Mercurial > libavcodec.hg
annotate qdm2.c @ 9677:592097778fa0 libavcodec
Use IF1 frame sizes in AMR-WB.
Patch by Martin Storsj: martin (remove that) at martin (oh and this too) st
author | benoit |
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date | Wed, 20 May 2009 08:57:32 +0000 |
parents | 87d4e5e27d9f |
children | 899237b1961f |
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 /** | |
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26 * @file libavcodec/qdm2.c |
2914 | 27 * QDM2 decoder |
28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni | |
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29 * The decoder is not perfect yet, there are still some distortions |
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30 * especially on files encoded with 16 or 8 subbands. |
2914 | 31 */ |
32 | |
33 #include <math.h> | |
34 #include <stddef.h> | |
35 #include <stdio.h> | |
36 | |
37 #define ALT_BITSTREAM_READER_LE | |
38 #include "avcodec.h" | |
9428 | 39 #include "get_bits.h" |
2914 | 40 #include "dsputil.h" |
41 #include "mpegaudio.h" | |
42 | |
43 #include "qdm2data.h" | |
44 | |
45 #undef NDEBUG | |
46 #include <assert.h> | |
47 | |
48 | |
49 #define SOFTCLIP_THRESHOLD 27600 | |
50 #define HARDCLIP_THRESHOLD 35716 | |
51 | |
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 BITS_LEFT(length,gb) ((length) - get_bits_count ((gb))) | |
73 | |
74 #define SAMPLES_NEEDED \ | |
75 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); | |
76 | |
77 #define SAMPLES_NEEDED_2(why) \ | |
78 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); | |
79 | |
80 | |
81 typedef int8_t sb_int8_array[2][30][64]; | |
82 | |
83 /** | |
84 * Subpacket | |
85 */ | |
86 typedef struct { | |
87 int type; ///< subpacket type | |
88 unsigned int size; ///< subpacket size | |
89 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) | |
90 } QDM2SubPacket; | |
91 | |
92 /** | |
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93 * A node in the subpacket list |
2914 | 94 */ |
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95 typedef struct QDM2SubPNode { |
2914 | 96 QDM2SubPacket *packet; ///< packet |
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97 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node |
2914 | 98 } QDM2SubPNode; |
99 | |
100 typedef struct { | |
8695 | 101 float re; |
102 float im; | |
103 } QDM2Complex; | |
104 | |
105 typedef struct { | |
2914 | 106 float level; |
8695 | 107 QDM2Complex *complex; |
6273 | 108 const float *table; |
2914 | 109 int phase; |
110 int phase_shift; | |
111 int duration; | |
112 short time_index; | |
113 short cutoff; | |
114 } FFTTone; | |
115 | |
116 typedef struct { | |
117 int16_t sub_packet; | |
118 uint8_t channel; | |
119 int16_t offset; | |
120 int16_t exp; | |
121 uint8_t phase; | |
122 } FFTCoefficient; | |
123 | |
124 typedef struct { | |
8695 | 125 DECLARE_ALIGNED_16(QDM2Complex, complex[MPA_MAX_CHANNELS][256]); |
2914 | 126 } QDM2FFT; |
127 | |
128 /** | |
129 * QDM2 decoder context | |
130 */ | |
131 typedef struct { | |
132 /// Parameters from codec header, do not change during playback | |
133 int nb_channels; ///< number of channels | |
134 int channels; ///< number of channels | |
135 int group_size; ///< size of frame group (16 frames per group) | |
136 int fft_size; ///< size of FFT, in complex numbers | |
137 int checksum_size; ///< size of data block, used also for checksum | |
138 | |
139 /// Parameters built from header parameters, do not change during playback | |
140 int group_order; ///< order of frame group | |
141 int fft_order; ///< order of FFT (actually fftorder+1) | |
142 int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im) | |
143 int frame_size; ///< size of data frame | |
144 int frequency_range; | |
145 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ | |
146 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 | |
147 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) | |
148 | |
149 /// Packets and packet lists | |
150 QDM2SubPacket sub_packets[16]; ///< the packets themselves | |
151 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets | |
152 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list | |
153 int sub_packets_B; ///< number of packets on 'B' list | |
154 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? | |
155 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets | |
156 | |
157 /// FFT and tones | |
158 FFTTone fft_tones[1000]; | |
159 int fft_tone_start; | |
160 int fft_tone_end; | |
161 FFTCoefficient fft_coefs[1000]; | |
162 int fft_coefs_index; | |
163 int fft_coefs_min_index[5]; | |
164 int fft_coefs_max_index[5]; | |
165 int fft_level_exp[6]; | |
8695 | 166 RDFTContext rdft_ctx; |
2914 | 167 QDM2FFT fft; |
168 | |
169 /// I/O data | |
6273 | 170 const uint8_t *compressed_data; |
2914 | 171 int compressed_size; |
172 float output_buffer[1024]; | |
173 | |
174 /// Synthesis filter | |
5009 | 175 DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]); |
2914 | 176 int synth_buf_offset[MPA_MAX_CHANNELS]; |
5009 | 177 DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]); |
2914 | 178 |
179 /// Mixed temporary data used in decoding | |
180 float tone_level[MPA_MAX_CHANNELS][30][64]; | |
181 int8_t coding_method[MPA_MAX_CHANNELS][30][64]; | |
182 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; | |
183 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; | |
184 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; | |
185 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; | |
186 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; | |
187 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; | |
188 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; | |
189 | |
190 // Flags | |
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191 int has_errors; ///< packet has errors |
2914 | 192 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type |
193 int do_synth_filter; ///< used to perform or skip synthesis filter | |
194 | |
195 int sub_packet; | |
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196 int noise_idx; ///< index for dithering noise table |
2914 | 197 } QDM2Context; |
198 | |
199 | |
200 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE]; | |
201 | |
202 static VLC vlc_tab_level; | |
203 static VLC vlc_tab_diff; | |
204 static VLC vlc_tab_run; | |
205 static VLC fft_level_exp_alt_vlc; | |
206 static VLC fft_level_exp_vlc; | |
207 static VLC fft_stereo_exp_vlc; | |
208 static VLC fft_stereo_phase_vlc; | |
209 static VLC vlc_tab_tone_level_idx_hi1; | |
210 static VLC vlc_tab_tone_level_idx_mid; | |
211 static VLC vlc_tab_tone_level_idx_hi2; | |
212 static VLC vlc_tab_type30; | |
213 static VLC vlc_tab_type34; | |
214 static VLC vlc_tab_fft_tone_offset[5]; | |
215 | |
216 static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1]; | |
217 static float noise_table[4096]; | |
218 static uint8_t random_dequant_index[256][5]; | |
219 static uint8_t random_dequant_type24[128][3]; | |
220 static float noise_samples[128]; | |
221 | |
5009 | 222 static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]); |
2914 | 223 |
224 | |
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225 static av_cold void softclip_table_init(void) { |
2914 | 226 int i; |
227 double dfl = SOFTCLIP_THRESHOLD - 32767; | |
228 float delta = 1.0 / -dfl; | |
229 for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++) | |
230 softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF); | |
231 } | |
232 | |
233 | |
234 // random generated table | |
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235 static av_cold void rnd_table_init(void) { |
2914 | 236 int i,j; |
237 uint32_t ldw,hdw; | |
238 uint64_t tmp64_1; | |
239 uint64_t random_seed = 0; | |
240 float delta = 1.0 / 16384.0; | |
241 for(i = 0; i < 4096 ;i++) { | |
242 random_seed = random_seed * 214013 + 2531011; | |
243 noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3; | |
244 } | |
245 | |
246 for (i = 0; i < 256 ;i++) { | |
247 random_seed = 81; | |
248 ldw = i; | |
249 for (j = 0; j < 5 ;j++) { | |
250 random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF); | |
251 ldw = (uint32_t)ldw % (uint32_t)random_seed; | |
252 tmp64_1 = (random_seed * 0x55555556); | |
253 hdw = (uint32_t)(tmp64_1 >> 32); | |
254 random_seed = (uint64_t)(hdw + (ldw >> 31)); | |
255 } | |
256 } | |
257 for (i = 0; i < 128 ;i++) { | |
258 random_seed = 25; | |
259 ldw = i; | |
260 for (j = 0; j < 3 ;j++) { | |
261 random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF); | |
262 ldw = (uint32_t)ldw % (uint32_t)random_seed; | |
263 tmp64_1 = (random_seed * 0x66666667); | |
264 hdw = (uint32_t)(tmp64_1 >> 33); | |
265 random_seed = hdw + (ldw >> 31); | |
266 } | |
267 } | |
268 } | |
269 | |
270 | |
