2 * QDM2 compatible decoder
3 * Copyright (c) 2003 Ewald Snel
4 * Copyright (c) 2005 Benjamin Larsson
5 * Copyright (c) 2005 Alex Beregszaszi
6 * Copyright (c) 2005 Roberto Togni
8 * This file is part of FFmpeg.
10 * FFmpeg is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Lesser General Public
12 * License as published by the Free Software Foundation; either
13 * version 2.1 of the License, or (at your option) any later version.
15 * FFmpeg is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Lesser General Public License for more details.
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with FFmpeg; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
30 * The decoder is not perfect yet, there are still some distortions
31 * especially on files encoded with 16 or 8 subbands.
38 #define BITSTREAM_READER_LE
39 #include "libavutil/channel_layout.h"
44 #include "mpegaudiodsp.h"
45 #include "mpegaudio.h"
48 #include "qdm2_tablegen.h"
54 #define QDM2_LIST_ADD(list, size, packet) \
57 list[size - 1].next = &list[size]; \
59 list[size].packet = packet; \
60 list[size].next = NULL; \
64 // Result is 8, 16 or 30
65 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
67 #define FIX_NOISE_IDX(noise_idx) \
68 if ((noise_idx) >= 3840) \
69 (noise_idx) -= 3840; \
71 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
73 #define SAMPLES_NEEDED \
74 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
76 #define SAMPLES_NEEDED_2(why) \
77 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
79 #define QDM2_MAX_FRAME_SIZE 512
81 typedef int8_t sb_int8_array
[2][30][64];
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)
93 * A node in the subpacket list
95 typedef struct QDM2SubPNode
{
96 QDM2SubPacket
*packet
; ///< packet
97 struct QDM2SubPNode
*next
; ///< pointer to next packet in the list, NULL if leaf node
107 QDM2Complex
*complex;
125 DECLARE_ALIGNED(32, QDM2Complex
, complex)[MPA_MAX_CHANNELS
][256];
129 * QDM2 decoder context
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
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 frame_size
; ///< size of data frame
144 int sub_sampling
; ///< subsampling: 0=25%, 1=50%, 2=100% */
145 int coeff_per_sb_select
; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
146 int cm_table_select
; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
148 /// Packets and packet lists
149 QDM2SubPacket sub_packets
[16]; ///< the packets themselves
150 QDM2SubPNode sub_packet_list_A
[16]; ///< list of all packets
151 QDM2SubPNode sub_packet_list_B
[16]; ///< FFT packets B are on list
152 int sub_packets_B
; ///< number of packets on 'B' list
153 QDM2SubPNode sub_packet_list_C
[16]; ///< packets with errors?
154 QDM2SubPNode sub_packet_list_D
[16]; ///< DCT packets
157 FFTTone fft_tones
[1000];
160 FFTCoefficient fft_coefs
[1000];
162 int fft_coefs_min_index
[5];
163 int fft_coefs_max_index
[5];
164 int fft_level_exp
[6];
165 RDFTContext rdft_ctx
;
169 const uint8_t *compressed_data
;
171 float output_buffer
[QDM2_MAX_FRAME_SIZE
* MPA_MAX_CHANNELS
* 2];
174 MPADSPContext mpadsp
;
175 DECLARE_ALIGNED(32, float, synth_buf
)[MPA_MAX_CHANNELS
][512*2];
176 int synth_buf_offset
[MPA_MAX_CHANNELS
];
177 DECLARE_ALIGNED(32, float, sb_samples
)[MPA_MAX_CHANNELS
][128][SBLIMIT
];
178 DECLARE_ALIGNED(32, float, samples
)[MPA_MAX_CHANNELS
* MPA_FRAME_SIZE
];
180 /// Mixed temporary data used in decoding
181 float tone_level
[MPA_MAX_CHANNELS
][30][64];
182 int8_t coding_method
[MPA_MAX_CHANNELS
][30][64];
183 int8_t quantized_coeffs
[MPA_MAX_CHANNELS
][10][8];
184 int8_t tone_level_idx_base
[MPA_MAX_CHANNELS
][30][8];
185 int8_t tone_level_idx_hi1
[MPA_MAX_CHANNELS
][3][8][8];
186 int8_t tone_level_idx_mid
[MPA_MAX_CHANNELS
][26][8];
187 int8_t tone_level_idx_hi2
[MPA_MAX_CHANNELS
][26];
188 int8_t tone_level_idx
[MPA_MAX_CHANNELS
][30][64];
189 int8_t tone_level_idx_temp
[MPA_MAX_CHANNELS
][30][64];
192 int has_errors
; ///< packet has errors
193 int superblocktype_2_3
; ///< select fft tables and some algorithm based on superblock type
194 int do_synth_filter
; ///< used to perform or skip synthesis filter
197 int noise_idx
; ///< index for dithering noise table
201 static VLC vlc_tab_level
;
202 static VLC vlc_tab_diff
;
203 static VLC vlc_tab_run
;
204 static VLC fft_level_exp_alt_vlc
;
205 static VLC fft_level_exp_vlc
;
206 static VLC fft_stereo_exp_vlc
;
207 static VLC fft_stereo_phase_vlc
;
208 static VLC vlc_tab_tone_level_idx_hi1
;
209 static VLC vlc_tab_tone_level_idx_mid
;
210 static VLC vlc_tab_tone_level_idx_hi2
;
211 static VLC vlc_tab_type30
;
212 static VLC vlc_tab_type34
;
213 static VLC vlc_tab_fft_tone_offset
[5];
215 static const uint16_t qdm2_vlc_offs
[] = {
216 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
219 static const int switchtable
[23] = {
220 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
223 static av_cold
void qdm2_init_vlc(void)
225 static VLC_TYPE qdm2_table
[3838][2];
227 vlc_tab_level
.table
= &qdm2_table
[qdm2_vlc_offs
[0]];
228 vlc_tab_level
.table_allocated
= qdm2_vlc_offs
[1] - qdm2_vlc_offs
[0];
229 init_vlc(&vlc_tab_level
, 8, 24,
230 vlc_tab_level_huffbits
, 1, 1,
231 vlc_tab_level_huffcodes
, 2, 2,
232 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
234 vlc_tab_diff
.table
= &qdm2_table
[qdm2_vlc_offs
[1]];
235 vlc_tab_diff
.table_allocated
= qdm2_vlc_offs
[2] - qdm2_vlc_offs
[1];
236 init_vlc(&vlc_tab_diff
, 8, 37,
237 vlc_tab_diff_huffbits
, 1, 1,
238 vlc_tab_diff_huffcodes
, 2, 2,
239 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,
246 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
248 fft_level_exp_alt_vlc
.table
= &qdm2_table
[qdm2_vlc_offs
[3]];
249 fft_level_exp_alt_vlc
.table_allocated
= qdm2_vlc_offs
[4] -
251 init_vlc(&fft_level_exp_alt_vlc
, 8, 28,
252 fft_level_exp_alt_huffbits
, 1, 1,
253 fft_level_exp_alt_huffcodes
, 2, 2,
254 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
256 fft_level_exp_vlc
.table
= &qdm2_table
[qdm2_vlc_offs
[4]];
257 fft_level_exp_vlc
.table_allocated
= qdm2_vlc_offs
[5] - qdm2_vlc_offs
[4];
258 init_vlc(&fft_level_exp_vlc
, 8, 20,
259 fft_level_exp_huffbits
, 1, 1,
260 fft_level_exp_huffcodes
, 2, 2,
261 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
263 fft_stereo_exp_vlc
.table
= &qdm2_table
[qdm2_vlc_offs
[5]];
264 fft_stereo_exp_vlc
.table_allocated
= qdm2_vlc_offs
[6] -
266 init_vlc(&fft_stereo_exp_vlc
, 6, 7,
267 fft_stereo_exp_huffbits
, 1, 1,
268 fft_stereo_exp_huffcodes
, 1, 1,
269 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
271 fft_stereo_phase_vlc
.table
= &qdm2_table
[qdm2_vlc_offs
[6]];
272 fft_stereo_phase_vlc
.table_allocated
= qdm2_vlc_offs
[7] -
274 init_vlc(&fft_stereo_phase_vlc
, 6, 9,
275 fft_stereo_phase_huffbits
, 1, 1,
276 fft_stereo_phase_huffcodes
, 1, 1,
277 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
