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1 | /* | ||
2 | * QDM2 compatible decoder | ||
3 | * Copyright (c) 2003 Ewald Snel | ||
4 | * Copyright (c) 2005 Benjamin Larsson | ||
5 | * Copyright (c) 2005 Alex Beregszaszi | ||
6 | * Copyright (c) 2005 Roberto Togni | ||
7 | * | ||
8 | * This file is part of FFmpeg. | ||
9 | * | ||
10 | * FFmpeg is free software; you can redistribute it and/or | ||
11 | * modify it under the terms of the GNU Lesser General Public | ||
12 | * License as published by the Free Software Foundation; either | ||
13 | * version 2.1 of the License, or (at your option) any later version. | ||
14 | * | ||
15 | * FFmpeg is distributed in the hope that it will be useful, | ||
16 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | ||
17 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | ||
18 | * Lesser General Public License for more details. | ||
19 | * | ||
20 | * You should have received a copy of the GNU Lesser General Public | ||
21 | * License along with FFmpeg; if not, write to the Free Software | ||
22 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | ||
23 | */ | ||
24 | |||
25 | /** | ||
26 | * @file | ||
27 | * QDM2 decoder | ||
28 | * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni | ||
29 | * | ||
30 | * The decoder is not perfect yet, there are still some distortions | ||
31 | * especially on files encoded with 16 or 8 subbands. | ||
32 | */ | ||
33 | |||
34 | #include <math.h> | ||
35 | #include <stddef.h> | ||
36 | |||
37 | #include "libavutil/channel_layout.h" | ||
38 | #include "libavutil/mem_internal.h" | ||
39 | #include "libavutil/thread.h" | ||
40 | #include "libavutil/tx.h" | ||
41 | |||
42 | #define BITSTREAM_READER_LE | ||
43 | #include "avcodec.h" | ||
44 | #include "get_bits.h" | ||
45 | #include "bytestream.h" | ||
46 | #include "codec_internal.h" | ||
47 | #include "decode.h" | ||
48 | #include "mpegaudio.h" | ||
49 | #include "mpegaudiodsp.h" | ||
50 | |||
51 | #include "qdm2_tablegen.h" | ||
52 | |||
53 | #define QDM2_LIST_ADD(list, size, packet) \ | ||
54 | do { \ | ||
55 | if (size > 0) { \ | ||
56 | list[size - 1].next = &list[size]; \ | ||
57 | } \ | ||
58 | list[size].packet = packet; \ | ||
59 | list[size].next = NULL; \ | ||
60 | size++; \ | ||
61 | } while(0) | ||
62 | |||
63 | // Result is 8, 16 or 30 | ||
64 | #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling)) | ||
65 | |||
66 | #define FIX_NOISE_IDX(noise_idx) \ | ||
67 | if ((noise_idx) >= 3840) \ | ||
68 | (noise_idx) -= 3840; \ | ||
69 | |||
70 | #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)]) | ||
71 | |||
72 | #define SAMPLES_NEEDED \ | ||
73 | av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); | ||
74 | |||
75 | #define SAMPLES_NEEDED_2(why) \ | ||
76 | av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); | ||
77 | |||
78 | #define QDM2_MAX_FRAME_SIZE 512 | ||
79 | |||
80 | typedef int8_t sb_int8_array[2][30][64]; | ||
81 | |||
82 | /** | ||
83 | * Subpacket | ||
84 | */ | ||
85 | typedef struct QDM2SubPacket { | ||
86 | int type; ///< subpacket type | ||
87 | unsigned int size; ///< subpacket size | ||
88 | const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) | ||
89 | } QDM2SubPacket; | ||
90 | |||
91 | /** | ||
92 | * A node in the subpacket list | ||
93 | */ | ||
94 | typedef struct QDM2SubPNode { | ||
95 | QDM2SubPacket *packet; ///< packet | ||
96 | struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node | ||
97 | } QDM2SubPNode; | ||
98 | |||
99 | typedef struct FFTTone { | ||
100 | float level; | ||
101 | AVComplexFloat *complex; | ||
102 | const float *table; | ||
103 | int phase; | ||
104 | int phase_shift; | ||
105 | int duration; | ||
106 | short time_index; | ||
107 | short cutoff; | ||
108 | } FFTTone; | ||
109 | |||
110 | typedef struct FFTCoefficient { | ||
111 | int16_t sub_packet; | ||
112 | uint8_t channel; | ||
113 | int16_t offset; | ||
114 | int16_t exp; | ||
115 | uint8_t phase; | ||
116 | } FFTCoefficient; | ||
117 | |||
118 | typedef struct QDM2FFT { | ||
119 | DECLARE_ALIGNED(32, AVComplexFloat, complex)[MPA_MAX_CHANNELS][256 + 1]; | ||
120 | DECLARE_ALIGNED(32, AVComplexFloat, temp)[MPA_MAX_CHANNELS][256]; | ||
121 | } QDM2FFT; | ||
122 | |||
123 | /** | ||
124 | * QDM2 decoder context | ||
125 | */ | ||
126 | typedef struct QDM2Context { | ||
127 | /// Parameters from codec header, do not change during playback | ||
128 | int nb_channels; ///< number of channels | ||
129 | int channels; ///< number of channels | ||
130 | int group_size; ///< size of frame group (16 frames per group) | ||
131 | int fft_size; ///< size of FFT, in complex numbers | ||
132 | int checksum_size; ///< size of data block, used also for checksum | ||
133 | |||
134 | /// Parameters built from header parameters, do not change during playback | ||
135 | int group_order; ///< order of frame group | ||
136 | int fft_order; ///< order of FFT (actually fftorder+1) | ||
137 | int frame_size; ///< size of data frame | ||
138 | int frequency_range; | ||
139 | int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ | ||
140 | int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 | ||
141 | int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) | ||
142 | |||
143 | /// Packets and packet lists | ||
144 | QDM2SubPacket sub_packets[16]; ///< the packets themselves | ||
145 | QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets | ||
146 | QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list | ||
147 | int sub_packets_B; ///< number of packets on 'B' list | ||
148 | QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? | ||
149 | QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets | ||
150 | |||
151 | /// FFT and tones | ||
152 | FFTTone fft_tones[1000]; | ||
153 | int fft_tone_start; | ||
154 | int fft_tone_end; | ||
155 | FFTCoefficient fft_coefs[1000]; | ||
156 | int fft_coefs_index; | ||
157 | int fft_coefs_min_index[5]; | ||
158 | int fft_coefs_max_index[5]; | ||
159 | int fft_level_exp[6]; | ||
160 | AVTXContext *rdft_ctx; | ||
161 | av_tx_fn rdft_fn; | ||
162 | QDM2FFT fft; | ||
163 | |||
164 | /// I/O data | ||
165 | const uint8_t *compressed_data; | ||
166 | int compressed_size; | ||
167 | float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2]; | ||
168 | |||
169 | /// Synthesis filter | ||
170 | MPADSPContext mpadsp; | ||
171 | DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2]; | ||
172 | int synth_buf_offset[MPA_MAX_CHANNELS]; | ||
173 | DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; | ||
174 | DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; | ||
175 | |||
176 | /// Mixed temporary data used in decoding | ||
177 | float tone_level[MPA_MAX_CHANNELS][30][64]; | ||
178 | int8_t coding_method[MPA_MAX_CHANNELS][30][64]; | ||
179 | int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; | ||
180 | int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; | ||
181 | int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; | ||
182 | int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; | ||
183 | int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; | ||
184 | int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; | ||
185 | int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; | ||
186 | |||
187 | // Flags | ||
188 | int has_errors; ///< packet has errors | ||
189 | int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type | ||
190 | int do_synth_filter; ///< used to perform or skip synthesis filter | ||
191 | |||
192 | int sub_packet; | ||
193 | int noise_idx; ///< index for dithering noise table | ||
194 | } QDM2Context; | ||
195 | |||
196 | static const int switchtable[23] = { | ||
197 | 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4 | ||
198 | }; | ||
199 | |||
200 | 67959 | static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth) | |
201 | { | ||
202 | int value; | ||
203 | |||
204 | 67959 | value = get_vlc2(gb, vlc->table, vlc->bits, depth); | |
205 | |||
206 | /* stage-2, 3 bits exponent escape sequence */ | ||
207 |
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67959 | if (value < 0) |
208 | 558 | value = get_bits(gb, get_bits(gb, 3) + 1); | |
209 | |||
210 | /* stage-3, optional */ | ||
211 |
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67959 | if (flag) { |
212 | int tmp; | ||
213 | |||
214 |
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18368 | if (value >= 60) { |
215 | ✗ | av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value); | |
216 | ✗ | return 0; | |
217 | } | ||
218 | |||
219 | 18368 | tmp= vlc_stage3_values[value]; | |
220 | |||
221 |
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18368 | if ((value & ~3) > 0) |
222 | 14255 | tmp += get_bits(gb, (value >> 2)); | |
223 | 18368 | value = tmp; | |
224 | } | ||
225 | |||
226 | 67959 | return value; | |
227 | } | ||
228 | |||
229 | 11196 | static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth) | |
230 | { | ||
231 | 11196 | int value = qdm2_get_vlc(gb, vlc, 0, depth); | |
232 | |||
233 |
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11196 | return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); |
234 | } | ||
235 | |||
236 | /** | ||
237 | * QDM2 checksum | ||
238 | * | ||
239 | * @param data pointer to data to be checksummed | ||
240 | * @param length data length | ||
241 | * @param value checksum value | ||
242 | * | ||
243 | * @return 0 if checksum is OK | ||
244 | */ | ||
245 | 141 | static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value) | |
246 | { | ||
247 | int i; | ||
248 | |||
249 |
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52311 | for (i = 0; i < length; i++) |
250 | 52170 | value -= data[i]; | |
251 | |||
252 | 141 | return (uint16_t)(value & 0xffff); | |
253 | } | ||
254 | |||
255 | /** | ||
256 | * Fill a QDM2SubPacket structure with packet type, size, and data pointer. | ||
257 | * | ||
258 | * @param gb bitreader context | ||
259 | * @param sub_packet packet under analysis | ||
