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dcadsp.c
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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 
24 #include "libavutil/attributes.h"
25 #include "libavutil/intreadwrite.h"
26 
27 #include "dcadsp.h"
28 
29 static void decode_hf_c(float dst[DCA_SUBBANDS][8],
30  const int32_t vq_num[DCA_SUBBANDS],
31  const int8_t hf_vq[1024][32], intptr_t vq_offset,
32  int32_t scale[DCA_SUBBANDS][2],
33  intptr_t start, intptr_t end)
34 {
35  int i, l;
36 
37  for (l = start; l < end; l++) {
38  /* 1 vector -> 32 samples but we only need the 8 samples
39  * for this subsubframe. */
40  const int8_t *ptr = &hf_vq[vq_num[l]][vq_offset];
41  float fscale = scale[l][0] * (1 / 16.0);
42  for (i = 0; i < 8; i++)
43  dst[l][i] = ptr[i] * fscale;
44  }
45 }
46 
47 static inline void dca_lfe_fir(float *out, const float *in, const float *coefs,
48  int decifactor)
49 {
50  float *out2 = out + 2 * decifactor - 1;
51  int num_coeffs = 256 / decifactor;
52  int j, k;
53 
54  /* One decimated sample generates 2*decifactor interpolated ones */
55  for (k = 0; k < decifactor; k++) {
56  float v0 = 0.0;
57  float v1 = 0.0;
58  for (j = 0; j < num_coeffs; j++, coefs++) {
59  v0 += in[-j] * *coefs;
60  v1 += in[j + 1 - num_coeffs] * *coefs;
61  }
62  *out++ = v0;
63  *out2-- = v1;
64  }
65 }
66 
67 static void dca_qmf_32_subbands(float samples_in[32][8], int sb_act,
68  SynthFilterContext *synth, FFTContext *imdct,
69  float synth_buf_ptr[512],
70  int *synth_buf_offset, float synth_buf2[32],
71  const float window[512], float *samples_out,
72  float raXin[32], float scale)
73 {
74  int i;
75  int subindex;
76 
77  for (i = sb_act; i < 32; i++)
78  raXin[i] = 0.0;
79 
80  /* Reconstructed channel sample index */
81  for (subindex = 0; subindex < 8; subindex++) {
82  /* Load in one sample from each subband and clear inactive subbands */
83  for (i = 0; i < sb_act; i++) {
84  unsigned sign = (i - 1) & 2;
85  uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
86  AV_WN32A(&raXin[i], v);
87  }
88 
89  synth->synth_filter_float(imdct, synth_buf_ptr, synth_buf_offset,
90  synth_buf2, window, samples_out, raXin,
91  scale);
92  samples_out += 32;
93  }
94 }
95 
96 static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs)
97 {
98  dca_lfe_fir(out, in, coefs, 32);
99 }
100 
101 static void dca_lfe_fir1_c(float *out, const float *in, const float *coefs)
102 {
103  dca_lfe_fir(out, in, coefs, 64);
104 }
105 
107 {
108  s->lfe_fir[0] = dca_lfe_fir0_c;
109  s->lfe_fir[1] = dca_lfe_fir1_c;
111  s->decode_hf = decode_hf_c;
112 
113  if (ARCH_ARM)
115  if (ARCH_X86)
117 }