diff options
Diffstat (limited to 'audio/filter/filter.c')
| -rw-r--r-- | audio/filter/filter.c | 72 |
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 |