Mercurial > libavcodec.hg
annotate qdm2.c @ 11577:e4a69ce0256f libavcodec
Optimize bidir_refine a bit.
compiled code is less tham 1/4 the size.
a tiny bit faster
author | michael |
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date | Fri, 02 Apr 2010 14:04:15 +0000 |
parents | 8a4984c5cacc |
children | 7dd2a45249a9 |
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" |
11370 | 41 #include "fft.h" |
2914 | 42 #include "mpegaudio.h" |
43 | |
44 #include "qdm2data.h" | |
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45 #include "qdm2_tablegen.h" |
2914 | 46 |
47 #undef NDEBUG | |
48 #include <assert.h> | |
49 | |
50 | |
51 #define QDM2_LIST_ADD(list, size, packet) \ | |
52 do { \ | |
53 if (size > 0) { \ | |
54 list[size - 1].next = &list[size]; \ | |
55 } \ | |
56 list[size].packet = packet; \ | |
57 list[size].next = NULL; \ | |
58 size++; \ | |
59 } while(0) | |
60 | |
61 // Result is 8, 16 or 30 | |
62 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling)) | |
63 | |
64 #define FIX_NOISE_IDX(noise_idx) \ | |
65 if ((noise_idx) >= 3840) \ | |
66 (noise_idx) -= 3840; \ | |
67 | |
68 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)]) | |
69 | |
70 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb))) | |
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 | |
79 typedef int8_t sb_int8_array[2][30][64]; | |
80 | |
81 /** | |
82 * Subpacket | |
83 */ | |
84 typedef struct { | |
85 int type; ///< subpacket type | |
86 unsigned int size; ///< subpacket size | |
87 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) | |
88 } QDM2SubPacket; | |
89 | |
90 /** | |
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91 * A node in the subpacket list |
2914 | 92 */ |
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93 typedef struct QDM2SubPNode { |
2914 | 94 QDM2SubPacket *packet; ///< packet |
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95 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node |
2914 | 96 } QDM2SubPNode; |
97 | |
98 typedef struct { | |
8695 | 99 float re; |
100 float im; | |
101 } QDM2Complex; | |
102 | |
103 typedef struct { | |
2914 | 104 float level; |
8695 | 105 QDM2Complex *complex; |
6273 | 106 const float *table; |
2914 | 107 int phase; |
108 int phase_shift; | |
109 int duration; | |
110 short time_index; | |
111 short cutoff; | |
112 } FFTTone; | |
113 | |
114 typedef struct { | |
115 int16_t sub_packet; | |
116 uint8_t channel; | |
117 int16_t offset; | |
118 int16_t exp; | |
119 uint8_t phase; | |
120 } FFTCoefficient; | |
121 | |
122 typedef struct { | |
11369 | 123 DECLARE_ALIGNED(16, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; |
2914 | 124 } QDM2FFT; |
125 | |
126 /** | |
127 * QDM2 decoder context | |
128 */ | |
129 typedef struct { | |
130 /// Parameters from codec header, do not change during playback | |
131 int nb_channels; ///< number of channels | |
132 int channels; ///< number of channels | |
133 int group_size; ///< size of frame group (16 frames per group) | |
134 int fft_size; ///< size of FFT, in complex numbers | |
135 int checksum_size; ///< size of data block, used also for checksum | |
136 | |
137 /// Parameters built from header parameters, do not change during playback | |
138 int group_order; ///< order of frame group | |
139 int fft_order; ///< order of FFT (actually fftorder+1) | |
140 int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im) | |
141 int frame_size; ///< size of data frame | |
142 int frequency_range; | |
143 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ | |
144 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 | |
145 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) | |
146 | |
147 /// Packets and packet lists | |
148 QDM2SubPacket sub_packets[16]; ///< the packets themselves | |
149 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets | |
150 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list | |
151 int sub_packets_B; ///< number of packets on 'B' list | |
152 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? | |
153 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets | |
154 | |
155 /// FFT and tones | |
156 FFTTone fft_tones[1000]; | |
157 int fft_tone_start; | |
158 int fft_tone_end; | |
159 FFTCoefficient fft_coefs[1000]; | |
160 int fft_coefs_index; | |
161 int fft_coefs_min_index[5]; | |
162 int fft_coefs_max_index[5]; | |
163 int fft_level_exp[6]; | |
8695 | 164 RDFTContext rdft_ctx; |
2914 | 165 QDM2FFT fft; |
166 | |
167 /// I/O data | |
6273 | 168 const uint8_t *compressed_data; |
2914 | 169 int compressed_size; |
170 float output_buffer[1024]; | |
171 | |
172 /// Synthesis filter | |
11369 | 173 DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2]; |
2914 | 174 int synth_buf_offset[MPA_MAX_CHANNELS]; |
11369 | 175 DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; |
2914 | 176 |
177 /// Mixed temporary data used in decoding | |
178 float tone_level[MPA_MAX_CHANNELS][30][64]; | |
179 int8_t coding_method[MPA_MAX_CHANNELS][30][64]; | |
180 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; | |
181 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; | |
182 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; | |
183 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; | |
184 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; | |
185 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; | |
186 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; | |
187 | |
188 // Flags | |
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189 int has_errors; ///< packet has errors |
2914 | 190 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type |
191 int do_synth_filter; ///< used to perform or skip synthesis filter | |
192 | |
193 int sub_packet; | |
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194 int noise_idx; ///< index for dithering noise table |
2914 | 195 } QDM2Context; |
196 | |
197 | |
198 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE]; | |
199 | |
200 static VLC vlc_tab_level; | |
201 static VLC vlc_tab_diff; | |
202 static VLC vlc_tab_run; | |
203 static VLC fft_level_exp_alt_vlc; | |
204 static VLC fft_level_exp_vlc; | |
205 static VLC fft_stereo_exp_vlc; | |
206 static VLC fft_stereo_phase_vlc; | |
207 static VLC vlc_tab_tone_level_idx_hi1; | |
208 static VLC vlc_tab_tone_level_idx_mid; | |
209 static VLC vlc_tab_tone_level_idx_hi2; | |
210 static VLC vlc_tab_type30; | |
211 static VLC vlc_tab_type34; | |
212 static VLC vlc_tab_fft_tone_offset[5]; | |
213 | |
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214 static const uint16_t qdm2_vlc_offs[] = { |
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215 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, |
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216 }; |
2914 | 217 |
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218 static av_cold void qdm2_init_vlc(void) |
2914 | 219 { |
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220 static int vlcs_initialized = 0; |
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221 static VLC_TYPE qdm2_table[3838][2]; |
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222 |
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223 if (!vlcs_initialized) { |
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224 |
9665 | 225 vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; |
226 vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; | |
227 init_vlc (&vlc_tab_level, 8, 24, | |
228 vlc_tab_level_huffbits, 1, 1, | |
229 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 230 |
9665 | 231 vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; |
232 vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; | |
233 init_vlc (&vlc_tab_diff, 8, 37, | |
234 vlc_tab_diff_huffbits, 1, 1, | |
235 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 236 |
9665 | 237 vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; |
238 vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; | |
239 init_vlc (&vlc_tab_run, 5, 6, | |
240 vlc_tab_run_huffbits, 1, 1, | |
241 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 242 |
9665 | 243 fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; |
244 fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3]; | |
245 init_vlc (&fft_level_exp_alt_vlc, 8, 28, | |
246 fft_level_exp_alt_huffbits, 1, 1, | |
247 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
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248 |
2914 | 249 |
9665 | 250 fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; |
