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FFmpeg/libavcodec/aacpsdsp_template.c
Rémi Denis-Courmont 08edacc248 lavc/aacpsdsp: precompute constant factors 
The input complex factors are constant for each iterations. This
substitudes 4 loads, 2 additions and 2 subtractions per iteration of
the inner-loop with another 4 loads. Thus effectively 4 arithmetic
operations per iteration of the inner loop are avoided, i.e. 24
operations per iteration of the outer loop, or 24 * (n - 1) operations
in total.

If the inner loop is not unrolled by the compiler, this also might
also save some pointer arithmetic as most instruction sets do not
have addressing modes with negated register offsets (12 - j). Unless
the compiler is optimising for code size, this is unlikely though.
2022-09-22 13:27:43 -03:00

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/*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
* Note: Rounding-to-nearest used unless otherwise stated
*
*/
#include <stdint.h>
#include "config.h"
#include "libavutil/attributes.h"
#include "aacpsdsp.h"
static void ps_add_squares_c(INTFLOAT *dst, const INTFLOAT (*src)[2], int n)
{
int i;
for (i = 0; i < n; i++)
dst[i] += (UINTFLOAT)AAC_MADD28(src[i][0], src[i][0], src[i][1], src[i][1]);
}
static void ps_mul_pair_single_c(INTFLOAT (*dst)[2], INTFLOAT (*src0)[2], INTFLOAT *src1,
int n)
{
int i;
for (i = 0; i < n; i++) {
dst[i][0] = AAC_MUL16(src0[i][0], src1[i]);
dst[i][1] = AAC_MUL16(src0[i][1], src1[i]);
}
}
static void ps_hybrid_analysis_c(INTFLOAT (*out)[2], INTFLOAT (*in)[2],
const INTFLOAT (*filter)[8][2],
ptrdiff_t stride, int n)
{
INT64FLOAT inre0[6], inre1[6], inim0[6], inim1[6];
for (int j = 0; j < 6; j++) {
inre0[j] = in[j][0] + in[12 - j][0];
inre1[j] = in[j][1] - in[12 - j][1];
inim0[j] = in[j][1] + in[12 - j][1];
inim1[j] = in[j][0] - in[12 - j][0];
}
for (int i = 0; i < n; i++) {
INT64FLOAT sum_re = (INT64FLOAT)filter[i][6][0] * in[6][0];
INT64FLOAT sum_im = (INT64FLOAT)filter[i][6][0] * in[6][1];
for (int j = 0; j < 6; j++) {
sum_re += (INT64FLOAT)filter[i][j][0] * inre0[j] -
(INT64FLOAT)filter[i][j][1] * inre1[j];
sum_im += (INT64FLOAT)filter[i][j][0] * inim0[j] +
(INT64FLOAT)filter[i][j][1] * inim1[j];
}
#if USE_FIXED
out[i * stride][0] = (int)((sum_re + 0x40000000) >> 31);
out[i * stride][1] = (int)((sum_im + 0x40000000) >> 31);
#else
out[i * stride][0] = sum_re;
out[i * stride][1] = sum_im;
#endif /* USE_FIXED */
}
}
static void ps_hybrid_analysis_ileave_c(INTFLOAT (*out)[32][2], INTFLOAT L[2][38][64],
int i, int len)
{
int j;
for (; i < 64; i++) {
for (j = 0; j < len; j++) {
out[i][j][0] = L[0][j][i];
out[i][j][1] = L[1][j][i];
}
}
}
static void ps_hybrid_synthesis_deint_c(INTFLOAT out[2][38][64],
INTFLOAT (*in)[32][2],
int i, int len)
{
int n;
for (; i < 64; i++) {
for (n = 0; n < len; n++) {
out[0][n][i] = in[i][n][0];
out[1][n][i] = in[i][n][1];
}
}
}
static void ps_decorrelate_c(INTFLOAT (*out)[2], INTFLOAT (*delay)[2],
INTFLOAT (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2],
const INTFLOAT phi_fract[2], const INTFLOAT (*Q_fract)[2],
const INTFLOAT *transient_gain,
INTFLOAT g_decay_slope,
int len)
{
static const INTFLOAT a[] = { Q31(0.65143905753106f),
Q31(0.56471812200776f),
Q31(0.48954165955695f) };
INTFLOAT ag[PS_AP_LINKS];
int m, n;
for (m = 0; m < PS_AP_LINKS; m++)
ag[m] = AAC_MUL30(a[m], g_decay_slope);
for (n = 0; n < len; n++) {
