Mercurial > libavcodec.hg
annotate qdm2.c @ 4576:ad02fe65f45a libavcodec
Move the av_realloc_static() declaration out of avcodec.h into bitstream.c.
author | takis |
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date | Thu, 22 Feb 2007 20:21:33 +0000 |
parents | 05e932ddaaa9 |
children | 4d10df33e823 |
rev | line source |
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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 * | |
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8 * This file is part of FFmpeg. |
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9 * |
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10 * FFmpeg is free software; you can redistribute it and/or |
2914 | 11 * modify it under the terms of the GNU Lesser General Public |
12 * License as published by the Free Software Foundation; either | |
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13 * version 2.1 of the License, or (at your option) any later version. |
2914 | 14 * |
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15 * FFmpeg is distributed in the hope that it will be useful, |
2914 | 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 | |
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21 * License along with FFmpeg; if not, write to the Free Software |
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22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
2914 | 23 * |
24 */ | |
25 | |
26 /** | |
27 * @file qdm2.c | |
28 * QDM2 decoder | |
29 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni | |
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30 * The decoder is not perfect yet, there are still some distortions |
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31 * especially on files encoded with 16 or 8 subbands. |
2914 | 32 */ |
33 | |
34 #include <math.h> | |
35 #include <stddef.h> | |
36 #include <stdio.h> | |
37 | |
38 #define ALT_BITSTREAM_READER_LE | |
39 #include "avcodec.h" | |
40 #include "bitstream.h" | |
41 #include "dsputil.h" | |
42 | |
43 #ifdef CONFIG_MPEGAUDIO_HP | |
44 #define USE_HIGHPRECISION | |
45 #endif | |
46 | |
47 #include "mpegaudio.h" | |
48 | |
49 #include "qdm2data.h" | |
50 | |
51 #undef NDEBUG | |
52 #include <assert.h> | |
53 | |
54 | |
55 #define SOFTCLIP_THRESHOLD 27600 | |
56 #define HARDCLIP_THRESHOLD 35716 | |
57 | |
58 | |
59 #define QDM2_LIST_ADD(list, size, packet) \ | |
60 do { \ | |
61 if (size > 0) { \ | |
62 list[size - 1].next = &list[size]; \ | |
63 } \ | |
64 list[size].packet = packet; \ | |
65 list[size].next = NULL; \ | |
66 size++; \ | |
67 } while(0) | |
68 | |
69 // Result is 8, 16 or 30 | |
70 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling)) | |
71 | |
72 #define FIX_NOISE_IDX(noise_idx) \ | |
73 if ((noise_idx) >= 3840) \ | |
74 (noise_idx) -= 3840; \ | |
75 | |
76 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)]) | |
77 | |
78 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb))) | |
79 | |
80 #define SAMPLES_NEEDED \ | |
81 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); | |
82 | |
83 #define SAMPLES_NEEDED_2(why) \ | |
84 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); | |
85 | |
86 | |
87 typedef int8_t sb_int8_array[2][30][64]; | |
88 | |
89 /** | |
90 * Subpacket | |
91 */ | |
92 typedef struct { | |
93 int type; ///< subpacket type | |
94 unsigned int size; ///< subpacket size | |
95 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) | |
96 } QDM2SubPacket; | |
97 | |
98 /** | |
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99 * A node in the subpacket list |
2914 | 100 */ |
101 typedef struct _QDM2SubPNode { | |
102 QDM2SubPacket *packet; ///< packet | |
103 struct _QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node | |
104 } QDM2SubPNode; | |
105 | |
106 typedef struct { | |
107 float level; | |
108 float *samples_im; | |
109 float *samples_re; | |
110 float *table; | |
111 int phase; | |
112 int phase_shift; | |
113 int duration; | |
114 short time_index; | |
115 short cutoff; | |
116 } FFTTone; | |
117 | |
118 typedef struct { | |
119 int16_t sub_packet; | |
120 uint8_t channel; | |
121 int16_t offset; | |
122 int16_t exp; | |
123 uint8_t phase; | |
124 } FFTCoefficient; | |
125 | |
126 typedef struct { | |
127 float re; | |
128 float im; | |
129 } QDM2Complex; | |
130 | |
131 typedef struct { | |
132 QDM2Complex complex[256 + 1] __attribute__((aligned(16))); | |
133 float samples_im[MPA_MAX_CHANNELS][256]; | |
134 float samples_re[MPA_MAX_CHANNELS][256]; | |
135 } QDM2FFT; | |
136 | |
137 /** | |
138 * QDM2 decoder context | |
139 */ | |
140 typedef struct { | |
141 /// Parameters from codec header, do not change during playback | |
142 int nb_channels; ///< number of channels | |
143 int channels; ///< number of channels | |
144 int group_size; ///< size of frame group (16 frames per group) | |
145 int fft_size; ///< size of FFT, in complex numbers | |
146 int checksum_size; ///< size of data block, used also for checksum | |
147 | |
148 /// Parameters built from header parameters, do not change during playback | |
149 int group_order; ///< order of frame group | |
150 int fft_order; ///< order of FFT (actually fftorder+1) | |
151 int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im) | |
152 int frame_size; ///< size of data frame | |
153 int frequency_range; | |
154 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ | |
155 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 | |
156 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) | |
157 | |
158 /// Packets and packet lists | |
159 QDM2SubPacket sub_packets[16]; ///< the packets themselves | |
160 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets | |
161 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list | |
162 int sub_packets_B; ///< number of packets on 'B' list | |
163 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? | |
164 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets | |
165 | |
166 /// FFT and tones | |
167 FFTTone fft_tones[1000]; | |
168 int fft_tone_start; | |
169 int fft_tone_end; | |
170 FFTCoefficient fft_coefs[1000]; | |
171 int fft_coefs_index; | |
172 int fft_coefs_min_index[5]; | |
173 int fft_coefs_max_index[5]; | |
174 int fft_level_exp[6]; | |
175 FFTContext fft_ctx; | |
176 FFTComplex exptab[128]; | |
177 QDM2FFT fft; | |
178 | |
179 /// I/O data | |
180 uint8_t *compressed_data; | |
181 int compressed_size; | |
182 float output_buffer[1024]; | |
183 | |
184 /// Synthesis filter | |
185 MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16))); | |
186 int synth_buf_offset[MPA_MAX_CHANNELS]; | |
187 int32_t sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT] __attribute__((aligned(16))); | |
188 | |
189 /// Mixed temporary data used in decoding | |
190 float tone_level[MPA_MAX_CHANNELS][30][64]; | |
191 int8_t coding_method[MPA_MAX_CHANNELS][30][64]; | |
192 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; | |
193 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; | |
194 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; | |
195 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; | |
196 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; | |
197 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; | |
198 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; | |
199 | |
200 // Flags | |
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201 int has_errors; ///< packet has errors |
2914 | 202 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type |
203 int do_synth_filter; ///< used to perform or skip synthesis filter | |
204 | |
205 int sub_packet; | |
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206 int noise_idx; ///< index for dithering noise table |
2914 | 207 } QDM2Context; |
208 | |
209 | |
210 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE]; | |
211 | |
212 static VLC vlc_tab_level; | |
213 static VLC vlc_tab_diff; | |
214 static VLC vlc_tab_run; | |
215 static VLC fft_level_exp_alt_vlc; | |
216 static VLC fft_level_exp_vlc; | |
217 static VLC fft_stereo_exp_vlc; | |
218 static VLC fft_stereo_phase_vlc; | |
219 static VLC vlc_tab_tone_level_idx_hi1; | |
220 static VLC vlc_tab_tone_level_idx_mid; | |
221 static VLC vlc_tab_tone_level_idx_hi2; | |
222 static VLC vlc_tab_type30; | |
223 static VLC vlc_tab_type34; | |
224 static VLC vlc_tab_fft_tone_offset[5]; | |
225 | |
226 static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1]; | |
227 static float noise_table[4096]; | |
228 static uint8_t random_dequant_index[256][5]; | |
229 static uint8_t random_dequant_type24[128][3]; | |
230 static float noise_samples[128]; | |
231 | |
