swr: split out DSP functions.

DSP bits of swri_resample go into their own mini-DSP functions; DSP
init goes from a per-call branch in multiple_resample to a proper
DSP init routine; x86 bits go into x86/; swri_resample() moves out of
resample_template.c into resample.c because it's independent of DSP
code or sample type; multiple_resample() is simplified.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
pull/76/merge
Ronald S. Bultje 11 years ago committed by Michael Niedermayer
parent 9236f7b5a2
commit 7128a35f8c
  1. 1
      libswresample/Makefile
  2. 149
      libswresample/resample.c
  3. 66
      libswresample/resample.h
  4. 69
      libswresample/resample_dsp.c
  5. 79
      libswresample/resample_template.c
  6. 2
      libswresample/x86/Makefile
  7. 89
      libswresample/x86/resample_x86_dsp.c

@ -10,6 +10,7 @@ OBJS = audioconvert.o \
dither.o \
rematrix.o \
resample.o \
resample_dsp.o \
swresample.o \
OBJS-$(CONFIG_LIBSOXR) += soxr_resample.o

@ -25,32 +25,8 @@
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "libavutil/log.h"
#include "libavutil/avassert.h"
#include "swresample_internal.h"
typedef struct ResampleContext {
const AVClass *av_class;
uint8_t *filter_bank;
int filter_length;
int filter_alloc;
int ideal_dst_incr;
int dst_incr;
int index;
int frac;
int src_incr;
int compensation_distance;
int phase_shift;
int phase_mask;
int linear;
enum SwrFilterType filter_type;
int kaiser_beta;
double factor;
enum AVSampleFormat format;
int felem_size;
int filter_shift;
} ResampleContext;
#include "resample.h"
/**
* 0th order modified bessel function of the first kind.
@ -197,7 +173,8 @@ static int build_filter(ResampleContext *c, void *filter, double factor, int tap
static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta,
double precision, int cheby){
double precision, int cheby)
{
double cutoff = cutoff0? cutoff0 : 0.97;
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
int phase_count= 1<<phase_shift;
@ -259,6 +236,8 @@ static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_r
c->index= -phase_count*((c->filter_length-1)/2);
c->frac= 0;
swresample_dsp_init(c);
return c;
error:
av_freep(&c->filter_bank);
@ -282,59 +261,53 @@ static int set_compensation(ResampleContext *c, int sample_delta, int compensati
return 0;
}
#define TEMPLATE_RESAMPLE_S16
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_S16
#define TEMPLATE_RESAMPLE_S32
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_S32
#define TEMPLATE_RESAMPLE_FLT
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_FLT
#define TEMPLATE_RESAMPLE_DBL
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_DBL
// XXX FIXME the whole C loop should be written in asm so this x86 specific code here isnt needed
#if HAVE_MMXEXT_INLINE
#include "x86/resample_mmx.h"
#define TEMPLATE_RESAMPLE_S16_MMX2
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_S16_MMX2
#if HAVE_SSE_INLINE
#define TEMPLATE_RESAMPLE_FLT_SSE
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_FLT_SSE
#endif
#if HAVE_SSE2_INLINE
#define TEMPLATE_RESAMPLE_S16_SSE2
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_S16_SSE2
#define TEMPLATE_RESAMPLE_DBL_SSE2
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_DBL_SSE2
#endif
#if HAVE_AVX_INLINE
#define TEMPLATE_RESAMPLE_FLT_AVX
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_FLT_AVX
#endif
static int swri_resample(ResampleContext *c,
uint8_t *dst, const uint8_t *src, int *consumed,
int src_size, int dst_size, int update_ctx)
{
int fn_idx = c->format - AV_SAMPLE_FMT_S16P;
if (c->filter_length == 1 && c->phase_shift == 0) {
int index= c->index;
int frac= c->frac;
int dst_incr_frac= c->dst_incr % c->src_incr;
int dst_incr= c->dst_incr / c->src_incr;
int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*index;
int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
int new_size = (src_size * (int64_t)c->src_incr - frac + c->dst_incr - 1) / c->dst_incr;
dst_size= FFMIN(dst_size, new_size);
c->dsp.resample_one[fn_idx](dst, src, dst_size, index2, incr);
index += dst_size * dst_incr;
index += (frac + dst_size * (int64_t)dst_incr_frac) / c->src_incr;
av_assert2(index >= 0);
*consumed= index;
if (update_ctx) {
c->frac = (frac + dst_size * (int64_t)dst_incr_frac) % c->src_incr;
c->index = 0;
}
} else {
int64_t end_index = (1LL + src_size - c->filter_length) << c->phase_shift;
int64_t delta_frac = (end_index - c->index) * c->src_incr - c->frac;
int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr;
dst_size = FFMIN(dst_size, delta_n);
if (!c->linear) {
*consumed = c->dsp.resample_common[fn_idx](c, dst, src, dst_size, update_ctx);
} else {
*consumed = c->dsp.resample_linear[fn_idx](c, dst, src, dst_size, update_ctx);
}
}
#endif // HAVE_MMXEXT_INLINE
return dst_size;
}
static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
int i, ret= -1;
int av_unused mm_flags = av_get_cpu_flags();
int need_emms= 0;
int need_emms = c->format == AV_SAMPLE_FMT_S16P && ARCH_X86_32 &&
(mm_flags & (AV_CPU_FLAG_MMX2 | AV_CPU_FLAG_SSE2)) == AV_CPU_FLAG_MMX2;
int64_t max_src_size = (INT64_MAX >> (c->phase_shift+1)) / c->src_incr;
if (c->compensation_distance)
@ -342,32 +315,8 @@ static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, A
src_size = FFMIN(src_size, max_src_size);
for(i=0; i<dst->ch_count; i++){
#if HAVE_MMXEXT_INLINE
#if HAVE_SSE2_INLINE
if(c->format == AV_SAMPLE_FMT_S16P && (mm_flags&AV_CPU_FLAG_SSE2)) ret= swri_resample_int16_sse2 (c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
else
#endif
if(c->format == AV_SAMPLE_FMT_S16P && (mm_flags&AV_CPU_FLAG_MMX2 )){
ret= swri_resample_int16_mmx2 (c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
need_emms= 1;
} else
#endif
if(c->format == AV_SAMPLE_FMT_S16P) ret= swri_resample_int16(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
else if(c->format == AV_SAMPLE_FMT_S32P) ret= swri_resample_int32(c, (int32_t*)dst->ch[i], (const int32_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
#if HAVE_AVX_INLINE
else if(c->format == AV_SAMPLE_FMT_FLTP && (mm_flags&AV_CPU_FLAG_AVX))
ret= swri_resample_float_avx (c, (float*)dst->ch[i], (const float*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
#endif
#if HAVE_SSE_INLINE
else if(c->format == AV_SAMPLE_FMT_FLTP && (mm_flags&AV_CPU_FLAG_SSE))
ret= swri_resample_float_sse (c, (float*)dst->ch[i], (const float*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
#endif
else if(c->format == AV_SAMPLE_FMT_FLTP) ret= swri_resample_float(c, (float *)dst->ch[i], (const float *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
#if HAVE_SSE2_INLINE
else if(c->format == AV_SAMPLE_FMT_DBLP && (mm_flags&AV_CPU_FLAG_SSE2))
ret= swri_resample_double_sse2(c,(double *)dst->ch[i], (const double *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
#endif
else if(c->format == AV_SAMPLE_FMT_DBLP) ret= swri_resample_double(c,(double *)dst->ch[i], (const double *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
ret= swri_resample(c, dst->ch[i], src->ch[i],
consumed, src_size, dst_size, i+1==dst->ch_count);
}
if(need_emms)
emms_c();

