Mercurial > libavcodec.hg
annotate qdm2.c @ 3357:649ebe86d779 libavcodec
10l
author | michael |
---|---|
date | Sun, 25 Jun 2006 21:22:25 +0000 |
parents | 6bfe6c09d837 |
children | c8c591fe26f8 |
rev | line source |
---|---|
2914 | 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 library is free software; you can redistribute it and/or | |
9 * modify it under the terms of the GNU Lesser General Public | |
10 * License as published by the Free Software Foundation; either | |
11 * version 2 of the License, or (at your option) any later version. | |
12 * | |
13 * This library is distributed in the hope that it will be useful, | |
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
16 * Lesser General Public License for more details. | |
17 * | |
18 * You should have received a copy of the GNU Lesser General Public | |
19 * License along with this library; if not, write to the Free Software | |
3036
0b546eab515d
Update licensing information: The FSF changed postal address.
diego
parents:
2967
diff
changeset
|
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
2914 | 21 * |
22 */ | |
23 | |
24 /** | |
25 * @file qdm2.c | |
26 * QDM2 decoder | |
27 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
28 * The decoder is not perfect yet, there are still some distortions |
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
29 * especially on files encoded with 16 or 8 subbands. |
2914 | 30 */ |
31 | |
32 #include <math.h> | |
33 #include <stddef.h> | |
34 #include <stdio.h> | |
35 | |
36 #define ALT_BITSTREAM_READER_LE | |
37 #include "avcodec.h" | |
38 #include "bitstream.h" | |
39 #include "dsputil.h" | |
40 | |
41 #ifdef CONFIG_MPEGAUDIO_HP | |
42 #define USE_HIGHPRECISION | |
43 #endif | |
44 | |
45 #include "mpegaudio.h" | |
46 | |
47 #include "qdm2data.h" | |
48 | |
49 #undef NDEBUG | |
50 #include <assert.h> | |
51 | |
52 | |
53 #define SOFTCLIP_THRESHOLD 27600 | |
54 #define HARDCLIP_THRESHOLD 35716 | |
55 | |
56 | |
57 #define QDM2_LIST_ADD(list, size, packet) \ | |
58 do { \ | |
59 if (size > 0) { \ | |
60 list[size - 1].next = &list[size]; \ | |
61 } \ | |
62 list[size].packet = packet; \ | |
63 list[size].next = NULL; \ | |
64 size++; \ | |
65 } while(0) | |
66 | |
67 // Result is 8, 16 or 30 | |
68 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling)) | |
69 | |
70 #define FIX_NOISE_IDX(noise_idx) \ | |
71 if ((noise_idx) >= 3840) \ | |
72 (noise_idx) -= 3840; \ | |
73 | |
74 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)]) | |
75 | |
76 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb))) | |
77 | |
78 #define SAMPLES_NEEDED \ | |
79 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); | |
80 | |
81 #define SAMPLES_NEEDED_2(why) \ | |
82 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); | |
83 | |
84 | |
85 typedef int8_t sb_int8_array[2][30][64]; | |
86 | |
87 /** | |
88 * Subpacket | |
89 */ | |
90 typedef struct { | |
91 int type; ///< subpacket type | |
92 unsigned int size; ///< subpacket size | |
93 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) | |
94 } QDM2SubPacket; | |
95 | |
96 /** | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
97 * A node in the subpacket list |
2914 | 98 */ |
99 typedef struct _QDM2SubPNode { | |
100 QDM2SubPacket *packet; ///< packet | |
101 struct _QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node | |
102 } QDM2SubPNode; | |
103 | |
104 typedef struct { | |
105 float level; | |
106 float *samples_im; | |
107 float *samples_re; | |
108 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 float re; | |
126 float im; | |
127 } QDM2Complex; | |
128 | |
129 typedef struct { | |
130 QDM2Complex complex[256 + 1] __attribute__((aligned(16))); | |
131 float samples_im[MPA_MAX_CHANNELS][256]; | |
132 float samples_re[MPA_MAX_CHANNELS][256]; | |
133 } QDM2FFT; | |
134 | |
135 /** | |
136 * QDM2 decoder context | |
137 */ | |
138 typedef struct { | |
139 /// Parameters from codec header, do not change during playback | |
140 int nb_channels; ///< number of channels | |
141 int channels; ///< number of channels | |
142 int group_size; ///< size of frame group (16 frames per group) | |
143 int fft_size; ///< size of FFT, in complex numbers | |
144 int checksum_size; ///< size of data block, used also for checksum | |
145 | |
146 /// Parameters built from header parameters, do not change during playback | |
147 int group_order; ///< order of frame group | |
148 int fft_order; ///< order of FFT (actually fftorder+1) | |
149 int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im) | |
150 int frame_size; ///< size of data frame | |
151 int frequency_range; | |
152 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ | |
153 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 | |
154 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) | |
155 | |
156 /// Packets and packet lists | |
157 QDM2SubPacket sub_packets[16]; ///< the packets themselves | |
158 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets | |
159 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list | |
160 int sub_packets_B; ///< number of packets on 'B' list | |
161 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? | |
162 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets | |
163 | |
164 /// FFT and tones | |
165 FFTTone fft_tones[1000]; | |
166 int fft_tone_start; | |
167 int fft_tone_end; | |
168 FFTCoefficient fft_coefs[1000]; | |
169 int fft_coefs_index; | |
170 int fft_coefs_min_index[5]; | |
171 int fft_coefs_max_index[5]; | |
172 int fft_level_exp[6]; | |
173 FFTContext fft_ctx; | |
174 FFTComplex exptab[128]; | |
175 QDM2FFT fft; | |
176 | |
177 /// I/O data | |
178 uint8_t *compressed_data; | |
179 int compressed_size; | |
180 float output_buffer[1024]; | |
181 | |
182 /// Synthesis filter | |
183 MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16))); | |
184 int synth_buf_offset[MPA_MAX_CHANNELS]; | |
185 int32_t sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT] __attribute__((aligned(16))); | |
186 | |
187 /// Mixed temporary data used in decoding | |
188 float tone_level[MPA_MAX_CHANNELS][30][64]; | |
189 int8_t coding_method[MPA_MAX_CHANNELS][30][64]; | |
190 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; | |
191 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; | |
192 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; | |
193 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; | |
194 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; | |
195 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; | |
196 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; | |
197 | |
198 // Flags | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
199 int has_errors; ///< packet has errors |
2914 | 200 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type |
201 int do_synth_filter; ///< used to perform or skip synthesis filter | |
202 | |
203 int sub_packet; | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
204 int noise_idx; ///< index for dithering noise table |
2914 | 205 } QDM2Context; |
206 | |
207 | |
208 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE]; | |
209 | |
210 static VLC vlc_tab_level; | |
211 static VLC vlc_tab_diff; | |
212 static VLC vlc_tab_run; | |
213 static VLC fft_level_exp_alt_vlc; | |
214 static VLC fft_level_exp_vlc; | |
215 static VLC fft_stereo_exp_vlc; | |
216 static VLC fft_stereo_phase_vlc; | |
217 static VLC vlc_tab_tone_level_idx_hi1; | |
218 static VLC vlc_tab_tone_level_idx_mid; | |
219 static VLC vlc_tab_tone_level_idx_hi2; | |
220 static VLC vlc_tab_type30; | |
221 static VLC vlc_tab_type34; | |
222 static VLC vlc_tab_fft_tone_offset[5]; | |
223 | |
224 static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1]; | |
225 static float noise_table[4096]; | |
226 static uint8_t random_dequant_index[256][5]; | |
227 static uint8_t random_dequant_type24[128][3]; | |
228 static float noise_samples[128]; | |
229 | |
230 static MPA_INT mpa_window[512] __attribute__((aligned(16))); | |
231 | |
232 | |
3076 | 233 static void softclip_table_init(void) { |
2914 | 234 int i; |
235 double dfl = SOFTCLIP_THRESHOLD - 32767; | |
236 float delta = 1.0 / -dfl; | |
237 for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++) | |
238 softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF); | |
239 } | |
240 | |
241 | |
242 // random generated table | |
3076 | 243 static void rnd_table_init(void) { |
2914 | 244 int i,j; |
245 uint32_t ldw,hdw; | |
246 uint64_t tmp64_1; | |
247 uint64_t random_seed = 0; | |
