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authorVitor Sessak <vitor1001@gmail.com>2010-07-01 21:04:12 +0000
committerVitor Sessak <vitor1001@gmail.com>2010-07-01 21:04:12 +0000
commit060dd9300041eff91e7fb5622c12f9dd4975502d (patch)
tree4de6f858261075ed807f8f5c31dd13fd11aa4c9f
parentde1807bbc4ea29e9492484e1104d4da33cd6ba28 (diff)
Altivec-optimized mp{1,2,3} windowing
Originally committed as revision 23943 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--libavcodec/mpegaudio.h1
-rw-r--r--libavcodec/mpegaudiodec.c2
-rw-r--r--libavcodec/ppc/Makefile6
-rw-r--r--libavcodec/ppc/mpegaudiodec_altivec.c130
4 files changed, 139 insertions, 0 deletions
diff --git a/libavcodec/mpegaudio.h b/libavcodec/mpegaudio.h
index 5a530b71e8..e2ad911b0c 100644
--- a/libavcodec/mpegaudio.h
+++ b/libavcodec/mpegaudio.h
@@ -188,6 +188,7 @@ void ff_mpa_synth_filter_float(MPADecodeContext *s,
INTFLOAT sb_samples[SBLIMIT]);
void ff_mpegaudiodec_init_mmx(MPADecodeContext *s);
+void ff_mpegaudiodec_init_altivec(MPADecodeContext *s);
/* fast header check for resync */
static inline int ff_mpa_check_header(uint32_t header){
diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c
index 0ae21e03fb..856eebd135 100644
--- a/libavcodec/mpegaudiodec.c
+++ b/libavcodec/mpegaudiodec.c
@@ -325,6 +325,8 @@ static av_cold int decode_init(AVCodecContext * avctx)
#if HAVE_MMX
ff_mpegaudiodec_init_mmx(s);
#endif
+ if (HAVE_ALTIVEC && CONFIG_FLOAT) ff_mpegaudiodec_init_altivec(s);
+
avctx->sample_fmt= OUT_FMT;
s->error_recognition= avctx->error_recognition;
diff --git a/libavcodec/ppc/Makefile b/libavcodec/ppc/Makefile
index 32c9082342..3bbd6c3630 100644
--- a/libavcodec/ppc/Makefile
+++ b/libavcodec/ppc/Makefile
@@ -7,6 +7,12 @@ ALTIVEC-OBJS-$(CONFIG_VP5_DECODER) += ppc/vp3dsp_altivec.o
ALTIVEC-OBJS-$(CONFIG_VP6_DECODER) += ppc/vp3dsp_altivec.o
ALTIVEC-OBJS-$(CONFIG_VP8_DECODER) += ppc/vp8dsp_altivec.o
+ALTIVEC-OBJS-$(CONFIG_MP1FLOAT_DECODER) += ppc/mpegaudiodec_altivec.o
+ALTIVEC-OBJS-$(CONFIG_MP2FLOAT_DECODER) += ppc/mpegaudiodec_altivec.o
+ALTIVEC-OBJS-$(CONFIG_MP3FLOAT_DECODER) += ppc/mpegaudiodec_altivec.o
+ALTIVEC-OBJS-$(CONFIG_MP3ON4FLOAT_DECODER) += ppc/mpegaudiodec_altivec.o
+ALTIVEC-OBJS-$(CONFIG_MP3ADUFLOAT_DECODER) += ppc/mpegaudiodec_altivec.o
+
OBJS-$(HAVE_ALTIVEC) += ppc/check_altivec.o \
ppc/dsputil_altivec.o \
ppc/fdct_altivec.o \
diff --git a/libavcodec/ppc/mpegaudiodec_altivec.c b/libavcodec/ppc/mpegaudiodec_altivec.c
new file mode 100644
index 0000000000..e087d4add1
--- /dev/null
+++ b/libavcodec/ppc/mpegaudiodec_altivec.c
@@ -0,0 +1,130 @@
+/*
+ * Altivec optimized MP3 decoding functions
