 0)
1387 if (q
->frequency_range
> (local_int_14
+ 1)) {
1388 int sub_packet
= (local_int_20
+ local_int_28
);
1390 qdm2_fft_init_coefficient(q
, sub_packet
, offset
, duration
, channel
, exp
, phase
);
1392 qdm2_fft_init_coefficient(q
, sub_packet
, offset
, duration
, (1 - channel
), stereo_exp
, stereo_phase
);
1400 static void qdm2_decode_fft_packets (QDM2Context
*q
)
1402 int i
, j
, min
, max
, value
, type
, unknown_flag
;
1405 if (q
->sub_packet_list_B
[0].packet
== NULL
)
1408 /* reset minimum indexes for FFT coefficients */
1409 q
->fft_coefs_index
= 0;
1410 for (i
=0; i
< 5; i
++)
1411 q
->fft_coefs_min_index
[i
] = -1;
1413 /* process subpackets ordered by type, largest type first */
1414 for (i
= 0, max
= 256; i
< q
->sub_packets_B
; i
++) {
1415 QDM2SubPacket
*packet
= NULL
;
1417 /* find subpacket with largest type less than max */
1418 for (j
= 0, min
= 0; j
< q
->sub_packets_B
; j
++) {
1419 value
= q
->sub_packet_list_B
[j
].packet
->type
;
1420 if (value
> min
&& value
< max
) {
1422 packet
= q
->sub_packet_list_B
[j
].packet
;
1428 /* check for errors (?) */
1432 if (i
== 0 && (packet
->type
< 16 || packet
->type
>= 48 || fft_subpackets
[packet
->type
- 16]))
1435 /* decode FFT tones */
1436 init_get_bits (&gb
, packet
->data
, packet
->size
*8);
1438 if (packet
->type
>= 32 && packet
->type
< 48 && !fft_subpackets
[packet
->type
- 16])
1443 type
= packet
->type
;
1445 if ((type
>= 17 && type
< 24) || (type
>= 33 && type
< 40)) {
1446 int duration
= q
->sub_sampling
+ 5 - (type
& 15);
1448 if (duration
>= 0 && duration
< 4)
1449 qdm2_fft_decode_tones(q
, duration
, &gb
, unknown_flag
);
1450 } else if (type
== 31) {
1451 for (j
=0; j
< 4; j
++)
1452 qdm2_fft_decode_tones(q
, j
, &gb
, unknown_flag
);
1453 } else if (type
== 46) {
1454 for (j
=0; j
< 6; j
++)
1455 q
->fft_level_exp
[j
] = get_bits(&gb
, 6);
1456 for (j
=0; j
< 4; j
++)
1457 qdm2_fft_decode_tones(q
, j
, &gb
, unknown_flag
);
1459 } // Loop on B packets
1461 /* calculate maximum indexes for FFT coefficients */
1462 for (i
= 0, j
= -1; i
< 5; i
++)
1463 if (q
->fft_coefs_min_index
[i
] >= 0) {
1465 q
->fft_coefs_max_index
[j
] = q
->fft_coefs_min_index
[i
];
1469 q
->fft_coefs_max_index
[j
] = q
->fft_coefs_index
;
1473 static void qdm2_fft_generate_tone (QDM2Context
*q
, FFTTone
*tone
)
1478 const double iscale
= 2.0*M_PI
/ 512.0;
1480 tone
->phase
+= tone
->phase_shift
;
1482 /* calculate current level (maximum amplitude) of tone */
1483 level
= fft_tone_envelope_table
[tone
->duration
][tone
->time_index
] * tone
->level
;
1484 c
.im
= level
* sin(tone
->phase
*iscale
);
1485 c
.re
= level
* cos(tone
->phase
*iscale
);
1487 /* generate FFT coefficients for tone */
1488 if (tone
->duration
>= 3 || tone
->cutoff
>= 3) {
1489 tone
->complex[0].im
+= c
.im
;
1490 tone
->complex[0].re
+= c
.re
;
1491 tone
->complex[1].im
-= c
.im
;
1492 tone
->complex[1].re
-= c
.re
;
1494 f
[1] = -tone
->table
[4];
1495 f
[0] = tone
->table
[3] - tone
->table
[0];
1496 f
