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Merge commit 'a24a252709dd38f12aa4929ce4981f87091a5113'

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* commit 'a24a252709dd38f12aa4929ce4981f87091a5113':
  aarch64: NEON optimized FIR audio resampling

Merged-by: Michael Niedermayer <michaelni@gmx.at>
pull/108/head
Michael Niedermayer 11 years ago
parent 26953ed2e3 a24a252709
commit e32fc9b45b
6 changed files with 342 additions and 2 deletions
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  1. 6
      libavresample/aarch64/Makefile
  2. 28
      libavresample/aarch64/asm-offsets.h
  3. 71
      libavresample/aarch64/resample_init.c
  4. 233
      libavresample/aarch64/resample_neon.S
  5. 3
      libavresample/internal.h
  6. 3
      libavresample/resample.c

6
libavresample/aarch64/Makefile
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@ -1,5 +1,7 @@
OBJS += aarch64/audio_convert_init.o OBJS += aarch64/audio_convert_init.o \
aarch64/resample_init.o \
OBJS-$(CONFIG_NEON_CLOBBER_TEST) += aarch64/neontest.o OBJS-$(CONFIG_NEON_CLOBBER_TEST) += aarch64/neontest.o
NEON-OBJS += aarch64/audio_convert_neon.o NEON-OBJS += aarch64/audio_convert_neon.o \
aarch64/resample_neon.o \

28
libavresample/aarch64/asm-offsets.h
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@ -0,0 +1,28 @@
/*
* 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 AVRESAMPLE_AARCH64_ASM_OFFSETS_H
#define AVRESAMPLE_AARCH64_ASM_OFFSETS_H
/* struct ResampleContext */
#define FILTER_BANK 0x10
#define FILTER_LENGTH 0x18
#define PHASE_SHIFT 0x34
#define PHASE_MASK (PHASE_SHIFT + 0x04) // loaded as pair
#endif /* AVRESAMPLE_AARCH64_ASM_OFFSETS_H */

71
libavresample/aarch64/resample_init.c
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@ -0,0 +1,71 @@
/*
* 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 <stdint.h>
#include "config.h"
#include "libavutil/cpu.h"
#include "libavutil/aarch64/cpu.h"
#include "libavutil/internal.h"
#include "libavutil/samplefmt.h"
#include "libavresample/resample.h"
#include "asm-offsets.h"
AV_CHECK_OFFSET(struct ResampleContext, filter_bank, FILTER_BANK);
AV_CHECK_OFFSET(struct ResampleContext, filter_length, FILTER_LENGTH);
AV_CHECK_OFFSET(struct ResampleContext, phase_shift, PHASE_SHIFT);
AV_CHECK_OFFSET(struct ResampleContext, phase_mask, PHASE_MASK);
void ff_resample_one_dbl_neon(struct ResampleContext *c, void *dst0,
int dst_index, const void *src0,
unsigned int index, int frac);
void ff_resample_one_flt_neon(struct ResampleContext *c, void *dst0,
int dst_index, const void *src0,
unsigned int index, int frac);
void ff_resample_one_s16_neon(struct ResampleContext *c, void *dst0,
int dst_index, const void *src0,
unsigned int index, int frac);
void ff_resample_one_s32_neon(struct ResampleContext *c, void *dst0,
int dst_index, const void *src0,
unsigned int index, int frac);
void ff_audio_resample_init_aarch64(ResampleContext *c,
enum AVSampleFormat sample_fmt)
{
int cpu_flags = av_get_cpu_flags();
if (have_neon(cpu_flags)) {
if (!c->linear) {
switch (sample_fmt) {
case AV_SAMPLE_FMT_DBLP:
c->resample_one = ff_resample_one_dbl_neon;
break;
case AV_SAMPLE_FMT_FLTP:
c->resample_one = ff_resample_one_flt_neon;
break;
case AV_SAMPLE_FMT_S16P:
c->resample_one = ff_resample_one_s16_neon;
break;
case AV_SAMPLE_FMT_S32P:
c->resample_one = ff_resample_one_s32_neon;
break;
}
}
}
}