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271 static av_cold void init_noise_samples(void) { |
2914 | 272 int i; |
273 int random_seed = 0; | |
274 float delta = 1.0 / 16384.0; | |
275 for (i = 0; i < 128;i++) { | |
276 random_seed = random_seed * 214013 + 2531011; | |
277 noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0); | |
278 } | |
279 } | |
280 | |
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281 static const uint16_t qdm2_vlc_offs[] = { |
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282 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, |
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283 }; |
2914 | 284 |
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285 static av_cold void qdm2_init_vlc(void) |
2914 | 286 { |
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287 static int vlcs_initialized = 0; |
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288 static VLC_TYPE qdm2_table[3838][2]; |
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289 |
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290 if (!vlcs_initialized) { |
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291 |
9665 | 292 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; |
293 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; | |
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_NEW_STATIC | INIT_VLC_LE); | |
2914 | 297 |
9665 | 298 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; |
299 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; | |
300 init_vlc (&vlc_tab_diff, 8, 37, | |
301 vlc_tab_diff_huffbits, 1, 1, | |
302 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 303 |
9665 | 304 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; |
305 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; | |
306 init_vlc (&vlc_tab_run, 5, 6, | |
307 vlc_tab_run_huffbits, 1, 1, | |
308 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 309 |
9665 | 310 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; |
311 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3]; | |
312 init_vlc (&fft_level_exp_alt_vlc, 8, 28, | |
313 fft_level_exp_alt_huffbits, 1, 1, | |
314 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
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315 |
2914 | 316 |
9665 | 317 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; |
318 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; | |
319 init_vlc (&fft_level_exp_vlc, 8, 20, | |
320 fft_level_exp_huffbits, 1, 1, | |
321 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 322 |
9665 | 323 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; |
324 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5]; | |
325 init_vlc (&fft_stereo_exp_vlc, 6, 7, | |
326 fft_stereo_exp_huffbits, 1, 1, | |
327 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 328 |
9665 | 329 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; |
330 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6]; | |
331 init_vlc (&fft_stereo_phase_vlc, 6, 9, | |
332 fft_stereo_phase_huffbits, 1, 1, | |
333 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 334 |
9665 | 335 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]]; |
336 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7]; | |
337 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, | |
338 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, | |
339 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 340 |
9665 | 341 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]]; |
342 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8]; | |
343 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, | |
344 vlc_tab_tone_level_idx_mid_huffbits, 1, 1, | |
345 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 346 |
9665 | 347 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]]; |
348 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9]; | |
349 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, | |
350 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, | |
351 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 352 |
9665 | 353 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; |
354 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; | |
355 init_vlc (&vlc_tab_type30, 6, 9, | |
356 vlc_tab_type30_huffbits, 1, 1, | |
357 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 358 |
9665 | 359 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; |
360 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; | |
361 init_vlc (&vlc_tab_type34, 5, 10, | |
362 vlc_tab_type34_huffbits, 1, 1, | |
363 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 364 |
9665 | 365 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]]; |
366 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12]; | |
367 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, | |
368 vlc_tab_fft_tone_offset_0_huffbits, 1, 1, | |
369 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 370 |
9665 | 371 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]]; |
372 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13]; | |
373 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, | |
374 vlc_tab_fft_tone_offset_1_huffbits, 1, 1, | |
375 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 376 |
9665 | 377 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]]; |
378 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14]; | |
379 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, | |
380 vlc_tab_fft_tone_offset_2_huffbits, 1, 1, | |
381 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 382 |
9665 | 383 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]]; |
384 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15]; | |
385 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, | |
386 vlc_tab_fft_tone_offset_3_huffbits, 1, 1, | |
387 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 388 |
9665 | 389 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]]; |
390 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16]; | |
391 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, | |
392 vlc_tab_fft_tone_offset_4_huffbits, 1, 1, | |
393 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
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394 |
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395 vlcs_initialized=1; |
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396 } |
2914 | 397 } |
398 | |
399 | |
400 /* for floating point to fixed point conversion */ | |
7129 | 401 static const float f2i_scale = (float) (1 << (FRAC_BITS - 15)); |
2914 | 402 |
403 | |
404 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) | |
405 { | |
406 int value; | |
407 | |
408 value = get_vlc2(gb, vlc->table, vlc->bits, depth); | |
409 | |
410 /* stage-2, 3 bits exponent escape sequence */ | |
411 if (value-- == 0) | |
412 value = get_bits (gb, get_bits (gb, 3) + 1); | |
413 | |
414 /* stage-3, optional */ | |
415 if (flag) { | |
416 int tmp = vlc_stage3_values[value]; | |
417 | |
418 if ((value & ~3) > 0) | |
419 tmp += get_bits (gb, (value >> 2)); | |
420 value = tmp; | |
421 } | |
422 | |
423 return value; | |
424 } | |
425 | |
426 | |
427 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) | |
428 { | |
429 int value = qdm2_get_vlc (gb, vlc, 0, depth); | |
430 | |
431 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); | |
432 } | |
433 | |
434 | |
435 /** | |
436 * QDM2 checksum | |
437 * | |
438 * @param data pointer to data to be checksum'ed | |
439 * @param length data length | |
440 * @param value checksum value | |
441 * | |
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442 * @return 0 if checksum is OK |
2914 | 443 */ |
6273 | 444 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) { |
2914 | 445 int i; |
446 | |
447 for (i=0; i < length; i++) | |
448 value -= data[i]; | |
449 | |
450 return (uint16_t)(value & 0xffff); | |
451 } | |
452 | |
453 | |
454 /** | |
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455 * Fills a QDM2SubPacket structure with packet type, size, and data pointer. |
2914 | 456 * |
457 * @param gb bitreader context | |
458 * @param sub_packet packet under analysis | |
459 */ | |
460 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) | |
461 { | |
462 sub_packet->type = get_bits (gb, 8); | |
463 | |
464 if (sub_packet->type == 0) { | |
465 sub_packet->size = 0; | |
466 sub_packet->data = NULL; | |
467 } else { | |
468 sub_packet->size = get_bits (gb, 8); | |
469 | |
470 if (sub_packet->type & 0x80) { | |
471 sub_packet->size <<= 8; | |
472 sub_packet->size |= get_bits (gb, 8); | |
473 sub_packet->type &= 0x7f; | |
474 } | |
475 | |
476 if (sub_packet->type == 0x7f) | |
477 sub_packet->type |= (get_bits (gb, 8) << 8); | |
478 | |
479 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data | |
