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1 | /* |
2 | * QDM2 compatible decoder | |
3 | * Copyright (c) 2003 Ewald Snel | |
4 | * Copyright (c) 2005 Benjamin Larsson | |
5 | * Copyright (c) 2005 Alex Beregszaszi | |
6 | * Copyright (c) 2005 Roberto Togni | |
7 | * | |
8 | * This file is part of FFmpeg. | |
9 | * | |
10 | * FFmpeg is free software; you can redistribute it and/or | |
11 | * modify it under the terms of the GNU Lesser General Public | |
12 | * License as published by the Free Software Foundation; either | |
13 | * version 2.1 of the License, or (at your option) any later version. | |
14 | * | |
15 | * FFmpeg is distributed in the hope that it will be useful, | |
16 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
17 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
18 | * Lesser General Public License for more details. | |
19 | * | |
20 | * You should have received a copy of the GNU Lesser General Public | |
21 | * License along with FFmpeg; if not, write to the Free Software | |
22 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
23 | */ | |
24 | ||
25 | /** | |
26 | * @file | |
27 | * QDM2 decoder | |
28 | * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni | |
29 | * | |
30 | * The decoder is not perfect yet, there are still some distortions | |
31 | * especially on files encoded with 16 or 8 subbands. | |
32 | */ | |
33 | ||
34 | #include <math.h> | |
35 | #include <stddef.h> | |
36 | #include <stdio.h> | |
37 | ||
38 | #define BITSTREAM_READER_LE | |
39 | #include "libavutil/channel_layout.h" | |
40 | #include "avcodec.h" | |
41 | #include "get_bits.h" | |
42 | #include "internal.h" | |
43 | #include "rdft.h" | |
44 | #include "mpegaudiodsp.h" | |
45 | #include "mpegaudio.h" | |
46 | ||
47 | #include "qdm2data.h" | |
48 | #include "qdm2_tablegen.h" | |
49 | ||
50 | #undef NDEBUG | |
51 | #include <assert.h> | |
52 | ||
53 | ||
54 | #define QDM2_LIST_ADD(list, size, packet) \ | |
55 | do { \ | |
56 | if (size > 0) { \ | |
57 | list[size - 1].next = &list[size]; \ | |
58 | } \ | |
59 | list[size].packet = packet; \ | |
60 | list[size].next = NULL; \ | |
61 | size++; \ | |
62 | } while(0) | |
63 | ||
64 | // Result is 8, 16 or 30 | |
65 | #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling)) | |
66 | ||
67 | #define FIX_NOISE_IDX(noise_idx) \ | |
68 | if ((noise_idx) >= 3840) \ | |
69 | (noise_idx) -= 3840; \ | |
70 | ||
71 | #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)]) | |
72 | ||
73 | #define SAMPLES_NEEDED \ | |
74 | av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); | |
75 | ||
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); | |
78 | ||
79 | #define QDM2_MAX_FRAME_SIZE 512 | |
80 | ||
81 | typedef int8_t sb_int8_array[2][30][64]; | |
82 | ||
83 | /** | |
84 | * Subpacket | |
85 | */ | |
86 | typedef struct { | |
87 | int type; ///< subpacket type | |
88 | unsigned int size; ///< subpacket size | |
89 | const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) | |
90 | } QDM2SubPacket; | |
91 | ||
92 | /** | |
93 | * A node in the subpacket list | |
94 | */ | |
95 | typedef struct QDM2SubPNode { | |
96 | QDM2SubPacket *packet; ///< packet | |
97 | struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node | |
98 | } QDM2SubPNode; | |
99 | ||
100 | typedef struct { | |
101 | float re; | |
102 | float im; | |
103 | } QDM2Complex; | |
104 | ||
105 | typedef struct { | |
106 | float level; | |
107 | QDM2Complex *complex; | |
108 | const float *table; | |
109 | int phase; | |
110 | int phase_shift; | |
111 | int duration; | |
112 | short time_index; | |
113 | short cutoff; | |
114 | } FFTTone; | |
115 | ||
116 | typedef struct { | |
117 | int16_t sub_packet; | |
118 | uint8_t channel; | |
119 | int16_t offset; | |
120 | int16_t exp; | |
121 | uint8_t phase; | |
122 | } FFTCoefficient; | |
123 | ||
124 | typedef struct { | |
125 | DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; | |
126 | } QDM2FFT; | |
127 | ||
128 | /** | |
129 | * QDM2 decoder context | |
130 | */ | |
131 | typedef struct { | |
132 | /// Parameters from codec header, do not change during playback | |
133 | int nb_channels; ///< number of channels | |
134 | int channels; ///< number of channels | |
135 | int group_size; ///< size of frame group (16 frames per group) | |
136 | int fft_size; ///< size of FFT, in complex numbers | |
137 | int checksum_size; ///< size of data block, used also for checksum | |
138 | ||
139 | /// Parameters built from header parameters, do not change during playback | |
140 | int group_order; ///< order of frame group | |
141 | int fft_order; ///< order of FFT (actually fftorder+1) | |
142 | int frame_size; ///< size of data frame | |
143 | int frequency_range; | |
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) | |
147 | ||
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 | |
155 | ||
156 | /// FFT and tones | |
157 | FFTTone fft_tones[1000]; | |
158 | int fft_tone_start; | |
159 | int fft_tone_end; | |
160 | FFTCoefficient fft_coefs[1000]; | |
161 | int fft_coefs_index; | |
162 | int fft_coefs_min_index[5]; | |
163 | int fft_coefs_max_index[5]; | |
164 | int fft_level_exp[6]; | |
165 | RDFTContext rdft_ctx; | |
166 | QDM2FFT fft; | |
167 | ||
168 | /// I/O data | |
169 | const uint8_t *compressed_data; | |
170 | int compressed_size; | |
171 | float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2]; | |
172 | ||
173 | /// Synthesis filter | |
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]; | |
179 | ||
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]; | |
190 | ||
191 | // Flags | |
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 | |
195 | ||
196 | int sub_packet; | |
197 | int noise_idx; ///< index for dithering noise table | |
198 | } QDM2Context; | |
199 | ||
200 | ||
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]; | |
214 | ||
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, | |
217 | }; | |
218 | ||
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 | |
221 | }; | |
222 | ||
223 | static av_cold void qdm2_init_vlc(void) | |
224 | { | |
225 | static VLC_TYPE qdm2_table[3838][2]; | |
226 | ||
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); | |
233 | ||
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); | |
240 | ||
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); | |
