History

aacdec: move spectrum decode and dequantization to a new file

Lynne committed 2 years ago

Commit 41ae2b01 parent 5c026e610 files · +626 −531Browse files

Showing the changes to libavcodec/aac (6 of 10 files). Show all

Unified · Split

6 files
  1. libavcodec/aac/aacdec.c +4 −0
  2. libavcodec/aac/aacdec_dsp_template.c +1 −1
  3. libavcodec/aac/aacdec_fixed.c +5 −0
  4. libavcodec/aac/aacdec_fixed_dequant.h +174 −0
  5. libavcodec/aac/aacdec_float.c +74 −0
  6. libavcodec/aac/aacdec_proc_template.c +354 −0
libavcodec/aac/aacdec.c+4−0
@@ -48,6 +48,9 @@ · 47 unmodified lines
48extern const AACDecDSP aac_dsp;48extern const AACDecDSP aac_dsp;
49extern const AACDecDSP aac_dsp_fixed;49extern const AACDecDSP aac_dsp_fixed;
50 50
51extern const AACDecProc aac_proc;
52extern const AACDecProc aac_proc_fixed;
53
51av_cold int ff_aac_decode_close(AVCodecContext *avctx)54av_cold int ff_aac_decode_close(AVCodecContext *avctx)
52{55{
53 AACDecContext *ac = avctx->priv_data;56 AACDecContext *ac = avctx->priv_data;
@@ -119,6 +122,7 @@ av_cold int ff_aac_decode_init_common(AVCodecContext *avctx) · 65 unmodified lines
119 return ret;122 return ret;
120 123
121 ac->dsp = is_fixed ? aac_dsp_fixed : aac_dsp;124 ac->dsp = is_fixed ? aac_dsp_fixed : aac_dsp;
125 ac->proc = is_fixed ? aac_proc_fixed : aac_proc;
122 126
123 ac->dsp.init_tables();127 ac->dsp.init_tables();
124 128
libavcodec/aac/aacdec_dsp_template.c+1−1
@@ -151,7 +151,7 @@ static void AAC_RENAME(apply_intensity_stereo)(AACDecContext *ac, · 150 unmodified lines
151 scale = c * sce1->AAC_RENAME(sf)[idx];151 scale = c * sce1->AAC_RENAME(sf)[idx];
152 for (group = 0; group < ics->group_len[g]; group++)152 for (group = 0; group < ics->group_len[g]; group++)
153#if USE_FIXED153#if USE_FIXED
154 ac->subband_scale(coef1 + group * 128 + offsets[i],154 subband_scale(coef1 + group * 128 + offsets[i],
155 coef0 + group * 128 + offsets[i],155 coef0 + group * 128 + offsets[i],
156 scale,156 scale,
157 23,157 23,
libavcodec/aac/aacdec_fixed.c+5−0
@@ -38,6 +38,7 @@ · 37 unmodified lines
38#include "libavcodec/aactab.h"38#include "libavcodec/aactab.h"
39#include "libavcodec/sinewin_fixed_tablegen.h"39#include "libavcodec/sinewin_fixed_tablegen.h"
40#include "libavcodec/kbdwin.h"40#include "libavcodec/kbdwin.h"
41#include "libavcodec/cbrt_data.h"
41 42
42DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME2(aac_kbd_long_1024))[1024];43DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME2(aac_kbd_long_1024))[1024];
43DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME2(aac_kbd_short_128))[128];44DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME2(aac_kbd_short_128))[128];
@@ -61,4 +62,8 @@ static void init_tables_fixed(void) · 17 unmodified lines
61 ff_thread_once(&init_fixed_once, init_tables_fixed_fn);62 ff_thread_once(&init_fixed_once, init_tables_fixed_fn);
62}63}
63 64
65/** Dequantization-related */
66#include "aacdec_fixed_dequant.h"
67
64#include "aacdec_dsp_template.c"68#include "aacdec_dsp_template.c"
69#include "aacdec_proc_template.c"
libavcodec/aac/aacdec_fixed_dequant.h+174−0
@@ -0,0 +1,174 @@
1/*
2 * AAC decoder
3 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
4 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
5 * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
6 *
