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* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
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* This file is part of FFmpeg.
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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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* 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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* 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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* Note: Rounding-to-nearest used unless otherwise stated
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#include "libavutil/attributes.h"
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static void ps_add_squares_c(INTFLOAT *dst, const INTFLOAT (*src)[2], int n)
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for (i = 0; i < n; i++)
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dst[i] += AAC_MADD28(src[i][0], src[i][0], src[i][1], src[i][1]);
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static void ps_mul_pair_single_c(INTFLOAT (*dst)[2], INTFLOAT (*src0)[2], INTFLOAT *src1,
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for (i = 0; i < n; i++) {
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dst[i][0] = AAC_MUL16(src0[i][0], src1[i]);
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dst[i][1] = AAC_MUL16(src0[i][1], src1[i]);
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static void ps_hybrid_analysis_c(INTFLOAT (*out)[2], INTFLOAT (*in)[2],
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const INTFLOAT (*filter)[8][2],
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for (i = 0; i < n; i++) {
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INT64FLOAT sum_re = (INT64FLOAT)filter[i][6][0] * in[6][0];
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INT64FLOAT sum_im = (INT64FLOAT)filter[i][6][0] * in[6][1];
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for (j = 0; j < 6; j++) {
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INTFLOAT in0_re = in[j][0];
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INTFLOAT in0_im = in[j][1];
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INTFLOAT in1_re = in[12-j][0];
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INTFLOAT in1_im = in[12-j][1];
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sum_re += (INT64FLOAT)filter[i][j][0] * (in0_re + in1_re) -
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(INT64FLOAT)filter[i][j][1] * (in0_im - in1_im);
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sum_im += (INT64FLOAT)filter[i][j][0] * (in0_im + in1_im) +
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(INT64FLOAT)filter[i][j][1] * (in0_re - in1_re);
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out[i * stride][0] = (int)((sum_re + 0x40000000) >> 31);
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out[i * stride][1] = (int)((sum_im + 0x40000000) >> 31);
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out[i * stride][0] = sum_re;
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out[i * stride][1] = sum_im;
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#endif /* USE_FIXED */
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static void ps_hybrid_analysis_ileave_c(INTFLOAT (*out)[32][2], INTFLOAT L[2][38][64],
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for (j = 0; j < len; j++) {
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out[i][j][0] = L[0][j][i];
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out[i][j][1] = L[1][j][i];
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static void ps_hybrid_synthesis_deint_c(INTFLOAT out[2][38][64],
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INTFLOAT (*in)[32][2],
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for (n = 0; n < len; n++) {
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out[0][n][i] = in[i][n][0];
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out[1][n][i] = in[i][n][1];
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static void ps_decorrelate_c(INTFLOAT (*out)[2], INTFLOAT (*delay)[2],
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INTFLOAT (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2],
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const INTFLOAT phi_fract[2], const INTFLOAT (*Q_fract)[2],
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const INTFLOAT *transient_gain,
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INTFLOAT g_decay_slope,
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static const INTFLOAT a[] = { Q31(0.65143905753106f),
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Q31(0.56471812200776f),
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Q31(0.48954165955695f) };
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INTFLOAT ag[PS_AP_LINKS];
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for (m = 0; m < PS_AP_LINKS; m++)
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ag[m] = AAC_MUL30(a[m], g_decay_slope);
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for (n = 0; n < len; n++) {
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INTFLOAT in_re = AAC_MSUB30(delay[n][0], phi_fract[0], delay[n][1], phi_fract[1]);
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INTFLOAT in_im = AAC_MADD30(delay[n][0], phi_fract[1], delay[n][1], phi_fract[0]);
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for (m = 0; m < PS_AP_LINKS; m++) {
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INTFLOAT a_re = AAC_MUL31(ag[m], in_re);
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INTFLOAT a_im = AAC_MUL31(ag[m], in_im);
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INTFLOAT link_delay_re = ap_delay[m][n+2-m][0];
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INTFLOAT link_delay_im = ap_delay[m][n+2-m][1];
