FFmpeg
 All Data Structures Namespaces Files Functions Variables Typedefs Enumerations Enumerator Macros Groups Pages
dcadsp.c
Go to the documentation of this file.
1 /*
2  * Copyright (c) 2004 Gildas Bazin
3  * Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "config.h"
23 #include "libavutil/attributes.h"
24 #include "libavutil/intreadwrite.h"
25 #include "dcadsp.h"
26 
27 static void decode_hf_c(float dst[DCA_SUBBANDS][8],
28  const int32_t vq_num[DCA_SUBBANDS],
29  const int8_t hf_vq[1024][32], intptr_t vq_offset,
30  int32_t scale[DCA_SUBBANDS][2],
31  intptr_t start, intptr_t end)
32 {
33  int i, l;
34 
35  for (l = start; l < end; l++) {
36  /* 1 vector -> 32 samples but we only need the 8 samples
37  * for this subsubframe. */
38  const int8_t *ptr = &hf_vq[vq_num[l]][vq_offset];
39  float fscale = scale[l][0] * (1 / 16.0);
40  for (i = 0; i < 8; i++)
41  dst[l][i] = ptr[i] * fscale;
42  }
43 }
44 
45 static inline void
46 dca_lfe_fir(float *out, const float *in, const float *coefs,
47  int decifactor)
48 {
49  float *out2 = out + 2 * decifactor - 1;
50  int num_coeffs = 256 / decifactor;
51  int j, k;
52 
53  /* One decimated sample generates 2*decifactor interpolated ones */
54  for (k = 0; k < decifactor; k++) {
55  float v0 = 0.0;
56  float v1 = 0.0;
57  for (j = 0; j < num_coeffs; j++, coefs++) {
58  v0 += in[-j] * *coefs;
59  v1 += in[j + 1 - num_coeffs] * *coefs;
60  }
61  *out++ = v0;
62  *out2-- = v1;
63  }
64 }
65 
66 static void dca_qmf_32_subbands(float samples_in[32][8], int sb_act,
67  SynthFilterContext *synth, FFTContext *imdct,
68  float synth_buf_ptr[512],
69  int *synth_buf_offset, float synth_buf2[32],
70  const float window[512], float *samples_out,
71  float raXin[32], float scale)
72 {
73  int i;
74  int subindex;
75 
76  for (i = sb_act; i < 32; i++)
77  raXin[i] = 0.0;
78 
79  /* Reconstructed channel sample index */
80  for (subindex = 0; subindex < 8; subindex++) {
81  /* Load in one sample from each subband and clear inactive subbands */
82  for (i = 0; i < sb_act; i++) {
83  unsigned sign = (i - 1) & 2;
84  uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
85  AV_WN32A(&raXin[i], v);
86  }
87 
88  synth->synth_filter_float(imdct, synth_buf_ptr, synth_buf_offset,
89  synth_buf2, window, samples_out, raXin, scale);
90  samples_out += 32;
91  }
92 }
93 
94 static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs)
95 {
96  dca_lfe_fir(out, in, coefs, 32);
97 }
98 
99 static void dca_lfe_fir1_c(float *out, const float *in, const float *coefs)
100 {
101  dca_lfe_fir(out, in, coefs, 64);
102 }
103 
105 {
106  s->lfe_fir[0] = dca_lfe_fir0_c;
107  s->lfe_fir[1] = dca_lfe_fir1_c;
109  s->decode_hf = decode_hf_c;
110  if (ARCH_ARM) ff_dcadsp_init_arm(s);
111  if (ARCH_X86) ff_dcadsp_init_x86(s);
112 }