247 | ||
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] - | |
250 | qdm2_vlc_offs[3]; | |
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); | |
255 | ||
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); | |
262 | ||
263 | fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; | |
264 | fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - | |
265 | qdm2_vlc_offs[5]; | |
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); | |
270 | ||
271 | fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; | |
272 | fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - | |
273 | qdm2_vlc_offs[6]; | |
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); | |
278 | ||
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] - | |
282 | qdm2_vlc_offs[7]; | |
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); | |
287 | ||
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] - | |
291 | qdm2_vlc_offs[8]; | |
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); | |
296 | ||
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] - | |
300 | qdm2_vlc_offs[9]; | |
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); | |
305 | ||
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); | |
312 | ||
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); | |
319 | ||
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] - | |
323 | qdm2_vlc_offs[12]; | |
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); | |
328 | ||
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] - | |
332 | qdm2_vlc_offs[13]; | |
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); | |
337 | ||
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] - | |
341 | qdm2_vlc_offs[14]; | |
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); | |
346 | ||
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] - | |
350 | qdm2_vlc_offs[15]; | |
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); | |
355 | ||
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] - | |
359 | qdm2_vlc_offs[16]; | |
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); | |
364 | } | |
365 | ||
366 | static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth) | |
367 | { | |
368 | int value; | |
369 | ||
370 | value = get_vlc2(gb, vlc->table, vlc->bits, depth); | |
371 | ||
372 | /* stage-2, 3 bits exponent escape sequence */ | |
373 | if (value-- == 0) | |
374 | value = get_bits(gb, get_bits(gb, 3) + 1); | |
375 | ||
376 | /* stage-3, optional */ | |
377 | if (flag) { | |
378 | int tmp; | |
379 | ||
380 | if (value >= 60) { | |
381 | av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value); | |
382 | return 0; | |
383 | } | |
384 | ||
385 | tmp= vlc_stage3_values[value]; | |
386 | ||
387 | if ((value & ~3) > 0) | |
388 | tmp += get_bits(gb, (value >> 2)); | |
389 | value = tmp; | |
390 | } | |
391 | ||
392 | return value; | |
393 | } | |
394 | ||
395 | static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth) | |
396 | { | |
397 | int value = qdm2_get_vlc(gb, vlc, 0, depth); | |
398 | ||
399 | return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); | |
400 | } | |
401 | ||
402 | /** | |
403 | * QDM2 checksum | |
404 | * | |
405 | * @param data pointer to data to be checksum'ed | |
406 | * @param length data length | |
407 | * @param value checksum value | |
408 | * | |
409 | * @return 0 if checksum is OK | |
410 | */ | |
411 | static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value) | |
412 | { | |
413 | int i; | |
414 | ||
415 | for (i = 0; i < length; i++) | |
416 | value -= data[i]; | |
417 | ||
418 | return (uint16_t)(value & 0xffff); | |
419 | } | |
420 | ||
421 | /** | |
422 | * Fill a QDM2SubPacket structure with packet type, size, and data pointer. | |
423 | * | |
424 | * @param gb bitreader context | |
425 | * @param sub_packet packet under analysis | |
426 | */ | |
427 | static void qdm2_decode_sub_packet_header(GetBitContext *gb, | |
428 | QDM2SubPacket *sub_packet) | |
429 | { | |
430 | sub_packet->type = get_bits(gb, 8); | |
431 | ||
432 | if (sub_packet->type == 0) { | |
433 | sub_packet->size = 0; | |
434 | sub_packet->data = NULL; | |
435 | } else { | |
436 | sub_packet->size = get_bits(gb, 8); | |
437 | ||
438 | if (sub_packet->type & 0x80) { | |
439 | sub_packet->size <<= 8; | |
440 | sub_packet->size |= get_bits(gb, 8); | |
441 | sub_packet->type &= 0x7f; | |
442 | } | |
443 | ||
444 | if (sub_packet->type == 0x7f) | |
445 | sub_packet->type |= (get_bits(gb, 8) << 8); | |
446 | ||
447 | // FIXME: this depends on bitreader-internal data | |
448 | sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; | |
449 | } | |
450 | ||
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); | |
453 | } | |
454 | ||
455 | /** | |
456 | * Return node pointer to first packet of requested type in list. | |
457 | * | |
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 | |
461 | */ | |
462 | static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list, | |
463 | int type) | |
464 | { | |
465 | while (list && list->packet) { | |
466 | if (list->packet->type == type) | |
467 | return list; | |
468 | list = list->next; | |
469 | } | |
470 | return NULL; | |
471 | } | |
472 | ||
473 | /** | |
474 | * Replace 8 elements with their average value. | |
475 | * Called by qdm2_decode_superblock before starting subblock decoding. | |
476 | * | |
477 | * @param q context | |
478 | */ | |
479 | static void average_quantized_coeffs(QDM2Context *q) | |
480 | { | |
481 | int i, j, n, ch, sum; | |
482 | ||
483 | n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; | |
484 | ||
485 | for (ch = 0; ch < q->nb_channels; ch++) | |
486 | for (i = 0; i < n; i++) { | |
487 | sum = 0; | |
488 | ||
489 | for (j = 0; j < 8; j++) | |
490 | sum += q->quantized_coeffs[ch][i][j]; | |
491 | ||
492 | sum /= 8; | |
493 | if (sum > 0) | |
494 | sum--; | |
495 | ||
496 | for (j = 0; j < 8; j++) | |
497 | q->quantized_coeffs[ch][i][j] = sum; | |
498 | } | |
499 | } | |
500 | ||
501 | /** | |
502 | * Build subband samples with noise weighted by q->tone_level. | |
503 | * Called by synthfilt_build_sb_samples. | |
504 | * | |
505 | * @param q context | |
506 | * @param sb subband index | |
507 | */ | |
508 | static void build_sb_samples_from_noise(QDM2Context *q, int sb) | |
509 | { | |
510 | int ch, j; | |
511 | ||
512 | FIX_NOISE_IDX(q->noise_idx); | |