233
libavresample/aarch64/resample_neon.S
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@ -0,0 +1,233 @@
/*
* Copyright (c) 2014 Janne Grunau <janne-libav@jannau.net>
*
* 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 "libavutil/aarch64/asm.S"
#include "asm-offsets.h"
.macro resample_one fmt, es=2
.ifnc \fmt, dbl
.macro M_MUL2 x:vararg
.endm
.macro M_MLA2 x:vararg
.endm
.endif
function ff_resample_one_\fmt\()_neon, export=1
sxtw x2, w2
ldr x9, [x0, #FILTER_BANK]
ldr w6, [x0, #FILTER_LENGTH]
ldp w7, w8, [x0, #PHASE_SHIFT] // and phase_mask
lsr x10, x4, x7 // sample_index
and x4, x4, x8
lsl x11, x6, #\es // filter_length * elem_size
add x3, x3, x10, lsl #\es // src[sample_index]
madd x9, x11, x4, x9 // filter
cmp w6, #16
b.lt 5f
8: // remaining filter_length at least 16
subs w6, w6, #16
LOAD8 v4, v5, v6, v7, x3
LOAD8 v16, v17, v18, v19, x9
M_MUL v0, v4, v16, v1
M_MUL2 v1, v6, v18
7:
LOAD8 v20, v21, v22, v23, x3
M_MLA v0, v5, v17, v1
M_MLA2 v1, v7, v19
LOAD8 v24, v25, v26, v27, x9
M_MLA v0, v20, v24, v1
M_MLA2 v1, v22, v26
b.eq 6f
cmp w6, #16
M_MLA v0, v21, v25, v1
M_MLA2 v1, v23, v27
b.lt 4f
subs w6, w6, #16
LOAD8 v4, v5, v6, v7, x3
LOAD8 v16, v17, v18, v19, x9
M_MLA v0, v4, v16, v1
M_MLA2 v1, v6, v18
b 7b
6:
M_MLA v0, v21, v25, v1
M_MLA2 v1, v23, v27
STORE_ONE 0, x1, x2, v1
ret
5:
movi v0.16b, #0
movi v1.16b, #0
4: // remaining filter_length 1-15
cmp w6, #4
b.lt 2f
subs w6, w6, #4
LOAD4 v4, v5, x3
LOAD4 v6, v7, x9
M_MLA v0, v4, v6, v1
M_MLA2 v1, v5, v7
b.eq 0f
b 4b
2: // remaining filter_length 1-3
cmp w6, #2
b.lt 1f
LOAD2 2, x3
LOAD2 3, x9
subs w6, w6, #2
M_MLA v0, v2, v3
b.eq 0f
1: // remaining filter_length 1
LOAD1 6, x3
LOAD1 7, x9
M_MLA v0, v6, v7
0:
STORE_ONE 0, x1, x2, v1
ret
endfunc
.purgem LOAD1
.purgem LOAD2
.purgem LOAD4
.purgem LOAD8
.purgem M_MLA
.purgem M_MLA2
.purgem M_MUL
.purgem M_MUL2
.purgem STORE_ONE
.endm
.macro LOAD1 d1, addr
ldr d\d1, [\addr], #8
.endm
.macro LOAD2 d1, addr
ld1 {v\d1\().2d}, [\addr], #16
.endm
.macro LOAD4 d1, d2, addr
ld1 {\d1\().2d,\d2\().2d}, [\addr], #32
.endm
.macro LOAD8 d1, d2, d3, d4, addr
ld1 {\d1\().2d,\d2\().2d,\d3\().2d,\d4\().2d}, [\addr], #64
.endm
.macro M_MLA d, r0, r1, d2:vararg
fmla \d\().2d, \r0\().2d, \r1\().2d
.endm
.macro M_MLA2 second:vararg
M_MLA \second
.endm
.macro M_MUL d, r0, r1, d2:vararg
fmul \d\().2d, \r0\().2d, \r1\().2d
.endm
.macro M_MUL2 second:vararg
M_MUL \second
.endm
.macro STORE_ONE rn, addr, idx, d2
fadd v\rn\().2d, v\rn\().2d, \d2\().2d
faddp d\rn\(), v\rn\().2d
str d\rn\(), [\addr, \idx, lsl #3]
.endm
resample_one dbl, 3
.macro LOAD1 d1, addr
ldr s\d1, [\addr], #4
.endm
.macro LOAD2 d1, addr
ld1 {v\d1\().2s}, [\addr], #8