260 | */ | ||
261 | 1128 | static void qdm2_decode_sub_packet_header(GetBitContext *gb, | |
262 | QDM2SubPacket *sub_packet) | ||
263 | { | ||
264 | 1128 | sub_packet->type = get_bits(gb, 8); | |
265 | |||
266 |
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1128 | if (sub_packet->type == 0) { |
267 | 141 | sub_packet->size = 0; | |
268 | 141 | sub_packet->data = NULL; | |
269 | } else { | ||
270 | 987 | sub_packet->size = get_bits(gb, 8); | |
271 | |||
272 |
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987 | if (sub_packet->type & 0x80) { |
273 | 141 | sub_packet->size <<= 8; | |
274 | 141 | sub_packet->size |= get_bits(gb, 8); | |
275 | 141 | sub_packet->type &= 0x7f; | |
276 | } | ||
277 | |||
278 |
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987 | if (sub_packet->type == 0x7f) |
279 | ✗ | sub_packet->type |= (get_bits(gb, 8) << 8); | |
280 | |||
281 | // FIXME: this depends on bitreader-internal data | ||
282 | 987 | sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; | |
283 | } | ||
284 | |||
285 | 1128 | av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n", | |
286 | 1128 | sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); | |
287 | 1128 | } | |
288 | |||
289 | /** | ||
290 | * Return node pointer to first packet of requested type in list. | ||
291 | * | ||
292 | * @param list list of subpackets to be scanned | ||
293 | * @param type type of searched subpacket | ||
294 | * @return node pointer for subpacket if found, else NULL | ||
295 | */ | ||
296 | 564 | static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list, | |
297 | int type) | ||
298 | { | ||
299 |
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987 | while (list && list->packet) { |
300 |
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564 | if (list->packet->type == type) |
301 | 141 | return list; | |
302 | 423 | list = list->next; | |
303 | } | ||
304 | 423 | return NULL; | |
305 | } | ||
306 | |||
307 | /** | ||
308 | * Replace 8 elements with their average value. | ||
309 | * Called by qdm2_decode_superblock before starting subblock decoding. | ||
310 | * | ||
311 | * @param q context | ||
312 | */ | ||
313 | 141 | static void average_quantized_coeffs(QDM2Context *q) | |
314 | { | ||
315 | int i, j, n, ch, sum; | ||
316 | |||
317 |
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141 | n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; |
318 | |||
319 |
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423 | for (ch = 0; ch < q->nb_channels; ch++) |
320 |
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3102 | for (i = 0; i < n; i++) { |
321 | 2820 | sum = 0; | |
322 | |||
323 |
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25380 | for (j = 0; j < 8; j++) |
324 | 22560 | sum += q->quantized_coeffs[ch][i][j]; | |
325 | |||
326 | 2820 | sum /= 8; | |
327 |
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2820 | if (sum > 0) |
328 | 2466 | sum--; | |
329 | |||
330 |
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25380 | for (j = 0; j < 8; j++) |
331 | 22560 | q->quantized_coeffs[ch][i][j] = sum; | |
332 | } | ||
333 | 141 | } | |
334 | |||
335 | /** | ||
336 | * Build subband samples with noise weighted by q->tone_level. | ||
337 | * Called by synthfilt_build_sb_samples. | ||
338 | * | ||
339 | * @param q context | ||
340 | * @param sb subband index | ||
341 | */ | ||
342 | 4230 | static void build_sb_samples_from_noise(QDM2Context *q, int sb) | |
343 | { | ||
344 | int ch, j; | ||
345 | |||
346 |
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4230 | FIX_NOISE_IDX(q->noise_idx); |
347 | |||
348 |
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4230 | if (!q->nb_channels) |
349 | ✗ | return; | |
350 | |||
351 |
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12690 | for (ch = 0; ch < q->nb_channels; ch++) { |
352 |
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549900 | for (j = 0; j < 64; j++) { |
353 | 541440 | q->sb_samples[ch][j * 2][sb] = | |
354 | 541440 | SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j]; | |
355 | 541440 | q->sb_samples[ch][j * 2 + 1][sb] = | |
356 | 541440 | SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j]; | |
357 | } | ||
358 | } | ||
359 | } | ||
360 | |||
361 | /** | ||
362 | * Called while processing data from subpackets 11 and 12. | ||
363 | * Used after making changes to coding_method array. | ||
364 | * | ||
365 | * @param sb subband index | ||
366 | * @param channels number of channels | ||
367 | * @param coding_method q->coding_method[0][0][0] | ||
368 | */ | ||
369 | ✗ | static int fix_coding_method_array(int sb, int channels, | |
370 | sb_int8_array coding_method) | ||
371 | { | ||
372 | int j, k; | ||
373 | int ch; | ||
374 | int run, case_val; | ||
375 | |||
376 | ✗ | for (ch = 0; ch < channels; ch++) { | |
377 | ✗ | for (j = 0; j < 64; ) { | |
378 | ✗ | if (coding_method[ch][sb][j] < 8) | |
379 | ✗ | return -1; | |
380 | ✗ | if ((coding_method[ch][sb][j] - 8) > 22) { | |
381 | ✗ | run = 1; | |
382 | ✗ | case_val = 8; | |
383 | } else { | ||
384 | ✗ | switch (switchtable[coding_method[ch][sb][j] - 8]) { | |
385 | ✗ | case 0: run = 10; | |
386 | ✗ | case_val = 10; | |
387 | ✗ | break; | |
388 | ✗ | case 1: run = 1; | |
389 | ✗ | case_val = 16; | |
390 | ✗ | break; | |
391 | ✗ | case 2: run = 5; | |
392 | ✗ | case_val = 24; | |
393 | ✗ | break; | |
394 | ✗ | case 3: run = 3; | |
395 | ✗ | case_val = 30; | |
396 | ✗ | break; | |
397 | ✗ | case 4: run = 1; | |
398 | ✗ | case_val = 30; | |
399 | ✗ | break; | |
400 | ✗ | case 5: run = 1; | |
401 | ✗ | case_val = 8; | |
402 | ✗ | break; | |
403 | ✗ | default: run = 1; | |
404 | ✗ | case_val = 8; | |
405 | ✗ | break; | |
406 | } | ||
407 | } | ||
408 | ✗ | for (k = 0; k < run; k++) { | |
409 | ✗ | if (j + k < 128) { | |
410 | ✗ | int sbjk = sb + (j + k) / 64; | |
411 | ✗ | if (sbjk > 29) { | |
412 | ✗ | SAMPLES_NEEDED | |
413 | ✗ | continue; | |
414 | } | ||
415 | ✗ | if (coding_method[ch][sbjk][(j + k) % 64] > coding_method[ch][sb][j]) { | |
416 | ✗ | if (k > 0) { | |
417 | ✗ | SAMPLES_NEEDED | |
418 | //not debugged, almost never used | ||
419 | ✗ | memset(&coding_method[ch][sb][j + k], case_val, | |
420 | k *sizeof(int8_t)); | ||
421 | ✗ | memset(&coding_method[ch][sb][j + k], case_val, | |
422 | 3 * sizeof(int8_t)); | ||
423 | } | ||
424 | } | ||
425 | } | ||
426 | } | ||
427 | ✗ | j += run; | |
428 | } | ||
429 | } | ||
430 | ✗ | return 0; | |
431 | } | ||
432 | |||
433 | /** | ||
434 | * Related to synthesis filter | ||
435 | * Called by process_subpacket_10 | ||
436 | * | ||
437 | * @param q context | ||
438 | * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 | ||
439 | */ | ||
440 | 141 | static void fill_tone_level_array(QDM2Context *q, int flag) | |
441 | { | ||
442 | int i, sb, ch, sb_used; | ||
443 | int tmp, tab; | ||
444 | |||
445 |
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423 | for (ch = 0; ch < q->nb_channels; ch++) |
446 |
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8742 | for (sb = 0; sb < 30; sb++) |
447 |
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76140 | for (i = 0; i < 8; i++) { |
448 |
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67680 | if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) |
449 | 54144 | tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ | |
450 | 54144 | q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; | |
451 | else | ||
452 | 13536 | tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; | |
453 |
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67680 | if(tmp < 0) |
454 | ✗ | tmp += 0xff; | |
455 | 67680 | q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; | |
456 | } | ||
457 | |||
458 |
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141 | sb_used = QDM2_SB_USED(q->sub_sampling); |
459 | |||
460 |
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141 | if ((q->superblocktype_2_3 != 0) && !flag) { |
461 |
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4371 | for (sb = 0; sb < sb_used; sb++) |
462 |
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12690 | for (ch = 0; ch < q->nb_channels; ch++) |
463 |
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549900 | for (i = 0; i < 64; i++) { |
464 | 541440 | q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; | |
465 |
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541440 | if (q->tone_level_idx[ch][sb][i] < 0) |
466 | ✗ | q->tone_level[ch][sb][i] = 0; | |
467 | else | ||
468 | 541440 | q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; | |
469 | } | ||
470 | } else { | ||
471 | ✗ | tab = q->superblocktype_2_3 ? 0 : 1; | |
472 | ✗ | for (sb = 0; sb < sb_used; sb++) { | |
473 | ✗ | if ((sb >= 4) && (sb <= 23)) { | |
474 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) | |
475 | ✗ | for (i = 0; i < 64; i++) { | |
476 | ✗ | tmp = q->tone_level_idx_base[ch][sb][i / 8] - | |
477 | ✗ | q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - | |
478 | ✗ | q->tone_level_idx_mid[ch][sb - 4][i / 8] - | |
479 | ✗ | q->tone_level_idx_hi2[ch][sb - 4]; | |
480 | ✗ | q->tone_level_idx[ch][sb][i] = tmp & 0xff; | |
481 | ✗ | if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
482 | ✗ | q->tone_level[ch][sb][i] = 0; | |
483 | else | ||
484 | ✗ | q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
485 | } | ||
486 | } else { | ||
487 | ✗ | if (sb > 4) { | |
488 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) | |
489 | ✗ | for (i = 0; i < 64; i++) { | |
490 | ✗ | tmp = q->tone_level_idx_base[ch][sb][i / 8] - | |
491 | ✗ | q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - | |
492 | ✗ | q->tone_level_idx_hi2[ch][sb - 4]; | |
493 | ✗ | q->tone_level_idx[ch][sb][i] = tmp & 0xff; | |
494 | ✗ | if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
495 | ✗ | q->tone_level[ch][sb][i] = 0; | |
496 | else | ||
497 | ✗ | q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
498 | } | ||
499 | } else { | ||
