mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-21 10:55:51 +02:00
5b59a9fc61
For the callable function (as opposed to the inline one):
C SSE SSE2 SSE4
Win32: 47 42 29 26
Win64: 30 33 25 23
The SSE version is neither compiled nor set for ARCH_X86_64, as the
inlinable function takes over.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
93 lines
3.0 KiB
C
93 lines
3.0 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 void dca_lfe_fir_c(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;
|
|
}
|
|
}
|
|
|
|
av_cold void ff_dcadsp_init(DCADSPContext *s)
|
|
{
|
|
s->lfe_fir = dca_lfe_fir_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);
|
|
}
|