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/*:filterkit.h */ #import "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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