279 vlc_tab_tone_level_idx_hi1
.table
=
280 &qdm2_table
[qdm2_vlc_offs
[7]];
281 vlc_tab_tone_level_idx_hi1
.table_allocated
= qdm2_vlc_offs
[8] -
283 init_vlc(&vlc_tab_tone_level_idx_hi1
, 8, 20,
284 vlc_tab_tone_level_idx_hi1_huffbits
, 1, 1,
285 vlc_tab_tone_level_idx_hi1_huffcodes
, 2, 2,
286 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
288 vlc_tab_tone_level_idx_mid
.table
=
289 &qdm2_table
[qdm2_vlc_offs
[8]];
290 vlc_tab_tone_level_idx_mid
.table_allocated
= qdm2_vlc_offs
[9] -
292 init_vlc(&vlc_tab_tone_level_idx_mid
, 8, 24,
293 vlc_tab_tone_level_idx_mid_huffbits
, 1, 1,
294 vlc_tab_tone_level_idx_mid_huffcodes
, 2, 2,
295 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
297 vlc_tab_tone_level_idx_hi2
.table
=
298 &qdm2_table
[qdm2_vlc_offs
[9]];
299 vlc_tab_tone_level_idx_hi2
.table_allocated
= qdm2_vlc_offs
[10] -
301 init_vlc(&vlc_tab_tone_level_idx_hi2
, 8, 24,
302 vlc_tab_tone_level_idx_hi2_huffbits
, 1, 1,
303 vlc_tab_tone_level_idx_hi2_huffcodes
, 2, 2,
304 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
306 vlc_tab_type30
.table
= &qdm2_table
[qdm2_vlc_offs
[10]];
307 vlc_tab_type30
.table_allocated
= qdm2_vlc_offs
[11] - qdm2_vlc_offs
[10];
308 init_vlc(&vlc_tab_type30
, 6, 9,
309 vlc_tab_type30_huffbits
, 1, 1,
310 vlc_tab_type30_huffcodes
, 1, 1,
311 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
313 vlc_tab_type34
.table
= &qdm2_table
[qdm2_vlc_offs
[11]];
314 vlc_tab_type34
.table_allocated
= qdm2_vlc_offs
[12] - qdm2_vlc_offs
[11];
315 init_vlc(&vlc_tab_type34
, 5, 10,
316 vlc_tab_type34_huffbits
, 1, 1,
317 vlc_tab_type34_huffcodes
, 1, 1,
318 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
320 vlc_tab_fft_tone_offset
[0].table
=
321 &qdm2_table
[qdm2_vlc_offs
[12]];
322 vlc_tab_fft_tone_offset
[0].table_allocated
= qdm2_vlc_offs
[13] -
324 init_vlc(&vlc_tab_fft_tone_offset
[0], 8, 23,
325 vlc_tab_fft_tone_offset_0_huffbits
, 1, 1,
326 vlc_tab_fft_tone_offset_0_huffcodes
, 2, 2,
327 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
329 vlc_tab_fft_tone_offset
[1].table
=
330 &qdm2_table
[qdm2_vlc_offs
[13]];
331 vlc_tab_fft_tone_offset
[1].table_allocated
= qdm2_vlc_offs
[14] -
333 init_vlc(&vlc_tab_fft_tone_offset
[1], 8, 28,
334 vlc_tab_fft_tone_offset_1_huffbits
, 1, 1,
335 vlc_tab_fft_tone_offset_1_huffcodes
, 2, 2,
336 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
338 vlc_tab_fft_tone_offset
[2].table
=
339 &qdm2_table
[qdm2_vlc_offs
[14]];
340 vlc_tab_fft_tone_offset
[2].table_allocated
= qdm2_vlc_offs
[15] -
342 init_vlc(&vlc_tab_fft_tone_offset
[2], 8, 32,
343 vlc_tab_fft_tone_offset_2_huffbits
, 1, 1,
344 vlc_tab_fft_tone_offset_2_huffcodes
, 2, 2,
345 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
347 vlc_tab_fft_tone_offset
[3].table
=
348 &qdm2_table
[qdm2_vlc_offs
[15]];
349 vlc_tab_fft_tone_offset
[3].table_allocated
= qdm2_vlc_offs
[16] -
351 init_vlc(&vlc_tab_fft_tone_offset
[3], 8, 35,
352 vlc_tab_fft_tone_offset_3_huffbits
, 1, 1,
353 vlc_tab_fft_tone_offset_3_huffcodes
, 2, 2,
354 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
356 vlc_tab_fft_tone_offset
[4].table
=
357 &qdm2_table
[qdm2_vlc_offs
[16]];
358 vlc_tab_fft_tone_offset
[4].table_allocated
= qdm2_vlc_offs
[17] -
360 init_vlc(&vlc_tab_fft_tone_offset
[4], 8, 38,
361 vlc_tab_fft_tone_offset_4_huffbits
, 1, 1,
362 vlc_tab_fft_tone_offset_4_huffcodes
, 2, 2,
363 INIT_VLC_USE_NEW_STATIC
| INIT_VLC_LE
);
366 static int qdm2_get_vlc(GetBitContext
*gb
, VLC
*vlc
, int flag
, int depth
)
370 value
= get_vlc2(gb
, vlc
->table
, vlc
->bits
, depth
);
372 /* stage-2, 3 bits exponent escape sequence */
374 value
= get_bits(gb
, get_bits(gb
, 3) + 1);
376 /* stage-3, optional */
381 av_log(NULL
, AV_LOG_ERROR
, "value %d in qdm2_get_vlc too large\n", value
);
385 tmp
= vlc_stage3_values
[value
];
387 if ((value
& ~3) > 0)
388 tmp
+= get_bits(gb
, (value
>> 2));
395 static int qdm2_get_se_vlc(VLC
*vlc
, GetBitContext
*gb
, int depth
)
397 int value
= qdm2_get_vlc(gb
, vlc
, 0, depth
);
399 return (value
& 1) ? ((value
+ 1) >> 1) : -(value
>> 1);
405 * @param data pointer to data to be checksum'ed
406 * @param length data length
407 * @param value checksum value
409 * @return 0 if checksum is OK
411 static uint16_t qdm2_packet_checksum(const uint8_t *data
, int length
, int value
)
415 for (i
= 0; i
< length
; i
++)
418 return (uint16_t)(value
& 0xffff);
422 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
424 * @param gb bitreader context
425 * @param sub_packet packet under analysis
427 static void qdm2_decode_sub_packet_header(GetBitContext
*gb
,
428 QDM2SubPacket
*sub_packet
)
430 sub_packet
->type
= get_bits(gb
, 8);
432 if (sub_packet
->type
== 0) {
433 sub_packet
->size
= 0;
434 sub_packet
->data
= NULL
;
436 sub_packet
->size
= get_bits(gb
, 8);
438 if (sub_packet
->type
& 0x80) {
439 sub_packet
->size
<<= 8;
440 sub_packet
->size
|= get_bits(gb
, 8);
441 sub_packet
->type
&= 0x7f;
444 if (sub_packet
->type
== 0x7f)
445 sub_packet
->type
|= (get_bits(gb
, 8) << 8);
447 // FIXME: this depends on bitreader-internal data
448 sub_packet
->data
= &gb
->buffer
[get_bits_count(gb
) / 8];
451 av_log(NULL
, AV_LOG_DEBUG
, "Subpacket: type=%d size=%d start_offs=%x\n",
452 sub_packet
->type
, sub_packet
->size
, get_bits_count(gb
) / 8);
456 * Return node pointer to first packet of requested type in list.
458 * @param list list of subpackets to be scanned
459 * @param type type of searched subpacket
460 * @return node pointer for subpacket if found, else NULL
462 static QDM2SubPNode
*qdm2_search_subpacket_type_in_list(QDM2SubPNode
*list
,
465 while (list
&& list
->packet
) {
466 if (list
->packet
->type
== type
)
474 * Replace 8 elements with their average value.
475 * Called by qdm2_decode_superblock before starting subblock decoding.
479 static void average_quantized_coeffs(QDM2Context
*q
)
481 int i
, j
, n
, ch
, sum
;
483 n
= coeff_per_sb_for_avg
[q
->coeff_per_sb_select
][QDM2_SB_USED(q
->sub_sampling
) - 1] + 1;
485 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
486 for (i
= 0; i
< n
; i
++) {
489 for (j
= 0; j
< 8; j
++)
490 sum
+= q
->quantized_coeffs
[ch
][i
][j
];
496 for (j
= 0; j
< 8; j
++)
497 q
->quantized_coeffs
[ch
][i
][j
] = sum
;
502 * Build subband samples with noise weighted by q->tone_level.
503 * Called by synthfilt_build_sb_samples.
506 * @param sb subband index
508 static void build_sb_samples_from_noise(QDM2Context
*q
, int sb
)
512 FIX_NOISE_IDX(q
->noise_idx
);
517 for (ch
= 0; ch
< q
->nb_channels
; ch
++) {
518 for (j
= 0; j
< 64; j
++) {
519 q
->sb_samples
[ch
][j
* 2][sb
] =
520 SB_DITHERING_NOISE(sb
, q
->noise_idx
) * q
->tone_level
[ch
][sb
][j
];
521 q
->sb_samples
[ch
][j
* 2 + 1][sb
] =
522 SB_DITHERING_NOISE(sb
, q
->noise_idx
) * q
->tone_level
[ch
][sb
][j
];
528 * Called while processing data from subpackets 11 and 12.