500 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) | |
501 | ✗ | for (i = 0; i < 64; i++) { | |
502 | ✗ | tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; | |
503 | ✗ | if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
504 | ✗ | q->tone_level[ch][sb][i] = 0; | |
505 | else | ||
506 | ✗ | q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
507 | } | ||
508 | } | ||
509 | } | ||
510 | } | ||
511 | } | ||
512 | 141 | } | |
513 | |||
514 | /** | ||
515 | * Related to synthesis filter | ||
516 | * Called by process_subpacket_11 | ||
517 | * c is built with data from subpacket 11 | ||
518 | * Most of this function is used only if superblock_type_2_3 == 0, | ||
519 | * never seen it in samples. | ||
520 | * | ||
521 | * @param tone_level_idx | ||
522 | * @param tone_level_idx_temp | ||
523 | * @param coding_method q->coding_method[0][0][0] | ||
524 | * @param nb_channels number of channels | ||
525 | * @param c coming from subpacket 11, passed as 8*c | ||
526 | * @param superblocktype_2_3 flag based on superblock packet type | ||
527 | * @param cm_table_select q->cm_table_select | ||
528 | */ | ||
529 | ✗ | static void fill_coding_method_array(sb_int8_array tone_level_idx, | |
530 | sb_int8_array tone_level_idx_temp, | ||
531 | sb_int8_array coding_method, | ||
532 | int nb_channels, | ||
533 | int c, int superblocktype_2_3, | ||
534 | int cm_table_select) | ||
535 | { | ||
536 | int ch, sb, j; | ||
537 | int tmp, acc, esp_40, comp; | ||
538 | int add1, add2, add3, add4; | ||
539 | int64_t multres; | ||
540 | |||
541 | ✗ | if (!superblocktype_2_3) { | |
542 | /* This case is untested, no samples available */ | ||
543 | ✗ | avpriv_request_sample(NULL, "!superblocktype_2_3"); | |
544 | ✗ | return; | |
545 | for (ch = 0; ch < nb_channels; ch++) { | ||
546 | for (sb = 0; sb < 30; sb++) { | ||
547 | for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer | ||
548 | add1 = tone_level_idx[ch][sb][j] - 10; | ||
549 | if (add1 < 0) | ||
550 | add1 = 0; | ||
551 | add2 = add3 = add4 = 0; | ||
552 | if (sb > 1) { | ||
553 | add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; | ||
554 | if (add2 < 0) | ||
555 | add2 = 0; | ||
556 | } | ||
557 | if (sb > 0) { | ||
558 | add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; | ||
559 | if (add3 < 0) | ||
560 | add3 = 0; | ||
561 | } | ||
562 | if (sb < 29) { | ||
563 | add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; | ||
564 | if (add4 < 0) | ||
565 | add4 = 0; | ||
566 | } | ||
567 | tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; | ||
568 | if (tmp < 0) | ||
569 | tmp = 0; | ||
570 | tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; | ||
571 | } | ||
572 | tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; | ||
573 | } | ||
574 | } | ||
575 | acc = 0; | ||
576 | for (ch = 0; ch < nb_channels; ch++) | ||
577 | for (sb = 0; sb < 30; sb++) | ||
578 | for (j = 0; j < 64; j++) | ||
579 | acc += tone_level_idx_temp[ch][sb][j]; | ||
580 | |||
581 | multres = 0x66666667LL * (acc * 10); | ||
582 | esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); | ||
583 | for (ch = 0; ch < nb_channels; ch++) | ||
584 | for (sb = 0; sb < 30; sb++) | ||
585 | for (j = 0; j < 64; j++) { | ||
586 | comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; | ||
587 | if (comp < 0) | ||
588 | comp += 0xff; | ||
589 | comp /= 256; // signed shift | ||
590 | switch(sb) { | ||
591 | case 0: | ||
592 | if (comp < 30) | ||
593 | comp = 30; | ||
594 | comp += 15; | ||
595 | break; | ||
596 | case 1: | ||
597 | if (comp < 24) | ||
598 | comp = 24; | ||
599 | comp += 10; | ||
600 | break; | ||
601 | case 2: | ||
602 | case 3: | ||
603 | case 4: | ||
604 | if (comp < 16) | ||
605 | comp = 16; | ||
606 | } | ||
607 | if (comp <= 5) | ||
608 | tmp = 0; | ||
609 | else if (comp <= 10) | ||
610 | tmp = 10; | ||
611 | else if (comp <= 16) | ||
612 | tmp = 16; | ||
613 | else if (comp <= 24) | ||
614 | tmp = -1; | ||
615 | else | ||
616 | tmp = 0; | ||
617 | coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; | ||
618 | } | ||
619 | for (sb = 0; sb < 30; sb++) | ||
620 | fix_coding_method_array(sb, nb_channels, coding_method); | ||
621 | for (ch = 0; ch < nb_channels; ch++) | ||
622 | for (sb = 0; sb < 30; sb++) | ||
623 | for (j = 0; j < 64; j++) | ||
624 | if (sb >= 10) { | ||
625 | if (coding_method[ch][sb][j] < 10) | ||
626 | coding_method[ch][sb][j] = 10; | ||
627 | } else { | ||
628 | if (sb >= 2) { | ||
629 | if (coding_method[ch][sb][j] < 16) | ||
630 | coding_method[ch][sb][j] = 16; | ||
631 | } else { | ||
632 | if (coding_method[ch][sb][j] < 30) | ||
633 | coding_method[ch][sb][j] = 30; | ||
634 | } | ||
635 | } | ||
636 | } else { // superblocktype_2_3 != 0 | ||
637 | ✗ | for (ch = 0; ch < nb_channels; ch++) | |
638 | ✗ | for (sb = 0; sb < 30; sb++) | |
639 | ✗ | for (j = 0; j < 64; j++) | |
640 | ✗ | coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; | |
641 | } | ||
642 | } | ||
643 | |||
644 | /** | ||
645 | * Called by process_subpacket_11 to process more data from subpacket 11 | ||
646 | * with sb 0-8. | ||
647 | * Called by process_subpacket_12 to process data from subpacket 12 with | ||
648 | * sb 8-sb_used. | ||
649 | * | ||
650 | * @param q context | ||
651 | * @param gb bitreader context | ||
652 | * @param length packet length in bits | ||
653 | * @param sb_min lower subband processed (sb_min included) | ||
654 | * @param sb_max higher subband processed (sb_max excluded) | ||
655 | */ | ||
656 | 282 | static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, | |
657 | int length, int sb_min, int sb_max) | ||
658 | { | ||
659 | int sb, j, k, n, ch, run, channels; | ||
660 | int joined_stereo, zero_encoding; | ||
661 | int type34_first; | ||
662 | 282 | float type34_div = 0; | |
663 | float type34_predictor; | ||
664 | float samples[10]; | ||
665 | 282 | int sign_bits[16] = {0}; | |
666 | |||
667 |
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282 | if (length == 0) { |
668 | // If no data use noise | ||
669 |
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4512 | for (sb=sb_min; sb < sb_max; sb++) |
670 | 4230 | build_sb_samples_from_noise(q, sb); | |
671 | |||
672 | 282 | return 0; | |
673 | } | ||
674 | |||
675 | ✗ | for (sb = sb_min; sb < sb_max; sb++) { | |
676 | ✗ | channels = q->nb_channels; | |
677 | |||
678 | ✗ | if (q->nb_channels <= 1 || sb < 12) | |
679 | ✗ | joined_stereo = 0; | |
680 | ✗ | else if (sb >= 24) | |
681 | ✗ | joined_stereo = 1; | |
682 | else | ||
683 | ✗ | joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0; | |
684 | |||
685 | ✗ | if (joined_stereo) { | |
686 | ✗ | if (get_bits_left(gb) >= 16) | |
687 | ✗ | for (j = 0; j < 16; j++) | |
688 | ✗ | sign_bits[j] = get_bits1(gb); | |
689 | |||
690 | ✗ | for (j = 0; j < 64; j++) | |
691 | ✗ | if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) | |
692 | ✗ | q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; | |
693 | |||
694 | ✗ | if (fix_coding_method_array(sb, q->nb_channels, | |
695 | ✗ | q->coding_method)) { | |
696 | ✗ | av_log(NULL, AV_LOG_ERROR, "coding method invalid\n"); | |
697 | ✗ | build_sb_samples_from_noise(q, sb); | |
698 | ✗ | continue; | |
699 | } | ||
700 | ✗ | channels = 1; | |
701 | } | ||
702 | |||
703 | ✗ | for (ch = 0; ch < channels; ch++) { | |
704 | ✗ | FIX_NOISE_IDX(q->noise_idx); | |
705 | ✗ | zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0; | |
706 | ✗ | type34_predictor = 0.0; | |
707 | ✗ | type34_first = 1; | |
708 | |||
709 | ✗ | for (j = 0; j < 128; ) { | |
710 | ✗ | switch (q->coding_method[ch][sb][j / 2]) { | |
711 | ✗ | case 8: | |
712 | ✗ | if (get_bits_left(gb) >= 10) { | |
713 | ✗ | if (zero_encoding) { | |
714 | ✗ | for (k = 0; k < 5; k++) { | |
715 | ✗ | if ((j + 2 * k) >= 128) | |
716 | ✗ | break; | |
717 | ✗ | samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; | |
718 | } | ||
719 | } else { | ||
720 | ✗ | n = get_bits(gb, 8); | |
721 | ✗ | if (n >= 243) { | |
722 | ✗ | av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); | |
723 | ✗ | return AVERROR_INVALIDDATA; | |
724 | } | ||
725 | |||
726 | ✗ | for (k = 0; k < 5; k++) | |
727 | ✗ | samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
728 | } | ||
729 | ✗ | for (k = 0; k < 5; k++) | |
730 | ✗ | samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
731 | } else { | ||
732 | ✗ | for (k = 0; k < 10; k++) | |
733 | ✗ | samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
734 | } | ||
735 | ✗ | run = 10; | |
736 | ✗ | break; | |
737 | |||
738 | ✗ | case 10: | |
739 | ✗ | if (get_bits_left(gb) >= 1) { | |
740 | ✗ | float f = 0.81; | |
741 | |||
742 | ✗ | if (get_bits1(gb)) | |
743 | ✗ | f = -f; | |
744 | ✗ | f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; | |
745 | ✗ | samples[0] = f; | |
746 | } else { | ||
747 | ✗ | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
748 | } | ||
749 | ✗ | run = 1; | |
750 | ✗ | break; | |
751 | |||
752 | ✗ | case 16: | |
753 | ✗ | if (get_bits_left(gb) >= 10) { | |
754 | ✗ | if (zero_encoding) { | |
755 | ✗ | for (k = 0; k < 5; k++) { | |
756 | ✗ | if ((j + k) >= 128) | |
757 | ✗ | break; | |
758 | ✗ | samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; | |
759 | } | ||
760 | } else { | ||
761 | ✗ | n = get_bits (gb, 8); | |
762 | ✗ | if (n >= 243) { | |
763 | ✗ | av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); | |
764 | ✗ | return AVERROR_INVALIDDATA; | |
765 | } | ||
766 | |||
767 | ✗ | for (k = 0; k < 5; k++) | |
768 | ✗ | samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
769 | } | ||
770 | } else { | ||
771 | ✗ | for (k = 0; k < 5; k++) | |
772 | ✗ | samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
773 | } | ||
774 | ✗ | run = 5; | |
775 | ✗ | break; | |
776 | |||
777 | ✗ | case 24: | |
778 | ✗ | if (get_bits_left(gb) >= 7) { | |
779 | ✗ | n = get_bits(gb, 7); | |
780 | ✗ | if (n >= 125) { | |
781 | ✗ | av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n"); | |
782 | ✗ | return AVERROR_INVALIDDATA; | |
