af: replace macros with too generic names

Defining names like min, max etc. in an often used header is not really
a good idea.

Somewhat similar to MPlayer svn commit 36491, but don't use libavutil,
because that typically causes us sorrow.

11 files changed, 33 insertions, 43 deletions

diff --git a/audio/filter/af.h b/audio/filter/af.h
index 088e02dc6a..dc776460aa 100644
--- a/audio/filter/af.h
+++ b/audio/filter/af.h
@@ -263,25 +263,4 @@ float af_softclip(float a);
 #define RESIZE_LOCAL_BUFFER(a, d) \
     ((a->data->len < af_lencalc(a->mul, d)) ? af_resize_local_buffer(a, d) : AF_OK)
 
-/* Some other useful macro definitions*/
-#ifndef min
-#define min(a, b)(((a) > (b)) ? (b) : (a))
-#endif
-
-#ifndef max
-#define max(a, b)(((a) > (b)) ? (a) : (b))
-#endif
-
-#ifndef clamp
-#define clamp(a, min, max) (((a) > (max)) ? (max) : (((a) < (min)) ? (min) : (a)))
-#endif
-
-#ifndef sign
-#define sign(a) (((a) > 0) ? (1) : (-1))
-#endif
-
-#ifndef lrnd
-#define lrnd(a, b) ((b)((a) >= 0.0 ? (a) + 0.5 : (a) - 0.5))
-#endif
-
 #endif /* MPLAYER_AF_H */
diff --git a/audio/filter/af_center.c b/audio/filter/af_center.c
index e04b5ceced..0cfdbc3b0e 100644
--- a/audio/filter/af_center.c
+++ b/audio/filter/af_center.c
@@ -29,6 +29,7 @@
 #include <stdlib.h>
 #include <string.h>
 
+#include "mpvcore/mp_common.h"
 #include "af.h"
 
 // Data for specific instances of this filter
@@ -48,7 +49,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     if(!arg) return AF_ERROR;
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
-    mp_audio_set_channels_old(af->data, max(s->ch+1,((struct mp_audio*)arg)->nch));
+    mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch));
     mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 
     return af_test_output(af,(struct mp_audio*)arg);
diff --git a/audio/filter/af_channels.c b/audio/filter/af_channels.c
index fc1a090f7f..27445aafe2 100644
--- a/audio/filter/af_channels.c
+++ b/audio/filter/af_channels.c
@@ -27,6 +27,7 @@
 #include <string.h>
 #include <inttypes.h>
 
+#include "mpvcore/mp_common.h"
 #include "af.h"
 
 #define FR 0
@@ -150,14 +151,14 @@ static int control(struct af_instance* af, int cmd, void* arg)
 
       // If mono: fake stereo
       if(((struct mp_audio*)arg)->nch == 1){
-	s->nr = min(af->data->nch,2);
+	s->nr = MPMIN(af->data->nch,2);
 	for(i=0;i<s->nr;i++){
 	  s->route[i][FR] = 0;
 	  s->route[i][TO] = i;
 	}
       }
       else{
-	s->nr = min(af->data->nch, ((struct mp_audio*)arg)->nch);
+	s->nr = MPMIN(af->data->nch, ((struct mp_audio*)arg)->nch);
 	for(i=0;i<s->nr;i++){
 	  s->route[i][FR] = i;
 	  s->route[i][TO] = i;
diff --git a/audio/filter/af_delay.c b/audio/filter/af_delay.c
index 5c34e33a92..a6515f84cf 100644
--- a/audio/filter/af_delay.c
+++ b/audio/filter/af_delay.c
@@ -27,6 +27,7 @@
 #include <string.h>
 #include <inttypes.h>
 
+#include "mpvcore/mp_common.h"
 #include "af.h"
 
 #define L 65536
@@ -85,7 +86,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     s->ri = 0;
     for(i=0;i<AF_NCH;i++){
       mp_msg(MSGT_AFILTER, MSGL_DBG2, "[delay] Channel %i delayed by %0.3fms\n",
-	     i,clamp(s->d[i],0.0,1000.0));
+	     i,MPCLAMP(s->d[i],0.0,1000.0));
       mp_msg(MSGT_AFILTER, MSGL_DBG3, "[delay] Channel %i delayed by %i samples\n",
 	     i,s->wi[i]);
     }
diff --git a/audio/filter/af_drc.c b/audio/filter/af_drc.c
index c0f7b5beca..9bbbde4831 100644
--- a/audio/filter/af_drc.c
+++ b/audio/filter/af_drc.c
@@ -26,6 +26,7 @@
 #include <math.h>
 #include <limits.h>
 