513 | ||
514 | if (!q->nb_channels) | |
515 | return; | |
516 | ||
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]; | |
523 | } | |
524 | } | |
525 | } | |
526 | ||
527 | /** | |
528 | * Called while processing data from subpackets 11 and 12. | |
529 | * Used after making changes to coding_method array. | |
530 | * | |
531 | * @param sb subband index | |
532 | * @param channels number of channels | |
533 | * @param coding_method q->coding_method[0][0][0] | |
534 | */ | |
535 | static int fix_coding_method_array(int sb, int channels, | |
536 | sb_int8_array coding_method) | |
537 | { | |
538 | int j, k; | |
539 | int ch; | |
540 | int run, case_val; | |
541 | ||
542 | for (ch = 0; ch < channels; ch++) { | |
543 | for (j = 0; j < 64; ) { | |
544 | if (coding_method[ch][sb][j] < 8) | |
545 | return -1; | |
546 | if ((coding_method[ch][sb][j] - 8) > 22) { | |
547 | run = 1; | |
548 | case_val = 8; | |
549 | } else { | |
550 | switch (switchtable[coding_method[ch][sb][j] - 8]) { | |
551 | case 0: run = 10; | |
552 | case_val = 10; | |
553 | break; | |
554 | case 1: run = 1; | |
555 | case_val = 16; | |
556 | break; | |
557 | case 2: run = 5; | |
558 | case_val = 24; | |
559 | break; | |
560 | case 3: run = 3; | |
561 | case_val = 30; | |
562 | break; | |
563 | case 4: run = 1; | |
564 | case_val = 30; | |
565 | break; | |
566 | case 5: run = 1; | |
567 | case_val = 8; | |
568 | break; | |
569 | default: run = 1; | |
570 | case_val = 8; | |
571 | break; | |
572 | } | |
573 | } | |
574 | for (k = 0; k < run; k++) { | |
575 | if (j + k < 128) { | |
576 | if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) { | |
577 | if (k > 0) { | |
578 | SAMPLES_NEEDED | |
579 | //not debugged, almost never used | |
580 | memset(&coding_method[ch][sb][j + k], case_val, | |
581 | k *sizeof(int8_t)); | |
582 | memset(&coding_method[ch][sb][j + k], case_val, | |
583 | 3 * sizeof(int8_t)); | |
584 | } | |
585 | } | |
586 | } | |
587 | } | |
588 | j += run; | |
589 | } | |
590 | } | |
591 | return 0; | |
592 | } | |
593 | ||
594 | /** | |
595 | * Related to synthesis filter | |
596 | * Called by process_subpacket_10 | |
597 | * | |
598 | * @param q context | |
599 | * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 | |
600 | */ | |
601 | static void fill_tone_level_array(QDM2Context *q, int flag) | |
602 | { | |
603 | int i, sb, ch, sb_used; | |
604 | int tmp, tab; | |
605 | ||
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]; | |
612 | else | |
613 | tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; | |
614 | if(tmp < 0) | |
615 | tmp += 0xff; | |
616 | q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; | |
617 | } | |
618 | ||
619 | sb_used = QDM2_SB_USED(q->sub_sampling); | |
620 | ||
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; | |
628 | else | |
629 | q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; | |
630 | } | |
631 | } else { | |
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; | |
644 | else | |
645 | q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
646 | } | |
647 | } else { | |
648 | if (sb > 4) { | |
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; | |
657 | else | |
658 | q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
659 | } | |
660 | } else { | |
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; | |
666 | else | |
667 | q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
668 | } | |
669 | } | |
670 | } | |
671 | } | |
672 | } | |
673 | } | |
674 | ||
675 | /** | |
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. | |
681 | * | |
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 | |
689 | */ | |
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, | |
693 | int nb_channels, | |
694 | int c, int superblocktype_2_3, | |
695 | int cm_table_select) | |
696 | { | |
697 | int ch, sb, j; | |
698 | int tmp, acc, esp_40, comp; | |
699 | int add1, add2, add3, add4; | |
700 | int64_t multres; | |
701 | ||
702 | if (!superblocktype_2_3) { | |
703 | /* This case is untested, no samples available */ | |
704 | avpriv_request_sample(NULL, "!superblocktype_2_3"); | |
705 | return; | |
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; | |
710 | if (add1 < 0) | |
711 | add1 = 0; | |
712 | add2 = add3 = add4 = 0; | |
713 | if (sb > 1) { | |
714 | add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; | |
715 | if (add2 < 0) | |
716 | add2 = 0; | |
717 | } | |
718 | if (sb > 0) { | |
719 | add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; | |
720 | if (add3 < 0) | |
721 | add3 = 0; | |
722 | } | |
723 | if (sb < 29) { | |
724 | add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; | |
725 | if (add4 < 0) | |
726 | add4 = 0; | |
727 | } | |
728 | tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; | |
729 | if (tmp < 0) | |
730 | tmp = 0; | |
731 | tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; | |
732 | } | |
733 | tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; | |
734 | } | |
735 | acc = 0; | |
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]; | |
740 | ||
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; | |
747 | if (comp < 0) | |
748 | comp += 0xff; | |
749 | comp /= 256; // signed shift | |
750 | switch(sb) { | |
751 | case 0: | |
752 | if (comp < 30) | |
753 | comp = 30; | |
754 | comp += 15; | |
755 | break; | |
756 | case 1: | |
757 | if (comp < 24) | |
758 | comp = 24; | |
759 | comp += 10; | |
760 | break; | |
761 | case 2: | |
762 | case 3: | |
763 | case 4: | |
764 | if (comp < 16) | |
765 | comp = 16; | |
766 | } | |
767 | if (comp <= 5) | |
768 | tmp = 0; | |
769 | else if (comp <= 10) | |
770 | tmp = 10; | |
771 | else if (comp <= 16) | |
772 | tmp = 16; | |
773 | else if (comp <= 24) | |
774 | tmp = -1; | |
775 | else | |
776 | tmp = 0; | |
777 | coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; | |
778 | } | |
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++) | |
784 | if (sb >= 10) { | |
785 | if (coding_method[ch][sb][j] < 10) | |
786 | coding_method[ch][sb][j] = 10; | |
787 | } else { | |
788 | if (sb >= 2) { | |
789 | if (coding_method[ch][sb][j] < 16) | |
790 | coding_method[ch][sb][j] = 16; | |
791 | } else { | |
792 | if (coding_method[ch][sb][j] < 30) | |
793 | coding_method[ch][sb][j] = 30; | |
794 | } | |
795 | } | |
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]; | |