783 | } | ||
784 | |||
785 | ✗ | for (k = 0; k < 3; k++) | |
786 | ✗ | samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; | |
787 | } else { | ||
788 | ✗ | for (k = 0; k < 3; k++) | |
789 | ✗ | samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
790 | } | ||
791 | ✗ | run = 3; | |
792 | ✗ | break; | |
793 | |||
794 | ✗ | case 30: | |
795 | ✗ | if (get_bits_left(gb) >= 4) { | |
796 | ✗ | unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1); | |
797 | ✗ | if (index >= FF_ARRAY_ELEMS(type30_dequant)) { | |
798 | ✗ | av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index); | |
799 | ✗ | return AVERROR_INVALIDDATA; | |
800 | } | ||
801 | ✗ | samples[0] = type30_dequant[index]; | |
802 | } else | ||
803 | ✗ | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
804 | |||
805 | ✗ | run = 1; | |
806 | ✗ | break; | |
807 | |||
808 | ✗ | case 34: | |
809 | ✗ | if (get_bits_left(gb) >= 7) { | |
810 | ✗ | if (type34_first) { | |
811 | ✗ | type34_div = (float)(1 << get_bits(gb, 2)); | |
812 | ✗ | samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; | |
813 | ✗ | type34_predictor = samples[0]; | |
814 | ✗ | type34_first = 0; | |
815 | } else { | ||
816 | ✗ | unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1); | |
817 | ✗ | if (index >= FF_ARRAY_ELEMS(type34_delta)) { | |
818 | ✗ | av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index); | |
819 | ✗ | return AVERROR_INVALIDDATA; | |
820 | } | ||
821 | ✗ | samples[0] = type34_delta[index] / type34_div + type34_predictor; | |
822 | ✗ | type34_predictor = samples[0]; | |
823 | } | ||
824 | } else { | ||
825 | ✗ | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
826 | } | ||
827 | ✗ | run = 1; | |
828 | ✗ | break; | |
829 | |||
830 | ✗ | default: | |
831 | ✗ | samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
832 | ✗ | run = 1; | |
833 | ✗ | break; | |
834 | } | ||
835 | |||
836 | ✗ | if (joined_stereo) { | |
837 | ✗ | for (k = 0; k < run && j + k < 128; k++) { | |
838 | ✗ | q->sb_samples[0][j + k][sb] = | |
839 | ✗ | q->tone_level[0][sb][(j + k) / 2] * samples[k]; | |
840 | ✗ | if (q->nb_channels == 2) { | |
841 | ✗ | if (sign_bits[(j + k) / 8]) | |
842 | ✗ | q->sb_samples[1][j + k][sb] = | |
843 | ✗ | q->tone_level[1][sb][(j + k) / 2] * -samples[k]; | |
844 | else | ||
845 | ✗ | q->sb_samples[1][j + k][sb] = | |
846 | ✗ | q->tone_level[1][sb][(j + k) / 2] * samples[k]; | |
847 | } | ||
848 | } | ||
849 | } else { | ||
850 | ✗ | for (k = 0; k < run; k++) | |
851 | ✗ | if ((j + k) < 128) | |
852 | ✗ | q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k]; | |
853 | } | ||
854 | |||
855 | ✗ | j += run; | |
856 | } // j loop | ||
857 | } // channel loop | ||
858 | } // subband loop | ||
859 | ✗ | return 0; | |
860 | } | ||
861 | |||
862 | /** | ||
863 | * Init the first element of a channel in quantized_coeffs with data | ||
864 | * from packet 10 (quantized_coeffs[ch][0]). | ||
865 | * This is similar to process_subpacket_9, but for a single channel | ||
866 | * and for element [0] | ||
867 | * same VLC tables as process_subpacket_9 are used. | ||
868 | * | ||
869 | * @param quantized_coeffs pointer to quantized_coeffs[ch][0] | ||
870 | * @param gb bitreader context | ||
871 | */ | ||
872 | ✗ | static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs, | |
873 | GetBitContext *gb) | ||
874 | { | ||
875 | int i, k, run, level, diff; | ||
876 | |||
877 | ✗ | if (get_bits_left(gb) < 16) | |
878 | ✗ | return -1; | |
879 | ✗ | level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); | |
880 | |||
881 | ✗ | quantized_coeffs[0] = level; | |
882 | |||
883 | ✗ | for (i = 0; i < 7; ) { | |
884 | ✗ | if (get_bits_left(gb) < 16) | |
885 | ✗ | return -1; | |
886 | ✗ | run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; | |
887 | |||
888 | ✗ | if (i + run >= 8) | |
889 | ✗ | return -1; | |
890 | |||
891 | ✗ | if (get_bits_left(gb) < 16) | |
892 | ✗ | return -1; | |
893 | ✗ | diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); | |
894 | |||
895 | ✗ | for (k = 1; k <= run; k++) | |
896 | ✗ | quantized_coeffs[i + k] = (level + ((k * diff) / run)); | |
897 | |||
898 | ✗ | level += diff; | |
899 | ✗ | i += run; | |
900 | } | ||
901 | ✗ | return 0; | |
902 | } | ||
903 | |||
904 | /** | ||
905 | * Related to synthesis filter, process data from packet 10 | ||
906 | * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 | ||
907 | * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with | ||
908 | * data from packet 10 | ||
909 | * | ||
910 | * @param q context | ||
911 | * @param gb bitreader context | ||
912 | */ | ||
913 | ✗ | static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb) | |
914 | { | ||
915 | int sb, j, k, n, ch; | ||
916 | |||
917 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) { | |
918 | ✗ | init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb); | |
919 | |||
920 | ✗ | if (get_bits_left(gb) < 16) { | |
921 | ✗ | memset(q->quantized_coeffs[ch][0], 0, 8); | |
922 | ✗ | break; | |
923 | } | ||
924 | } | ||
925 | |||
926 | ✗ | n = q->sub_sampling + 1; | |
927 | |||
928 | ✗ | for (sb = 0; sb < n; sb++) | |
929 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) | |
930 | ✗ | for (j = 0; j < 8; j++) { | |
931 | ✗ | if (get_bits_left(gb) < 1) | |
932 | ✗ | break; | |
933 | ✗ | if (get_bits1(gb)) { | |
934 | ✗ | for (k=0; k < 8; k++) { | |
935 | ✗ | if (get_bits_left(gb) < 16) | |
936 | ✗ | break; | |
937 | ✗ | q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); | |
938 | } | ||
939 | } else { | ||
940 | ✗ | for (k=0; k < 8; k++) | |
941 | ✗ | q->tone_level_idx_hi1[ch][sb][j][k] = 0; | |
942 | } | ||
943 | } | ||
944 | |||
945 | ✗ | n = QDM2_SB_USED(q->sub_sampling) - 4; | |
946 | |||
947 | ✗ | for (sb = 0; sb < n; sb++) | |
948 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) { | |
949 | ✗ | if (get_bits_left(gb) < 16) | |
950 | ✗ | break; | |
951 | ✗ | q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); | |
952 | ✗ | if (sb > 19) | |
953 | ✗ | q->tone_level_idx_hi2[ch][sb] -= 16; | |
954 | else | ||
955 | ✗ | for (j = 0; j < 8; j++) | |
956 | ✗ | q->tone_level_idx_mid[ch][sb][j] = -16; | |
957 | } | ||
958 | |||
959 | ✗ | n = QDM2_SB_USED(q->sub_sampling) - 5; | |
960 | |||
961 | ✗ | for (sb = 0; sb < n; sb++) | |
962 | ✗ | for (ch = 0; ch < q->nb_channels; ch++) | |
963 | ✗ | for (j = 0; j < 8; j++) { | |
964 | ✗ | if (get_bits_left(gb) < 16) | |
965 | ✗ | break; | |
966 | ✗ | q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; | |
967 | } | ||
968 | ✗ | } | |
969 | |||
970 | /** | ||
971 | * Process subpacket 9, init quantized_coeffs with data from it | ||
972 | * | ||
973 | * @param q context | ||
974 | * @param node pointer to node with packet | ||
975 | */ | ||
976 | 141 | static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node) | |
977 | { | ||
978 | GetBitContext gb; | ||
979 | int i, j, k, n, ch, run, level, diff; | ||
980 | |||
981 | 141 | init_get_bits(&gb, node->packet->data, node->packet->size * 8); | |
982 | |||
983 |
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141 | n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; |
984 | |||
985 |
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1410 | for (i = 1; i < n; i++) |
986 |
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3807 | for (ch = 0; ch < q->nb_channels; ch++) { |
987 | 2538 | level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); | |
988 | 2538 | q->quantized_coeffs[ch][i][0] = level; | |
989 | |||
990 |
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13734 | for (j = 0; j < (8 - 1); ) { |
991 | 11196 | run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; | |
992 | 11196 | diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); | |
993 | |||
994 |
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11196 | if (j + run >= 8) |
995 | ✗ | return -1; | |
996 | |||
997 |
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28962 | for (k = 1; k <= run; k++) |
998 | 17766 | q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run)); | |
999 | |||
1000 | 11196 | level += diff; | |
1001 | 11196 | j += run; | |
1002 | } | ||
1003 | } | ||
1004 | |||
1005 |
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423 | for (ch = 0; ch < q->nb_channels; ch++) |
1006 |
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2538 | for (i = 0; i < 8; i++) |
1007 | 2256 | q->quantized_coeffs[ch][0][i] = 0; | |
1008 | |||
1009 | 141 | return 0; | |
1010 | } | ||
1011 | |||
1012 | /** | ||
1013 | * Process subpacket 10 if not null, else | ||
1014 | * | ||
1015 | * @param q context | ||
1016 | * @param node pointer to node with packet | ||
1017 | */ | ||
1018 | 141 | static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node) | |
1019 | { | ||
1020 | GetBitContext gb; | ||
1021 | |||
1022 |
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|
141 | if (node) { |
1023 | ✗ | init_get_bits(&gb, node->packet->data, node->packet->size * 8); | |
1024 | ✗ | init_tone_level_dequantization(q, &gb); | |
1025 | ✗ | fill_tone_level_array(q, 1); | |
1026 | } else { | ||
1027 | 141 | fill_tone_level_array(q, 0); | |
1028 | } | ||
1029 | 141 | } | |
1030 | |||
1031 | /** | ||
1032 | * Process subpacket 11 | ||
1033 | * | ||
1034 | * @param q context | ||
1035 | * @param node pointer to node with packet | ||
1036 | */ | ||
1037 | 141 | static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node) | |
1038 | { | ||
1039 | GetBitContext gb; | ||
1040 | 141 | int length = 0; | |
1041 | |||
1042 |
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|
141 | if (node) { |
1043 | ✗ | length = node->packet->size * 8; | |
1044 | ✗ | init_get_bits(&gb, node->packet->data, length); | |
1045 | } | ||
1046 | |||
1047 |
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141 | if (length >= 32) { |
1048 | ✗ | int c = get_bits(&gb, 13); | |
1049 | |||
1050 | ✗ | if (c > 3) | |
1051 | ✗ | fill_coding_method_array(q->tone_level_idx, | |
1052 | ✗ | q->tone_level_idx_temp, q->coding_method, | |