480 } | |
481 | |
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482 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", |
2914 | 483 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); |
484 } | |
485 | |
486 | |
487 /** | |
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488 * Return node pointer to first packet of requested type in list. |
2914 | 489 * |
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490 * @param list list of subpackets to be scanned |
2914 | 491 * @param type type of searched subpacket |
492 * @return node pointer for subpacket if found, else NULL | |
493 */ | |
494 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) | |
495 { | |
496 while (list != NULL && list->packet != NULL) { | |
497 if (list->packet->type == type) | |
498 return list; | |
499 list = list->next; | |
500 } | |
501 return NULL; | |
502 } | |
503 | |
504 | |
505 /** | |
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506 * Replaces 8 elements with their average value. |
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507 * Called by qdm2_decode_superblock before starting subblock decoding. |
2914 | 508 * |
509 * @param q context | |
510 */ | |
511 static void average_quantized_coeffs (QDM2Context *q) | |
512 { | |
513 int i, j, n, ch, sum; | |
514 | |
515 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; | |
516 | |
517 for (ch = 0; ch < q->nb_channels; ch++) | |
518 for (i = 0; i < n; i++) { | |
519 sum = 0; | |
520 | |
521 for (j = 0; j < 8; j++) | |
522 sum += q->quantized_coeffs[ch][i][j]; | |
523 | |
524 sum /= 8; | |
525 if (sum > 0) | |
526 sum--; | |
527 | |
528 for (j=0; j < 8; j++) | |
529 q->quantized_coeffs[ch][i][j] = sum; | |
530 } | |
531 } | |
532 | |
533 | |
534 /** | |
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535 * Build subband samples with noise weighted by q->tone_level. |
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536 * Called by synthfilt_build_sb_samples. |
2914 | 537 * |
538 * @param q context | |
539 * @param sb subband index | |
540 */ | |
541 static void build_sb_samples_from_noise (QDM2Context *q, int sb) | |
542 { | |
543 int ch, j; | |
544 | |
545 FIX_NOISE_IDX(q->noise_idx); | |
546 | |
547 if (!q->nb_channels) | |
548 return; | |
549 | |
550 for (ch = 0; ch < q->nb_channels; ch++) | |
551 for (j = 0; j < 64; j++) { | |
552 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); | |
553 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); | |
554 } | |
555 } | |
556 | |
557 | |
558 /** | |
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559 * Called while processing data from subpackets 11 and 12. |
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560 * Used after making changes to coding_method array. |
2914 | 561 * |
562 * @param sb subband index | |
563 * @param channels number of channels | |
564 * @param coding_method q->coding_method[0][0][0] | |
565 */ | |
3076 | 566 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) |
2914 | 567 { |
568 int j,k; | |
569 int ch; | |
570 int run, case_val; | |
571 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}; | |
572 | |
573 for (ch = 0; ch < channels; ch++) { | |
574 for (j = 0; j < 64; ) { | |
575 if((coding_method[ch][sb][j] - 8) > 22) { | |
576 run = 1; | |
577 case_val = 8; | |
578 } else { | |
3333 | 579 switch (switchtable[coding_method[ch][sb][j]-8]) { |
2914 | 580 case 0: run = 10; case_val = 10; break; |
581 case 1: run = 1; case_val = 16; break; | |
582 case 2: run = 5; case_val = 24; break; | |
583 case 3: run = 3; case_val = 30; break; | |
584 case 4: run = 1; case_val = 30; break; | |
585 case 5: run = 1; case_val = 8; break; | |
586 default: run = 1; case_val = 8; break; | |
587 } | |
588 } | |
589 for (k = 0; k < run; k++) | |
590 if (j + k < 128) | |
591 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) | |
592 if (k > 0) { | |
593 SAMPLES_NEEDED | |
594 //not debugged, almost never used | |
595 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t)); | |
596 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); | |
597 } | |
598 j += run; | |
599 } | |
600 } | |
601 } | |
602 | |
603 | |
604 /** | |
605 * Related to synthesis filter | |
606 * Called by process_subpacket_10 | |
607 * | |
608 * @param q context | |
609 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 | |
610 */ | |
611 static void fill_tone_level_array (QDM2Context *q, int flag) | |
612 { | |
613 int i, sb, ch, sb_used; | |
614 int tmp, tab; | |
615 | |
616 // This should never happen | |
617 if (q->nb_channels <= 0) | |
618 return; | |
619 | |
620 for (ch = 0; ch < q->nb_channels; ch++) | |
621 for (sb = 0; sb < 30; sb++) | |
622 for (i = 0; i < 8; i++) { | |
623 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) | |
624 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ | |
625 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; | |
626 else | |
627 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; | |
628 if(tmp < 0) | |
629 tmp += 0xff; | |
630 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; | |
631 } | |
632 | |
633 sb_used = QDM2_SB_USED(q->sub_sampling); | |
634 | |
635 if ((q->superblocktype_2_3 != 0) && !flag) { | |
636 for (sb = 0; sb < sb_used; sb++) | |
637 for (ch = 0; ch < q->nb_channels; ch++) | |
638 for (i = 0; i < 64; i++) { | |
639 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; | |
640 if (q->tone_level_idx[ch][sb][i] < 0) | |
641 q->tone_level[ch][sb][i] = 0; | |
642 else | |
643 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; | |
644 } | |
645 } else { | |
646 tab = q->superblocktype_2_3 ? 0 : 1; | |
647 for (sb = 0; sb < sb_used; sb++) { | |
648 if ((sb >= 4) && (sb <= 23)) { | |
649 for (ch = 0; ch < q->nb_channels; ch++) | |
650 for (i = 0; i < 64; i++) { | |
651 tmp = q->tone_level_idx_base[ch][sb][i / 8] - | |
652 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - | |
653 q->tone_level_idx_mid[ch][sb - 4][i / 8] - | |
654 q->tone_level_idx_hi2[ch][sb - 4]; | |
655 q->tone_level_idx[ch][sb][i] = tmp & 0xff; | |
656 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
657 q->tone_level[ch][sb][i] = 0; | |
658 else | |
659 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
660 } | |
661 } else { | |
662 if (sb > 4) { | |
663 for (ch = 0; ch < q->nb_channels; ch++) | |
664 for (i = 0; i < 64; i++) { | |
665 tmp = q->tone_level_idx_base[ch][sb][i / 8] - | |
666 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - | |
667 q->tone_level_idx_hi2[ch][sb - 4]; | |
668 q->tone_level_idx[ch][sb][i] = tmp & 0xff; | |
669 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
670 q->tone_level[ch][sb][i] = 0; | |
671 else | |
672 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
673 } | |
674 } else { | |
675 for (ch = 0; ch < q->nb_channels; ch++) | |
676 for (i = 0; i < 64; i++) { | |
677 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; | |
678 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
679 q->tone_level[ch][sb][i] = 0; | |
680 else | |
681 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
682 } | |
683 } | |
684 } | |
685 } | |
686 } | |
687 | |
688 return; | |
689 } | |
690 | |
691 | |
692 /** | |
693 * Related to synthesis filter | |
694 * Called by process_subpacket_11 | |
695 * c is built with data from subpacket 11 | |
696 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples | |
697 * | |
2967 | 698 * @param tone_level_idx |
2914 | 699 * @param tone_level_idx_temp |
700 * @param coding_method q->coding_method[0][0][0] | |
701 * @param nb_channels number of channels | |
702 * @param c coming from subpacket 11, passed as 8*c | |
703 * @param superblocktype_2_3 flag based on superblock packet type | |
704 * @param cm_table_select q->cm_table_select | |
705 */ | |
706 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, | |
707 sb_int8_array coding_method, int nb_channels, | |
708 int c, int superblocktype_2_3, int cm_table_select) | |
709 { | |
710 int ch, sb, j; | |
711 int tmp, acc, esp_40, comp; | |
712 int add1, add2, add3, add4; | |
713 int64_t multres; | |
714 | |
715 // This should never happen | |
716 if (nb_channels <= 0) | |
717 return; | |
718 | |
719 if (!superblocktype_2_3) { | |
720 /* This case is untested, no samples available */ | |
721 SAMPLES_NEEDED | |
722 for (ch = 0; ch < nb_channels; ch++) | |
723 for (sb = 0; sb < 30; sb++) { | |
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724 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer |
2914 | 725 add1 = tone_level_idx[ch][sb][j] - 10; |
726 if (add1 < 0) | |
727 add1 = 0; | |
728 add2 = add3 = add4 = 0; | |
729 if (sb > 1) { | |
730 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; | |
731 if (add2 < 0) | |
732 add2 = 0; | |
733 } | |
734 if (sb > 0) { | |
735 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; | |
