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diff --git a/third_party/resample/man/resample.1 b/third_party/resample/man/resample.1 deleted file mode 100644 index 7a85565b..00000000 --- a/third_party/resample/man/resample.1 +++ /dev/null @@ -1,115 +0,0 @@ -.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. |