1053 | q->nb_channels, 8 * c, | ||
1054 | q->superblocktype_2_3, q->cm_table_select); | ||
1055 | } | ||
1056 | |||
1057 | 141 | synthfilt_build_sb_samples(q, &gb, length, 0, 8); | |
1058 | 141 | } | |
1059 | |||
1060 | /** | ||
1061 | * Process subpacket 12 | ||
1062 | * | ||
1063 | * @param q context | ||
1064 | * @param node pointer to node with packet | ||
1065 | */ | ||
1066 | 141 | static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node) | |
1067 | { | ||
1068 | GetBitContext gb; | ||
1069 | 141 | int length = 0; | |
1070 | |||
1071 |
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|
141 | if (node) { |
1072 | ✗ | length = node->packet->size * 8; | |
1073 | ✗ | init_get_bits(&gb, node->packet->data, length); | |
1074 | } | ||
1075 | |||
1076 |
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|
141 | synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); |
1077 | 141 | } | |
1078 | |||
1079 | /** | ||
1080 | * Process new subpackets for synthesis filter | ||
1081 | * | ||
1082 | * @param q context | ||
1083 | * @param list list with synthesis filter packets (list D) | ||
1084 | */ | ||
1085 | 141 | static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list) | |
1086 | { | ||
1087 | QDM2SubPNode *nodes[4]; | ||
1088 | |||
1089 | 141 | nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); | |
1090 |
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|
141 | if (nodes[0]) |
1091 | 141 | process_subpacket_9(q, nodes[0]); | |
1092 | |||
1093 | 141 | nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); | |
1094 |
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141 | if (nodes[1]) |
1095 | ✗ | process_subpacket_10(q, nodes[1]); | |
1096 | else | ||
1097 | 141 | process_subpacket_10(q, NULL); | |
1098 | |||
1099 | 141 | nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); | |
1100 |
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141 | if (nodes[0] && nodes[1] && nodes[2]) |
1101 | ✗ | process_subpacket_11(q, nodes[2]); | |
1102 | else | ||
1103 | 141 | process_subpacket_11(q, NULL); | |
1104 | |||
1105 | 141 | nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); | |
1106 |
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141 | if (nodes[0] && nodes[1] && nodes[3]) |
1107 | ✗ | process_subpacket_12(q, nodes[3]); | |
1108 | else | ||
1109 | 141 | process_subpacket_12(q, NULL); | |
1110 | 141 | } | |
1111 | |||
1112 | /** | ||
1113 | * Decode superblock, fill packet lists. | ||
1114 | * | ||
1115 | * @param q context | ||
1116 | */ | ||
1117 | 141 | static void qdm2_decode_super_block(QDM2Context *q) | |
1118 | { | ||
1119 | GetBitContext gb; | ||
1120 | QDM2SubPacket header, *packet; | ||
1121 | int i, packet_bytes, sub_packet_size, sub_packets_D; | ||
1122 | 141 | unsigned int next_index = 0; | |
1123 | |||
1124 | 141 | memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); | |
1125 | 141 | memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); | |
1126 | 141 | memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); | |
1127 | |||
1128 | 141 | q->sub_packets_B = 0; | |
1129 | 141 | sub_packets_D = 0; | |
1130 | |||
1131 | 141 | average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] | |
1132 | |||
1133 | 141 | init_get_bits(&gb, q->compressed_data, q->compressed_size * 8); | |
1134 | 141 | qdm2_decode_sub_packet_header(&gb, &header); | |
1135 | |||
1136 |
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141 | if (header.type < 2 || header.type >= 8) { |
1137 | ✗ | q->has_errors = 1; | |
1138 | ✗ | av_log(NULL, AV_LOG_ERROR, "bad superblock type\n"); | |
1139 | ✗ | return; | |
1140 | } | ||
1141 | |||
1142 |
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141 | q->superblocktype_2_3 = (header.type == 2 || header.type == 3); |
1143 | 141 | packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); | |
1144 | |||
1145 | 141 | init_get_bits(&gb, header.data, header.size * 8); | |
1146 | |||
1147 |
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141 | if (header.type == 2 || header.type == 4 || header.type == 5) { |
1148 | 141 | int csum = 257 * get_bits(&gb, 8); | |
1149 | 141 | csum += 2 * get_bits(&gb, 8); | |
1150 | |||
1151 | 141 | csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); | |
1152 | |||
1153 |
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|
141 | if (csum != 0) { |
1154 | ✗ | q->has_errors = 1; | |
1155 | ✗ | av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n"); | |
1156 | ✗ | return; | |
1157 | } | ||
1158 | } | ||
1159 | |||
1160 | 141 | q->sub_packet_list_B[0].packet = NULL; | |
1161 | 141 | q->sub_packet_list_D[0].packet = NULL; | |
1162 | |||
1163 |
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|
987 | for (i = 0; i < 6; i++) |
1164 |
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|
846 | if (--q->fft_level_exp[i] < 0) |
1165 | 846 | q->fft_level_exp[i] = 0; | |
1166 | |||
1167 |
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987 | for (i = 0; packet_bytes > 0; i++) { |
1168 | int j; | ||
1169 | |||
1170 |
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|
987 | if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) { |
1171 | ✗ | SAMPLES_NEEDED_2("too many packet bytes"); | |
1172 | ✗ | return; | |
1173 | } | ||
1174 | |||
1175 | 987 | q->sub_packet_list_A[i].next = NULL; | |
1176 | |||
1177 |
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|
987 | if (i > 0) { |
1178 | 846 | q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; | |
1179 | |||
1180 | /* seek to next block */ | ||
1181 | 846 | init_get_bits(&gb, header.data, header.size * 8); | |
1182 | 846 | skip_bits(&gb, next_index * 8); | |
1183 | |||
1184 |
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|
846 | if (next_index >= header.size) |
1185 | ✗ | break; | |
1186 | } | ||
1187 | |||
1188 | /* decode subpacket */ | ||
1189 | 987 | packet = &q->sub_packets[i]; | |
1190 | 987 | qdm2_decode_sub_packet_header(&gb, packet); | |
1191 | 987 | next_index = packet->size + get_bits_count(&gb) / 8; | |
1192 |
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987 | sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; |
1193 | |||
1194 |
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987 | if (packet->type == 0) |
1195 | 141 | break; | |
1196 | |||
1197 |
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846 | if (sub_packet_size > packet_bytes) { |
1198 | ✗ | if (packet->type != 10 && packet->type != 11 && packet->type != 12) | |
1199 | ✗ | break; | |
1200 | ✗ | packet->size += packet_bytes - sub_packet_size; | |
1201 | } | ||
1202 | |||
1203 | 846 | packet_bytes -= sub_packet_size; | |
1204 | |||
1205 | /* add subpacket to 'all subpackets' list */ | ||
1206 | 846 | q->sub_packet_list_A[i].packet = packet; | |
1207 | |||
1208 | /* add subpacket to related list */ | ||
1209 |
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846 | if (packet->type == 8) { |
1210 | ✗ | SAMPLES_NEEDED_2("packet type 8"); | |
1211 | ✗ | return; | |
1212 |
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846 | } else if (packet->type >= 9 && packet->type <= 12) { |
1213 | /* packets for MPEG Audio like Synthesis Filter */ | ||
1214 |
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|
141 | QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); |
1215 |
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|
705 | } else if (packet->type == 13) { |
1216 | ✗ | for (j = 0; j < 6; j++) | |
1217 | ✗ | q->fft_level_exp[j] = get_bits(&gb, 6); | |
1218 |
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|
705 | } else if (packet->type == 14) { |
1219 | ✗ | for (j = 0; j < 6; j++) | |
1220 | ✗ | q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); | |
1221 |
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705 | } else if (packet->type == 15) { |
1222 | ✗ | SAMPLES_NEEDED_2("packet type 15") | |
1223 | ✗ | return; | |
1224 |
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705 | } else if (packet->type >= 16 && packet->type < 48 && |
1225 |
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705 | !fft_subpackets[packet->type - 16]) { |
1226 | /* packets for FFT */ | ||
1227 |
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705 | QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); |
1228 | } | ||
1229 | } // Packet bytes loop | ||
1230 | |||
1231 |
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141 | if (q->sub_packet_list_D[0].packet) { |
1232 | 141 | process_synthesis_subpackets(q, q->sub_packet_list_D); | |
1233 | 141 | q->do_synth_filter = 1; | |
1234 | ✗ | } else if (q->do_synth_filter) { | |
1235 | ✗ | process_subpacket_10(q, NULL); | |
1236 | ✗ | process_subpacket_11(q, NULL); | |
1237 | ✗ | process_subpacket_12(q, NULL); | |
1238 | } | ||
1239 | } | ||
1240 | |||
1241 | 20217 | static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet, | |
1242 | int offset, int duration, int channel, | ||
1243 | int exp, int phase) | ||
1244 | { | ||
1245 |
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20217 | if (q->fft_coefs_min_index[duration] < 0) |
1246 | 551 | q->fft_coefs_min_index[duration] = q->fft_coefs_index; | |
1247 | |||
1248 |
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20217 | q->fft_coefs[q->fft_coefs_index].sub_packet = |
1249 | 296 | ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); | |
1250 | 20217 | q->fft_coefs[q->fft_coefs_index].channel = channel; | |
1251 | 20217 | q->fft_coefs[q->fft_coefs_index].offset = offset; | |
1252 | 20217 | q->fft_coefs[q->fft_coefs_index].exp = exp; | |
1253 | 20217 | q->fft_coefs[q->fft_coefs_index].phase = phase; | |
1254 | 20217 | q->fft_coefs_index++; | |
1255 | 20217 | } | |
1256 | |||
1257 | 564 | static void qdm2_fft_decode_tones(QDM2Context *q, int duration, | |
1258 | GetBitContext *gb, int b) | ||
1259 | { | ||
1260 | int channel, stereo, phase, exp; | ||
1261 | int local_int_4, local_int_8, stereo_phase, local_int_10; | ||