529 * Used after making changes to coding_method array.
531 * @param sb subband index
532 * @param channels number of channels
533 * @param coding_method q->coding_method[0][0][0]
535 static int fix_coding_method_array(int sb
, int channels
,
536 sb_int8_array coding_method
)
542 for (ch
= 0; ch
< channels
; ch
++) {
543 for (j
= 0; j
< 64; ) {
544 if (coding_method
[ch
][sb
][j
] < 8)
546 if ((coding_method
[ch
][sb
][j
] - 8) > 22) {
550 switch (switchtable
[coding_method
[ch
][sb
][j
] - 8]) {
574 for (k
= 0; k
< run
; k
++) {
576 if (coding_method
[ch
][sb
+ (j
+ k
) / 64][(j
+ k
) % 64] > coding_method
[ch
][sb
][j
]) {
579 //not debugged, almost never used
580 memset(&coding_method
[ch
][sb
][j
+ k
], case_val
,
582 memset(&coding_method
[ch
][sb
][j
+ k
], case_val
,
595 * Related to synthesis filter
596 * Called by process_subpacket_10
599 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
601 static void fill_tone_level_array(QDM2Context
*q
, int flag
)
603 int i
, sb
, ch
, sb_used
;
606 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
607 for (sb
= 0; sb
< 30; sb
++)
608 for (i
= 0; i
< 8; i
++) {
609 if ((tab
=coeff_per_sb_for_dequant
[q
->coeff_per_sb_select
][sb
]) < (last_coeff
[q
->coeff_per_sb_select
] - 1))
610 tmp
= q
->quantized_coeffs
[ch
][tab
+ 1][i
] * dequant_table
[q
->coeff_per_sb_select
][tab
+ 1][sb
]+
611 q
->quantized_coeffs
[ch
][tab
][i
] * dequant_table
[q
->coeff_per_sb_select
][tab
][sb
];
613 tmp
= q
->quantized_coeffs
[ch
][tab
][i
] * dequant_table
[q
->coeff_per_sb_select
][tab
][sb
];
616 q
->tone_level_idx_base
[ch
][sb
][i
] = (tmp
/ 256) & 0xff;
619 sb_used
= QDM2_SB_USED(q
->sub_sampling
);
621 if ((q
->superblocktype_2_3
!= 0) && !flag
) {
622 for (sb
= 0; sb
< sb_used
; sb
++)
623 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
624 for (i
= 0; i
< 64; i
++) {
625 q
->tone_level_idx
[ch
][sb
][i
] = q
->tone_level_idx_base
[ch
][sb
][i
/ 8];
626 if (q
->tone_level_idx
[ch
][sb
][i
] < 0)
627 q
->tone_level
[ch
][sb
][i
] = 0;
629 q
->tone_level
[ch
][sb
][i
] = fft_tone_level_table
[0][q
->tone_level_idx
[ch
][sb
][i
] & 0x3f];
632 tab
= q
->superblocktype_2_3
? 0 : 1;
633 for (sb
= 0; sb
< sb_used
; sb
++) {
634 if ((sb
>= 4) && (sb
<= 23)) {
635 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
636 for (i
= 0; i
< 64; i
++) {
637 tmp
= q
->tone_level_idx_base
[ch
][sb
][i
/ 8] -
638 q
->tone_level_idx_hi1
[ch
][sb
/ 8][i
/ 8][i
% 8] -
639 q
->tone_level_idx_mid
[ch
][sb
- 4][i
/ 8] -
640 q
->tone_level_idx_hi2
[ch
][sb
- 4];
641 q
->tone_level_idx
[ch
][sb
][i
] = tmp
& 0xff;
642 if ((tmp
< 0) || (!q
->superblocktype_2_3
&& !tmp
))
643 q
->tone_level
[ch
][sb
][i
] = 0;
645 q
->tone_level
[ch
][sb
][i
] = fft_tone_level_table
[tab
][tmp
& 0x3f];
649 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
650 for (i
= 0; i
< 64; i
++) {
651 tmp
= q
->tone_level_idx_base
[ch
][sb
][i
/ 8] -
652 q
->tone_level_idx_hi1
[ch
][2][i
/ 8][i
% 8] -
653 q
->tone_level_idx_hi2
[ch
][sb
- 4];
654 q
->tone_level_idx
[ch
][sb
][i
] = tmp
& 0xff;
655 if ((tmp
< 0) || (!q
->superblocktype_2_3
&& !tmp
))
656 q
->tone_level
[ch
][sb
][i
] = 0;
658 q
->tone_level
[ch
][sb
][i
] = fft_tone_level_table
[tab
][tmp
& 0x3f];
661 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
662 for (i
= 0; i
< 64; i
++) {
663 tmp
= q
->tone_level_idx
[ch
][sb
][i
] = q
->tone_level_idx_base
[ch
][sb
][i
/ 8];
664 if ((tmp
< 0) || (!q
->superblocktype_2_3
&& !tmp
))
665 q
->tone_level
[ch
][sb
][i
] = 0;
667 q
->tone_level
[ch
][sb
][i
] = fft_tone_level_table
[tab
][tmp
& 0x3f];
676 * Related to synthesis filter
677 * Called by process_subpacket_11
678 * c is built with data from subpacket 11
679 * Most of this function is used only if superblock_type_2_3 == 0,
680 * never seen it in samples.
682 * @param tone_level_idx
683 * @param tone_level_idx_temp
684 * @param coding_method q->coding_method[0][0][0]
685 * @param nb_channels number of channels
686 * @param c coming from subpacket 11, passed as 8*c
687 * @param superblocktype_2_3 flag based on superblock packet type
688 * @param cm_table_select q->cm_table_select
690 static void fill_coding_method_array(sb_int8_array tone_level_idx
,
691 sb_int8_array tone_level_idx_temp
,
692 sb_int8_array coding_method
,
694 int c
, int superblocktype_2_3
,
698 int tmp
, acc
, esp_40
, comp
;
699 int add1
, add2
, add3
, add4
;
702 if (!superblocktype_2_3
) {
703 /* This case is untested, no samples available */
704 avpriv_request_sample(NULL
, "!superblocktype_2_3");
706 for (ch
= 0; ch
< nb_channels
; ch
++)
707 for (sb
= 0; sb
< 30; sb
++) {
708 for (j
= 1; j
< 63; j
++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
709 add1
= tone_level_idx
[ch
][sb
][j
] - 10;
712 add2
= add3
= add4
= 0;
714 add2
= tone_level_idx
[ch
][sb
- 2][j
] + tone_level_idx_offset_table
[sb
][0] - 6;
719 add3
= tone_level_idx
[ch
][sb
- 1][j
] + tone_level_idx_offset_table
[sb
][1] - 6;
724 add4
= tone_level_idx
[ch
][sb
+ 1][j
] + tone_level_idx_offset_table
[sb
][3] - 6;
728 tmp
= tone_level_idx
[ch
][sb
][j
+ 1] * 2 - add4
- add3
- add2
- add1
;
731 tone_level_idx_temp
[ch
][sb
][j
+ 1] = tmp
& 0xff;
733 tone_level_idx_temp
[ch
][sb
][0] = tone_level_idx_temp
[ch
][sb
][1];
736 for (ch
= 0; ch
< nb_channels
; ch
++)
737 for (sb
= 0; sb
< 30; sb
++)
738 for (j
= 0; j
< 64; j
++)
739 acc
+= tone_level_idx_temp
[ch
][sb
][j
];
741 multres
= 0x66666667LL
* (acc
* 10);
742 esp_40
= (multres
>> 32) / 8 + ((multres
& 0xffffffff) >> 31);
743 for (ch
= 0; ch
< nb_channels
; ch
++)
744 for (sb
= 0; sb
< 30; sb
++)
745 for (j
= 0; j
< 64; j
++) {
746 comp
= tone_level_idx_temp
[ch
][sb
][j
]* esp_40
* 10;
749 comp
/= 256; // signed shift
777 coding_method
[ch
][sb
][j
] = ((tmp
& 0xfffa) + 30 )& 0xff;