248 float delta = 1.0 / 16384.0; | |
249 for(i = 0; i < 4096 ;i++) { | |
250 random_seed = random_seed * 214013 + 2531011; | |
251 noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3; | |
252 } | |
253 | |
254 for (i = 0; i < 256 ;i++) { | |
255 random_seed = 81; | |
256 ldw = i; | |
257 for (j = 0; j < 5 ;j++) { | |
258 random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF); | |
259 ldw = (uint32_t)ldw % (uint32_t)random_seed; | |
260 tmp64_1 = (random_seed * 0x55555556); | |
261 hdw = (uint32_t)(tmp64_1 >> 32); | |
262 random_seed = (uint64_t)(hdw + (ldw >> 31)); | |
263 } | |
264 } | |
265 for (i = 0; i < 128 ;i++) { | |
266 random_seed = 25; | |
267 ldw = i; | |
268 for (j = 0; j < 3 ;j++) { | |
269 random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF); | |
270 ldw = (uint32_t)ldw % (uint32_t)random_seed; | |
271 tmp64_1 = (random_seed * 0x66666667); | |
272 hdw = (uint32_t)(tmp64_1 >> 33); | |
273 random_seed = hdw + (ldw >> 31); | |
274 } | |
275 } | |
276 } | |
277 | |
278 | |
3076 | 279 static void init_noise_samples(void) { |
2914 | 280 int i; |
281 int random_seed = 0; | |
282 float delta = 1.0 / 16384.0; | |
283 for (i = 0; i < 128;i++) { | |
284 random_seed = random_seed * 214013 + 2531011; | |
285 noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0); | |
286 } | |
287 } | |
288 | |
289 | |
3076 | 290 static void qdm2_init_vlc(void) |
2914 | 291 { |
292 init_vlc (&vlc_tab_level, 8, 24, | |
293 vlc_tab_level_huffbits, 1, 1, | |
294 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
295 | |
296 init_vlc (&vlc_tab_diff, 8, 37, | |
297 vlc_tab_diff_huffbits, 1, 1, | |
298 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
299 | |
300 init_vlc (&vlc_tab_run, 5, 6, | |
301 vlc_tab_run_huffbits, 1, 1, | |
302 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
303 | |
304 init_vlc (&fft_level_exp_alt_vlc, 8, 28, | |
305 fft_level_exp_alt_huffbits, 1, 1, | |
306 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
307 | |
308 init_vlc (&fft_level_exp_vlc, 8, 20, | |
309 fft_level_exp_huffbits, 1, 1, | |
310 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
311 | |
312 init_vlc (&fft_stereo_exp_vlc, 6, 7, | |
313 fft_stereo_exp_huffbits, 1, 1, | |
314 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
315 | |
316 init_vlc (&fft_stereo_phase_vlc, 6, 9, | |
317 fft_stereo_phase_huffbits, 1, 1, | |
318 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
319 | |
320 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, | |
321 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, | |
322 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
323 | |
324 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, | |
325 vlc_tab_tone_level_idx_mid_huffbits, 1, 1, | |
326 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
327 | |
328 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, | |
329 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, | |
330 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
331 | |
332 init_vlc (&vlc_tab_type30, 6, 9, | |
333 vlc_tab_type30_huffbits, 1, 1, | |
334 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
335 | |
336 init_vlc (&vlc_tab_type34, 5, 10, | |
337 vlc_tab_type34_huffbits, 1, 1, | |
338 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
339 | |
340 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, | |
341 vlc_tab_fft_tone_offset_0_huffbits, 1, 1, | |
342 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
343 | |
344 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, | |
345 vlc_tab_fft_tone_offset_1_huffbits, 1, 1, | |
346 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
347 | |
348 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, | |
349 vlc_tab_fft_tone_offset_2_huffbits, 1, 1, | |
350 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
351 | |
352 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, | |
353 vlc_tab_fft_tone_offset_3_huffbits, 1, 1, | |
354 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
355 | |
356 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, | |
357 vlc_tab_fft_tone_offset_4_huffbits, 1, 1, | |
358 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
359 } | |
360 | |
361 | |
362 /* for floating point to fixed point conversion */ | |
363 static float f2i_scale = (float) (1 << (FRAC_BITS - 15)); | |
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 = vlc_stage3_values[value]; | |
379 | |
380 if ((value & ~3) > 0) | |
381 tmp += get_bits (gb, (value >> 2)); | |
382 value = tmp; | |
383 } | |
384 | |
385 return value; | |
386 } | |
387 | |
388 | |
389 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) | |
390 { | |
391 int value = qdm2_get_vlc (gb, vlc, 0, depth); | |
392 | |
393 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); | |
394 } | |
395 | |
396 | |
397 /** | |
398 * QDM2 checksum | |
399 * | |
400 * @param data pointer to data to be checksum'ed | |
401 * @param length data length | |
402 * @param value checksum value | |
403 * | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
404 * @return 0 if checksum is OK |
2914 | 405 */ |
406 static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) { | |
407 int i; | |
408 | |
409 for (i=0; i < length; i++) | |
410 value -= data[i]; | |
411 | |
412 return (uint16_t)(value & 0xffff); | |
413 } | |
414 | |
415 | |
416 /** | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
417 * Fills a QDM2SubPacket structure with packet type, size, and data pointer. |
2914 | 418 * |
419 * @param gb bitreader context | |
420 * @param sub_packet packet under analysis | |
421 */ | |
422 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) | |
423 { | |
424 sub_packet->type = get_bits (gb, 8); | |
425 | |
426 if (sub_packet->type == 0) { | |
427 sub_packet->size = 0; | |
428 sub_packet->data = NULL; | |
429 } else { | |
430 sub_packet->size = get_bits (gb, 8); | |
431 | |
432 if (sub_packet->type & 0x80) { | |
433 sub_packet->size <<= 8; | |
434 sub_packet->size |= get_bits (gb, 8); | |
435 sub_packet->type &= 0x7f; | |
436 } | |
437 | |
438 if (sub_packet->type == 0x7f) | |
439 sub_packet->type |= (get_bits (gb, 8) << 8); | |
440 | |
441 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data | |
442 } | |
443 | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
444 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", |
2914 | 445 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); |
446 } | |
447 | |
448 | |
449 /** | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
450 * Return node pointer to first packet of requested type in list. |
2914 | 451 * |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
452 * @param list list of subpackets to be scanned |
2914 | 453 * @param type type of searched subpacket |
454 * @return node pointer for subpacket if found, else NULL | |
455 */ | |
456 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) | |
457 { | |
458 while (list != NULL && list->packet != NULL) { | |
459 if (list->packet->type == type) | |
460 return list; | |
461 list = list->next; | |
462 } | |
463 return NULL; | |
464 } | |
465 | |
466 | |
467 /** | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
468 * Replaces 8 elements with their average value. |
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
469 * Called by qdm2_decode_superblock before starting subblock decoding. |
2914 | 470 * |
471 * @param q context | |
472 */ | |
473 static void average_quantized_coeffs (QDM2Context *q) | |
474 { | |
475 int i, j, n, ch, sum; | |
476 | |
477 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; | |
478 | |
479 for (ch = 0; ch < q->nb_channels; ch++) | |
480 for (i = 0; i < n; i++) { | |
481 sum = 0; | |
482 | |
483 for (j = 0; j < 8; j++) | |
484 sum += q->quantized_coeffs[ch][i][j]; | |
485 | |
486 sum /= 8; | |
487 if (sum > 0) | |
488 sum--; | |
489 | |
490 for (j=0; j < 8; j++) | |
491 q->quantized_coeffs[ch][i][j] = sum; | |
492 } | |
493 } | |
494 | |
495 | |
496 /** | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
497 * Build subband samples with noise weighted by q->tone_level. |
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
498 * Called by synthfilt_build_sb_samples. |
2914 | 499 * |
500 * @param q context | |
501 * @param sb subband index | |
502 */ | |
503 static void build_sb_samples_from_noise (QDM2Context *q, int sb) | |
504 { | |
505 int ch, j; | |
506 | |
507 FIX_NOISE_IDX(q->noise_idx); | |
508 | |
509 if (!q->nb_channels) | |
510 return; | |
511 | |