@ -0,0 +1,66 @@
/*
* audio resampling
* Copyright (c) 2004-2012 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
*/
#ifndef SWRESAMPLE_RESAMPLE_H
#define SWRESAMPLE_RESAMPLE_H
#include "libavutil/log.h"
#include "libavutil/samplefmt.h"
#include "swresample_internal.h"
typedef void (*resample_one_fn)(uint8_t *dst, const uint8_t *src,
int n, int64_t index, int64_t incr);
typedef int (*resample_fn)(struct ResampleContext *c, uint8_t *dst,
const uint8_t *src, int n, int update_ctx);
typedef struct ResampleContext {
const AVClass *av_class;
uint8_t *filter_bank;
int filter_length;
int filter_alloc;
int ideal_dst_incr;
int dst_incr;
int index;
int frac;
int src_incr;
int compensation_distance;
int phase_shift;
int phase_mask;
int linear;
enum SwrFilterType filter_type;
int kaiser_beta;
double factor;
enum AVSampleFormat format;
int felem_size;
int filter_shift;
struct {
resample_one_fn resample_one[AV_SAMPLE_FMT_NB - AV_SAMPLE_FMT_S16P];
resample_fn resample_common[AV_SAMPLE_FMT_NB - AV_SAMPLE_FMT_S16P];
resample_fn resample_linear[AV_SAMPLE_FMT_NB - AV_SAMPLE_FMT_S16P];
} dsp;
} ResampleContext;
void swresample_dsp_init(ResampleContext *c);
void swresample_dsp_x86_init(ResampleContext *c);
#endif /* SWRESAMPLE_RESAMPLE_H */