251 fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; | |
252 init_vlc (&fft_level_exp_vlc, 8, 20, | |
253 fft_level_exp_huffbits, 1, 1, | |
254 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 255 |
9665 | 256 fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; |
257 fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5]; | |
258 init_vlc (&fft_stereo_exp_vlc, 6, 7, | |
259 fft_stereo_exp_huffbits, 1, 1, | |
260 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 261 |
9665 | 262 fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; |
263 fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6]; | |
264 init_vlc (&fft_stereo_phase_vlc, 6, 9, | |
265 fft_stereo_phase_huffbits, 1, 1, | |
266 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 267 |
9665 | 268 vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]]; |
269 vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7]; | |
270 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20, | |
271 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1, | |
272 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 273 |
9665 | 274 vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]]; |
275 vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8]; | |
276 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24, | |
277 vlc_tab_tone_level_idx_mid_huffbits, 1, 1, | |
278 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 279 |
9665 | 280 vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]]; |
281 vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9]; | |
282 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24, | |
283 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1, | |
284 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 285 |
9665 | 286 vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; |
287 vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; | |
288 init_vlc (&vlc_tab_type30, 6, 9, | |
289 vlc_tab_type30_huffbits, 1, 1, | |
290 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 291 |
9665 | 292 vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; |
293 vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; | |
294 init_vlc (&vlc_tab_type34, 5, 10, | |
295 vlc_tab_type34_huffbits, 1, 1, | |
296 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 297 |
9665 | 298 vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]]; |
299 vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12]; | |
300 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23, | |
301 vlc_tab_fft_tone_offset_0_huffbits, 1, 1, | |
302 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 303 |
9665 | 304 vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]]; |
305 vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13]; | |
306 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28, | |
307 vlc_tab_fft_tone_offset_1_huffbits, 1, 1, | |
308 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 309 |
9665 | 310 vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]]; |
311 vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14]; | |
312 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32, | |
313 vlc_tab_fft_tone_offset_2_huffbits, 1, 1, | |
314 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 315 |
9665 | 316 vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]]; |
317 vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15]; | |
318 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35, | |
319 vlc_tab_fft_tone_offset_3_huffbits, 1, 1, | |
320 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
2914 | 321 |
9665 | 322 vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]]; |
323 vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16]; | |
324 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38, | |
325 vlc_tab_fft_tone_offset_4_huffbits, 1, 1, | |
326 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); | |
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327 |
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328 vlcs_initialized=1; |
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329 } |
2914 | 330 } |
331 | |
332 | |
333 /* for floating point to fixed point conversion */ | |
7129 | 334 static const float f2i_scale = (float) (1 << (FRAC_BITS - 15)); |
2914 | 335 |
336 | |
337 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth) | |
338 { | |
339 int value; | |
340 | |
341 value = get_vlc2(gb, vlc->table, vlc->bits, depth); | |
342 | |
343 /* stage-2, 3 bits exponent escape sequence */ | |
344 if (value-- == 0) | |
345 value = get_bits (gb, get_bits (gb, 3) + 1); | |
346 | |
347 /* stage-3, optional */ | |
348 if (flag) { | |
349 int tmp = vlc_stage3_values[value]; | |
350 | |
351 if ((value & ~3) > 0) | |
352 tmp += get_bits (gb, (value >> 2)); | |
353 value = tmp; | |
354 } | |
355 | |
356 return value; | |
357 } | |
358 | |
359 | |
360 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth) | |
361 { | |
362 int value = qdm2_get_vlc (gb, vlc, 0, depth); | |
363 | |
364 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); | |
365 } | |
366 | |
367 | |
368 /** | |
369 * QDM2 checksum | |
370 * | |
371 * @param data pointer to data to be checksum'ed | |
372 * @param length data length | |
373 * @param value checksum value | |
374 * | |
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375 * @return 0 if checksum is OK |
2914 | 376 */ |
6273 | 377 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) { |
2914 | 378 int i; |
379 | |
380 for (i=0; i < length; i++) | |
381 value -= data[i]; | |
382 | |
383 return (uint16_t)(value & 0xffff); | |
384 } | |
385 | |
386 | |
387 /** | |
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388 * Fills a QDM2SubPacket structure with packet type, size, and data pointer. |
2914 | 389 * |
390 * @param gb bitreader context | |
391 * @param sub_packet packet under analysis | |
392 */ | |
393 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet) | |
394 { | |
395 sub_packet->type = get_bits (gb, 8); | |
396 | |
397 if (sub_packet->type == 0) { | |
398 sub_packet->size = 0; | |
399 sub_packet->data = NULL; | |
400 } else { | |
401 sub_packet->size = get_bits (gb, 8); | |
402 | |
403 if (sub_packet->type & 0x80) { | |
404 sub_packet->size <<= 8; | |
405 sub_packet->size |= get_bits (gb, 8); | |
406 sub_packet->type &= 0x7f; | |
407 } | |
408 | |
409 if (sub_packet->type == 0x7f) | |
410 sub_packet->type |= (get_bits (gb, 8) << 8); | |
411 | |
412 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data | |
413 } | |
414 | |
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415 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n", |
2914 | 416 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); |
417 } | |
418 | |
419 | |
420 /** | |
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421 * Return node pointer to first packet of requested type in list. |
2914 | 422 * |
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423 * @param list list of subpackets to be scanned |
2914 | 424 * @param type type of searched subpacket |
425 * @return node pointer for subpacket if found, else NULL | |
426 */ | |
427 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type) | |
428 { | |
429 while (list != NULL && list->packet != NULL) { | |
430 if (list->packet->type == type) | |
431 return list; | |
432 list = list->next; | |
433 } | |
434 return NULL; | |
435 } | |
436 | |
437 | |
438 /** | |
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439 * Replaces 8 elements with their average value. |
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440 * Called by qdm2_decode_superblock before starting subblock decoding. |
2914 | 441 * |
442 * @param q context | |
443 */ | |
444 static void average_quantized_coeffs (QDM2Context *q) | |
445 { | |
446 int i, j, n, ch, sum; | |
447 | |
448 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; | |
449 | |
450 for (ch = 0; ch < q->nb_channels; ch++) | |
451 for (i = 0; i < n; i++) { | |
452 sum = 0; | |
453 | |
454 for (j = 0; j < 8; j++) | |
455 sum += q->quantized_coeffs[ch][i][j]; | |
456 | |
457 sum /= 8; | |
458 if (sum > 0) | |
459 sum--; | |
460 | |
461 for (j=0; j < 8; j++) | |
462 q->quantized_coeffs[ch][i][j] = sum; | |
463 } | |
464 } | |
465 | |
466 | |
467 /** | |
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468 * Build subband samples with noise weighted by q->tone_level. |
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469 * Called by synthfilt_build_sb_samples. |