512 for (ch = 0; ch < q->nb_channels; ch++) | |
513 for (j = 0; j < 64; j++) { | |
514 q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5); | |
515 q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5); | |
516 } | |
517 } | |
518 | |
519 | |
520 /** | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
521 * Called while processing data from subpackets 11 and 12. |
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
522 * Used after making changes to coding_method array. |
2914 | 523 * |
524 * @param sb subband index | |
525 * @param channels number of channels | |
526 * @param coding_method q->coding_method[0][0][0] | |
527 */ | |
3076 | 528 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) |
2914 | 529 { |
530 int j,k; | |
531 int ch; | |
532 int run, case_val; | |
533 int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4}; | |
534 | |
535 for (ch = 0; ch < channels; ch++) { | |
536 for (j = 0; j < 64; ) { | |
537 if((coding_method[ch][sb][j] - 8) > 22) { | |
538 run = 1; | |
539 case_val = 8; | |
540 } else { | |
3333 | 541 switch (switchtable[coding_method[ch][sb][j]-8]) { |
2914 | 542 case 0: run = 10; case_val = 10; break; |
543 case 1: run = 1; case_val = 16; break; | |
544 case 2: run = 5; case_val = 24; break; | |
545 case 3: run = 3; case_val = 30; break; | |
546 case 4: run = 1; case_val = 30; break; | |
547 case 5: run = 1; case_val = 8; break; | |
548 default: run = 1; case_val = 8; break; | |
549 } | |
550 } | |
551 for (k = 0; k < run; k++) | |
552 if (j + k < 128) | |
553 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) | |
554 if (k > 0) { | |
555 SAMPLES_NEEDED | |
556 //not debugged, almost never used | |
557 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t)); | |
558 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); | |
559 } | |
560 j += run; | |
561 } | |
562 } | |
563 } | |
564 | |
565 | |
566 /** | |
567 * Related to synthesis filter | |
568 * Called by process_subpacket_10 | |
569 * | |
570 * @param q context | |
571 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 | |
572 */ | |
573 static void fill_tone_level_array (QDM2Context *q, int flag) | |
574 { | |
575 int i, sb, ch, sb_used; | |
576 int tmp, tab; | |
577 | |
578 // This should never happen | |
579 if (q->nb_channels <= 0) | |
580 return; | |
581 | |
582 for (ch = 0; ch < q->nb_channels; ch++) | |
583 for (sb = 0; sb < 30; sb++) | |
584 for (i = 0; i < 8; i++) { | |
585 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) | |
586 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ | |
587 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; | |
588 else | |
589 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; | |
590 if(tmp < 0) | |
591 tmp += 0xff; | |
592 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; | |
593 } | |
594 | |
595 sb_used = QDM2_SB_USED(q->sub_sampling); | |
596 | |
597 if ((q->superblocktype_2_3 != 0) && !flag) { | |
598 for (sb = 0; sb < sb_used; sb++) | |
599 for (ch = 0; ch < q->nb_channels; ch++) | |
600 for (i = 0; i < 64; i++) { | |
601 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; | |
602 if (q->tone_level_idx[ch][sb][i] < 0) | |
603 q->tone_level[ch][sb][i] = 0; | |
604 else | |
605 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; | |
606 } | |
607 } else { | |
608 tab = q->superblocktype_2_3 ? 0 : 1; | |
609 for (sb = 0; sb < sb_used; sb++) { | |
610 if ((sb >= 4) && (sb <= 23)) { | |
611 for (ch = 0; ch < q->nb_channels; ch++) | |
612 for (i = 0; i < 64; i++) { | |
613 tmp = q->tone_level_idx_base[ch][sb][i / 8] - | |
614 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - | |
615 q->tone_level_idx_mid[ch][sb - 4][i / 8] - | |
616 q->tone_level_idx_hi2[ch][sb - 4]; | |
617 q->tone_level_idx[ch][sb][i] = tmp & 0xff; | |
618 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
619 q->tone_level[ch][sb][i] = 0; | |
620 else | |
621 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
622 } | |
623 } else { | |
624 if (sb > 4) { | |
625 for (ch = 0; ch < q->nb_channels; ch++) | |
626 for (i = 0; i < 64; i++) { | |
627 tmp = q->tone_level_idx_base[ch][sb][i / 8] - | |
628 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - | |
629 q->tone_level_idx_hi2[ch][sb - 4]; | |
630 q->tone_level_idx[ch][sb][i] = tmp & 0xff; | |
631 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
632 q->tone_level[ch][sb][i] = 0; | |
633 else | |
634 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
635 } | |
636 } else { | |
637 for (ch = 0; ch < q->nb_channels; ch++) | |
638 for (i = 0; i < 64; i++) { | |
639 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; | |
640 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
641 q->tone_level[ch][sb][i] = 0; | |
642 else | |
643 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
644 } | |
645 } | |
646 } | |
647 } | |
648 } | |
649 | |
650 return; | |
651 } | |
652 | |
653 | |
654 /** | |
655 * Related to synthesis filter | |
656 * Called by process_subpacket_11 | |
657 * c is built with data from subpacket 11 | |
658 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples | |
659 * | |
2967 | 660 * @param tone_level_idx |
2914 | 661 * @param tone_level_idx_temp |
662 * @param coding_method q->coding_method[0][0][0] | |
663 * @param nb_channels number of channels | |
664 * @param c coming from subpacket 11, passed as 8*c | |
665 * @param superblocktype_2_3 flag based on superblock packet type | |
666 * @param cm_table_select q->cm_table_select | |
667 */ | |
668 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, | |
669 sb_int8_array coding_method, int nb_channels, | |
670 int c, int superblocktype_2_3, int cm_table_select) | |
671 { | |
672 int ch, sb, j; | |
673 int tmp, acc, esp_40, comp; | |
674 int add1, add2, add3, add4; | |
675 int64_t multres; | |
676 | |
677 // This should never happen | |
678 if (nb_channels <= 0) | |
679 return; | |
680 | |
681 if (!superblocktype_2_3) { | |
682 /* This case is untested, no samples available */ | |
683 SAMPLES_NEEDED | |
684 for (ch = 0; ch < nb_channels; ch++) | |
685 for (sb = 0; sb < 30; sb++) { | |
686 for (j = 1; j < 64; j++) { | |
687 add1 = tone_level_idx[ch][sb][j] - 10; | |
688 if (add1 < 0) | |
689 add1 = 0; | |
690 add2 = add3 = add4 = 0; | |
691 if (sb > 1) { | |
692 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; | |
693 if (add2 < 0) | |
694 add2 = 0; | |
695 } | |
696 if (sb > 0) { | |
697 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; | |
698 if (add3 < 0) | |
699 add3 = 0; | |
700 } | |
701 if (sb < 29) { | |
702 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; | |
703 if (add4 < 0) | |
704 add4 = 0; | |
705 } | |
706 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; | |
707 if (tmp < 0) | |
708 tmp = 0; | |
709 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; | |
710 } | |
711 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; | |
712 } | |
713 acc = 0; | |
714 for (ch = 0; ch < nb_channels; ch++) | |
715 for (sb = 0; sb < 30; sb++) | |
716 for (j = 0; j < 64; j++) | |
717 acc += tone_level_idx_temp[ch][sb][j]; | |
718 if (acc) | |
719 tmp = c * 256 / (acc & 0xffff); | |
720 multres = 0x66666667 * (acc * 10); | |
721 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); | |
722 for (ch = 0; ch < nb_channels; ch++) | |
723 for (sb = 0; sb < 30; sb++) | |
724 for (j = 0; j < 64; j++) { | |
725 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; | |
726 if (comp < 0) | |
727 comp += 0xff; | |
728 comp /= 256; // signed shift | |
729 switch(sb) { | |
730 case 0: | |
731 if (comp < 30) | |
732 comp = 30; | |
733 comp += 15; | |
734 break; | |
735 case 1: | |
736 if (comp < 24) | |
737 comp = 24; | |
738 comp += 10; | |
739 break; | |
740 case 2: | |
741 case 3: | |
742 case 4: | |
743 if (comp < 16) | |
744 comp = 16; | |
745 } | |
746 if (comp <= 5) | |
747 tmp = 0; | |
748 else if (comp <= 10) | |
749 tmp = 10; | |
750 else if (comp <= 16) | |
751 tmp = 16; | |
752 else if (comp <= 24) | |
753 tmp = -1; | |
754 else | |
755 tmp = 0; | |
756 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; | |
757 } | |
758 for (sb = 0; sb < 30; sb++) | |
759 fix_coding_method_array(sb, nb_channels, coding_method); | |
760 for (ch = 0; ch < nb_channels; ch++) | |
761 for (sb = 0; sb < 30; sb++) | |
762 for (j = 0; j < 64; j++) | |
763 if (sb >= 10) { | |
764 if (coding_method[ch][sb][j] < 10) | |
765 coding_method[ch][sb][j] = 10; | |
766 } else { | |
767 if (sb >= 2) { | |
768 if (coding_method[ch][sb][j] < 16) | |
769 coding_method[ch][sb][j] = 16; | |