736 if (add3 < 0) | |
737 add3 = 0; | |
738 } | |
739 if (sb < 29) { | |
740 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; | |
741 if (add4 < 0) | |
742 add4 = 0; | |
743 } | |
744 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; | |
745 if (tmp < 0) | |
746 tmp = 0; | |
747 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; | |
748 } | |
749 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; | |
750 } | |
751 acc = 0; | |
752 for (ch = 0; ch < nb_channels; ch++) | |
753 for (sb = 0; sb < 30; sb++) | |
754 for (j = 0; j < 64; j++) | |
755 acc += tone_level_idx_temp[ch][sb][j]; | |
9538 | 756 |
2914 | 757 multres = 0x66666667 * (acc * 10); |
758 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); | |
759 for (ch = 0; ch < nb_channels; ch++) | |
760 for (sb = 0; sb < 30; sb++) | |
761 for (j = 0; j < 64; j++) { | |
762 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; | |
763 if (comp < 0) | |
764 comp += 0xff; | |
765 comp /= 256; // signed shift | |
766 switch(sb) { | |
767 case 0: | |
768 if (comp < 30) | |
769 comp = 30; | |
770 comp += 15; | |
771 break; | |
772 case 1: | |
773 if (comp < 24) | |
774 comp = 24; | |
775 comp += 10; | |
776 break; | |
777 case 2: | |
778 case 3: | |
779 case 4: | |
780 if (comp < 16) | |
781 comp = 16; | |
782 } | |
783 if (comp <= 5) | |
784 tmp = 0; | |
785 else if (comp <= 10) | |
786 tmp = 10; | |
787 else if (comp <= 16) | |
788 tmp = 16; | |
789 else if (comp <= 24) | |
790 tmp = -1; | |
791 else | |
792 tmp = 0; | |
793 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; | |
794 } | |
795 for (sb = 0; sb < 30; sb++) | |
796 fix_coding_method_array(sb, nb_channels, coding_method); | |
797 for (ch = 0; ch < nb_channels; ch++) | |
798 for (sb = 0; sb < 30; sb++) | |
799 for (j = 0; j < 64; j++) | |
800 if (sb >= 10) { | |
801 if (coding_method[ch][sb][j] < 10) | |
802 coding_method[ch][sb][j] = 10; | |
803 } else { | |
804 if (sb >= 2) { | |
805 if (coding_method[ch][sb][j] < 16) | |
806 coding_method[ch][sb][j] = 16; | |
807 } else { | |
808 if (coding_method[ch][sb][j] < 30) | |
809 coding_method[ch][sb][j] = 30; | |
810 } | |
811 } | |
812 } else { // superblocktype_2_3 != 0 | |
813 for (ch = 0; ch < nb_channels; ch++) | |
814 for (sb = 0; sb < 30; sb++) | |
815 for (j = 0; j < 64; j++) | |
816 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; | |
817 } | |
818 | |
819 return; | |
820 } | |
821 | |
822 | |
823 /** | |
824 * | |
825 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8 | |
826 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used | |
827 * | |
828 * @param q context | |
829 * @param gb bitreader context | |
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830 * @param length packet length in bits |
2914 | 831 * @param sb_min lower subband processed (sb_min included) |
832 * @param sb_max higher subband processed (sb_max excluded) | |
833 */ | |
834 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) | |
835 { | |
836 int sb, j, k, n, ch, run, channels; | |
837 int joined_stereo, zero_encoding, chs; | |
838 int type34_first; | |
839 float type34_div = 0; | |
840 float type34_predictor; | |
841 float samples[10], sign_bits[16]; | |
842 | |
843 if (length == 0) { | |
844 // If no data use noise | |
845 for (sb=sb_min; sb < sb_max; sb++) | |
846 build_sb_samples_from_noise (q, sb); | |
847 | |
848 return; | |
849 } | |
850 | |
851 for (sb = sb_min; sb < sb_max; sb++) { | |
852 FIX_NOISE_IDX(q->noise_idx); | |
853 | |
854 channels = q->nb_channels; | |
855 | |
856 if (q->nb_channels <= 1 || sb < 12) | |
857 joined_stereo = 0; | |
858 else if (sb >= 24) | |
859 joined_stereo = 1; | |
860 else | |
861 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0; | |
862 | |
863 if (joined_stereo) { | |
864 if (BITS_LEFT(length,gb) >= 16) | |
865 for (j = 0; j < 16; j++) | |
866 sign_bits[j] = get_bits1 (gb); | |
867 | |
868 for (j = 0; j < 64; j++) | |
869 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) | |
870 q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; | |
871 | |
872 fix_coding_method_array(sb, q->nb_channels, q->coding_method); | |
873 channels = 1; | |
874 } | |
875 | |
876 for (ch = 0; ch < channels; ch++) { | |
877 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; | |
878 type34_predictor = 0.0; | |
879 type34_first = 1; | |
880 | |
881 for (j = 0; j < 128; ) { | |
882 switch (q->coding_method[ch][sb][j / 2]) { | |
883 case 8: | |
884 if (BITS_LEFT(length,gb) >= 10) { | |
885 if (zero_encoding) { | |
886 for (k = 0; k < 5; k++) { | |
887 if ((j + 2 * k) >= 128) | |
888 break; | |
889 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; | |
890 } | |
891 } else { | |
892 n = get_bits(gb, 8); | |
893 for (k = 0; k < 5; k++) | |
894 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
895 } | |
896 for (k = 0; k < 5; k++) | |
897 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
898 } else { | |
899 for (k = 0; k < 10; k++) | |
900 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
901 } | |
902 run = 10; | |
903 break; | |
904 | |
905 case 10: | |
906 if (BITS_LEFT(length,gb) >= 1) { | |
907 float f = 0.81; | |
908 | |
909 if (get_bits1(gb)) | |
910 f = -f; | |
911 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; | |
912 samples[0] = f; | |
913 } else { | |
914 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
915 } | |
916 run = 1; | |
917 break; | |
918 | |
919 case 16: | |
920 if (BITS_LEFT(length,gb) >= 10) { | |
921 if (zero_encoding) { | |
922 for (k = 0; k < 5; k++) { | |
923 if ((j + k) >= 128) | |
924 break; | |
925 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; | |
926 } | |
927 } else { | |
928 n = get_bits (gb, 8); | |
929 for (k = 0; k < 5; k++) | |
930 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
931 } | |
932 } else { | |
933 for (k = 0; k < 5; k++) | |
934 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
935 } | |
936 run = 5; | |
937 break; | |
938 | |
939 case 24: | |
940 if (BITS_LEFT(length,gb) >= 7) { | |
941 n = get_bits(gb, 7); | |
942 for (k = 0; k < 3; k++) | |
943 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; | |
944 } else { | |
945 for (k = 0; k < 3; k++) | |
946 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
947 } | |
948 run = 3; | |
949 break; | |
950 | |
951 case 30: | |
952 if (BITS_LEFT(length,gb) >= 4) | |
953 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)]; | |
954 else | |
955 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
2967 | 956 |
2914 | 957 run = 1; |
958 break; | |
959 | |
960 case 34: | |
961 if (BITS_LEFT(length,gb) >= 7) { | |
962 if (type34_first) { | |
963 type34_div = (float)(1 << get_bits(gb, 2)); | |
964 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; | |
965 type34_predictor = samples[0]; | |
966 type34_first = 0; | |
967 } else { | |
968 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor; | |
969 type34_predictor = samples[0]; | |
970 } | |
971 } else { | |
972 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
973 } | |
974 run = 1; | |
975 break; | |
976 | |
977 default: | |
978 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
979 run = 1; | |
980 break; | |
981 } | |
982 | |
983 if (joined_stereo) { | |
984 float tmp[10][MPA_MAX_CHANNELS]; | |
985 | |
986 for (k = 0; k < run; k++) { | |
987 tmp[k][0] = samples[k]; | |
988 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; | |
989 } | |
990 for (chs = 0; chs < q->nb_channels; chs++) | |
991 for (k = 0; k < run; k++) | |
992 if ((j + k) < 128) | |
993 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5); | |
994 } else { | |
995 for (k = 0; k < run; k++) | |
996 if ((j + k) < 128) | |
997 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5); | |
998 } | |
999 | |
1000 j += run; | |
1001 } // j loop | |
1002 } // channel loop | |
1003 } // subband loop | |
1004 } | |
1005 | |
1006 | |
1007 /** | |
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1008 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]). |
2914 | 1009 * This is similar to process_subpacket_9, but for a single channel and for element [0] |
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1010 * same VLC tables as process_subpacket_9 are used. |
2914 | 1011 * |
1012 * @param q context | |
1013 * @param quantized_coeffs pointer to quantized_coeffs[ch][0] | |
1014 * @param gb bitreader context | |
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1015 * @param length packet length in bits |
2914 | 1016 */ |
1017 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) | |
1018 { | |
1019 int i, k, run, level, diff; | |
1020 | |
1021 if (BITS_LEFT(length,gb) < 16) | |
1022 return; | |
1023 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); | |
1024 | |
1025 quantized_coeffs[0] = level; | |
1026 | |
1027 for (i = 0; i < 7; ) { | |