232 static MPA_INT mpa_window[512] __attribute__((aligned(16))); | |
233 | |
234 | |
3076 | 235 static void softclip_table_init(void) { |
2914 | 236 int i; |
237 double dfl = SOFTCLIP_THRESHOLD - 32767; | |
238 float delta = 1.0 / -dfl; | |
239 for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++) | |
240 softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF); | |
241 } | |
242 | |
243 | |
244 // random generated table | |
3076 | 245 static void rnd_table_init(void) { |
2914 | 246 int i,j; |
247 uint32_t ldw,hdw; | |
248 uint64_t tmp64_1; | |
249 uint64_t random_seed = 0; | |
250 float delta = 1.0 / 16384.0; | |
251 for(i = 0; i < 4096 ;i++) { | |
252 random_seed = random_seed * 214013 + 2531011; | |
253 noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3; | |
254 } | |
255 | |
256 for (i = 0; i < 256 ;i++) { | |
257 random_seed = 81; | |
258 ldw = i; | |
259 for (j = 0; j < 5 ;j++) { | |
260 random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF); | |
261 ldw = (uint32_t)ldw % (uint32_t)random_seed; | |
262 tmp64_1 = (random_seed * 0x55555556); | |
263 hdw = (uint32_t)(tmp64_1 >> 32); | |
264 random_seed = (uint64_t)(hdw + (ldw >> 31)); | |
265 } | |
266 } | |
267 for (i = 0; i < 128 ;i++) { | |
268 random_seed = 25; | |
269 ldw = i; | |
270 for (j = 0; j < 3 ;j++) { | |
271 random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF); | |
272 ldw = (uint32_t)ldw % (uint32_t)random_seed; | |
273 tmp64_1 = (random_seed * 0x66666667); | |
274 hdw = (uint32_t)(tmp64_1 >> 33); | |
275 random_seed = hdw + (ldw >> 31); | |
276 } | |
277 } | |
278 } | |
279 | |
280 | |
3076 | 281 static void init_noise_samples(void) { |
2914 | 282 int i; |
283 int random_seed = 0; | |
284 float delta = 1.0 / 16384.0; | |
285 for (i = 0; i < 128;i++) { | |
286 random_seed = random_seed * 214013 + 2531011; | |
287 noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0); | |
288 } | |
289 } | |
290 | |
291 | |
3076 | 292 static void qdm2_init_vlc(void) |
2914 | 293 { |
294 init_vlc (&vlc_tab_level, 8, 24, | |
295 vlc_tab_level_huffbits, 1, 1, | |
296 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
297 | |
298 init_vlc (&vlc_tab_diff, 8, 37, | |
299 vlc_tab_diff_huffbits, 1, 1, | |
300 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
301 | |
302 init_vlc (&vlc_tab_run, 5, 6, | |
303 vlc_tab_run_huffbits, 1, 1, | |
304 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
305 | |
306 init_vlc (&fft_level_exp_alt_vlc, 8, 28, | |
307 fft_level_exp_alt_huffbits, 1, 1, | |
308 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
309 | |
310 init_vlc (&fft_level_exp_vlc, 8, 20, | |
311 fft_level_exp_huffbits, 1, 1, | |
312 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
313 | |
314 init_vlc (&fft_stereo_exp_vlc, 6, 7, | |
315 fft_stereo_exp_huffbits, 1, 1, | |
316 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
317 | |
318 init_vlc (&fft_stereo_phase_vlc, 6, 9, | |
319 fft_stereo_phase_huffbits, 1, 1, | |
320 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
321 | |
322 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, | |
323 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, | |
324 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
325 | |
326 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, | |
327 vlc_tab_tone_level_idx_mid_huffbits, 1, 1, | |
328 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
329 | |
330 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, | |
331 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, | |
332 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
333 | |
334 init_vlc (&vlc_tab_type30, 6, 9, | |
335 vlc_tab_type30_huffbits, 1, 1, | |
336 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
337 | |
338 init_vlc (&vlc_tab_type34, 5, 10, | |
339 vlc_tab_type34_huffbits, 1, 1, | |
340 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
341 | |
342 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, | |
343 vlc_tab_fft_tone_offset_0_huffbits, 1, 1, | |
344 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
345 | |
346 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, | |
347 vlc_tab_fft_tone_offset_1_huffbits, 1, 1, | |
348 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
349 | |
350 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, | |
351 vlc_tab_fft_tone_offset_2_huffbits, 1, 1, | |
352 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
353 | |
354 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, | |
355 vlc_tab_fft_tone_offset_3_huffbits, 1, 1, | |
356 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
357 | |
358 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, | |
359 vlc_tab_fft_tone_offset_4_huffbits, 1, 1, | |
360 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE); | |
361 } | |
362 | |
363 | |
364 /* for floating point to fixed point conversion */ | |
365 static float f2i_scale = (float) (1 << (FRAC_BITS - 15)); | |
366 | |
367 | |
368 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) | |
369 { | |
370 int value; | |
371 | |
372 value = get_vlc2(gb, vlc->table, vlc->bits, depth); | |
373 | |
374 /* stage-2, 3 bits exponent escape sequence */ | |
375 if (value-- == 0) | |
376 value = get_bits (gb, get_bits (gb, 3) + 1); | |
377 | |
378 /* stage-3, optional */ | |
379 if (flag) { | |
380 int tmp = vlc_stage3_values[value]; | |
381 | |
382 if ((value & ~3) > 0) | |
383 tmp += get_bits (gb, (value >> 2)); | |
384 value = tmp; | |
385 } | |
386 | |
387 return value; | |
388 } | |
389 | |
390 | |
391 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) | |
392 { | |
393 int value = qdm2_get_vlc (gb, vlc, 0, depth); | |
394 | |
395 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); | |
396 } | |
397 | |
398 | |
399 /** | |
400 * QDM2 checksum | |
401 * | |
402 * @param data pointer to data to be checksum'ed | |
403 * @param length data length | |
404 * @param value checksum value | |
405 * | |
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406 * @return 0 if checksum is OK |
2914 | 407 */ |
408 static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) { | |
409 int i; | |
410 | |
411 for (i=0; i < length; i++) | |
412 value -= data[i]; | |
413 | |
414 return (uint16_t)(value & 0xffff); | |
415 } | |
416 | |
417 | |
418 /** | |
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419 * Fills a QDM2SubPacket structure with packet type, size, and data pointer. |
2914 | 420 * |
421 * @param gb bitreader context | |
422 * @param sub_packet packet under analysis | |
423 */ | |
424 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) | |
425 { | |
426 sub_packet->type = get_bits (gb, 8); | |
427 | |
428 if (sub_packet->type == 0) { | |
429 sub_packet->size = 0; | |
430 sub_packet->data = NULL; | |
431 } else { | |
432 sub_packet->size = get_bits (gb, 8); | |
433 | |
434 if (sub_packet->type & 0x80) { | |
435 sub_packet->size <<= 8; | |
436 sub_packet->size |= get_bits (gb, 8); | |
437 sub_packet->type &= 0x7f; | |
438 } | |
439 | |
440 if (sub_packet->type == 0x7f) | |
441 sub_packet->type |= (get_bits (gb, 8) << 8); | |
442 | |
443 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data | |
444 } | |
445 | |
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446 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", |
2914 | 447 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); |
448 } | |
449 | |
450 | |
451 /** | |
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452 * Return node pointer to first packet of requested type in list. |
2914 | 453 * |
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454 * @param list list of subpackets to be scanned |
2914 | 455 * @param type type of searched subpacket |
456 * @return node pointer for subpacket if found, else NULL | |
457 */ | |
458 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) | |
459 { | |
460 while (list != NULL && list->packet != NULL) { | |
461 if (list->packet->type == type) | |
462 return list; | |
463 list = list->next; | |
464 } | |
465 return NULL; | |
466 } | |
467 | |
468 | |
469 /** | |
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470 * Replaces 8 elements with their average value. |
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471 * Called by qdm2_decode_superblock before starting subblock decoding. |
2914 | 472 * |
473 * @param q context | |
474 */ | |
475 static void average_quantized_coeffs (QDM2Context *q) | |
476 { | |
477 int i, j, n, ch, sum; | |
478 | |
479 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; | |
480 | |
481 for (ch = 0; ch < q->nb_channels; ch++) | |
482 for (i = 0; i < n; i++) { | |
483 sum = 0; | |
484 | |
485 for (j = 0; j < 8; j++) | |