2914 | 470 * |
471 * @param q context | |
472 * @param sb subband index | |
473 */ | |
474 static void build_sb_samples_from_noise (QDM2Context *q, int sb) | |
475 { | |
476 int ch, j; | |
477 | |
478 FIX_NOISE_IDX(q->noise_idx); | |
479 | |
480 if (!q->nb_channels) | |
481 return; | |
482 | |
483 for (ch = 0; ch < q->nb_channels; ch++) | |
484 for (j = 0; j < 64; j++) { | |
485 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); | |
486 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); | |
487 } | |
488 } | |
489 | |
490 | |
491 /** | |
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492 * Called while processing data from subpackets 11 and 12. |
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493 * Used after making changes to coding_method array. |
2914 | 494 * |
495 * @param sb subband index | |
496 * @param channels number of channels | |
497 * @param coding_method q->coding_method[0][0][0] | |
498 */ | |
3076 | 499 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method) |
2914 | 500 { |
501 int j,k; | |
502 int ch; | |
503 int run, case_val; | |
504 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}; | |
505 | |
506 for (ch = 0; ch < channels; ch++) { | |
507 for (j = 0; j < 64; ) { | |
508 if((coding_method[ch][sb][j] - 8) > 22) { | |
509 run = 1; | |
510 case_val = 8; | |
511 } else { | |
3333 | 512 switch (switchtable[coding_method[ch][sb][j]-8]) { |
2914 | 513 case 0: run = 10; case_val = 10; break; |
514 case 1: run = 1; case_val = 16; break; | |
515 case 2: run = 5; case_val = 24; break; | |
516 case 3: run = 3; case_val = 30; break; | |
517 case 4: run = 1; case_val = 30; break; | |
518 case 5: run = 1; case_val = 8; break; | |
519 default: run = 1; case_val = 8; break; | |
520 } | |
521 } | |
522 for (k = 0; k < run; k++) | |
523 if (j + k < 128) | |
524 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) | |
525 if (k > 0) { | |
526 SAMPLES_NEEDED | |
527 //not debugged, almost never used | |
528 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t)); | |
529 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t)); | |
530 } | |
531 j += run; | |
532 } | |
533 } | |
534 } | |
535 | |
536 | |
537 /** | |
538 * Related to synthesis filter | |
539 * Called by process_subpacket_10 | |
540 * | |
541 * @param q context | |
542 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 | |
543 */ | |
544 static void fill_tone_level_array (QDM2Context *q, int flag) | |
545 { | |
546 int i, sb, ch, sb_used; | |
547 int tmp, tab; | |
548 | |
549 // This should never happen | |
550 if (q->nb_channels <= 0) | |
551 return; | |
552 | |
553 for (ch = 0; ch < q->nb_channels; ch++) | |
554 for (sb = 0; sb < 30; sb++) | |
555 for (i = 0; i < 8; i++) { | |
556 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) | |
557 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ | |
558 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; | |
559 else | |
560 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; | |
561 if(tmp < 0) | |
562 tmp += 0xff; | |
563 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; | |
564 } | |
565 | |
566 sb_used = QDM2_SB_USED(q->sub_sampling); | |
567 | |
568 if ((q->superblocktype_2_3 != 0) && !flag) { | |
569 for (sb = 0; sb < sb_used; sb++) | |
570 for (ch = 0; ch < q->nb_channels; ch++) | |
571 for (i = 0; i < 64; i++) { | |
572 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; | |
573 if (q->tone_level_idx[ch][sb][i] < 0) | |
574 q->tone_level[ch][sb][i] = 0; | |
575 else | |
576 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; | |
577 } | |
578 } else { | |
579 tab = q->superblocktype_2_3 ? 0 : 1; | |
580 for (sb = 0; sb < sb_used; sb++) { | |
581 if ((sb >= 4) && (sb <= 23)) { | |
582 for (ch = 0; ch < q->nb_channels; ch++) | |
583 for (i = 0; i < 64; i++) { | |
584 tmp = q->tone_level_idx_base[ch][sb][i / 8] - | |
585 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - | |
586 q->tone_level_idx_mid[ch][sb - 4][i / 8] - | |
587 q->tone_level_idx_hi2[ch][sb - 4]; | |
588 q->tone_level_idx[ch][sb][i] = tmp & 0xff; | |
589 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
590 q->tone_level[ch][sb][i] = 0; | |
591 else | |
592 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
593 } | |
594 } else { | |
595 if (sb > 4) { | |
596 for (ch = 0; ch < q->nb_channels; ch++) | |
597 for (i = 0; i < 64; i++) { | |
598 tmp = q->tone_level_idx_base[ch][sb][i / 8] - | |
599 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - | |
600 q->tone_level_idx_hi2[ch][sb - 4]; | |
601 q->tone_level_idx[ch][sb][i] = tmp & 0xff; | |
602 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
603 q->tone_level[ch][sb][i] = 0; | |
604 else | |
605 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
606 } | |
607 } else { | |
608 for (ch = 0; ch < q->nb_channels; ch++) | |
609 for (i = 0; i < 64; i++) { | |
610 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; | |
611 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) | |
612 q->tone_level[ch][sb][i] = 0; | |
613 else | |
614 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; | |
615 } | |
616 } | |
617 } | |
618 } | |
619 } | |
620 | |
621 return; | |
622 } | |
623 | |
624 | |
625 /** | |
626 * Related to synthesis filter | |
627 * Called by process_subpacket_11 | |
628 * c is built with data from subpacket 11 | |
629 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples | |
630 * | |
2967 | 631 * @param tone_level_idx |
2914 | 632 * @param tone_level_idx_temp |
633 * @param coding_method q->coding_method[0][0][0] | |
634 * @param nb_channels number of channels | |
635 * @param c coming from subpacket 11, passed as 8*c | |
636 * @param superblocktype_2_3 flag based on superblock packet type | |
637 * @param cm_table_select q->cm_table_select | |
638 */ | |
639 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, | |
640 sb_int8_array coding_method, int nb_channels, | |
641 int c, int superblocktype_2_3, int cm_table_select) | |
642 { | |
643 int ch, sb, j; | |
644 int tmp, acc, esp_40, comp; | |
645 int add1, add2, add3, add4; | |
646 int64_t multres; | |
647 | |
648 // This should never happen | |
649 if (nb_channels <= 0) | |
650 return; | |
651 | |
652 if (!superblocktype_2_3) { | |
653 /* This case is untested, no samples available */ | |
654 SAMPLES_NEEDED | |
655 for (ch = 0; ch < nb_channels; ch++) | |
656 for (sb = 0; sb < 30; sb++) { | |
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657 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer |
2914 | 658 add1 = tone_level_idx[ch][sb][j] - 10; |
659 if (add1 < 0) | |
660 add1 = 0; | |
661 add2 = add3 = add4 = 0; | |
662 if (sb > 1) { | |
663 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; | |
664 if (add2 < 0) | |
665 add2 = 0; | |
666 } | |
667 if (sb > 0) { | |
668 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; | |
669 if (add3 < 0) | |
670 add3 = 0; | |
671 } | |
672 if (sb < 29) { | |
673 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; | |
674 if (add4 < 0) | |
675 add4 = 0; | |
676 } | |
677 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; | |
678 if (tmp < 0) | |
679 tmp = 0; | |
680 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; | |
681 } | |
682 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; | |
683 } | |
684 acc = 0; | |
685 for (ch = 0; ch < nb_channels; ch++) | |
686 for (sb = 0; sb < 30; sb++) | |
687 for (j = 0; j < 64; j++) | |
688 acc += tone_level_idx_temp[ch][sb][j]; | |
9538 | 689 |
2914 | 690 multres = 0x66666667 * (acc * 10); |
691 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); | |
692 for (ch = 0; ch < nb_channels; ch++) | |
693 for (sb = 0; sb < 30; sb++) | |
694 for (j = 0; j < 64; j++) { | |
695 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; | |
696 if (comp < 0) | |
697 comp += 0xff; | |
698 comp /= 256; // signed shift | |
699 switch(sb) { | |
700 case 0: | |
701 if (comp < 30) | |
702 comp = 30; | |
703 comp += 15; | |
704 break; | |
705 case 1: | |
706 if (comp < 24) | |
707 comp = 24; | |
708 comp += 10; | |
709 break; | |
710 case 2: | |
711 case 3: | |
712 case 4: | |
713 if (comp < 16) | |
714 comp = 16; | |
715 } | |
716 if (comp <= 5) | |
717 tmp = 0; | |
718 else if (comp <= 10) | |
719 tmp = 10; | |
720 else if (comp <= 16) | |
721 tmp = 16; | |
722 else if (comp <= 24) | |
723 tmp = -1; | |
724 else | |
725 tmp = 0; | |
726 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; | |
727 } | |
728 for (sb = 0; sb < 30; sb++) | |
729 fix_coding_method_array(sb, nb_channels, coding_method); | |