801 | } | |
802 | } | |
803 | ||
804 | /** | |
805 | * | |
806 | * Called by process_subpacket_11 to process more data from subpacket 11 | |
807 | * with sb 0-8. | |
808 | * Called by process_subpacket_12 to process data from subpacket 12 with | |
809 | * sb 8-sb_used. | |
810 | * | |
811 | * @param q context | |
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) | |
816 | */ | |
817 | static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, | |
818 | int length, int sb_min, int sb_max) | |
819 | { | |
820 | int sb, j, k, n, ch, run, channels; | |
821 | int joined_stereo, zero_encoding; | |
822 | int type34_first; | |
823 | float type34_div = 0; | |
824 | float type34_predictor; | |
825 | float samples[10]; | |
826 | int sign_bits[16] = {0}; | |
827 | ||
828 | if (length == 0) { | |
829 | // If no data use noise | |
830 | for (sb=sb_min; sb < sb_max; sb++) | |
831 | build_sb_samples_from_noise(q, sb); | |
832 | ||
833 | return 0; | |
834 | } | |
835 | ||
836 | for (sb = sb_min; sb < sb_max; sb++) { | |
837 | channels = q->nb_channels; | |
838 | ||
839 | if (q->nb_channels <= 1 || sb < 12) | |
840 | joined_stereo = 0; | |
841 | else if (sb >= 24) | |
842 | joined_stereo = 1; | |
843 | else | |
844 | joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0; | |
845 | ||
846 | if (joined_stereo) { | |
847 | if (get_bits_left(gb) >= 16) | |
848 | for (j = 0; j < 16; j++) | |
849 | sign_bits[j] = get_bits1(gb); | |
850 | ||
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]; | |
854 | ||
855 | if (fix_coding_method_array(sb, q->nb_channels, | |
856 | q->coding_method)) { | |
857 | av_log(NULL, AV_LOG_ERROR, "coding method invalid\n"); | |
858 | build_sb_samples_from_noise(q, sb); | |
859 | continue; | |
860 | } | |
861 | channels = 1; | |
862 | } | |
863 | ||
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; | |
868 | type34_first = 1; | |
869 | ||
870 | for (j = 0; j < 128; ) { | |
871 | switch (q->coding_method[ch][sb][j / 2]) { | |
872 | case 8: | |
873 | if (get_bits_left(gb) >= 10) { | |
874 | if (zero_encoding) { | |
875 | for (k = 0; k < 5; k++) { | |
876 | if ((j + 2 * k) >= 128) | |
877 | break; | |
878 | samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; | |
879 | } | |
880 | } else { | |
881 | n = get_bits(gb, 8); | |
882 | if (n >= 243) { | |
883 | av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); | |
884 | return AVERROR_INVALIDDATA; | |
885 | } | |
886 | ||
887 | for (k = 0; k < 5; k++) | |
888 | samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
889 | } | |
890 | for (k = 0; k < 5; k++) | |
891 | samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
892 | } else { | |
893 | for (k = 0; k < 10; k++) | |
894 | samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
895 | } | |
896 | run = 10; | |
897 | break; | |
898 | ||
899 | case 10: | |
900 | if (get_bits_left(gb) >= 1) { | |
901 | float f = 0.81; | |
902 | ||
903 | if (get_bits1(gb)) | |
904 | f = -f; | |
905 | f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; | |
906 | samples[0] = f; | |
907 | } else { | |
908 | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
909 | } | |
910 | run = 1; | |
911 | break; | |
912 | ||
913 | case 16: | |
914 | if (get_bits_left(gb) >= 10) { | |
915 | if (zero_encoding) { | |
916 | for (k = 0; k < 5; k++) { | |
917 | if ((j + k) >= 128) | |
918 | break; | |
919 | samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; | |
920 | } | |
921 | } else { | |
922 | n = get_bits (gb, 8); | |
923 | if (n >= 243) { | |
924 | av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); | |
925 | return AVERROR_INVALIDDATA; | |
926 | } | |
927 | ||
928 | for (k = 0; k < 5; k++) | |
929 | samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
930 | } | |
931 | } else { | |
932 | for (k = 0; k < 5; k++) | |
933 | samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
934 | } | |
935 | run = 5; | |
936 | break; | |
937 | ||
938 | case 24: | |
939 | if (get_bits_left(gb) >= 7) { | |
940 | n = get_bits(gb, 7); | |
941 | if (n >= 125) { | |
942 | av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n"); | |
943 | return AVERROR_INVALIDDATA; | |
944 | } | |
945 | ||
946 | for (k = 0; k < 3; k++) | |
947 | samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; | |
948 | } else { | |
949 | for (k = 0; k < 3; k++) | |
950 | samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
951 | } | |
952 | run = 3; | |
953 | break; | |
954 | ||
955 | case 30: | |
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; | |
961 | } | |
962 | samples[0] = type30_dequant[index]; | |
963 | } else | |
964 | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
965 | ||
966 | run = 1; | |
967 | break; | |
968 | ||
969 | case 34: | |
970 | if (get_bits_left(gb) >= 7) { | |
971 | if (type34_first) { | |
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]; | |
975 | type34_first = 0; | |
976 | } else { | |
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; | |
981 | } | |
982 | samples[0] = type34_delta[index] / type34_div + type34_predictor; | |
983 | type34_predictor = samples[0]; | |
984 | } | |
985 | } else { | |
986 | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
987 | } | |
988 | run = 1; | |
989 | break; | |
990 | ||
991 | default: | |
992 | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
993 | run = 1; | |
994 | break; | |
995 | } | |
996 | ||
997 | if (joined_stereo) { | |
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]; | |
1005 | else | |
1006 | q->sb_samples[1][j + k][sb] = | |
1007 | q->tone_level[1][sb][(j + k) / 2] * samples[k]; | |
1008 | } | |
1009 | } | |
1010 | } else { | |
1011 | for (k = 0; k < run; k++) | |
1012 | if ((j + k) < 128) | |
1013 | q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k]; | |
1014 | } | |
1015 | ||
1016 | j += run; | |
1017 | } // j loop | |
1018 | } // channel loop | |
1019 | } // subband loop | |
1020 | return 0; | |
1021 | } | |
1022 | ||
1023 | /** | |
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. | |
1029 | * | |
1030 | * @param quantized_coeffs pointer to quantized_coeffs[ch][0] | |
1031 | * @param gb bitreader context | |
1032 | */ | |
1033 | static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs, | |
1034 | GetBitContext *gb) | |
1035 | { | |
1036 | int i, k, run, level, diff; | |
1037 | ||
1038 | if (get_bits_left(gb) < 16) | |