486 sum += q->quantized_coeffs[ch][i][j]; | |
487 | |
488 sum /= 8; | |
489 if (sum > 0) | |
490 sum--; | |
491 | |
492 for (j=0; j < 8; j++) | |
493 q->quantized_coeffs[ch][i][j] = sum; | |
494 } | |
495 } | |
496 | |
497 | |
498 /** | |
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499 * Build subband samples with noise weighted by q->tone_level. |
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500 * Called by synthfilt_build_sb_samples. |
2914 | 501 * |
502 * @param q context | |
503 * @param sb subband index | |
504 */ | |
505 static void build_sb_samples_from_noise (QDM2Context *q, int sb) | |
506 { | |
507 int ch, j; | |
508 | |
509 FIX_NOISE_IDX(q->noise_idx); | |
510 | |
511 if (!q->nb_channels) | |
512 return; | |
513 | |
514 for (ch = 0; ch < q->nb_channels; ch++) | |
515 for (j = 0; j < 64; j++) { | |
516 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); | |
517 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); | |
518 } | |
519 } | |
520 | |
521 | |
522 /** | |
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523 * Called while processing data from subpackets 11 and 12. |
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524 * Used after making changes to coding_method array. |
2914 | 525 * |
526 * @param sb subband index | |
527 * @param channels number of channels | |
528 * @param coding_method q->coding_method[0][0][0] | |
529 */ | |
3076 | 530 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) |
2914 | 531 { |
532 int j,k; | |
533 int ch; | |
534 int run, case_val; | |
535 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}; | |
536 | |
537 for (ch = 0; ch < channels; ch++) { | |
538 for (j = 0; j < 64; ) { | |
539 if((coding_method[ch][sb][j] - 8) > 22) { | |
540 run = 1; | |
541 case_val = 8; | |
542 } else { | |
3333 | 543 switch (switchtable[coding_method[ch][sb][j]-8]) { |
2914 | 544 case 0: run = 10; case_val = 10; break; |
545 case 1: run = 1; case_val = 16; break; | |
546 case 2: run = 5; case_val = 24; break; | |
547 case 3: run = 3; case_val = 30; break; | |
548 case 4: run = 1; case_val = 30; break; | |
549 case 5: run = 1; case_val = 8; break; | |
550 default: run = 1; case_val = 8; break; | |
551 } | |
552 } | |
553 for (k = 0; k < run; k++) | |
554 if (j + k < 128) | |
555 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) | |
556 if (k > 0) { | |
557 SAMPLES_NEEDED | |
558 //not debugged, almost never used | |
559 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t)); | |
560 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); | |
561 } | |
562 j += run; | |
563 } | |
564 } | |
565 } | |
566 | |
567 | |
568 /** | |
569 * Related to synthesis filter | |
570 * Called by process_subpacket_10 | |
571 * | |
572 * @param q context | |
573 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 | |
574 */ | |
575 static void fill_tone_level_array (QDM2Context *q, int flag) | |
576 { | |
577 int i, sb, ch, sb_used; | |
578 int tmp, tab; | |
579 | |
580 // This should never happen | |
581 if (q->nb_channels <= 0) | |
582 return; | |
583 | |
584 for (ch = 0; ch < q->nb_channels; ch++) | |
585 for (sb = 0; sb < 30; sb++) | |
586 for (i = 0; i < 8; i++) { | |
587 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) | |
588 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ | |
589 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; | |
590 else | |
591 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; | |
592 if(tmp < 0) | |
593 tmp += 0xff; | |
594 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; | |
595 } | |
596 | |
597 sb_used = QDM2_SB_USED(q->sub_sampling); | |
598 | |
599 if ((q->superblocktype_2_3 != 0) && !flag) { | |
600 for (sb = 0; sb < sb_used; sb++) | |
601 for (ch = 0; ch < q->nb_channels; ch++) | |
602 for (i = 0; i < 64; i++) { | |
603 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; | |
604 if (q->tone_level_idx[ch][sb][i] < 0) | |
605 q->tone_level[ch][sb][i] = 0; | |
606 else | |
607 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; | |
608 } | |
609 } else { | |
610 tab = q->superblocktype_2_3 ? 0 : 1; | |
611 for (sb = 0; sb < sb_used; sb++) { | |
612 if ((sb >= 4) && (sb <= 23)) { | |
613 for (ch = 0; ch < q->nb_channels; ch++) | |
614 for (i = 0; i < 64; i++) { | |
615 tmp = q->tone_level_idx_base[ch][sb][i / 8] - | |
616 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - | |
617 q->tone_level_idx_mid[ch][sb - 4][i / 8] - | |
618 q->tone_level_idx_hi2[ch][sb - 4]; | |
619 q->tone_level_idx[ch][sb][i] = tmp & 0xff; | |
620 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
621 q->tone_level[ch][sb][i] = 0; | |
622 else | |
623 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
624 } | |
625 } else { | |
626 if (sb > 4) { | |
627 for (ch = 0; ch < q->nb_channels; ch++) | |
628 for (i = 0; i < 64; i++) { | |
629 tmp = q->tone_level_idx_base[ch][sb][i / 8] - | |
630 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - | |
631 q->tone_level_idx_hi2[ch][sb - 4]; | |
632 q->tone_level_idx[ch][sb][i] = tmp & 0xff; | |
633 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
634 q->tone_level[ch][sb][i] = 0; | |
635 else | |
636 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
637 } | |
638 } else { | |
639 for (ch = 0; ch < q->nb_channels; ch++) | |
640 for (i = 0; i < 64; i++) { | |
641 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; | |
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 } | |
648 } | |
649 } | |
650 } | |
651 | |
652 return; | |
653 } | |
654 | |
655 | |
656 /** | |
657 * Related to synthesis filter | |
658 * Called by process_subpacket_11 | |
659 * c is built with data from subpacket 11 | |
660 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples | |
661 * | |
2967 | 662 * @param tone_level_idx |
2914 | 663 * @param tone_level_idx_temp |
664 * @param coding_method q->coding_method[0][0][0] | |
665 * @param nb_channels number of channels | |
666 * @param c coming from subpacket 11, passed as 8*c | |
667 * @param superblocktype_2_3 flag based on superblock packet type | |
668 * @param cm_table_select q->cm_table_select | |
669 */ | |
670 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, | |
671 sb_int8_array coding_method, int nb_channels, | |
672 int c, int superblocktype_2_3, int cm_table_select) | |
673 { | |
674 int ch, sb, j; | |
675 int tmp, acc, esp_40, comp; | |
676 int add1, add2, add3, add4; | |
677 int64_t multres; | |
678 | |
679 // This should never happen | |
680 if (nb_channels <= 0) | |
681 return; | |
682 | |
683 if (!superblocktype_2_3) { | |
684 /* This case is untested, no samples available */ | |
685 SAMPLES_NEEDED | |
686 for (ch = 0; ch < nb_channels; ch++) | |
687 for (sb = 0; sb < 30; sb++) { | |
688 for (j = 1; j < 64; j++) { | |
689 add1 = tone_level_idx[ch][sb][j] - 10; | |
690 if (add1 < 0) | |
691 add1 = 0; | |
692 add2 = add3 = add4 = 0; | |
693 if (sb > 1) { | |
694 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; | |
695 if (add2 < 0) | |
696 add2 = 0; | |
697 } | |
698 if (sb > 0) { | |
699 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; | |
700 if (add3 < 0) | |
701 add3 = 0; | |
702 } | |
703 if (sb < 29) { | |
704 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; | |
705 if (add4 < 0) | |
706 add4 = 0; | |
707 } | |
708 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; | |
709 if (tmp < 0) | |
710 tmp = 0; | |
711 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; | |
712 } | |
713 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; | |
714 } | |
715 acc = 0; | |
716 for (ch = 0; ch < nb_channels; ch++) | |
717 for (sb = 0; sb < 30; sb++) | |
718 for (j = 0; j < 64; j++) | |
719 acc += tone_level_idx_temp[ch][sb][j]; | |
720 if (acc) | |
721 tmp = c * 256 / (acc & 0xffff); | |
722 multres = 0x66666667 * (acc * 10); | |
723 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); | |
724 for (ch = 0; ch < nb_channels; ch++) | |
725 for (sb = 0; sb < 30; sb++) | |
726 for (j = 0; j < 64; j++) { | |
727 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; | |
728 if (comp < 0) | |
729 comp += 0xff; | |
730 comp /= 256; // signed shift | |
731 switch(sb) { | |
732 case 0: | |
733 if (comp < 30) | |
734 comp = 30; | |
735 comp += 15; | |
736 break; | |
737 case 1: | |
738 if (comp < 24) | |
739 comp = 24; | |
740 comp += 10; | |
741 break; | |
742 case 2: | |
743 case 3: | |
744 case 4: | |
745 if (comp < 16) | |
746 comp = 16; | |
747 } | |
748 if (comp <= 5) | |
749 tmp = 0; | |
750 else if (comp <= 10) | |
751 tmp = 10; | |
752 else if (comp <= 16) | |
753 tmp = 16; | |
754 else if (comp <= 24) | |
755 tmp = -1; | |
756 else | |
757 tmp = 0; | |