730 for (ch = 0; ch < nb_channels; ch++) | |
731 for (sb = 0; sb < 30; sb++) | |
732 for (j = 0; j < 64; j++) | |
733 if (sb >= 10) { | |
734 if (coding_method[ch][sb][j] < 10) | |
735 coding_method[ch][sb][j] = 10; | |
736 } else { | |
737 if (sb >= 2) { | |
738 if (coding_method[ch][sb][j] < 16) | |
739 coding_method[ch][sb][j] = 16; | |
740 } else { | |
741 if (coding_method[ch][sb][j] < 30) | |
742 coding_method[ch][sb][j] = 30; | |
743 } | |
744 } | |
745 } else { // superblocktype_2_3 != 0 | |
746 for (ch = 0; ch < nb_channels; ch++) | |
747 for (sb = 0; sb < 30; sb++) | |
748 for (j = 0; j < 64; j++) | |
749 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; | |
750 } | |
751 | |
752 return; | |
753 } | |
754 | |
755 | |
756 /** | |
757 * | |
758 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8 | |
759 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used | |
760 * | |
761 * @param q context | |
762 * @param gb bitreader context | |
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763 * @param length packet length in bits |
2914 | 764 * @param sb_min lower subband processed (sb_min included) |
765 * @param sb_max higher subband processed (sb_max excluded) | |
766 */ | |
767 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max) | |
768 { | |
769 int sb, j, k, n, ch, run, channels; | |
770 int joined_stereo, zero_encoding, chs; | |
771 int type34_first; | |
772 float type34_div = 0; | |
773 float type34_predictor; | |
774 float samples[10], sign_bits[16]; | |
775 | |
776 if (length == 0) { | |
777 // If no data use noise | |
778 for (sb=sb_min; sb < sb_max; sb++) | |
779 build_sb_samples_from_noise (q, sb); | |
780 | |
781 return; | |
782 } | |
783 | |
784 for (sb = sb_min; sb < sb_max; sb++) { | |
785 FIX_NOISE_IDX(q->noise_idx); | |
786 | |
787 channels = q->nb_channels; | |
788 | |
789 if (q->nb_channels <= 1 || sb < 12) | |
790 joined_stereo = 0; | |
791 else if (sb >= 24) | |
792 joined_stereo = 1; | |
793 else | |
794 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0; | |
795 | |
796 if (joined_stereo) { | |
797 if (BITS_LEFT(length,gb) >= 16) | |
798 for (j = 0; j < 16; j++) | |
799 sign_bits[j] = get_bits1 (gb); | |
800 | |
801 for (j = 0; j < 64; j++) | |
802 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) | |
803 q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; | |
804 | |
805 fix_coding_method_array(sb, q->nb_channels, q->coding_method); | |
806 channels = 1; | |
807 } | |
808 | |
809 for (ch = 0; ch < channels; ch++) { | |
810 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0; | |
811 type34_predictor = 0.0; | |
812 type34_first = 1; | |
813 | |
814 for (j = 0; j < 128; ) { | |
815 switch (q->coding_method[ch][sb][j / 2]) { | |
816 case 8: | |
817 if (BITS_LEFT(length,gb) >= 10) { | |
818 if (zero_encoding) { | |
819 for (k = 0; k < 5; k++) { | |
820 if ((j + 2 * k) >= 128) | |
821 break; | |
822 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; | |
823 } | |
824 } else { | |
825 n = get_bits(gb, 8); | |
826 for (k = 0; k < 5; k++) | |
827 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
828 } | |
829 for (k = 0; k < 5; k++) | |
830 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
831 } else { | |
832 for (k = 0; k < 10; k++) | |
833 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
834 } | |
835 run = 10; | |
836 break; | |
837 | |
838 case 10: | |
839 if (BITS_LEFT(length,gb) >= 1) { | |
840 float f = 0.81; | |
841 | |
842 if (get_bits1(gb)) | |
843 f = -f; | |
844 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; | |
845 samples[0] = f; | |
846 } else { | |
847 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
848 } | |
849 run = 1; | |
850 break; | |
851 | |
852 case 16: | |
853 if (BITS_LEFT(length,gb) >= 10) { | |
854 if (zero_encoding) { | |
855 for (k = 0; k < 5; k++) { | |
856 if ((j + k) >= 128) | |
857 break; | |
858 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; | |
859 } | |
860 } else { | |
861 n = get_bits (gb, 8); | |
862 for (k = 0; k < 5; k++) | |
863 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; | |
864 } | |
865 } else { | |
866 for (k = 0; k < 5; k++) | |
867 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
868 } | |
869 run = 5; | |
870 break; | |
871 | |
872 case 24: | |
873 if (BITS_LEFT(length,gb) >= 7) { | |
874 n = get_bits(gb, 7); | |
875 for (k = 0; k < 3; k++) | |
876 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; | |
877 } else { | |
878 for (k = 0; k < 3; k++) | |
879 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
880 } | |
881 run = 3; | |
882 break; | |
883 | |
884 case 30: | |
885 if (BITS_LEFT(length,gb) >= 4) | |
886 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)]; | |
887 else | |
888 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
2967 | 889 |
2914 | 890 run = 1; |
891 break; | |
892 | |
893 case 34: | |
894 if (BITS_LEFT(length,gb) >= 7) { | |
895 if (type34_first) { | |
896 type34_div = (float)(1 << get_bits(gb, 2)); | |
897 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; | |
898 type34_predictor = samples[0]; | |
899 type34_first = 0; | |
900 } else { | |
901 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor; | |
902 type34_predictor = samples[0]; | |
903 } | |
904 } else { | |
905 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
906 } | |
907 run = 1; | |
908 break; | |
909 | |
910 default: | |
911 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); | |
912 run = 1; | |
913 break; | |
914 } | |
915 | |
916 if (joined_stereo) { | |
917 float tmp[10][MPA_MAX_CHANNELS]; | |
918 | |
919 for (k = 0; k < run; k++) { | |
920 tmp[k][0] = samples[k]; | |
921 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k]; | |
922 } | |
923 for (chs = 0; chs < q->nb_channels; chs++) | |
924 for (k = 0; k < run; k++) | |
925 if ((j + k) < 128) | |
926 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5); | |
927 } else { | |
928 for (k = 0; k < run; k++) | |
929 if ((j + k) < 128) | |
930 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5); | |
931 } | |
932 | |
933 j += run; | |
934 } // j loop | |
935 } // channel loop | |
936 } // subband loop | |
937 } | |
938 | |
939 | |
940 /** | |
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941 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]). |
2914 | 942 * This is similar to process_subpacket_9, but for a single channel and for element [0] |
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943 * same VLC tables as process_subpacket_9 are used. |
2914 | 944 * |
945 * @param q context | |
946 * @param quantized_coeffs pointer to quantized_coeffs[ch][0] | |
947 * @param gb bitreader context | |
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948 * @param length packet length in bits |
2914 | 949 */ |
950 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length) | |
951 { | |
952 int i, k, run, level, diff; | |
953 | |
954 if (BITS_LEFT(length,gb) < 16) | |
955 return; | |
956 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); | |
957 | |
958 quantized_coeffs[0] = level; | |
959 | |
960 for (i = 0; i < 7; ) { | |
961 if (BITS_LEFT(length,gb) < 16) | |
962 break; | |
963 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; | |
964 | |
965 if (BITS_LEFT(length,gb) < 16) | |
966 break; | |
967 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); | |
2967 | 968 |
2914 | 969 for (k = 1; k <= run; k++) |
970 quantized_coeffs[i + k] = (level + ((k * diff) / run)); | |
2967 | 971 |
2914 | 972 level += diff; |
973 i += run; | |
974 } | |
975 } | |
976 | |
977 | |
978 /** | |
979 * Related to synthesis filter, process data from packet 10 | |
980 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0 | |
981 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10 | |
982 * | |
983 * @param q context | |
984 * @param gb bitreader context | |
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985 * @param length packet length in bits |
2914 | 986 */ |
987 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length) | |
988 { | |
989 int sb, j, k, n, ch; | |
990 | |
991 for (ch = 0; ch < q->nb_channels; ch++) { | |
992 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length); | |
993 | |
994 if (BITS_LEFT(length,gb) < 16) { | |
995 memset(q->quantized_coeffs[ch][0], 0, 8); | |
996 break; | |
997 } | |
998 } | |
999 | |
1000 n = q->sub_sampling + 1; | |
1001 | |
1002 for (sb = 0; sb < n; sb++) | |
1003 for (ch = 0; ch < q->nb_channels; ch++) | |
1004 for (j = 0; j < 8; j++) { | |
