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author | Michael Niedermayer <michaelni@gmx.at> | 2014-06-22 17:58:28 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2014-06-22 17:58:28 +0200 |
commit | 99497b4683e5054bcdc5b6802a27d717df9e04f3 (patch) | |
tree | 130022374c1a92b72288272bd0927ae6ac7d825b /libavcodec/audiodsp.h | |
parent | 0dae193d3ecf5d0dc687f5ad708419bf7600de9a (diff) | |
parent | 9a9e2f1c8aa4539a261625145e5c1f46a8106ac2 (diff) | |
download | ffmpeg-99497b4683e5054bcdc5b6802a27d717df9e04f3.tar.gz |
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.h | 59 |
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 */ |