+#include "mpvcore/mp_common.h"
 #include "af.h"
 
 // Methods:
@@ -139,14 +140,14 @@ static void method1_int16(af_drc_t *s, struct mp_audio *c)
     s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
 
     // clamp the mul coefficient
-    s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
+    s->mul = MPCLAMP(s->mul, MUL_MIN, MUL_MAX);
   }
 
   // Scale & clamp the samples
   for (i = 0; i < len; i++)
   {
     tmp = s->mul * data[i];
-    tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
+    tmp = MPCLAMP(tmp, SHRT_MIN, SHRT_MAX);
     data[i] = tmp;
   }
 
@@ -180,7 +181,7 @@ static void method1_float(af_drc_t *s, struct mp_audio *c)
     s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
 
     // clamp the mul coefficient
-    s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
+    s->mul = MPCLAMP(s->mul, MUL_MIN, MUL_MAX);
   }
 
   // Scale & clamp the samples
@@ -223,7 +224,7 @@ static void method2_int16(af_drc_t *s, struct mp_audio *c)
     if (avg >= SIL_S16)
     {
 	s->mul = s->mid_s16 / avg;
-	s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
+	s->mul = MPCLAMP(s->mul, MUL_MIN, MUL_MAX);
     }
   }
 
@@ -231,7 +232,7 @@ static void method2_int16(af_drc_t *s, struct mp_audio *c)
   for (i = 0; i < len; i++)
   {
     tmp = s->mul * data[i];
-    tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
+    tmp = MPCLAMP(tmp, SHRT_MIN, SHRT_MAX);
     data[i] = tmp;
   }
 
@@ -273,7 +274,7 @@ static void method2_float(af_drc_t *s, struct mp_audio *c)
     if (avg >= SIL_FLOAT)
     {
 	s->mul = s->mid_float / avg;
-	s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
+	s->mul = MPCLAMP(s->mul, MUL_MIN, MUL_MAX);
     }
   }
 
diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c
index 8bc1507584..cbdcd3f84a 100644
--- a/audio/filter/af_equalizer.c
+++ b/audio/filter/af_equalizer.c
@@ -29,6 +29,7 @@
 #include <inttypes.h>
 #include <math.h>
 
+#include "mpvcore/mp_common.h"
 #include "af.h"
 
 #define L   	2      // Storage for filter taps
@@ -143,7 +144,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     for(i=0;i<AF_NCH;i++){
       for(j=0;j<KM;j++){
 	((af_equalizer_t*)af->setup)->g[i][j] =
-	  pow(10.0,clamp(g[j],G_MIN,G_MAX)/20.0)-1.0;
+	  pow(10.0,MPCLAMP(g[j],G_MIN,G_MAX)/20.0)-1.0;
       }
     }
     return AF_OK;
diff --git a/audio/filter/af_extrastereo.c b/audio/filter/af_extrastereo.c
index 75570c7648..4cf27f2724 100644
--- a/audio/filter/af_extrastereo.c
+++ b/audio/filter/af_extrastereo.c
@@ -26,6 +26,7 @@
 #include <math.h>
 #include <limits.h>
 
+#include "mpvcore/mp_common.h"
 #include "af.h"
 
 // Data for specific instances of this filter
@@ -93,8 +94,8 @@ static struct mp_audio* play_s16(struct af_instance* af, struct mp_audio* data)
     l = avg + (int)(s->mul * (a[i] - avg));
     r = avg + (int)(s->mul * (a[i + 1] - avg));
 
-    a[i] = clamp(l, SHRT_MIN, SHRT_MAX);
-    a[i + 1] = clamp(r, SHRT_MIN, SHRT_MAX);
+    a[i] = MPCLAMP(l, SHRT_MIN, SHRT_MAX);
+    a[i + 1] = MPCLAMP(r, SHRT_MIN, SHRT_MAX);
   }
 
   return data;
diff --git a/audio/filter/af_pan.c b/audio/filter/af_pan.c
index bbd754647a..b3e31dab98 100644
--- a/audio/filter/af_pan.c
+++ b/audio/filter/af_pan.c
@@ -25,6 +25,7 @@
 #include <math.h>
 #include <limits.h>
 