1005 if (BITS_LEFT(length,gb) < 1) | |
1006 break; | |
1007 if (get_bits1(gb)) { | |
1008 for (k=0; k < 8; k++) { | |
1009 if (BITS_LEFT(length,gb) < 16) | |
1010 break; | |
1011 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); | |
1012 } | |
1013 } else { | |
1014 for (k=0; k < 8; k++) | |
1015 q->tone_level_idx_hi1[ch][sb][j][k] = 0; | |
1016 } | |
1017 } | |
1018 | |
1019 n = QDM2_SB_USED(q->sub_sampling) - 4; | |
1020 | |
1021 for (sb = 0; sb < n; sb++) | |
1022 for (ch = 0; ch < q->nb_channels; ch++) { | |
1023 if (BITS_LEFT(length,gb) < 16) | |
1024 break; | |
1025 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); | |
1026 if (sb > 19) | |
1027 q->tone_level_idx_hi2[ch][sb] -= 16; | |
1028 else | |
1029 for (j = 0; j < 8; j++) | |
1030 q->tone_level_idx_mid[ch][sb][j] = -16; | |
1031 } | |
1032 | |
1033 n = QDM2_SB_USED(q->sub_sampling) - 5; | |
1034 | |
1035 for (sb = 0; sb < n; sb++) | |
1036 for (ch = 0; ch < q->nb_channels; ch++) | |
1037 for (j = 0; j < 8; j++) { | |
1038 if (BITS_LEFT(length,gb) < 16) | |
1039 break; | |
1040 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; | |
1041 } | |
1042 } | |
1043 | |
1044 /** | |
1045 * Process subpacket 9, init quantized_coeffs with data from it | |
1046 * | |
1047 * @param q context | |
1048 * @param node pointer to node with packet | |
1049 */ | |
1050 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node) | |
1051 { | |
1052 GetBitContext gb; | |
1053 int i, j, k, n, ch, run, level, diff; | |
1054 | |
2916 | 1055 init_get_bits(&gb, node->packet->data, node->packet->size*8); |
2914 | 1056 |
1057 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function | |
1058 | |
1059 for (i = 1; i < n; i++) | |
1060 for (ch=0; ch < q->nb_channels; ch++) { | |
1061 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); | |
1062 q->quantized_coeffs[ch][i][0] = level; | |
1063 | |
1064 for (j = 0; j < (8 - 1); ) { | |
1065 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; | |
1066 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); | |
1067 | |
1068 for (k = 1; k <= run; k++) | |
1069 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run)); | |
1070 | |
1071 level += diff; | |
1072 j += run; | |
1073 } | |
1074 } | |
1075 | |
1076 for (ch = 0; ch < q->nb_channels; ch++) | |
1077 for (i = 0; i < 8; i++) | |
1078 q->quantized_coeffs[ch][0][i] = 0; | |
1079 } | |
1080 | |
1081 | |
1082 /** | |
1083 * Process subpacket 10 if not null, else | |
1084 * | |
1085 * @param q context | |
1086 * @param node pointer to node with packet | |
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1087 * @param length packet length in bits |
2914 | 1088 */ |
1089 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length) | |
1090 { | |
1091 GetBitContext gb; | |
1092 | |
2916 | 1093 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
2914 | 1094 |
1095 if (length != 0) { | |
1096 init_tone_level_dequantization(q, &gb, length); | |
1097 fill_tone_level_array(q, 1); | |
1098 } else { | |
1099 fill_tone_level_array(q, 0); | |
1100 } | |
1101 } | |
1102 | |
1103 | |
1104 /** | |
1105 * Process subpacket 11 | |
1106 * | |
1107 * @param q context | |
1108 * @param node pointer to node with packet | |
1109 * @param length packet length in bit | |
1110 */ | |
1111 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length) | |
1112 { | |
1113 GetBitContext gb; | |
1114 | |
2916 | 1115 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
2914 | 1116 if (length >= 32) { |
1117 int c = get_bits (&gb, 13); | |
1118 | |
1119 if (c > 3) | |
1120 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method, | |
1121 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select); | |
1122 } | |
1123 | |
1124 synthfilt_build_sb_samples(q, &gb, length, 0, 8); | |
1125 } | |
1126 | |
1127 | |
1128 /** | |
1129 * Process subpacket 12 | |
1130 * | |
1131 * @param q context | |
1132 * @param node pointer to node with packet | |
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1133 * @param length packet length in bits |
2914 | 1134 */ |
1135 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length) | |
1136 { | |
1137 GetBitContext gb; | |
1138 | |
2916 | 1139 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8)); |
2914 | 1140 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); |
1141 } | |
1142 | |
1143 /* | |
1144 * Process new subpackets for synthesis filter | |
1145 * | |
1146 * @param q context | |
1147 * @param list list with synthesis filter packets (list D) | |
1148 */ | |
1149 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list) | |
1150 { | |
1151 QDM2SubPNode *nodes[4]; | |
1152 | |
1153 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); | |
1154 if (nodes[0] != NULL) | |
1155 process_subpacket_9(q, nodes[0]); | |
1156 | |
1157 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); | |
1158 if (nodes[1] != NULL) | |
1159 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3); | |
1160 else | |
1161 process_subpacket_10(q, NULL, 0); | |
1162 | |
1163 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); | |
1164 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL) | |
1165 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3)); | |
1166 else | |
1167 process_subpacket_11(q, NULL, 0); | |
1168 | |
1169 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); | |
1170 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL) | |
1171 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3)); | |
1172 else | |
1173 process_subpacket_12(q, NULL, 0); | |
1174 } | |
1175 | |
1176 | |
1177 /* | |
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1178 * Decode superblock, fill packet lists. |
2914 | 1179 * |
1180 * @param q context | |
1181 */ | |
1182 static void qdm2_decode_super_block (QDM2Context *q) | |
1183 { | |
1184 GetBitContext gb; | |
1185 QDM2SubPacket header, *packet; | |
1186 int i, packet_bytes, sub_packet_size, sub_packets_D; | |
1187 unsigned int next_index = 0; | |
1188 | |
1189 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); | |
1190 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); | |
1191 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); | |
1192 | |
1193 q->sub_packets_B = 0; | |
1194 sub_packets_D = 0; | |
1195 | |
1196 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] | |
1197 | |
2916 | 1198 init_get_bits(&gb, q->compressed_data, q->compressed_size*8); |
2914 | 1199 qdm2_decode_sub_packet_header(&gb, &header); |
1200 | |
1201 if (header.type < 2 || header.type >= 8) { | |
1202 q->has_errors = 1; | |
1203 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n"); | |
1204 return; | |
1205 } | |
1206 | |
1207 q->superblocktype_2_3 = (header.type == 2 || header.type == 3); | |
1208 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); | |
1209 | |
2916 | 1210 init_get_bits(&gb, header.data, header.size*8); |
2914 | 1211 |
1212 if (header.type == 2 || header.type == 4 || header.type == 5) { | |
1213 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8); | |
1214 | |
1215 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); | |
1216 | |
1217 if (csum != 0) { | |
1218 q->has_errors = 1; | |
1219 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n"); | |
1220 return; | |
1221 } | |
1222 } | |
1223 | |
1224 q->sub_packet_list_B[0].packet = NULL; | |
1225 q->sub_packet_list_D[0].packet = NULL; | |
1226 | |
1227 for (i = 0; i < 6; i++) | |
1228 if (--q->fft_level_exp[i] < 0) | |
1229 q->fft_level_exp[i] = 0; | |
1230 | |
1231 for (i = 0; packet_bytes > 0; i++) { | |
1232 int j; | |
1233 | |
1234 q->sub_packet_list_A[i].next = NULL; | |
1235 | |
1236 if (i > 0) { | |
1237 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; | |
1238 | |
1239 /* seek to next block */ | |
2916 | 1240 init_get_bits(&gb, header.data, header.size*8); |
2914 | 1241 skip_bits(&gb, next_index*8); |
1242 | |
1243 if (next_index >= header.size) | |
1244 break; | |
1245 } | |
1246 | |
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1247 /* decode subpacket */ |
2914 | 1248 packet = &q->sub_packets[i]; |
1249 qdm2_decode_sub_packet_header(&gb, packet); | |
1250 next_index = packet->size + get_bits_count(&gb) / 8; | |
1251 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; | |
1252 | |
1253 if (packet->type == 0) | |
1254 break; | |
1255 | |
1256 if (sub_packet_size > packet_bytes) { | |
1257 if (packet->type != 10 && packet->type != 11 && packet->type != 12) | |
1258 break; | |
1259 packet->size += packet_bytes - sub_packet_size; | |
