Merge commit '12004a9a'

* commit '12004a9a':
  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 12004a9a?
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 1d36defe
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>

libavcodec/ac3enc_float.c

@@ -121,7 +121,7 @@ static void sum_square_butterfly(AC3EncodeContext *s, float sum[4],
 static void clip_coefficients(AudioDSPContext *adsp, float *coef,
                               unsigned int len)
 {
-    adsp->vector_clipf(coef, coef, COEF_MIN, COEF_MAX, len);
+    adsp->vector_clipf(coef, coef, len, COEF_MIN, COEF_MAX);
 }
 
 

libavcodec/arm/audiodsp_init_neon.c

@@ -25,8 +25,7 @@
 #include "libavcodec/audiodsp.h"
 #include "audiodsp_arm.h"
 
-void ff_vector_clipf_neon(float *dst, const float *src, float min, float max,
-                          int len);
+void ff_vector_clipf_neon(float *dst, const float *src, int len, float min, float max);
 void ff_vector_clip_int32_neon(int32_t *dst, const int32_t *src, int32_t min,
                                int32_t max, unsigned int len);
 

libavcodec/arm/audiodsp_neon.S

@@ -24,9 +24,8 @@
 function ff_vector_clipf_neon, export=1
 VFP     vdup.32         q1,  d0[1]
 VFP     vdup.32         q0,  d0[0]
-NOVFP   vdup.32         q0,  r2
-NOVFP   vdup.32         q1,  r3
-NOVFP   ldr             r2,  [sp]
+NOVFP   vdup.32         q0,  r3
+NOVFP   vld1.32         {d2[],d3[]}, [sp]
         vld1.f32        {q2},[r1,:128]!
         vmin.f32        q10, q2,  q1
         vld1.f32        {q3},[r1,:128]!

libavcodec/audiodsp.c

@@ -55,8 +55,8 @@ static void vector_clipf_c_opposite_sign(float *dst, const float *src,
     }
 }
 
-static void vector_clipf_c(float *dst, const float *src,
-                           float min, float max, int len)
+static void vector_clipf_c(float *dst, const float *src, int len,
+                           float min, float max)
 {
     int i;
 

libavcodec/audiodsp.h

@@ -48,7 +48,8 @@ typedef struct AudioDSPContext {
     /* 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 */,
-                         float min, float max, int len /* align 16 */);
+                         int len /* align 16 */,
+                         float min, float max);
 } AudioDSPContext;
 
 void ff_audiodsp_init(AudioDSPContext *c);

libavcodec/cook.c

@@ -882,7 +882,7 @@ static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
 static void saturate_output_float(COOKContext *q, float *out)
 {
     q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
-                         -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
+                         FFALIGN(q->samples_per_channel, 8), -1.0f, 1.0f);
 }
 
 