1039 | return -1; | |
1040 | level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); | |
1041 | ||
1042 | quantized_coeffs[0] = level; | |
1043 | ||
1044 | for (i = 0; i < 7; ) { | |
1045 | if (get_bits_left(gb) < 16) | |
1046 | return -1; | |
1047 | run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; | |
1048 | ||
1049 | if (i + run >= 8) | |
1050 | return -1; | |
1051 | ||
1052 | if (get_bits_left(gb) < 16) | |
1053 | return -1; | |
1054 | diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); | |
1055 | ||
1056 | for (k = 1; k <= run; k++) | |
1057 | quantized_coeffs[i + k] = (level + ((k * diff) / run)); | |
1058 | ||
1059 | level += diff; | |
1060 | i += run; | |
1061 | } | |
1062 | return 0; | |
1063 | } | |
1064 | ||
1065 | /** | |
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 | |
1070 | * | |
1071 | * @param q context | |
1072 | * @param gb bitreader context | |
1073 | */ | |
1074 | static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb) | |
1075 | { | |
1076 | int sb, j, k, n, ch; | |
1077 | ||
1078 | for (ch = 0; ch < q->nb_channels; ch++) { | |
1079 | init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb); | |
1080 | ||
1081 | if (get_bits_left(gb) < 16) { | |
1082 | memset(q->quantized_coeffs[ch][0], 0, 8); | |
1083 | break; | |
1084 | } | |
1085 | } | |
1086 | ||
1087 | n = q->sub_sampling + 1; | |
1088 | ||
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) | |
1093 | break; | |
1094 | if (get_bits1(gb)) { | |
1095 | for (k=0; k < 8; k++) { | |
1096 | if (get_bits_left(gb) < 16) | |
1097 | break; | |
1098 | q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); | |
1099 | } | |
1100 | } else { | |
1101 | for (k=0; k < 8; k++) | |
1102 | q->tone_level_idx_hi1[ch][sb][j][k] = 0; | |
1103 | } | |
1104 | } | |
1105 | ||
1106 | n = QDM2_SB_USED(q->sub_sampling) - 4; | |
1107 | ||
1108 | for (sb = 0; sb < n; sb++) | |
1109 | for (ch = 0; ch < q->nb_channels; ch++) { | |
1110 | if (get_bits_left(gb) < 16) | |
1111 | break; | |
1112 | q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); | |
1113 | if (sb > 19) | |
1114 | q->tone_level_idx_hi2[ch][sb] -= 16; | |
1115 | else | |
1116 | for (j = 0; j < 8; j++) | |
1117 | q->tone_level_idx_mid[ch][sb][j] = -16; | |
1118 | } | |
1119 | ||
1120 | n = QDM2_SB_USED(q->sub_sampling) - 5; | |
1121 | ||
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) | |
1126 | break; | |
1127 | q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; | |
1128 | } | |
1129 | } | |
1130 | ||
1131 | /** | |
1132 | * Process subpacket 9, init quantized_coeffs with data from it | |
1133 | * | |
1134 | * @param q context | |
1135 | * @param node pointer to node with packet | |
1136 | */ | |
1137 | static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node) | |
1138 | { | |
1139 | GetBitContext gb; | |
1140 | int i, j, k, n, ch, run, level, diff; | |
1141 | ||
1142 | init_get_bits(&gb, node->packet->data, node->packet->size * 8); | |
1143 | ||
1144 | n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; | |
1145 | ||
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; | |
1150 | ||
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); | |
1154 | ||
1155 | if (j + run >= 8) | |
1156 | return -1; | |
1157 | ||
1158 | for (k = 1; k <= run; k++) | |
1159 | q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run)); | |
1160 | ||
1161 | level += diff; | |
1162 | j += run; | |
1163 | } | |
1164 | } | |
1165 | ||
1166 | for (ch = 0; ch < q->nb_channels; ch++) | |
1167 | for (i = 0; i < 8; i++) | |
1168 | q->quantized_coeffs[ch][0][i] = 0; | |
1169 | ||
1170 | return 0; | |
1171 | } | |
1172 | ||
1173 | /** | |
1174 | * Process subpacket 10 if not null, else | |
1175 | * | |
1176 | * @param q context | |
1177 | * @param node pointer to node with packet | |
1178 | */ | |
1179 | static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node) | |
1180 | { | |
1181 | GetBitContext gb; | |
1182 | ||
1183 | if (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); | |
1187 | } else { | |
1188 | fill_tone_level_array(q, 0); | |
1189 | } | |
1190 | } | |
1191 | ||
1192 | /** | |
1193 | * Process subpacket 11 | |
1194 | * | |
1195 | * @param q context | |
1196 | * @param node pointer to node with packet | |
1197 | */ | |
1198 | static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node) | |
1199 | { | |
1200 | GetBitContext gb; | |
1201 | int length = 0; | |
1202 | ||
1203 | if (node) { | |
1204 | length = node->packet->size * 8; | |
1205 | init_get_bits(&gb, node->packet->data, length); | |
1206 | } | |
1207 | ||
1208 | if (length >= 32) { | |
1209 | int c = get_bits(&gb, 13); | |
1210 | ||
1211 | if (c > 3) | |
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); | |
1216 | } | |
1217 | ||
1218 | synthfilt_build_sb_samples(q, &gb, length, 0, 8); | |
1219 | } | |
1220 | ||
1221 | /** | |
1222 | * Process subpacket 12 | |
1223 | * | |
1224 | * @param q context | |
1225 | * @param node pointer to node with packet | |
1226 | */ | |
1227 | static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node) | |
1228 | { | |
1229 | GetBitContext gb; | |
1230 | int length = 0; | |
1231 | ||
1232 | if (node) { | |
1233 | length = node->packet->size * 8; | |
1234 | init_get_bits(&gb, node->packet->data, length); | |
1235 | } | |
1236 | ||
1237 | synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); | |
1238 | } | |
1239 | ||
1240 | /** | |
1241 | * Process new subpackets for synthesis filter | |
1242 | * | |
1243 | * @param q context | |
1244 | * @param list list with synthesis filter packets (list D) | |
1245 | */ | |
1246 | static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list) | |
1247 | { | |
1248 | QDM2SubPNode *nodes[4]; | |
1249 | ||
1250 | nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); | |
1251 | if (nodes[0]) | |
1252 | process_subpacket_9(q, nodes[0]); | |
1253 | ||
1254 | nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); | |
1255 | if (nodes[1]) | |
1256 | process_subpacket_10(q, nodes[1]); | |
1257 | else | |
1258 | process_subpacket_10(q, NULL); | |
1259 | ||
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]); | |
1263 | else | |
1264 | process_subpacket_11(q, NULL); | |
1265 | ||
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]); | |
1269 | else | |
1270 | process_subpacket_12(q, NULL); | |
1271 | } | |
