whitespace cosmetics: Remove all trailing whitespace.

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@29305 b3059339-0415-0410-9bf9-f77b7e298cf2

12 files changed, 110 insertions, 110 deletions

diff --git a/mp3lib/dct12.c b/mp3lib/dct12.c
index 61f2b29900..5ba45af389 100644
--- a/mp3lib/dct12.c
+++ b/mp3lib/dct12.c
@@ -39,7 +39,7 @@ static void dct12(real *in,real *rawout1,real *rawout2,register real *wi,registe
      register real *out1 = rawout1;
      ts[SBLIMIT*0] = out1[0]; ts[SBLIMIT*1] = out1[1]; ts[SBLIMIT*2] = out1[2];
      ts[SBLIMIT*3] = out1[3]; ts[SBLIMIT*4] = out1[4]; ts[SBLIMIT*5] = out1[5];
- 
+
      DCT12_PART1
 
      {
@@ -73,7 +73,7 @@ static void dct12(real *in,real *rawout1,real *rawout2,register real *wi,registe
   {
      real in0,in1,in2,in3,in4,in5;
      register real *out2 = rawout2;
- 
+
      DCT12_PART1
 
      {
@@ -102,7 +102,7 @@ static void dct12(real *in,real *rawout1,real *rawout2,register real *wi,registe
      ts[(17-2)*SBLIMIT] += in4 * wi[5-2];
   }
 
-  in++; 
+  in++;
 
   {
      real in0,in1,in2,in3,in4,in5;
diff --git a/mp3lib/dct36.c b/mp3lib/dct36.c
index bc9ea25a93..753c2f25d5 100644
--- a/mp3lib/dct36.c
+++ b/mp3lib/dct36.c
@@ -4,7 +4,7 @@
  * $Id$
  */
 
-/* 
+/*
 // This is an optimized DCT from Jeff Tsay's maplay 1.2+ package.
 // Saved one multiplication by doing the 'twiddle factor' stuff
 // together with the window mul. (MH)
@@ -187,7 +187,7 @@ static void dct36(real *inbuf,real *o1,real *o2,real *wintab,real *tsbuf)
     out2[8-(v)] = tmp * w[26-(v)];  } \
     sum0 -= sum1; \
     ts[SBLIMIT*(8-(v))] = out1[8-(v)] + sum0 * w[8-(v)]; \
-    ts[SBLIMIT*(9+(v))] = out1[9+(v)] + sum0 * w[9+(v)]; 
+    ts[SBLIMIT*(9+(v))] = out1[9+(v)] + sum0 * w[9+(v)];
 #define MACRO1(v) { \
 	real sum0,sum1; \
     sum0 = tmp1a + tmp2a; \
@@ -212,7 +212,7 @@ static void dct36(real *inbuf,real *o1,real *o2,real *wintab,real *tsbuf)
     tb33 = in[2*3+1] * c[3];
     tb66 = in[2*6+1] * c[6];
 
-    { 
+    {
       real tmp1a,tmp2a,tmp1b,tmp2b;
       tmp1a =             in[2*1+0] * c[1] + ta33 + in[2*5+0] * c[5] + in[2*7+0] * c[7];
       tmp1b =             in[2*1+1] * c[1] + tb33 + in[2*5+1] * c[5] + in[2*7+1] * c[7];
diff --git a/mp3lib/dct64_altivec.c b/mp3lib/dct64_altivec.c
index 3f3eb0fedf..21a7b88699 100644
--- a/mp3lib/dct64_altivec.c
+++ b/mp3lib/dct64_altivec.c
@@ -46,7 +46,7 @@ void dct64_altivec(real *a,real *b,real *c)
 {
   real __attribute__ ((aligned(16))) b1[0x20];
   real __attribute__ ((aligned(16))) b2[0x20];
-  
+
   real *out0 = a;
   real *out1 = b;
   real *samples = c;
@@ -57,7 +57,7 @@ void dct64_altivec(real *a,real *b,real *c)
 
   if (((unsigned long)b1 & 0x0000000F) ||
       ((unsigned long)b2 & 0x0000000F))
-    
+
   {
     printf("MISALIGNED:\t%p\t%p\t%p\t%p\t%p\n",
            b1, b2, a, b, samples);
@@ -65,7 +65,7 @@ void dct64_altivec(real *a,real *b,real *c)
 
 
 #ifdef ALTIVEC_USE_REFERENCE_C_CODE
-  
+
   {
     register real *costab = mp3lib_pnts[0];
 