758 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; | |
759 } | |
760 for (sb = 0; sb < 30; sb++) | |
761 fix_coding_method_array(sb, nb_channels, coding_method); | |
762 for (ch = 0; ch < nb_channels; ch++) | |
763 for (sb = 0; sb < 30; sb++) | |
764 for (j = 0; j < 64; j++) | |
765 if (sb >= 10) { | |
766 if (coding_method[ch][sb][j] < 10) | |
767 coding_method[ch][sb][j] = 10; | |
768 } else { | |
769 if (sb >= 2) { | |
770 if (coding_method[ch][sb][j] < 16) | |
771 coding_method[ch][sb][j] = 16; | |
772 } else { | |
773 if (coding_method[ch][sb][j] < 30) | |
774 coding_method[ch][sb][j] = 30; | |
775 } | |
776 } | |
777 } else { // superblocktype_2_3 != 0 | |
778 for (ch = 0; ch < nb_channels; ch++) | |
779 for (sb = 0; sb < 30; sb++) | |
780 for (j = 0; j < 64; j++) | |
781 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; | |
782 } | |
783 | |
784 return; | |
785 } | |
786 | |
787 | |
788 /** | |
789 * | |
790 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8 | |
791 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used | |
792 * | |
793 * @param q context | |
794 * @param gb bitreader context | |
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795 * @param length packet length in bits |
2914 | 796 * @param sb_min lower subband processed (sb_min included) |
797 * @param sb_max higher subband processed (sb_max excluded) | |
798 */ | |
799 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) | |
800 { | |
801 int sb, j, k, n, ch, run, channels; | |
802 int joined_stereo, zero_encoding, chs; | |
803 int type34_first; | |
804 float type34_div = 0; | |
805 float type34_predictor; | |
806 float samples[10], sign_bits[16]; | |
807 | |
808 if (length == 0) { | |
809 // If no data use noise | |
810 for (sb=sb_min; sb < sb_max; sb++) | |
811 build_sb_samples_from_noise (q, sb); | |
812 | |
813 return; | |
814 } | |
815 | |
816 for (sb = sb_min; sb < sb_max; sb++) { | |
817 FIX_NOISE_IDX(q->noise_idx); | |
818 | |
819 channels = q->nb_channels; | |
820 | |
821 if (q->nb_channels <= 1 || sb < 12) | |
822 joined_stereo = 0; | |
823 else if (sb >= 24) | |
824 joined_stereo = 1; | |
825 else | |
826 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0; | |
827 | |
828 if (joined_stereo) { | |
829 if (BITS_LEFT(length,gb) >= 16) | |
830 for (j = 0; j < 16; j++) | |
831 sign_bits[j] = get_bits1 (gb); | |
832 | |
833 for (j = 0; j < 64; j++) | |
834 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) | |
835 q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; | |
836 | |
837 fix_coding_method_array(sb, q->nb_channels, q->coding_method); | |
838 channels = 1; | |
839 } | |
840 | |
841 for (ch = 0; ch < channels; ch++) { | |
842 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; | |
843 type34_predictor = 0.0; | |
844 type34_first = 1; | |
845 | |
846 for (j = 0; j < 128; ) { | |
847 switch (q->coding_method[ch][sb][j / 2]) { | |
848 case 8: | |
849 if (BITS_LEFT(length,gb) >= 10) { | |
850 if (zero_encoding) { | |
851 for (k = 0; k < 5; k++) { | |
852 if ((j + 2 * k) >= 128) | |
853 break; | |
854 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; | |
855 } | |
856 } else { | |
857 n = get_bits(gb, 8); | |
858 for (k = 0; k < 5; k++) | |
859 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
860 } | |
861 for (k = 0; k < 5; k++) | |
862 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
863 } else { | |
864 for (k = 0; k < 10; k++) | |
865 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
866 } | |
867 run = 10; | |
868 break; | |
869 | |
870 case 10: | |
871 if (BITS_LEFT(length,gb) >= 1) { | |
872 float f = 0.81; | |
873 | |
874 if (get_bits1(gb)) | |
875 f = -f; | |
876 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; | |
877 samples[0] = f; | |
878 } else { | |
879 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
880 } | |
881 run = 1; | |
882 break; | |
883 | |
884 case 16: | |
885 if (BITS_LEFT(length,gb) >= 10) { | |
886 if (zero_encoding) { | |
887 for (k = 0; k < 5; k++) { | |
888 if ((j + k) >= 128) | |
889 break; | |
890 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; | |
891 } | |
892 } else { | |
893 n = get_bits (gb, 8); | |
894 for (k = 0; k < 5; k++) | |
895 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
896 } | |
897 } else { | |
898 for (k = 0; k < 5; k++) | |
899 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
900 } | |
901 run = 5; | |
902 break; | |
903 | |
904 case 24: | |
905 if (BITS_LEFT(length,gb) >= 7) { | |
906 n = get_bits(gb, 7); | |
907 for (k = 0; k < 3; k++) | |
908 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; | |
909 } else { | |
910 for (k = 0; k < 3; k++) | |
911 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
912 } | |
913 run = 3; | |
914 break; | |
915 | |
916 case 30: | |
917 if (BITS_LEFT(length,gb) >= 4) | |
918 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)]; | |
919 else | |
920 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
2967 | 921 |
2914 | 922 run = 1; |
923 break; | |
924 | |
925 case 34: | |
926 if (BITS_LEFT(length,gb) >= 7) { | |
927 if (type34_first) { | |
928 type34_div = (float)(1 << get_bits(gb, 2)); | |
929 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; | |
930 type34_predictor = samples[0]; | |
931 type34_first = 0; | |
932 } else { | |
933 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor; | |
934 type34_predictor = samples[0]; | |
935 } | |
936 } else { | |
937 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
938 } | |
939 run = 1; | |
940 break; | |
941 | |
942 default: | |
943 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
944 run = 1; | |
945 break; | |
946 } | |
947 | |
948 if (joined_stereo) { | |
949 float tmp[10][MPA_MAX_CHANNELS]; | |
950 | |
951 for (k = 0; k < run; k++) { | |
952 tmp[k][0] = samples[k]; | |
953 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; | |
954 } | |
955 for (chs = 0; chs < q->nb_channels; chs++) | |
956 for (k = 0; k < run; k++) | |
957 if ((j + k) < 128) | |
958 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5); | |
959 } else { | |
960 for (k = 0; k < run; k++) | |
961 if ((j + k) < 128) | |
962 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5); | |
963 } | |
964 | |
965 j += run; | |
966 } // j loop | |
967 } // channel loop | |
968 } // subband loop | |
969 } | |
970 | |
971 | |
972 /** | |
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973 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]). |
2914 | 974 * This is similar to process_subpacket_9, but for a single channel and for element [0] |
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975 * same VLC tables as process_subpacket_9 are used. |
2914 | 976 * |
977 * @param q context | |
978 * @param quantized_coeffs pointer to quantized_coeffs[ch][0] | |
979 * @param gb bitreader context | |
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980 * @param length packet length in bits |
2914 | 981 */ |
982 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) | |
983 { | |
984 int i, k, run, level, diff; | |
985 | |
986 if (BITS_LEFT(length,gb) < 16) | |
987 return; | |
988 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); | |
989 | |
990 quantized_coeffs[0] = level; | |
991 | |
992 for (i = 0; i < 7; ) { | |
993 if (BITS_LEFT(length,gb) < 16) | |
994 break; | |
995 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; | |
996 | |
997 if (BITS_LEFT(length,gb) < 16) | |
998 break; | |
999 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); | |
2967 | 1000 |
2914 | 1001 for (k = 1; k <= run; k++) |
1002 quantized_coeffs[i + k] = (level + ((k * diff) / run)); | |
2967 | 1003 |
2914 | 1004 level += diff; |
1005 i += run; | |
1006 } | |
1007 } | |
1008 | |
1009 | |
1010 /** | |
1011 * Related to synthesis filter, process data from packet 10 | |
1012 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 | |
1013 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10 | |
1014 * | |
1015 * @param q context | |
1016 * @param gb bitreader context | |
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1017 * @param length packet length in bits |
2914 | 1018 */ |
1019 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) | |
1020 { | |
1021 int sb, j, k, n, ch; | |
1022 | |
1023 for (ch = 0; ch < q->nb_channels; ch++) { | |
1024 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length); | |
1025 | |
1026 if (BITS_LEFT(length,gb) < 16) { | |
1027 memset(q->quantized_coeffs[ch][0], 0, 8); | |
1028 break; | |
1029 } | |
1030 } | |
1031 | |
1032 n = q->sub_sampling + 1; | |
1033 | |
1034 for (sb = 0; sb < n; sb++) | |
1035 for (ch = 0; ch < q->nb_channels; ch++) | |
1036 for (j = 0; j < 8; j++) { | |