1262 | int local_int_14, stereo_exp, local_int_20, local_int_28; | ||
1263 | int n, offset; | ||
1264 | |||
1265 | 564 | local_int_4 = 0; | |
1266 | 564 | local_int_28 = 0; | |
1267 | 564 | local_int_20 = 2; | |
1268 | 564 | local_int_8 = (4 - duration); | |
1269 | 564 | local_int_10 = 1 << (q->group_order - duration - 1); | |
1270 | 564 | offset = 1; | |
1271 | |||
1272 |
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16337 | while (get_bits_left(gb)>0) { |
1273 |
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16337 | if (q->superblocktype_2_3) { |
1274 |
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18368 | while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { |
1275 |
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2098 | if (get_bits_left(gb)<0) { |
1276 |
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|
67 | if(local_int_4 < q->group_size) |
1277 | ✗ | av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n"); | |
1278 | 67 | return; | |
1279 | } | ||
1280 | 2031 | offset = 1; | |
1281 |
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|
2031 | if (n == 0) { |
1282 | 2019 | local_int_4 += local_int_10; | |
1283 | 2019 | local_int_28 += (1 << local_int_8); | |
1284 | } else { | ||
1285 | 12 | local_int_4 += 8 * local_int_10; | |
1286 | 12 | local_int_28 += (8 << local_int_8); | |
1287 | } | ||
1288 | } | ||
1289 | 16270 | offset += (n - 2); | |
1290 | } else { | ||
1291 | ✗ | if (local_int_10 <= 2) { | |
1292 | ✗ | av_log(NULL, AV_LOG_ERROR, "qdm2_fft_decode_tones() stuck\n"); | |
1293 | ✗ | return; | |
1294 | } | ||
1295 | ✗ | offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); | |
1296 | ✗ | while (offset >= (local_int_10 - 1)) { | |
1297 | ✗ | offset += (1 - (local_int_10 - 1)); | |
1298 | ✗ | local_int_4 += local_int_10; | |
1299 | ✗ | local_int_28 += (1 << local_int_8); | |
1300 | } | ||
1301 | } | ||
1302 | |||
1303 |
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16270 | if (local_int_4 >= q->group_size) |
1304 | 497 | return; | |
1305 | |||
1306 | 15773 | local_int_14 = (offset >> local_int_8); | |
1307 |
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15773 | if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table)) |
1308 | ✗ | return; | |
1309 | |||
1310 |
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15773 | if (q->nb_channels > 1) { |
1311 | 15773 | channel = get_bits1(gb); | |
1312 | 15773 | stereo = get_bits1(gb); | |
1313 | } else { | ||
1314 | ✗ | channel = 0; | |
1315 | ✗ | stereo = 0; | |
1316 | } | ||
1317 | |||
1318 |
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|
15773 | exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); |
1319 | 15773 | exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; | |
1320 | 15773 | exp = (exp < 0) ? 0 : exp; | |
1321 | |||
1322 | 15773 | phase = get_bits(gb, 3); | |
1323 | 15773 | stereo_exp = 0; | |
1324 | 15773 | stereo_phase = 0; | |
1325 | |||
1326 |
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15773 | if (stereo) { |
1327 | 4444 | stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); | |
1328 | 4444 | stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); | |
1329 |
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4444 | if (stereo_phase < 0) |
1330 | 497 | stereo_phase += 8; | |
1331 | } | ||
1332 | |||
1333 |
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15773 | if (q->frequency_range > (local_int_14 + 1)) { |
1334 | 15773 | int sub_packet = (local_int_20 + local_int_28); | |
1335 | |||
1336 |
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15773 | if (q->fft_coefs_index + stereo >= FF_ARRAY_ELEMS(q->fft_coefs)) |
1337 | ✗ | return; | |
1338 | |||
1339 | 15773 | qdm2_fft_init_coefficient(q, sub_packet, offset, duration, | |
1340 | channel, exp, phase); | ||
1341 |
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|
15773 | if (stereo) |
1342 | 4444 | qdm2_fft_init_coefficient(q, sub_packet, offset, duration, | |
1343 | 1 - channel, | ||
1344 | stereo_exp, stereo_phase); | ||
1345 | } | ||
1346 | 15773 | offset++; | |
1347 | } | ||
1348 | } | ||
1349 | |||
1350 | 141 | static void qdm2_decode_fft_packets(QDM2Context *q) | |
1351 | { | ||
1352 | int i, j, min, max, value, type, unknown_flag; | ||
1353 | GetBitContext gb; | ||
1354 | |||
1355 |
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141 | if (!q->sub_packet_list_B[0].packet) |
1356 | ✗ | return; | |
1357 | |||
1358 | /* reset minimum indexes for FFT coefficients */ | ||
1359 | 141 | q->fft_coefs_index = 0; | |
1360 |
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846 | for (i = 0; i < 5; i++) |
1361 | 705 | q->fft_coefs_min_index[i] = -1; | |
1362 | |||
1363 | /* process subpackets ordered by type, largest type first */ | ||
1364 |
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846 | for (i = 0, max = 256; i < q->sub_packets_B; i++) { |
1365 | 705 | QDM2SubPacket *packet = NULL; | |
1366 | |||
1367 | /* find subpacket with largest type less than max */ | ||
1368 |
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4230 | for (j = 0, min = 0; j < q->sub_packets_B; j++) { |
1369 | 3525 | value = q->sub_packet_list_B[j].packet->type; | |
1370 |
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|
3525 | if (value > min && value < max) { |
1371 | 705 | min = value; | |
1372 | 705 | packet = q->sub_packet_list_B[j].packet; | |
1373 | } | ||
1374 | } | ||
1375 | |||
1376 | 705 | max = min; | |
1377 | |||
1378 | /* check for errors (?) */ | ||
1379 |
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705 | if (!packet) |
1380 | ✗ | return; | |
1381 | |||
1382 |
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|
705 | if (i == 0 && |
1383 |
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141 | (packet->type < 16 || packet->type >= 48 || |
1384 |
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|
141 | fft_subpackets[packet->type - 16])) |
1385 | ✗ | return; | |
1386 | |||
1387 | /* decode FFT tones */ | ||
1388 | 705 | init_get_bits(&gb, packet->data, packet->size * 8); | |
1389 | |||
1390 |
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705 | if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) |
1391 | ✗ | unknown_flag = 1; | |
1392 | else | ||
1393 | 705 | unknown_flag = 0; | |
1394 | |||
1395 | 705 | type = packet->type; | |
1396 | |||
1397 |
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|
1410 | if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { |
1398 | 705 | int duration = q->sub_sampling + 5 - (type & 15); | |
1399 | |||
1400 |
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705 | if (duration >= 0 && duration < 4) |
1401 | 564 | qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); | |
1402 | ✗ | } else if (type == 31) { | |
1403 | ✗ | for (j = 0; j < 4; j++) | |
1404 | ✗ | qdm2_fft_decode_tones(q, j, &gb, unknown_flag); | |
1405 | ✗ | } else if (type == 46) { | |
1406 | ✗ | for (j = 0; j < 6; j++) | |
1407 | ✗ | q->fft_level_exp[j] = get_bits(&gb, 6); | |
1408 | ✗ | for (j = 0; j < 4; j++) | |
1409 | ✗ | qdm2_fft_decode_tones(q, j, &gb, unknown_flag); | |
1410 | } | ||
1411 | } // Loop on B packets | ||
1412 | |||
1413 | /* calculate maximum indexes for FFT coefficients */ | ||
1414 |
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846 | for (i = 0, j = -1; i < 5; i++) |
1415 |
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|
705 | if (q->fft_coefs_min_index[i] >= 0) { |
1416 |
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|
551 | if (j >= 0) |
1417 | 410 | q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; | |
1418 | 551 | j = i; | |
1419 | } | ||
1420 |
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141 | if (j >= 0) |
1421 | 141 | q->fft_coefs_max_index[j] = q->fft_coefs_index; | |
1422 | } | ||
1423 | |||
1424 | 341696 | static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone) | |
1425 | { | ||
1426 | float level, f[6]; | ||
1427 | int i; | ||
1428 | AVComplexFloat c; | ||
1429 | 341696 | const double iscale = 2.0 * M_PI / 512.0; | |
1430 | |||
1431 | 341696 | tone->phase += tone->phase_shift; | |
1432 | |||
1433 | /* calculate current level (maximum amplitude) of tone */ | ||
1434 | 341696 | level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; | |
1435 | 341696 | c.im = level * sin(tone->phase * iscale); | |
1436 | 341696 | c.re = level * cos(tone->phase * iscale); | |
1437 | |||
1438 | /* generate FFT coefficients for tone */ | ||
1439 |
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|
341696 | if (tone->duration >= 3 || tone->cutoff >= 3) { |
1440 | 46655 | tone->complex[0].im += c.im; | |
1441 | 46655 | tone->complex[0].re += c.re; | |
1442 | 46655 | tone->complex[1].im -= c.im; | |
1443 | 46655 | tone->complex[1].re -= c.re; | |
1444 | } else { | ||
1445 | 295041 | f[1] = -tone->table[4]; | |
1446 | 295041 | f[0] = tone->table[3] - tone->table[0]; | |
1447 | 295041 | f[2] = 1.0 - tone->table[2] - tone->table[3]; | |
1448 | 295041 | f[3] = tone->table[1] + tone->table[4] - 1.0; | |
1449 | 295041 | f[4] = tone->table[0] - tone->table[1]; | |
1450 | 295041 | f[5] = tone->table[2]; | |
1451 |
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885123 | for (i = 0; i < 2; i++) { |
1452 | 590082 | tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += | |
1453 | 590082 | c.re * f[i]; | |
1454 | 1180164 | tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += | |
1455 |
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590082 | c.im * ((tone->cutoff <= i) ? -f[i] : f[i]); |
1456 | } | ||
1457 |
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1475205 | for (i = 0; i < 4; i++) { |
1458 | 1180164 | tone->complex[i].re += c.re * f[i + 2]; | |
1459 | 1180164 | tone->complex[i].im += c.im * f[i + 2]; | |
1460 | } | ||
1461 | } | ||
1462 | |||
1463 | /* copy the tone if it has not yet died out */ | ||
1464 |
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341696 | if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { |
1465 | 321878 | memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); | |
1466 | 321878 | q->fft_tone_end = (q->fft_tone_end + 1) % 1000; | |
1467 | } | ||
1468 | 341696 | } | |
1469 | |||
1470 | 2256 | static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet) | |
1471 | { | ||
1472 | int i, j, ch; | ||
1473 | 2256 | const double iscale = 0.25 * M_PI; | |
1474 | |||
1475 |
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6768 | for (ch = 0; ch < q->channels; ch++) { |
1476 | 4512 | memset(q->fft.complex[ch], 0, q->fft_size * sizeof(AVComplexFloat)); | |
1477 | } | ||
1478 | |||
1479 | |||
1480 | /* apply FFT tones with duration 4 (1 FFT period) */ | ||
1481 |
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2256 | if (q->fft_coefs_min_index[4] >= 0) |
1482 |
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8 | for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { |
1483 | float level; | ||
1484 | AVComplexFloat c; | ||
1485 | |||
1486 | ✗ | if (q->fft_coefs[i].sub_packet != sub_packet) | |
1487 | ✗ | break; | |
1488 | |||
1489 | ✗ | ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; | |
1490 | ✗ | 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]; | |
1491 | |||
1492 | ✗ | c.re = level * cos(q->fft_coefs[i].phase * iscale); | |
1493 | ✗ | c.im = level * sin(q->fft_coefs[i].phase * iscale); | |
1494 | ✗ | q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; | |
1495 | ✗ | q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; | |
1496 | ✗ | q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; | |
1497 | ✗ | q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; | |
1498 | } | ||
1499 | |||
1500 | /* generate existing FFT tones */ | ||
1501 |
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323747 | for (i = q->fft_tone_end; i != q->fft_tone_start; ) { |
1502 | 321491 | qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); | |
1503 | 321491 | q->fft_tone_start = (q->fft_tone_start + 1) % 1000; | |
1504 | } | ||
1505 | |||
1506 | /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ | ||
1507 |
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11280 | for (i = 0; i < 4; i++) |
1508 |
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|
9024 | if (q->fft_coefs_min_index[i] >= 0) { |
1509 |
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29029 | for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { |
1510 | int offset, four_i; | ||
1511 | FFTTone tone; | ||
1512 | |||
1513 |
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24549 | if (q->fft_coefs[j].sub_packet != sub_packet) |
1514 | 4344 | break; | |
1515 | |||
1516 | 20205 | four_i = (4 - i); | |
1517 | 20205 | offset = q->fft_coefs[j].offset >> four_i; | |
1518 |
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20205 | ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; |
1519 | |||
1520 |
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20205 | if (offset < q->frequency_range) { |
1521 |
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20205 | if (offset < 2) |
1522 | 2795 | tone.cutoff = offset; | |
1523 | else | ||
1524 |
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17410 | tone.cutoff = (offset >= 60) ? 3 : 2; |
1525 | |||
1526 |
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|
20205 | 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]; |
1527 | 20205 | tone.complex = &q->fft.complex[ch][offset]; | |
1528 | 20205 | tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; | |
1529 | 20205 | tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; | |
1530 | 20205 | tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); | |
1531 | 20205 | tone.duration = i; | |
1532 | 20205 | tone.time_index = 0; | |
1533 | |||
1534 | 20205 | qdm2_fft_generate_tone(q, &tone); | |
1535 | } | ||
1536 | } | ||
1537 | 8824 | q->fft_coefs_min_index[i] = j; | |
1538 | } | ||
1539 | 2256 | } | |
1540 | |||
1541 | 4512 | static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet) | |
1542 | { | ||
1543 |
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4512 | const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; |
1544 | 4512 | float *out = q->output_buffer + channel; | |
1545 | |||
1546 | 4512 | q->fft.complex[channel][0].re *= 2.0f; | |
1547 | 4512 | q->fft.complex[channel][0].im = 0.0f; | |
1548 | 4512 | q->fft.complex[channel][q->fft_size].re = 0.0f; | |
1549 | 4512 | q->fft.complex[channel][q->fft_size].im = 0.0f; | |
1550 | |||
1551 | 4512 | q->rdft_fn(q->rdft_ctx, q->fft.temp[channel], q->fft.complex[channel], | |
1552 | sizeof(AVComplexFloat)); | ||
1553 | |||
1554 | /* add samples to output buffer */ | ||
1555 |
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1159584 | for (int i = 0; i < FFALIGN(q->fft_size, 8); i++) { |
1556 | 1155072 | out[0] += q->fft.temp[channel][i].re * gain; | |
1557 | 1155072 | out[q->channels] += q->fft.temp[channel][i].im * gain; | |
1558 | 1155072 | out += 2 * q->channels; | |
1559 | } | ||
1560 | 4512 | } | |
1561 | |||
1562 | /** | ||
1563 | * @param q context | ||
1564 | * @param index subpacket number | ||
1565 | */ | ||
1566 | 2256 | static void qdm2_synthesis_filter(QDM2Context *q, int index) | |
1567 | { | ||
1568 | 2256 | int i, k, ch, sb_used, sub_sampling, dither_state = 0; | |
1569 | |||
1570 | /* copy sb_samples */ | ||
1571 |
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2256 | sb_used = QDM2_SB_USED(q->sub_sampling); |
1572 | |||
1573 |
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6768 | for (ch = 0; ch < q->channels; ch++) |
1574 |
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40608 | for (i = 0; i < 8; i++) |
1575 |
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108288 | for (k = sb_used; k < SBLIMIT; k++) |
1576 | 72192 | q->sb_samples[ch][(8 * index) + i][k] = 0; | |
1577 | |||
1578 |
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6768 | for (ch = 0; ch < q->nb_channels; ch++) { |
1579 | 4512 | float *samples_ptr = q->samples + ch; | |
1580 | |||
1581 |
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40608 | for (i = 0; i < 8; i++) { |
1582 | 36096 | ff_mpa_synth_filter_float(&q->mpadsp, | |
1583 | 36096 | q->synth_buf[ch], &(q->synth_buf_offset[ch]), | |
1584 | ff_mpa_synth_window_float, &dither_state, | ||
1585 | 36096 | samples_ptr, q->nb_channels, | |
1586 | 36096 | q->sb_samples[ch][(8 * index) + i]); | |
1587 | 36096 | samples_ptr += 32 * q->nb_channels; | |
1588 | } | ||
1589 | } | ||
1590 | |||
1591 | /* add samples to output buffer */ | ||
1592 | 2256 | sub_sampling = (4 >> q->sub_sampling); | |
1593 | |||
1594 |
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6768 | for (ch = 0; ch < q->channels; ch++) |
1595 |
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1159584 | for (i = 0; i < q->frame_size; i++) |
1596 | 1155072 | q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch]; | |
1597 | 2256 | } | |
1598 | |||
1599 | /** | ||
1600 | * Init static data (does not depend on specific file) | ||
1601 | */ | ||
1602 | 3 | static av_cold void qdm2_init_static_data(void) { | |
1603 | 3 | qdm2_init_vlc(); | |
1604 | 3 | softclip_table_init(); | |
1605 | 3 | rnd_table_init(); | |
1606 | 3 | init_noise_samples(); | |
1607 | |||
1608 | 3 | ff_mpa_synth_init_float(); | |
1609 | 3 | } | |
1610 | |||
1611 | /** | ||
1612 | * Init parameters from codec extradata | ||
1613 | */ | ||
1614 | 4 | static av_cold int qdm2_decode_init(AVCodecContext *avctx) | |
1615 | { | ||
1616 | static AVOnce init_static_once = AV_ONCE_INIT; | ||
1617 | 4 | QDM2Context *s = avctx->priv_data; | |
1618 | int ret, tmp_val, tmp, size; | ||
1619 | 4 | float scale = 1.0f / 2.0f; | |
1620 | GetByteContext gb; | ||
1621 | |||
1622 | /* extradata parsing | ||
1623 | |||
1624 | Structure: | ||
1625 | wave { | ||
1626 | frma (QDM2) | ||
1627 | QDCA | ||
1628 | QDCP | ||
1629 | } | ||
1630 | |||
1631 | 32 size (including this field) | ||
1632 | 32 tag (=frma) | ||
1633 | 32 type (=QDM2 or QDMC) | ||
1634 | |||
1635 | 32 size (including this field, in bytes) | ||
1636 | 32 tag (=QDCA) // maybe mandatory parameters | ||
1637 | 32 unknown (=1) | ||
1638 | 32 channels (=2) | ||
1639 | 32 samplerate (=44100) | ||
1640 | 32 bitrate (=96000) | ||
1641 | 32 block size (=4096) | ||
1642 | 32 frame size (=256) (for one channel) | ||
1643 | 32 packet size (=1300) | ||
1644 | |||
1645 | 32 size (including this field, in bytes) | ||
1646 | 32 tag (=QDCP) // maybe some tuneable parameters | ||
1647 | 32 float1 (=1.0) | ||
1648 | 32 zero ? | ||
1649 | 32 float2 (=1.0) | ||
1650 | 32 float3 (=1.0) | ||
1651 | 32 unknown (27) | ||
1652 | 32 unknown (8) | ||
1653 | 32 zero ? | ||
1654 | */ | ||
1655 | |||
1656 |
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4 | if (!avctx->extradata || (avctx->extradata_size < 48)) { |
1657 | ✗ | av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); | |
1658 | ✗ | return AVERROR_INVALIDDATA; | |
1659 | } | ||
1660 | |||
1661 | 4 | bytestream2_init(&gb, avctx->extradata, avctx->extradata_size); | |
1662 | |||
1663 |
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20 | while (bytestream2_get_bytes_left(&gb) > 8) { |
1664 |
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20 | if (bytestream2_peek_be64(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) | |
1665 | (uint64_t)MKBETAG('Q','D','M','2'))) | ||
1666 | 4 | break; | |
1667 | 16 | bytestream2_skip(&gb, 1); | |
1668 | } | ||
1669 | |||
1670 |
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4 | if (bytestream2_get_bytes_left(&gb) < 12) { |
1671 | ✗ | av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", | |
1672 | bytestream2_get_bytes_left(&gb)); | ||
1673 | ✗ | return AVERROR_INVALIDDATA; | |
1674 | } | ||
1675 | |||
1676 | 4 | bytestream2_skip(&gb, 8); | |
1677 | 4 | size = bytestream2_get_be32(&gb); | |
1678 | |||
1679 |
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4 | if (size > bytestream2_get_bytes_left(&gb)) { |
1680 | ✗ | av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", | |