1028 if (BITS_LEFT(length,gb) < 16) | |
1029 break; | |
1030 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; | |
1031 | |
1032 if (BITS_LEFT(length,gb) < 16) | |
1033 break; | |
1034 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); | |
2967 | 1035 |
2914 | 1036 for (k = 1; k <= run; k++) |
1037 quantized_coeffs[i + k] = (level + ((k * diff) / run)); | |
2967 | 1038 |
2914 | 1039 level += diff; |
1040 i += run; | |
1041 } | |
1042 } | |
1043 | |
1044 | |
1045 /** | |
1046 * Related to synthesis filter, process data from packet 10 | |
1047 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 | |
1048 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10 | |
1049 * | |
1050 * @param q context | |
1051 * @param gb bitreader context | |
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1052 * @param length packet length in bits |
2914 | 1053 */ |
1054 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) | |
1055 { | |
1056 int sb, j, k, n, ch; | |
1057 | |
1058 for (ch = 0; ch < q->nb_channels; ch++) { | |
1059 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length); | |
1060 | |
1061 if (BITS_LEFT(length,gb) < 16) { | |
1062 memset(q->quantized_coeffs[ch][0], 0, 8); | |
1063 break; | |
1064 } | |
1065 } | |
1066 | |
1067 n = q->sub_sampling + 1; | |
1068 | |
1069 for (sb = 0; sb < n; sb++) | |
1070 for (ch = 0; ch < q->nb_channels; ch++) | |
1071 for (j = 0; j < 8; j++) { | |
1072 if (BITS_LEFT(length,gb) < 1) | |
1073 break; | |
1074 if (get_bits1(gb)) { | |
1075 for (k=0; k < 8; k++) { | |
1076 if (BITS_LEFT(length,gb) < 16) | |
1077 break; | |
1078 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); | |
1079 } | |
1080 } else { | |
1081 for (k=0; k < 8; k++) | |
1082 q->tone_level_idx_hi1[ch][sb][j][k] = 0; | |
1083 } | |
1084 } | |
1085 | |
1086 n = QDM2_SB_USED(q->sub_sampling) - 4; | |
1087 | |
1088 for (sb = 0; sb < n; sb++) | |
1089 for (ch = 0; ch < q->nb_channels; ch++) { | |
1090 if (BITS_LEFT(length,gb) < 16) | |
1091 break; | |
1092 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); | |
1093 if (sb > 19) | |
1094 q->tone_level_idx_hi2[ch][sb] -= 16; | |
1095 else | |
1096 for (j = 0; j < 8; j++) | |
1097 q->tone_level_idx_mid[ch][sb][j] = -16; | |
1098 } | |
1099 | |
1100 n = QDM2_SB_USED(q->sub_sampling) - 5; | |
1101 | |
1102 for (sb = 0; sb < n; sb++) | |
1103 for (ch = 0; ch < q->nb_channels; ch++) | |
1104 for (j = 0; j < 8; j++) { | |
1105 if (BITS_LEFT(length,gb) < 16) | |
1106 break; | |
1107 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; | |
1108 } | |
1109 } | |
1110 | |
1111 /** | |
1112 * Process subpacket 9, init quantized_coeffs with data from it | |
1113 * | |
1114 * @param q context | |
1115 * @param node pointer to node with packet | |
1116 */ | |
1117 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) | |
1118 { | |
1119 GetBitContext gb; | |
1120 int i, j, k, n, ch, run, level, diff; | |
1121 | |
2916 | 1122 init_get_bits(&gb, node->packet->data, node->packet->size*8); |
2914 | 1123 |
1124 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function | |
1125 | |
1126 for (i = 1; i < n; i++) | |
1127 for (ch=0; ch < q->nb_channels; ch++) { | |
1128 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); | |
1129 q->quantized_coeffs[ch][i][0] = level; | |
1130 | |
1131 for (j = 0; j < (8 - 1); ) { | |
1132 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; | |
1133 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); | |
1134 | |
1135 for (k = 1; k <= run; k++) | |
1136 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); | |
1137 | |
1138 level += diff; | |
1139 j += run; | |
1140 } | |
1141 } | |
1142 | |
1143 for (ch = 0; ch < q->nb_channels; ch++) | |
1144 for (i = 0; i < 8; i++) | |
1145 q->quantized_coeffs[ch][0][i] = 0; | |
1146 } | |
1147 | |
1148 | |
1149 /** | |
1150 * Process subpacket 10 if not null, else | |
1151 * | |
1152 * @param q context | |
1153 * @param node pointer to node with packet | |
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1154 * @param length packet length in bits |
2914 | 1155 */ |
1156 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) | |
1157 { | |
1158 GetBitContext gb; | |
1159 | |
2916 | 1160 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
2914 | 1161 |
1162 if (length != 0) { | |
1163 init_tone_level_dequantization(q, &gb, length); | |
1164 fill_tone_level_array(q, 1); | |
1165 } else { | |
1166 fill_tone_level_array(q, 0); | |
1167 } | |
1168 } | |
1169 | |
1170 | |
1171 /** | |
1172 * Process subpacket 11 | |
1173 * | |
1174 * @param q context | |
1175 * @param node pointer to node with packet | |
1176 * @param length packet length in bit | |
1177 */ | |
1178 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length) | |
1179 { | |
1180 GetBitContext gb; | |
1181 | |
2916 | 1182 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
2914 | 1183 if (length >= 32) { |
1184 int c = get_bits (&gb, 13); | |
1185 | |
1186 if (c > 3) | |
1187 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, | |
1188 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select); | |
1189 } | |
1190 | |
1191 synthfilt_build_sb_samples(q, &gb, length, 0, 8); | |
1192 } | |
1193 | |
1194 | |
1195 /** | |
1196 * Process subpacket 12 | |
1197 * | |
1198 * @param q context | |
1199 * @param node pointer to node with packet | |
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1200 * @param length packet length in bits |
2914 | 1201 */ |
1202 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) | |
1203 { | |
1204 GetBitContext gb; | |
1205 | |
2916 | 1206 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
2914 | 1207 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); |
1208 } | |
1209 | |
1210 /* | |
1211 * Process new subpackets for synthesis filter | |
1212 * | |
1213 * @param q context | |
1214 * @param list list with synthesis filter packets (list D) | |
1215 */ | |
1216 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) | |
1217 { | |
1218 QDM2SubPNode *nodes[4]; | |
1219 | |
1220 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); | |
1221 if (nodes[0] != NULL) | |
1222 process_subpacket_9(q, nodes[0]); | |
1223 | |
1224 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); | |
1225 if (nodes[1] != NULL) | |
1226 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3); | |
1227 else | |
1228 process_subpacket_10(q, NULL, 0); | |
1229 | |
1230 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); | |
1231 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) | |
1232 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3)); | |
1233 else | |
1234 process_subpacket_11(q, NULL, 0); | |
1235 | |
1236 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); | |
1237 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) | |
1238 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3)); | |
1239 else | |
1240 process_subpacket_12(q, NULL, 0); | |
1241 } | |
1242 | |
1243 | |
1244 /* | |
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1245 * Decode superblock, fill packet lists. |
2914 | 1246 * |
1247 * @param q context | |
1248 */ | |
1249 static void qdm2_decode_super_block (QDM2Context *q) | |
1250 { | |
1251 GetBitContext gb; | |
1252 QDM2SubPacket header, *packet; | |
1253 int i, packet_bytes, sub_packet_size, sub_packets_D; | |
1254 unsigned int next_index = 0; | |
1255 | |
1256 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); | |
1257 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); | |
1258 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); | |
1259 | |
1260 q->sub_packets_B = 0; | |
1261 sub_packets_D = 0; | |
1262 | |
1263 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] | |
1264 | |
2916 | 1265 init_get_bits(&gb, q->compressed_data, q->compressed_size*8); |
2914 | 1266 qdm2_decode_sub_packet_header(&gb, &header); |
1267 | |
1268 if (header.type < 2 || header.type >= 8) { | |
1269 q->has_errors = 1; | |
1270 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); | |
1271 return; | |
1272 } | |
1273 | |
1274 q->superblocktype_2_3 = (header.type == 2 || header.type == 3); | |
1275 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); | |
1276 | |
2916 | 1277 init_get_bits(&gb, header.data, header.size*8); |
2914 | 1278 |
1279 if (header.type == 2 || header.type == 4 || header.type == 5) { | |
1280 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8); | |
1281 | |
1282 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); | |
1283 | |
1284 if (csum != 0) { | |
1285 q->has_errors = 1; | |
1286 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); | |
1287 return; | |
1288 } | |
1289 } | |
1290 | |
1291 q->sub_packet_list_B[0].packet = NULL; | |
1292 q->sub_packet_list_D[0].packet = NULL; | |
1293 | |
1294 for (i = 0; i < 6; i++) | |
1295 if (--q->fft_level_exp[i] < 0) | |