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INTFLOAT fractional_delay_re = Q_fract[m][0];
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INTFLOAT fractional_delay_im = Q_fract[m][1];
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INTFLOAT apd_re = in_re;
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INTFLOAT apd_im = in_im;
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in_re = AAC_MSUB30(link_delay_re, fractional_delay_re,
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link_delay_im, fractional_delay_im);
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in_im = AAC_MADD30(link_delay_re, fractional_delay_im,
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link_delay_im, fractional_delay_re);
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ap_delay[m][n+5][0] = apd_re + AAC_MUL31(ag[m], in_re);
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ap_delay[m][n+5][1] = apd_im + AAC_MUL31(ag[m], in_im);
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out[n][0] = AAC_MUL16(transient_gain[n], in_re);
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out[n][1] = AAC_MUL16(transient_gain[n], in_im);
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static void ps_stereo_interpolate_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
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INTFLOAT h[2][4], INTFLOAT h_step[2][4],
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INTFLOAT h0 = h[0][0];
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INTFLOAT h1 = h[0][1];
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INTFLOAT h2 = h[0][2];
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INTFLOAT h3 = h[0][3];
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INTFLOAT hs0 = h_step[0][0];
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INTFLOAT hs1 = h_step[0][1];
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INTFLOAT hs2 = h_step[0][2];
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INTFLOAT hs3 = h_step[0][3];
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for (n = 0; n < len; n++) {
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INTFLOAT l_re = l[n][0];
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INTFLOAT l_im = l[n][1];
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INTFLOAT r_re = r[n][0];
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INTFLOAT r_im = r[n][1];
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l[n][0] = AAC_MADD30(h0, l_re, h2, r_re);
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l[n][1] = AAC_MADD30(h0, l_im, h2, r_im);
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r[n][0] = AAC_MADD30(h1, l_re, h3, r_re);
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r[n][1] = AAC_MADD30(h1, l_im, h3, r_im);
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static void ps_stereo_interpolate_ipdopd_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
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INTFLOAT h[2][4], INTFLOAT h_step[2][4],
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INTFLOAT h00 = h[0][0], h10 = h[1][0];
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INTFLOAT h01 = h[0][1], h11 = h[1][1];
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INTFLOAT h02 = h[0][2], h12 = h[1][2];
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INTFLOAT h03 = h[0][3], h13 = h[1][3];
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INTFLOAT hs00 = h_step[0][0], hs10 = h_step[1][0];
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INTFLOAT hs01 = h_step[0][1], hs11 = h_step[1][1];
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INTFLOAT hs02 = h_step[0][2], hs12 = h_step[1][2];
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INTFLOAT hs03 = h_step[0][3], hs13 = h_step[1][3];
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for (n = 0; n < len; n++) {
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INTFLOAT l_re = l[n][0];
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INTFLOAT l_im = l[n][1];
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INTFLOAT r_re = r[n][0];
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INTFLOAT r_im = r[n][1];
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l[n][0] = AAC_MSUB30_V8(h00, l_re, h02, r_re, h10, l_im, h12, r_im);
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l[n][1] = AAC_MADD30_V8(h00, l_im, h02, r_im, h10, l_re, h12, r_re);
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r[n][0] = AAC_MSUB30_V8(h01, l_re, h03, r_re, h11, l_im, h13, r_im);
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r[n][1] = AAC_MADD30_V8(h01, l_im, h03, r_im, h11, l_re, h13, r_re);
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av_cold void AAC_RENAME(ff_psdsp_init)(PSDSPContext *s)
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s->add_squares = ps_add_squares_c;
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s->mul_pair_single = ps_mul_pair_single_c;
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s->hybrid_analysis = ps_hybrid_analysis_c;
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s->hybrid_analysis_ileave = ps_hybrid_analysis_ileave_c;
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s->hybrid_synthesis_deint = ps_hybrid_synthesis_deint_c;
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s->decorrelate = ps_decorrelate_c;
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s->stereo_interpolate[0] = ps_stereo_interpolate_c;
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s->stereo_interpolate[1] = ps_stereo_interpolate_ipdopd_c;
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ff_psdsp_init_arm(s);
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ff_psdsp_init_mips(s);
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ff_psdsp_init_x86(s);
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#endif /* !USE_FIXED */