1260 } | |
1261 | |
1262 packet_bytes -= sub_packet_size; | |
1263 | |
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1264 /* add subpacket to 'all subpackets' list */ |
2914 | 1265 q->sub_packet_list_A[i].packet = packet; |
1266 | |
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1267 /* add subpacket to related list */ |
2914 | 1268 if (packet->type == 8) { |
1269 SAMPLES_NEEDED_2("packet type 8"); | |
1270 return; | |
1271 } else if (packet->type >= 9 && packet->type <= 12) { | |
1272 /* packets for MPEG Audio like Synthesis Filter */ | |
1273 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); | |
1274 } else if (packet->type == 13) { | |
1275 for (j = 0; j < 6; j++) | |
1276 q->fft_level_exp[j] = get_bits(&gb, 6); | |
1277 } else if (packet->type == 14) { | |
1278 for (j = 0; j < 6; j++) | |
1279 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); | |
1280 } else if (packet->type == 15) { | |
1281 SAMPLES_NEEDED_2("packet type 15") | |
1282 return; | |
1283 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) { | |
1284 /* packets for FFT */ | |
1285 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); | |
1286 } | |
1287 } // Packet bytes loop | |
1288 | |
1289 /* **************************************************************** */ | |
1290 if (q->sub_packet_list_D[0].packet != NULL) { | |
1291 process_synthesis_subpackets(q, q->sub_packet_list_D); | |
1292 q->do_synth_filter = 1; | |
1293 } else if (q->do_synth_filter) { | |
1294 process_subpacket_10(q, NULL, 0); | |
1295 process_subpacket_11(q, NULL, 0); | |
1296 process_subpacket_12(q, NULL, 0); | |
1297 } | |
1298 /* **************************************************************** */ | |
1299 } | |
1300 | |
1301 | |
1302 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, | |
1303 int offset, int duration, int channel, | |
1304 int exp, int phase) | |
1305 { | |
1306 if (q->fft_coefs_min_index[duration] < 0) | |
1307 q->fft_coefs_min_index[duration] = q->fft_coefs_index; | |
1308 | |
1309 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); | |
1310 q->fft_coefs[q->fft_coefs_index].channel = channel; | |
1311 q->fft_coefs[q->fft_coefs_index].offset = offset; | |
1312 q->fft_coefs[q->fft_coefs_index].exp = exp; | |
1313 q->fft_coefs[q->fft_coefs_index].phase = phase; | |
1314 q->fft_coefs_index++; | |
1315 } | |
1316 | |
1317 | |
1318 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) | |
1319 { | |
1320 int channel, stereo, phase, exp; | |
1321 int local_int_4, local_int_8, stereo_phase, local_int_10; | |
1322 int local_int_14, stereo_exp, local_int_20, local_int_28; | |
1323 int n, offset; | |
1324 | |
1325 local_int_4 = 0; | |
1326 local_int_28 = 0; | |
1327 local_int_20 = 2; | |
1328 local_int_8 = (4 - duration); | |
1329 local_int_10 = 1 << (q->group_order - duration - 1); | |
1330 offset = 1; | |
1331 | |
1332 while (1) { | |
1333 if (q->superblocktype_2_3) { | |
1334 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { | |
1335 offset = 1; | |
1336 if (n == 0) { | |
1337 local_int_4 += local_int_10; | |
1338 local_int_28 += (1 << local_int_8); | |
1339 } else { | |
1340 local_int_4 += 8*local_int_10; | |
1341 local_int_28 += (8 << local_int_8); | |
1342 } | |
1343 } | |
1344 offset += (n - 2); | |
1345 } else { | |
1346 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); | |
1347 while (offset >= (local_int_10 - 1)) { | |
1348 offset += (1 - (local_int_10 - 1)); | |
1349 local_int_4 += local_int_10; | |
1350 local_int_28 += (1 << local_int_8); | |
1351 } | |
1352 } | |
1353 | |
1354 if (local_int_4 >= q->group_size) | |
1355 return; | |
1356 | |
1357 local_int_14 = (offset >> local_int_8); | |
1358 | |
1359 if (q->nb_channels > 1) { | |
1360 channel = get_bits1(gb); | |
1361 stereo = get_bits1(gb); | |
1362 } else { | |
1363 channel = 0; | |
1364 stereo = 0; | |
1365 } | |
1366 | |
1367 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); | |
1368 exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; | |
1369 exp = (exp < 0) ? 0 : exp; | |
1370 | |
1371 phase = get_bits(gb, 3); | |
1372 stereo_exp = 0; | |
1373 stereo_phase = 0; | |
1374 | |
1375 if (stereo) { | |
1376 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); | |
1377 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); | |
1378 if (stereo_phase < 0) | |
1379 stereo_phase += 8; | |
1380 } | |
1381 | |
1382 if (q->frequency_range > (local_int_14 + 1)) { | |
1383 int sub_packet = (local_int_20 + local_int_28); | |
1384 | |
1385 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); | |
1386 if (stereo) | |
1387 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); | |
1388 } | |
1389 | |
1390 offset++; | |
1391 } | |
1392 } | |
1393 | |
1394 | |
1395 static void qdm2_decode_fft_packets (QDM2Context *q) | |
1396 { | |
1397 int i, j, min, max, value, type, unknown_flag; | |
1398 GetBitContext gb; | |
1399 | |
1400 if (q->sub_packet_list_B[0].packet == NULL) | |
1401 return; | |
1402 | |
6903 | 1403 /* reset minimum indexes for FFT coefficients */ |
2914 | 1404 q->fft_coefs_index = 0; |
1405 for (i=0; i < 5; i++) | |
1406 q->fft_coefs_min_index[i] = -1; | |
1407 | |
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1408 /* process subpackets ordered by type, largest type first */ |
2914 | 1409 for (i = 0, max = 256; i < q->sub_packets_B; i++) { |
7306 | 1410 QDM2SubPacket *packet= NULL; |
2914 | 1411 |
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1412 /* find subpacket with largest type less than max */ |
7306 | 1413 for (j = 0, min = 0; j < q->sub_packets_B; j++) { |
2914 | 1414 value = q->sub_packet_list_B[j].packet->type; |
1415 if (value > min && value < max) { | |
1416 min = value; | |
1417 packet = q->sub_packet_list_B[j].packet; | |
1418 } | |
1419 } | |
1420 | |
1421 max = min; | |
1422 | |
1423 /* check for errors (?) */ | |
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1424 if (!packet) |
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1425 return; |
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1426 |
2914 | 1427 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) |
1428 return; | |
1429 | |
1430 /* decode FFT tones */ | |
2916 | 1431 init_get_bits (&gb, packet->data, packet->size*8); |
2914 | 1432 |
1433 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) | |
1434 unknown_flag = 1; | |
1435 else | |
1436 unknown_flag = 0; | |
1437 | |
1438 type = packet->type; | |
1439 | |
1440 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { | |
1441 int duration = q->sub_sampling + 5 - (type & 15); | |
1442 | |
1443 if (duration >= 0 && duration < 4) | |
1444 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); | |
1445 } else if (type == 31) { | |
3320 | 1446 for (j=0; j < 4; j++) |
1447 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); | |
2914 | 1448 } else if (type == 46) { |
3320 | 1449 for (j=0; j < 6; j++) |
1450 q->fft_level_exp[j] = get_bits(&gb, 6); | |
1451 for (j=0; j < 4; j++) | |
1452 qdm2_fft_decode_tones(q, j, &gb, unknown_flag); | |
2914 | 1453 } |
1454 } // Loop on B packets | |
1455 | |
6903 | 1456 /* calculate maximum indexes for FFT coefficients */ |
2914 | 1457 for (i = 0, j = -1; i < 5; i++) |
1458 if (q->fft_coefs_min_index[i] >= 0) { | |
1459 if (j >= 0) | |
1460 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; | |
1461 j = i; | |
1462 } | |
1463 if (j >= 0) | |
1464 q->fft_coefs_max_index[j] = q->fft_coefs_index; | |
1465 } | |
1466 | |
1467 | |
1468 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) | |
1469 { | |
1470 float level, f[6]; | |
1471 int i; | |
1472 QDM2Complex c; | |
1473 const double iscale = 2.0*M_PI / 512.0; | |
1474 | |
1475 tone->phase += tone->phase_shift; | |
1476 | |
1477 /* calculate current level (maximum amplitude) of tone */ | |
1478 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; | |
1479 c.im = level * sin(tone->phase*iscale); | |
1480 c.re = level * cos(tone->phase*iscale); | |
1481 | |
1482 /* generate FFT coefficients for tone */ | |
1483 if (tone->duration >= 3 || tone->cutoff >= 3) { | |
8695 | 1484 tone->complex[0].im += c.im; |
1485 tone->complex[0].re += c.re; | |
1486 tone->complex[1].im -= c.im; | |
1487 tone->complex[1].re -= c.re; | |
2914 | 1488 } else { |
1489 f[1] = -tone->table[4]; | |
1490 f[0] = tone->table[3] - tone->table[0]; | |
1491 f[2] = 1.0 - tone->table[2] - tone->table[3]; | |
1492 f[3] = tone->table[1] + tone->table[4] - 1.0; | |
1493 f[4] = tone->table[0] - tone->table[1]; | |
1494 f[5] = tone->table[2]; | |
1495 for (i = 0; i < 2; i++) { | |
