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synced 2025-02-07 15:22:57 +00:00
Split synth filter out of dca.c.
Originally committed as revision 20396 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
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638783bf81
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4f99c31c39
@ -73,7 +73,7 @@ OBJS-$(CONFIG_CLJR_ENCODER) += cljr.o
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OBJS-$(CONFIG_COOK_DECODER) += cook.o
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OBJS-$(CONFIG_CSCD_DECODER) += cscd.o
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OBJS-$(CONFIG_CYUV_DECODER) += cyuv.o
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OBJS-$(CONFIG_DCA_DECODER) += dca.o
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OBJS-$(CONFIG_DCA_DECODER) += dca.o synth_filter.o
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OBJS-$(CONFIG_DNXHD_DECODER) += dnxhddec.o dnxhddata.o
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OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o \
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mpegvideo_enc.o motion_est.o \
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@ -37,6 +37,7 @@
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#include "dcadata.h"
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#include "dcahuff.h"
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#include "dca.h"
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#include "synth_filter.h"
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//#define TRACE
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@ -753,9 +754,6 @@ static void qmf_32_subbands(DCAContext * s, int chans,
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const float *prCoeff;
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int i, j;
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int hist_index= s->hist_index[chans];
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float *subband_fir_hist2 = s->subband_fir_noidea[chans];
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int subindex;
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scale *= sqrt(1/8.0);
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@ -768,7 +766,6 @@ static void qmf_32_subbands(DCAContext * s, int chans,
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/* Reconstructed channel sample index */
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for (subindex = 0; subindex < 8; subindex++) {
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float *subband_fir_hist = s->subband_fir_hist[chans] + hist_index;
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/* Load in one sample from each subband and clear inactive subbands */
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for (i = 0; i < s->subband_activity[chans]; i++){
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if((i-1)&2) s->raXin[i] = -samples_in[i][subindex];
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@ -777,36 +774,13 @@ static void qmf_32_subbands(DCAContext * s, int chans,
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for (; i < 32; i++)
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s->raXin[i] = 0.0;
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ff_imdct_half(&s->imdct, subband_fir_hist, s->raXin);
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/* Multiply by filter coefficients */
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for (i = 0; i < 16; i++){
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float a= subband_fir_hist2[i ];
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float b= subband_fir_hist2[i+16];
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float c= 0;
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float d= 0;
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for (j = 0; j < 512-hist_index; j += 64){
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a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j]);
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b += prCoeff[i+j+16]*( subband_fir_hist[ i+j]);
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c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j]);
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d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j]);
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}
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for ( ; j < 512; j += 64){
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a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j-512]);
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b += prCoeff[i+j+16]*( subband_fir_hist[ i+j-512]);
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c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j-512]);
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d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j-512]);
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}
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samples_out[i ] = a * scale + bias;
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samples_out[i+16] = b * scale + bias;
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subband_fir_hist2[i ] = c;
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subband_fir_hist2[i+16] = d;
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}
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ff_synth_filter_float(&s->imdct,
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s->subband_fir_hist[chans], &s->hist_index[chans],
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s->subband_fir_noidea[chans], prCoeff,
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samples_out, s->raXin, scale, bias);
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samples_out+= 32;
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hist_index = (hist_index-32)&511;
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}
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s->hist_index[chans]= hist_index;
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}
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static void lfe_interpolation_fir(int decimation_select,
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56
libavcodec/synth_filter.c
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56
libavcodec/synth_filter.c
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@ -0,0 +1,56 @@
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/*
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* copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at>
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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 "synth_filter.h"
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void ff_synth_filter_float(FFTContext *imdct,
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float *synth_buf_ptr, int *synth_buf_offset,
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float synth_buf2[32], const float window[512],
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float out[32], const float in[32], float scale, float bias)
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{
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float *synth_buf= synth_buf_ptr + *synth_buf_offset;
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int i, j;
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ff_imdct_half(imdct, synth_buf, in);
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for (i = 0; i < 16; i++){
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float a= synth_buf2[i ];
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float b= synth_buf2[i+16];
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float c= 0;
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float d= 0;
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for (j = 0; j < 512 - *synth_buf_offset; j += 64){
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a += window[i+j ]*(-synth_buf[15-i+j]);
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b += window[i+j+16]*( synth_buf[ i+j]);
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c += window[i+j+32]*( synth_buf[16+i+j]);
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d += window[i+j+48]*( synth_buf[31-i+j]);
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}
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for ( ; j < 512; j += 64){
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a += window[i+j ]*(-synth_buf[15-i+j-512]);
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b += window[i+j+16]*( synth_buf[ i+j-512]);
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c += window[i+j+32]*( synth_buf[16+i+j-512]);
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d += window[i+j+48]*( synth_buf[31-i+j-512]);
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}
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out[i ] = a * scale + bias;
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out[i+16] = b * scale + bias;
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synth_buf2[i ] = c;
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synth_buf2[i+16] = d;
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}
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*synth_buf_offset= (*synth_buf_offset-32)&511;
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}
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27
libavcodec/synth_filter.h
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27
libavcodec/synth_filter.h
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@ -0,0 +1,27 @@
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/*
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* copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at>
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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 "dsputil.h"
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void ff_synth_filter_float(FFTContext *imdct,
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float *synth_buf_ptr, int *synth_buf_offset,
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float synth_buf2[32], const float window[512],
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float out[32], const float in[32], float scale, float bias);
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