@@ -249,7 +249,7 @@ void dct64_altivec(real *a,real *b,real *c)
       costabv3 = vec_perm(costabv3, costabv4, costab_perm);
       costabv5 = vec_ld(64, costab);
       costabv4 = vec_perm(costabv4, costabv5, costab_perm);
-    
+
       temp1 = vec_sub(vec_perm(samplesv4, samplesv4, reverse),
                       samplesv5);
       temp2 = vec_madd(temp1,
@@ -257,7 +257,7 @@ void dct64_altivec(real *a,real *b,real *c)
                        vczero);
       //vec_st(temp2, 64, b1);
       b1v4 = temp2;
-    
+
       temp1 = vec_sub(vec_perm(samplesv3, samplesv3, reverse),
                       samplesv6);
       temp2 = vec_madd(temp1,
@@ -272,7 +272,7 @@ void dct64_altivec(real *a,real *b,real *c)
                        vczero);
       //vec_st(temp2, 96, b1);
       b1v6 = temp2;
-    
+
       temp1 = vec_sub(vec_perm(samplesv1, samplesv1, reverse),
                       samplesv8);
       temp2 = vec_madd(temp1,
@@ -299,7 +299,7 @@ void dct64_altivec(real *a,real *b,real *c)
         costabv2 = vec_perm(costabv2, costabv3 , costab_perm);
         costabv1r = vec_perm(costabv1, costabv1, reverse);
         costabv2r = vec_perm(costabv2, costabv2, reverse);
-    
+
         temp1 = vec_add(b1v0, vec_perm(b1v3, b1v3, reverse));
         //vec_st(temp1, 0, b2);
         b2v0 = temp1;
@@ -333,7 +333,7 @@ void dct64_altivec(real *a,real *b,real *c)
       {
         register real *costab = mp3lib_pnts[2];
 
-    
+
         vector float costabv1r, costabv1, costabv2;
         vector unsigned char costab_perm = vec_lvsl(0, costab);
 
@@ -341,13 +341,13 @@ void dct64_altivec(real *a,real *b,real *c)
         costabv2 = vec_ld(16, costab);
         costabv1 = vec_perm(costabv1, costabv2, costab_perm);
         costabv1r = vec_perm(costabv1, costabv1, reverse);
-    
+
         temp1 = vec_add(b2v0, vec_perm(b2v1, b2v1, reverse));
         vec_st(temp1, 0, b1);
         temp2 = vec_sub(vec_perm(b2v0, b2v0, reverse), b2v1);
         temp1 = vec_madd(temp2, costabv1r, vczero);
         vec_st(temp1, 16, b1);
-    
+
         temp1 = vec_add(b2v2, vec_perm(b2v3, b2v3, reverse));
         vec_st(temp1, 32, b1);
         temp2 = vec_sub(b2v3, vec_perm(b2v2, b2v2, reverse));
@@ -365,7 +365,7 @@ void dct64_altivec(real *a,real *b,real *c)
         temp2 = vec_sub(b2v7, vec_perm(b2v6, b2v6, reverse));
         temp1 = vec_madd(temp2, costabv1r, vczero);
         vec_st(temp1, 112, b1);
-    
+
       }
     }
   }
diff --git a/mp3lib/decod386.c b/mp3lib/decod386.c
index 841cf6981d..4b6d69bc4d 100644
--- a/mp3lib/decod386.c
+++ b/mp3lib/decod386.c
@@ -107,7 +107,7 @@ static synth_func_t synth_func;
 #else /* HAVE_ALTIVEC */
 #define dct64_base(a,b,c) dct64(a,b,c)
 #endif /* HAVE_ALTIVEC */
- 
+
 static int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt)
 {
   static real buffs[2][2][0x110];
diff --git a/mp3lib/huffman.h b/mp3lib/huffman.h
index 7fec0d589e..aa9e7b91fa 100644
--- a/mp3lib/huffman.h
+++ b/mp3lib/huffman.h
@@ -1,19 +1,19 @@
 /*
  * huffman tables ... recalcualted to work with my optimzed
  * decoder scheme (MH)
- * 
- * probably we could save a few bytes of memory, because the 
+ *
+ * probably we could save a few bytes of memory, because the
  * smaller tables are often the part of a bigger table
  */
 
-struct newhuff 
+struct newhuff
 {
   unsigned int linbits;
   short *table;
 };
 
-static short tab0[] = 
-{ 
+static short tab0[] =
+{
    0
 };
 
@@ -286,7 +286,7 @@ static short tab_c1[] =
 
 
 
