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-rw-r--r--libaf/filter.c61
1 files changed, 30 insertions, 31 deletions
diff --git a/libaf/filter.c b/libaf/filter.c
index 78d5e86e2d..c5ab039130 100644
--- a/libaf/filter.c
+++ b/libaf/filter.c
@@ -32,13 +32,13 @@
n number of filter taps, where mod(n,4)==0
w filter taps
- x input signal must be a circular buffer which is indexed backwards
+ x input signal must be a circular buffer which is indexed backwards
*/
inline FLOAT_TYPE af_filter_fir(register unsigned int n, const FLOAT_TYPE* w,
const FLOAT_TYPE* x)
{
register FLOAT_TYPE y; // Output
- y = 0.0;
+ y = 0.0;
do{
n--;
y+=w[n]*x[n];
@@ -52,7 +52,7 @@ inline FLOAT_TYPE af_filter_fir(register unsigned int n, const FLOAT_TYPE* w,
d number of filters
xi current index in xq
w filter taps k by n big
- x input signal must be a circular buffers which are indexed backwards
+ x input signal must be a circular buffers which are indexed backwards
y output buffer
s output buffer stride
*/
@@ -82,7 +82,7 @@ int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi,
{
register FLOAT_TYPE* txq = *xq + xi;
register int nt = n*2;
-
+
while(d-- >0){
*txq= *(txq+n) = *in;
txq+=nt;
@@ -99,13 +99,13 @@ int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi,
n filter length must be odd for HP and BS filters
w buffer for the filter taps (must be n long)
- fc cutoff frequencies (1 for LP and HP, 2 for BP and BS)
+ fc cutoff frequencies (1 for LP and HP, 2 for BP and BS)
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
+ variables are ored together: i.e. LP|HAMMING will give a
+ 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
*/
int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
@@ -142,10 +142,10 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
case(KAISER):
af_window_kaiser(n,w,opt); break;
default:
- return -1;
+ return -1;
}
- if(flags & (LP | HP)){
+ if(flags & (LP | HP)){
fc1=*fc;
// Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2
fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25;
@@ -154,7 +154,7 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
if(flags & LP){ // Low pass filter
// If the filter length is odd, there is one point which is exactly
- // in the middle. The value at this point is 2*fCutoff*sin(x)/x,
+ // in the middle. The value at this point is 2*fCutoff*sin(x)/x,
// where x is zero. To make sure nothing strange happens, we set this
// value separately.
if (o){
@@ -206,9 +206,9 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
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);
+ 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;
@@ -220,7 +220,7 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
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);
+ w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
g += 2*w[end-i-1]; // Total gain in filter
}
}
@@ -228,9 +228,9 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
// Normalize gain
g=1/g;
- for (i=0; i<n; i++)
+ for (i=0; i<n; i++)
w[i] *= g;
-
+
return 0;
}
@@ -239,7 +239,7 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
n length of prototype filter
k number of polyphase components
w prototype filter taps
- pw Parallel FIR filter
+ pw Parallel FIR filter
g Filter gain
flags FWD forward indexing
REW reverse indexing
@@ -254,7 +254,7 @@ int af_filter_design_pfir(unsigned int n, unsigned int k, const FLOAT_TYPE* w,
int i; // Counters
int j;
FLOAT_TYPE t; // g * w[i]
-
+
// Sanity check
if(l<1 || k<1 || !w || !pw)
return -1;
@@ -287,7 +287,7 @@ int af_filter_design_pfir(unsigned int n, unsigned int k, const FLOAT_TYPE* w,
/* Pre-warp the coefficients of a numerator or denominator.
Note that a0 is assumed to be 1, so there is no wrapping
- of it.
+ of it.
*/
static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs)
{
@@ -299,7 +299,7 @@ static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs)
/* Transform the numerator and denominator coefficients of s-domain
biquad section into corresponding z-domain coefficients.
-
+
The transfer function for z-domain is:
1 + alpha1 * z^(-1) + alpha2 * z^(-2)
@@ -310,7 +310,7 @@ static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs)
order:
beta1, beta2 (denominator)
alpha1, alpha2 (numerator)
-
+
Arguments:
a - s-domain numerator coefficients
b - s-domain denominator coefficients
@@ -318,10 +318,10 @@ static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs)
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.
+ specified in initial value of k.
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
required to maintain overall gain = 1.0;
*/
@@ -352,26 +352,26 @@ static void af_filter_bilinear(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_T
/* IIR filter design using bilinear transform and prewarp. Transforms
2nd order s domain analog filter into a digital IIR biquad link. To
create a filter fill in a, b, Q and fs and make space for coef and k.
-
- Example Butterworth design:
+
+ Example Butterworth design:
Below are Butterworth polynomials, arranged as a series of 2nd
order sections:
Note: n is filter order.
-
+
n Polynomials
-------------------------------------------------------------------
2 s^2 + 1.4142s + 1
4 (s^2 + 0.765367s + 1) * (s^2 + 1.847759s + 1)
6 (s^2 + 0.5176387s + 1) * (s^2 + 1.414214 + 1) * (s^2 + 1.931852s + 1)
-
+
For n=4 we have following equation for the filter transfer function:
1 1
T(s) = --------------------------- * ----------------------------
s^2 + (1/Q) * 0.765367s + 1 s^2 + (1/Q) * 1.847759s + 1
-
+
The filter consists of two 2nd order sections since highest s power
is 2. Now we can take the coefficients, or the numbers by which s
is multiplied and plug them into a standard formula to be used by
@@ -414,7 +414,7 @@ static void af_filter_bilinear(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_T
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.
+ specified in initial value of k.
fs - sampling rate (Hz)
coef - array of z-domain coefficients to be filled in.
@@ -432,7 +432,7 @@ int af_filter_szxform(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE Q, FL
FLOAT_TYPE at[3];
FLOAT_TYPE bt[3];
- if(!a || !b || !k || !coef || (Q>1000.0 || Q< 1.0))
+ if(!a || !b || !k || !coef || (Q>1000.0 || Q< 1.0))
return -1;
memcpy(at,a,3*sizeof(FLOAT_TYPE));
@@ -448,4 +448,3 @@ int af_filter_szxform(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE Q, FL
return 0;
}
-