7 * AAC LATM decoder
8 * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
9 * Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
10 *
11 * AAC decoder fixed-point implementation
12 * Copyright (c) 2013
13 * MIPS Technologies, Inc., California.
14 *
15 * This file is part of FFmpeg.
16 *
17 * FFmpeg is free software; you can redistribute it and/or
18 * modify it under the terms of the GNU Lesser General Public
19 * License as published by the Free Software Foundation; either
20 * version 2.1 of the License, or (at your option) any later version.
21 *
22 * FFmpeg is distributed in the hope that it will be useful,
23 * but WITHOUT ANY WARRANTY; without even the implied warranty of
24 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
25 * Lesser General Public License for more details.
26 *
27 * You should have received a copy of the GNU Lesser General Public
28 * License along with FFmpeg; if not, write to the Free Software
29 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
30 */
31
32#ifndef AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H
33#define AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H
34
35#include "aacdec_tab.h"
36
37static void inline vector_pow43(int *coefs, int len)
38{
39 int i, coef;
40
41 for (i=0; i<len; i++) {
42 coef = coefs[i];
43 if (coef < 0)
44 coef = -(int)ff_cbrt_tab_fixed[(-coef) & 8191];
45 else
46 coef = (int)ff_cbrt_tab_fixed[ coef & 8191];
47 coefs[i] = coef;
48 }
49}
50
51/* 2^0, 2^0.25, 2^0.5, 2^0.75 */
52static const int exp2tab[4] = {
53 Q31(1.0000000000/2), Q31(1.1892071150/2),
54 Q31(1.4142135624/2), Q31(1.6817928305/2)
55};
56
57static void inline subband_scale(int *dst, int *src, int scale,
58 int offset, int len, void *log_context)
59{
60 int ssign = scale < 0 ? -1 : 1;
61 int s = FFABS(scale);
62 unsigned int round;
63 int i, out, c = exp2tab[s & 3];
64
65 s = offset - (s >> 2);
66
67 if (s > 31) {
68 for (i=0; i<len; i++) {
69 dst[i] = 0;
70 }
71 } else if (s > 0) {
72 round = 1 << (s-1);
73 for (i=0; i<len; i++) {
74 out = (int)(((int64_t)src[i] * c) >> 32);
75 dst[i] = ((int)(out+round) >> s) * ssign;
76 }
77 } else if (s > -32) {
78 s = s + 32;
79 round = 1U << (s-1);
80 for (i=0; i<len; i++) {
81 out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
82 dst[i] = out * (unsigned)ssign;
83 }
84 } else {
85 av_log(log_context, AV_LOG_ERROR, "Overflow in subband_scale()\n");
86 }
87}
88
89static void noise_scale(int *coefs, int scale, int band_energy, int len)
90{
91 int s = -scale;
92 unsigned int round;
93 int i, out, c = exp2tab[s & 3];
94 int nlz = 0;
95
96 av_assert0(s >= 0);
97 while (band_energy > 0x7fff) {
98 band_energy >>= 1;
99 nlz++;
100 }
101 c /= band_energy;
102 s = 21 + nlz - (s >> 2);
103
104 if (s > 31) {
105 for (i=0; i<len; i++) {
106 coefs[i] = 0;
107 }
108 } else if (s >= 0) {
109 round = s ? 1 << (s-1) : 0;
110 for (i=0; i<len; i++) {
111 out = (int)(((int64_t)coefs[i] * c) >> 32);
112 coefs[i] = -((int)(out+round) >> s);
113 }
114 }
115 else {
116 s = s + 32;
117 if (s > 0) {
118 round = 1 << (s-1);
119 for (i=0; i<len; i++) {
120 out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
121 coefs[i] = -out;
122 }
123 } else {
124 for (i=0; i<len; i++)
125 coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
126 }
127 }
128}
129
130static inline int *DEC_SPAIR(int *dst, unsigned idx)