.endm
.macro LOAD4 d1, d2, addr
ld1 {\d1\().4s}, [\addr], #16
.endm
.macro LOAD8 d1, d2, d3, d4, addr
ld1 {\d1\().4s,\d2\().4s}, [\addr], #32
.endm
.macro M_MLA d, r0, r1, d2:vararg
fmla \d\().4s, \r0\().4s, \r1\().4s
.endm
.macro M_MUL d, r0, r1, d2:vararg
fmul \d\().4s, \r0\().4s, \r1\().4s
.endm
.macro STORE_ONE rn, addr, idx, d2
faddp v\rn\().4s, v\rn\().4s, v\rn\().4s
faddp s\rn\(), v\rn\().2s
str s\rn\(), [\addr, \idx, lsl #2]
.endm
resample_one flt
.macro LOAD1 d1, addr
ldr h\d1, [\addr], #2
.endm
.macro LOAD2 d1, addr
ldr s\d1, [\addr], #4
.endm
.macro LOAD4 d1, d2, addr
ld1 {\d1\().4h}, [\addr], #8
.endm
.macro LOAD8 d1, d2, d3, d4, addr
ld1 {\d1\().4h,\d2\().4h}, [\addr], #16
.endm
.macro M_MLA d, r0, r1, d2:vararg
smlal \d\().4s, \r0\().4h, \r1\().4h
.endm
.macro M_MUL d, r0, r1, d2:vararg
smull \d\().4s, \r0\().4h, \r1\().4h
.endm
.macro STORE_ONE rn, addr, idx, d2
addp v\rn\().4s, v\rn\().4s, v\rn\().4s
addp v\rn\().4s, v\rn\().4s, v\rn\().4s
sqrshrn v\rn\().4h, v\rn\().4s, #15
str h\rn\(), [\addr, \idx, lsl #1]
.endm
resample_one s16, 1
.macro LOAD1 d1, addr
ldr s\d1, [\addr], #4
.endm
.macro LOAD2 d1, addr
ld1 {v\d1\().2s}, [\addr], #8
.endm
.macro LOAD4 d1, d2, addr
ld1 {\d1\().4s}, [\addr], #16
.endm
.macro LOAD8 d1, d2, d3, d4, addr
ld1 {\d1\().4s,\d2\().4s}, [\addr], #32
.endm
.macro M_MLA d1, r0, r1, d2:vararg
smlal \d1\().2d, \r0\().2s, \r1\().2s
.ifnb \d2
smlal2 \d2\().2d, \r0\().4s, \r1\().4s
.endif
.endm
.macro M_MUL d1, r0, r1, d2:vararg
smull \d1\().2d, \r0\().2s, \r1\().2s
.ifnb \d2
smull2 \d2\().2d, \r0\().4s, \r1\().4s
.endif
.endm
.macro STORE_ONE rn, addr, idx, d2
add v\rn\().2d, v\rn\().2d, \d2\().2d
addp d\rn\(), v\rn\().2d
sqrshrn v\rn\().2s, v\rn\().2d, #30
str s\rn\(), [\addr, \idx, lsl #2]
.endm
resample_one s32

3
libavresample/internal.h
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@ -107,4 +107,7 @@ struct AVAudioResampleContext {
ChannelMapInfo ch_map_info; ChannelMapInfo ch_map_info;
}; };
void ff_audio_resample_init_aarch64(ResampleContext *c,
enum AVSampleFormat sample_fmt);
#endif /* AVRESAMPLE_INTERNAL_H */ #endif /* AVRESAMPLE_INTERNAL_H */

3
libavresample/resample.c
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@ -170,6 +170,9 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
break; break;
} }
if (ARCH_AARCH64)
ff_audio_resample_init_aarch64(c, avr->internal_sample_fmt);
felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt); felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size); c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
if (!c->filter_bank) if (!c->filter_bank)

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