770 } else { | |
771 if (coding_method[ch][sb][j] < 30) | |
772 coding_method[ch][sb][j] = 30; | |
773 } | |
774 } | |
775 } else { // superblocktype_2_3 != 0 | |
776 for (ch = 0; ch < nb_channels; ch++) | |
777 for (sb = 0; sb < 30; sb++) | |
778 for (j = 0; j < 64; j++) | |
779 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; | |
780 } | |
781 | |
782 return; | |
783 } | |
784 | |
785 | |
786 /** | |
787 * | |
788 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8 | |
789 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used | |
790 * | |
791 * @param q context | |
792 * @param gb bitreader context | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
793 * @param length packet length in bits |
2914 | 794 * @param sb_min lower subband processed (sb_min included) |
795 * @param sb_max higher subband processed (sb_max excluded) | |
796 */ | |
797 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) | |
798 { | |
799 int sb, j, k, n, ch, run, channels; | |
800 int joined_stereo, zero_encoding, chs; | |
801 int type34_first; | |
802 float type34_div = 0; | |
803 float type34_predictor; | |
804 float samples[10], sign_bits[16]; | |
805 | |
806 if (length == 0) { | |
807 // If no data use noise | |
808 for (sb=sb_min; sb < sb_max; sb++) | |
809 build_sb_samples_from_noise (q, sb); | |
810 | |
811 return; | |
812 } | |
813 | |
814 for (sb = sb_min; sb < sb_max; sb++) { | |
815 FIX_NOISE_IDX(q->noise_idx); | |
816 | |
817 channels = q->nb_channels; | |
818 | |
819 if (q->nb_channels <= 1 || sb < 12) | |
820 joined_stereo = 0; | |
821 else if (sb >= 24) | |
822 joined_stereo = 1; | |
823 else | |
824 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0; | |
825 | |
826 if (joined_stereo) { | |
827 if (BITS_LEFT(length,gb) >= 16) | |
828 for (j = 0; j < 16; j++) | |
829 sign_bits[j] = get_bits1 (gb); | |
830 | |
831 for (j = 0; j < 64; j++) | |
832 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) | |
833 q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; | |
834 | |
835 fix_coding_method_array(sb, q->nb_channels, q->coding_method); | |
836 channels = 1; | |
837 } | |
838 | |
839 for (ch = 0; ch < channels; ch++) { | |
840 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; | |
841 type34_predictor = 0.0; | |
842 type34_first = 1; | |
843 | |
844 for (j = 0; j < 128; ) { | |
845 switch (q->coding_method[ch][sb][j / 2]) { | |
846 case 8: | |
847 if (BITS_LEFT(length,gb) >= 10) { | |
848 if (zero_encoding) { | |
849 for (k = 0; k < 5; k++) { | |
850 if ((j + 2 * k) >= 128) | |
851 break; | |
852 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; | |
853 } | |
854 } else { | |
855 n = get_bits(gb, 8); | |
856 for (k = 0; k < 5; k++) | |
857 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
858 } | |
859 for (k = 0; k < 5; k++) | |
860 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
861 } else { | |
862 for (k = 0; k < 10; k++) | |
863 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
864 } | |
865 run = 10; | |
866 break; | |
867 | |
868 case 10: | |
869 if (BITS_LEFT(length,gb) >= 1) { | |
870 float f = 0.81; | |
871 | |
872 if (get_bits1(gb)) | |
873 f = -f; | |
874 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; | |
875 samples[0] = f; | |
876 } else { | |
877 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
878 } | |
879 run = 1; | |
880 break; | |
881 | |
882 case 16: | |
883 if (BITS_LEFT(length,gb) >= 10) { | |
884 if (zero_encoding) { | |
885 for (k = 0; k < 5; k++) { | |
886 if ((j + k) >= 128) | |
887 break; | |
888 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; | |
889 } | |
890 } else { | |
891 n = get_bits (gb, 8); | |
892 for (k = 0; k < 5; k++) | |
893 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
894 } | |
895 } else { | |
896 for (k = 0; k < 5; k++) | |
897 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
898 } | |
899 run = 5; | |
900 break; | |
901 | |
902 case 24: | |
903 if (BITS_LEFT(length,gb) >= 7) { | |
904 n = get_bits(gb, 7); | |
905 for (k = 0; k < 3; k++) | |
906 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; | |
907 } else { | |
908 for (k = 0; k < 3; k++) | |
909 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
910 } | |
911 run = 3; | |
912 break; | |
913 | |
914 case 30: | |
915 if (BITS_LEFT(length,gb) >= 4) | |
916 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)]; | |
917 else | |
918 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
2967 | 919 |
2914 | 920 run = 1; |
921 break; | |
922 | |
923 case 34: | |
924 if (BITS_LEFT(length,gb) >= 7) { | |
925 if (type34_first) { | |
926 type34_div = (float)(1 << get_bits(gb, 2)); | |
927 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; | |
928 type34_predictor = samples[0]; | |
929 type34_first = 0; | |
930 } else { | |
931 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor; | |
932 type34_predictor = samples[0]; | |
933 } | |
934 } else { | |
935 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
936 } | |
937 run = 1; | |
938 break; | |
939 | |
940 default: | |
941 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
942 run = 1; | |
943 break; | |
944 } | |
945 | |
946 if (joined_stereo) { | |
947 float tmp[10][MPA_MAX_CHANNELS]; | |
948 | |
949 for (k = 0; k < run; k++) { | |
950 tmp[k][0] = samples[k]; | |
951 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; | |
952 } | |
953 for (chs = 0; chs < q->nb_channels; chs++) | |
954 for (k = 0; k < run; k++) | |
955 if ((j + k) < 128) | |
956 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5); | |
957 } else { | |
958 for (k = 0; k < run; k++) | |
959 if ((j + k) < 128) | |
960 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5); | |
961 } | |
962 | |
963 j += run; | |
964 } // j loop | |
965 } // channel loop | |
966 } // subband loop | |
967 } | |
968 | |
969 | |
970 /** | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
971 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]). |
2914 | 972 * This is similar to process_subpacket_9, but for a single channel and for element [0] |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
973 * same VLC tables as process_subpacket_9 are used. |
2914 | 974 * |
975 * @param q context | |
976 * @param quantized_coeffs pointer to quantized_coeffs[ch][0] | |
977 * @param gb bitreader context | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
978 * @param length packet length in bits |
2914 | 979 */ |
980 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) | |
981 { | |
982 int i, k, run, level, diff; | |
983 | |
984 if (BITS_LEFT(length,gb) < 16) | |
985 return; | |
986 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); | |
987 | |
988 quantized_coeffs[0] = level; | |
989 | |
990 for (i = 0; i < 7; ) { | |
991 if (BITS_LEFT(length,gb) < 16) | |
992 break; | |
993 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; | |
994 | |
995 if (BITS_LEFT(length,gb) < 16) | |
996 break; | |
997 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); | |
2967 | 998 |
2914 | 999 for (k = 1; k <= run; k++) |
1000 quantized_coeffs[i + k] = (level + ((k * diff) / run)); | |
2967 | 1001 |
2914 | 1002 level += diff; |
1003 i += run; | |
1004 } | |
1005 } | |
1006 | |
1007 | |
1008 /** | |
1009 * Related to synthesis filter, process data from packet 10 | |
1010 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 | |
1011 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10 | |
1012 * | |
1013 * @param q context | |
1014 * @param gb bitreader context | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
1015 * @param length packet length in bits |
2914 | 1016 */ |
1017 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) | |
1018 { | |
1019 int sb, j, k, n, ch; | |
1020 | |
1021 for (ch = 0; ch < q->nb_channels; ch++) { | |
1022 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length); | |
1023 | |
1024 if (BITS_LEFT(length,gb) < 16) { | |
1025 memset(q->quantized_coeffs[ch][0], 0, 8); | |
1026 break; | |
1027 } | |
1028 } | |
1029 | |
1030 n = q->sub_sampling + 1; | |
1031 | |
1032 for (sb = 0; sb < n; sb++) | |
1033 for (ch = 0; ch < q->nb_channels; ch++) | |
1034 for (j = 0; j < 8; j++) { | |
1035 if (BITS_LEFT(length,gb) < 1) | |
1036 break; | |
1037 if (get_bits1(gb)) { | |
1038 for (k=0; k < 8; k++) { | |
1039 if (BITS_LEFT(length,gb) < 16) | |
1040 break; | |
1041 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); | |
1042 } | |
1043 } else { | |
1044 for (k=0; k < 8; k++) | |
1045 q->tone_level_idx_hi1[ch][sb][j][k] = 0; | |
1046 } | |
1047 } | |