131{
132 dst[0] = (idx & 15) - 4;
133 dst[1] = (idx >> 4 & 15) - 4;
134
135 return dst + 2;
136}
137
138static inline int *DEC_SQUAD(int *dst, unsigned idx)
139{
140 dst[0] = (idx & 3) - 1;
141 dst[1] = (idx >> 2 & 3) - 1;
142 dst[2] = (idx >> 4 & 3) - 1;
143 dst[3] = (idx >> 6 & 3) - 1;
144
145 return dst + 4;
146}
147
148static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
149{
150 dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
151 dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
152
153 return dst + 2;
154}
155
156static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
157{
158 unsigned nz = idx >> 12;
159
160 dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
161 sign <<= nz & 1;
162 nz >>= 1;
163 dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
164 sign <<= nz & 1;
165 nz >>= 1;
166 dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
167 sign <<= nz & 1;
168 nz >>= 1;
169 dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
170
171 return dst + 4;
172}
173
174#endif /* AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H */
libavcodec/aac/aacdec_float.c+74−0
@@ -38,6 +38,7 @@ · 37 unmodified lines
38#include "libavcodec/aactab.h"38#include "libavcodec/aactab.h"
39#include "libavcodec/sinewin.h"39#include "libavcodec/sinewin.h"
40#include "libavcodec/kbdwin.h"40#include "libavcodec/kbdwin.h"
41#include "libavcodec/cbrt_data.h"
41 42
42DECLARE_ALIGNED(32, static float, sine_120)[120];43DECLARE_ALIGNED(32, static float, sine_120)[120];
43DECLARE_ALIGNED(32, static float, sine_960)[960];44DECLARE_ALIGNED(32, static float, sine_960)[960];
@@ -63,4 +64,77 @@ static void init_tables(void) · 19 unmodified lines
63 ff_thread_once(&init_float_once, init_tables_float_fn);64 ff_thread_once(&init_float_once, init_tables_float_fn);
64}65}
65 66
67/** Dequantization-related **/
68#include "aacdec_tab.h"
69#include "libavutil/intfloat.h"
70
71#ifndef VMUL2
72static inline float *VMUL2(float *dst, const float *v, unsigned idx,
73 const float *scale)
74{
75 float s = *scale;
76 *dst++ = v[idx & 15] * s;
77 *dst++ = v[idx>>4 & 15] * s;
78 return dst;
79}
80#endif
81
82#ifndef VMUL4
83static inline float *VMUL4(float *dst, const float *v, unsigned idx,
84 const float *scale)
85{
86 float s = *scale;
87 *dst++ = v[idx & 3] * s;
88 *dst++ = v[idx>>2 & 3] * s;
89 *dst++ = v[idx>>4 & 3] * s;
90 *dst++ = v[idx>>6 & 3] * s;
91 return dst;
92}
93#endif
94
95#ifndef VMUL2S
96static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
97 unsigned sign, const float *scale)
98{
99 union av_intfloat32 s0, s1;
100
101 s0.f = s1.f = *scale;
102 s0.i ^= sign >> 1 << 31;
103 s1.i ^= sign << 31;
104
105 *dst++ = v[idx & 15] * s0.f;
106 *dst++ = v[idx>>4 & 15] * s1.f;
107
108 return dst;
109}
110#endif
111
112#ifndef VMUL4S
113static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
114 unsigned sign, const float *scale)
115{
116 unsigned nz = idx >> 12;
117 union av_intfloat32 s = { .f = *scale };
118 union av_intfloat32 t;
119
120 t.i = s.i ^ (sign & 1U<<31);
121 *dst++ = v[idx & 3] * t.f;
122
123 sign <<= nz & 1; nz >>= 1;
124 t.i = s.i ^ (sign & 1U<<31);
125 *dst++ = v[idx>>2 & 3] * t.f;
126
127 sign <<= nz & 1; nz >>= 1;
128 t.i = s.i ^ (sign & 1U<<31);
129 *dst++ = v[idx>>4 & 3] * t.f;
130
131 sign <<= nz & 1;