1037 if (BITS_LEFT(length,gb) < 1) | |
1038 break; | |
1039 if (get_bits1(gb)) { | |
1040 for (k=0; k < 8; k++) { | |
1041 if (BITS_LEFT(length,gb) < 16) | |
1042 break; | |
1043 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); | |
1044 } | |
1045 } else { | |
1046 for (k=0; k < 8; k++) | |
1047 q->tone_level_idx_hi1[ch][sb][j][k] = 0; | |
1048 } | |
1049 } | |
1050 | |
1051 n = QDM2_SB_USED(q->sub_sampling) - 4; | |
1052 | |
1053 for (sb = 0; sb < n; sb++) | |
1054 for (ch = 0; ch < q->nb_channels; ch++) { | |
1055 if (BITS_LEFT(length,gb) < 16) | |
1056 break; | |
1057 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); | |
1058 if (sb > 19) | |
1059 q->tone_level_idx_hi2[ch][sb] -= 16; | |
1060 else | |
1061 for (j = 0; j < 8; j++) | |
1062 q->tone_level_idx_mid[ch][sb][j] = -16; | |
1063 } | |
1064 | |
1065 n = QDM2_SB_USED(q->sub_sampling) - 5; | |
1066 | |
1067 for (sb = 0; sb < n; sb++) | |
1068 for (ch = 0; ch < q->nb_channels; ch++) | |
1069 for (j = 0; j < 8; j++) { | |
1070 if (BITS_LEFT(length,gb) < 16) | |
1071 break; | |
1072 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; | |
1073 } | |
1074 } | |
1075 | |
1076 /** | |
1077 * Process subpacket 9, init quantized_coeffs with data from it | |
1078 * | |
1079 * @param q context | |
1080 * @param node pointer to node with packet | |
1081 */ | |
1082 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) | |
1083 { | |
1084 GetBitContext gb; | |
1085 int i, j, k, n, ch, run, level, diff; | |
1086 | |
2916 | 1087 init_get_bits(&gb, node->packet->data, node->packet->size*8); |
2914 | 1088 |
1089 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function | |
1090 | |
1091 for (i = 1; i < n; i++) | |
1092 for (ch=0; ch < q->nb_channels; ch++) { | |
1093 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); | |
1094 q->quantized_coeffs[ch][i][0] = level; | |
1095 | |
1096 for (j = 0; j < (8 - 1); ) { | |
1097 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; | |
1098 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); | |
1099 | |
1100 for (k = 1; k <= run; k++) | |
1101 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); | |
1102 | |
1103 level += diff; | |
1104 j += run; | |
1105 } | |
1106 } | |
1107 | |
1108 for (ch = 0; ch < q->nb_channels; ch++) | |
1109 for (i = 0; i < 8; i++) | |
1110 q->quantized_coeffs[ch][0][i] = 0; | |
1111 } | |
1112 | |
1113 | |
1114 /** | |
1115 * Process subpacket 10 if not null, else | |
1116 * | |
1117 * @param q context | |
1118 * @param node pointer to node with packet | |
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1119 * @param length packet length in bits |
2914 | 1120 */ |
1121 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) | |
1122 { | |
1123 GetBitContext gb; | |
1124 | |
2916 | 1125 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
2914 | 1126 |
1127 if (length != 0) { | |
1128 init_tone_level_dequantization(q, &gb, length); | |
1129 fill_tone_level_array(q, 1); | |
1130 } else { | |
1131 fill_tone_level_array(q, 0); | |
1132 } | |
1133 } | |
1134 | |
1135 | |
1136 /** | |
1137 * Process subpacket 11 | |
1138 * | |
1139 * @param q context | |
1140 * @param node pointer to node with packet | |
1141 * @param length packet length in bit | |
1142 */ | |
1143 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length) | |
1144 { | |
1145 GetBitContext gb; | |
1146 | |
2916 | 1147 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
2914 | 1148 if (length >= 32) { |
1149 int c = get_bits (&gb, 13); | |
1150 | |
1151 if (c > 3) | |
1152 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, | |
1153 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select); | |
1154 } | |
1155 | |
1156 synthfilt_build_sb_samples(q, &gb, length, 0, 8); | |
1157 } | |
1158 | |
1159 | |
1160 /** | |
1161 * Process subpacket 12 | |
1162 * | |
1163 * @param q context | |
1164 * @param node pointer to node with packet | |
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1165 * @param length packet length in bits |
2914 | 1166 */ |
1167 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) | |
1168 { | |
1169 GetBitContext gb; | |
1170 | |
2916 | 1171 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
2914 | 1172 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); |
1173 } | |
1174 | |
1175 /* | |
1176 * Process new subpackets for synthesis filter | |
1177 * | |
1178 * @param q context | |
1179 * @param list list with synthesis filter packets (list D) | |
1180 */ | |
1181 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) | |
1182 { | |
1183 QDM2SubPNode *nodes[4]; | |
1184 | |
1185 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); | |
1186 if (nodes[0] != NULL) | |
1187 process_subpacket_9(q, nodes[0]); | |
1188 | |
1189 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); | |
1190 if (nodes[1] != NULL) | |
1191 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3); | |
1192 else | |
1193 process_subpacket_10(q, NULL, 0); | |
1194 | |
1195 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); | |
1196 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) | |
1197 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3)); | |
1198 else | |
1199 process_subpacket_11(q, NULL, 0); | |
1200 | |
1201 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); | |
1202 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) | |
1203 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3)); | |
1204 else | |
1205 process_subpacket_12(q, NULL, 0); | |
1206 } | |
1207 | |
1208 | |
1209 /* | |
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1210 * Decode superblock, fill packet lists. |
2914 | 1211 * |
1212 * @param q context | |
1213 */ | |
1214 static void qdm2_decode_super_block (QDM2Context *q) | |
1215 { | |
1216 GetBitContext gb; | |
1217 QDM2SubPacket header, *packet; | |
1218 int i, packet_bytes, sub_packet_size, sub_packets_D; | |
1219 unsigned int next_index = 0; | |
1220 | |
1221 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); | |
1222 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); | |
1223 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); | |
1224 | |
1225 q->sub_packets_B = 0; | |
1226 sub_packets_D = 0; | |
1227 | |
1228 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] | |
1229 | |
2916 | 1230 init_get_bits(&gb, q->compressed_data, q->compressed_size*8); |
2914 | 1231 qdm2_decode_sub_packet_header(&gb, &header); |
1232 | |
1233 if (header.type < 2 || header.type >= 8) { | |
1234 q->has_errors = 1; | |
1235 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); | |
1236 return; | |
1237 } | |
1238 | |
1239 q->superblocktype_2_3 = (header.type == 2 || header.type == 3); | |
1240 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); | |
1241 | |
2916 | 1242 init_get_bits(&gb, header.data, header.size*8); |
2914 | 1243 |
1244 if (header.type == 2 || header.type == 4 || header.type == 5) { | |
1245 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8); | |
1246 | |
1247 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); | |
1248 | |
1249 if (csum != 0) { | |
1250 q->has_errors = 1; | |
1251 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); | |
1252 return; | |
1253 } | |
1254 } | |
1255 | |
1256 q->sub_packet_list_B[0].packet = NULL; | |
1257 q->sub_packet_list_D[0].packet = NULL; | |
1258 | |
1259 for (i = 0; i < 6; i++) | |
1260 if (--q->fft_level_exp[i] < 0) | |
1261 q->fft_level_exp[i] = 0; | |
1262 | |
1263 for (i = 0; packet_bytes > 0; i++) { | |
1264 int j; | |
1265 | |
1266 q->sub_packet_list_A[i].next = NULL; | |
1267 | |
1268 if (i > 0) { | |
1269 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; | |
1270 | |
1271 /* seek to next block */ | |
2916 | 1272 init_get_bits(&gb, header.data, header.size*8); |
2914 | 1273 skip_bits(&gb, next_index*8); |
1274 | |
1275 if (next_index >= header.size) | |
1276 break; | |
1277 } | |
1278 | |
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1279 /* decode subpacket */ |
2914 | 1280 packet = &q->sub_packets[i]; |
1281 qdm2_decode_sub_packet_header(&gb, packet); | |
1282 next_index = packet->size + get_bits_count(&gb) / 8; | |
1283 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; | |
1284 | |
1285 if (packet->type == 0) | |
1286 break; | |
1287 | |
1288 if (sub_packet_size > packet_bytes) { | |
1289 if (packet->type != 10 && packet->type != 11 && packet->type != 12) | |
1290 break; | |
1291 packet->size += packet_bytes - sub_packet_size; | |
1292 } | |
1293 | |
1294 packet_bytes -= sub_packet_size; | |
1295 | |
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1296 /* add subpacket to 'all subpackets' list */ |