1296 q->fft_level_exp[i] = 0; | |
1297 | |
1298 for (i = 0; packet_bytes > 0; i++) { | |
1299 int j; | |
1300 | |
1301 q->sub_packet_list_A[i].next = NULL; | |
1302 | |
1303 if (i > 0) { | |
1304 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; | |
1305 | |
1306 /* seek to next block */ | |
2916 | 1307 init_get_bits(&gb, header.data, header.size*8); |
2914 | 1308 skip_bits(&gb, next_index*8); |
1309 | |
1310 if (next_index >= header.size) | |
1311 break; | |
1312 } | |
1313 | |
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1314 /* decode subpacket */ |
2914 | 1315 packet = &q->sub_packets[i]; |
1316 qdm2_decode_sub_packet_header(&gb, packet); | |
1317 next_index = packet->size + get_bits_count(&gb) / 8; | |
1318 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; | |
1319 | |
1320 if (packet->type == 0) | |
1321 break; | |
1322 | |
1323 if (sub_packet_size > packet_bytes) { | |
1324 if (packet->type != 10 && packet->type != 11 && packet->type != 12) | |
1325 break; | |
1326 packet->size += packet_bytes - sub_packet_size; | |
1327 } | |
1328 | |
1329 packet_bytes -= sub_packet_size; | |
1330 | |
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1331 /* add subpacket to 'all subpackets' list */ |
2914 | 1332 q->sub_packet_list_A[i].packet = packet; |
1333 | |
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1334 /* add subpacket to related list */ |
2914 | 1335 if (packet->type == 8) { |
1336 SAMPLES_NEEDED_2("packet type 8"); | |
1337 return; | |
1338 } else if (packet->type >= 9 && packet->type <= 12) { | |
1339 /* packets for MPEG Audio like Synthesis Filter */ | |
1340 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); | |
1341 } else if (packet->type == 13) { | |
1342 for (j = 0; j < 6; j++) | |
1343 q->fft_level_exp[j] = get_bits(&gb, 6); | |
1344 } else if (packet->type == 14) { | |
1345 for (j = 0; j < 6; j++) | |
1346 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); | |
1347 } else if (packet->type == 15) { | |
1348 SAMPLES_NEEDED_2("packet type 15") | |
1349 return; | |
1350 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { | |
1351 /* packets for FFT */ | |
1352 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); | |
1353 } | |
1354 } // Packet bytes loop | |
1355 | |
1356 /* **************************************************************** */ | |
1357 if (q->sub_packet_list_D[0].packet != NULL) { | |
1358 process_synthesis_subpackets(q, q->sub_packet_list_D); | |
1359 q->do_synth_filter = 1; | |
1360 } else if (q->do_synth_filter) { | |
1361 process_subpacket_10(q, NULL, 0); | |
1362 process_subpacket_11(q, NULL, 0); | |
1363 process_subpacket_12(q, NULL, 0); | |
1364 } | |
1365 /* **************************************************************** */ | |
1366 } | |
1367 | |
1368 | |
1369 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, | |
1370 int offset, int duration, int channel, | |
1371 int exp, int phase) | |
1372 { | |
1373 if (q->fft_coefs_min_index[duration] < 0) | |
1374 q->fft_coefs_min_index[duration] = q->fft_coefs_index; | |
1375 | |
1376 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); | |
1377 q->fft_coefs[q->fft_coefs_index].channel = channel; | |
1378 q->fft_coefs[q->fft_coefs_index].offset = offset; | |
1379 q->fft_coefs[q->fft_coefs_index].exp = exp; | |
1380 q->fft_coefs[q->fft_coefs_index].phase = phase; | |
1381 q->fft_coefs_index++; | |
1382 } | |
1383 | |
1384 | |
1385 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) | |
1386 { | |
1387 int channel, stereo, phase, exp; | |
1388 int local_int_4, local_int_8, stereo_phase, local_int_10; | |
1389 int local_int_14, stereo_exp, local_int_20, local_int_28; | |
1390 int n, offset; | |
1391 | |
1392 local_int_4 = 0; | |
1393 local_int_28 = 0; | |
1394 local_int_20 = 2; | |
1395 local_int_8 = (4 - duration); | |
1396 local_int_10 = 1 << (q->group_order - duration - 1); | |
1397 offset = 1; | |
1398 | |
1399 while (1) { | |
1400 if (q->superblocktype_2_3) { | |
1401 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { | |
1402 offset = 1; | |
1403 if (n == 0) { | |
1404 local_int_4 += local_int_10; | |
1405 local_int_28 += (1 << local_int_8); | |
1406 } else { | |
1407 local_int_4 += 8*local_int_10; | |
1408 local_int_28 += (8 << local_int_8); | |
1409 } | |
1410 } | |
1411 offset += (n - 2); | |
1412 } else { | |
1413 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); | |
1414 while (offset >= (local_int_10 - 1)) { | |
1415 offset += (1 - (local_int_10 - 1)); | |
1416 local_int_4 += local_int_10; | |
1417 local_int_28 += (1 << local_int_8); | |
1418 } | |
1419 } | |
1420 | |
1421 if (local_int_4 >= q->group_size) | |
1422 return; | |
1423 | |
1424 local_int_14 = (offset >> local_int_8); | |
1425 | |
1426 if (q->nb_channels > 1) { | |
1427 channel = get_bits1(gb); | |
1428 stereo = get_bits1(gb); | |
1429 } else { | |
1430 channel = 0; | |
1431 stereo = 0; | |
1432 } | |
1433 | |
1434 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); | |
1435 exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; | |
1436 exp = (exp < 0) ? 0 : exp; | |
1437 | |
1438 phase = get_bits(gb, 3); | |
1439 stereo_exp = 0; | |
1440 stereo_phase = 0; | |
1441 | |
1442 if (stereo) { | |
1443 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); | |
1444 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); | |
1445 if (stereo_phase < 0) | |
1446 stereo_phase += 8; | |
1447 } | |
1448 | |
1449 if (q->frequency_range > (local_int_14 + 1)) { | |
1450 int sub_packet = (local_int_20 + local_int_28); | |
1451 | |
1452 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); | |
1453 if (stereo) | |
1454 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); | |
1455 } | |
1456 | |
1457 offset++; | |
1458 } | |
1459 } | |
1460 | |
1461 | |
1462 static void qdm2_decode_fft_packets (QDM2Context *q) | |
1463 { | |
1464 int i, j, min, max, value, type, unknown_flag; | |
1465 GetBitContext gb; | |
1466 | |
1467 if (q->sub_packet_list_B[0].packet == NULL) | |
1468 return; | |
1469 | |
6903 | 1470 /* reset minimum indexes for FFT coefficients */ |
2914 | 1471 q->fft_coefs_index = 0; |
1472 for (i=0; i < 5; i++) | |
1473 q->fft_coefs_min_index[i] = -1; | |
1474 | |
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1475 /* process subpackets ordered by type, largest type first */ |
2914 | 1476 for (i = 0, max = 256; i < q->sub_packets_B; i++) { |
7306 | 1477 QDM2SubPacket *packet= NULL; |
2914 | 1478 |
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1479 /* find subpacket with largest type less than max */ |
7306 | 1480 for (j = 0, min = 0; j < q->sub_packets_B; j++) { |
2914 | 1481 value = q->sub_packet_list_B[j].packet->type; |
1482 if (value > min && value < max) { | |
1483 min = value; | |
1484 packet = q->sub_packet_list_B[j].packet; | |
1485 } | |
1486 } | |
1487 | |
1488 max = min; | |
1489 | |
1490 /* check for errors (?) */ | |
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1491 if (!packet) |
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1492 return; |
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1493 |
2914 | 1494 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) |
1495 return; | |
1496 | |
1497 /* decode FFT tones */ | |
2916 | 1498 init_get_bits (&gb, packet->data, packet->size*8); |
2914 | 1499 |
1500 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) | |
1501 unknown_flag = 1; | |
1502 else | |
1503 unknown_flag = 0; | |
1504 | |
1505 type = packet->type; | |
1506 | |
1507 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { | |
1508 int duration = q->sub_sampling + 5 - (type & 15); | |
1509 | |
1510 if (duration >= 0 && duration < 4) | |
1511 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); | |
1512 } else if (type == 31) { | |
3320 | 1513 for (j=0; j < 4; j++) |
1514 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); | |
2914 | 1515 } else if (type == 46) { |
3320 | 1516 for (j=0; j < 6; j++) |
1517 q->fft_level_exp[j] = get_bits(&gb, 6); | |
1518 for (j=0; j < 4; j++) | |
1519 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); | |
2914 | 1520 } |
1521 } // Loop on B packets | |
1522 | |
6903 | 1523 /* calculate maximum indexes for FFT coefficients */ |
2914 | 1524 for (i = 0, j = -1; i < 5; i++) |
1525 if (q->fft_coefs_min_index[i] >= 0) { | |
1526 if (j >= 0) | |
1527 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; | |
1528 j = i; | |
1529 } | |
1530 if (j >= 0) | |
1531 q->fft_coefs_max_index[j] = q->fft_coefs_index; | |
1532 } | |
1533 | |
1534 | |
1535 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) | |
1536 { | |
1537 float level, f[6]; | |
1538 int i; | |
1539 QDM2Complex c; | |
1540 const double iscale = 2.0*M_PI / 512.0; | |
1541 | |
1542 tone->phase += tone->phase_shift; | |
1543 | |
1544 /* calculate current level (maximum amplitude) of tone */ | |