779 for (sb
= 0; sb
< 30; sb
++)
780 fix_coding_method_array(sb
, nb_channels
, coding_method
);
781 for (ch
= 0; ch
< nb_channels
; ch
++)
782 for (sb
= 0; sb
< 30; sb
++)
783 for (j
= 0; j
< 64; j
++)
785 if (coding_method
[ch
][sb
][j
] < 10)
786 coding_method
[ch
][sb
][j
] = 10;
789 if (coding_method
[ch
][sb
][j
] < 16)
790 coding_method
[ch
][sb
][j
] = 16;
792 if (coding_method
[ch
][sb
][j
] < 30)
793 coding_method
[ch
][sb
][j
] = 30;
796 } else { // superblocktype_2_3 != 0
797 for (ch
= 0; ch
< nb_channels
; ch
++)
798 for (sb
= 0; sb
< 30; sb
++)
799 for (j
= 0; j
< 64; j
++)
800 coding_method
[ch
][sb
][j
] = coding_method_table
[cm_table_select
][sb
];
806 * Called by process_subpacket_11 to process more data from subpacket 11
808 * Called by process_subpacket_12 to process data from subpacket 12 with
812 * @param gb bitreader context
813 * @param length packet length in bits
814 * @param sb_min lower subband processed (sb_min included)
815 * @param sb_max higher subband processed (sb_max excluded)
817 static int synthfilt_build_sb_samples(QDM2Context
*q
, GetBitContext
*gb
,
818 int length
, int sb_min
, int sb_max
)
820 int sb
, j
, k
, n
, ch
, run
, channels
;
821 int joined_stereo
, zero_encoding
;
823 float type34_div
= 0;
824 float type34_predictor
;
826 int sign_bits
[16] = {0};
829 // If no data use noise
830 for (sb
=sb_min
; sb
< sb_max
; sb
++)
831 build_sb_samples_from_noise(q
, sb
);
836 for (sb
= sb_min
; sb
< sb_max
; sb
++) {
837 channels
= q
->nb_channels
;
839 if (q
->nb_channels
<= 1 || sb
< 12)
844 joined_stereo
= (get_bits_left(gb
) >= 1) ? get_bits1(gb
) : 0;
847 if (get_bits_left(gb
) >= 16)
848 for (j
= 0; j
< 16; j
++)
849 sign_bits
[j
] = get_bits1(gb
);
851 for (j
= 0; j
< 64; j
++)
852 if (q
->coding_method
[1][sb
][j
] > q
->coding_method
[0][sb
][j
])
853 q
->coding_method
[0][sb
][j
] = q
->coding_method
[1][sb
][j
];
855 if (fix_coding_method_array(sb
, q
->nb_channels
,
857 av_log(NULL
, AV_LOG_ERROR
, "coding method invalid\n");
858 build_sb_samples_from_noise(q
, sb
);
864 for (ch
= 0; ch
< channels
; ch
++) {
865 FIX_NOISE_IDX(q
->noise_idx
);
866 zero_encoding
= (get_bits_left(gb
) >= 1) ? get_bits1(gb
) : 0;
867 type34_predictor
= 0.0;
870 for (j
= 0; j
< 128; ) {
871 switch (q
->coding_method
[ch
][sb
][j
/ 2]) {
873 if (get_bits_left(gb
) >= 10) {
875 for (k
= 0; k
< 5; k
++) {
876 if ((j
+ 2 * k
) >= 128)
878 samples
[2 * k
] = get_bits1(gb
) ? dequant_1bit
[joined_stereo
][2 * get_bits1(gb
)] : 0;
883 av_log(NULL
, AV_LOG_ERROR
, "Invalid 8bit codeword\n");
884 return AVERROR_INVALIDDATA
;
887 for (k
= 0; k
< 5; k
++)
888 samples
[2 * k
] = dequant_1bit
[joined_stereo
][random_dequant_index
[n
][k
]];
890 for (k
= 0; k
< 5; k
++)
891 samples
[2 * k
+ 1] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
893 for (k
= 0; k
< 10; k
++)
894 samples
[k
] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
900 if (get_bits_left(gb
) >= 1) {
905 f
-= noise_samples
[((sb
+ 1) * (j
+5 * ch
+ 1)) & 127] * 9.0 / 40.0;
908 samples
[0] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
914 if (get_bits_left(gb
) >= 10) {
916 for (k
= 0; k
< 5; k
++) {
919 samples
[k
] = (get_bits1(gb
) == 0) ? 0 : dequant_1bit
[joined_stereo
][2 * get_bits1(gb
)];
922 n
= get_bits (gb
, 8);
924 av_log(NULL
, AV_LOG_ERROR
, "Invalid 8bit codeword\n");
925 return AVERROR_INVALIDDATA
;
928 for (k
= 0; k
< 5; k
++)
929 samples
[k
] = dequant_1bit
[joined_stereo
][random_dequant_index
[n
][k
]];
932 for (k
= 0; k
< 5; k
++)
933 samples
[k
] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
939 if (get_bits_left(gb
) >= 7) {
942 av_log(NULL
, AV_LOG_ERROR
, "Invalid 7bit codeword\n");
943 return AVERROR_INVALIDDATA
;
946 for (k
= 0; k
< 3; k
++)
947 samples
[k
] = (random_dequant_type24
[n
][k
] - 2.0) * 0.5;
949 for (k
= 0; k
< 3; k
++)
950 samples
[k
] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
956 if (get_bits_left(gb
) >= 4) {
957 unsigned index
= qdm2_get_vlc(gb
, &vlc_tab_type30
, 0, 1);
958 if (index
>= FF_ARRAY_ELEMS(type30_dequant
)) {
959 av_log(NULL
, AV_LOG_ERROR
, "index %d out of type30_dequant array\n", index
);
960 return AVERROR_INVALIDDATA
;
962 samples
[0] = type30_dequant
[index
];
964 samples
[0] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
970 if (get_bits_left(gb
) >= 7) {
972 type34_div
= (float)(1 << get_bits(gb
, 2));
973 samples
[0] = ((float)get_bits(gb
, 5) - 16.0) / 15.0;
974 type34_predictor
= samples
[0];
977 unsigned index
= qdm2_get_vlc(gb
, &vlc_tab_type34
, 0, 1);
978 if (index
>= FF_ARRAY_ELEMS(type34_delta
)) {
979 av_log(NULL
, AV_LOG_ERROR
, "index %d out of type34_delta array\n", index
);
980 return AVERROR_INVALIDDATA
;
982 samples
[0] = type34_delta
[index
] / type34_div
+ type34_predictor
;
983 type34_predictor
= samples
[0];
986 samples
[0] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
992 samples
[0] = SB_DITHERING_NOISE(sb
,q
->noise_idx
);
998 for (k
= 0; k
< run
&& j
+ k
< 128; k
++) {
999 q
->sb_samples
[0][j
+ k
][sb
] =
1000 q
->tone_level
[0][sb
][(j
+ k
) / 2] * samples
[k
];
1001 if (q
->nb_channels
== 2) {
1002 if (sign_bits
[(j
+ k
) / 8])
1003 q
->sb_samples
[1][j
+ k
][sb
] =
1004 q
->tone_level
[1][sb
][(j
+ k
) / 2] * -samples
[k
];
1006 q
->sb_samples
[1][j
+ k
][sb
] =
1007 q
->tone_level
[1][sb
][(j
+ k
) / 2] * samples
[k
];
1011 for (k
= 0; k
< run
; k
++)
1013 q
->sb_samples
[ch
][j
+ k
][sb
] = q
->tone_level
[ch
][sb
][(j
+ k
)/2] * samples
[k
];
1024 * Init the first element of a channel in quantized_coeffs with data
1025 * from packet 10 (quantized_coeffs[ch][0]).
1026 * This is similar to process_subpacket_9, but for a single channel
1027 * and for element [0]
1028 * same VLC tables as process_subpacket_9 are used.