INTFLOAT in_re = AAC_MSUB30(delay[n][0], phi_fract[0], delay[n][1], phi_fract[1]);
INTFLOAT in_im = AAC_MADD30(delay[n][0], phi_fract[1], delay[n][1], phi_fract[0]);
for (m = 0; m < PS_AP_LINKS; m++) {
INTFLOAT a_re = AAC_MUL31(ag[m], in_re);
INTFLOAT a_im = AAC_MUL31(ag[m], in_im);
INTFLOAT link_delay_re = ap_delay[m][n+2-m][0];
INTFLOAT link_delay_im = ap_delay[m][n+2-m][1];
INTFLOAT fractional_delay_re = Q_fract[m][0];
INTFLOAT fractional_delay_im = Q_fract[m][1];
INTFLOAT apd_re = in_re;
INTFLOAT apd_im = in_im;
in_re = AAC_MSUB30(link_delay_re, fractional_delay_re,
link_delay_im, fractional_delay_im);
in_re -= (UINTFLOAT)a_re;
in_im = AAC_MADD30(link_delay_re, fractional_delay_im,
link_delay_im, fractional_delay_re);
in_im -= (UINTFLOAT)a_im;
ap_delay[m][n+5][0] = apd_re + (UINTFLOAT)AAC_MUL31(ag[m], in_re);
ap_delay[m][n+5][1] = apd_im + (UINTFLOAT)AAC_MUL31(ag[m], in_im);
}
out[n][0] = AAC_MUL16(transient_gain[n], in_re);
out[n][1] = AAC_MUL16(transient_gain[n], in_im);
}
}
static void ps_stereo_interpolate_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
INTFLOAT h[2][4], INTFLOAT h_step[2][4],
int len)
{
INTFLOAT h0 = h[0][0];
INTFLOAT h1 = h[0][1];
INTFLOAT h2 = h[0][2];
INTFLOAT h3 = h[0][3];
UINTFLOAT hs0 = h_step[0][0];
UINTFLOAT hs1 = h_step[0][1];
UINTFLOAT hs2 = h_step[0][2];
UINTFLOAT hs3 = h_step[0][3];
int n;
for (n = 0; n < len; n++) {
//l is s, r is d
INTFLOAT l_re = l[n][0];
INTFLOAT l_im = l[n][1];
INTFLOAT r_re = r[n][0];
INTFLOAT r_im = r[n][1];
h0 += hs0;
h1 += hs1;
h2 += hs2;
h3 += hs3;
l[n][0] = AAC_MADD30(h0, l_re, h2, r_re);
l[n][1] = AAC_MADD30(h0, l_im, h2, r_im);
r[n][0] = AAC_MADD30(h1, l_re, h3, r_re);
r[n][1] = AAC_MADD30(h1, l_im, h3, r_im);
}
}
static void ps_stereo_interpolate_ipdopd_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
INTFLOAT h[2][4], INTFLOAT h_step[2][4],
int len)
{
INTFLOAT h00 = h[0][0], h10 = h[1][0];
INTFLOAT h01 = h[0][1], h11 = h[1][1];
INTFLOAT h02 = h[0][2], h12 = h[1][2];
INTFLOAT h03 = h[0][3], h13 = h[1][3];
UINTFLOAT hs00 = h_step[0][0], hs10 = h_step[1][0];
UINTFLOAT hs01 = h_step[0][1], hs11 = h_step[1][1];
UINTFLOAT hs02 = h_step[0][2], hs12 = h_step[1][2];
UINTFLOAT hs03 = h_step[0][3], hs13 = h_step[1][3];
int n;
for (n = 0; n < len; n++) {
//l is s, r is d
INTFLOAT l_re = l[n][0];
INTFLOAT l_im = l[n][1];
INTFLOAT r_re = r[n][0];
INTFLOAT r_im = r[n][1];
h00 += hs00;
h01 += hs01;
h02 += hs02;
h03 += hs03;
h10 += hs10;
h11 += hs11;
h12 += hs12;
h13 += hs13;
l[n][0] = AAC_MSUB30_V8(h00, l_re, h02, r_re, h10, l_im, h12, r_im);
l[n][1] = AAC_MADD30_V8(h00, l_im, h02, r_im, h10, l_re, h12, r_re);
r[n][0] = AAC_MSUB30_V8(h01, l_re, h03, r_re, h11, l_im, h13, r_im);
r[n][1] = AAC_MADD30_V8(h01, l_im, h03, r_im, h11, l_re, h13, r_re);
}
}
av_cold void AAC_RENAME(ff_psdsp_init)(PSDSPContext *s)
{
s->add_squares = ps_add_squares_c;
s->mul_pair_single = ps_mul_pair_single_c;
s->hybrid_analysis = ps_hybrid_analysis_c;
s->hybrid_analysis_ileave = ps_hybrid_analysis_ileave_c;
s->hybrid_synthesis_deint = ps_hybrid_synthesis_deint_c;
s->decorrelate = ps_decorrelate_c;
s->stereo_interpolate[0] = ps_stereo_interpolate_c;
s->stereo_interpolate[1] = ps_stereo_interpolate_ipdopd_c;
#if !USE_FIXED
#if ARCH_ARM
ff_psdsp_init_arm(s);
#elif ARCH_AARCH64
ff_psdsp_init_aarch64(s);
#elif ARCH_MIPS
ff_psdsp_init_mips(s);
#elif ARCH_X86
ff_psdsp_init_x86(s);
#endif
#endif /* !USE_FIXED */
}
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