1048 | |
1049 n = QDM2_SB_USED(q->sub_sampling) - 4; | |
1050 | |
1051 for (sb = 0; sb < n; sb++) | |
1052 for (ch = 0; ch < q->nb_channels; ch++) { | |
1053 if (BITS_LEFT(length,gb) < 16) | |
1054 break; | |
1055 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); | |
1056 if (sb > 19) | |
1057 q->tone_level_idx_hi2[ch][sb] -= 16; | |
1058 else | |
1059 for (j = 0; j < 8; j++) | |
1060 q->tone_level_idx_mid[ch][sb][j] = -16; | |
1061 } | |
1062 | |
1063 n = QDM2_SB_USED(q->sub_sampling) - 5; | |
1064 | |
1065 for (sb = 0; sb < n; sb++) | |
1066 for (ch = 0; ch < q->nb_channels; ch++) | |
1067 for (j = 0; j < 8; j++) { | |
1068 if (BITS_LEFT(length,gb) < 16) | |
1069 break; | |
1070 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; | |
1071 } | |
1072 } | |
1073 | |
1074 /** | |
1075 * Process subpacket 9, init quantized_coeffs with data from it | |
1076 * | |
1077 * @param q context | |
1078 * @param node pointer to node with packet | |
1079 */ | |
1080 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) | |
1081 { | |
1082 GetBitContext gb; | |
1083 int i, j, k, n, ch, run, level, diff; | |
1084 | |
2916 | 1085 init_get_bits(&gb, node->packet->data, node->packet->size*8); |
2914 | 1086 |
1087 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function | |
1088 | |
1089 for (i = 1; i < n; i++) | |
1090 for (ch=0; ch < q->nb_channels; ch++) { | |
1091 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); | |
1092 q->quantized_coeffs[ch][i][0] = level; | |
1093 | |
1094 for (j = 0; j < (8 - 1); ) { | |
1095 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; | |
1096 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); | |
1097 | |
1098 for (k = 1; k <= run; k++) | |
1099 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); | |
1100 | |
1101 level += diff; | |
1102 j += run; | |
1103 } | |
1104 } | |
1105 | |
1106 for (ch = 0; ch < q->nb_channels; ch++) | |
1107 for (i = 0; i < 8; i++) | |
1108 q->quantized_coeffs[ch][0][i] = 0; | |
1109 } | |
1110 | |
1111 | |
1112 /** | |
1113 * Process subpacket 10 if not null, else | |
1114 * | |
1115 * @param q context | |
1116 * @param node pointer to node with packet | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
1117 * @param length packet length in bits |
2914 | 1118 */ |
1119 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) | |
1120 { | |
1121 GetBitContext gb; | |
1122 | |
2916 | 1123 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
2914 | 1124 |
1125 if (length != 0) { | |
1126 init_tone_level_dequantization(q, &gb, length); | |
1127 fill_tone_level_array(q, 1); | |
1128 } else { | |
1129 fill_tone_level_array(q, 0); | |
1130 } | |
1131 } | |
1132 | |
1133 | |
1134 /** | |
1135 * Process subpacket 11 | |
1136 * | |
1137 * @param q context | |
1138 * @param node pointer to node with packet | |
1139 * @param length packet length in bit | |
1140 */ | |
1141 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length) | |
1142 { | |
1143 GetBitContext gb; | |
1144 | |
2916 | 1145 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
2914 | 1146 if (length >= 32) { |
1147 int c = get_bits (&gb, 13); | |
1148 | |
1149 if (c > 3) | |
1150 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, | |
1151 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select); | |
1152 } | |
1153 | |
1154 synthfilt_build_sb_samples(q, &gb, length, 0, 8); | |
1155 } | |
1156 | |
1157 | |
1158 /** | |
1159 * Process subpacket 12 | |
1160 * | |
1161 * @param q context | |
1162 * @param node pointer to node with packet | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
1163 * @param length packet length in bits |
2914 | 1164 */ |
1165 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) | |
1166 { | |
1167 GetBitContext gb; | |
1168 | |
2916 | 1169 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
2914 | 1170 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); |
1171 } | |
1172 | |
1173 /* | |
1174 * Process new subpackets for synthesis filter | |
1175 * | |
1176 * @param q context | |
1177 * @param list list with synthesis filter packets (list D) | |
1178 */ | |
1179 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) | |
1180 { | |
1181 QDM2SubPNode *nodes[4]; | |
1182 | |
1183 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); | |
1184 if (nodes[0] != NULL) | |
1185 process_subpacket_9(q, nodes[0]); | |
1186 | |
1187 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); | |
1188 if (nodes[1] != NULL) | |
1189 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3); | |
1190 else | |
1191 process_subpacket_10(q, NULL, 0); | |
1192 | |
1193 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); | |
1194 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) | |
1195 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3)); | |
1196 else | |
1197 process_subpacket_11(q, NULL, 0); | |
1198 | |
1199 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); | |
1200 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) | |
1201 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3)); | |
1202 else | |
1203 process_subpacket_12(q, NULL, 0); | |
1204 } | |
1205 | |
1206 | |
1207 /* | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
1208 * Decode superblock, fill packet lists. |
2914 | 1209 * |
1210 * @param q context | |
1211 */ | |
1212 static void qdm2_decode_super_block (QDM2Context *q) | |
1213 { | |
1214 GetBitContext gb; | |
1215 QDM2SubPacket header, *packet; | |
1216 int i, packet_bytes, sub_packet_size, sub_packets_D; | |
1217 unsigned int next_index = 0; | |
1218 | |
1219 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); | |
1220 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); | |
1221 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); | |
1222 | |
1223 q->sub_packets_B = 0; | |
1224 sub_packets_D = 0; | |
1225 | |
1226 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] | |
1227 | |
2916 | 1228 init_get_bits(&gb, q->compressed_data, q->compressed_size*8); |
2914 | 1229 qdm2_decode_sub_packet_header(&gb, &header); |
1230 | |
1231 if (header.type < 2 || header.type >= 8) { | |
1232 q->has_errors = 1; | |
1233 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); | |
1234 return; | |
1235 } | |
1236 | |
1237 q->superblocktype_2_3 = (header.type == 2 || header.type == 3); | |
1238 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); | |
1239 | |
2916 | 1240 init_get_bits(&gb, header.data, header.size*8); |
2914 | 1241 |
1242 if (header.type == 2 || header.type == 4 || header.type == 5) { | |
1243 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8); | |
1244 | |
1245 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); | |
1246 | |
1247 if (csum != 0) { | |
1248 q->has_errors = 1; | |
1249 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); | |
1250 return; | |
1251 } | |
1252 } | |
1253 | |
1254 q->sub_packet_list_B[0].packet = NULL; | |
1255 q->sub_packet_list_D[0].packet = NULL; | |
1256 | |
1257 for (i = 0; i < 6; i++) | |
1258 if (--q->fft_level_exp[i] < 0) | |
1259 q->fft_level_exp[i] = 0; | |
1260 | |
1261 for (i = 0; packet_bytes > 0; i++) { | |
1262 int j; | |
1263 | |
1264 q->sub_packet_list_A[i].next = NULL; | |
1265 | |
1266 if (i > 0) { | |
1267 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; | |
1268 | |
1269 /* seek to next block */ | |
2916 | 1270 init_get_bits(&gb, header.data, header.size*8); |
2914 | 1271 skip_bits(&gb, next_index*8); |
1272 | |
1273 if (next_index >= header.size) | |
1274 break; | |
1275 } | |
1276 | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
1277 /* decode subpacket */ |
2914 | 1278 packet = &q->sub_packets[i]; |
1279 qdm2_decode_sub_packet_header(&gb, packet); | |
1280 next_index = packet->size + get_bits_count(&gb) / 8; | |
1281 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; | |
1282 | |
1283 if (packet->type == 0) | |
1284 break; | |
1285 | |
1286 if (sub_packet_size > packet_bytes) { | |
1287 if (packet->type != 10 && packet->type != 11 && packet->type != 12) | |
1288 break; | |
1289 packet->size += packet_bytes - sub_packet_size; | |
1290 } | |
1291 | |
1292 packet_bytes -= sub_packet_size; | |
1293 | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
1294 /* add subpacket to 'all subpackets' list */ |
2914 | 1295 q->sub_packet_list_A[i].packet = packet; |
1296 | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
1297 /* add subpacket to related list */ |
2914 | 1298 if (packet->type == 8) { |
1299 SAMPLES_NEEDED_2("packet type 8"); | |
1300 return; | |