-static struct newhuff ht[] = 
+static struct newhuff ht[] =
 {
  { /* 0 */ 0 , tab0  } ,
  { /* 2 */ 0 , tab1  } ,
@@ -323,7 +323,7 @@ static struct newhuff ht[] =
  { /* 16 */ 13, tab24 }
 };
 
-static struct newhuff htc[] = 
+static struct newhuff htc[] =
 {
  { /* 1 , 1 , */ 0 , tab_c0 } ,
  { /* 1 , 1 , */ 0 , tab_c1 }
diff --git a/mp3lib/l2tables.h b/mp3lib/l2tables.h
index 662241742d..f62a546088 100644
--- a/mp3lib/l2tables.h
+++ b/mp3lib/l2tables.h
@@ -5,9 +5,9 @@
  */
 
 /*
- * Layer 2 Alloc tables .. 
+ * Layer 2 Alloc tables ..
  * most other tables are calculated on program start (which is (of course)
- * not ISO-conform) .. 
+ * not ISO-conform) ..
  * Layer-3 huffman table is in huffman.h
  */
 
diff --git a/mp3lib/layer1.c b/mp3lib/layer1.c
index 444dab9a4d..9a50b5df47 100644
--- a/mp3lib/layer1.c
+++ b/mp3lib/layer1.c
@@ -1,10 +1,10 @@
-/* 
- * Mpeg Layer-1 audio decoder 
+/*
+ * Mpeg Layer-1 audio decoder
  * --------------------------
  * copyright (c) 1995 by Michael Hipp, All rights reserved. See also 'README'
  * near unoptimzed ...
  *
- * may have a few bugs after last optimization ... 
+ * may have a few bugs after last optimization ...
  *
  */
 
@@ -28,7 +28,7 @@ static void I_step_one(unsigned int balloc[], unsigned int scale_index[2][SBLIMI
   if(fr->stereo == 2) {
     int i;
     int jsbound = fr->jsbound;
-    for (i=0;i<jsbound;i++) { 
+    for (i=0;i<jsbound;i++) {
       *ba++ = getbits(4);
       *ba++ = getbits(4);
     }
@@ -80,7 +80,7 @@ static void I_step_two(real fraction[2][SBLIMIT],unsigned int balloc[2*SBLIMIT],
       if ((n = *ba++))
         *sample++ = getbits(n+1);
     }
-    for (i=jsbound;i<SBLIMIT;i++) 
+    for (i=jsbound;i<SBLIMIT;i++)
       if ((n = *ba++))
         *sample++ = getbits(n+1);
 
@@ -137,7 +137,7 @@ static int do_layer1(struct frame *fr,int single)
 //  printf("do_layer1(0x%02X 0x%02X 0x%02X 0x%02X 0x%02X 0x%02X 0x%02X 0x%02X )\n",
 //    wordpointer[0],wordpointer[1],wordpointer[2],wordpointer[3],wordpointer[4],wordpointer[5],wordpointer[6],wordpointer[7]);
 
-  fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ? 
+  fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ?
                          (fr->mode_ext<<2)+4 : 32;
 
   if(stereo == 1 || single == 3)
diff --git a/mp3lib/layer2.c b/mp3lib/layer2.c
index a1d3fa43fe..f2c134c827 100644
--- a/mp3lib/layer2.c
+++ b/mp3lib/layer2.c
@@ -4,8 +4,8 @@
  * $Id$
  */
 
-/* 
- * Mpeg Layer-2 audio decoder 
+/*
+ * Mpeg Layer-2 audio decoder
  * --------------------------
  * copyright (c) 1995 by Michael Hipp, All rights reserved. See also 'README'
  *
@@ -56,7 +56,7 @@ static void init_layer2(void)
   {
     double m=mulmul[k];
     table = muls[k];
-    if(_has_mmx) 
+    if(_has_mmx)
     {
         for(j=3,i=0;i<63;i++,j--)
 	  *table++ = 16384 * m * pow(2.0,(double) j / 3.0);
@@ -116,21 +116,21 @@ static void II_step_one(unsigned int *bit_alloc,int *scale,struct frame *fr)
 