+#include "mpvcore/mp_common.h"
 #include "af.h"
 
 // Data for specific instances of this filter
@@ -126,10 +127,10 @@ static int control(struct af_instance* af, int cmd, void* arg)
     if (s->nch)
       return AF_ERROR;
     if (af->data->nch >= 2) {
-      s->level[0][0] = min(1.f, 1.f - val);
-      s->level[0][1] = max(0.f, val);
-      s->level[1][0] = max(0.f, -val);
-      s->level[1][1] = min(1.f, 1.f + val);
+      s->level[0][0] = MPMIN(1.f, 1.f - val);
+      s->level[0][1] = MPMAX(0.f, val);
+      s->level[1][0] = MPMAX(0.f, -val);
+      s->level[1][1] = MPMIN(1.f, 1.f + val);
     }
     return AF_OK;
   }
diff --git a/audio/filter/af_sub.c b/audio/filter/af_sub.c
index 6fd492db95..7778e80b64 100644
--- a/audio/filter/af_sub.c
+++ b/audio/filter/af_sub.c
@@ -32,6 +32,7 @@
 #include <stdlib.h>
 #include <string.h>
 
+#include "mpvcore/mp_common.h"
 #include "af.h"
 #include "dsp.h"
 
@@ -70,7 +71,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     if(!arg) return AF_ERROR;
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
-    mp_audio_set_channels_old(af->data, max(s->ch+1,((struct mp_audio*)arg)->nch));
+    mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch));
     mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 
     // Design low-pass filter
diff --git a/audio/filter/af_tools.c b/audio/filter/af_tools.c
index 77fdad55f2..77638ed9d6 100644
--- a/audio/filter/af_tools.c
+++ b/audio/filter/af_tools.c
@@ -18,6 +18,8 @@
 
 #include <math.h>
 #include <string.h>
+
+#include "mpvcore/mp_common.h"
 #include "af.h"
 
 /* Convert to gain value from dB. Returns AF_OK if of and AF_ERROR if
@@ -33,7 +35,7 @@ int af_from_dB(int n, float* in, float* out, float k, float mi, float ma)
     if(in[i]<=-200)
       out[i]=0.0;
     else
-      out[i]=pow(10.0,clamp(in[i],mi,ma)/k);
+      out[i]=pow(10.0,MPCLAMP(in[i],mi,ma)/k);
   }
   return AF_OK;
 }
@@ -65,7 +67,7 @@ int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
     return AF_ERROR;
 
   for(i=0;i<n;i++)
-    out[i]=(int)((float)rate * clamp(in[i],mi,ma)/1000.0);
+    out[i]=(int)((float)rate * MPCLAMP(in[i],mi,ma)/1000.0);
 
   return AF_OK;
 }
diff --git a/audio/filter/af_volume.c b/audio/filter/af_volume.c
index 08c7789374..fa505a3dab 100644
--- a/audio/filter/af_volume.c
+++ b/audio/filter/af_volume.c
@@ -26,6 +26,7 @@
 #include <math.h>
 #include <limits.h>
 
+#include "mpvcore/mp_common.h"
 #include "af.h"
 
 struct priv {
@@ -73,7 +74,7 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data)
             if (vol != 256) {
                 for (int i = ch; i < len; i += nch) {
                     int x = (a[i] * vol) >> 8;
-                    a[i] = clamp(x, SHRT_MIN, SHRT_MAX);
+                    a[i] = MPCLAMP(x, SHRT_MIN, SHRT_MAX);
                 }
             }
         }
@@ -85,7 +86,7 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data)
                 for (int i = ch; i < len; i += nch) {
                     float x = a[i];
                     x *= s->level[ch];
-                    a[i] = s->soft ? af_softclip(x) : clamp(x, -1.0, 1.0);
+                    a[i] = s->soft ? af_softclip(x) : MPCLAMP(x, -1.0, 1.0);
                 }
             }
         }