1301 } else if (packet->type >= 9 && packet->type <= 12) { | |
1302 /* packets for MPEG Audio like Synthesis Filter */ | |
1303 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); | |
1304 } else if (packet->type == 13) { | |
1305 for (j = 0; j < 6; j++) | |
1306 q->fft_level_exp[j] = get_bits(&gb, 6); | |
1307 } else if (packet->type == 14) { | |
1308 for (j = 0; j < 6; j++) | |
1309 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); | |
1310 } else if (packet->type == 15) { | |
1311 SAMPLES_NEEDED_2("packet type 15") | |
1312 return; | |
1313 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { | |
1314 /* packets for FFT */ | |
1315 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); | |
1316 } | |
1317 } // Packet bytes loop | |
1318 | |
1319 /* **************************************************************** */ | |
1320 if (q->sub_packet_list_D[0].packet != NULL) { | |
1321 process_synthesis_subpackets(q, q->sub_packet_list_D); | |
1322 q->do_synth_filter = 1; | |
1323 } else if (q->do_synth_filter) { | |
1324 process_subpacket_10(q, NULL, 0); | |
1325 process_subpacket_11(q, NULL, 0); | |
1326 process_subpacket_12(q, NULL, 0); | |
1327 } | |
1328 /* **************************************************************** */ | |
1329 } | |
1330 | |
1331 | |
1332 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, | |
1333 int offset, int duration, int channel, | |
1334 int exp, int phase) | |
1335 { | |
1336 if (q->fft_coefs_min_index[duration] < 0) | |
1337 q->fft_coefs_min_index[duration] = q->fft_coefs_index; | |
1338 | |
1339 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); | |
1340 q->fft_coefs[q->fft_coefs_index].channel = channel; | |
1341 q->fft_coefs[q->fft_coefs_index].offset = offset; | |
1342 q->fft_coefs[q->fft_coefs_index].exp = exp; | |
1343 q->fft_coefs[q->fft_coefs_index].phase = phase; | |
1344 q->fft_coefs_index++; | |
1345 } | |
1346 | |
1347 | |
1348 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) | |
1349 { | |
1350 int channel, stereo, phase, exp; | |
1351 int local_int_4, local_int_8, stereo_phase, local_int_10; | |
1352 int local_int_14, stereo_exp, local_int_20, local_int_28; | |
1353 int n, offset; | |
1354 | |
1355 local_int_4 = 0; | |
1356 local_int_28 = 0; | |
1357 local_int_20 = 2; | |
1358 local_int_8 = (4 - duration); | |
1359 local_int_10 = 1 << (q->group_order - duration - 1); | |
1360 offset = 1; | |
1361 | |
1362 while (1) { | |
1363 if (q->superblocktype_2_3) { | |
1364 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { | |
1365 offset = 1; | |
1366 if (n == 0) { | |
1367 local_int_4 += local_int_10; | |
1368 local_int_28 += (1 << local_int_8); | |
1369 } else { | |
1370 local_int_4 += 8*local_int_10; | |
1371 local_int_28 += (8 << local_int_8); | |
1372 } | |
1373 } | |
1374 offset += (n - 2); | |
1375 } else { | |
1376 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); | |
1377 while (offset >= (local_int_10 - 1)) { | |
1378 offset += (1 - (local_int_10 - 1)); | |
1379 local_int_4 += local_int_10; | |
1380 local_int_28 += (1 << local_int_8); | |
1381 } | |
1382 } | |
1383 | |
1384 if (local_int_4 >= q->group_size) | |
1385 return; | |
1386 | |
1387 local_int_14 = (offset >> local_int_8); | |
1388 | |
1389 if (q->nb_channels > 1) { | |
1390 channel = get_bits1(gb); | |
1391 stereo = get_bits1(gb); | |
1392 } else { | |
1393 channel = 0; | |
1394 stereo = 0; | |
1395 } | |
1396 | |
1397 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); | |
1398 exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; | |
1399 exp = (exp < 0) ? 0 : exp; | |
1400 | |
1401 phase = get_bits(gb, 3); | |
1402 stereo_exp = 0; | |
1403 stereo_phase = 0; | |
1404 | |
1405 if (stereo) { | |
1406 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); | |
1407 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); | |
1408 if (stereo_phase < 0) | |
1409 stereo_phase += 8; | |
1410 } | |
1411 | |
1412 if (q->frequency_range > (local_int_14 + 1)) { | |
1413 int sub_packet = (local_int_20 + local_int_28); | |
1414 | |
1415 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); | |
1416 if (stereo) | |
1417 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); | |
1418 } | |
1419 | |
1420 offset++; | |
1421 } | |
1422 } | |
1423 | |
1424 | |
1425 static void qdm2_decode_fft_packets (QDM2Context *q) | |
1426 { | |
1427 int i, j, min, max, value, type, unknown_flag; | |
1428 GetBitContext gb; | |
1429 | |
1430 if (q->sub_packet_list_B[0].packet == NULL) | |
1431 return; | |
1432 | |
1433 /* reset minimum indices for FFT coefficients */ | |
1434 q->fft_coefs_index = 0; | |
1435 for (i=0; i < 5; i++) | |
1436 q->fft_coefs_min_index[i] = -1; | |
1437 | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
1438 /* process subpackets ordered by type, largest type first */ |
2914 | 1439 for (i = 0, max = 256; i < q->sub_packets_B; i++) { |
1440 QDM2SubPacket *packet; | |
1441 | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
1442 /* find subpacket with largest type less than max */ |
2914 | 1443 for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) { |
1444 value = q->sub_packet_list_B[j].packet->type; | |
1445 if (value > min && value < max) { | |
1446 min = value; | |
1447 packet = q->sub_packet_list_B[j].packet; | |
1448 } | |
1449 } | |
1450 | |
1451 max = min; | |
1452 | |
1453 /* check for errors (?) */ | |
1454 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) | |
1455 return; | |
1456 | |
1457 /* decode FFT tones */ | |
2916 | 1458 init_get_bits (&gb, packet->data, packet->size*8); |
2914 | 1459 |
1460 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) | |
1461 unknown_flag = 1; | |
1462 else | |
1463 unknown_flag = 0; | |
1464 | |
1465 type = packet->type; | |
1466 | |
1467 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { | |
1468 int duration = q->sub_sampling + 5 - (type & 15); | |
1469 | |
1470 if (duration >= 0 && duration < 4) | |
1471 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); | |
1472 } else if (type == 31) { | |
3320 | 1473 for (j=0; j < 4; j++) |
1474 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); | |
2914 | 1475 } else if (type == 46) { |
3320 | 1476 for (j=0; j < 6; j++) |
1477 q->fft_level_exp[j] = get_bits(&gb, 6); | |
1478 for (j=0; j < 4; j++) | |
1479 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); | |
2914 | 1480 } |
1481 } // Loop on B packets | |
1482 | |
1483 /* calculate maximum indices for FFT coefficients */ | |
1484 for (i = 0, j = -1; i < 5; i++) | |
1485 if (q->fft_coefs_min_index[i] >= 0) { | |
1486 if (j >= 0) | |
1487 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; | |
1488 j = i; | |
1489 } | |
1490 if (j >= 0) | |
1491 q->fft_coefs_max_index[j] = q->fft_coefs_index; | |
1492 } | |
1493 | |
1494 | |
1495 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) | |
1496 { | |
1497 float level, f[6]; | |
1498 int i; | |
1499 QDM2Complex c; | |
1500 const double iscale = 2.0*M_PI / 512.0; | |
1501 | |
1502 tone->phase += tone->phase_shift; | |
1503 | |
1504 /* calculate current level (maximum amplitude) of tone */ | |
1505 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; | |
1506 c.im = level * sin(tone->phase*iscale); | |
1507 c.re = level * cos(tone->phase*iscale); | |
1508 | |
1509 /* generate FFT coefficients for tone */ | |
1510 if (tone->duration >= 3 || tone->cutoff >= 3) { | |
1511 tone->samples_im[0] += c.im; | |
1512 tone->samples_re[0] += c.re; | |
1513 tone->samples_im[1] -= c.im; | |
1514 tone->samples_re[1] -= c.re; | |
1515 } else { | |
1516 f[1] = -tone->table[4]; | |
1517 f[0] = tone->table[3] - tone->table[0]; | |
1518 f[2] = 1.0 - tone->table[2] - tone->table[3]; | |
1519 f[3] = tone->table[1] + tone->table[4] - 1.0; | |
1520 f[4] = tone->table[0] - tone->table[1]; | |
1521 f[5] = tone->table[2]; | |
1522 for (i = 0; i < 2; i++) { | |
1523 tone->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i]; | |
1524 tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); | |
1525 } | |
1526 for (i = 0; i < 4; i++) { | |
1527 tone->samples_re[i] += c.re * f[i+2]; | |
1528 tone->samples_im[i] += c.im * f[i+2]; | |
1529 } | |
1530 } | |
1531 | |
1532 /* copy the tone if it has not yet died out */ | |
1533 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { | |
1534 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); | |
1535 q->fft_tone_end = (q->fft_tone_end + 1) % 1000; | |
1536 } | |
1537 } | |
1538 | |
1539 | |
1540 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) | |
1541 { | |
1542 int i, j, ch; | |
