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>

7 files changed, 325 insertions, 148 deletions

diff --git a/libswresample/Makefile b/libswresample/Makefile
index 953c945..ca59d54 100644
--- a/libswresample/Makefile
+++ b/libswresample/Makefile
@@ -10,6 +10,7 @@ OBJS = audioconvert.o                        \
        dither.o                              \
        rematrix.o                            \
        resample.o                            \
+       resample_dsp.o                        \
        swresample.o                          \
 
 OBJS-$(CONFIG_LIBSOXR) += soxr_resample.o
diff --git a/libswresample/resample.c b/libswresample/resample.c
index 5b7a0f9..9a67e63 100644
--- a/libswresample/resample.c
+++ b/libswresample/resample.c
@@ -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();
diff --git a/libswresample/resample.h b/libswresample/resample.h
new file mode 100644
index 0000000..b159c7b
--- /dev/null
+++ b/libswresample/resample.h
@@ -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 */
diff --git a/libswresample/resample_dsp.c b/libswresample/resample_dsp.c
new file mode 100644
index 0000000..a48a5a9
--- /dev/null
+++ b/libswresample/resample_dsp.c
@@ -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);
+}
diff --git a/libswresample/resample_template.c b/libswresample/resample_template.c
index 40def7a..6773e3e 100644
--- a/libswresample/resample_template.c
+++ b/libswresample/resample_template.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
-    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;
-
-    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;
+
+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;
+
             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;
-    }
+            if(update_ctx){
+                c->frac= frac;
+                c->index= index;
+            }
 
-    return dst_index;
+    return sample_index;
 }
 
 #undef COMMON_CORE
diff --git a/libswresample/x86/Makefile b/libswresample/x86/Makefile
index 1d1ab6e..cc3e65f 100644
--- a/libswresample/x86/Makefile
+++ b/libswresample/x86/Makefile
@@ -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
diff --git a/libswresample/x86/resample_x86_dsp.c b/libswresample/x86/resample_x86_dsp.c
new file mode 100644
index 0000000..02a7db5
--- /dev/null
+++ b/libswresample/x86/resample_x86_dsp.c
@@ -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;
+    }
+}