8695 | 1496 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; |
1497 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); | |
2914 | 1498 } |
1499 for (i = 0; i < 4; i++) { | |
8695 | 1500 tone->complex[i].re += c.re * f[i+2]; |
1501 tone->complex[i].im += c.im * f[i+2]; | |
2914 | 1502 } |
1503 } | |
1504 | |
1505 /* copy the tone if it has not yet died out */ | |
1506 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { | |
1507 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); | |
1508 q->fft_tone_end = (q->fft_tone_end + 1) % 1000; | |
1509 } | |
1510 } | |
1511 | |
1512 | |
1513 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) | |
1514 { | |
1515 int i, j, ch; | |
1516 const double iscale = 0.25 * M_PI; | |
1517 | |
1518 for (ch = 0; ch < q->channels; ch++) { | |
8695 | 1519 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); |
2914 | 1520 } |
1521 | |
1522 | |
1523 /* apply FFT tones with duration 4 (1 FFT period) */ | |
1524 if (q->fft_coefs_min_index[4] >= 0) | |
1525 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { | |
1526 float level; | |
1527 QDM2Complex c; | |
1528 | |
1529 if (q->fft_coefs[i].sub_packet != sub_packet) | |
1530 break; | |
1531 | |
1532 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; | |
1533 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]; | |
1534 | |
1535 c.re = level * cos(q->fft_coefs[i].phase * iscale); | |
1536 c.im = level * sin(q->fft_coefs[i].phase * iscale); | |
8695 | 1537 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; |
1538 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; | |
1539 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; | |
1540 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; | |
2914 | 1541 } |
1542 | |
1543 /* generate existing FFT tones */ | |
1544 for (i = q->fft_tone_end; i != q->fft_tone_start; ) { | |
1545 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); | |
1546 q->fft_tone_start = (q->fft_tone_start + 1) % 1000; | |
1547 } | |
1548 | |
1549 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ | |
1550 for (i = 0; i < 4; i++) | |
1551 if (q->fft_coefs_min_index[i] >= 0) { | |
1552 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { | |
1553 int offset, four_i; | |
1554 FFTTone tone; | |
1555 | |
1556 if (q->fft_coefs[j].sub_packet != sub_packet) | |
1557 break; | |
1558 | |
1559 four_i = (4 - i); | |
1560 offset = q->fft_coefs[j].offset >> four_i; | |
1561 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; | |
1562 | |
1563 if (offset < q->frequency_range) { | |
1564 if (offset < 2) | |
1565 tone.cutoff = offset; | |
1566 else | |
1567 tone.cutoff = (offset >= 60) ? 3 : 2; | |
1568 | |
1569 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 | 1570 tone.complex = &q->fft.complex[ch][offset]; |
6273 | 1571 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; |
2914 | 1572 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; |
1573 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); | |
1574 tone.duration = i; | |
1575 tone.time_index = 0; | |
1576 | |
1577 qdm2_fft_generate_tone(q, &tone); | |
1578 } | |
1579 } | |
1580 q->fft_coefs_min_index[i] = j; | |
1581 } | |
1582 } | |
1583 | |
1584 | |
1585 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) | |
1586 { | |
8695 | 1587 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; |
1588 int i; | |
1589 q->fft.complex[channel][0].re *= 2.0f; | |
1590 q->fft.complex[channel][0].im = 0.0f; | |
1591 ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); | |
2914 | 1592 /* add samples to output buffer */ |
1593 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++) | |
8695 | 1594 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain; |
2914 | 1595 } |
1596 | |
1597 | |
1598 /** | |
1599 * @param q context | |
1600 * @param index subpacket number | |
1601 */ | |
1602 static void qdm2_synthesis_filter (QDM2Context *q, int index) | |
1603 { | |
1604 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; | |
1605 int i, k, ch, sb_used, sub_sampling, dither_state = 0; | |
1606 | |
1607 /* copy sb_samples */ | |
1608 sb_used = QDM2_SB_USED(q->sub_sampling); | |
1609 | |
1610 for (ch = 0; ch < q->channels; ch++) | |
1611 for (i = 0; i < 8; i++) | |
1612 for (k=sb_used; k < SBLIMIT; k++) | |
1613 q->sb_samples[ch][(8 * index) + i][k] = 0; | |
1614 | |
1615 for (ch = 0; ch < q->nb_channels; ch++) { | |
1616 OUT_INT *samples_ptr = samples + ch; | |
1617 | |
1618 for (i = 0; i < 8; i++) { | |
1619 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]), | |
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1620 ff_mpa_synth_window, &dither_state, |
2914 | 1621 samples_ptr, q->nb_channels, |
1622 q->sb_samples[ch][(8 * index) + i]); | |
1623 samples_ptr += 32 * q->nb_channels; | |
1624 } | |
1625 } | |
1626 | |
1627 /* add samples to output buffer */ | |
1628 sub_sampling = (4 >> q->sub_sampling); | |
1629 | |
1630 for (ch = 0; ch < q->channels; ch++) | |
1631 for (i = 0; i < q->frame_size; i++) | |
1632 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16)); | |
1633 } | |
1634 | |
1635 | |
1636 /** | |
1637 * Init static data (does not depend on specific file) | |
1638 * | |
1639 * @param q context | |
1640 */ | |
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1641 static av_cold void qdm2_init(QDM2Context *q) { |
6350 | 1642 static int initialized = 0; |
2914 | 1643 |
6350 | 1644 if (initialized != 0) |
2914 | 1645 return; |
6350 | 1646 initialized = 1; |
2914 | 1647 |
1648 qdm2_init_vlc(); | |
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1649 ff_mpa_synth_init(ff_mpa_synth_window); |
2914 | 1650 softclip_table_init(); |
1651 rnd_table_init(); | |
1652 init_noise_samples(); | |
1653 | |
1654 av_log(NULL, AV_LOG_DEBUG, "init done\n"); | |
1655 } | |
1656 | |
1657 | |
1658 #if 0 | |
1659 static void dump_context(QDM2Context *q) | |
1660 { | |
1661 int i; | |
1662 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b); | |
1663 PRINT("compressed_data",q->compressed_data); | |
1664 PRINT("compressed_size",q->compressed_size); | |
1665 PRINT("frame_size",q->frame_size); | |
1666 PRINT("checksum_size",q->checksum_size); | |
1667 PRINT("channels",q->channels); | |
1668 PRINT("nb_channels",q->nb_channels); | |
1669 PRINT("fft_frame_size",q->fft_frame_size); | |
1670 PRINT("fft_size",q->fft_size); | |
1671 PRINT("sub_sampling",q->sub_sampling); | |
1672 PRINT("fft_order",q->fft_order); | |
1673 PRINT("group_order",q->group_order); | |
1674 PRINT("group_size",q->group_size); | |
1675 PRINT("sub_packet",q->sub_packet); | |
1676 PRINT("frequency_range",q->frequency_range); | |
1677 PRINT("has_errors",q->has_errors); | |
1678 PRINT("fft_tone_end",q->fft_tone_end); | |
1679 PRINT("fft_tone_start",q->fft_tone_start); | |
1680 PRINT("fft_coefs_index",q->fft_coefs_index); | |
1681 PRINT("coeff_per_sb_select",q->coeff_per_sb_select); | |
1682 PRINT("cm_table_select",q->cm_table_select); | |
1683 PRINT("noise_idx",q->noise_idx); | |
1684 | |
1685 for (i = q->fft_tone_start; i < q->fft_tone_end; i++) | |
1686 { | |
1687 FFTTone *t = &q->fft_tones[i]; | |
2967 | 1688 |
2914 | 1689 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i); |
1690 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level); | |
1691 // PRINT(" level", t->level); | |
1692 PRINT(" phase", t->phase); | |
1693 PRINT(" phase_shift", t->phase_shift); | |
1694 PRINT(" duration", t->duration); | |
1695 PRINT(" samples_im", t->samples_im); | |
1696 PRINT(" samples_re", t->samples_re); | |
1697 PRINT(" table", t->table); | |
1698 } | |
1699 | |
1700 } | |
1701 #endif | |
1702 | |
1703 | |
1704 /** | |
1705 * Init parameters from codec extradata | |
1706 */ | |
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1707 static av_cold int qdm2_decode_init(AVCodecContext *avctx) |
2914 | 1708 { |
1709 QDM2Context *s = avctx->priv_data; | |
1710 uint8_t *extradata; | |
1711 int extradata_size; | |
1712 int tmp_val, tmp, size; | |
2967 | 1713 |
2914 | 1714 /* extradata parsing |
2967 | 1715 |
2914 | 1716 Structure: |
1717 wave { | |
1718 frma (QDM2) | |
1719 QDCA | |
1720 QDCP | |
1721 } | |
2967 | 1722 |
2914 | 1723 32 size (including this field) |
1724 32 tag (=frma) | |
1725 32 type (=QDM2 or QDMC) | |
2967 | 1726 |
2914 | 1727 32 size (including this field, in bytes) |
1728 32 tag (=QDCA) // maybe mandatory parameters | |
1729 32 unknown (=1) | |
1730 32 channels (=2) | |
1731 32 samplerate (=44100) | |
1732 32 bitrate (=96000) | |
1733 32 block size (=4096) | |