1681 | bytestream2_get_bytes_left(&gb), size); | ||
1682 | ✗ | return AVERROR_INVALIDDATA; | |
1683 | } | ||
1684 | |||
1685 | 4 | av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); | |
1686 |
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4 | if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) { |
1687 | ✗ | av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); | |
1688 | ✗ | return AVERROR_INVALIDDATA; | |
1689 | } | ||
1690 | |||
1691 | 4 | bytestream2_skip(&gb, 4); | |
1692 | |||
1693 | 4 | s->nb_channels = s->channels = bytestream2_get_be32(&gb); | |
1694 |
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4 | if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) { |
1695 | ✗ | av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n"); | |
1696 | ✗ | return AVERROR_INVALIDDATA; | |
1697 | } | ||
1698 | 4 | av_channel_layout_uninit(&avctx->ch_layout); | |
1699 | 4 | av_channel_layout_default(&avctx->ch_layout, s->channels); | |
1700 | |||
1701 | 4 | avctx->sample_rate = bytestream2_get_be32(&gb); | |
1702 | 4 | avctx->bit_rate = bytestream2_get_be32(&gb); | |
1703 | 4 | s->group_size = bytestream2_get_be32(&gb); | |
1704 | 4 | s->fft_size = bytestream2_get_be32(&gb); | |
1705 | 4 | s->checksum_size = bytestream2_get_be32(&gb); | |
1706 |
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4 | if (s->checksum_size >= 1U << 28 || s->checksum_size <= 1) { |
1707 | ✗ | av_log(avctx, AV_LOG_ERROR, "data block size invalid (%u)\n", s->checksum_size); | |
1708 | ✗ | return AVERROR_INVALIDDATA; | |
1709 | } | ||
1710 | |||
1711 | 4 | s->fft_order = av_log2(s->fft_size) + 1; | |
1712 | |||
1713 | // Fail on unknown fft order | ||
1714 |
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4 | if ((s->fft_order < 7) || (s->fft_order > 9)) { |
1715 | ✗ | avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order); | |
1716 | ✗ | return AVERROR_PATCHWELCOME; | |
1717 | } | ||
1718 | |||
1719 | // something like max decodable tones | ||
1720 | 4 | s->group_order = av_log2(s->group_size) + 1; | |
1721 | 4 | s->frame_size = s->group_size / 16; // 16 iterations per super block | |
1722 | |||
1723 |
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4 | if (s->frame_size > QDM2_MAX_FRAME_SIZE) |
1724 | ✗ | return AVERROR_INVALIDDATA; | |
1725 | |||
1726 | 4 | s->sub_sampling = s->fft_order - 7; | |
1727 | 4 | s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); | |
1728 | |||
1729 |
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4 | if (s->frame_size * 4 >> s->sub_sampling > MPA_FRAME_SIZE) { |
1730 | ✗ | avpriv_request_sample(avctx, "large frames"); | |
1731 | ✗ | return AVERROR_PATCHWELCOME; | |
1732 | } | ||
1733 | |||
1734 |
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4 | switch ((s->sub_sampling * 2 + s->channels - 1)) { |
1735 | ✗ | case 0: tmp = 40; break; | |
1736 | ✗ | case 1: tmp = 48; break; | |
1737 | ✗ | case 2: tmp = 56; break; | |
1738 | ✗ | case 3: tmp = 72; break; | |
1739 | ✗ | case 4: tmp = 80; break; | |
1740 | 4 | case 5: tmp = 100;break; | |
1741 | ✗ | default: tmp=s->sub_sampling; break; | |
1742 | } | ||
1743 | 4 | tmp_val = 0; | |
1744 |
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4 | if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; |
1745 |
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4 | if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; |
1746 |
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4 | if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; |
1747 |
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4 | if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; |
1748 | 4 | s->cm_table_select = tmp_val; | |
1749 | |||
1750 |
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4 | if (avctx->bit_rate <= 8000) |
1751 | ✗ | s->coeff_per_sb_select = 0; | |
1752 |
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4 | else if (avctx->bit_rate < 16000) |
1753 | ✗ | s->coeff_per_sb_select = 1; | |
1754 | else | ||
1755 | 4 | s->coeff_per_sb_select = 2; | |
1756 | |||
1757 |
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4 | if (s->fft_size != (1 << (s->fft_order - 1))) { |
1758 | ✗ | av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size); | |
1759 | ✗ | return AVERROR_INVALIDDATA; | |
1760 | } | ||
1761 | |||
1762 | 4 | ret = av_tx_init(&s->rdft_ctx, &s->rdft_fn, AV_TX_FLOAT_RDFT, 1, 2*s->fft_size, &scale, 0); | |
1763 |
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4 | if (ret < 0) |
1764 | ✗ | return ret; | |
1765 | |||
1766 | 4 | ff_mpadsp_init(&s->mpadsp); | |
1767 | |||
1768 | 4 | avctx->sample_fmt = AV_SAMPLE_FMT_S16; | |
1769 | |||
1770 | 4 | ff_thread_once(&init_static_once, qdm2_init_static_data); | |
1771 | |||
1772 | 4 | return 0; | |
1773 | } | ||
1774 | |||
1775 | 4 | static av_cold int qdm2_decode_close(AVCodecContext *avctx) | |
1776 | { | ||
1777 | 4 | QDM2Context *s = avctx->priv_data; | |
1778 | |||
1779 | 4 | av_tx_uninit(&s->rdft_ctx); | |
1780 | |||
1781 | 4 | return 0; | |
1782 | } | ||
1783 | |||
1784 | 2256 | static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out) | |
1785 | { | ||
1786 | int ch, i; | ||
1787 | 2256 | const int frame_size = (q->frame_size * q->channels); | |
1788 | |||
1789 |
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2256 | if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2) |
1790 | ✗ | return -1; | |
1791 | |||
1792 | /* select input buffer */ | ||
1793 | 2256 | q->compressed_data = in; | |
1794 | 2256 | q->compressed_size = q->checksum_size; | |
1795 | |||
1796 | /* copy old block, clear new block of output samples */ | ||
1797 | 2256 | memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); | |
1798 | 2256 | memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); | |
1799 | |||
1800 | /* decode block of QDM2 compressed data */ | ||
1801 |
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2256 | if (q->sub_packet == 0) { |
1802 | 141 | q->has_errors = 0; // zero it for a new super block | |
1803 | 141 | av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); | |
1804 | 141 | qdm2_decode_super_block(q); | |
1805 | } | ||
1806 | |||
1807 | /* parse subpackets */ | ||
1808 |
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2256 | if (!q->has_errors) { |
1809 |
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2256 | if (q->sub_packet == 2) |
1810 | 141 | qdm2_decode_fft_packets(q); | |
1811 | |||
1812 | 2256 | qdm2_fft_tone_synthesizer(q, q->sub_packet); | |
1813 | } | ||
1814 | |||
1815 | /* sound synthesis stage 1 (FFT) */ | ||
1816 |
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6768 | for (ch = 0; ch < q->channels; ch++) { |
1817 | 4512 | qdm2_calculate_fft(q, ch, q->sub_packet); | |
1818 | |||
1819 |
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4512 | if (!q->has_errors && q->sub_packet_list_C[0].packet) { |
1820 | ✗ | SAMPLES_NEEDED_2("has errors, and C list is not empty") | |
1821 | ✗ | return -1; | |
1822 | } | ||
1823 | } | ||
1824 | |||
1825 | /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ | ||
1826 |
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2256 | if (!q->has_errors && q->do_synth_filter) |
1827 | 2256 | qdm2_synthesis_filter(q, q->sub_packet); | |
1828 | |||
1829 | 2256 | q->sub_packet = (q->sub_packet + 1) % 16; | |
1830 | |||
1831 | /* clip and convert output float[] to 16-bit signed samples */ | ||
1832 |
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1157328 | for (i = 0; i < frame_size; i++) { |
1833 | 1155072 | int value = (int)q->output_buffer[i]; | |
1834 | |||
1835 |
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1155072 | if (value > SOFTCLIP_THRESHOLD) |
1836 |
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245 | value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; |
1837 |
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1154827 | else if (value < -SOFTCLIP_THRESHOLD) |
1838 |
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839 | value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; |
1839 | |||
1840 | 1155072 | out[i] = value; | |
1841 | } | ||
1842 | |||
1843 | 2256 | return 0; | |
1844 | } | ||
1845 | |||
1846 | 141 | static int qdm2_decode_frame(AVCodecContext *avctx, AVFrame *frame, | |
1847 | int *got_frame_ptr, AVPacket *avpkt) | ||
1848 | { | ||
1849 | 141 | const uint8_t *buf = avpkt->data; | |
1850 | 141 | int buf_size = avpkt->size; | |
1851 | 141 | QDM2Context *s = avctx->priv_data; | |
1852 | int16_t *out; | ||
1853 | int i, ret; | ||
1854 | |||
1855 |
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141 | if(!buf) |
1856 | ✗ | return 0; | |
1857 |
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141 | if(buf_size < s->checksum_size) |
1858 | ✗ | return -1; | |
1859 | |||
1860 | /* get output buffer */ | ||
1861 | 141 | frame->nb_samples = 16 * s->frame_size; | |
1862 |
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141 | if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
1863 | ✗ | return ret; | |
1864 | 141 | out = (int16_t *)frame->data[0]; | |
1865 | |||
1866 |
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2397 | for (i = 0; i < 16; i++) { |
1867 |
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2256 | if ((ret = qdm2_decode(s, buf, out)) < 0) |
1868 | ✗ | return ret; | |
1869 | 2256 | out += s->channels * s->frame_size; | |
1870 | } | ||
1871 | |||
1872 | 141 | *got_frame_ptr = 1; | |
1873 | |||
1874 | 141 | return s->checksum_size; | |
1875 | } | ||
1876 | |||
1877 | const FFCodec ff_qdm2_decoder = { | ||
1878 | .p.name = "qdm2", | ||
1879 | CODEC_LONG_NAME("QDesign Music Codec 2"), | ||
1880 | .p.type = AVMEDIA_TYPE_AUDIO, | ||
1881 | .p.id = AV_CODEC_ID_QDM2, | ||
1882 | .priv_data_size = sizeof(QDM2Context), | ||
1883 | .init = qdm2_decode_init, | ||
1884 | .close = qdm2_decode_close, | ||
1885 | FF_CODEC_DECODE_CB(qdm2_decode_frame), | ||
1886 | .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, | ||
1887 | }; | ||
1888 |