132 t.i = s.i ^ (sign & 1U<<31);
133 *dst++ = v[idx>>6 & 3] * t.f;
134
135 return dst;
136}
137#endif
138
66#include "aacdec_dsp_template.c"139#include "aacdec_dsp_template.c"
140#include "aacdec_proc_template.c"
libavcodec/aac/aacdec_proc_template.c+354−0
@@ -0,0 +1,354 @@
1/*
2 * AAC decoder
3 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
4 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
5 * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
6 *
7 * AAC LATM decoder
8 * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
9 * Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
10 *
11 * AAC decoder fixed-point implementation
12 * Copyright (c) 2013
13 * MIPS Technologies, Inc., California.
14 *
15 * This file is part of FFmpeg.
16 *
17 * FFmpeg is free software; you can redistribute it and/or
18 * modify it under the terms of the GNU Lesser General Public
19 * License as published by the Free Software Foundation; either
20 * version 2.1 of the License, or (at your option) any later version.
21 *
22 * FFmpeg is distributed in the hope that it will be useful,
23 * but WITHOUT ANY WARRANTY; without even the implied warranty of
24 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
25 * Lesser General Public License for more details.
26 *
27 * You should have received a copy of the GNU Lesser General Public
28 * License along with FFmpeg; if not, write to the Free Software
29 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
30 */
31
32/**
33 * linear congruential pseudorandom number generator
34 *
35 * @param previous_val pointer to the current state of the generator
36 *
37 * @return Returns a 32-bit pseudorandom integer
38 */
39static av_always_inline int lcg_random(unsigned previous_val)
40{
41 union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
42 return v.s;
43}
44
45/**
46 * Decode spectral data; reference: table 4.50.
47 * Dequantize and scale spectral data; reference: 4.6.3.3.
48 *
49 * @param coef array of dequantized, scaled spectral data
50 * @param sf array of scalefactors or intensity stereo positions
51 * @param pulse_present set if pulses are present
52 * @param pulse pointer to pulse data struct
53 * @param band_type array of the used band type
54 *
55 * @return Returns error status. 0 - OK, !0 - error
56 */
57static int AAC_RENAME(decode_spectrum_and_dequant)(AACDecContext *ac,
58 GetBitContext *gb,
59 const Pulse *pulse,
60 SingleChannelElement *sce)
61{
62 int i, k, g, idx = 0;
63 INTFLOAT *coef = sce->AAC_RENAME(coeffs);
64 IndividualChannelStream *ics = &sce->ics;
65 const int c = 1024 / ics->num_windows;
66 const uint16_t *offsets = ics->swb_offset;
67 const INTFLOAT *sf = sce->AAC_RENAME(sf);
68 const enum BandType *band_type = sce->band_type;
69 INTFLOAT *coef_base = coef;
70
71 for (g = 0; g < ics->num_windows; g++)
72 memset(coef + g * 128 + offsets[ics->max_sfb], 0,
73 sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
74
75 for (g = 0; g < ics->num_window_groups; g++) {
76 unsigned g_len = ics->group_len[g];
77
78 for (i = 0; i < ics->max_sfb; i++, idx++) {
79 const unsigned cbt_m1 = band_type[idx] - 1;
80 INTFLOAT *cfo = coef + offsets[i];
81 int off_len = offsets[i + 1] - offsets[i];
82 int group;
83
84 if (cbt_m1 >= INTENSITY_BT2 - 1) {
85 for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
