changeset 24890:a54a25221b79

Remove some pointless 'inline' qualifiers Most of these functions aren't even used in the same translation unit.

--- a/libaf/af.c	Thu Nov 01 06:52:06 2007 +0000
+++ b/libaf/af.c	Thu Nov 01 06:52:10 2007 +0000
@@ -572,7 +572,7 @@
 
 /* Helper function called by the macro with the same name this
    function should not be called directly */
-inline int af_resize_local_buffer(af_instance_t* af, af_data_t* data)
+int af_resize_local_buffer(af_instance_t* af, af_data_t* data)
 {
   // Calculate new length
   register int len = af_lencalc(af->mul,data);

--- a/libaf/af_tools.c	Thu Nov 01 06:52:06 2007 +0000
+++ b/libaf/af_tools.c	Thu Nov 01 06:52:10 2007 +0000
@@ -4,7 +4,7 @@
 
 /* Convert to gain value from dB. Returns AF_OK if of and AF_ERROR if
    fail */
-inline int af_from_dB(int n, float* in, float* out, float k, float mi, float ma)
+int af_from_dB(int n, float* in, float* out, float k, float mi, float ma)
 {
   int i = 0; 
   // Sanity check
@@ -22,7 +22,7 @@
 
 /* Convert from gain value to dB. Returns AF_OK if of and AF_ERROR if
    fail */
-inline int af_to_dB(int n, float* in, float* out, float k)
+int af_to_dB(int n, float* in, float* out, float k)
 {
   int i = 0; 
   // Sanity check
@@ -39,7 +39,7 @@
 }
 
 /* Convert from ms to sample time */
-inline int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
+int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
 {
   int i = 0; 
   // Sanity check
@@ -53,7 +53,7 @@
 }
 
 /* Convert from sample time to ms */
-inline int af_to_ms(int n, int* in, float* out, int rate)
+int af_to_ms(int n, int* in, float* out, int rate)
 {
   int i = 0; 
   // Sanity check
@@ -67,7 +67,7 @@
 }
 
 /* Helper function for testing the output format */
-inline int af_test_output(struct af_instance_s* af, af_data_t* out)
+int af_test_output(struct af_instance_s* af, af_data_t* out)
 {
   if((af->data->format != out->format) || 
      (af->data->bps    != out->bps)    ||
@@ -81,7 +81,7 @@
 
 /* Soft clipping, the sound of a dream, thanks to Jon Wattes
    post to Musicdsp.org */
-inline float af_softclip(float a)
+float af_softclip(float a)
 {
     if (a >= M_PI/2)
 	return 1.0;

--- a/libaf/filter.c	Thu Nov 01 06:52:06 2007 +0000
+++ b/libaf/filter.c	Thu Nov 01 06:52:10 2007 +0000
@@ -46,7 +46,7 @@
    y  output buffer
    s  output buffer stride
 */
-inline _ftype_t* af_filter_pfir(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** w, _ftype_t** x, _ftype_t* y, unsigned int s)
+_ftype_t* af_filter_pfir(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** w, _ftype_t** x, _ftype_t* y, unsigned int s)
 {
   register _ftype_t* xt = *x + xi;
   register _ftype_t* wt = *w;
@@ -65,7 +65,7 @@
    at the new samples, xi current index in xq and n the length of the
    filter. xq must be n*2 by k big, s is the index for in.
 */
-inline int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** xq, _ftype_t* in, unsigned int s)  
+int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** xq, _ftype_t* in, unsigned int s)
 {
   register _ftype_t* txq = *xq + xi;
   register int nt = n*2;

--- a/libaf/format.c	Thu Nov 01 06:52:06 2007 +0000
+++ b/libaf/format.c	Thu Nov 01 06:52:10 2007 +0000
@@ -62,7 +62,7 @@
   return format;
 }
 
-inline int af_fmt2bits(int format)
+int af_fmt2bits(int format)
 {
     return (format & AF_FORMAT_BITS_MASK)+8;
 //    return (((format & AF_FORMAT_BITS_MASK)>>3)+1) * 8;
@@ -79,7 +79,7 @@
     return -1;
 }
 
-inline int af_bits2fmt(int bits)
+int af_bits2fmt(int bits)
 {
     return (bits/8 - 1) << 3;
 }

--- a/libvo/x11_common.c	Thu Nov 01 06:52:06 2007 +0000
+++ b/libvo/x11_common.c	Thu Nov 01 06:52:10 2007 +0000
@@ -2566,7 +2566,7 @@
  * It doesn't call XFlush.
  *
  */
-inline void vo_xv_draw_colorkey(  int32_t x,  int32_t y,
+void vo_xv_draw_colorkey(  int32_t x,  int32_t y,
                                   int32_t w,  int32_t h  )
 {
   if( xv_ck_info.method == CK_METHOD_MANUALFILL ||