README file for resample-1.x.tgz from the Digital Audio Resampling Home Page located at http://ccrma.stanford.edu/~jos/resample/. SOFTWARE FOR SAMPLING-RATE CONVERSION AND FIR DIGITAL FILTER DESIGN The resample program "resamples" a soundfile to change its sampling rate. For example, it can be used to convert the sampling rate from 48 kHz (used by DAT machines) to 44.1 kHz (the standard sampling rate for Compact Discs). The command line for this operation would look something like resample -by 0.91875 dat.snd cd.snd or, more simply, resample -to 44100 dat.snd cd.snd Any reasonable sampling rate can be converted to any other. The windowfilter program designs Finite-Impulse-Response (FIR) digital filters by the so-called "window method." In this method, the ideal impulse response (a sinc function) is "windowed" by a Kaiser window (a popular window used in spectrum analysis). The resample program uses 32-bit fixed-point arithmetic: 16-bits data and 16-bits coefficients. The input soundfile may contain 16-bit mono or stereo (interleaved) audio data. The Linux version uses Bill Schottstaedt's sndlib for soundfile i/o, so it supports all soundfile header formats supported by that package. Everything is written in plain C, so porting to various platforms, even DSP chips, should be straightforward. CONTENTS resample.c Sampling-rate conversion program. resample.1 Manual page for resample. Try "nroff -man resample.1". resamplesubs.c Subroutines used by resample. resample.h Configuration constants for the sampling rate converter. stdefs.h Machine-dependent definitions, useful constants and macros. windowfilter.c Program for designing FIR digital filters used by resample. windowfilter.1 Manual page for windowfilter. filterkit.c Library for filter design, application, and file management. filterkit.h Declarations (procedure prototypes) for the filterkit library. The NeXT distribution further contains some filter tests and program tests based on NeXT sound facilities and Mathematica. FILTERKIT CONTENTS 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() - A section of the original SAIL program. Calls LpFilter() to create a filter, then scales the double coeffs into a 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. Both are repoductions of the original SAIL program. 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. FILTER FILE FORMAT File Name: "F" Nmult "T" Nhc ".filter" example: "F13T8.filter" and "F27T8.filter" Structure of File: "ScaleFactor" LpScl "Length" Nwing "Coeffs:" Imp[0] Imp[1] : Imp[Nwing-1] "Differences:" ImpD[0] ImpD[1] : ImpD[Nwing-1] EOF where: Something enclosed in "" indicates specific characters in the file. Nmult, Nwing, Imp[], and ImpD[] are variables (HWORD) Npc is a conversion constant. EOF is the end of the file. See writeFilter() and readFilter() in "filterkit.c" for more details. -------------------------- This is the README file for software package resample-n.m obtained from the Digital Audio Resampling Home Page located at http://ccrma.stanford.edu/~jos/resample/ See the INSTALL file in the latest source package for installation instructions.