2914 | 1297 q->sub_packet_list_A[i].packet = packet; |
1298 | |
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1299 /* add subpacket to related list */ |
2914 | 1300 if (packet->type == 8) { |
1301 SAMPLES_NEEDED_2("packet type 8"); | |
1302 return; | |
1303 } else if (packet->type >= 9 && packet->type <= 12) { | |
1304 /* packets for MPEG Audio like Synthesis Filter */ | |
1305 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); | |
1306 } else if (packet->type == 13) { | |
1307 for (j = 0; j < 6; j++) | |
1308 q->fft_level_exp[j] = get_bits(&gb, 6); | |
1309 } else if (packet->type == 14) { | |
1310 for (j = 0; j < 6; j++) | |
1311 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); | |
1312 } else if (packet->type == 15) { | |
1313 SAMPLES_NEEDED_2("packet type 15") | |
1314 return; | |
1315 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { | |
1316 /* packets for FFT */ | |
1317 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); | |
1318 } | |
1319 } // Packet bytes loop | |
1320 | |
1321 /* **************************************************************** */ | |
1322 if (q->sub_packet_list_D[0].packet != NULL) { | |
1323 process_synthesis_subpackets(q, q->sub_packet_list_D); | |
1324 q->do_synth_filter = 1; | |
1325 } else if (q->do_synth_filter) { | |
1326 process_subpacket_10(q, NULL, 0); | |
1327 process_subpacket_11(q, NULL, 0); | |
1328 process_subpacket_12(q, NULL, 0); | |
1329 } | |
1330 /* **************************************************************** */ | |
1331 } | |
1332 | |
1333 | |
1334 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, | |
1335 int offset, int duration, int channel, | |
1336 int exp, int phase) | |
1337 { | |
1338 if (q->fft_coefs_min_index[duration] < 0) | |
1339 q->fft_coefs_min_index[duration] = q->fft_coefs_index; | |
1340 | |
1341 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); | |
1342 q->fft_coefs[q->fft_coefs_index].channel = channel; | |
1343 q->fft_coefs[q->fft_coefs_index].offset = offset; | |
1344 q->fft_coefs[q->fft_coefs_index].exp = exp; | |
1345 q->fft_coefs[q->fft_coefs_index].phase = phase; | |
1346 q->fft_coefs_index++; | |
1347 } | |
1348 | |
1349 | |
1350 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) | |
1351 { | |
1352 int channel, stereo, phase, exp; | |
1353 int local_int_4, local_int_8, stereo_phase, local_int_10; | |
1354 int local_int_14, stereo_exp, local_int_20, local_int_28; | |
1355 int n, offset; | |
1356 | |
1357 local_int_4 = 0; | |
1358 local_int_28 = 0; | |
1359 local_int_20 = 2; | |
1360 local_int_8 = (4 - duration); | |
1361 local_int_10 = 1 << (q->group_order - duration - 1); | |
1362 offset = 1; | |
1363 | |
1364 while (1) { | |
1365 if (q->superblocktype_2_3) { | |
1366 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { | |
1367 offset = 1; | |
1368 if (n == 0) { | |
1369 local_int_4 += local_int_10; | |
1370 local_int_28 += (1 << local_int_8); | |
1371 } else { | |
1372 local_int_4 += 8*local_int_10; | |
1373 local_int_28 += (8 << local_int_8); | |
1374 } | |
1375 } | |
1376 offset += (n - 2); | |
1377 } else { | |
1378 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); | |
1379 while (offset >= (local_int_10 - 1)) { | |
1380 offset += (1 - (local_int_10 - 1)); | |
1381 local_int_4 += local_int_10; | |
1382 local_int_28 += (1 << local_int_8); | |
1383 } | |
1384 } | |
1385 | |
1386 if (local_int_4 >= q->group_size) | |
1387 return; | |
1388 | |
1389 local_int_14 = (offset >> local_int_8); | |
1390 | |
1391 if (q->nb_channels > 1) { | |
1392 channel = get_bits1(gb); | |
1393 stereo = get_bits1(gb); | |
1394 } else { | |
1395 channel = 0; | |
1396 stereo = 0; | |
1397 } | |
1398 | |
1399 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); | |
1400 exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; | |
1401 exp = (exp < 0) ? 0 : exp; | |
1402 | |
1403 phase = get_bits(gb, 3); | |
1404 stereo_exp = 0; | |
1405 stereo_phase = 0; | |
1406 | |
1407 if (stereo) { | |
1408 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); | |
1409 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); | |
1410 if (stereo_phase < 0) | |
1411 stereo_phase += 8; | |
1412 } | |
1413 | |
1414 if (q->frequency_range > (local_int_14 + 1)) { | |
1415 int sub_packet = (local_int_20 + local_int_28); | |
1416 | |
1417 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); | |
1418 if (stereo) | |
1419 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); | |
1420 } | |
1421 | |
1422 offset++; | |
1423 } | |
1424 } | |
1425 | |
1426 | |
1427 static void qdm2_decode_fft_packets (QDM2Context *q) | |
1428 { | |
1429 int i, j, min, max, value, type, unknown_flag; | |
1430 GetBitContext gb; | |
1431 | |
1432 if (q->sub_packet_list_B[0].packet == NULL) | |
1433 return; | |
1434 | |
1435 /* reset minimum indices for FFT coefficients */ | |
1436 q->fft_coefs_index = 0; | |
1437 for (i=0; i < 5; i++) | |
1438 q->fft_coefs_min_index[i] = -1; | |
1439 | |
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1440 /* process subpackets ordered by type, largest type first */ |
2914 | 1441 for (i = 0, max = 256; i < q->sub_packets_B; i++) { |
1442 QDM2SubPacket *packet; | |
1443 | |
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1444 /* find subpacket with largest type less than max */ |
2914 | 1445 for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) { |
1446 value = q->sub_packet_list_B[j].packet->type; | |
1447 if (value > min && value < max) { | |
1448 min = value; | |
1449 packet = q->sub_packet_list_B[j].packet; | |
1450 } | |
1451 } | |
1452 | |
1453 max = min; | |
1454 | |
1455 /* check for errors (?) */ | |
1456 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) | |
1457 return; | |
1458 | |
1459 /* decode FFT tones */ | |
2916 | 1460 init_get_bits (&gb, packet->data, packet->size*8); |
2914 | 1461 |
1462 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) | |
1463 unknown_flag = 1; | |
1464 else | |
1465 unknown_flag = 0; | |
1466 | |
1467 type = packet->type; | |
1468 | |
1469 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { | |
1470 int duration = q->sub_sampling + 5 - (type & 15); | |
1471 | |
1472 if (duration >= 0 && duration < 4) | |
1473 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); | |
1474 } else if (type == 31) { | |
3320 | 1475 for (j=0; j < 4; j++) |
1476 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); | |
2914 | 1477 } else if (type == 46) { |
3320 | 1478 for (j=0; j < 6; j++) |
1479 q->fft_level_exp[j] = get_bits(&gb, 6); | |
1480 for (j=0; j < 4; j++) | |
1481 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); | |
2914 | 1482 } |
1483 } // Loop on B packets | |
1484 | |
1485 /* calculate maximum indices for FFT coefficients */ | |
1486 for (i = 0, j = -1; i < 5; i++) | |
1487 if (q->fft_coefs_min_index[i] >= 0) { | |
1488 if (j >= 0) | |
1489 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; | |
1490 j = i; | |
1491 } | |
1492 if (j >= 0) | |
1493 q->fft_coefs_max_index[j] = q->fft_coefs_index; | |
1494 } | |
1495 | |
1496 | |
1497 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) | |
1498 { | |
1499 float level, f[6]; | |
1500 int i; | |
1501 QDM2Complex c; | |
1502 const double iscale = 2.0*M_PI / 512.0; | |
1503 | |
1504 tone->phase += tone->phase_shift; | |
1505 | |
1506 /* calculate current level (maximum amplitude) of tone */ | |
1507 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; | |
1508 c.im = level * sin(tone->phase*iscale); | |
1509 c.re = level * cos(tone->phase*iscale); | |
1510 | |
1511 /* generate FFT coefficients for tone */ | |
1512 if (tone->duration >= 3 || tone->cutoff >= 3) { | |
1513 tone->samples_im[0] += c.im; | |
1514 tone->samples_re[0] += c.re; | |
1515 tone->samples_im[1] -= c.im; | |
1516 tone->samples_re[1] -= c.re; | |
1517 } else { | |
1518 f[1] = -tone->table[4]; | |
1519 f[0] = tone->table[3] - tone->table[0]; | |
1520 f[2] = 1.0 - tone->table[2] - tone->table[3]; | |
1521 f[3] = tone->table[1] + tone->table[4] - 1.0; | |
1522 f[4] = tone->table[0] - tone->table[1]; | |
1523 f[5] = tone->table[2]; | |
1524 for (i = 0; i < 2; i++) { | |
1525 tone->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i]; | |
1526 tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); | |
1527 } | |
1528 for (i = 0; i < 4; i++) { | |
1529 tone->samples_re[i] += c.re * f[i+2]; | |
1530 tone->samples_im[i] += c.im * f[i+2]; | |
1531 } | |
1532 } | |
1533 | |
1534 /* copy the tone if it has not yet died out */ | |
1535 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { | |
1536 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); | |
1537 q->fft_tone_end = (q->fft_tone_end + 1) % 1000; | |
1538 } | |
1539 } | |
