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diff --git a/third_party/resample/man/resample.1 b/third_party/resample/man/resample.1 new file mode 100644 index 00000000..7a85565b --- /dev/null +++ b/third_party/resample/man/resample.1 @@ -0,0 +1,115 @@ +.TH RESAMPLE 1 "5 Jan 2006" "CCRMA" +.SH NAME +resample \- resample a 16-bit mono or stereo sound file by an arbitrary factor + +.SH SYNOPSIS +\fBresample\fR +[-by factor] +[-to newSrate] +[-f filterFile] +[-n] +[-l] +[-trace] +[-version] +inputFile +[outputFile] + +.SH DESCRIPTION +The \fBresample\fR program takes a 16-bit mono or stereo sound file +and performs bandlimited interpolation to produce an output sound file +have a desired new sampling rate. The output file is in the same +format as the input. + +.SH OPTIONS + +.IP \fB\-toSrate\fR +This option or "-byFactor" is required. Specify new sampling rate in +samples per second. The conversion factor is implied and will be set +to the new sampling rate divided by the sampling rate of the input +soundfile. + +.IP \fB\-byFactor\fR +Specify conversion factor. This option or "-toSrate" is required. +The conversion factor is the amount by which the sampling rate is +changed. If the sampling rate of the input signal is Srate1, then the +sampling rate of the output is factor*Srate1. For example, a factor +of 2.0 increases the sampling rate by a factor of 2, giving twice as +many samples in the output signal as in the input. The fractional +part of the conversion factor is accurate to 15 bits. This is +sufficiently accurate that humans should not be able to hear any error +whatsoever in the pitch of resampled sounds. + +.IP \fB\-filterFile\fR +Change the resampling filter from its default. Such a filter file can +be designed by the \fBwindowfilter (1)\fR program (included with the +\fBresample\fR distribution). The preloaded filter file requires an +oversampling factor of at least 20% to avoid aliasing (in other words, +its "transition band" as a lowpass filter is at least 20% of the +useable frequency range in the sampled signal); the stop-band +attenuation is approximately 80 dB. + +.IP \fB\-noFilterInterp\fR +By default, the resampling filter table is linearly interpolated to +provide high audio quality at arbitrary sampling-rate conversion +factors. This option turns off filter interpolation, thus cutting the +number of multiply-adds in half in the inner loop (for most conversion +factors). + +.IP \fB\-linearInterpolation\fR +Select plain linear interpolation for resampling (which means +resampling filter table is not used at all). This option is very fast, +but the output quality is poor unless the signal is already heavily +oversampled. Do not confuse linear interpolation of the signal with +linear interpolation of the resampling-filter-table which is +controlled by the "noFilterInterp" option. + +.IP \fB\-terse\fR +Disable informational printout. + +.IP \fB\-version\fR +Print program version. + +.SH EXAMPLE +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 would look something like + + resample -to 44100 dat.snd cd.snd +or + resample -by 0.91875 dat.snd cd.snd + +Any reasonable sampling rate can be converted to any other. (Note +that, in this example, if you have obtained a direct-digital transfer +from DAT or CD, you probably have some pre-emphasis filtering which +should be canceled using a digital filter. See README.deemph in the +\fBresample\fR release for further information) + +.SH REFERENCES +Source code and further documentation may be found at the Digital +Audio Resampling Home Page (DARHP) located at + + http://ccrma.stanford.edu/~jos/resample/ + +.SH HISTORY +The first version of this software was written by Julius O. Smith III +<jos /at/ ccrma /dot/ stanford /dot/ edu> at CCRMA +<http://ccrma.stanford.edu> in 1981. It was called SRCONV and was +written in SAIL for PDP-10 compatible machines (see the DARHP for that +code). 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 tutorial based on this paper is available at the DARHP. + +Circa 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. + +Sndlib support was added 6/99 by John Gibson <jgg9c@virginia.edu>. + +The \fBresample\fR program is free software distributed in accordance +with the Lesser GNU Public License (LGPL). There is NO warranty; not +even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. |