[2] = 1.0 - tone
->table
[2] - tone
->table
[3];
1497 f
[3] = tone
->table
[1] + tone
->table
[4] - 1.0;
1498 f
[4] = tone
->table
[0] - tone
->table
[1];
1499 f
[5] = tone
->table
[2];
1500 for (i
= 0; i
< 2; i
++) {
1501 tone
->complex[fft_cutoff_index_table
[tone
->cutoff
][i
]].re
+= c
.re
* f
[i
];
1502 tone
->complex[fft_cutoff_index_table
[tone
->cutoff
][i
]].im
+= c
.im
*((tone
->cutoff
<= i
) ? -f
[i
] : f
[i
]);
1504 for (i
= 0; i
< 4; i
++) {
1505 tone
->complex[i
].re
+= c
.re
* f
[i
+2];
1506 tone
->complex[i
].im
+= c
.im
* f
[i
+2];
1510 /* copy the tone if it has not yet died out */
1511 if (++tone
->time_index
< ((1 << (5 - tone
->duration
)) - 1)) {
1512 memcpy(&q
->fft_tones
[q
->fft_tone_end
], tone
, sizeof(FFTTone
));
1513 q
->fft_tone_end
= (q
->fft_tone_end
+ 1) % 1000;
1518 static void qdm2_fft_tone_synthesizer (QDM2Context
*q
, int sub_packet
)
1521 const double iscale
= 0.25 * M_PI
;
1523 for (ch
= 0; ch
< q
->channels
; ch
++) {
1524 memset(q
->fft
.complex[ch
], 0, q
->fft_size
* sizeof(QDM2Complex
));
1528 /* apply FFT tones with duration 4 (1 FFT period) */
1529 if (q
->fft_coefs_min_index
[4] >= 0)
1530 for (i
= q
->fft_coefs_min_index
[4]; i
< q
->fft_coefs_max_index
[4]; i
++) {
1534 if (q
->fft_coefs
[i
].sub_packet
!= sub_packet
)
1537 ch
= (q
->channels
== 1) ? 0 : q
->fft_coefs
[i
].channel
;
1538 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];
1540 c
.re
= level
* cos(q
->fft_coefs
[i
].phase
* iscale
);
1541 c
.im
= level
* sin(q
->fft_coefs
[i
].phase
* iscale
);
1542 q
->fft
.complex[ch
][q
->fft_coefs
[i
].offset
+ 0].re
+= c
.re
;
1543 q
->fft
.complex[ch
][q
->fft_coefs
[i
].offset
+ 0].im
+= c
.im
;
1544 q
->fft
.complex[ch
][q
->fft_coefs
[i
].offset
+ 1].re
-= c
.re
;
1545 q
->fft
.complex[ch
][q
->fft_coefs
[i
].offset
+ 1].im
-= c
.im
;
1548 /* generate existing FFT tones */
1549 for (i
= q
->fft_tone_end
; i
!= q
->fft_tone_start
; ) {
1550 qdm2_fft_generate_tone(q
, &q
->fft_tones
[q
->fft_tone_start
]);
1551 q
->fft_tone_start
= (q
->fft_tone_start
+ 1) % 1000;
1554 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1555 for (i
= 0; i
< 4; i
++)
1556 if (q
->fft_coefs_min_index
[i
] >= 0) {
1557 for (j
= q
->fft_coefs_min_index
[i
]; j
< q
->fft_coefs_max_index
[i
]; j
++) {
1561 if (q
->fft_coefs
[j
].sub_packet
!= sub_packet
)
1565 offset
= q
->fft_coefs
[j
].offset
>> four_i
;
1566 ch
= (q
->channels
== 1) ? 0 : q
->fft_coefs
[j
].channel
;
1568 if (offset
< q
->frequency_range
) {
1570 tone
.cutoff
= offset
;
1572 tone
.cutoff
= (offset
>= 60) ? 3 : 2;
1574 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];
1575 tone
.complex = &q
->fft
.complex[ch
][offset
];
1576 tone
.table
= fft_tone_sample_table
[i
][q
->fft_coefs
[j
].offset
- (offset
<< four_i
)];
1577 tone
.phase
= 64 * q
->fft_coefs
[j
].phase
- (offset
<< 8) - 128;
1578 tone
.phase_shift
= (2 * q
->fft_coefs
[j
].offset
+ 1) << (7 - four_i
);
1580 tone
.time_index
= 0;
1582 qdm2_fft_generate_tone(q
, &tone
);
1585 q
