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FFmpeg/libavcodec/dcadsp.c
Christophe Gisquet 5fdbfcb5b7 dcadsp: split lfe_dir cases
The x86 runs short on registers because numerous elements are not static.
In addition, splitting them allows more optimized code, at least for x86.

Arm asm changes by Janne Grunau.

Signed-off-by: Janne Grunau <janne-libav@jannau.net>
2014-02-07 22:54:18 +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 Libav.
*
* Libav 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.
*
* Libav 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 Libav; 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 + decifactor;
const float *cf0 = coefs;
const float *cf1 = coefs + 256;
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 < 256 / decifactor; j++) {
float s = in[-j];
v0 += s * *cf0++;
v1 += s * *--cf1;
}
*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);
}