     bita = bit_alloc;
     scfsi=scfsi_buf;
-    for (i=sblimit2;i>0;i--) 
+    for (i=sblimit2;i>0;i--)
       if (*bita++)
-        switch (*scfsi++) 
+        switch (*scfsi++)
         {
-          case 0: 
+          case 0:
                 *scale++ = getbits_fast(6);
                 *scale++ = getbits_fast(6);
                 *scale++ = getbits_fast(6);
                 break;
-          case 1 : 
+          case 1 :
                 *scale++ = sc = getbits_fast(6);
                 *scale++ = sc;
                 *scale++ = getbits_fast(6);
                 break;
-          case 2: 
+          case 2:
                 *scale++ = sc = getbits_fast(6);
                 *scale++ = sc;
                 *scale++ = sc;
@@ -159,17 +159,17 @@ static void II_step_two(unsigned int *bit_alloc,real fraction[2][4][SBLIMIT],int
       step = alloc1->bits;
       for (j=0;j<stereo;j++)
       {
-        if ( (ba=*bita++) ) 
+        if ( (ba=*bita++) )
         {
           k=(alloc2 = alloc1+ba)->bits;
-          if( (d1=alloc2->d) < 0) 
+          if( (d1=alloc2->d) < 0)
           {
             real cm=muls[k][scale[x1]];
             fraction[j][0][i] = ((real) ((int)getbits(k) + d1)) * cm;
             fraction[j][1][i] = ((real) ((int)getbits(k) + d1)) * cm;
             fraction[j][2][i] = ((real) ((int)getbits(k) + d1)) * cm;
-          }        
-          else 
+          }
+          else
           {
             static int *table[] = { 0,0,0,grp_3tab,0,grp_5tab,0,0,0,grp_9tab };
             unsigned int idx,*tab,m=scale[x1];
@@ -177,7 +177,7 @@ static void II_step_two(unsigned int *bit_alloc,real fraction[2][4][SBLIMIT],int
             tab = (unsigned int *) (table[d1] + idx + idx + idx);
             fraction[j][0][i] = muls[*tab++][m];
             fraction[j][1][i] = muls[*tab++][m];
-            fraction[j][2][i] = muls[*tab][m];  
+            fraction[j][2][i] = muls[*tab][m];
           }
           scale+=3;
         }
@@ -220,13 +220,13 @@ static void II_step_two(unsigned int *bit_alloc,real fraction[2][4][SBLIMIT],int
         fraction[0][0][i] = fraction[0][1][i] = fraction[0][2][i] =
         fraction[1][0][i] = fraction[1][1][i] = fraction[1][2][i] = 0.0;
       }
-/* 
+/*
    should we use individual scalefac for channel 2 or
    is the current way the right one , where we just copy channel 1 to
-   channel 2 ?? 
+   channel 2 ??
    The current 'strange' thing is, that we throw away the scalefac
    values for the second channel ...!!
--> changed .. now we use the scalefac values of channel one !! 
+-> changed .. now we use the scalefac values of channel one !!
 */
     }
 
@@ -299,10 +299,10 @@ static int do_layer2(struct frame *fr,int outmode)
 
   II_step_one(bit_alloc, scale, fr);
 
-  for (i=0;i<SCALE_BLOCK;i++) 
+  for (i=0;i<SCALE_BLOCK;i++)
   {
     II_step_two(bit_alloc,fraction,scale,fr,i>>2);
-    for (j=0;j<3;j++) 
+    for (j=0;j<3;j++)
     {
       if(single >= 0)
       {
diff --git a/mp3lib/layer3.c b/mp3lib/layer3.c
index 71dec48663..398e631913 100644
--- a/mp3lib/layer3.c
+++ b/mp3lib/layer3.c
@@ -4,16 +4,16 @@
  * $Id$
  */
 
-/* 
- * Mpeg Layer-3 audio decoder 
+/*
+ * Mpeg Layer-3 audio decoder
  * --------------------------
  * copyright (c) 1995-1999 by Michael Hipp.
  * All rights reserved. See also 'README'
  *
- * Optimize-TODO: put short bands into the band-field without the stride 
+ * Optimize-TODO: put short bands into the band-field without the stride
  *                of 3 reals
  * Length-optimze: unify long and short band code where it is possible
- */ 
+ */
 
 #if 0
 #define L3_DEBUG 1
@@ -53,7 +53,7 @@ struct bandInfoStruct {
 static int longLimit[9][23];
 static int shortLimit[9][14];
 
-static const struct bandInfoStruct bandInfo[9] = { 
+static const struct bandInfoStruct bandInfo[9] = {
 