@ -0,0 +1,69 @@
/*
* audio resampling
* Copyright (c) 2004-2012 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
*/
/**
* @file
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "resample.h"
#define DO_RESAMPLE_ONE 1
#define TEMPLATE_RESAMPLE_S16
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_S16
#define TEMPLATE_RESAMPLE_S32
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_S32
#define TEMPLATE_RESAMPLE_FLT
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_FLT
#define TEMPLATE_RESAMPLE_DBL
#include "resample_template.c"
#undef TEMPLATE_RESAMPLE_DBL
#undef DO_RESAMPLE_ONE
void swresample_dsp_init(ResampleContext *c)
{
#define FNIDX(fmt) (AV_SAMPLE_FMT_##fmt - AV_SAMPLE_FMT_S16P)
c->dsp.resample_one[FNIDX(S16P)] = (resample_one_fn) resample_one_int16;
c->dsp.resample_one[FNIDX(S32P)] = (resample_one_fn) resample_one_int32;
c->dsp.resample_one[FNIDX(FLTP)] = (resample_one_fn) resample_one_float;
c->dsp.resample_one[FNIDX(DBLP)] = (resample_one_fn) resample_one_double;
c->dsp.resample_common[FNIDX(S16P)] = (resample_fn) resample_common_int16;
c->dsp.resample_common[FNIDX(S32P)] = (resample_fn) resample_common_int32;
c->dsp.resample_common[FNIDX(FLTP)] = (resample_fn) resample_common_float;
c->dsp.resample_common[FNIDX(DBLP)] = (resample_fn) resample_common_double;
c->dsp.resample_linear[FNIDX(S16P)] = (resample_fn) resample_linear_int16;
c->dsp.resample_linear[FNIDX(S32P)] = (resample_fn) resample_linear_int32;
c->dsp.resample_linear[FNIDX(FLTP)] = (resample_fn) resample_linear_float;
c->dsp.resample_linear[FNIDX(DBLP)] = (resample_fn) resample_linear_double;
if (ARCH_X86) swresample_dsp_x86_init(c);
}

@ -106,45 +106,30 @@
#endif
int RENAME(swri_resample)(ResampleContext *c, DELEM *dst, const DELEM *src, int *consumed, int src_size, int dst_size, int update_ctx){
#if DO_RESAMPLE_ONE
static void RENAME(resample_one)(DELEM *dst, const DELEM *src,
int dst_size, int64_t index2, int64_t incr)
{
int dst_index;
#if !defined(COMMON_CORE) || !defined(LINEAR_CORE)
int i;
for (dst_index = 0; dst_index < dst_size; dst_index++) {
dst[dst_index] = src[index2 >> 32];
index2 += incr;
}
}
#endif
static int RENAME(resample_common)(ResampleContext *c,
DELEM *dst, const DELEM *src,
int n, int update_ctx)
{
int dst_index;
int index= c->index;
int frac= c->frac;
int dst_incr_frac= c->dst_incr % c->src_incr;
int dst_incr= c->dst_incr / c->src_incr;
int sample_index = index >> c->phase_shift;
av_assert1(c->filter_shift == FILTER_SHIFT);
av_assert1(c->felem_size == sizeof(FELEM));
if (c->filter_length == 1 && c->phase_shift == 0) {
int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*index;
int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
int new_size = (src_size * (int64_t)c->src_incr - frac + c->dst_incr - 1) / c->dst_incr;
dst_size= FFMIN(dst_size, new_size);
for(dst_index=0; dst_index < dst_size; dst_index++){
dst[dst_index] = src[index2>>32];
index2 += incr;
}
index += dst_index * dst_incr;
index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
av_assert2(index >= 0);
*consumed= index;
index = 0;
} else {
int64_t end_index = (1LL + src_size - c->filter_length) << c->phase_shift;
int64_t delta_frac = (end_index - index) * c->src_incr - c->frac;
int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr;
int n = FFMIN(dst_size, delta_n);
int sample_index;
if (!c->linear) {
sample_index = index >> c->phase_shift;
index &= c->phase_mask;
for (dst_index = 0; dst_index < n; dst_index++) {
FELEM *filter = ((FELEM *) c->filter_bank) + c->filter_alloc * index;
@ -153,6 +138,7 @@ int RENAME(swri_resample)(ResampleContext *c, DELEM *dst, const DELEM *src, int
COMMON_CORE
#else
FELEM2 val=0;
int i;
for (i = 0; i < c->filter_length; i++) {
val += src[sample_index + i] * (FELEM2)filter[i];
}
@ -168,8 +154,26 @@ int RENAME(swri_resample)(ResampleContext *c, DELEM *dst, const DELEM *src, int
sample_index += index >> c->phase_shift;
index &= c->phase_mask;
}
} else {
sample_index = index >> c->phase_shift;
if(update_ctx){
c->frac= frac;
c->index= index;
}
return sample_index;
}
static int RENAME(resample_linear)(ResampleContext *c,
DELEM *dst, const DELEM *src,
int n, int update_ctx)
{
int dst_index;
int index= c->index;
int frac= c->frac;
int dst_incr_frac= c->dst_incr % c->src_incr;
int dst_incr= c->dst_incr / c->src_incr;
int sample_index = index >> c->phase_shift;
index &= c->phase_mask;
for (dst_index = 0; dst_index < n; dst_index++) {
FELEM *filter = ((FELEM *) c->filter_bank) + c->filter_alloc * index;
@ -178,6 +182,7 @@ int RENAME(swri_resample)(ResampleContext *c, DELEM *dst, const DELEM *src, int
#ifdef LINEAR_CORE
LINEAR_CORE
#else
int i;
for (i = 0; i < c->filter_length; i++) {
val += src[sample_index + i] * (FELEM2)filter[i];
v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_alloc];
@ -195,17 +200,13 @@ int RENAME(swri_resample)(ResampleContext *c, DELEM *dst, const DELEM *src, int
sample_index += index >> c->phase_shift;
index &= c->phase_mask;
}
}
*consumed = sample_index;
}
if(update_ctx){
c->frac= frac;
c->index= index;
}
return dst_index;
return sample_index;
}
#undef COMMON_CORE

