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authorMichael Niedermayer <michaelni@gmx.at>2014-06-22 19:58:28 +0400
committerMichael Niedermayer <michaelni@gmx.at>2014-06-22 19:58:28 +0400
commit99497b4683e5054bcdc5b6802a27d717df9e04f3 (patch)
tree130022374c1a92b72288272bd0927ae6ac7d825b /libavcodec/audiodsp.h
parent0dae193d3ecf5d0dc687f5ad708419bf7600de9a (diff)
parent9a9e2f1c8aa4539a261625145e5c1f46a8106ac2 (diff)
Merge commit '9a9e2f1c8aa4539a261625145e5c1f46a8106ac2'
* commit '9a9e2f1c8aa4539a261625145e5c1f46a8106ac2': dsputil: Split audio operations off into a separate context Conflicts: configure libavcodec/takdec.c libavcodec/x86/Makefile libavcodec/x86/dsputil.asm libavcodec/x86/dsputil_init.c libavcodec/x86/dsputil_mmx.c libavcodec/x86/dsputil_x86.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/audiodsp.h')
-rw-r--r--libavcodec/audiodsp.h59
1 files changed, 59 insertions, 0 deletions
diff --git a/libavcodec/audiodsp.h b/libavcodec/audiodsp.h
new file mode 100644
index 0000000000..b55bf858e0
--- /dev/null
+++ b/libavcodec/audiodsp.h
@@ -0,0 +1,59 @@
+/*
+ * 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
+ */
+
+#ifndef AVCODEC_AUDIODSP_H
+#define AVCODEC_AUDIODSP_H
+
+#include <stdint.h>
+
+typedef struct AudioDSPContext {
+ /**
+ * Calculate scalar product of two vectors.
+ * @param len length of vectors, should be multiple of 16
+ */
+ int32_t (*scalarproduct_int16)(const int16_t *v1,
+ const int16_t *v2 /* align 16 */, int len);
+
+ /**
+ * Clip each element in an array of int32_t to a given minimum and
+ * maximum value.
+ * @param dst destination array
+ * constraints: 16-byte aligned
+ * @param src source array
+ * constraints: 16-byte aligned
+ * @param min minimum value
+ * constraints: must be in the range [-(1 << 24), 1 << 24]
+ * @param max maximum value
+ * constraints: must be in the range [-(1 << 24), 1 << 24]
+ * @param len number of elements in the array
+ * constraints: multiple of 32 greater than zero
+ */
+ void (*vector_clip_int32)(int32_t *dst, const int32_t *src, int32_t min,
+ int32_t max, unsigned int len);
+ /* assume len is a multiple of 8, and arrays are 16-byte aligned */
+ void (*vector_clipf)(float *dst /* align 16 */,
+ const float *src /* align 16 */,
+ float min, float max, int len /* align 16 */);
+} AudioDSPContext;
+
+void ff_audiodsp_init(AudioDSPContext *c);
+void ff_audiodsp_init_arm(AudioDSPContext *c);
+void ff_audiodsp_init_ppc(AudioDSPContext *c);
+void ff_audiodsp_init_x86(AudioDSPContext *c);
+
+#endif /* AVCODEC_AUDIODSP_H */