 /* MPEG 1.0 */
  { {0,4,8,12,16,20,24,30,36,44,52,62,74, 90,110,134,162,196,238,288,342,418,576},
@@ -331,11 +331,11 @@ static int III_get_side_info(struct III_sideinfo *si,int stereo,
 
    static const int tabs[2][5] = { { 2,9,5,3,4 } , { 1,8,1,2,9 } };
    const int *tab = tabs[lsf];
-   
+
    si->main_data_begin = getbits(tab[1]);
    if (stereo == 1)
      si->private_bits = getbits_fast(tab[2]);
-   else 
+   else
      si->private_bits = getbits_fast(tab[3]);
 
    if(!lsf) {
@@ -383,8 +383,8 @@ if(2*gr_info->big_values > bandInfo[sfreq].shortIdx[12])
              fprintf(stderr,"Blocktype == 0 and window-switching == 1 not allowed.\n");
            return 0;
          }
-      
-         /* region_count/start parameters are implicit in this case. */       
+
+         /* region_count/start parameters are implicit in this case. */
          if(!lsf || gr_info->block_type == 2)
            gr_info->region1start = 36>>1;
          else {
@@ -474,7 +474,7 @@ static int III_get_scale_factors_1(int *scf,struct gr_info_s *gr_info)
           numbits += num0 * 6;
         }
         else {
-          scf += 6; 
+          scf += 6;
         }
 
         if(!(scfsi & 0x4)) {
@@ -492,7 +492,7 @@ static int III_get_scale_factors_1(int *scf,struct gr_info_s *gr_info)
           numbits += num1 * 5;
         }
         else {
-          scf += 5; 
+          scf += 5;
         }
 
         if(!(scfsi & 0x1)) {
@@ -523,7 +523,7 @@ static int III_get_scale_factors_2(int *scf,struct gr_info_s *gr_info,int i_ster
    { { 9, 9, 9,9 } , { 9, 9,12,6 } , { 18,18,0,0} ,
      {12,12,12,0 } , {12, 9, 9,6 } , { 15,12,9,0} } ,
    { { 6, 9, 9,9 } , { 6, 9,12,6 } , { 15,18,0,0} ,
-     { 6,15,12,0 } , { 6,12, 9,6 } , {  6,18,9,0} } }; 
+     { 6,15,12,0 } , { 6,12, 9,6 } , {  6,18,9,0} } };
 
   if(i_stereo) /* i_stereo AND second channel -> do_layer3() checks this */
     slen = i_slen2[gr_info->scalefac_compress>>1];
@@ -532,7 +532,7 @@ static int III_get_scale_factors_2(int *scf,struct gr_info_s *gr_info,int i_ster
 
   gr_info->preflag = (slen>>15) & 0x1;
 
-  n = 0;  
+  n = 0;
   if( gr_info->block_type == 2 ) {
     n++;
     if(gr_info->mixed_block_flag) n++;
@@ -551,7 +551,7 @@ static int III_get_scale_factors_2(int *scf,struct gr_info_s *gr_info,int i_ster
       for(j=0;j<(int)(pnt[i]);j++) *scf++ = 0;
     }
   }
-  
+
   n = (n << 1) + 1;
   for(i=0;i<n;i++) *scf++ = 0;
 
@@ -596,10 +596,10 @@ static int III_dequantize_sample(real xr[SBLIMIT][SSLIMIT],int *scf,
     int region1  = gr_info->region1start;
     int region2  = gr_info->region2start;
 
-    l3 = ((576>>1)-bv)>>1;   
+    l3 = ((576>>1)-bv)>>1;
 /*
- * we may lose the 'odd' bit here !! 
- * check this later again 
+ * we may lose the 'odd' bit here !!
+ * check this later again
  */
     if(bv <= region1) {
       l[0] = bv; l[1] = l[2] = 0;
@@ -614,10 +614,10 @@ static int III_dequantize_sample(real xr[SBLIMIT][SSLIMIT],int *scf,
       }
     }
   }
- 
+
   if(gr_info->block_type == 2) {
     /*
-     * decoding with short or mixed mode BandIndex table 
+     * decoding with short or mixed mode BandIndex table
      */
     int i,max[4];
     int step=0,lwin=3,cb=0;
@@ -759,7 +759,7 @@ static int III_dequantize_sample(real xr[SBLIMIT][SSLIMIT],int *scf,
           if(part2remain+num <= 0) {
             break;
           }
-          if(mask < 0) 
+          if(mask < 0)
             *xrpnt = -v;
           else
             *xrpnt = v;
@@ -922,7 +922,7 @@ static int III_dequantize_sample(real xr[SBLIMIT][SSLIMIT],int *scf,
             mc = *m++;
             cb = *m++;
 #ifdef CUT_HF
-            if(cb == 21) { 
+            if(cb == 21) {
               fprintf(stderr,"c");
               v = 0.0;
             }
@@ -958,7 +958,7 @@ static int III_dequantize_sample(real xr[SBLIMIT][SSLIMIT],int *scf,
   bitindex -= num; wordpointer += (bitindex>>3); bitindex &= 0x7;
   num = 0;
 