1540 | |
1541 | |
1542 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) | |
1543 { | |
1544 int i, j, ch; | |
1545 const double iscale = 0.25 * M_PI; | |
1546 | |
1547 for (ch = 0; ch < q->channels; ch++) { | |
1548 memset(q->fft.samples_im[ch], 0, q->fft_size * sizeof(float)); | |
1549 memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float)); | |
1550 } | |
1551 | |
1552 | |
1553 /* apply FFT tones with duration 4 (1 FFT period) */ | |
1554 if (q->fft_coefs_min_index[4] >= 0) | |
1555 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { | |
1556 float level; | |
1557 QDM2Complex c; | |
1558 | |
1559 if (q->fft_coefs[i].sub_packet != sub_packet) | |
1560 break; | |
1561 | |
1562 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; | |
1563 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]; | |
1564 | |
1565 c.re = level * cos(q->fft_coefs[i].phase * iscale); | |
1566 c.im = level * sin(q->fft_coefs[i].phase * iscale); | |
1567 q->fft.samples_re[ch][q->fft_coefs[i].offset + 0] += c.re; | |
1568 q->fft.samples_im[ch][q->fft_coefs[i].offset + 0] += c.im; | |
1569 q->fft.samples_re[ch][q->fft_coefs[i].offset + 1] -= c.re; | |
1570 q->fft.samples_im[ch][q->fft_coefs[i].offset + 1] -= c.im; | |
1571 } | |
1572 | |
1573 /* generate existing FFT tones */ | |
1574 for (i = q->fft_tone_end; i != q->fft_tone_start; ) { | |
1575 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); | |
1576 q->fft_tone_start = (q->fft_tone_start + 1) % 1000; | |
1577 } | |
1578 | |
1579 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ | |
1580 for (i = 0; i < 4; i++) | |
1581 if (q->fft_coefs_min_index[i] >= 0) { | |
1582 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { | |
1583 int offset, four_i; | |
1584 FFTTone tone; | |
1585 | |
1586 if (q->fft_coefs[j].sub_packet != sub_packet) | |
1587 break; | |
1588 | |
1589 four_i = (4 - i); | |
1590 offset = q->fft_coefs[j].offset >> four_i; | |
1591 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; | |
1592 | |
1593 if (offset < q->frequency_range) { | |
1594 if (offset < 2) | |
1595 tone.cutoff = offset; | |
1596 else | |
1597 tone.cutoff = (offset >= 60) ? 3 : 2; | |
1598 | |
1599 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]; | |
1600 tone.samples_im = &q->fft.samples_im[ch][offset]; | |
1601 tone.samples_re = &q->fft.samples_re[ch][offset]; | |
1602 tone.table = (float*)fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; | |
1603 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; | |
1604 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); | |
1605 tone.duration = i; | |
1606 tone.time_index = 0; | |
1607 | |
1608 qdm2_fft_generate_tone(q, &tone); | |
1609 } | |
1610 } | |
1611 q->fft_coefs_min_index[i] = j; | |
1612 } | |
1613 } | |
1614 | |
1615 | |
1616 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) | |
1617 { | |
1618 const int n = 1 << (q->fft_order - 1); | |
1619 const int n2 = n >> 1; | |
1620 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f; | |
1621 float c, s, f0, f1, f2, f3; | |
1622 int i, j; | |
1623 | |
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1624 /* prerotation (or something like that) */ |
2914 | 1625 for (i=1; i < n2; i++) { |
1626 j = (n - i); | |
1627 c = q->exptab[i].re; | |
1628 s = -q->exptab[i].im; | |
1629 f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain; | |
1630 f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain; | |
1631 f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain; | |
1632 f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain; | |
1633 q->fft.complex[i].re = s * f0 - c * f1 + f2; | |
1634 q->fft.complex[i].im = c * f0 + s * f1 + f3; | |
1635 q->fft.complex[j].re = -s * f0 + c * f1 + f2; | |
1636 q->fft.complex[j].im = c * f0 + s * f1 - f3; | |
1637 } | |
1638 | |
1639 q->fft.complex[ 0].re = q->fft.samples_re[channel][ 0] * gain * 2.0; | |
1640 q->fft.complex[ 0].im = q->fft.samples_re[channel][ 0] * gain * 2.0; | |
1641 q->fft.complex[n2].re = q->fft.samples_re[channel][n2] * gain * 2.0; | |
1642 q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0; | |
1643 | |
1644 ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex); | |
1645 ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex); | |
1646 /* add samples to output buffer */ | |
1647 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) | |
1648 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex)[i]; | |
1649 } | |
1650 | |
1651 | |
1652 /** | |
1653 * @param q context | |
1654 * @param index subpacket number | |
1655 */ | |
1656 static void qdm2_synthesis_filter (QDM2Context *q, int index) | |
1657 { | |
1658 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; | |
1659 int i, k, ch, sb_used, sub_sampling, dither_state = 0; | |
1660 | |
1661 /* copy sb_samples */ | |
1662 sb_used = QDM2_SB_USED(q->sub_sampling); | |
1663 | |
1664 for (ch = 0; ch < q->channels; ch++) | |
1665 for (i = 0; i < 8; i++) | |
1666 for (k=sb_used; k < SBLIMIT; k++) | |
1667 q->sb_samples[ch][(8 * index) + i][k] = 0; | |
1668 | |
1669 for (ch = 0; ch < q->nb_channels; ch++) { | |
1670 OUT_INT *samples_ptr = samples + ch; | |
1671 | |
1672 for (i = 0; i < 8; i++) { | |
1673 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]), | |
1674 mpa_window, &dither_state, | |
1675 samples_ptr, q->nb_channels, | |
1676 q->sb_samples[ch][(8 * index) + i]); | |
1677 samples_ptr += 32 * q->nb_channels; | |
1678 } | |
1679 } | |
1680 | |
1681 /* add samples to output buffer */ | |
1682 sub_sampling = (4 >> q->sub_sampling); | |
1683 | |
1684 for (ch = 0; ch < q->channels; ch++) | |
1685 for (i = 0; i < q->frame_size; i++) | |
1686 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16)); | |
1687 } | |
1688 | |
1689 | |
1690 /** | |
1691 * Init static data (does not depend on specific file) | |
1692 * | |
1693 * @param q context | |
1694 */ | |
3076 | 1695 static void qdm2_init(QDM2Context *q) { |
2914 | 1696 static int inited = 0; |
1697 | |
1698 if (inited != 0) | |
1699 return; | |
1700 inited = 1; | |
1701 | |
1702 qdm2_init_vlc(); | |
1703 ff_mpa_synth_init(mpa_window); | |
1704 softclip_table_init(); | |
1705 rnd_table_init(); | |
1706 init_noise_samples(); | |
1707 | |
1708 av_log(NULL, AV_LOG_DEBUG, "init done\n"); | |
1709 } | |
1710 | |
1711 | |
1712 #if 0 | |
1713 static void dump_context(QDM2Context *q) | |
1714 { | |
1715 int i; | |
1716 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b); | |
1717 PRINT("compressed_data",q->compressed_data); | |
1718 PRINT("compressed_size",q->compressed_size); | |
1719 PRINT("frame_size",q->frame_size); | |
1720 PRINT("checksum_size",q->checksum_size); | |
1721 PRINT("channels",q->channels); | |
1722 PRINT("nb_channels",q->nb_channels); | |
1723 PRINT("fft_frame_size",q->fft_frame_size); | |
1724 PRINT("fft_size",q->fft_size); | |
1725 PRINT("sub_sampling",q->sub_sampling); | |
1726 PRINT("fft_order",q->fft_order); | |
1727 PRINT("group_order",q->group_order); | |
1728 PRINT("group_size",q->group_size); | |
1729 PRINT("sub_packet",q->sub_packet); | |
1730 PRINT("frequency_range",q->frequency_range); | |
1731 PRINT("has_errors",q->has_errors); | |
1732 PRINT("fft_tone_end",q->fft_tone_end); | |
1733 PRINT("fft_tone_start",q->fft_tone_start); | |
1734 PRINT("fft_coefs_index",q->fft_coefs_index); | |
1735 PRINT("coeff_per_sb_select",q->coeff_per_sb_select); | |
1736 PRINT("cm_table_select",q->cm_table_select); | |
1737 PRINT("noise_idx",q->noise_idx); | |
1738 | |
1739 for (i = q->fft_tone_start; i < q->fft_tone_end; i++) | |
1740 { | |
1741 FFTTone *t = &q->fft_tones[i]; | |
2967 | 1742 |
2914 | 1743 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i); |
1744 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level); | |
1745 // PRINT(" level", t->level); | |
1746 PRINT(" phase", t->phase); | |
1747 PRINT(" phase_shift", t->phase_shift); | |
1748 PRINT(" duration", t->duration); | |
1749 PRINT(" samples_im", t->samples_im); | |
1750 PRINT(" samples_re", t->samples_re); | |
1751 PRINT(" table", t->table); | |
1752 } | |
1753 | |
1754 } | |
1755 #endif | |
1756 | |
1757 | |
1758 /** | |
1759 * Init parameters from codec extradata | |
1760 */ | |
1761 static int qdm2_decode_init(AVCodecContext *avctx) | |
1762 { | |
1763 QDM2Context *s = avctx->priv_data; | |
1764 uint8_t *extradata; | |
1765 int extradata_size; | |
1766 int tmp_val, tmp, size; | |
1767 int i; | |
1768 float alpha; | |
2967 | 1769 |
2914 | 1770 /* extradata parsing |
2967 | 1771 |
2914 | 1772 Structure: |
1773 wave { | |
1774 frma (QDM2) | |
1775 QDCA | |
1776 QDCP | |
1777 } | |
2967 | 1778 |
2914 | 1779 32 size (including this field) |
1780 32 tag (=frma) | |