@ -2,4 +2,6 @@ YASM-OBJS += x86/swresample_x86.o\
x86/audio_convert.o\
x86/rematrix.o\
OBJS += x86/resample_x86_dsp.o\
OBJS-$(CONFIG_XMM_CLOBBER_TEST) += x86/w64xmmtest.o

@ -0,0 +1,89 @@
/*
* audio resampling
* Copyright (c) 2004-2012 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
*/
/**
* @file
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "libswresample/resample.h"
#if HAVE_MMXEXT_INLINE
#define DO_RESAMPLE_ONE 0
#include "resample_mmx.h"
#define TEMPLATE_RESAMPLE_S16_MMX2
#include "libswresample/resample_template.c"
#undef TEMPLATE_RESAMPLE_S16_MMX2
#if HAVE_SSE_INLINE
#define TEMPLATE_RESAMPLE_FLT_SSE
#include "libswresample/resample_template.c"
#undef TEMPLATE_RESAMPLE_FLT_SSE
#endif
#if HAVE_SSE2_INLINE
#define TEMPLATE_RESAMPLE_S16_SSE2
#include "libswresample/resample_template.c"
#undef TEMPLATE_RESAMPLE_S16_SSE2
#define TEMPLATE_RESAMPLE_DBL_SSE2
#include "libswresample/resample_template.c"
#undef TEMPLATE_RESAMPLE_DBL_SSE2
#endif
#if HAVE_AVX_INLINE
#define TEMPLATE_RESAMPLE_FLT_AVX
#include "libswresample/resample_template.c"
#undef TEMPLATE_RESAMPLE_FLT_AVX
#endif
#undef DO_RESAMPLE_ONE
#endif // HAVE_MMXEXT_INLINE
void swresample_dsp_x86_init(ResampleContext *c)
{
int av_unused mm_flags = av_get_cpu_flags();
#define FNIDX(fmt) (AV_SAMPLE_FMT_##fmt - AV_SAMPLE_FMT_S16P)
if (ARCH_X86_32 && HAVE_MMXEXT_INLINE && mm_flags & AV_CPU_FLAG_MMX2) {
c->dsp.resample_common[FNIDX(S16P)] = (resample_fn) resample_common_int16_mmx2;
c->dsp.resample_linear[FNIDX(S16P)] = (resample_fn) resample_linear_int16_mmx2;
}
if (HAVE_SSE_INLINE && mm_flags & AV_CPU_FLAG_SSE) {
c->dsp.resample_common[FNIDX(FLTP)] = (resample_fn) resample_common_float_sse;
c->dsp.resample_linear[FNIDX(FLTP)] = (resample_fn) resample_linear_float_sse;
}
if (HAVE_SSE2_INLINE && mm_flags & AV_CPU_FLAG_SSE2) {
c->dsp.resample_common[FNIDX(S16P)] = (resample_fn) resample_common_int16_sse2;
c->dsp.resample_linear[FNIDX(S16P)] = (resample_fn) resample_linear_int16_sse2;
c->dsp.resample_common[FNIDX(DBLP)] = (resample_fn) resample_common_double_sse2;
c->dsp.resample_linear[FNIDX(DBLP)] = (resample_fn) resample_linear_double_sse2;
}
if (HAVE_AVX_INLINE && mm_flags & AV_CPU_FLAG_AVX) {
c->dsp.resample_common[FNIDX(FLTP)] = (resample_fn) resample_common_float_avx;
c->dsp.resample_linear[FNIDX(FLTP)] = (resample_fn) resample_linear_float_avx;
}
}
Loading…
Cancel
Save