1545 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; | |
1546 c.im = level * sin(tone->phase*iscale); | |
1547 c.re = level * cos(tone->phase*iscale); | |
1548 | |
1549 /* generate FFT coefficients for tone */ | |
1550 if (tone->duration >= 3 || tone->cutoff >= 3) { | |
8695 | 1551 tone->complex[0].im += c.im; |
1552 tone->complex[0].re += c.re; | |
1553 tone->complex[1].im -= c.im; | |
1554 tone->complex[1].re -= c.re; | |
2914 | 1555 } else { |
1556 f[1] = -tone->table[4]; | |
1557 f[0] = tone->table[3] - tone->table[0]; | |
1558 f[2] = 1.0 - tone->table[2] - tone->table[3]; | |
1559 f[3] = tone->table[1] + tone->table[4] - 1.0; | |
1560 f[4] = tone->table[0] - tone->table[1]; | |
1561 f[5] = tone->table[2]; | |
1562 for (i = 0; i < 2; i++) { | |
8695 | 1563 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; |
1564 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); | |
2914 | 1565 } |
1566 for (i = 0; i < 4; i++) { | |
8695 | 1567 tone->complex[i].re += c.re * f[i+2]; |
1568 tone->complex[i].im += c.im * f[i+2]; | |
2914 | 1569 } |
1570 } | |
1571 | |
1572 /* copy the tone if it has not yet died out */ | |
1573 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { | |
1574 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); | |
1575 q->fft_tone_end = (q->fft_tone_end + 1) % 1000; | |
1576 } | |
1577 } | |
1578 | |
1579 | |
1580 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) | |
1581 { | |
1582 int i, j, ch; | |
1583 const double iscale = 0.25 * M_PI; | |
1584 | |
1585 for (ch = 0; ch < q->channels; ch++) { | |
8695 | 1586 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); |
2914 | 1587 } |
1588 | |
1589 | |
1590 /* apply FFT tones with duration 4 (1 FFT period) */ | |
1591 if (q->fft_coefs_min_index[4] >= 0) | |
1592 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { | |
1593 float level; | |
1594 QDM2Complex c; | |
1595 | |
1596 if (q->fft_coefs[i].sub_packet != sub_packet) | |
1597 break; | |
1598 | |
1599 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; | |
1600 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]; | |
1601 | |
1602 c.re = level * cos(q->fft_coefs[i].phase * iscale); | |
1603 c.im = level * sin(q->fft_coefs[i].phase * iscale); | |
8695 | 1604 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; |
1605 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; | |
1606 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; | |
1607 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; | |
2914 | 1608 } |
1609 | |
1610 /* generate existing FFT tones */ | |
1611 for (i = q->fft_tone_end; i != q->fft_tone_start; ) { | |
1612 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); | |
1613 q->fft_tone_start = (q->fft_tone_start + 1) % 1000; | |
1614 } | |
1615 | |
1616 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ | |
1617 for (i = 0; i < 4; i++) | |
1618 if (q->fft_coefs_min_index[i] >= 0) { | |
1619 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { | |
1620 int offset, four_i; | |
1621 FFTTone tone; | |
1622 | |
1623 if (q->fft_coefs[j].sub_packet != sub_packet) | |
1624 break; | |
1625 | |
1626 four_i = (4 - i); | |
1627 offset = q->fft_coefs[j].offset >> four_i; | |
1628 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; | |
1629 | |
1630 if (offset < q->frequency_range) { | |
1631 if (offset < 2) | |
1632 tone.cutoff = offset; | |
1633 else | |
1634 tone.cutoff = (offset >= 60) ? 3 : 2; | |
1635 | |
1636 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]; | |
8695 | 1637 tone.complex = &q->fft.complex[ch][offset]; |
6273 | 1638 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; |
2914 | 1639 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; |
1640 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); | |
1641 tone.duration = i; | |
1642 tone.time_index = 0; | |
1643 | |
1644 qdm2_fft_generate_tone(q, &tone); | |
1645 } | |
1646 } | |
1647 q->fft_coefs_min_index[i] = j; | |
1648 } | |
1649 } | |
1650 | |
1651 | |
1652 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) | |
1653 { | |
8695 | 1654 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; |
1655 int i; | |
1656 q->fft.complex[channel][0].re *= 2.0f; | |
1657 q->fft.complex[channel][0].im = 0.0f; | |
1658 ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); | |
2914 | 1659 /* add samples to output buffer */ |
1660 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) | |
8695 | 1661 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; |
2914 | 1662 } |
1663 | |
1664 | |
1665 /** | |
1666 * @param q context | |
1667 * @param index subpacket number | |
1668 */ | |
1669 static void qdm2_synthesis_filter (QDM2Context *q, int index) | |
1670 { | |
1671 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; | |
1672 int i, k, ch, sb_used, sub_sampling, dither_state = 0; | |
1673 | |
1674 /* copy sb_samples */ | |
1675 sb_used = QDM2_SB_USED(q->sub_sampling); | |
1676 | |
1677 for (ch = 0; ch < q->channels; ch++) | |
1678 for (i = 0; i < 8; i++) | |
1679 for (k=sb_used; k < SBLIMIT; k++) | |
1680 q->sb_samples[ch][(8 * index) + i][k] = 0; | |
1681 | |
1682 for (ch = 0; ch < q->nb_channels; ch++) { | |
1683 OUT_INT *samples_ptr = samples + ch; | |
1684 | |
1685 for (i = 0; i < 8; i++) { | |
1686 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]), | |
1687 mpa_window, &dither_state, | |
1688 samples_ptr, q->nb_channels, | |
1689 q->sb_samples[ch][(8 * index) + i]); | |
1690 samples_ptr += 32 * q->nb_channels; | |
1691 } | |
1692 } | |
1693 | |
1694 /* add samples to output buffer */ | |
1695 sub_sampling = (4 >> q->sub_sampling); | |
1696 | |
1697 for (ch = 0; ch < q->channels; ch++) | |
1698 for (i = 0; i < q->frame_size; i++) | |
1699 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16)); | |
1700 } | |
1701 | |
1702 | |
1703 /** | |
1704 * Init static data (does not depend on specific file) | |
1705 * | |
1706 * @param q context | |
1707 */ | |
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1708 static av_cold void qdm2_init(QDM2Context *q) { |
6350 | 1709 static int initialized = 0; |
2914 | 1710 |
6350 | 1711 if (initialized != 0) |
2914 | 1712 return; |
6350 | 1713 initialized = 1; |
2914 | 1714 |
1715 qdm2_init_vlc(); | |
1716 ff_mpa_synth_init(mpa_window); | |
1717 softclip_table_init(); | |
1718 rnd_table_init(); | |
1719 init_noise_samples(); | |
1720 | |
1721 av_log(NULL, AV_LOG_DEBUG, "init done\n"); | |
1722 } | |
1723 | |
1724 | |
1725 #if 0 | |
1726 static void dump_context(QDM2Context *q) | |
1727 { | |
1728 int i; | |
1729 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b); | |
1730 PRINT("compressed_data",q->compressed_data); | |
1731 PRINT("compressed_size",q->compressed_size); | |
1732 PRINT("frame_size",q->frame_size); | |
1733 PRINT("checksum_size",q->checksum_size); | |
1734 PRINT("channels",q->channels); | |
1735 PRINT("nb_channels",q->nb_channels); | |
1736 PRINT("fft_frame_size",q->fft_frame_size); | |
1737 PRINT("fft_size",q->fft_size); | |
1738 PRINT("sub_sampling",q->sub_sampling); | |
1739 PRINT("fft_order",q->fft_order); | |
1740 PRINT("group_order",q->group_order); | |
1741 PRINT("group_size",q->group_size); | |
1742 PRINT("sub_packet",q->sub_packet); | |
1743 PRINT("frequency_range",q->frequency_range); | |
1744 PRINT("has_errors",q->has_errors); | |
1745 PRINT("fft_tone_end",q->fft_tone_end); | |
1746 PRINT("fft_tone_start",q->fft_tone_start); | |
1747 PRINT("fft_coefs_index",q->fft_coefs_index); | |
1748 PRINT("coeff_per_sb_select",q->coeff_per_sb_select); | |
1749 PRINT("cm_table_select",q->cm_table_select); | |
1750 PRINT("noise_idx",q->noise_idx); | |
1751 | |
1752 for (i = q->fft_tone_start; i < q->fft_tone_end; i++) | |
1753 { | |
1754 FFTTone *t = &q->fft_tones[i]; | |
2967 | 1755 |
2914 | 1756 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i); |
1757 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level); | |
1758 // PRINT(" level", t->level); | |
1759 PRINT(" phase", t->phase); | |
1760 PRINT(" phase_shift", t->phase_shift); | |
1761 PRINT(" duration", t->duration); | |
1762 PRINT(" samples_im", t->samples_im); | |
1763 PRINT(" samples_re", t->samples_re); | |
1764 PRINT(" table", t->table); | |
1765 } | |
1766 | |
1767 } | |
1768 #endif | |
1769 | |
1770 | |
1771 /** | |
1772 * Init parameters from codec extradata | |
1773 */ | |
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1774 static av_cold int qdm2_decode_init(AVCodecContext *avctx) |
2914 | 1775 { |
1776 QDM2Context *s = avctx->priv_data; | |
1777 uint8_t *extradata; | |
1778 int extradata_size; | |
1779 int tmp_val, tmp, size; | |
2967 | 1780 |
2914 | 1781 /* extradata parsing |
2967 | 1782 |
2914 | 1783 Structure: |
1784 wave { | |
1785 frma (QDM2) | |
1786 QDCA | |
1787 QDCP | |
1788 } | |
2967 | 1789 |
2914 | 1790 32 size (including this field) |