-  while(xrpnt < &xr[SBLIMIT][0]) 
+  while(xrpnt < &xr[SBLIMIT][0])
     *xrpnt++ = 0.0;
 
   while( part2remain > 16 ) {
@@ -977,7 +977,7 @@ static int III_dequantize_sample(real xr[SBLIMIT][SSLIMIT],int *scf,
 
 
 
-/* 
+/*
  * III_stereo: calculate real channel values for Joint-I-Stereo-mode
  */
 static void III_i_stereo(real xr_buf[2][SBLIMIT][SSLIMIT],int *scalefac,
@@ -989,10 +989,10 @@ static void III_i_stereo(real xr_buf[2][SBLIMIT][SSLIMIT],int *scalefac,
       const real *tab1,*tab2;
 
       int tab;
-      static const real *tabs[3][2][2] = { 
+      static const real *tabs[3][2][2] = {
          { { tan1_1,tan2_1 }     , { tan1_2,tan2_2 } },
          { { pow1_1[0],pow2_1[0] } , { pow1_2[0],pow2_2[0] } } ,
-         { { pow1_1[1],pow2_1[1] } , { pow1_2[1],pow2_2[1] } } 
+         { { pow1_1[1],pow2_1[1] } , { pow1_2[1],pow2_2[1] } }
       };
 
       tab = lsf + (gr_info->scalefac_compress & lsf);
@@ -1032,7 +1032,7 @@ static void III_i_stereo(real xr_buf[2][SBLIMIT][SSLIMIT],int *scalefac,
              do_l = 0;
 
            for(;sfb<12;sfb++) {
-             is_p = scalefac[sfb*3+lwin-gr_info->mixed_block_flag]; /* scale: 0-15 */ 
+             is_p = scalefac[sfb*3+lwin-gr_info->mixed_block_flag]; /* scale: 0-15 */
              if(is_p != 7) {
                real t1,t2;
                sb  = bi->shortDiff[sfb];
@@ -1047,7 +1047,7 @@ static void III_i_stereo(real xr_buf[2][SBLIMIT][SSLIMIT],int *scalefac,
            }
 
 #if 1
-/* in the original: copy 10 to 11 , here: copy 11 to 12 
+/* in the original: copy 10 to 11 , here: copy 11 to 12
 maybe still wrong??? (copy 12 to 13?) */
            is_p = scalefac[11*3+lwin-gr_info->mixed_block_flag]; /* scale: 0-15 */
            sb   = bi->shortDiff[12];
@@ -1060,7 +1060,7 @@ maybe still wrong??? (copy 12 to 13?) */
            if(is_p != 7) {
              real t1,t2;
              t1 = tab1[is_p]; t2 = tab2[is_p];
-             for ( ; sb > 0; sb--,idx+=3 ) {  
+             for ( ; sb > 0; sb--,idx+=3 ) {
                real v = xr[0][idx];
                xr[0][idx] = REAL_MUL(v, t1);
                xr[1][idx] = REAL_MUL(v, t2);
@@ -1069,7 +1069,7 @@ maybe still wrong??? (copy 12 to 13?) */
          } /* end for(lwin; .. ; . ) */
 
 /* also check l-part, if ALL bands in the three windows are 'empty'
- * and mode = mixed_mode 
+ * and mode = mixed_mode
  */
          if (do_l) {
            int sfb = gr_info->maxbandl;
@@ -1087,17 +1087,17 @@ maybe still wrong??? (copy 12 to 13?) */
                  xr[1][idx] = REAL_MUL(v, t2);
                }
              }
-             else 
+             else
                idx += sb;
            }
-         }     
-      } 
+         }
+      }
       else { /* ((gr_info->block_type != 2)) */
        int sfb = gr_info->maxbandl;
        int is_p,idx = bi->longIdx[sfb];
 