1543 const double iscale = 0.25 * M_PI; | |
1544 | |
1545 for (ch = 0; ch < q->channels; ch++) { | |
1546 memset(q->fft.samples_im[ch], 0, q->fft_size * sizeof(float)); | |
1547 memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float)); | |
1548 } | |
1549 | |
1550 | |
1551 /* apply FFT tones with duration 4 (1 FFT period) */ | |
1552 if (q->fft_coefs_min_index[4] >= 0) | |
1553 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { | |
1554 float level; | |
1555 QDM2Complex c; | |
1556 | |
1557 if (q->fft_coefs[i].sub_packet != sub_packet) | |
1558 break; | |
1559 | |
1560 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; | |
1561 level = (q->fft_coefs[i].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[i].exp & 63]; | |
1562 | |
1563 c.re = level * cos(q->fft_coefs[i].phase * iscale); | |
1564 c.im = level * sin(q->fft_coefs[i].phase * iscale); | |
1565 q->fft.samples_re[ch][q->fft_coefs[i].offset + 0] += c.re; | |
1566 q->fft.samples_im[ch][q->fft_coefs[i].offset + 0] += c.im; | |
1567 q->fft.samples_re[ch][q->fft_coefs[i].offset + 1] -= c.re; | |
1568 q->fft.samples_im[ch][q->fft_coefs[i].offset + 1] -= c.im; | |
1569 } | |
1570 | |
1571 /* generate existing FFT tones */ | |
1572 for (i = q->fft_tone_end; i != q->fft_tone_start; ) { | |
1573 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); | |
1574 q->fft_tone_start = (q->fft_tone_start + 1) % 1000; | |
1575 } | |
1576 | |
1577 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ | |
1578 for (i = 0; i < 4; i++) | |
1579 if (q->fft_coefs_min_index[i] >= 0) { | |
1580 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { | |
1581 int offset, four_i; | |
1582 FFTTone tone; | |
1583 | |
1584 if (q->fft_coefs[j].sub_packet != sub_packet) | |
1585 break; | |
1586 | |
1587 four_i = (4 - i); | |
1588 offset = q->fft_coefs[j].offset >> four_i; | |
1589 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; | |
1590 | |
1591 if (offset < q->frequency_range) { | |
1592 if (offset < 2) | |
1593 tone.cutoff = offset; | |
1594 else | |
1595 tone.cutoff = (offset >= 60) ? 3 : 2; | |
1596 | |
1597 tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63]; | |
1598 tone.samples_im = &q->fft.samples_im[ch][offset]; | |
1599 tone.samples_re = &q->fft.samples_re[ch][offset]; | |
1600 tone.table = (float*)fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; | |
1601 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; | |
1602 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); | |
1603 tone.duration = i; | |
1604 tone.time_index = 0; | |
1605 | |
1606 qdm2_fft_generate_tone(q, &tone); | |
1607 } | |
1608 } | |
1609 q->fft_coefs_min_index[i] = j; | |
1610 } | |
1611 } | |
1612 | |
1613 | |
1614 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) | |
1615 { | |
1616 const int n = 1 << (q->fft_order - 1); | |
1617 const int n2 = n >> 1; | |
1618 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f; | |
1619 float c, s, f0, f1, f2, f3; | |
1620 int i, j; | |
1621 | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
1622 /* prerotation (or something like that) */ |
2914 | 1623 for (i=1; i < n2; i++) { |
1624 j = (n - i); | |
1625 c = q->exptab[i].re; | |
1626 s = -q->exptab[i].im; | |
1627 f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain; | |
1628 f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain; | |
1629 f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain; | |
1630 f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain; | |
1631 q->fft.complex[i].re = s * f0 - c * f1 + f2; | |
1632 q->fft.complex[i].im = c * f0 + s * f1 + f3; | |
1633 q->fft.complex[j].re = -s * f0 + c * f1 + f2; | |
1634 q->fft.complex[j].im = c * f0 + s * f1 - f3; | |
1635 } | |
1636 | |
1637 q->fft.complex[ 0].re = q->fft.samples_re[channel][ 0] * gain * 2.0; | |
1638 q->fft.complex[ 0].im = q->fft.samples_re[channel][ 0] * gain * 2.0; | |
1639 q->fft.complex[n2].re = q->fft.samples_re[channel][n2] * gain * 2.0; | |
1640 q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0; | |
1641 | |
1642 ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex); | |
1643 ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex); | |
1644 /* add samples to output buffer */ | |
1645 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) | |
1646 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex)[i]; | |
1647 } | |
1648 | |
1649 | |
1650 /** | |
1651 * @param q context | |
1652 * @param index subpacket number | |
1653 */ | |
1654 static void qdm2_synthesis_filter (QDM2Context *q, int index) | |
1655 { | |
1656 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; | |
1657 int i, k, ch, sb_used, sub_sampling, dither_state = 0; | |
1658 | |
1659 /* copy sb_samples */ | |
1660 sb_used = QDM2_SB_USED(q->sub_sampling); | |
1661 | |
1662 for (ch = 0; ch < q->channels; ch++) | |
1663 for (i = 0; i < 8; i++) | |
1664 for (k=sb_used; k < SBLIMIT; k++) | |
1665 q->sb_samples[ch][(8 * index) + i][k] = 0; | |
1666 | |
1667 for (ch = 0; ch < q->nb_channels; ch++) { | |
1668 OUT_INT *samples_ptr = samples + ch; | |
1669 | |
1670 for (i = 0; i < 8; i++) { | |
1671 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]), | |
1672 mpa_window, &dither_state, | |
1673 samples_ptr, q->nb_channels, | |
1674 q->sb_samples[ch][(8 * index) + i]); | |
1675 samples_ptr += 32 * q->nb_channels; | |
1676 } | |
1677 } | |
1678 | |
1679 /* add samples to output buffer */ | |
1680 sub_sampling = (4 >> q->sub_sampling); | |
1681 | |
1682 for (ch = 0; ch < q->channels; ch++) | |
1683 for (i = 0; i < q->frame_size; i++) | |
1684 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16)); | |
1685 } | |
1686 | |
1687 | |
1688 /** | |
1689 * Init static data (does not depend on specific file) | |
1690 * | |
1691 * @param q context | |
1692 */ | |
3076 | 1693 static void qdm2_init(QDM2Context *q) { |
2914 | 1694 static int inited = 0; |
1695 | |
1696 if (inited != 0) | |
1697 return; | |
1698 inited = 1; | |
1699 | |
1700 qdm2_init_vlc(); | |
1701 ff_mpa_synth_init(mpa_window); | |
1702 softclip_table_init(); | |
1703 rnd_table_init(); | |
1704 init_noise_samples(); | |
1705 | |
1706 av_log(NULL, AV_LOG_DEBUG, "init done\n"); | |
1707 } | |
1708 | |
1709 | |
1710 #if 0 | |
1711 static void dump_context(QDM2Context *q) | |
1712 { | |
1713 int i; | |
1714 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b); | |
1715 PRINT("compressed_data",q->compressed_data); | |
1716 PRINT("compressed_size",q->compressed_size); | |
1717 PRINT("frame_size",q->frame_size); | |
1718 PRINT("checksum_size",q->checksum_size); | |
1719 PRINT("channels",q->channels); | |
1720 PRINT("nb_channels",q->nb_channels); | |
1721 PRINT("fft_frame_size",q->fft_frame_size); | |
1722 PRINT("fft_size",q->fft_size); | |
1723 PRINT("sub_sampling",q->sub_sampling); | |
1724 PRINT("fft_order",q->fft_order); | |
1725 PRINT("group_order",q->group_order); | |
1726 PRINT("group_size",q->group_size); | |
1727 PRINT("sub_packet",q->sub_packet); | |
1728 PRINT("frequency_range",q->frequency_range); | |
1729 PRINT("has_errors",q->has_errors); | |
1730 PRINT("fft_tone_end",q->fft_tone_end); | |
1731 PRINT("fft_tone_start",q->fft_tone_start); | |
1732 PRINT("fft_coefs_index",q->fft_coefs_index); | |
1733 PRINT("coeff_per_sb_select",q->coeff_per_sb_select); | |
1734 PRINT("cm_table_select",q->cm_table_select); | |
1735 PRINT("noise_idx",q->noise_idx); | |
1736 | |
1737 for (i = q->fft_tone_start; i < q->fft_tone_end; i++) | |
1738 { | |
1739 FFTTone *t = &q->fft_tones[i]; | |
2967 | 1740 |
2914 | 1741 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i); |
1742 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level); | |
1743 // PRINT(" level", t->level); | |
1744 PRINT(" phase", t->phase); | |
1745 PRINT(" phase_shift", t->phase_shift); | |
1746 PRINT(" duration", t->duration); | |
1747 PRINT(" samples_im", t->samples_im); | |
1748 PRINT(" samples_re", t->samples_re); | |
1749 PRINT(" table", t->table); | |
1750 } | |
1751 | |
1752 } | |
1753 #endif | |
1754 | |
1755 | |
1756 /** | |
1757 * Init parameters from codec extradata | |
1758 */ | |
1759 static int qdm2_decode_init(AVCodecContext *avctx) | |
1760 { | |
1761 QDM2Context *s = avctx->priv_data; | |
1762 uint8_t *extradata; | |
1763 int extradata_size; | |
1764 int tmp_val, tmp, size; | |
1765 int i; | |
1766 float alpha; | |
2967 | 1767 |
2914 | 1768 /* extradata parsing |
2967 | 1769 |
2914 | 1770 Structure: |
1771 wave { | |
1772 frma (QDM2) | |
1773 QDCA | |
1774 QDCP | |
1775 } | |
2967 | 1776 |