1030 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
1031 * @param gb bitreader context
1033 static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs
,
1036 int i
, k
, run
, level
, diff
;
1038 if (get_bits_left(gb
) < 16)
1040 level
= qdm2_get_vlc(gb
, &vlc_tab_level
, 0, 2);
1042 quantized_coeffs
[0] = level
;
1044 for (i
= 0; i
< 7; ) {
1045 if (get_bits_left(gb
) < 16)
1047 run
= qdm2_get_vlc(gb
, &vlc_tab_run
, 0, 1) + 1;
1052 if (get_bits_left(gb
) < 16)
1054 diff
= qdm2_get_se_vlc(&vlc_tab_diff
, gb
, 2);
1056 for (k
= 1; k
<= run
; k
++)
1057 quantized_coeffs
[i
+ k
] = (level
+ ((k
* diff
) / run
));
1066 * Related to synthesis filter, process data from packet 10
1067 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1068 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
1069 * data from packet 10
1072 * @param gb bitreader context
1074 static void init_tone_level_dequantization(QDM2Context
*q
, GetBitContext
*gb
)
1076 int sb
, j
, k
, n
, ch
;
1078 for (ch
= 0; ch
< q
->nb_channels
; ch
++) {
1079 init_quantized_coeffs_elem0(q
->quantized_coeffs
[ch
][0], gb
);
1081 if (get_bits_left(gb
) < 16) {
1082 memset(q
->quantized_coeffs
[ch
][0], 0, 8);
1087 n
= q
->sub_sampling
+ 1;
1089 for (sb
= 0; sb
< n
; sb
++)
1090 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
1091 for (j
= 0; j
< 8; j
++) {
1092 if (get_bits_left(gb
) < 1)
1094 if (get_bits1(gb
)) {
1095 for (k
=0; k
< 8; k
++) {
1096 if (get_bits_left(gb
) < 16)
1098 q
->tone_level_idx_hi1
[ch
][sb
][j
][k
] = qdm2_get_vlc(gb
, &vlc_tab_tone_level_idx_hi1
, 0, 2);
1101 for (k
=0; k
< 8; k
++)
1102 q
->tone_level_idx_hi1
[ch
][sb
][j
][k
] = 0;
1106 n
= QDM2_SB_USED(q
->sub_sampling
) - 4;
1108 for (sb
= 0; sb
< n
; sb
++)
1109 for (ch
= 0; ch
< q
->nb_channels
; ch
++) {
1110 if (get_bits_left(gb
) < 16)
1112 q
->tone_level_idx_hi2
[ch
][sb
] = qdm2_get_vlc(gb
, &vlc_tab_tone_level_idx_hi2
, 0, 2);
1114 q
->tone_level_idx_hi2
[ch
][sb
] -= 16;
1116 for (j
= 0; j
< 8; j
++)
1117 q
->tone_level_idx_mid
[ch
][sb
][j
] = -16;
1120 n
= QDM2_SB_USED(q
->sub_sampling
) - 5;
1122 for (sb
= 0; sb
< n
; sb
++)
1123 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
1124 for (j
= 0; j
< 8; j
++) {
1125 if (get_bits_left(gb
) < 16)
1127 q
->tone_level_idx_mid
[ch
][sb
][j
] = qdm2_get_vlc(gb
, &vlc_tab_tone_level_idx_mid
, 0, 2) - 32;
1132 * Process subpacket 9, init quantized_coeffs with data from it
1135 * @param node pointer to node with packet
1137 static int process_subpacket_9(QDM2Context
*q
, QDM2SubPNode
*node
)
1140 int i
, j
, k
, n
, ch
, run
, level
, diff
;
1142 init_get_bits(&gb
, node
->packet
->data
, node
->packet
->size
* 8);
1144 n
= coeff_per_sb_for_avg
[q
->coeff_per_sb_select
][QDM2_SB_USED(q
->sub_sampling
) - 1] + 1;
1146 for (i
= 1; i
< n
; i
++)
1147 for (ch
= 0; ch
< q
->nb_channels
; ch
++) {
1148 level
= qdm2_get_vlc(&gb
, &vlc_tab_level
, 0, 2);
1149 q
->quantized_coeffs
[ch
][i
][0] = level
;
1151 for (j
= 0; j
< (8 - 1); ) {
1152 run
= qdm2_get_vlc(&gb
, &vlc_tab_run
, 0, 1) + 1;
1153 diff
= qdm2_get_se_vlc(&vlc_tab_diff
, &gb
, 2);
1158 for (k
= 1; k
<= run
; k
++)
1159 q
->quantized_coeffs
[ch
][i
][j
+ k
] = (level
+ ((k
* diff
) / run
));
1166 for (ch
= 0; ch
< q
->nb_channels
; ch
++)
1167 for (i
= 0; i
< 8; i
++)
1168 q
->quantized_coeffs
[ch
][0][i
] = 0;
1174 * Process subpacket 10 if not null, else
1177 * @param node pointer to node with packet
1179 static void process_subpacket_10(QDM2Context
*q
, QDM2SubPNode
*node
)
1184 init_get_bits(&gb
, node
->packet
->data
, node
->packet
->size
* 8);
1185 init_tone_level_dequantization(q
, &gb
);
1186 fill_tone_level_array(q
, 1);
1188 fill_tone_level_array(q
, 0);
1193 * Process subpacket 11
1196 * @param node pointer to node with packet
1198 static void process_subpacket_11(QDM2Context
*q
, QDM2SubPNode
*node
)
1204 length
= node
->packet
->size
* 8;
1205 init_get_bits(&gb
, node
->packet
->data
, length
);
1209 int c
= get_bits(&gb
, 13);
1212 fill_coding_method_array(q
->tone_level_idx
,
1213 q
->tone_level_idx_temp
, q
->coding_method
,
1214 q
->nb_channels
, 8 * c
,
1215 q
->superblocktype_2_3
, q
->cm_table_select
);
1218 synthfilt_build_sb_samples(q
, &gb
, length
, 0, 8);
1222 * Process subpacket 12
1225 * @param node pointer to node with packet
1227 static void process_subpacket_12(QDM2Context
*q
, QDM2SubPNode
*node
)
1233 length
= node
->packet
->size
* 8;
1234 init_get_bits(&gb
, node
->packet
->data
, length
);
1237 synthfilt_build_sb_samples(q
, &gb
, length
, 8, QDM2_SB_USED(q
->sub_sampling
));
1241 * Process new subpackets for synthesis filter
1244 * @param list list with synthesis filter packets (list D)
1246 static void process_synthesis_subpackets(QDM2Context
*q
, QDM2SubPNode
*list
)
1248 QDM2SubPNode
*nodes
[4];
1250 nodes
[0] = qdm2_search_subpacket_type_in_list(list
, 9);
1252 process_subpacket_9(q
, nodes
[0]);
1254 nodes
[1] = qdm2_search_subpacket_type_in_list(list
, 10);
1256 process_subpacket_10(q
, nodes
[1]);
1258 process_subpacket_10(q
, NULL
);
1260 nodes
[2] = qdm2_search_subpacket_type_in_list(list
, 11);
1261 if (nodes
[0] && nodes
[1] && nodes
[2])
1262 process_subpacket_11(q
, nodes
[2]);
1264 process_subpacket_11(q
, NULL
);
1266 nodes
[3] = qdm2_search_subpacket_type_in_list(list
, 12);
1267 if (nodes
[0] && nodes
[1] && nodes
[3])
1268 process_subpacket_12(q
, nodes
[3]);
1270 process_subpacket_12(q
, NULL
);
1274 * Decode superblock, fill packet lists.
1278 static void qdm2_decode_super_block(QDM2Context
*q
)
1281 QDM2SubPacket header
, *packet
;
1282 int i
, packet_bytes
, sub_packet_size
, sub_packets_D
;
1283 unsigned int next_index
= 0;
1285 memset(q
->tone_level_idx_hi1
, 0, sizeof(q
->tone_level_idx_hi1
));
1286 memset(q
->tone_level_idx_mid
, 0, sizeof(q
->tone_level_idx_mid
));
1287 memset(q
->tone_level_idx_hi2
, 0, sizeof(q
->tone_level_idx_hi2
));
1289 q
->sub_packets_B
= 0;
1292 average_quantized_coeffs(q
); // average elements in quantized_coeffs[max_ch][10][8]
1294 init_get_bits(&gb
, q
->compressed_data
, q
->compressed_size
* 8);
1295 qdm2_decode_sub_packet_header(&gb
, &header
);
1297 if (header
.type
< 2 || header
.type
>= 8) {
1299 av_log(NULL
, AV_LOG_ERROR
, "bad superblock type\n");
1303 q
->superblocktype_2_3
= (header
.type
== 2 || header
.type
== 3);
1304 packet_bytes
= (q
->compressed_size
- get_bits_count(&gb
) / 8);
1306 init_get_bits(&gb
, header
.data
, header
.size
* 8);
1308 if (header
.type
== 2 || header
.type
== 4 || header
.type
== 5) {
1309 int csum
= 257 * get_bits(&gb
, 8);
1310 csum
+= 2 * get_bits(&gb
, 8);
1312 csum
= qdm2_packet_checksum(q