 /* hmm ... maybe the maxbandl stuff for i-stereo is buggy? */
-       if(sfb <= 21) { 
+       if(sfb <= 21) {
         for ( ; sfb<21; sfb++) {
           int sb = bi->longDiff[sfb];
           is_p = scalefac[sfb]; /* scale: 0-15 */
@@ -1117,7 +1117,7 @@ maybe still wrong??? (copy 12 to 13?) */
         is_p = scalefac[20];
         if(is_p != 7) {  /* copy l-band 20 to l-band 21 */
           int sb;
-          real t1 = tab1[is_p],t2 = tab2[is_p]; 
+          real t1 = tab1[is_p],t2 = tab2[is_p];
 
           for ( sb = bi->longDiff[21]; sb > 0; sb--,idx++ ) {
             real v = xr[0][idx];
@@ -1133,9 +1133,9 @@ static void III_antialias(real xr[SBLIMIT][SSLIMIT],struct gr_info_s *gr_info) {
    int sblim;
 
    if(gr_info->block_type == 2) {
-      if(!gr_info->mixed_block_flag) 
+      if(!gr_info->mixed_block_flag)
         return;
-      sblim = 1; 
+      sblim = 1;
    }
    else {
      sblim = gr_info->maxb-1;
@@ -1159,7 +1159,7 @@ static void III_antialias(real xr[SBLIMIT][SSLIMIT],struct gr_info_s *gr_info) {
          *xr1++ = (bd * (*cs++) ) + (bu * (*ca++) );
        }
      }
-     
+
   }
 }
 
@@ -1172,9 +1172,9 @@ static void III_antialias(real xr[SBLIMIT][SSLIMIT],struct gr_info_s *gr_info) {
 /*
  * III_hybrid
  */
- 
+
 static dct36_func_t dct36_func;
-  
+
 static void III_hybrid(real fsIn[SBLIMIT][SSLIMIT],real tsOut[SSLIMIT][SBLIMIT],
    int ch,struct gr_info_s *gr_info)
 {
@@ -1199,7 +1199,7 @@ static void III_hybrid(real fsIn[SBLIMIT][SSLIMIT],real tsOut[SSLIMIT][SBLIMIT],
      (*dct36_func)(fsIn[1],rawout1+18,rawout2+18,win1[0],tspnt+1);
      rawout1 += 36; rawout2 += 36; tspnt += 2;
    }
- 
+
    bt = gr_info->block_type;
    if(bt == 2) {
      for (; sb<gr_info->maxb; sb+=2,tspnt+=2,rawout1+=36,rawout2+=36) {
@@ -1255,7 +1255,7 @@ static int do_layer3(struct frame *fr,int single){
 
   if(!III_get_side_info(&sideinfo,stereo,ms_stereo,sfreq,single,fr->lsf))
     return -1;
-  
+
   set_pointer(sideinfo.main_data_begin);
 
   granules = (fr->lsf) ? 1 : 2;
@@ -1275,9 +1275,9 @@ static int do_layer3(struct frame *fr,int single){
 
     if(stereo == 2) {
       struct gr_info_s *gr_info = &(sideinfo.ch[1].gr[gr]);
-      
+
       int part2bits;
-      if(fr->lsf) 
+      if(fr->lsf)
         part2bits = III_get_scale_factors_2(scalefacs[1],gr_info,i_stereo);
       else
         part2bits = III_get_scale_factors_1(scalefacs[1],gr_info);
@@ -1302,7 +1302,7 @@ static int do_layer3(struct frame *fr,int single){
         III_i_stereo(hybridIn,scalefacs[1],gr_info,sfreq,ms_stereo,fr->lsf);
 
       if(ms_stereo || i_stereo || (single == 3) ) {
-        if(gr_info->maxb > sideinfo.ch[0].gr[gr].maxb) 
+        if(gr_info->maxb > sideinfo.ch[0].gr[gr].maxb)
           sideinfo.ch[0].gr[gr].maxb = gr_info->maxb;
         else
           gr_info->maxb = sideinfo.ch[0].gr[gr].maxb;
@@ -1313,7 +1313,7 @@ static int do_layer3(struct frame *fr,int single){
           register int i;
           register real *in0 = (real *) hybridIn[0],*in1 = (real *) hybridIn[1];
           for(i=0;i<SSLIMIT*gr_info->maxb;i++,in0++)
-            *in0 = (*in0 + *in1++); /* *0.5 done by pow-scale */ 
+            *in0 = (*in0 + *in1++); /* *0.5 done by pow-scale */
           break; }
         case 1: {
           register int i;
@@ -1340,7 +1340,7 @@ static int do_layer3(struct frame *fr,int single){
 		    clip += (fr->synth)(hybridOut[1][ss],1,pcm_sample,&pcm_point);
       }
     }
-    
+
   }
 
