avcodec: Implementation of AAC_fixed_decoder (PS-module)

Add fixed point implementation.

Signed-off-by: Nedeljko Babic <nedeljko.babic@imgtec.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

1 files changed, 228 insertions, 0 deletions

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 <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);
+#endif /* !USE_FIXED */
+}