1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-18 03:19:31 +02:00
FFmpeg/libavcodec/audiodsp.h
Clément Bœsch 83cd80d10a Merge commit '12004a9a7f20e44f4da2ee6c372d5e1794c8d6c5'
* commit '12004a9a7f20e44f4da2ee6c372d5e1794c8d6c5':
  audiodsp/x86: yasmify vector_clipf_sse
  audiodsp: reorder arguments for vector_clipf

Merged the version from Libav after a discussion with James Almer on
IRC:

19:22 <ubitux> jamrial: opinion on 12004a9a7f20e44f4da2ee6c372d5e1794c8d6c5?
19:23 <ubitux> it was apparently yasmified differently
19:23 <ubitux> (it depends on the previous commit arg shuffle)
19:24 <ubitux> i don't see the magic movsxdifnidn in your port btw
19:24 <ubitux> it's a port from 1d36defe94
19:25 <jamrial> seems better thanks to said arg shuffle
19:25 <jamrial> the loop is the same, but init is simpler
19:25 <jamrial> probably worth merging
19:25 <ubitux> OK
19:25 <ubitux> thanks
19:26 <jamrial> curious they didn't make len ptrdiff_t after the previous bunch of commits, heh
19:26 <ubitux> yeah indeed

Both commits are merged at the same time to prevent a conflict with our
existing yasmified ff_vector_clipf_sse.

Merged-by: Clément Bœsch <u@pkh.me>
2017-03-20 22:35:07 +01:00

61 lines
2.3 KiB
C

/*
* 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 16, and arrays are 16-byte aligned */
void (*vector_clipf)(float *dst /* align 16 */,
const float *src /* align 16 */,
int len /* align 16 */,
float min, float max);
} 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 */