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/*:filterkit.h */
#include "stdefs.h"
/*
* LpFilter() - Calculates the filter coeffs for a Kaiser-windowed low-pass
* filter with a given roll-off frequency. These coeffs
* are stored into a array of doubles.
* writeFilter() - Writes a filter to a file.
* makeFilter() - Calls LpFilter() to create a filter, then scales the double
* coeffs into an array of half words.
* readFilter() - Reads a filter from a file.
* FilterUp() - Applies a filter to a given sample when up-converting.
* FilterUD() - Applies a filter to a given sample when up- or down-
* converting.
* initZerox() - Initialization routine for the zerox() function. Must
* be called before zerox() is called. This routine loads
* the correct filter so zerox() can use it.
* zerox() - Given a pointer into a sample, finds a zero-crossing on the
* interval [pointer-1:pointer+2] by iteration.
* Query() - Ask the user for a yes/no question with prompt, default,
* and optional help.
* GetUShort() - Ask the user for a unsigned short with prompt, default,
* and optional help.
* GetDouble() - Ask the user for a double with prompt, default, and
* optional help.
* GetString() - Ask the user for a string with prompt, default, and
* optional help.
*/
void LpFilter(double c[], int N, double frq, double Beta, int Num);
/*
* reference: "Digital Filters, 2nd edition"
* R.W. Hamming, pp. 178-179
*
* LpFilter() computes the coeffs of a Kaiser-windowed low pass filter with
* the following characteristics:
*
* c[] = array in which to store computed coeffs
* frq = roll-off frequency of filter
* N = Half the window length in number of coeffs
* Beta = parameter of Kaiser window
* Num = number of coeffs before 1/frq
*
* Beta trades the rejection of the lowpass filter against the transition
* width from passband to stopband. Larger Beta means a slower
* transition and greater stopband rejection. See Rabiner and Gold
* (Theory and Application of DSP) under Kaiser windows for more about
* Beta. The following table from Rabiner and Gold gives some feel
* for the effect of Beta:
*
* All ripples in dB, width of transition band = D*N where N = window length
*
* BETA D PB RIP SB RIP
* 2.120 1.50 +-0.27 -30
* 3.384 2.23 0.0864 -40
* 4.538 2.93 0.0274 -50
* 5.658 3.62 0.00868 -60
* 6.764 4.32 0.00275 -70
* 7.865 5.0 0.000868 -80
* 8.960 5.7 0.000275 -90
* 10.056 6.4 0.000087 -100
*/
int writeFilter(HWORD Imp[], HWORD ImpD[], UHWORD LpScl, UHWORD Nmult, UHWORD Nwing);
/*
* Write a filter to a file
* Filter file format:
* file name: "F" Nmult "T" Nhc ".filter"
* 1st line: the string "ScaleFactor" followed by its value.
* 2nd line: the string "Length" followed by Nwing's value.
* 3rd line: the string "Coeffs:" on a separate line.
* following lines: Nwing number of 16-bit impulse response values
* in the right wing of the impulse response (the Imp[] array).
* (Nwing is equal to Npc*(Nmult+1)/2+1, where Npc is defined in the
* file "resample.h".) Each coefficient is on a separate line.
* next line: the string "Differences:" on a separate line.
* following lines: Nwing number of 16-bit impulse-response
* successive differences: ImpD[i] = Imp[i+1] - Imp[i].
* ERROR codes:
* 0 - no error
* 1 - could not open file
*/
int makeFilter(HWORD Imp[], HWORD ImpD[], UHWORD *LpScl, UHWORD Nwing,
double Froll, double Beta);
/*
* makeFilter
* ERROR return codes:
* 0 - no error
* 1 - Nwing too large (Nwing is > MAXNWING)
* 2 - Froll is not in interval [0:1)
* 3 - Beta is < 1.0
* 4 - LpScl will not fit in 16-bits
*/
int readFilter(char *filterFile,
HWORD **ImpP, HWORD **ImpDP, UHWORD *LpScl,
UHWORD *Nmult, UHWORD *Nwing);
/*
* Read-in a filter
* Filter file format:
* Default file name: "F" Nmult "T" Nhc ".filter"
* 1st line: the string "ScaleFactor" followed by its value.
* 2nd line: the string "Length" followed by Nwing's value.
* 3rd line: the string "Coeffs:" on separate line.
* Nwing number of 16-bit impulse response values in the right
* wing of the impulse response. (Length=Npc*(Nmult+1)/2+1,
* where originally Npc=2^9, and Nmult=13.) Each on separate line.
* The string "Differences:" on separate line.
* Nwing number of 16-bit impulse-response successive differences:
* ImpDiff[i] = Imp[i+1] - Imp[i].
*
* ERROR return codes:
* 0 - no error
* 1 - file not found
* 2 - invalid ScaleFactor in file
* 3 - invalid Length in file
*/
WORD FilterUp(HWORD Imp[], HWORD ImpD[], UHWORD Nwing, BOOL Interp,
HWORD *Xp, HWORD Inc, HWORD Ph);
WORD FilterUD(HWORD Imp[], HWORD ImpD[], UHWORD Nwing, BOOL Interp,
HWORD *Xp, HWORD Ph, HWORD Inc, UHWORD dhb);
int initZerox(UHWORD tempNmult);
/*
* initZerox
* ERROR return values:
* 0 - no error
* 1 - Nmult is even (should be odd)
* 2 - filter file not found
* 3 - invalid ScaleFactor in input file
* 4 - invalid Length in file
*/
/*
* zerox
* Given a pointer into a sound sample, this function uses a low-pass
* filter to estimate the x coordinate of the zero-crossing which must ocurr
* between Data[0] and Data[1]. This value is returned as the value of the
* function. A return value of -100 indicates there was no zero-crossing in
* the x interval [-1,2]. Factor is the resampling factor: Rate(out) /
* Rate(in). Nmult (which determines which filter is used) is passed the
* zerox's initialization routine: initZerox(Nmult)
*/
double zerox(HWORD *Data, double Factor);
BOOL Query(char *prompt, BOOL deflt, char *help);
unsigned short GetUShort(char *title, unsigned short deflt, char *help);
double GetDouble(char *title, double deflt, char *help);
char *GetString(char *prompt, char *deflt, char *help);
#define GetUHWORD(x,y,z) GetUShort(x,y,z)
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