1781 32 type (=QDM2 or QDMC) | |
2967 | 1782 |
2914 | 1783 32 size (including this field, in bytes) |
1784 32 tag (=QDCA) // maybe mandatory parameters | |
1785 32 unknown (=1) | |
1786 32 channels (=2) | |
1787 32 samplerate (=44100) | |
1788 32 bitrate (=96000) | |
1789 32 block size (=4096) | |
1790 32 frame size (=256) (for one channel) | |
1791 32 packet size (=1300) | |
2967 | 1792 |
2914 | 1793 32 size (including this field, in bytes) |
1794 32 tag (=QDCP) // maybe some tuneable parameters | |
1795 32 float1 (=1.0) | |
1796 32 zero ? | |
1797 32 float2 (=1.0) | |
1798 32 float3 (=1.0) | |
1799 32 unknown (27) | |
1800 32 unknown (8) | |
1801 32 zero ? | |
1802 */ | |
1803 | |
1804 if (!avctx->extradata || (avctx->extradata_size < 48)) { | |
1805 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); | |
1806 return -1; | |
1807 } | |
1808 | |
1809 extradata = avctx->extradata; | |
1810 extradata_size = avctx->extradata_size; | |
1811 | |
1812 while (extradata_size > 7) { | |
1813 if (!memcmp(extradata, "frmaQDM", 7)) | |
1814 break; | |
1815 extradata++; | |
1816 extradata_size--; | |
1817 } | |
1818 | |
1819 if (extradata_size < 12) { | |
1820 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", | |
1821 extradata_size); | |
1822 return -1; | |
1823 } | |
1824 | |
1825 if (memcmp(extradata, "frmaQDM", 7)) { | |
1826 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); | |
1827 return -1; | |
1828 } | |
1829 | |
1830 if (extradata[7] == 'C') { | |
1831 // s->is_qdmc = 1; | |
1832 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); | |
1833 return -1; | |
1834 } | |
1835 | |
1836 extradata += 8; | |
1837 extradata_size -= 8; | |
1838 | |
4364 | 1839 size = AV_RB32(extradata); |
2914 | 1840 |
1841 if(size > extradata_size){ | |
1842 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", | |
1843 extradata_size, size); | |
1844 return -1; | |
1845 } | |
1846 | |
1847 extradata += 4; | |
1848 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); | |
4364 | 1849 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { |
2914 | 1850 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); |
1851 return -1; | |
1852 } | |
1853 | |
1854 extradata += 8; | |
1855 | |
4364 | 1856 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); |
2914 | 1857 extradata += 4; |
1858 | |
4364 | 1859 avctx->sample_rate = AV_RB32(extradata); |
2914 | 1860 extradata += 4; |
1861 | |
4364 | 1862 avctx->bit_rate = AV_RB32(extradata); |
2914 | 1863 extradata += 4; |
1864 | |
4364 | 1865 s->group_size = AV_RB32(extradata); |
2914 | 1866 extradata += 4; |
1867 | |
4364 | 1868 s->fft_size = AV_RB32(extradata); |
2914 | 1869 extradata += 4; |
1870 | |
4364 | 1871 s->checksum_size = AV_RB32(extradata); |
2914 | 1872 extradata += 4; |
1873 | |
1874 s->fft_order = av_log2(s->fft_size) + 1; | |
1875 s->fft_frame_size = 2 * s->fft_size; // complex has two floats | |
1876 | |
1877 // something like max decodable tones | |
1878 s->group_order = av_log2(s->group_size) + 1; | |
1879 s->frame_size = s->group_size / 16; // 16 iterations per super block | |
1880 | |
2954 | 1881 s->sub_sampling = s->fft_order - 7; |
2914 | 1882 s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); |
2967 | 1883 |
2914 | 1884 switch ((s->sub_sampling * 2 + s->channels - 1)) { |
1885 case 0: tmp = 40; break; | |
1886 case 1: tmp = 48; break; | |
1887 case 2: tmp = 56; break; | |
1888 case 3: tmp = 72; break; | |
1889 case 4: tmp = 80; break; | |
1890 case 5: tmp = 100;break; | |
1891 default: tmp=s->sub_sampling; break; | |
1892 } | |
1893 tmp_val = 0; | |
1894 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; | |
1895 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; | |
1896 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; | |
1897 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; | |
1898 s->cm_table_select = tmp_val; | |
1899 | |
1900 if (s->sub_sampling == 0) | |
2954 | 1901 tmp = 7999; |
2914 | 1902 else |
1903 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; | |
1904 /* | |
2954 | 1905 0: 7999 -> 0 |
2914 | 1906 1: 20000 -> 2 |
1907 2: 28000 -> 2 | |
1908 */ | |
1909 if (tmp < 8000) | |
1910 s->coeff_per_sb_select = 0; | |
1911 else if (tmp <= 16000) | |
1912 s->coeff_per_sb_select = 1; | |
1913 else | |
1914 s->coeff_per_sb_select = 2; | |
1915 | |
2954 | 1916 // Fail on unknown fft order, if it's > 9 it can overflow s->exptab[] |
1917 if ((s->fft_order < 7) || (s->fft_order > 9)) { | |
2914 | 1918 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); |
2954 | 1919 return -1; |
1920 } | |
2914 | 1921 |
1922 ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1); | |
1923 | |
1924 for (i = 1; i < (1 << (s->fft_order - 2)); i++) { | |
1925 alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1)); | |
1926 s->exptab[i].re = cos(alpha); | |
1927 s->exptab[i].im = sin(alpha); | |
1928 } | |
1929 | |
1930 qdm2_init(s); | |
2967 | 1931 |
2914 | 1932 // dump_context(s); |
1933 return 0; | |
1934 } | |
1935 | |
1936 | |
1937 static int qdm2_decode_close(AVCodecContext *avctx) | |
1938 { | |
1939 QDM2Context *s = avctx->priv_data; | |
1940 | |
1941 ff_fft_end(&s->fft_ctx); | |
2967 | 1942 |
2914 | 1943 return 0; |
1944 } | |
1945 | |
1946 | |
3076 | 1947 static void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out) |
2914 | 1948 { |
1949 int ch, i; | |
1950 const int frame_size = (q->frame_size * q->channels); | |
2967 | 1951 |
2914 | 1952 /* select input buffer */ |
1953 q->compressed_data = in; | |
1954 q->compressed_size = q->checksum_size; | |
1955 | |
1956 // dump_context(q); | |
1957 | |
1958 /* copy old block, clear new block of output samples */ | |
1959 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); | |
1960 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); | |
1961 | |
1962 /* decode block of QDM2 compressed data */ | |
1963 if (q->sub_packet == 0) { | |
1964 q->has_errors = 0; // zero it for a new super block | |
3043
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changeset
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1965 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); |
2914 | 1966 qdm2_decode_super_block(q); |
1967 } | |
1968 | |
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diego
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changeset
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1969 /* parse subpackets */ |
2914 | 1970 if (!q->has_errors) { |
1971 if (q->sub_packet == 2) | |
1972 qdm2_decode_fft_packets(q); | |
1973 | |
1974 qdm2_fft_tone_synthesizer(q, q->sub_packet); | |
1975 } | |
1976 | |
1977 /* sound synthesis stage 1 (FFT) */ | |
1978 for (ch = 0; ch < q->channels; ch++) { | |
1979 qdm2_calculate_fft(q, ch, q->sub_packet); | |
1980 | |
1981 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { | |
1982 SAMPLES_NEEDED_2("has errors, and C list is not empty") | |
1983 return; | |
1984 } | |
1985 } | |
1986 | |
1987 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ | |
1988 if (!q->has_errors && q->do_synth_filter) | |
1989 qdm2_synthesis_filter(q, q->sub_packet); | |
1990 | |
1991 q->sub_packet = (q->sub_packet + 1) % 16; | |
1992 | |
1993 /* clip and convert output float[] to 16bit signed samples */ | |
1994 for (i = 0; i < frame_size; i++) { | |
1995 int value = (int)q->output_buffer[i]; | |
1996 | |
1997 if (value > SOFTCLIP_THRESHOLD) | |
1998 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; | |
1999 else if (value < -SOFTCLIP_THRESHOLD) | |
2000 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; | |
2001 | |
2002 out[i] = value; | |
2003 } | |
2004 } | |
2005 | |
2006 | |
2007 static int qdm2_decode_frame(AVCodecContext *avctx, | |
2008 void *data, int *data_size, | |
2009 uint8_t *buf, int buf_size) | |
2010 { | |
2011 QDM2Context *s = avctx->priv_data; | |
2012 | |
3158 | 2013 if(!buf) |
2914 | 2014 return 0; |
3158 | 2015 if(buf_size < s->checksum_size) |
2016 return -1; | |
2914 | 2017 |
2018 *data_size = s->channels * s->frame_size * sizeof(int16_t); | |
2019 | |
2020 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", | |
2021 buf_size, buf, s->checksum_size, data, *data_size); | |
2022 | |
2023 qdm2_decode(s, buf, data); | |
2024 | |
2025 // reading only when next superblock found | |
2026 if (s->sub_packet == 0) { | |
2027 return s->checksum_size; | |
2028 } | |
2029 | |
2030 return 0; | |
2031 } | |
2032 | |
2033 AVCodec qdm2_decoder = | |
2034 { | |
2035 .name = "qdm2", | |
2036 .type = CODEC_TYPE_AUDIO, | |
2037 .id = CODEC_ID_QDM2, | |
2038 .priv_data_size = sizeof(QDM2Context), | |
2039 .init = qdm2_decode_init, | |
2040 .close = qdm2_decode_close, | |
2041 .decode = qdm2_decode_frame, | |
2042 }; |