Separate format conversion DSP functions from DSPContext.

This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.

Signed-off-by: Mans Rullgard <mans@mansr.com>

1 files changed, 68 insertions, 0 deletions

diff --git a/libavcodec/fmtconvert.c b/libavcodec/fmtconvert.c
new file mode 100644
index 0000000000..e26b8997ab
--- /dev/null
+++ b/libavcodec/fmtconvert.c
@@ -0,0 +1,68 @@
+/*
+ * Format Conversion Utils
+ * Copyright (c) 2000, 2001 Fabrice Bellard
+ * Copyright (c) 2002-2004 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * 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 "avcodec.h"
+#include "fmtconvert.h"
+
+static void int32_to_float_fmul_scalar_c(float *dst, const int *src, float mul, int len){
+    int i;
+    for(i=0; i<len; i++)
+        dst[i] = src[i] * mul;
+}
+
+static av_always_inline int float_to_int16_one(const float *src){
+    return av_clip_int16(lrintf(*src));
+}
+
+static void float_to_int16_c(int16_t *dst, const float *src, long len)
+{
+    int i;
+    for(i=0; i<len; i++)
+        dst[i] = float_to_int16_one(src+i);
+}
+
+static void float_to_int16_interleave_c(int16_t *dst, const float **src,
+                                        long len, int channels)
+{
+    int i,j,c;
+    if(channels==2){
+        for(i=0; i<len; i++){
+            dst[2*i]   = float_to_int16_one(src[0]+i);
+            dst[2*i+1] = float_to_int16_one(src[1]+i);
+        }
+    }else{
+        for(c=0; c<channels; c++)
+            for(i=0, j=c; i<len; i++, j+=channels)
+                dst[j] = float_to_int16_one(src[c]+i);
+    }
+}
+
+av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
+{
+    c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_c;
+    c->float_to_int16             = float_to_int16_c;
+    c->float_to_int16_interleave  = float_to_int16_interleave_c;
+
+    if (ARCH_ARM) ff_fmt_convert_init_arm(c, avctx);
+    if (ARCH_PPC) ff_fmt_convert_init_ppc(c, avctx);
+    if (HAVE_MMX) ff_fmt_convert_init_x86(c, avctx);
+}