1734 32 frame size (=256) (for one channel) | |
1735 32 packet size (=1300) | |
2967 | 1736 |
2914 | 1737 32 size (including this field, in bytes) |
1738 32 tag (=QDCP) // maybe some tuneable parameters | |
1739 32 float1 (=1.0) | |
1740 32 zero ? | |
1741 32 float2 (=1.0) | |
1742 32 float3 (=1.0) | |
1743 32 unknown (27) | |
1744 32 unknown (8) | |
1745 32 zero ? | |
1746 */ | |
1747 | |
1748 if (!avctx->extradata || (avctx->extradata_size < 48)) { | |
1749 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); | |
1750 return -1; | |
1751 } | |
1752 | |
1753 extradata = avctx->extradata; | |
1754 extradata_size = avctx->extradata_size; | |
1755 | |
1756 while (extradata_size > 7) { | |
1757 if (!memcmp(extradata, "frmaQDM", 7)) | |
1758 break; | |
1759 extradata++; | |
1760 extradata_size--; | |
1761 } | |
1762 | |
1763 if (extradata_size < 12) { | |
1764 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", | |
1765 extradata_size); | |
1766 return -1; | |
1767 } | |
1768 | |
1769 if (memcmp(extradata, "frmaQDM", 7)) { | |
1770 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); | |
1771 return -1; | |
1772 } | |
1773 | |
1774 if (extradata[7] == 'C') { | |
1775 // s->is_qdmc = 1; | |
1776 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); | |
1777 return -1; | |
1778 } | |
1779 | |
1780 extradata += 8; | |
1781 extradata_size -= 8; | |
1782 | |
4364 | 1783 size = AV_RB32(extradata); |
2914 | 1784 |
1785 if(size > extradata_size){ | |
1786 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", | |
1787 extradata_size, size); | |
1788 return -1; | |
1789 } | |
1790 | |
1791 extradata += 4; | |
1792 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); | |
4364 | 1793 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { |
2914 | 1794 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); |
1795 return -1; | |
1796 } | |
1797 | |
1798 extradata += 8; | |
1799 | |
4364 | 1800 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); |
2914 | 1801 extradata += 4; |
1802 | |
4364 | 1803 avctx->sample_rate = AV_RB32(extradata); |
2914 | 1804 extradata += 4; |
1805 | |
4364 | 1806 avctx->bit_rate = AV_RB32(extradata); |
2914 | 1807 extradata += 4; |
1808 | |
4364 | 1809 s->group_size = AV_RB32(extradata); |
2914 | 1810 extradata += 4; |
1811 | |
4364 | 1812 s->fft_size = AV_RB32(extradata); |
2914 | 1813 extradata += 4; |
1814 | |
4364 | 1815 s->checksum_size = AV_RB32(extradata); |
2914 | 1816 |
1817 s->fft_order = av_log2(s->fft_size) + 1; | |
1818 s->fft_frame_size = 2 * s->fft_size; // complex has two floats | |
1819 | |
1820 // something like max decodable tones | |
1821 s->group_order = av_log2(s->group_size) + 1; | |
1822 s->frame_size = s->group_size / 16; // 16 iterations per super block | |
1823 | |
2954 | 1824 s->sub_sampling = s->fft_order - 7; |
2914 | 1825 s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); |
2967 | 1826 |
2914 | 1827 switch ((s->sub_sampling * 2 + s->channels - 1)) { |
1828 case 0: tmp = 40; break; | |
1829 case 1: tmp = 48; break; | |
1830 case 2: tmp = 56; break; | |
1831 case 3: tmp = 72; break; | |
1832 case 4: tmp = 80; break; | |
1833 case 5: tmp = 100;break; | |
1834 default: tmp=s->sub_sampling; break; | |
1835 } | |
1836 tmp_val = 0; | |
1837 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; | |
1838 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; | |
1839 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; | |
1840 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; | |
1841 s->cm_table_select = tmp_val; | |
1842 | |
1843 if (s->sub_sampling == 0) | |
2954 | 1844 tmp = 7999; |
2914 | 1845 else |
1846 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; | |
1847 /* | |
2954 | 1848 0: 7999 -> 0 |
2914 | 1849 1: 20000 -> 2 |
1850 2: 28000 -> 2 | |
1851 */ | |
1852 if (tmp < 8000) | |
1853 s->coeff_per_sb_select = 0; | |
1854 else if (tmp <= 16000) | |
1855 s->coeff_per_sb_select = 1; | |
1856 else | |
1857 s->coeff_per_sb_select = 2; | |
1858 | |
8695 | 1859 // Fail on unknown fft order |
2954 | 1860 if ((s->fft_order < 7) || (s->fft_order > 9)) { |
2914 | 1861 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); |
2954 | 1862 return -1; |
1863 } | |
2914 | 1864 |
11391 | 1865 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); |
2914 | 1866 |
1867 qdm2_init(s); | |
2967 | 1868 |
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1869 avctx->sample_fmt = SAMPLE_FMT_S16; |
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1870 |
2914 | 1871 // dump_context(s); |
1872 return 0; | |
1873 } | |
1874 | |
1875 | |
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1876 static av_cold int qdm2_decode_close(AVCodecContext *avctx) |
2914 | 1877 { |
1878 QDM2Context *s = avctx->priv_data; | |
1879 | |
8695 | 1880 ff_rdft_end(&s->rdft_ctx); |
2967 | 1881 |
2914 | 1882 return 0; |
1883 } | |
1884 | |
1885 | |
6273 | 1886 static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) |
2914 | 1887 { |
1888 int ch, i; | |
1889 const int frame_size = (q->frame_size * q->channels); | |
2967 | 1890 |
2914 | 1891 /* select input buffer */ |
1892 q->compressed_data = in; | |
1893 q->compressed_size = q->checksum_size; | |
1894 | |
1895 // dump_context(q); | |
1896 | |
1897 /* copy old block, clear new block of output samples */ | |
1898 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); | |
1899 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); | |
1900 | |
1901 /* decode block of QDM2 compressed data */ | |
1902 if (q->sub_packet == 0) { | |
1903 q->has_errors = 0; // zero it for a new super block | |
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1904 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); |
2914 | 1905 qdm2_decode_super_block(q); |
1906 } | |
1907 | |
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1908 /* parse subpackets */ |
2914 | 1909 if (!q->has_errors) { |
1910 if (q->sub_packet == 2) | |
1911 qdm2_decode_fft_packets(q); | |
1912 | |
1913 qdm2_fft_tone_synthesizer(q, q->sub_packet); | |
1914 } | |
1915 | |
1916 /* sound synthesis stage 1 (FFT) */ | |
1917 for (ch = 0; ch < q->channels; ch++) { | |
1918 qdm2_calculate_fft(q, ch, q->sub_packet); | |
1919 | |
1920 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { | |
1921 SAMPLES_NEEDED_2("has errors, and C list is not empty") | |
1922 return; | |
1923 } | |
1924 } | |
1925 | |
1926 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ | |
1927 if (!q->has_errors && q->do_synth_filter) | |
1928 qdm2_synthesis_filter(q, q->sub_packet); | |
1929 | |
1930 q->sub_packet = (q->sub_packet + 1) % 16; | |
1931 | |
1932 /* clip and convert output float[] to 16bit signed samples */ | |
1933 for (i = 0; i < frame_size; i++) { | |
1934 int value = (int)q->output_buffer[i]; | |
1935 | |
1936 if (value > SOFTCLIP_THRESHOLD) | |
1937 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; | |
1938 else if (value < -SOFTCLIP_THRESHOLD) | |
1939 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; | |
1940 | |
1941 out[i] = value; | |
1942 } | |
1943 } | |
1944 | |
1945 | |
1946 static int qdm2_decode_frame(AVCodecContext *avctx, | |
1947 void *data, int *data_size, | |
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1948 AVPacket *avpkt) |
2914 | 1949 { |
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1950 const uint8_t *buf = avpkt->data; |
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1951 int buf_size = avpkt->size; |
2914 | 1952 QDM2Context *s = avctx->priv_data; |
1953 | |
3158 | 1954 if(!buf) |
2914 | 1955 return 0; |
3158 | 1956 if(buf_size < s->checksum_size) |
1957 return -1; | |
2914 | 1958 |
1959 *data_size = s->channels * s->frame_size * sizeof(int16_t); | |
1960 | |
1961 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", | |
1962 buf_size, buf, s->checksum_size, data, *data_size); | |
1963 | |
1964 qdm2_decode(s, buf, data); | |
1965 | |
1966 // reading only when next superblock found | |
1967 if (s->sub_packet == 0) { | |
1968 return s->checksum_size; | |
1969 } | |
1970 | |
1971 return 0; | |
1972 } | |
1973 | |
1974 AVCodec qdm2_decoder = | |
1975 { | |
1976 .name = "qdm2", | |
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1977 .type = AVMEDIA_TYPE_AUDIO, |
2914 | 1978 .id = CODEC_ID_QDM2, |
1979 .priv_data_size = sizeof(QDM2Context), | |
1980 .init = qdm2_decode_init, | |
1981 .close = qdm2_decode_close, | |
1982 .decode = qdm2_decode_frame, | |
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1983 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), |
2914 | 1984 }; |