1272 | ||
1273 | /** | |
1274 | * Decode superblock, fill packet lists. | |
1275 | * | |
1276 | * @param q context | |
1277 | */ | |
1278 | static void qdm2_decode_super_block(QDM2Context *q) | |
1279 | { | |
1280 | GetBitContext gb; | |
1281 | QDM2SubPacket header, *packet; | |
1282 | int i, packet_bytes, sub_packet_size, sub_packets_D; | |
1283 | unsigned int next_index = 0; | |
1284 | ||
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)); | |
1288 | ||
1289 | q->sub_packets_B = 0; | |
1290 | sub_packets_D = 0; | |
1291 | ||
1292 | average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] | |
1293 | ||
1294 | init_get_bits(&gb, q->compressed_data, q->compressed_size * 8); | |
1295 | qdm2_decode_sub_packet_header(&gb, &header); | |
1296 | ||
1297 | if (header.type < 2 || header.type >= 8) { | |
1298 | q->has_errors = 1; | |
1299 | av_log(NULL, AV_LOG_ERROR, "bad superblock type\n"); | |
1300 | return; | |
1301 | } | |
1302 | ||
1303 | q->superblocktype_2_3 = (header.type == 2 || header.type == 3); | |
1304 | packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); | |
1305 | ||
1306 | init_get_bits(&gb, header.data, header.size * 8); | |
1307 | ||
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); | |
1311 | ||
1312 | csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); | |
1313 | ||
1314 | if (csum != 0) { | |
1315 | q->has_errors = 1; | |
1316 | av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n"); | |
1317 | return; | |
1318 | } | |
1319 | } | |
1320 | ||
1321 | q->sub_packet_list_B[0].packet = NULL; | |
1322 | q->sub_packet_list_D[0].packet = NULL; | |
1323 | ||
1324 | for (i = 0; i < 6; i++) | |
1325 | if (--q->fft_level_exp[i] < 0) | |
1326 | q->fft_level_exp[i] = 0; | |
1327 | ||
1328 | for (i = 0; packet_bytes > 0; i++) { | |
1329 | int j; | |
1330 | ||
1331 | if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) { | |
1332 | SAMPLES_NEEDED_2("too many packet bytes"); | |
1333 | return; | |
1334 | } | |
1335 | ||
1336 | q->sub_packet_list_A[i].next = NULL; | |
1337 | ||
1338 | if (i > 0) { | |
1339 | q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; | |
1340 | ||
1341 | /* seek to next block */ | |
1342 | init_get_bits(&gb, header.data, header.size * 8); | |
1343 | skip_bits(&gb, next_index * 8); | |
1344 | ||
1345 | if (next_index >= header.size) | |
1346 | break; | |
1347 | } | |
1348 | ||
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; | |
1354 | ||
1355 | if (packet->type == 0) | |
1356 | break; | |
1357 | ||
1358 | if (sub_packet_size > packet_bytes) { | |
1359 | if (packet->type != 10 && packet->type != 11 && packet->type != 12) | |
1360 | break; | |
1361 | packet->size += packet_bytes - sub_packet_size; | |
1362 | } | |
1363 | ||
1364 | packet_bytes -= sub_packet_size; | |
1365 | ||
1366 | /* add subpacket to 'all subpackets' list */ | |
1367 | q->sub_packet_list_A[i].packet = packet; | |
1368 | ||
1369 | /* add subpacket to related list */ | |
1370 | if (packet->type == 8) { | |
1371 | SAMPLES_NEEDED_2("packet type 8"); | |
1372 | return; | |
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") | |
1384 | return; | |
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); | |
1389 | } | |
1390 | } // Packet bytes loop | |
1391 | ||
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); | |
1399 | } | |
1400 | } | |
1401 | ||
1402 | static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet, | |
1403 | int offset, int duration, int channel, | |
1404 | int exp, int phase) | |
1405 | { | |
1406 | if (q->fft_coefs_min_index[duration] < 0) | |
1407 | q->fft_coefs_min_index[duration] = q->fft_coefs_index; | |
1408 | ||
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++; | |
1416 | } | |
1417 | ||
1418 | static void qdm2_fft_decode_tones(QDM2Context *q, int duration, | |
1419 | GetBitContext *gb, int b) | |
1420 | { | |
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; | |
1424 | int n, offset; | |
1425 | ||
1426 | local_int_4 = 0; | |
1427 | local_int_28 = 0; | |
1428 | local_int_20 = 2; | |
1429 | local_int_8 = (4 - duration); | |
1430 | local_int_10 = 1 << (q->group_order - duration - 1); | |
1431 | offset = 1; | |
1432 | ||
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"); | |
1439 | return; | |
1440 | } | |
1441 | offset = 1; | |
1442 | if (n == 0) { | |
1443 | local_int_4 += local_int_10; | |
1444 | local_int_28 += (1 << local_int_8); | |
1445 | } else { | |
1446 | local_int_4 += 8 * local_int_10; | |
1447 | local_int_28 += (8 << local_int_8); | |
1448 | } | |
1449 | } | |
1450 | offset += (n - 2); | |
1451 | } else { | |
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); | |
1457 | } | |
1458 | } | |
1459 | ||
1460 | if (local_int_4 >= q->group_size) | |
1461 | return; | |
1462 | ||
1463 | local_int_14 = (offset >> local_int_8); | |
1464 | if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table)) | |
1465 | return; | |
1466 | ||
1467 | if (q->nb_channels > 1) { | |
1468 | channel = get_bits1(gb); | |
1469 | stereo = get_bits1(gb); | |
1470 | } else { | |
1471 | channel = 0; | |
1472 | stereo = 0; | |
1473 | } | |
1474 | ||
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; | |
1478 | ||
1479 | phase = get_bits(gb, 3); | |
1480 | stereo_exp = 0; | |
1481 | stereo_phase = 0; | |
1482 | ||
1483 | if (stereo) { | |
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) | |
1487 | stereo_phase += 8; | |
1488 | } | |
1489 | ||
1490 | if (q->frequency_range > (local_int_14 + 1)) { | |
1491 | int sub_packet = (local_int_20 + local_int_28); | |
1492 | ||
1493 | qdm2_fft_init_coefficient(q, sub_packet, offset, duration, | |
1494 | channel, exp, phase); | |
1495 | if (stereo) | |
1496 | qdm2_fft_init_coefficient(q, sub_packet, offset, duration, | |
1497 | 1 - channel, | |
1498 | stereo_exp, stereo_phase); | |
1499 | } | |
1500 | offset++; | |
1501 | } | |
1502 | } | |
1503 | ||
1504 | static void qdm2_decode_fft_packets(QDM2Context *q) | |
1505 | { | |
1506 | int i, j, min, max, value, type, unknown_flag; | |
1507 | GetBitContext gb; | |
1508 | ||
1509 | if (!q->sub_packet_list_B[0].packet) | |
1510 | return; | |
1511 | ||
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; | |
1516 | ||
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; | |
1520 | ||
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) { | |
1525 | min = value; | |
