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/*
* 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"
#include "dcamath.h"
static void decode_hf_c(int32_t dst[DCA_SUBBANDS][SAMPLES_PER_SUBBAND],
const int32_t vq_num[DCA_SUBBANDS],
const int8_t hf_vq[1024][32], intptr_t vq_offset,
int32_t scale[DCA_SUBBANDS][2],
intptr_t start, intptr_t end)
{
int i, j;
for (j = start; j < end; j++) {
const int8_t *ptr = &hf_vq[vq_num[j]][vq_offset];
for (i = 0; i < 8; i++)
dst[j][i] = ptr[i] * scale[j][0] + 8 >> 4;
}
}
static inline void dca_lfe_fir(float *out, const float *in, const float *coefs,
int decifactor)
{
float *out2 = out + 2 * decifactor - 1;
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++, coefs++) {
v0 += in[-j] * *coefs;
v1 += in[j + 1 - num_coeffs] * *coefs;
}
*out++ = v0;
*out2-- = v1;
}
}
static void dca_qmf_32_subbands(float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], 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 dequantize_c(int32_t *samples, uint32_t step_size, uint32_t scale)
{
int64_t step = (int64_t)step_size * scale;
int shift, i;
int32_t step_scale;
if (step > (1 << 23))
shift = av_log2(step >> 23) + 1;
else
shift = 0;
step_scale = (int32_t)(step >> shift);
for (i = 0; i < SAMPLES_PER_SUBBAND; i++)
samples[i] = dca_clip23(dca_norm((int64_t)samples[i] * step_scale, 22 - shift));
}
static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs)
{
dca_lfe_fir(out, in, coefs, 32);
}
static void dca_lfe_fir1_c(float *out, const float *in, const float *coefs)
{
dca_lfe_fir(out, in, coefs, 64);
}
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->decode_hf = decode_hf_c;
s->dequantize = dequantize_c;
if (ARCH_AARCH64)
ff_dcadsp_init_aarch64(s);
if (ARCH_ARM)
ff_dcadsp_init_arm(s);
if (ARCH_X86)
ff_dcadsp_init_x86(s);
}
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