->fft_coefs_min_index
[i
] = j
;
1590 static void qdm2_calculate_fft (QDM2Context
*q
, int channel
, int sub_packet
)
1592 const float gain
= (q
->channels
== 1 && q
->nb_channels
== 2) ? 0.5f
: 1.0f
;
1593 float *out
= q
->output_buffer
+ channel
;
1595 q
->fft
.complex[channel
][0].re
*= 2.0f
;
1596 q
->fft
.complex[channel
][0].im
= 0.0f
;
1597 q
->rdft_ctx
.rdft_calc(&q
->rdft_ctx
, (FFTSample
*)q
->fft
.complex[channel
]);
1598 /* add samples to output buffer */
1599 for (i
= 0; i
< FFALIGN(q
->fft_size
, 8); i
++) {
1600 out
[0] += q
->fft
.complex[channel
][i
].re
* gain
;
1601 out
[q
->channels
] += q
->fft
.complex[channel
][i
].im
* gain
;
1602 out
+= 2 * q
->channels
;
1609 * @param index subpacket number
1611 static void qdm2_synthesis_filter (QDM2Context
*q
, int index
)
1613 int i
, k
, ch
, sb_used
, sub_sampling
, dither_state
= 0;
1615 /* copy sb_samples */
1616 sb_used
= QDM2_SB_USED(q
->sub_sampling
);
1618 for (ch
= 0; ch
< q
->channels
; ch
++)
1619 for (i
= 0; i
< 8; i
++)
1620 for (k
=sb_used
; k
< SBLIMIT
; k
++)
1621 q
->sb_samples
[ch
][(8 * index
) + i
][k
] = 0;
1623 for (ch
= 0; ch
< q
->nb_channels
; ch
++) {
1624 float *samples_ptr
= q
->samples
+ ch
;
1626 for (i
= 0; i
< 8; i
++) {
1627 ff_mpa_synth_filter_float(&q
->mpadsp
,
1628 q
->synth_buf
[ch
], &(q
->synth_buf_offset
[ch
]),
1629 ff_mpa_synth_window_float
, &dither_state
,
1630 samples_ptr
, q
->nb_channels
,
1631 q
->sb_samples
[ch
][(8 * index
) + i
]);
1632 samples_ptr
+= 32 * q
->nb_channels
;
1636 /* add samples to output buffer */
1637 sub_sampling
= (4 >> q
->sub_sampling
);
1639 for (ch
= 0; ch
< q
->channels
; ch
++)
1640 for (i
= 0; i
< q
->frame_size
; i
++)
1641 q
->output_buffer
[q
->channels
* i
+ ch
] += (1 << 23) * q
->samples
[q
->nb_channels
* sub_sampling
* i
+ ch
];
1646 * Init static data (does not depend on specific file)
1650 static av_cold
void qdm2_init(QDM2Context
*q
) {
1651 static int initialized
= 0;
1653 if (initialized
!= 0)
1658 ff_mpa_synth_init_float(ff_mpa_synth_window_float
);
1659 softclip_table_init();
1661 init_noise_samples();
1663 av_log(NULL
, AV_LOG_DEBUG
, "init done\n");
1668 * Init parameters from codec extradata
1670 static av_cold
int qdm2_decode_init(AVCodecContext
*avctx
)
1672 QDM2Context
*s
= avctx
->priv_data
;
1675 int tmp_val
, tmp
, size
;
1677 /* extradata parsing
1686 32 size (including this field)
1688 32 type (=QDM2 or QDMC)
1690 32 size (including this field, in bytes)
1691 32 tag (=QDCA) // maybe mandatory parameters
1694 32 samplerate (=44100)
1696 32 block size (=4096)
1697 32 frame size (=256) (for one channel)
1698 32 packet size (=1300)
1700 32 size (including this field, in bytes)
1701 32 tag (=QDCP) // maybe some tuneable parameters
1711 if (!avctx
->extradata
|| (avctx
->extradata_size
< 48)) {
1712 av_log(avctx
, AV_LOG_ERROR
, "extradata missing or truncated\n");
1716 extradata
= avctx
->extradata
;
1717 extradata_size
= avctx
->extradata_size
;
1719 while (extradata_size
> 7) {