1526 | packet = q->sub_packet_list_B[j].packet; | |
1527 | } | |
1528 | } | |
1529 | ||
1530 | max = min; | |
1531 | ||
1532 | /* check for errors (?) */ | |
1533 | if (!packet) | |
1534 | return; | |
1535 | ||
1536 | if (i == 0 && | |
1537 | (packet->type < 16 || packet->type >= 48 || | |
1538 | fft_subpackets[packet->type - 16])) | |
1539 | return; | |
1540 | ||
1541 | /* decode FFT tones */ | |
1542 | init_get_bits(&gb, packet->data, packet->size * 8); | |
1543 | ||
1544 | if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) | |
1545 | unknown_flag = 1; | |
1546 | else | |
1547 | unknown_flag = 0; | |
1548 | ||
1549 | type = packet->type; | |
1550 | ||
1551 | if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { | |
1552 | int duration = q->sub_sampling + 5 - (type & 15); | |
1553 | ||
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); | |
1564 | } | |
1565 | } // Loop on B packets | |
1566 | ||
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) { | |
1570 | if (j >= 0) | |
1571 | q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; | |
1572 | j = i; | |
1573 | } | |
1574 | if (j >= 0) | |
1575 | q->fft_coefs_max_index[j] = q->fft_coefs_index; | |
1576 | } | |
1577 | ||
1578 | static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone) | |
1579 | { | |
1580 | float level, f[6]; | |
1581 | int i; | |
1582 | QDM2Complex c; | |
1583 | const double iscale = 2.0 * M_PI / 512.0; | |
1584 | ||
1585 | tone->phase += tone->phase_shift; | |
1586 | ||
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); | |
1591 | ||
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; | |
1598 | } else { | |
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 += | |
1607 | c.re * f[i]; | |
1608 | tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += | |
1609 | c.im * ((tone->cutoff <= i) ? -f[i] : f[i]); | |
1610 | } | |
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]; | |
1614 | } | |
1615 | } | |
1616 | ||
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; | |
1621 | } | |
1622 | } | |
1623 | ||
1624 | static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet) | |
1625 | { | |
1626 | int i, j, ch; | |
1627 | const double iscale = 0.25 * M_PI; | |
1628 | ||
1629 | for (ch = 0; ch < q->channels; ch++) { | |
1630 | memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); | |
1631 | } | |
1632 | ||
1633 | ||
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++) { | |
1637 | float level; | |
1638 | QDM2Complex c; | |
1639 | ||
1640 | if (q->fft_coefs[i].sub_packet != sub_packet) | |
1641 | break; | |
1642 | ||
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]; | |
1645 | ||
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; | |
1652 | } | |
1653 | ||
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; | |
1658 | } | |
1659 | ||
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++) { | |
1664 | int offset, four_i; | |
1665 | FFTTone tone; | |
1666 | ||
1667 | if (q->fft_coefs[j].sub_packet != sub_packet) | |
1668 | break; | |
1669 | ||
1670 | four_i = (4 - i); | |
1671 | offset = q->fft_coefs[j].offset >> four_i; | |
1672 | ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; | |
1673 | ||
1674 | if (offset < q->frequency_range) { | |
1675 | if (offset < 2) | |
1676 | tone.cutoff = offset; | |
1677 | else | |
1678 | tone.cutoff = (offset >= 60) ? 3 : 2; | |
1679 | ||
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); | |
1685 | tone.duration = i; | |
1686 | tone.time_index = 0; | |
1687 | ||
1688 | qdm2_fft_generate_tone(q, &tone); | |
1689 | } | |
1690 | } | |
1691 | q->fft_coefs_min_index[i] = j; | |
1692 | } | |
1693 | } | |
1694 | ||
1695 | static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet) | |
1696 | { | |
1697 | const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; | |
1698 | float *out = q->output_buffer + channel; | |
1699 | int i; | |
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; | |
1708 | } | |
1709 | } | |
1710 | ||
1711 | /** | |
1712 | * @param q context | |
1713 | * @param index subpacket number | |
1714 | */ | |
1715 | static void qdm2_synthesis_filter(QDM2Context *q, int index) | |
1716 | { | |
1717 | int i, k, ch, sb_used, sub_sampling, dither_state = 0; | |
1718 | ||
1719 | /* copy sb_samples */ | |
1720 | sb_used = QDM2_SB_USED(q->sub_sampling); | |
1721 | ||
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; | |
1726 | ||
1727 | for (ch = 0; ch < q->nb_channels; ch++) { | |
1728 | float *samples_ptr = q->samples + ch; | |
1729 | ||
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; | |
1737 | } | |
1738 | } | |
1739 | ||
1740 | /* add samples to output buffer */ | |
1741 | sub_sampling = (4 >> q->sub_sampling); | |
1742 | ||
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]; | |
1746 | } | |
1747 | ||
1748 | /** | |
1749 | * Init static data (does not depend on specific file) | |
1750 | * | |
1751 | * @param q context | |
1752 | */ | |
1753 | static av_cold void qdm2_init_static_data(void) { | |
1754 | static int done; | |
1755 | ||
1756 | if(done) | |
1757 | return; | |
1758 | ||
1759 | qdm2_init_vlc(); | |
1760 | ff_mpa_synth_init_float(ff_mpa_synth_window_float); | |
1761 | softclip_table_init(); | |
1762 | rnd_table_init(); | |
1763 | init_noise_samples(); | |
1764 | ||
1765 | done = 1; | |
1766 | } | |
1767 | ||
1768 | /** | |
1769 | * Init parameters from codec extradata | |
1770 | */ | |
1771 | static av_cold int qdm2_decode_init(AVCodecContext *avctx) | |
1772 | { | |
1773 | QDM2Context *s = avctx->priv_data; | |
1774 | uint8_t *extradata; | |
1775 | int extradata_size; | |
1776 | int tmp_val, tmp, size; | |
1777 | ||
1778 | qdm2_init_static_data(); | |
1779 | ||
1780 | /* extradata parsing | |
1781 | ||
1782 | Structure: | |
1783 | wave { | |
1784 | frma (QDM2) | |
1785 | QDCA | |
1786 | QDCP | |
1787 | } | |
1788 | ||
1789 | 32 size (including this field) | |
1790 | 32 tag (=frma) | |
1791 | 32 type (=QDM2 or QDMC) | |
1792 | ||
1793 | 32 size (including this field, in bytes) | |
1794 | 32 tag (=QDCA) // maybe mandatory parameters | |
1795 | 32 unknown (=1) | |
1796 | 32 channels (=2) | |
1797 | 32 samplerate (=44100) | |
1798 | 32 bitrate (=96000) | |
1799 | 32 block size (=4096) | |
1800 | 32 frame size (=256) (for one channel) | |
1801 | 32 packet size (=1300) | |
1802 | ||
1803 | 32 size (including this field, in bytes) | |