+ * Copyright (c) 2010 Vitor Sessak
+ *
+ * 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
+ */
+
+#include "dsputil_altivec.h"
+#include "util_altivec.h"
+
+#define CONFIG_FLOAT 1
+#include "libavcodec/mpegaudio.h"
+
+#define MACS(rt, ra, rb) rt+=(ra)*(rb)
+#define MLSS(rt, ra, rb) rt-=(ra)*(rb)
+
+#define SUM8(op, sum, w, p) \
+{ \
+ op(sum, (w)[0 * 64], (p)[0 * 64]); \
+ op(sum, (w)[1 * 64], (p)[1 * 64]); \
+ op(sum, (w)[2 * 64], (p)[2 * 64]); \
+ op(sum, (w)[3 * 64], (p)[3 * 64]); \
+ op(sum, (w)[4 * 64], (p)[4 * 64]); \
+ op(sum, (w)[5 * 64], (p)[5 * 64]); \
+ op(sum, (w)[6 * 64], (p)[6 * 64]); \
+ op(sum, (w)[7 * 64], (p)[7 * 64]); \
+}
+
+static void apply_window(const float *buf, const float *win1,
+ const float *win2, float *sum1, float *sum2, int len)
+{
+ const vector float *win1a = (const vector float *) win1;
+ const vector float *win2a = (const vector float *) win2;
+ const vector float *bufa = (const vector float *) buf;
+ vector float *sum1a = (vector float *) sum1;
+ vector float *sum2a = (vector float *) sum2;
+ vector float av_uninit(v0), av_uninit(v4);
+ vector float v1, v2, v3;
+
+ len = len >> 2;
+
+#define MULT(a, b) \
+ { \
+ v1 = vec_ld(a, win1a); \
+ v2 = vec_ld(b, win2a); \
+ v3 = vec_ld(a, bufa); \
+ v0 = vec_madd(v3, v1, v0); \
+ v4 = vec_madd(v2, v3, v4); \
+ }
+
+ while (len--) {
+ v0 = vec_xor(v0, v0);
+ v4 = vec_xor(v4, v4);
+
+ MULT( 0, 0);
+ MULT( 256, 64);
+ MULT( 512, 128);
+ MULT( 768, 192);
+ MULT(1024, 256);
+ MULT(1280, 320);
+ MULT(1536, 384);
+ MULT(1792, 448);
+
+ vec_st(v0, 0, sum1a);
+ vec_st(v4, 0, sum2a);
+ sum1a++;
+ sum2a++;
+ win1a++;
+ win2a++;
+ bufa++;
+ }
+}
+
+static void apply_window_mp3(float *in, float *win, int *unused, float *out,
+ int incr)
+{
+ LOCAL_ALIGNED_16(float, suma, [17]);
+ LOCAL_ALIGNED_16(float, sumb, [17]);
+ LOCAL_ALIGNED_16(float, sumc, [17]);
+ LOCAL_ALIGNED_16(float, sumd, [17]);
+
+ float sum;
+ int j;
+ float *out2 = out + 32 * incr;
+
+ /* copy to avoid wrap */
+ memcpy(in + 512, in, 32 * sizeof(*in));
+
+ apply_window(in + 16, win , win + 512, suma, sumc, 16);
+ apply_window(in + 32, win + 48, win + 640, sumb, sumd, 16);
+
+ SUM8(MLSS, suma[0], win + 32, in + 48);
+
+ sumc[ 0] = 0;
+ sumb[16] = 0;
+ sumd[16] = 0;
+
+ out[0 ] = suma[ 0];
+ out += incr;
+ out2 -= incr;
+ for(j=1;j<16;j++) {
+ *out = suma[ j] - sumd[16-j];
+ *out2 = -sumb[16-j] - sumc[ j];
+ out += incr;
+ out2 -= incr;
+ }
+
+ sum = 0;
+ SUM8(MLSS, sum, win + 16 + 32, in + 32);
+ *out = sum;
+}
+
+void ff_mpegaudiodec_init_altivec(MPADecodeContext *s)
+{
+ s->apply_window_mp3 = apply_window_mp3;
+}