1720 if (!memcmp(extradata
, "frmaQDM", 7))
1726 if (extradata_size
< 12) {
1727 av_log(avctx
, AV_LOG_ERROR
, "not enough extradata (%i)\n",
1732 if (memcmp(extradata
, "frmaQDM", 7)) {
1733 av_log(avctx
, AV_LOG_ERROR
, "invalid headers, QDM? not found\n");
1737 if (extradata
[7] == 'C') {
1739 av_log(avctx
, AV_LOG_ERROR
, "stream is QDMC version 1, which is not supported\n");
1744 extradata_size
-= 8;
1746 size
= AV_RB32(extradata
);
1748 if(size
> extradata_size
){
1749 av_log(avctx
, AV_LOG_ERROR
, "extradata size too small, %i < %i\n",
1750 extradata_size
, size
);
1755 av_log(avctx
, AV_LOG_DEBUG
, "size: %d\n", size
);
1756 if (AV_RB32(extradata
) != MKBETAG('Q','D','C','A')) {
1757 av_log(avctx
, AV_LOG_ERROR
, "invalid extradata, expecting QDCA\n");
1763 avctx
->channels
= s
->nb_channels
= s
->channels
= AV_RB32(extradata
);
1765 if (s
->channels
<= 0 || s
->channels
> MPA_MAX_CHANNELS
)
1766 return AVERROR_INVALIDDATA
;
1767 avctx
->channel_layout
= avctx
->channels
== 2 ? AV_CH_LAYOUT_STEREO
:
1770 avctx
->sample_rate
= AV_RB32(extradata
);
1773 avctx
->bit_rate
= AV_RB32(extradata
);
1776 s
->group_size
= AV_RB32(extradata
);
1779 s
->fft_size
= AV_RB32(extradata
);
1782 s
->checksum_size
= AV_RB32(extradata
);
1783 if (s
->checksum_size
>= 1U << 28) {
1784 av_log(avctx
, AV_LOG_ERROR
, "data block size too large (%u)\n", s
->checksum_size
);
1785 return AVERROR_INVALIDDATA
;
1788 s
->fft_order
= av_log2(s
->fft_size
) + 1;
1790 // something like max decodable tones
1791 s
->group_order
= av_log2(s
->group_size
) + 1;
1792 s
->frame_size
= s
->group_size
/ 16; // 16 iterations per super block
1793 if (s
->frame_size
> QDM2_MAX_FRAME_SIZE
)
1794 return AVERROR_INVALIDDATA
;
1796 s
->sub_sampling
= s
->fft_order
- 7;
1797 s
->frequency_range
= 255 / (1 << (2 - s
->sub_sampling
));
1799 switch ((s
->sub_sampling
* 2 + s
->channels
- 1)) {
1800 case 0: tmp
= 40; break;
1801 case 1: tmp
= 48; break;
1802 case 2: tmp
= 56; break;
1803 case 3: tmp
= 72; break;
1804 case 4: tmp
= 80; break;
1805 case 5: tmp
= 100;break;
1806 default: tmp
=s
->sub_sampling
; break;
1809 if ((tmp
* 1000) < avctx
->bit_rate
) tmp_val
= 1;
1810 if ((tmp
* 1440) < avctx
->bit_rate
) tmp_val
= 2;
1811 if ((tmp
* 1760) < avctx
->bit_rate
) tmp_val
= 3;
1812 if ((tmp
* 2240) < avctx
->bit_rate
) tmp_val
= 4;
1813 s
->cm_table_select
= tmp_val
;
1815 if (s
->sub_sampling
== 0)
1818 tmp
= ((-(s
->sub_sampling
-1)) & 8000) + 20000;
1825 s
->coeff_per_sb_select
= 0;
1826 else if (tmp
<= 16000)
1827 s
->coeff_per_sb_select
= 1;
1829 s
->coeff_per_sb_select
= 2;
1831 // Fail on unknown fft order
1832 if ((s
->fft_order
< 7) || (s
->fft_order
> 9)) {
1833 av_log(avctx
, AV_LOG_ERROR
, "Unknown FFT order (%d), contact the developers!\n", s
->fft_order
);
1837 ff_rdft_init(&s
->rdft_ctx
, s
->fft_order
, IDFT_C2R
);
1838 ff_mpadsp_init(&s
->mpadsp
);
1842 avctx
->sample_fmt
= AV_SAMPLE_FMT_S16
;
1844 avcodec_get_frame_defaults(&s
->frame
);
1845 avctx
->coded_frame
= &s
->frame
;