1804 | 32 tag (=QDCP) // maybe some tuneable parameters | |
1805 | 32 float1 (=1.0) | |
1806 | 32 zero ? | |
1807 | 32 float2 (=1.0) | |
1808 | 32 float3 (=1.0) | |
1809 | 32 unknown (27) | |
1810 | 32 unknown (8) | |
1811 | 32 zero ? | |
1812 | */ | |
1813 | ||
1814 | if (!avctx->extradata || (avctx->extradata_size < 48)) { | |
1815 | av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); | |
1816 | return -1; | |
1817 | } | |
1818 | ||
1819 | extradata = avctx->extradata; | |
1820 | extradata_size = avctx->extradata_size; | |
1821 | ||
1822 | while (extradata_size > 7) { | |
1823 | if (!memcmp(extradata, "frmaQDM", 7)) | |
1824 | break; | |
1825 | extradata++; | |
1826 | extradata_size--; | |
1827 | } | |
1828 | ||
1829 | if (extradata_size < 12) { | |
1830 | av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", | |
1831 | extradata_size); | |
1832 | return -1; | |
1833 | } | |
1834 | ||
1835 | if (memcmp(extradata, "frmaQDM", 7)) { | |
1836 | av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); | |
1837 | return -1; | |
1838 | } | |
1839 | ||
1840 | if (extradata[7] == 'C') { | |
1841 | // s->is_qdmc = 1; | |
1842 | av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); | |
1843 | return -1; | |
1844 | } | |
1845 | ||
1846 | extradata += 8; | |
1847 | extradata_size -= 8; | |
1848 | ||
1849 | size = AV_RB32(extradata); | |
1850 | ||
1851 | if(size > extradata_size){ | |
1852 | av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", | |
1853 | extradata_size, size); | |
1854 | return -1; | |
1855 | } | |
1856 | ||
1857 | extradata += 4; | |
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"); | |
1861 | return -1; | |
1862 | } | |
1863 | ||
1864 | extradata += 8; | |
1865 | ||
1866 | avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); | |
1867 | extradata += 4; | |
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; | |
1871 | } | |
1872 | avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO : | |
1873 | AV_CH_LAYOUT_MONO; | |
1874 | ||
1875 | avctx->sample_rate = AV_RB32(extradata); | |
1876 | extradata += 4; | |
1877 | ||
1878 | avctx->bit_rate = AV_RB32(extradata); | |
1879 | extradata += 4; | |
1880 | ||
1881 | s->group_size = AV_RB32(extradata); | |
1882 | extradata += 4; | |
1883 | ||
1884 | s->fft_size = AV_RB32(extradata); | |
1885 | extradata += 4; | |
1886 | ||
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; | |
1891 | } | |
1892 | ||
1893 | s->fft_order = av_log2(s->fft_size) + 1; | |
1894 | ||
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 | |
1898 | ||
1899 | if (s->frame_size > QDM2_MAX_FRAME_SIZE) | |
1900 | return AVERROR_INVALIDDATA; | |
1901 | ||
1902 | s->sub_sampling = s->fft_order - 7; | |
1903 | s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); | |
1904 | ||
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; | |
1913 | } | |
1914 | tmp_val = 0; | |
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; | |
1920 | ||
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; | |
1925 | else | |
1926 | s->coeff_per_sb_select = 2; | |
1927 | ||
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); | |
1931 | return -1; | |
1932 | } | |
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; | |
1936 | } | |
1937 | ||
1938 | ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); | |
1939 | ff_mpadsp_init(&s->mpadsp); | |
1940 | ||
1941 | avctx->sample_fmt = AV_SAMPLE_FMT_S16; | |
1942 | ||
1943 | return 0; | |
1944 | } | |
1945 | ||
1946 | static av_cold int qdm2_decode_close(AVCodecContext *avctx) | |
1947 | { | |
1948 | QDM2Context *s = avctx->priv_data; | |
1949 | ||
1950 | ff_rdft_end(&s->rdft_ctx); | |
1951 | ||
1952 | return 0; | |
1953 | } | |
1954 | ||
1955 | static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out) | |
1956 | { | |
1957 | int ch, i; | |
1958 | const int frame_size = (q->frame_size * q->channels); | |
1959 | ||
1960 | if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2) | |
1961 | return -1; | |
1962 | ||
1963 | /* select input buffer */ | |
1964 | q->compressed_data = in; | |
1965 | q->compressed_size = q->checksum_size; | |
1966 | ||
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)); | |
1970 | ||
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); | |
1976 | } | |
1977 | ||
1978 | /* parse subpackets */ | |
1979 | if (!q->has_errors) { | |
1980 | if (q->sub_packet == 2) | |
1981 | qdm2_decode_fft_packets(q); | |
1982 | ||
1983 | qdm2_fft_tone_synthesizer(q, q->sub_packet); | |
1984 | } | |
1985 | ||
1986 | /* sound synthesis stage 1 (FFT) */ | |
1987 | for (ch = 0; ch < q->channels; ch++) { | |
1988 | qdm2_calculate_fft(q, ch, q->sub_packet); | |
1989 | ||
1990 | if (!q->has_errors && q->sub_packet_list_C[0].packet) { | |
1991 | SAMPLES_NEEDED_2("has errors, and C list is not empty") | |
1992 | return -1; | |
1993 | } | |
1994 | } | |
1995 | ||
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); | |
1999 | ||
2000 | q->sub_packet = (q->sub_packet + 1) % 16; | |
2001 | ||
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]; | |
2005 | ||
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]; | |
2010 | ||
2011 | out[i] = value; | |
2012 | } | |
2013 | ||
2014 | return 0; | |
2015 | } | |
2016 | ||
2017 | static int qdm2_decode_frame(AVCodecContext *avctx, void *data, | |
2018 | int *got_frame_ptr, AVPacket *avpkt) | |
2019 | { | |
2020 | AVFrame *frame = data; | |
2021 | const uint8_t *buf = avpkt->data; | |
2022 | int buf_size = avpkt->size; | |
2023 | QDM2Context *s = avctx->priv_data; | |
2024 | int16_t *out; | |
2025 | int i, ret; | |
2026 | ||
2027 | if(!buf) | |
2028 | return 0; | |
2029 | if(buf_size < s->checksum_size) | |
2030 | return -1; | |
2031 | ||
2032 | /* get output buffer */ | |
2033 | frame->nb_samples = 16 * s->frame_size; | |
2034 | if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) | |
2035 | return ret; | |
2036 | out = (int16_t *)frame->data[0]; | |
2037 | ||
2038 | for (i = 0; i < 16; i++) { | |
2039 | if (qdm2_decode(s, buf, out) < 0) | |
2040 | return -1; | |
2041 | out += s->channels * s->frame_size; | |
2042 | } | |
2043 | ||
2044 | *got_frame_ptr = 1; | |
2045 | ||
2046 | return s->checksum_size; | |
2047 | } | |
2048 | ||
2049 | AVCodec ff_qdm2_decoder = { | |
2050 | .name = "qdm2", | |
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, | |
2059 | }; |