->compressed_data
, q
->checksum_size
, csum
);
1316 av_log(NULL
, AV_LOG_ERROR
, "bad packet checksum\n");
1321 q
->sub_packet_list_B
[0].packet
= NULL
;
1322 q
->sub_packet_list_D
[0].packet
= NULL
;
1324 for (i
= 0; i
< 6; i
++)
1325 if (--q
->fft_level_exp
[i
] < 0)
1326 q
->fft_level_exp
[i
] = 0;
1328 for (i
= 0; packet_bytes
> 0; i
++) {
1331 if (i
>= FF_ARRAY_ELEMS(q
->sub_packet_list_A
)) {
1332 SAMPLES_NEEDED_2("too many packet bytes");
1336 q
->sub_packet_list_A
[i
].next
= NULL
;
1339 q
->sub_packet_list_A
[i
- 1].next
= &q
->sub_packet_list_A
[i
];
1341 /* seek to next block */
1342 init_get_bits(&gb
, header
.data
, header
.size
* 8);
1343 skip_bits(&gb
, next_index
* 8);
1345 if (next_index
>= header
.size
)
1349 /* decode subpacket */
1350 packet
= &q
->sub_packets
[i
];
1351 qdm2_decode_sub_packet_header(&gb
, packet
);
1352 next_index
= packet
->size
+ get_bits_count(&gb
) / 8;
1353 sub_packet_size
= ((packet
->size
> 0xff) ? 1 : 0) + packet
->size
+ 2;
1355 if (packet
->type
== 0)
1358 if (sub_packet_size
> packet_bytes
) {
1359 if (packet
->type
!= 10 && packet
->type
!= 11 && packet
->type
!= 12)
1361 packet
->size
+= packet_bytes
- sub_packet_size
;
1364 packet_bytes
-= sub_packet_size
;
1366 /* add subpacket to 'all subpackets' list */
1367 q
->sub_packet_list_A
[i
].packet
= packet
;
1369 /* add subpacket to related list */
1370 if (packet
->type
== 8) {
1371 SAMPLES_NEEDED_2("packet type 8");
1373 } else if (packet
->type
>= 9 && packet
->type
<= 12) {
1374 /* packets for MPEG Audio like Synthesis Filter */
1375 QDM2_LIST_ADD(q
->sub_packet_list_D
, sub_packets_D
, packet
);
1376 } else if (packet
->type
== 13) {
1377 for (j
= 0; j
< 6; j
++)
1378 q
->fft_level_exp
[j
] = get_bits(&gb
, 6);
1379 } else if (packet
->type
== 14) {
1380 for (j
= 0; j
< 6; j
++)
1381 q
->fft_level_exp
[j
] = qdm2_get_vlc(&gb
, &fft_level_exp_vlc
, 0, 2);
1382 } else if (packet
->type
== 15) {
1383 SAMPLES_NEEDED_2("packet type 15")
1385 } else if (packet
->type
>= 16 && packet
->type
< 48 &&
1386 !fft_subpackets
[packet
->type
- 16]) {
1387 /* packets for FFT */
1388 QDM2_LIST_ADD(q
->sub_packet_list_B
, q
->sub_packets_B
, packet
);
1390 } // Packet bytes loop
1392 if (q
->sub_packet_list_D
[0].packet
) {
1393 process_synthesis_subpackets(q
, q
->sub_packet_list_D
);
1394 q
->do_synth_filter
= 1;
1395 } else if (q
->do_synth_filter
) {
1396 process_subpacket_10(q
, NULL
);
1397 process_subpacket_11(q
, NULL
);
1398 process_subpacket_12(q
, NULL
);
1402 static void qdm2_fft_init_coefficient(QDM2Context
*q
, int sub_packet
,
1403 int offset
, int duration
, int channel
,
1406 if (q
->fft_coefs_min_index
[duration
] < 0)
1407 q
->fft_coefs_min_index
[duration
] = q
->fft_coefs_index
;
1409 q
->fft_coefs
[q
->fft_coefs_index
].sub_packet
=
1410 ((sub_packet
>= 16) ? (sub_packet
- 16) : sub_packet
);
1411 q
->fft_coefs
[q
->fft_coefs_index
].channel
= channel
;
1412 q
->fft_coefs
[q
->fft_coefs_index
].offset
= offset
;
1413 q
->fft_coefs
[q
->fft_coefs_index
].exp
= exp
;
1414 q
->fft_coefs
[q
->fft_coefs_index
].phase
= phase
;
1415 q
->fft_coefs_index
++;
1418 static void qdm2_fft_decode_tones(QDM2Context
*q
, int duration
,
1419 GetBitContext
*gb
, int b
)
1421 int channel
, stereo
, phase
, exp
;
1422 int local_int_4
, local_int_8
, stereo_phase
, local_int_10
;
1423 int local_int_14
, stereo_exp
, local_int_20
, local_int_28
;
1429 local_int_8
= (4 - duration
);
1430 local_int_10
= 1 << (q
->group_order
- duration
- 1);
1433 while (get_bits_left(gb
)>0) {
1434 if (q
->superblocktype_2_3
) {
1435 while ((n
= qdm2_get_vlc(gb
, &vlc_tab_fft_tone_offset
[local_int_8
], 1, 2)) < 2) {
1436 if (get_bits_left(gb
)<0) {
1437 if(local_int_4
< q
->group_size
)
1438 av_log(NULL
, AV_LOG_ERROR
, "overread in qdm2_fft_decode_tones()\n");
1443 local_int_4
+= local_int_10
;
1444 local_int_28
+= (1 << local_int_8
);
1446 local_int_4
+= 8 * local_int_10
;
1447 local_int_28
+= (8 << local_int_8
);
1452 offset
+= qdm2_get_vlc(gb
, &vlc_tab_fft_tone_offset
[local_int_8
], 1, 2);
1453 while (offset
>= (local_int_10
- 1)) {
1454 offset
+= (1 - (local_int_10
- 1));
1455 local_int_4
+= local_int_10
;
1456 local_int_28
+= (1 << local_int_8
);
1460 if (local_int_4
>= q
->group_size
)
1463 local_int_14
= (offset
>> local_int_8
);
1464 if (local_int_14
>= FF_ARRAY_ELEMS(fft_level_index_table
))
1467 if (q
->nb_channels
> 1) {
1468 channel
= get_bits1(gb
);
1469 stereo
= get_bits1(gb
);
1475 exp
= qdm2_get_vlc(gb
, (b
? &fft_level_exp_vlc
: &fft_level_exp_alt_vlc
), 0, 2);
1476 exp
+= q
->fft_level_exp
[fft_level_index_table
[local_int_14
]];
1477 exp
= (exp
< 0) ? 0 : exp
;
1479 phase
= get_bits(gb
, 3);
1484 stereo_exp
= (exp
- qdm2_get_vlc(gb
, &fft_stereo_exp_vlc
, 0, 1));
1485 stereo_phase
= (phase
- qdm2_get_vlc(gb
, &fft_stereo_phase_vlc
, 0, 1));
1486 if (stereo_phase
< 0)
1490 if (q
->frequency_range
> (local_int_14
+ 1)) {
1491 int sub_packet
= (local_int_20
+ local_int_28
);
1493 qdm2_fft_init_coefficient(q
, sub_packet
, offset
, duration
,
1494 channel
, exp
, phase
);
1496 qdm2_fft_init_coefficient(q
, sub_packet
, offset
, duration
,
1498 stereo_exp
, stereo_phase
);
1504 static void qdm2_decode_fft_packets(QDM2Context
*q
)
1506 int i
, j
, min
, max
, value
, type
, unknown_flag
;
1509 if (!q
->sub_packet_list_B
[0].packet
)
1512 /* reset minimum indexes for FFT coefficients */
1513 q
->fft_coefs_index
= 0;
1514 for (i
= 0; i
< 5; i
++)
1515 q
->fft_coefs_min_index
[i
] = -1;
1517 /* process subpackets ordered by type, largest type first */
1518 for (i
= 0, max
= 256; i
< q
->sub_packets_B
; i
++) {
1519 QDM2SubPacket
*packet
= NULL
;
1521 /* find subpacket with largest type less than max */
1522 for (j
= 0, min
= 0; j
< q
->sub_packets_B
; j
++) {
1523 value
= q
->sub_packet_list_B
[j
].packet
->type
;
1524 if (value
> min
&& value
< max
) {
1526 packet
= q
->sub_packet_list_B
[j
].packet
;
1532 /* check for errors (?) */
1537 (packet
->type
< 16 || packet
->type
>= 48 ||
1538 fft_subpackets
[packet
->type
- 16]))
1541 /* decode FFT tones */
1542 init_get_bits(&gb
, packet
->data
, packet
->size
* 8);
1544 if (packet
->type
>= 32 && packet
->type
< 48 && !fft_subpackets
[packet
->type
- 16])
1549 type
= packet
->type
;
1551 if ((type
>= 17 && type
< 24) || (type
>= 33 && type
< 40)) {
1552 int duration
= q
->sub_sampling
+ 5 - (type
& 15);
1554 if (duration
>= 0 && duration
< 4)
1555 qdm2_fft_decode_tones(q
, duration
, &gb
, unknown_flag
);
1556 } else if (type
== 31) {
1557 for (j
= 0; j
< 4; j
++)
1558 qdm2_fft_decode_tones(q
, j
, &gb
, unknown_flag
);
1559 } else if (type
== 46) {
1560 for (j
