ffmpeg/libavcodec/dcadsp.c
Christophe Gisquet 9ae8e23188 dcadsp: scan coefficients linearly instead.
This change is inspired by x86 asm, where this frees a register.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-02-15 20:08:08 +01:00

107 lines
3.4 KiB
C

/*
* Copyright (c) 2004 Gildas Bazin
* Copyright (c) 2010 Mans Rullgard <mans@mansr.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
*/
#include "config.h"
#include "libavutil/attributes.h"
#include "libavutil/intreadwrite.h"
#include "dcadsp.h"
static void int8x8_fmul_int32_c(float *dst, const int8_t *src, int scale)
{
float fscale = scale / 16.0;
int i;
for (i = 0; i < 8; i++)
dst[i] = src[i] * fscale;
}
static inline void
dca_lfe_fir(float *out, const float *in, const float *coefs,
int decifactor, float scale)
{
float *out2 = out + 2*decifactor-1;
const float *cf0 = coefs;
int num_coeffs = 256 / decifactor;
int j, k;
/* One decimated sample generates 2*decifactor interpolated ones */
for (k = 0; k < decifactor; k++) {
float v0 = 0.0;
float v1 = 0.0;
for (j = 0; j < num_coeffs; j++, cf0++) {
v0 += in[-j] * *cf0;
v1 += in[1+j-num_coeffs] * *cf0;
}
*out++ = v0 * scale;
*out2-- = v1 * scale;
}
}
static void dca_qmf_32_subbands(float samples_in[32][8], int sb_act,
SynthFilterContext *synth, FFTContext *imdct,
float synth_buf_ptr[512],
int *synth_buf_offset, float synth_buf2[32],
const float window[512], float *samples_out,
float raXin[32], float scale)
{
int i;
int subindex;
for (i = sb_act; i < 32; i++)
raXin[i] = 0.0;
/* Reconstructed channel sample index */
for (subindex = 0; subindex < 8; subindex++) {
/* Load in one sample from each subband and clear inactive subbands */
for (i = 0; i < sb_act; i++) {
unsigned sign = (i - 1) & 2;
uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
AV_WN32A(&raXin[i], v);
}
synth->synth_filter_float(imdct, synth_buf_ptr, synth_buf_offset,
synth_buf2, window, samples_out, raXin, scale);
samples_out += 32;
}
}
static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs,
float scale)
{
dca_lfe_fir(out, in, coefs, 32, scale);
}
static void dca_lfe_fir1_c(float *out, const float *in, const float *coefs,
float scale)
{
dca_lfe_fir(out, in, coefs, 64, scale);
}
av_cold void ff_dcadsp_init(DCADSPContext *s)
{
s->lfe_fir[0] = dca_lfe_fir0_c;
s->lfe_fir[1] = dca_lfe_fir1_c;
s->qmf_32_subbands = dca_qmf_32_subbands;
s->int8x8_fmul_int32 = int8x8_fmul_int32_c;
if (ARCH_ARM) ff_dcadsp_init_arm(s);
if (ARCH_X86) ff_dcadsp_init_x86(s);
}