2914 | 1777 32 size (including this field) |
1778 32 tag (=frma) | |
1779 32 type (=QDM2 or QDMC) | |
2967 | 1780 |
2914 | 1781 32 size (including this field, in bytes) |
1782 32 tag (=QDCA) // maybe mandatory parameters | |
1783 32 unknown (=1) | |
1784 32 channels (=2) | |
1785 32 samplerate (=44100) | |
1786 32 bitrate (=96000) | |
1787 32 block size (=4096) | |
1788 32 frame size (=256) (for one channel) | |
1789 32 packet size (=1300) | |
2967 | 1790 |
2914 | 1791 32 size (including this field, in bytes) |
1792 32 tag (=QDCP) // maybe some tuneable parameters | |
1793 32 float1 (=1.0) | |
1794 32 zero ? | |
1795 32 float2 (=1.0) | |
1796 32 float3 (=1.0) | |
1797 32 unknown (27) | |
1798 32 unknown (8) | |
1799 32 zero ? | |
1800 */ | |
1801 | |
1802 if (!avctx->extradata || (avctx->extradata_size < 48)) { | |
1803 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); | |
1804 return -1; | |
1805 } | |
1806 | |
1807 extradata = avctx->extradata; | |
1808 extradata_size = avctx->extradata_size; | |
1809 | |
1810 while (extradata_size > 7) { | |
1811 if (!memcmp(extradata, "frmaQDM", 7)) | |
1812 break; | |
1813 extradata++; | |
1814 extradata_size--; | |
1815 } | |
1816 | |
1817 if (extradata_size < 12) { | |
1818 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", | |
1819 extradata_size); | |
1820 return -1; | |
1821 } | |
1822 | |
1823 if (memcmp(extradata, "frmaQDM", 7)) { | |
1824 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); | |
1825 return -1; | |
1826 } | |
1827 | |
1828 if (extradata[7] == 'C') { | |
1829 // s->is_qdmc = 1; | |
1830 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); | |
1831 return -1; | |
1832 } | |
1833 | |
1834 extradata += 8; | |
1835 extradata_size -= 8; | |
1836 | |
1837 size = BE_32(extradata); | |
1838 | |
1839 if(size > extradata_size){ | |
1840 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", | |
1841 extradata_size, size); | |
1842 return -1; | |
1843 } | |
1844 | |
1845 extradata += 4; | |
1846 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); | |
1847 if (BE_32(extradata) != MKBETAG('Q','D','C','A')) { | |
1848 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); | |
1849 return -1; | |
1850 } | |
1851 | |
1852 extradata += 8; | |
1853 | |
1854 avctx->channels = s->nb_channels = s->channels = BE_32(extradata); | |
1855 extradata += 4; | |
1856 | |
1857 avctx->sample_rate = BE_32(extradata); | |
1858 extradata += 4; | |
1859 | |
1860 avctx->bit_rate = BE_32(extradata); | |
1861 extradata += 4; | |
1862 | |
1863 s->group_size = BE_32(extradata); | |
1864 extradata += 4; | |
1865 | |
1866 s->fft_size = BE_32(extradata); | |
1867 extradata += 4; | |
1868 | |
1869 s->checksum_size = BE_32(extradata); | |
1870 extradata += 4; | |
1871 | |
1872 s->fft_order = av_log2(s->fft_size) + 1; | |
1873 s->fft_frame_size = 2 * s->fft_size; // complex has two floats | |
1874 | |
1875 // something like max decodable tones | |
1876 s->group_order = av_log2(s->group_size) + 1; | |
1877 s->frame_size = s->group_size / 16; // 16 iterations per super block | |
1878 | |
2954 | 1879 s->sub_sampling = s->fft_order - 7; |
2914 | 1880 s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); |
2967 | 1881 |
2914 | 1882 switch ((s->sub_sampling * 2 + s->channels - 1)) { |
1883 case 0: tmp = 40; break; | |
1884 case 1: tmp = 48; break; | |
1885 case 2: tmp = 56; break; | |
1886 case 3: tmp = 72; break; | |
1887 case 4: tmp = 80; break; | |
1888 case 5: tmp = 100;break; | |
1889 default: tmp=s->sub_sampling; break; | |
1890 } | |
1891 tmp_val = 0; | |
1892 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; | |
1893 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; | |
1894 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; | |
1895 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; | |
1896 s->cm_table_select = tmp_val; | |
1897 | |
1898 if (s->sub_sampling == 0) | |
2954 | 1899 tmp = 7999; |
2914 | 1900 else |
1901 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; | |
1902 /* | |
2954 | 1903 0: 7999 -> 0 |
2914 | 1904 1: 20000 -> 2 |
1905 2: 28000 -> 2 | |
1906 */ | |
1907 if (tmp < 8000) | |
1908 s->coeff_per_sb_select = 0; | |
1909 else if (tmp <= 16000) | |
1910 s->coeff_per_sb_select = 1; | |
1911 else | |
1912 s->coeff_per_sb_select = 2; | |
1913 | |
2954 | 1914 // Fail on unknown fft order, if it's > 9 it can overflow s->exptab[] |
1915 if ((s->fft_order < 7) || (s->fft_order > 9)) { | |
2914 | 1916 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); |
2954 | 1917 return -1; |
1918 } | |
2914 | 1919 |
1920 ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1); | |
1921 | |
1922 for (i = 1; i < (1 << (s->fft_order - 2)); i++) { | |
1923 alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1)); | |
1924 s->exptab[i].re = cos(alpha); | |
1925 s->exptab[i].im = sin(alpha); | |
1926 } | |
1927 | |
1928 qdm2_init(s); | |
2967 | 1929 |
2914 | 1930 // dump_context(s); |
1931 return 0; | |
1932 } | |
1933 | |
1934 | |
1935 static int qdm2_decode_close(AVCodecContext *avctx) | |
1936 { | |
1937 QDM2Context *s = avctx->priv_data; | |
1938 | |
1939 ff_fft_end(&s->fft_ctx); | |
2967 | 1940 |
2914 | 1941 return 0; |
1942 } | |
1943 | |
1944 | |
3076 | 1945 static void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out) |
2914 | 1946 { |
1947 int ch, i; | |
1948 const int frame_size = (q->frame_size * q->channels); | |
2967 | 1949 |
2914 | 1950 /* select input buffer */ |
1951 q->compressed_data = in; | |
1952 q->compressed_size = q->checksum_size; | |
1953 | |
1954 // dump_context(q); | |
1955 | |
1956 /* copy old block, clear new block of output samples */ | |
1957 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); | |
1958 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); | |
1959 | |
1960 /* decode block of QDM2 compressed data */ | |
1961 if (q->sub_packet == 0) { | |
1962 q->has_errors = 0; // zero it for a new super block | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
1963 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); |
2914 | 1964 qdm2_decode_super_block(q); |
1965 } | |
1966 | |
3043
583020ce54a8
Fix a bunch of spelling/grammar mistakes in doxygen comments and output.
diego
parents:
3036
diff
changeset
|
1967 /* parse subpackets */ |
2914 | 1968 if (!q->has_errors) { |
1969 if (q->sub_packet == 2) | |
1970 qdm2_decode_fft_packets(q); | |
1971 | |
1972 qdm2_fft_tone_synthesizer(q, q->sub_packet); | |
1973 } | |
1974 | |
1975 /* sound synthesis stage 1 (FFT) */ | |
1976 for (ch = 0; ch < q->channels; ch++) { | |
1977 qdm2_calculate_fft(q, ch, q->sub_packet); | |
1978 | |
1979 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { | |
1980 SAMPLES_NEEDED_2("has errors, and C list is not empty") | |
1981 return; | |
1982 } | |
1983 } | |
1984 | |
1985 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ | |
1986 if (!q->has_errors && q->do_synth_filter) | |
1987 qdm2_synthesis_filter(q, q->sub_packet); | |
1988 | |
1989 q->sub_packet = (q->sub_packet + 1) % 16; | |
1990 | |
1991 /* clip and convert output float[] to 16bit signed samples */ | |
1992 for (i = 0; i < frame_size; i++) { | |
1993 int value = (int)q->output_buffer[i]; | |
1994 | |
1995 if (value > SOFTCLIP_THRESHOLD) | |
1996 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; | |
1997 else if (value < -SOFTCLIP_THRESHOLD) | |
1998 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; | |
1999 | |
2000 out[i] = value; | |
2001 } | |
2002 } | |
2003 | |
2004 | |
2005 static int qdm2_decode_frame(AVCodecContext *avctx, | |
2006 void *data, int *data_size, | |
2007 uint8_t *buf, int buf_size) | |
2008 { | |
2009 QDM2Context *s = avctx->priv_data; | |
2010 | |
3158 | 2011 if(!buf) |
2914 | 2012 return 0; |
3158 | 2013 if(buf_size < s->checksum_size) |
2014 return -1; | |
2914 | 2015 |
2016 *data_size = s->channels * s->frame_size * sizeof(int16_t); | |
2017 | |
2018 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", | |
2019 buf_size, buf, s->checksum_size, data, *data_size); | |
2020 | |
2021 qdm2_decode(s, buf, data); | |
2022 | |
2023 // reading only when next superblock found | |
2024 if (s->sub_packet == 0) { | |
2025 return s->checksum_size; | |
2026 } | |
2027 | |
2028 return 0; | |
2029 } | |
2030 | |
2031 AVCodec qdm2_decoder = | |
2032 { | |
2033 .name = "qdm2", | |
2034 .type = CODEC_TYPE_AUDIO, | |
2035 .id = CODEC_ID_QDM2, | |
2036 .priv_data_size = sizeof(QDM2Context), | |
2037 .init = qdm2_decode_init, | |
2038 .close = qdm2_decode_close, | |
2039 .decode = qdm2_decode_frame, | |
2040 }; |