= 0; j
< 6; j
++)
1561 q
->fft_level_exp
[j
] = get_bits(&gb
, 6);
1562 for (j
= 0; j
< 4; j
++)
1563 qdm2_fft_decode_tones(q
, j
, &gb
, unknown_flag
);
1565 } // Loop on B packets
1567 /* calculate maximum indexes for FFT coefficients */
1568 for (i
= 0, j
= -1; i
< 5; i
++)
1569 if (q
->fft_coefs_min_index
[i
] >= 0) {
1571 q
->fft_coefs_max_index
[j
] = q
->fft_coefs_min_index
[i
];
1575 q
->fft_coefs_max_index
[j
] = q
->fft_coefs_index
;
1578 static void qdm2_fft_generate_tone(QDM2Context
*q
, FFTTone
*tone
)
1583 const double iscale
= 2.0 * M_PI
/ 512.0;
1585 tone
->phase
+= tone
->phase_shift
;
1587 /* calculate current level (maximum amplitude) of tone */
1588 level
= fft_tone_envelope_table
[tone
->duration
][tone
->time_index
] * tone
->level
;
1589 c
.im
= level
* sin(tone
->phase
* iscale
);
1590 c
.re
= level
* cos(tone
->phase
* iscale
);
1592 /* generate FFT coefficients for tone */
1593 if (tone
->duration
>= 3 || tone
->cutoff
>= 3) {
1594 tone
->complex[0].im
+= c
.im
;
1595 tone
->complex[0].re
+= c
.re
;
1596 tone
->complex[1].im
-= c
.im
;
1597 tone
->complex[1].re
-= c
.re
;
1599 f
[1] = -tone
->table
[4];
1600 f
[0] = tone
->table
[3] - tone
->table
[0];
1601 f
[2] = 1.0 - tone
->table
[2] - tone
->table
[3];
1602 f
[3] = tone
->table
[1] + tone
->table
[4] - 1.0;
1603 f
[4] = tone
->table
[0] - tone
->table
[1];
1604 f
[5] = tone
->table
[2];
1605 for (i
= 0; i
< 2; i
++) {
1606 tone
->complex[fft_cutoff_index_table
[tone
->cutoff
][i
]].re
+=
1608 tone
->complex[fft_cutoff_index_table
[tone
->cutoff
][i
]].im
+=
1609 c
.im
* ((tone
->cutoff
<= i
) ? -f
[i
] : f
[i
]);
1611 for (i
= 0; i
< 4; i
++) {
1612 tone
->complex[i
].re
+= c
.re
* f
[i
+ 2];
1613 tone
->complex[i
].im
+= c
.im
* f
[i
+ 2];
1617 /* copy the tone if it has not yet died out */
1618 if (++tone
->time_index
< ((1 << (5 - tone
->duration
)) - 1)) {
1619 memcpy(&q
->fft_tones
[q
->fft_tone_end
], tone
, sizeof(FFTTone
));
1620 q
->fft_tone_end
= (q
->fft_tone_end
+ 1) % 1000;
1624 static void qdm2_fft_tone_synthesizer(QDM2Context
*q
, int sub_packet
)
1627 const double iscale
= 0.25 * M_PI
;
1629 for (ch
= 0; ch
< q
->channels
; ch
++) {
1630 memset(q
->fft
.complex[ch
], 0, q
->fft_size
* sizeof(QDM2Complex
));
1634 /* apply FFT tones with duration 4 (1 FFT period) */
1635 if (q
->fft_coefs_min_index
[4] >= 0)
1636 for (i
= q
->fft_coefs_min_index
[4]; i
< q
->fft_coefs_max_index
[4]; i
++) {
1640 if (q
->fft_coefs
[i
].sub_packet
!= sub_packet
)
1643 ch
= (q
->channels
== 1) ? 0 : q
->fft_coefs
[i
].channel
;
1644 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];
1646 c
.re
= level
* cos(q
->fft_coefs
[i
].phase
* iscale
);
1647 c
.im
= level
* sin(q
->fft_coefs
[i
].phase
* iscale
);
1648 q
->fft
.complex[ch
][q
->fft_coefs
[i
].offset
+ 0].re
+= c
.re
;
1649 q
->fft
.complex[ch
][q
->fft_coefs
[i
].offset
+ 0].im
+= c
.im
;
1650 q
->fft
.complex[ch
][q
->fft_coefs
[i
].offset
+ 1].re
-= c
.re
;
1651 q
->fft
.complex[ch
][q
->fft_coefs
[i
].offset
+ 1].im
-= c
.im
;
1654 /* generate existing FFT tones */
1655 for (i
= q
->fft_tone_end
; i
!= q
->fft_tone_start
; ) {
1656 qdm2_fft_generate_tone(q
, &q
->fft_tones
[q
->fft_tone_start
]);
1657 q
->fft_tone_start
= (q
->fft_tone_start
+ 1) % 1000;
1660 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1661 for (i
= 0; i
< 4; i
++)
1662 if (q
->fft_coefs_min_index
[i
] >= 0) {
1663 for (j
= q
->fft_coefs_min_index
[i
]; j
< q
->fft_coefs_max_index
[i
]; j
++) {
1667 if (q
->fft_coefs
[j
].sub_packet
!= sub_packet
)
1671 offset
= q
->fft_coefs
[j
].offset
>> four_i
;
1672 ch
= (q
->channels
== 1) ? 0 : q
->fft_coefs
[j
].channel
;
1674 if (offset
< q
->frequency_range
) {
1676 tone
.cutoff
= offset
;
1678 tone
.cutoff
= (offset
>= 60) ? 3 : 2;
1680 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];
1681 tone
.complex = &q
->fft
.complex[ch
][offset
];
1682 tone
.table
= fft_tone_sample_table
[i
][q
->fft_coefs
[j
].offset
- (offset
<< four_i
)];
1683 tone
.phase
= 64 * q
->fft_coefs
[j
].phase
- (offset
<< 8) - 128;
1684 tone
.phase_shift
= (2 * q
->fft_coefs
[j
].offset
+ 1) << (7 - four_i
);
1686 tone
.time_index
= 0;
1688 qdm2_fft_generate_tone(q
, &tone
);
1691 q
->fft_coefs_min_index
[i
] = j
;
1695 static void qdm2_calculate_fft(QDM2Context
*q
, int channel
, int sub_packet
)
1697 const float gain
= (q
->channels
== 1 && q
->nb_channels
== 2) ? 0.5f
: 1.0f
;
1698 float *out
= q
->output_buffer
+ channel
;
1700 q
->fft
.complex[channel
][0].re
*= 2.0f
;
1701 q
->fft
.complex[channel
][0].im
= 0.0f
;
1702 q
->rdft_ctx
.rdft_calc(&q
->rdft_ctx
, (FFTSample
*)q
->fft
.complex[channel
]);
1703 /* add samples to output buffer */
1704 for (i
= 0; i
< FFALIGN(q
->fft_size
, 8); i
++) {
1705 out
[0] += q
->fft
.complex[channel
][i
].re
* gain
;
1706 out
[q
->channels
] += q
->fft
.complex[channel
][i
].im
* gain
;
1707 out
+= 2 * q
->channels
;
1713 * @param index subpacket number
1715 static void qdm2_synthesis_filter(QDM2Context
*q
, int index
)
1717 int i
, k
, ch
, sb_used
, sub_sampling
, dither_state
= 0;
1719 /* copy sb_samples */
1720 sb_used
= QDM2_SB_USED(q
->sub_sampling
);
1722 for (ch
= 0; ch
< q
->channels
; ch
++)
1723 for (i
= 0; i
< 8; i
++)
1724 for (k
= sb_used
; k
< SBLIMIT
; k
++)
1725 q
->sb_samples
[ch
][(8 * index
) + i
][k
] = 0;
1727 for (ch
= 0; ch
< q
->nb_channels
; ch
++) {
1728 float *samples_ptr
= q
->samples
+ ch
;
1730 for (i
= 0; i
< 8; i
++) {
1731 ff_mpa_synth_filter_float(&q
->mpadsp
,
1732 q
->synth_buf
[ch
], &(q
->synth_buf_offset
[ch
]),
1733 ff_mpa_synth_window_float
, &dither_state
,
1734 samples_ptr
, q
->nb_channels
,
1735 q
->sb_samples
[ch
][(8 * index
) + i
]);
1736 samples_ptr
+= 32 * q
->nb_channels
;
1740 /* add samples to output buffer */
1741 sub_sampling
= (4 >> q
->sub_sampling
);
1743 for (ch
= 0; ch
< q
->channels
; ch
++)
1744 for (i
= 0; i
< q
->frame_size
; i
++)
1745 q
->output_buffer
[q
->channels
* i
+ ch
] += (1 << 23) * q
->samples
[q
->nb_channels
* sub_sampling
* i
+ ch
];
1749 * Init static data (does not depend on specific file)
1753 static av_cold
void qdm2_init_static_data(void) {
1760 ff_mpa_synth_init_float(ff_mpa_synth_window_float
);
1761 softclip_table_init();
1763 init_noise_samples();
1769 * Init parameters from codec extradata
1771 static av_cold
int qdm2_decode_init(AVCodecContext
*avctx
)
1773 QDM2Context
*s
= avctx
->priv_data
;
1776 int tmp_val
, tmp
, size
;
1778 qdm2_init_static_data();
1780 /* extradata parsing
1789 32 size (including this field)
1791 32 type (=QDM2 or QDMC)
1793 32 size (including this field, in bytes)
1794 32 tag (=QDCA) // maybe mandatory parameters