1791 32 tag (=frma) | |
1792 32 type (=QDM2 or QDMC) | |
2967 | 1793 |
2914 | 1794 32 size (including this field, in bytes) |
1795 32 tag (=QDCA) // maybe mandatory parameters | |
1796 32 unknown (=1) | |
1797 32 channels (=2) | |
1798 32 samplerate (=44100) | |
1799 32 bitrate (=96000) | |
1800 32 block size (=4096) | |
1801 32 frame size (=256) (for one channel) | |
1802 32 packet size (=1300) | |
2967 | 1803 |
2914 | 1804 32 size (including this field, in bytes) |
1805 32 tag (=QDCP) // maybe some tuneable parameters | |
1806 32 float1 (=1.0) | |
1807 32 zero ? | |
1808 32 float2 (=1.0) | |
1809 32 float3 (=1.0) | |
1810 32 unknown (27) | |
1811 32 unknown (8) | |
1812 32 zero ? | |
1813 */ | |
1814 | |
1815 if (!avctx->extradata || (avctx->extradata_size < 48)) { | |
1816 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); | |
1817 return -1; | |
1818 } | |
1819 | |
1820 extradata = avctx->extradata; | |
1821 extradata_size = avctx->extradata_size; | |
1822 | |
1823 while (extradata_size > 7) { | |
1824 if (!memcmp(extradata, "frmaQDM", 7)) | |
1825 break; | |
1826 extradata++; | |
1827 extradata_size--; | |
1828 } | |
1829 | |
1830 if (extradata_size < 12) { | |
1831 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", | |
1832 extradata_size); | |
1833 return -1; | |
1834 } | |
1835 | |
1836 if (memcmp(extradata, "frmaQDM", 7)) { | |
1837 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); | |
1838 return -1; | |
1839 } | |
1840 | |
1841 if (extradata[7] == 'C') { | |
1842 // s->is_qdmc = 1; | |
1843 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); | |
1844 return -1; | |
1845 } | |
1846 | |
1847 extradata += 8; | |
1848 extradata_size -= 8; | |
1849 | |
4364 | 1850 size = AV_RB32(extradata); |
2914 | 1851 |
1852 if(size > extradata_size){ | |
1853 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", | |
1854 extradata_size, size); | |
1855 return -1; | |
1856 } | |
1857 | |
1858 extradata += 4; | |
1859 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); | |
4364 | 1860 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { |
2914 | 1861 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); |
1862 return -1; | |
1863 } | |
1864 | |
1865 extradata += 8; | |
1866 | |
4364 | 1867 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); |
2914 | 1868 extradata += 4; |
1869 | |
4364 | 1870 avctx->sample_rate = AV_RB32(extradata); |
2914 | 1871 extradata += 4; |
1872 | |
4364 | 1873 avctx->bit_rate = AV_RB32(extradata); |
2914 | 1874 extradata += 4; |
1875 | |
4364 | 1876 s->group_size = AV_RB32(extradata); |
2914 | 1877 extradata += 4; |
1878 | |
4364 | 1879 s->fft_size = AV_RB32(extradata); |
2914 | 1880 extradata += 4; |
1881 | |
4364 | 1882 s->checksum_size = AV_RB32(extradata); |
2914 | 1883 |
1884 s->fft_order = av_log2(s->fft_size) + 1; | |
1885 s->fft_frame_size = 2 * s->fft_size; // complex has two floats | |
1886 | |
1887 // something like max decodable tones | |
1888 s->group_order = av_log2(s->group_size) + 1; | |
1889 s->frame_size = s->group_size / 16; // 16 iterations per super block | |
1890 | |
2954 | 1891 s->sub_sampling = s->fft_order - 7; |
2914 | 1892 s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); |
2967 | 1893 |
2914 | 1894 switch ((s->sub_sampling * 2 + s->channels - 1)) { |
1895 case 0: tmp = 40; break; | |
1896 case 1: tmp = 48; break; | |
1897 case 2: tmp = 56; break; | |
1898 case 3: tmp = 72; break; | |
1899 case 4: tmp = 80; break; | |
1900 case 5: tmp = 100;break; | |
1901 default: tmp=s->sub_sampling; break; | |
1902 } | |
1903 tmp_val = 0; | |
1904 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; | |
1905 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; | |
1906 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; | |
1907 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; | |
1908 s->cm_table_select = tmp_val; | |
1909 | |
1910 if (s->sub_sampling == 0) | |
2954 | 1911 tmp = 7999; |
2914 | 1912 else |
1913 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; | |
1914 /* | |
2954 | 1915 0: 7999 -> 0 |
2914 | 1916 1: 20000 -> 2 |
1917 2: 28000 -> 2 | |
1918 */ | |
1919 if (tmp < 8000) | |
1920 s->coeff_per_sb_select = 0; | |
1921 else if (tmp <= 16000) | |
1922 s->coeff_per_sb_select = 1; | |
1923 else | |
1924 s->coeff_per_sb_select = 2; | |
1925 | |
8695 | 1926 // Fail on unknown fft order |
2954 | 1927 if ((s->fft_order < 7) || (s->fft_order > 9)) { |
2914 | 1928 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); |
2954 | 1929 return -1; |
1930 } | |
2914 | 1931 |
8695 | 1932 ff_rdft_init(&s->rdft_ctx, s->fft_order, IRDFT); |
2914 | 1933 |
1934 qdm2_init(s); | |
2967 | 1935 |
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1936 avctx->sample_fmt = SAMPLE_FMT_S16; |
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1937 |
2914 | 1938 // dump_context(s); |
1939 return 0; | |
1940 } | |
1941 | |
1942 | |
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1943 static av_cold int qdm2_decode_close(AVCodecContext *avctx) |
2914 | 1944 { |
1945 QDM2Context *s = avctx->priv_data; | |
1946 | |
8695 | 1947 ff_rdft_end(&s->rdft_ctx); |
2967 | 1948 |
2914 | 1949 return 0; |
1950 } | |
1951 | |
1952 | |
6273 | 1953 static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) |
2914 | 1954 { |
1955 int ch, i; | |
1956 const int frame_size = (q->frame_size * q->channels); | |
2967 | 1957 |
2914 | 1958 /* select input buffer */ |
1959 q->compressed_data = in; | |
1960 q->compressed_size = q->checksum_size; | |
1961 | |
1962 // dump_context(q); | |
1963 | |
1964 /* copy old block, clear new block of output samples */ | |
1965 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); | |
1966 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); | |
1967 | |
1968 /* decode block of QDM2 compressed data */ | |
1969 if (q->sub_packet == 0) { | |
1970 q->has_errors = 0; // zero it for a new super block | |
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1971 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); |
2914 | 1972 qdm2_decode_super_block(q); |
1973 } | |
1974 | |
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1975 /* parse subpackets */ |
2914 | 1976 if (!q->has_errors) { |
1977 if (q->sub_packet == 2) | |
1978 qdm2_decode_fft_packets(q); | |
1979 | |
1980 qdm2_fft_tone_synthesizer(q, q->sub_packet); | |
1981 } | |
1982 | |
1983 /* sound synthesis stage 1 (FFT) */ | |
1984 for (ch = 0; ch < q->channels; ch++) { | |
1985 qdm2_calculate_fft(q, ch, q->sub_packet); | |
1986 | |
1987 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { | |
1988 SAMPLES_NEEDED_2("has errors, and C list is not empty") | |
1989 return; | |
1990 } | |
1991 } | |
1992 | |
1993 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ | |
1994 if (!q->has_errors && q->do_synth_filter) | |
1995 qdm2_synthesis_filter(q, q->sub_packet); | |
1996 | |
1997 q->sub_packet = (q->sub_packet + 1) % 16; | |
1998 | |
1999 /* clip and convert output float[] to 16bit signed samples */ | |
2000 for (i = 0; i < frame_size; i++) { | |
2001 int value = (int)q->output_buffer[i]; | |
2002 | |
2003 if (value > SOFTCLIP_THRESHOLD) | |
2004 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; | |
2005 else if (value < -SOFTCLIP_THRESHOLD) | |
2006 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; | |
2007 | |
2008 out[i] = value; | |
2009 } | |
2010 } | |
2011 | |
2012 | |
2013 static int qdm2_decode_frame(AVCodecContext *avctx, | |
2014 void *data, int *data_size, | |
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2015 AVPacket *avpkt) |
2914 | 2016 { |
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2017 const uint8_t *buf = avpkt->data; |
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2018 int buf_size = avpkt->size; |
2914 | 2019 QDM2Context *s = avctx->priv_data; |
2020 | |
3158 | 2021 if(!buf) |
2914 | 2022 return 0; |
3158 | 2023 if(buf_size < s->checksum_size) |
2024 return -1; | |
2914 | 2025 |
2026 *data_size = s->channels * s->frame_size * sizeof(int16_t); | |
2027 | |
2028 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", | |
2029 buf_size, buf, s->checksum_size, data, *data_size); | |
2030 | |
2031 qdm2_decode(s, buf, data); | |
2032 | |
2033 // reading only when next superblock found | |
2034 if (s->sub_packet == 0) { | |
2035 return s->checksum_size; | |
2036 } | |
2037 | |
2038 return 0; | |
2039 } | |
2040 | |
2041 AVCodec qdm2_decoder = | |
2042 { | |
2043 .name = "qdm2", | |
2044 .type = CODEC_TYPE_AUDIO, | |
2045 .id = CODEC_ID_QDM2, | |
2046 .priv_data_size = sizeof(QDM2Context), | |
2047 .init = qdm2_decode_init, | |
2048 .close = qdm2_decode_close, | |
2049 .decode = qdm2_decode_frame, | |
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2050 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), |
2914 | 2051 }; |