README file for resample-1.2.tar.Z (from ftp://ccrma-ftp.stanford.edu/pub/NeXT) 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). A Mathematica notebook is provided for display the frequency response of the designed filter. The resample program uses 32-bit fixed-point arithmetic: 16-bits data and 16-bits coefficients. The input soundfile must be 16-bit mono or stereo (interleaved) audio data. While everything is written in plain C, the I/O routines assume NeXT soundfile format (a header, usually 28 bytes, followed by 16-bit, two's-complement samples, interleaved for stereo). 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. testResample Shell script for testing resample at various conversion ratios. testStereo Shell script for testing resample on a stereo input soundfile. 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. testFilter Shell script for testing a filter written by windowfilter. Note: The filter in largefilter.h (obtained using "resample -aaa") is designed by windowfilter when you take all the defaults. The filter in smallfilter.h (used when there is no "-aaa" option) is designed by windowfilter when you take all the defaults except NMULT which is set to 13. i.snd Soundfile containing an "impulse" used to test filters. testFilter.ma Mathematica file for displaying filter test results. warp.c Program for time-varying warping of a sampling rate (NOT TESTED). COPYING This software package is Copyright 1994 by Julius O. Smith (jos@ccrma.stanford.edu), all rights reserved. Permission to use and copy is granted subject to the terms of the "GNU Software General Public License" (see ftp://prep.ai.mit.edu/pub/gnu/COPYING). In addition, we request that a copy of any modified files be sent by email to jos@ccrma.stanford.edu so that we may incorporate them in the CCRMA version. TEST SCRIPTS The test scripts testFilter and testResample make use of various utilities found on NeXT computers: Mathematica Obtainable from Wolfram Research, Champain-Urbana, IL sndutil Soundfile utility package (a copy may be picked up from the directory ftp://ccrma-ftp.stanford.edu/pub/NeXT) Music Kit Computer music utility package (a copy may be picked up from the directory ftp://ccrma-ftp.stanford.edu/pub/NeXT/MusicKit) The soundfile utilities are also written in plain C and support NeXT-format soundfiles. The only Music Kit utility used is the program playscore which is used to generate a test tone. The following utilities and Mathematica are needed by the filter test script testFilter: sndtoascii Utility for converting a sound file to an ascii file (for MMA) sndtrim Utility for stripping leading and trailing zeros in a .snd file The following additional utilities are needed by the sampling-rate convertsion test script testResample: playscore Utility for converting a scorefile to a sound file sndtomono Utility for converting a stereo sound file to a mono sound file 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. HISTORY AND REFERENCES The first version of this software was finished by Julius O. Smith at CCRMA in 1981. It was called SRCONV and was written in SAIL for PDP-10 compatible machines. The algorithm was first published in Smith, Julius O. and Phil Gossett. ``A Flexible Sampling-Rate Conversion Method,'' Proceedings (2): 19.4.1-19.4.4, IEEE Conference on Acoustics, Speech, and Signal Processing, San Diego, March 1984. An expanded version of this paper is available via ftp (file BandlimitedInterpolation.eps.Z in directory ftp://ccrma-ftp.stanford.edu/pub/DSP/Tutorials). Ca. 1988, the SRCONV program was translated from SAIL to C by Christopher Lee Fraley working with Roger Dannenberg at CMU. Since then, the C version has been maintained by JOS.
These are the contents of the former NiCE NeXT User Group NeXTSTEP/OpenStep software archive, currently hosted by Netfuture.ch.