libavcodec/aacpsdsp_template.c - nest-android-app/ffmpeg - Git at Google

 /*
  * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
  *
  * 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 <stdint.h>

 #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);
     if (ARCH_X86)
         ff_psdsp_init_x86(s);
 #endif /* !USE_FIXED */
 }
	/*
	* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
	*
	* 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 <stdint.h>

	#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);
	if (ARCH_X86)
	ff_psdsp_init_x86(s);
	#endif /* !USE_FIXED */
	}