From 5fd81cf6f082ed00878a5898f47550cb1646d219 Mon Sep 17 00:00:00 2001 From: Djordje Pesut Date: Mon, 20 Jul 2015 13:36:19 +0200 Subject: avcodec: Implementation of AAC_fixed_decoder (PS-module) Add fixed point implementation. Signed-off-by: Nedeljko Babic Signed-off-by: Michael Niedermayer --- libavcodec/aacpsdsp_template.c | 228 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 228 insertions(+) create mode 100644 libavcodec/aacpsdsp_template.c (limited to 'libavcodec/aacpsdsp_template.c') diff --git a/libavcodec/aacpsdsp_template.c b/libavcodec/aacpsdsp_template.c new file mode 100644 index 0000000000..bfec828cf6 --- /dev/null +++ b/libavcodec/aacpsdsp_template.c @@ -0,0 +1,228 @@ +/* + * Copyright (c) 2010 Alex Converse + * + * This file is part of FFmpeg. + * + * FFmpeg 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. + * + * FFmpeg 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + * + * Note: Rounding-to-nearest used unless otherwise stated + * + */ +#include + +#include "config.h" +#include "libavutil/attributes.h" +#include "aacpsdsp.h" + +static void ps_add_squares_c(INTFLOAT *dst, const INTFLOAT (*src)[2], int n) +{ + int i; + for (i = 0; i < n; i++) + dst[i] += AAC_MADD28(src[i][0], src[i][0], src[i][1], src[i][1]); +} + +static void ps_mul_pair_single_c(INTFLOAT (*dst)[2], INTFLOAT (*src0)[2], INTFLOAT *src1, + int n) +{ + int i; + for (i = 0; i < n; i++) { + dst[i][0] = AAC_MUL16(src0[i][0], src1[i]); + dst[i][1] = AAC_MUL16(src0[i][1], src1[i]); + } +} + +static void ps_hybrid_analysis_c(INTFLOAT (*out)[2], INTFLOAT (*in)[2], + const INTFLOAT (*filter)[8][2], + int stride, int n) +{ + int i, j; + + for (i = 0; i < n; i++) { + INT64FLOAT sum_re = (INT64FLOAT)filter[i][6][0] * in[6][0]; + INT64FLOAT sum_im = (INT64FLOAT)filter[i][6][0] * in[6][1]; + + for (j = 0; j < 6; j++) { + INTFLOAT in0_re = in[j][0]; + INTFLOAT in0_im = in[j][1]; + INTFLOAT in1_re = in[12-j][0]; + INTFLOAT in1_im = in[12-j][1]; + sum_re += (INT64FLOAT)filter[i][j][0] * (in0_re + in1_re) - + (INT64FLOAT)filter[i][j][1] * (in0_im - in1_im); + sum_im += (INT64FLOAT)filter[i][j][0] * (in0_im + in1_im) + + (INT64FLOAT)filter[i][j][1] * (in0_re - in1_re); + } +#if USE_FIXED + out[i * stride][0] = (int)((sum_re + 0x40000000) >> 31); + out[i * stride][1] = (int)((sum_im + 0x40000000) >> 31); +#else + out[i * stride][0] = sum_re; + out[i * stride][1] = sum_im; +#endif /* USE_FIXED */ + } +} +static void ps_hybrid_analysis_ileave_c(INTFLOAT (*out)[32][2], INTFLOAT L[2][38][64], + int i, int len) +{ + int j; + + for (; i < 64; i++) { + for (j = 0; j < len; j++) { + out[i][j][0] = L[0][j][i]; + out[i][j][1] = L[1][j][i]; + } + } +} + +static void ps_hybrid_synthesis_deint_c(INTFLOAT out[2][38][64], + INTFLOAT (*in)[32][2], + int i, int len) +{ + int n; + + for (; i < 64; i++) { + for (n = 0; n < len; n++) { + out[0][n][i] = in[i][n][0]; + out[1][n][i] = in[i][n][1]; + } + } +} + +static void ps_decorrelate_c(INTFLOAT (*out)[2], INTFLOAT (*delay)[2], + INTFLOAT (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2], + const INTFLOAT phi_fract[2], const INTFLOAT (*Q_fract)[2], + const INTFLOAT *transient_gain, + INTFLOAT g_decay_slope, + int len) +{ + static const INTFLOAT a[] = { Q31(0.65143905753106f), + Q31(0.56471812200776f), + Q31(0.48954165955695f) }; + INTFLOAT ag[PS_AP_LINKS]; + int m, n; + + for (m = 0; m < PS_AP_LINKS; m++) + ag[m] = AAC_MUL30(a[m], g_decay_slope); + + for (n = 0; n < len; n++) { + INTFLOAT in_re = AAC_MSUB30(delay[n][0], phi_fract[0], delay[n][1], phi_fract[1]); + INTFLOAT in_im = AAC_MADD30(delay[n][0], phi_fract[1], delay[n][1], phi_fract[0]); + for (m = 0; m < PS_AP_LINKS; m++) { + INTFLOAT a_re = AAC_MUL31(ag[m], in_re); + INTFLOAT a_im = AAC_MUL31(ag[m], in_im); + INTFLOAT link_delay_re = ap_delay[m][n+2-m][0]; + INTFLOAT link_delay_im = ap_delay[m][n+2-m][1]; + INTFLOAT fractional_delay_re = Q_fract[m][0]; + INTFLOAT fractional_delay_im = Q_fract[m][1]; + INTFLOAT apd_re = in_re; + INTFLOAT apd_im = in_im; + in_re = AAC_MSUB30(link_delay_re, fractional_delay_re, + link_delay_im, fractional_delay_im); + in_re -= a_re; + in_im = AAC_MADD30(link_delay_re, fractional_delay_im, + link_delay_im, fractional_delay_re); + in_im -= a_im; + ap_delay[m][n+5][0] = apd_re + AAC_MUL31(ag[m], in_re); + ap_delay[m][n+5][1] = apd_im + AAC_MUL31(ag[m], in_im); + } + out[n][0] = AAC_MUL16(transient_gain[n], in_re); + out[n][1] = AAC_MUL16(transient_gain[n], in_im); + } +} + +static void ps_stereo_interpolate_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2], + INTFLOAT h[2][4], INTFLOAT h_step[2][4], + int len) +{ + INTFLOAT h0 = h[0][0]; + INTFLOAT h1 = h[0][1]; + INTFLOAT h2 = h[0][2]; + INTFLOAT h3 = h[0][3]; + INTFLOAT hs0 = h_step[0][0]; + INTFLOAT hs1 = h_step[0][1]; + INTFLOAT hs2 = h_step[0][2]; + INTFLOAT hs3 = h_step[0][3]; + int n; + + for (n = 0; n < len; n++) { + //l is s, r is d + INTFLOAT l_re = l[n][0]; + INTFLOAT l_im = l[n][1]; + INTFLOAT r_re = r[n][0]; + INTFLOAT r_im = r[n][1]; + h0 += hs0; + h1 += hs1; + h2 += hs2; + h3 += hs3; + l[n][0] = AAC_MADD30(h0, l_re, h2, r_re); + l[n][1] = AAC_MADD30(h0, l_im, h2, r_im); + r[n][0] = AAC_MADD30(h1, l_re, h3, r_re); + r[n][1] = AAC_MADD30(h1, l_im, h3, r_im); + } +} + +static void ps_stereo_interpolate_ipdopd_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2], + INTFLOAT h[2][4], INTFLOAT h_step[2][4], + int len) +{ + INTFLOAT h00 = h[0][0], h10 = h[1][0]; + INTFLOAT h01 = h[0][1], h11 = h[1][1]; + INTFLOAT h02 = h[0][2], h12 = h[1][2]; + INTFLOAT h03 = h[0][3], h13 = h[1][3]; + INTFLOAT hs00 = h_step[0][0], hs10 = h_step[1][0]; + INTFLOAT hs01 = h_step[0][1], hs11 = h_step[1][1]; + INTFLOAT hs02 = h_step[0][2], hs12 = h_step[1][2]; + INTFLOAT hs03 = h_step[0][3], hs13 = h_step[1][3]; + int n; + + for (n = 0; n < len; n++) { + //l is s, r is d + INTFLOAT l_re = l[n][0]; + INTFLOAT l_im = l[n][1]; + INTFLOAT r_re = r[n][0]; + INTFLOAT r_im = r[n][1]; + h00 += hs00; + h01 += hs01; + h02 += hs02; + h03 += hs03; + h10 += hs10; + h11 += hs11; + h12 += hs12; + h13 += hs13; + + l[n][0] = AAC_MSUB30_V8(h00, l_re, h02, r_re, h10, l_im, h12, r_im); + l[n][1] = AAC_MADD30_V8(h00, l_im, h02, r_im, h10, l_re, h12, r_re); + r[n][0] = AAC_MSUB30_V8(h01, l_re, h03, r_re, h11, l_im, h13, r_im); + r[n][1] = AAC_MADD30_V8(h01, l_im, h03, r_im, h11, l_re, h13, r_re); + } +} + +av_cold void AAC_RENAME(ff_psdsp_init)(PSDSPContext *s) +{ + s->add_squares = ps_add_squares_c; + s->mul_pair_single = ps_mul_pair_single_c; + s->hybrid_analysis = ps_hybrid_analysis_c; + s->hybrid_analysis_ileave = ps_hybrid_analysis_ileave_c; + s->hybrid_synthesis_deint = ps_hybrid_synthesis_deint_c; + s->decorrelate = ps_decorrelate_c; + s->stereo_interpolate[0] = ps_stereo_interpolate_c; + s->stereo_interpolate[1] = ps_stereo_interpolate_ipdopd_c; + +#if !USE_FIXED + if (ARCH_ARM) + ff_psdsp_init_arm(s); + if (ARCH_MIPS) + ff_psdsp_init_mips(s); +#endif /* !USE_FIXED */ +} -- cgit v1.2.3