86 memset(cfo, 0, off_len * sizeof(*cfo));
87 }
88 } else if (cbt_m1 == NOISE_BT - 1) {
89 for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
90 INTFLOAT band_energy;
91#if USE_FIXED
92 for (k = 0; k < off_len; k++) {
93 ac->random_state = lcg_random(ac->random_state);
94 cfo[k] = ac->random_state >> 3;
95 }
96
97 band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
98 band_energy = fixed_sqrt(band_energy, 31);
99 noise_scale(cfo, sf[idx], band_energy, off_len);
100#else
101 float scale;
102
103 for (k = 0; k < off_len; k++) {
104 ac->random_state = lcg_random(ac->random_state);
105 cfo[k] = ac->random_state;
106 }
107
108 band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
109 scale = sf[idx] / sqrtf(band_energy);
110 ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
111#endif /* USE_FIXED */
112 }
113 } else {
114#if !USE_FIXED
115 const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
116#endif /* !USE_FIXED */
117 const VLCElem *vlc_tab = ff_vlc_spectral[cbt_m1];
118 OPEN_READER(re, gb);
119
120 switch (cbt_m1 >> 1) {
121 case 0:
122 for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
123 INTFLOAT *cf = cfo;
124 int len = off_len;
125
126 do {
127 int code;
128 unsigned cb_idx;
129
130 UPDATE_CACHE(re, gb);
131 GET_VLC(code, re, gb, vlc_tab, 8, 2);
132 cb_idx = code;
133#if USE_FIXED
134 cf = DEC_SQUAD(cf, cb_idx);
135#else
136 cf = VMUL4(cf, vq, cb_idx, sf + idx);
137#endif /* USE_FIXED */
138 } while (len -= 4);
139 }
140 break;
141
142 case 1:
143 for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
144 INTFLOAT *cf = cfo;
145 int len = off_len;
146
147 do {
148 int code;
149 unsigned nnz;
150 unsigned cb_idx;
151 uint32_t bits;
152
153 UPDATE_CACHE(re, gb);
154 GET_VLC(code, re, gb, vlc_tab, 8, 2);
155 cb_idx = code;
156 nnz = cb_idx >> 8 & 15;
157 bits = nnz ? GET_CACHE(re, gb) : 0;
158 LAST_SKIP_BITS(re, gb, nnz);
159#if USE_FIXED
160 cf = DEC_UQUAD(cf, cb_idx, bits);
161#else
162 cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
163#endif /* USE_FIXED */
164 } while (len -= 4);
165 }
166 break;
167
168 case 2:
169 for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
170 INTFLOAT *cf = cfo;
171 int len = off_len;
172
173 do {
174 int code;
175 unsigned cb_idx;
176
177 UPDATE_CACHE(re, gb);
178 GET_VLC(code, re, gb, vlc_tab, 8, 2);
179 cb_idx = code;
180#if USE_FIXED
181 cf = DEC_SPAIR(cf, cb_idx);
182#else
183 cf = VMUL2(cf, vq, cb_idx, sf + idx);
184#endif /* USE_FIXED */
185 } while (len -= 2);
186 }
187 break;
188
189 case 3:
190 case 4:
191 for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
192 INTFLOAT *cf = cfo;
193 int len = off_len;
194
195 do {
196 int code;
197 unsigned nnz;
198 unsigned cb_idx;
199 unsigned sign;
200
201 UPDATE_CACHE(re, gb);
202 GET_VLC(code, re, gb, vlc_tab, 8, 2);
203 cb_idx = code;
204 nnz = cb_idx >> 8 & 15;
205 sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
206 LAST_SKIP_BITS(re, gb, nnz);
207#if USE_FIXED
208 cf = DEC_UPAIR(cf, cb_idx, sign);
209#else
210 cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
211#endif /* USE_FIXED */
212 } while (len -= 2);
213 }
214 break;
215
216 default:
217 for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
218#if USE_FIXED
219 int *icf = cfo;
220 int v;
221#else
222 float *cf = cfo;
223 uint32_t *icf = (uint32_t *) cf;