1851 static av_cold
int qdm2_decode_close(AVCodecContext
*avctx
)
1853 QDM2Context
*s
= avctx
->priv_data
;
1855 ff_rdft_end(&s
->rdft_ctx
);
1861 static int qdm2_decode (QDM2Context
*q
, const uint8_t *in
, int16_t *out
)
1864 const int frame_size
= (q
->frame_size
* q
->channels
);
1866 /* select input buffer */
1867 q
->compressed_data
= in
;
1868 q
->compressed_size
= q
->checksum_size
;
1870 /* copy old block, clear new block of output samples */
1871 memmove(q
->output_buffer
, &q
->output_buffer
[frame_size
], frame_size
* sizeof(float));
1872 memset(&q
->output_buffer
[frame_size
], 0, frame_size
* sizeof(float));
1874 /* decode block of QDM2 compressed data */
1875 if (q
->sub_packet
== 0) {
1876 q
->has_errors
= 0; // zero it for a new super block
1877 av_log(NULL
,AV_LOG_DEBUG
,"Superblock follows\n");
1878 qdm2_decode_super_block(q
);
1881 /* parse subpackets */
1882 if (!q
->has_errors
) {
1883 if (q
->sub_packet
== 2)
1884 qdm2_decode_fft_packets(q
);
1886 qdm2_fft_tone_synthesizer(q
, q
->sub_packet
);
1889 /* sound synthesis stage 1 (FFT) */
1890 for (ch
= 0; ch
< q
->channels
; ch
++) {
1891 qdm2_calculate_fft(q
, ch
, q
->sub_packet
);
1893 if (!q
->has_errors
&& q
->sub_packet_list_C
[0].packet
!= NULL
) {
1894 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1899 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1900 if (!q
->has_errors
&& q
->do_synth_filter
)
1901 qdm2_synthesis_filter(q
, q
->sub_packet
);
1903 q
->sub_packet
= (q
->sub_packet
+ 1) % 16;
1905 /* clip and convert output float[] to 16bit signed samples */
1906 for (i
= 0; i
< frame_size
; i
++) {
1907 int value
= (int)q
->output_buffer
[i
];
1909 if (value
> SOFTCLIP_THRESHOLD
)
1910 value
= (value
> HARDCLIP_THRESHOLD
) ? 32767 : softclip_table
[ value
- SOFTCLIP_THRESHOLD
];
1911 else if (value
< -SOFTCLIP_THRESHOLD
)
1912 value
= (value
< -HARDCLIP_THRESHOLD
) ? -32767 : -softclip_table
[-value
- SOFTCLIP_THRESHOLD
];
1921 static int qdm2_decode_frame(AVCodecContext
*avctx
, void *data
,
1922 int *got_frame_ptr
, AVPacket
*avpkt
)
1924 const uint8_t *buf
= avpkt
->data
;
1925 int buf_size
= avpkt
->size
;
1926 QDM2Context
*s
= avctx
->priv_data
;
1932 if(buf_size
< s
->checksum_size
)
1935 /* get output buffer */
1936 s
->frame
.nb_samples
= 16 * s
->frame_size
;
1937 if ((ret
= ff_get_buffer(avctx
, &s
->frame
)) < 0) {
1938 av_log(avctx
, AV_LOG_ERROR
, "get_buffer() failed\n");
1941 out
= (int16_t *)s
->frame
.data
[0];
1943 for (i
= 0; i
< 16; i
++) {
1944 if (qdm2_decode(s
, buf
, out
) < 0)
1946 out
+= s
->channels
* s
->frame_size
;
1950 *(AVFrame
*)data
= s
->frame
;
1952 return s
->checksum_size
;
1955 AVCodec ff_qdm2_decoder
=
1958 .type
= AVMEDIA_TYPE_AUDIO
,
1959 .id
= AV_CODEC_ID_QDM2
,
1960 .priv_data_size
= sizeof(QDM2Context
),
1961 .init
= qdm2_decode_init
,
1962 .close
= qdm2_decode_close
,
1963 .decode
= qdm2_decode_frame
,
1964 .capabilities
= CODEC_CAP_DR1
,
1965 .long_name
= NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),