   return clip;
diff --git a/mp3lib/sr1.c b/mp3lib/sr1.c
index 9aa4dec827..224f838c7d 100644
--- a/mp3lib/sr1.c
+++ b/mp3lib/sr1.c
@@ -191,7 +191,7 @@ LOCAL int stream_head_read(unsigned char *hbuf,uint32_t *newhead){
    * we may not be able to address unaligned 32-bit data on non-x86 cpus.
    * Fall back to some portable code.
    */
-  *newhead = 
+  *newhead =
       hbuf[0] << 24 |
       hbuf[1] << 16 |
       hbuf[2] <<  8 |
@@ -215,7 +215,7 @@ LOCAL int stream_head_shift(unsigned char *hbuf,uint32_t *head){
 LOCAL int decode_header(struct frame *fr,uint32_t newhead){
 
     // head_check:
-    if( (newhead & 0xffe00000) != 0xffe00000 ||  
+    if( (newhead & 0xffe00000) != 0xffe00000 ||
         (newhead & 0x0000fc00) == 0x0000fc00) return FALSE;
 
     fr->lay = 4-((newhead>>17)&3);
diff --git a/mp3lib/test.c b/mp3lib/test.c
index ea0169185b..5246ad3ff8 100644
--- a/mp3lib/test.c
+++ b/mp3lib/test.c
@@ -18,7 +18,7 @@ static inline unsigned int GetTimer(void){
   gettimeofday(&tv,&tz);
 //  s=tv.tv_usec;s*=0.000001;s+=tv.tv_sec;
   return (tv.tv_sec*1000000+tv.tv_usec);
-}  
+}
 
 static FILE* mp3file=NULL;
 
@@ -38,10 +38,10 @@ int main(int argc,char* argv[]){
   FILE *f=NULL;
   f=fopen("test.pcm","wb");
 #endif
-  
+
   mp3file=fopen((argc>1)?argv[1]:"test.mp3","rb");
   if(!mp3file){  printf("file not found\n");  exit(1); }
-  
+
   GetCpuCaps(&gCpuCaps);
 
   // MPEG Audio:
@@ -51,7 +51,7 @@ int main(int argc,char* argv[]){
   MP3_Init();
 #endif
   MP3_samplerate=MP3_channels=0;
-  
+
   time1=GetTimer();
   while((len=MP3_DecodeFrame(buffer,-1))>0 && total<2000000){
       total+=len;
@@ -66,7 +66,7 @@ int main(int argc,char* argv[]){
   printf("\nDecoding time: %8.6f\n",(float)time1*0.000001f);
   printf("Uncompressed size: %d bytes  (%8.3f secs)\n",total,length);
   printf("CPU usage at normal playback: %5.2f %%\n",time1*0.0001f/length);
-  
+
   fclose(mp3file);
   return 0;
 }
diff --git a/mp3lib/test2.c b/mp3lib/test2.c
index 0b6a7cbb25..f9bcfc073f 100644
--- a/mp3lib/test2.c
+++ b/mp3lib/test2.c
@@ -26,10 +26,10 @@ int main(int argc,char* argv[]){
   int total=0;
   int r;
   int audio_fd;
-  
+
   mp3file=fopen((argc>1)?argv[1]:"test.mp3","rb");
   if(!mp3file){  printf("file not found\n");  exit(1); }
-  
+
   GetCpuCaps(&gCpuCaps);
 
   // MPEG Audio:
@@ -40,19 +40,19 @@ int main(int argc,char* argv[]){
 #endif
   MP3_samplerate=MP3_channels=0;
   len=MP3_DecodeFrame(buffer,-1);
-  
+
   audio_fd=open("/dev/dsp", O_WRONLY);
   if(audio_fd<0){  printf("Can't open audio device\n");exit(1); }
   r=AFMT_S16_LE;ioctl (audio_fd, SNDCTL_DSP_SETFMT, &r);
   r=MP3_channels-1;ioctl (audio_fd, SNDCTL_DSP_STEREO, &r);
   r=MP3_samplerate;ioctl (audio_fd, SNDCTL_DSP_SPEED, &r);
   printf("audio_setup: using %d Hz samplerate (requested: %d)\n",r,MP3_samplerate);
-  
+
   while(1){
       int len2;
       if(len==0) len=MP3_DecodeFrame(buffer,-1);
       if(len<=0) break; // EOF
-      
+
       // play it
       len2=write(audio_fd,buffer,len);
       if(len2<0) break; // ERROR?
@@ -63,7 +63,7 @@ int main(int argc,char* argv[]){
           putchar('!');fflush(stdout);
       }
   }
-  
+
   fclose(mp3file);
   return 0;
 }