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fb3c33f3cd
* commit '4cb6964244fd6c099383d8b7e99731e72cc844b9': dcadec: simplify decoding of VQ high frequencies Conflicts: configure libavcodec/dcadec.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
113 lines
3.7 KiB
C
113 lines
3.7 KiB
C
/*
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* Copyright (c) 2004 Gildas Bazin
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* Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "config.h"
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#include "libavutil/attributes.h"
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#include "libavutil/intreadwrite.h"
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#include "dcadsp.h"
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static void decode_hf_c(float dst[DCA_SUBBANDS][8],
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const int32_t vq_num[DCA_SUBBANDS],
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const int8_t hf_vq[1024][32], intptr_t vq_offset,
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int32_t scale[DCA_SUBBANDS][2],
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intptr_t start, intptr_t end)
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{
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int i, l;
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for (l = start; l < end; l++) {
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/* 1 vector -> 32 samples but we only need the 8 samples
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* for this subsubframe. */
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const int8_t *ptr = &hf_vq[vq_num[l]][vq_offset];
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float fscale = scale[l][0] * (1 / 16.0);
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for (i = 0; i < 8; i++)
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dst[l][i] = ptr[i] * fscale;
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}
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}
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static inline void
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dca_lfe_fir(float *out, const float *in, const float *coefs,
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int decifactor)
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{
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float *out2 = out + 2 * decifactor - 1;
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int num_coeffs = 256 / decifactor;
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int j, k;
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/* One decimated sample generates 2*decifactor interpolated ones */
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for (k = 0; k < decifactor; k++) {
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float v0 = 0.0;
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float v1 = 0.0;
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for (j = 0; j < num_coeffs; j++, coefs++) {
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v0 += in[-j] * *coefs;
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v1 += in[j + 1 - num_coeffs] * *coefs;
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}
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*out++ = v0;
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*out2-- = v1;
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}
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}
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static void dca_qmf_32_subbands(float samples_in[32][8], int sb_act,
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SynthFilterContext *synth, FFTContext *imdct,
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float synth_buf_ptr[512],
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int *synth_buf_offset, float synth_buf2[32],
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const float window[512], float *samples_out,
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float raXin[32], float scale)
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{
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int i;
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int subindex;
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for (i = sb_act; i < 32; i++)
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raXin[i] = 0.0;
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/* Reconstructed channel sample index */
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for (subindex = 0; subindex < 8; subindex++) {
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/* Load in one sample from each subband and clear inactive subbands */
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for (i = 0; i < sb_act; i++) {
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unsigned sign = (i - 1) & 2;
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uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
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AV_WN32A(&raXin[i], v);
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}
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synth->synth_filter_float(imdct, synth_buf_ptr, synth_buf_offset,
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synth_buf2, window, samples_out, raXin, scale);
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samples_out += 32;
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}
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}
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static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs)
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{
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dca_lfe_fir(out, in, coefs, 32);
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}
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static void dca_lfe_fir1_c(float *out, const float *in, const float *coefs)
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{
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dca_lfe_fir(out, in, coefs, 64);
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}
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av_cold void ff_dcadsp_init(DCADSPContext *s)
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{
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s->lfe_fir[0] = dca_lfe_fir0_c;
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s->lfe_fir[1] = dca_lfe_fir1_c;
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s->qmf_32_subbands = dca_qmf_32_subbands;
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s->decode_hf = decode_hf_c;
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if (ARCH_ARM) ff_dcadsp_init_arm(s);
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if (ARCH_X86) ff_dcadsp_init_x86(s);
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}
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