Kill all tabs

I hate tabs.

This replaces all tabs in all source files with spaces. The only
exception is old-makefile. The replacement was made by running the
GNU coreutils "expand" command on every file. Since the replacement was
automatic, it's possible that some formatting was destroyed (but perhaps
only if it was assuming that the end of a tab does not correspond to
aligning the end to multiples of 8 spaces).

1 files changed, 36 insertions, 36 deletions

diff --git a/audio/filter/filter.c b/audio/filter/filter.c
index b272125fd8..dbbe743a79 100644
--- a/audio/filter/filter.c
+++ b/audio/filter/filter.c
@@ -58,7 +58,7 @@ inline FLOAT_TYPE af_filter_fir(register unsigned int n, const FLOAT_TYPE* w,
          0 < fc < 1 where 1 <=> Fs/2
    flags window and filter type as defined in filter.h
          variables are ored together: i.e. LP|HAMMING will give a
-	 low pass filter designed using a hamming window
+         low pass filter designed using a hamming window
    opt   beta constant used only when designing using kaiser windows
 
    returns 0 if OK, -1 if fail
@@ -66,16 +66,16 @@ inline FLOAT_TYPE af_filter_fir(register unsigned int n, const FLOAT_TYPE* w,
 int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
                          unsigned int flags, FLOAT_TYPE opt)
 {
-  unsigned int	o   = n & 1;          	// Indicator for odd filter length
-  unsigned int	end = ((n + 1) >> 1) - o;       // Loop end
-  unsigned int	i;			// Loop index
+  unsigned int  o   = n & 1;            // Indicator for odd filter length
+  unsigned int  end = ((n + 1) >> 1) - o;       // Loop end
+  unsigned int  i;                      // Loop index
 
-  FLOAT_TYPE k1 = 2 * M_PI;		// 2*pi*fc1
+  FLOAT_TYPE k1 = 2 * M_PI;             // 2*pi*fc1
   FLOAT_TYPE k2 = 0.5 * (FLOAT_TYPE)(1 - o);// Constant used if the filter has even length
-  FLOAT_TYPE k3;		        // 2*pi*fc2 Constant used in BP and BS design
-  FLOAT_TYPE g  = 0.0;     		// Gain
-  FLOAT_TYPE t1,t2,t3;     		// Temporary variables
-  FLOAT_TYPE fc1,fc2;			// Cutoff frequencies
+  FLOAT_TYPE k3;                        // 2*pi*fc2 Constant used in BP and BS design
+  FLOAT_TYPE g  = 0.0;                  // Gain
+  FLOAT_TYPE t1,t2,t3;                  // Temporary variables
+  FLOAT_TYPE fc1,fc2;                   // Cutoff frequencies
 
   // Sanity check
   if(!w || (n == 0)) return -1;
@@ -113,28 +113,28 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
       // where x is zero. To make sure nothing strange happens, we set this
       // value separately.
       if (o){
-	w[end] = fc1 * w[end] * 2.0;
-	g=w[end];
+        w[end] = fc1 * w[end] * 2.0;
+        g=w[end];
       }
 
       // Create filter
       for (i=0 ; i<end ; i++){
-	t1 = (FLOAT_TYPE)(i+1) - k2;
-	w[end-i-1] = w[n-end+i] = w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
-	g += 2*w[end-i-1]; // Total gain in filter
+        t1 = (FLOAT_TYPE)(i+1) - k2;
+        w[end-i-1] = w[n-end+i] = w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
+        g += 2*w[end-i-1]; // Total gain in filter
       }
     }
     else{ // High pass filter
       if (!o) // High pass filters must have odd length
-	return -1;
+        return -1;
       w[end] = 1.0 - (fc1 * w[end] * 2.0);
       g= w[end];
 
       // Create filter
       for (i=0 ; i<end ; i++){
-	t1 = (FLOAT_TYPE)(i+1);
-	w[end-i-1] = w[n-end+i] = -1 * w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
-	g += ((i&1) ? (2*w[end-i-1]) : (-2*w[end-i-1])); // Total gain in filter
+        t1 = (FLOAT_TYPE)(i+1);
+        w[end-i-1] = w[n-end+i] = -1 * w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
+        g += ((i&1) ? (2*w[end-i-1]) : (-2*w[end-i-1])); // Total gain in filter
       }
     }
   }
@@ -151,32 +151,32 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
     if(flags & BP){ // Band pass
       // Calculate center tap
       if (o){
-	g=w[end]*(fc1+fc2);
-	w[end] = (fc2 - fc1) * w[end] * 2.0;
+        g=w[end]*(fc1+fc2);
+        w[end] = (fc2 - fc1) * w[end] * 2.0;
       }
 
       // Create filter
       for (i=0 ; i<end ; i++){
-	t1 = (FLOAT_TYPE)(i+1) - k2;
-	t2 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
-	t3 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
-	g += w[end-i-1] * (t3 + t2);   // Total gain in filter
-	w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
+        t1 = (FLOAT_TYPE)(i+1) - k2;
+        t2 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
+        t3 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
+        g += w[end-i-1] * (t3 + t2);   // Total gain in filter
+        w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
       }
     }
     else{ // Band stop
       if (!o) // Band stop filters must have odd length
-	return -1;
+        return -1;
       w[end] = 1.0 - (fc2 - fc1) * w[end] * 2.0;
       g= w[end];
 
       // Create filter
       for (i=0 ; i<end ; i++){
-	t1 = (FLOAT_TYPE)(i+1);
-	t2 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
-	t3 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
-	w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
-	g += 2*w[end-i-1]; // Total gain in filter
+        t1 = (FLOAT_TYPE)(i+1);
+        t2 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
+        t3 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
+        w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
+        g += 2*w[end-i-1]; // Total gain in filter
       }
     }
   }
@@ -224,12 +224,12 @@ static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs)
    Arguments:
    a       - s-domain numerator coefficients
    b       - s-domain denominator coefficients
-   k 	   - filter gain factor. Initially set to 1 and modified by each
+   k       - filter gain factor. Initially set to 1 and modified by each
              biquad section in such a way, as to make it the
              coefficient by which to multiply the overall filter gain
              in order to achieve a desired overall filter gain,
              specified in initial value of k.
-   fs 	   - sampling rate (Hz)
+   fs      - sampling rate (Hz)
    coef    - array of z-domain coefficients to be filled in.
 
    Return: On return, set coef z-domain coefficients and k to the gain
@@ -319,13 +319,13 @@ static void af_filter_bilinear(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_T
    Arguments:
    a       - s-domain numerator coefficients, a[1] is always assumed to be 1.0
    b       - s-domain denominator coefficients
-   Q	   - Q value for the filter
-   k 	   - filter gain factor. Initially set to 1 and modified by each
+   Q       - Q value for the filter
+   k       - filter gain factor. Initially set to 1 and modified by each
              biquad section in such a way, as to make it the
              coefficient by which to multiply the overall filter gain
              in order to achieve a desired overall filter gain,
              specified in initial value of k.
-   fs 	   - sampling rate (Hz)
+   fs      - sampling rate (Hz)
    coef    - array of z-domain coefficients to be filled in.
 
    Note: Upon return from each call, the k argument will be set to a