1797 32 samplerate (=44100)
1799 32 block size (=4096)
1800 32 frame size (=256) (for one channel)
1801 32 packet size (=1300)
1803 32 size (including this field, in bytes)
1804 32 tag (=QDCP) // maybe some tuneable parameters
1814 if (!avctx
->extradata
|| (avctx
->extradata_size
< 48)) {
1815 av_log(avctx
, AV_LOG_ERROR
, "extradata missing or truncated\n");
1819 extradata
= avctx
->extradata
;
1820 extradata_size
= avctx
->extradata_size
;
1822 while (extradata_size
> 7) {
1823 if (!memcmp(extradata
, "frmaQDM", 7))
1829 if (extradata_size
< 12) {
1830 av_log(avctx
, AV_LOG_ERROR
, "not enough extradata (%i)\n",
1835 if (memcmp(extradata
, "frmaQDM", 7)) {
1836 av_log(avctx
, AV_LOG_ERROR
, "invalid headers, QDM? not found\n");
1840 if (extradata
[7] == 'C') {
1842 av_log(avctx
, AV_LOG_ERROR
, "stream is QDMC version 1, which is not supported\n");
1847 extradata_size
-= 8;
1849 size
= AV_RB32(extradata
);
1851 if(size
> extradata_size
){
1852 av_log(avctx
, AV_LOG_ERROR
, "extradata size too small, %i < %i\n",
1853 extradata_size
, size
);
1858 av_log(avctx
, AV_LOG_DEBUG
, "size: %d\n", size
);
1859 if (AV_RB32(extradata
) != MKBETAG('Q','D','C','A')) {
1860 av_log(avctx
, AV_LOG_ERROR
, "invalid extradata, expecting QDCA\n");
1866 avctx
->channels
= s
->nb_channels
= s
->channels
= AV_RB32(extradata
);
1868 if (s
->channels
<= 0 || s
->channels
> MPA_MAX_CHANNELS
) {
1869 av_log(avctx
, AV_LOG_ERROR
, "Invalid number of channels\n");
1870 return AVERROR_INVALIDDATA
;
1872 avctx
->channel_layout
= avctx
->channels
== 2 ? AV_CH_LAYOUT_STEREO
:
1875 avctx
->sample_rate
= AV_RB32(extradata
);
1878 avctx
->bit_rate
= AV_RB32(extradata
);
1881 s
->group_size
= AV_RB32(extradata
);
1884 s
->fft_size
= AV_RB32(extradata
);
1887 s
->checksum_size
= AV_RB32(extradata
);
1888 if (s
->checksum_size
>= 1U << 28) {
1889 av_log(avctx
, AV_LOG_ERROR
, "data block size too large (%u)\n", s
->checksum_size
);
1890 return AVERROR_INVALIDDATA
;
1893 s
->fft_order
= av_log2(s
->fft_size
) + 1;
1895 // something like max decodable tones
1896 s
->group_order
= av_log2(s
->group_size
) + 1;
1897 s
->frame_size
= s
->group_size
/ 16; // 16 iterations per super block
1899 if (s
->frame_size
> QDM2_MAX_FRAME_SIZE
)
1900 return AVERROR_INVALIDDATA
;
1902 s
->sub_sampling
= s
->fft_order
- 7;
1903 s
->frequency_range
= 255 / (1 << (2 - s
->sub_sampling
));
1905 switch ((s
->sub_sampling
* 2 + s
->channels
- 1)) {
1906 case 0: tmp
= 40; break;
1907 case 1: tmp
= 48; break;
1908 case 2: tmp
= 56; break;
1909 case 3: tmp
= 72; break;
1910 case 4: tmp
= 80; break;
1911 case 5: tmp
= 100;break;
1912 default: tmp
=s
->sub_sampling
; break;
1915 if ((tmp
* 1000) < avctx
->bit_rate
) tmp_val
= 1;
1916 if ((tmp
* 1440) < avctx
->bit_rate
) tmp_val
= 2;
1917 if ((tmp
* 1760) < avctx
->bit_rate
) tmp_val
= 3;
1918 if ((tmp
* 2240) < avctx
->bit_rate
) tmp_val
= 4;
1919 s
->cm_table_select
= tmp_val
;
1921 if (avctx
->bit_rate
<= 8000)
1922 s
->coeff_per_sb_select
= 0;
1923 else if (avctx
->bit_rate
< 16000)
1924 s
->coeff_per_sb_select
= 1;
1926 s
->coeff_per_sb_select
= 2;
1928 // Fail on unknown fft order
1929 if ((s
->fft_order
< 7) || (s
->fft_order
> 9)) {
1930 av_log(avctx
, AV_LOG_ERROR
, "Unknown FFT order (%d), contact the developers!\n", s
->fft_order
);
1933 if (s
->fft_size
!= (1 << (s
->fft_order
- 1))) {
1934 av_log(avctx
, AV_LOG_ERROR
, "FFT size %d not power of 2.\n", s
->fft_size
);
1935 return AVERROR_INVALIDDATA
;
1938 ff_rdft_init(&s
->rdft_ctx
, s
->fft_order
, IDFT_C2R
);
1939 ff_mpadsp_init(&s
->mpadsp
);
1941 avctx
->sample_fmt
= AV_SAMPLE_FMT_S16
;
1946 static av_cold
int qdm2_decode_close(AVCodecContext
*avctx
)
1948 QDM2Context
*s
= avctx
->priv_data
;
1950 ff_rdft_end(&s
->rdft_ctx
);
1955 static int qdm2_decode(QDM2Context
*q
, const uint8_t *in
, int16_t *out
)
1958 const int frame_size
= (q
->frame_size
* q
->channels
);
1960 if((unsigned)frame_size
> FF_ARRAY_ELEMS(q
->output_buffer
)/2)
1963 /* select input buffer */
1964 q
->compressed_data
= in
;
1965 q
->compressed_size
= q
->checksum_size
;
1967 /* copy old block, clear new block of output samples */
1968 memmove(q
->output_buffer
, &q
->output_buffer
[frame_size
], frame_size
* sizeof(float));
1969 memset(&q
->output_buffer
[frame_size
], 0, frame_size
* sizeof(float));
1971 /* decode block of QDM2 compressed data */
1972 if (q
->sub_packet
== 0) {
1973 q
->has_errors
= 0; // zero it for a new super block
1974 av_log(NULL
,AV_LOG_DEBUG
,"Superblock follows\n");
1975 qdm2_decode_super_block(q
);
1978 /* parse subpackets */
1979 if (!q
->has_errors
) {
1980 if (q
->sub_packet
== 2)
1981 qdm2_decode_fft_packets(q
);
1983 qdm2_fft_tone_synthesizer(q
, q
->sub_packet
);
1986 /* sound synthesis stage 1 (FFT) */
1987 for (ch
= 0; ch
< q
->channels
; ch
++) {
1988 qdm2_calculate_fft(q
, ch
, q
->sub_packet
);
1990 if (!q
->has_errors
&& q
->sub_packet_list_C
[0].packet
) {
1991 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1996 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1997 if (!q
->has_errors
&& q
->do_synth_filter
)
1998 qdm2_synthesis_filter(q
, q
->sub_packet
);
2000 q
->sub_packet
= (q
->sub_packet
+ 1) % 16;
2002 /* clip and convert output float[] to 16bit signed samples */
2003 for (i
= 0; i
< frame_size
; i
++) {
2004 int value
= (int)q
->output_buffer
[i
];
2006 if (value
> SOFTCLIP_THRESHOLD
)
2007 value
= (value
> HARDCLIP_THRESHOLD
) ? 32767 : softclip_table
[ value
- SOFTCLIP_THRESHOLD
];
2008 else if (value
< -SOFTCLIP_THRESHOLD
)
2009 value
= (value
< -HARDCLIP_THRESHOLD
) ? -32767 : -softclip_table
[-value
- SOFTCLIP_THRESHOLD
];
2017 static int qdm2_decode_frame(AVCodecContext
*avctx
, void *data
,
2018 int *got_frame_ptr
, AVPacket
*avpkt
)
2020 AVFrame
*frame
= data
;
2021 const uint8_t *buf
= avpkt
->data
;
2022 int buf_size
= avpkt
->size
;
2023 QDM2Context
*s
= avctx
->priv_data
;
2029 if(buf_size
< s
->checksum_size
)
2032 /* get output buffer */
2033 frame
->nb_samples
= 16 * s
->frame_size
;
2034 if ((ret
= ff_get_buffer(avctx
, frame
, 0)) < 0)
2036 out
= (int16_t *)frame
->data
[0];
2038 for (i
= 0; i
< 16; i
++) {
2039 if (qdm2_decode(s
, buf
, out
) < 0)
2041 out
+= s
->channels
* s
->frame_size
;
2046 return s
->checksum_size
;
2049 AVCodec ff_qdm2_decoder
= {
2051 .long_name
= NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
2052 .type
= AVMEDIA_TYPE_AUDIO
,
2053 .id
= AV_CODEC_ID_QDM2
,
2054 .priv_data_size
= sizeof(QDM2Context
),
2055 .init
= qdm2_decode_init
,
2056 .close
= qdm2_decode_close
,
2057 .decode
= qdm2_decode_frame
,
2058 .capabilities
= CODEC_CAP_DR1
,