libavcodec/x86/audiodsp.asm

@@ -132,46 +132,45 @@ VECTOR_CLIP_INT32 11, 1, 1, 0
 VECTOR_CLIP_INT32 6, 1, 0, 0
 %endif
 
-;-----------------------------------------------------
-;void ff_vector_clipf(float *dst, const float *src,
-;                     float min, float max, int len)
-;-----------------------------------------------------
+; void ff_vector_clipf_sse(float *dst, const float *src,
+;                          int len, float min, float max)
 INIT_XMM sse
-%if UNIX64
-cglobal vector_clipf, 3,3,6, dst, src, len
-%else
-cglobal vector_clipf, 5,5,6, dst, src, min, max, len
-%endif
-%if WIN64
-    SWAP 0, 2
-    SWAP 1, 3
-%elif ARCH_X86_32
-    movss   m0, minm
-    movss   m1, maxm
+cglobal vector_clipf, 3, 3, 6, dst, src, len, min, max
+%if ARCH_X86_32
+    VBROADCASTSS m0, minm
+    VBROADCASTSS m1, maxm
+%elif WIN64
+    VBROADCASTSS m0, m3
+    VBROADCASTSS m1, maxm
+%else ; 64bit sysv
+    VBROADCASTSS m0, m0
+    VBROADCASTSS m1, m1
 %endif
-    SPLATD  m0
-    SPLATD  m1
-        shl lend, 2
-        add srcq, lenq
-        add dstq, lenq
-        neg lenq
-.loop:
-    mova    m2,  [srcq+lenq+mmsize*0]
-    mova    m3,  [srcq+lenq+mmsize*1]
-    mova    m4,  [srcq+lenq+mmsize*2]
-    mova    m5,  [srcq+lenq+mmsize*3]
-    maxps   m2, m0
-    maxps   m3, m0
-    maxps   m4, m0
-    maxps   m5, m0
-    minps   m2, m1
-    minps   m3, m1
-    minps   m4, m1
-    minps   m5, m1
-    mova    [dstq+lenq+mmsize*0], m2
-    mova    [dstq+lenq+mmsize*1], m3
-    mova    [dstq+lenq+mmsize*2], m4
-    mova    [dstq+lenq+mmsize*3], m5
-    add     lenq, mmsize*4
-    jl .loop
-    REP_RET
+
+    movsxdifnidn lenq, lend
+
+.loop
+    mova m2, [srcq + 4 * lenq - 4 * mmsize]
+    mova m3, [srcq + 4 * lenq - 3 * mmsize]
+    mova m4, [srcq + 4 * lenq - 2 * mmsize]
+    mova m5, [srcq + 4 * lenq - 1 * mmsize]
+
+    maxps m2, m0
+    maxps m3, m0
+    maxps m4, m0
+    maxps m5, m0
+
+    minps m2, m1
+    minps m3, m1
+    minps m4, m1
+    minps m5, m1
+
+    mova [dstq + 4 * lenq - 4 * mmsize], m2
+    mova [dstq + 4 * lenq - 3 * mmsize], m3
+    mova [dstq + 4 * lenq - 2 * mmsize], m4
+    mova [dstq + 4 * lenq - 1 * mmsize], m5
+
+    sub lenq, mmsize
+    jg .loop
+
+    RET

libavcodec/x86/audiodsp_init.c

@@ -38,7 +38,7 @@ void ff_vector_clip_int32_int_sse2(int32_t *dst, const int32_t *src,
 void ff_vector_clip_int32_sse4(int32_t *dst, const int32_t *src,
                                int32_t min, int32_t max, unsigned int len);
 void ff_vector_clipf_sse(float *dst, const float *src,
-                         float min, float max, int len);
+                         int len, float min, float max);
 
 av_cold void ff_audiodsp_init_x86(AudioDSPContext *c)
 {

tests/checkasm/audiodsp.c

@@ -120,7 +120,7 @@ void checkasm_check_audiodsp(void)
         int i, len;
 
         declare_func_emms(AV_CPU_FLAG_MMX, void, float *dst, const float *src,
-                          float min, float max, unsigned int len);
+                          int len, float min, float max);
 
         val1 = (float)rnd() / (UINT_MAX >> 1) - 1.0f;
         val2 = (float)rnd() / (UINT_MAX >> 1) - 1.0f;
@@ -133,13 +133,13 @@ void checkasm_check_audiodsp(void)
         len = rnd() % 128;
         len = 16 * FFMAX(len, 1);
 
-        call_ref(dst0, src, min, max, len);
-        call_new(dst1, src, min, max, len);
+        call_ref(dst0, src, len, min, max);
+        call_new(dst1, src, len, min, max);
         for (i = 0; i < len; i++) {
             if (!float_near_ulp_array(dst0, dst1, 3, len))
                 fail();
         }
-        bench_new(dst1, src, min, max, MAX_SIZE);
+        bench_new(dst1, src, MAX_SIZE, min, max);
     }
 
     report("audiodsp");