224#endif /* USE_FIXED */
225 int len = off_len;
226
227 do {
228 int code;
229 unsigned nzt, nnz;
230 unsigned cb_idx;
231 uint32_t bits;
232 int j;
233
234 UPDATE_CACHE(re, gb);
235 GET_VLC(code, re, gb, vlc_tab, 8, 2);
236 cb_idx = code;
237
238 if (cb_idx == 0x0000) {
239 *icf++ = 0;
240 *icf++ = 0;
241 continue;
242 }
243
244 nnz = cb_idx >> 12;
245 nzt = cb_idx >> 8;
246 bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
247 LAST_SKIP_BITS(re, gb, nnz);
248
249 for (j = 0; j < 2; j++) {
250 if (nzt & 1<<j) {
251 uint32_t b;
252 int n;
253 /* The total length of escape_sequence must be < 22 bits according
254 to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
255 UPDATE_CACHE(re, gb);
256 b = GET_CACHE(re, gb);
257 b = 31 - av_log2(~b);
258
259 if (b > 8) {
260 av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
261 return AVERROR_INVALIDDATA;
262 }
263
264 SKIP_BITS(re, gb, b + 1);
265 b += 4;
266 n = (1 << b) + SHOW_UBITS(re, gb, b);
267 LAST_SKIP_BITS(re, gb, b);
268#if USE_FIXED
269 v = n;
270 if (bits & 1U<<31)
271 v = -v;
272 *icf++ = v;
273#else
274 *icf++ = ff_cbrt_tab[n] | (bits & 1U<<31);
275#endif /* USE_FIXED */
276 bits <<= 1;
277 } else {
278#if USE_FIXED
279 v = cb_idx & 15;
280 if (bits & 1U<<31)
281 v = -v;
282 *icf++ = v;
283#else
284 unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
285 *icf++ = (bits & 1U<<31) | v;
286#endif /* USE_FIXED */
287 bits <<= !!v;
288 }
289 cb_idx >>= 4;
290 }
291 } while (len -= 2);
292#if !USE_FIXED
293 ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
294#endif /* !USE_FIXED */
295 }
296 }
297
298 CLOSE_READER(re, gb);
299 }
300 }
301 coef += g_len << 7;
302 }
303
304 if (pulse) {
305 idx = 0;
306 for (i = 0; i < pulse->num_pulse; i++) {
307 INTFLOAT co = coef_base[ pulse->pos[i] ];
308 while (offsets[idx + 1] <= pulse->pos[i])
309 idx++;
310 if (band_type[idx] != NOISE_BT && sf[idx]) {
311 INTFLOAT ico = -pulse->amp[i];
312#if USE_FIXED
313 if (co) {
314 ico = co + (co > 0 ? -ico : ico);
315 }
316 coef_base[ pulse->pos[i] ] = ico;
317#else
318 if (co) {
319 co /= sf[idx];
320 ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
321 }
322 coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
323#endif /* USE_FIXED */
324 }
325 }
326 }
327#if USE_FIXED
328 coef = coef_base;
329 idx = 0;
330 for (g = 0; g < ics->num_window_groups; g++) {
331 unsigned g_len = ics->group_len[g];
332
333 for (i = 0; i < ics->max_sfb; i++, idx++) {
334 const unsigned cbt_m1 = band_type[idx] - 1;
335 int *cfo = coef + offsets[i];
336 int off_len = offsets[i + 1] - offsets[i];
337 int group;
338
339 if (cbt_m1 < NOISE_BT - 1) {
340 for (group = 0; group < (int)g_len; group++, cfo+=128) {
341 vector_pow43(cfo, off_len);
342 subband_scale(cfo, cfo, sf[idx], 34, off_len, ac->avctx);
343 }
344 }
345 }
346 coef += g_len << 7;
347 }
348#endif /* USE_FIXED */
349 return 0;
350}
351
352const AACDecProc AAC_RENAME(aac_proc) = {
353 .decode_spectrum_and_dequant = AAC_RENAME(decode_spectrum_and_dequant),
354};

Keyboard shortcuts

General

Show keyboard shortcuts
?
Focus the current search
/

Issues and pull requests

Open filters
f

Issues, pull requests and Inbox

Select next item
j↓
Select previous item
k↑
Open selected item
Enter
Dismiss filters or clear selection
Esc

Repository code

Go to file
t