From c0c0582462720ed597b00e116506570577614e89 Mon Sep 17 00:00:00 2001
From: shamikam
Date: Tue, 7 Nov 2017 15:59:48 +0530
Subject: initial commit
---
help/en_US/ac2poly.xml | 62 +
help/en_US/ac2rc.xml | 31 +
help/en_US/arParEst.xml | 23 +
help/en_US/arburg.xml | 69 ++
help/en_US/arcov.xml | 23 +
help/en_US/armcov.xml | 23 +
help/en_US/aryule.xml | 63 +
help/en_US/barthannwin.xml | 60 +
help/en_US/bartlett.xml | 56 +
help/en_US/besself.xml | 66 ++
help/en_US/bitrevorder.xml | 57 +
help/en_US/blackman.xml | 60 +
help/en_US/blackmanharris.xml | 65 ++
help/en_US/blackmannuttall.xml | 56 +
help/en_US/bohmanwin.xml | 60 +
help/en_US/boxcar.xml | 62 +
help/en_US/buffer.xml | 66 ++
help/en_US/build_help.sce | 16 +
help/en_US/butter.xml | 71 ++
help/en_US/buttord.xml | 70 ++
help/en_US/cconv.xml | 23 +
help/en_US/cell2sos.xml | 100 ++
help/en_US/cheb1ord.xml | 67 ++
help/en_US/cheb2ord.xml | 67 ++
help/en_US/chebwin.xml | 67 ++
help/en_US/cheby1.xml | 74 ++
help/en_US/cheby2.xml | 74 ++
help/en_US/check.xml | 23 +
help/en_US/chirp.xml | 75 ++
help/en_US/clustersegment.xml | 60 +
help/en_US/cmorwavf.xml | 23 +
help/en_US/convmtx.xml | 23 +
help/en_US/corrmtx.xml | 129 +++
help/en_US/cummax.xml | 70 ++
help/en_US/cummin.xml | 70 ++
help/en_US/db.xml | 23 +
help/en_US/db2pow.xml | 53 +
help/en_US/dctmtx.xml | 58 +
help/en_US/decimate.xml | 23 +
help/en_US/dftmtx.xml | 56 +
help/en_US/diric.xml | 58 +
help/en_US/downsample.xml | 61 +
help/en_US/dutycycle.xml | 23 +
help/en_US/ellip.xml | 74 ++
help/en_US/ellipord.xml | 68 ++
help/en_US/enbw.xml | 23 +
help/en_US/eqtflength.xml | 31 +
help/en_US/falltime.xml | 73 ++
help/en_US/fftfilt.xml | 74 ++
help/en_US/filternorm.xml | 75 ++
help/en_US/filtfilt.xml | 58 +
help/en_US/filtic.xml | 65 ++
help/en_US/filtord.xml | 23 +
help/en_US/firpmord.xml | 91 ++
help/en_US/firtype.xml | 23 +
help/en_US/flattopwin.xml | 69 ++
help/en_US/fracshift.xml | 62 +
help/en_US/fwhm.xml | 43 +
help/en_US/fwhmjlt.xml | 23 +
help/en_US/gauspuls.xml | 62 +
help/en_US/gaussdesign.xml | 23 +
help/en_US/gaussian.xml | 65 ++
help/en_US/gausswin.xml | 63 +
help/en_US/gmonopuls.xml | 51 +
help/en_US/goertzel.xml | 67 ++
help/en_US/grpdelay.xml | 35 +
help/en_US/hann.xml | 67 ++
help/en_US/helperHarmonicDistortionAmplifier.xml | 23 +
help/en_US/icceps.xml | 23 +
help/en_US/iirlp2mb.xml | 67 ++
help/en_US/impinvar.xml | 68 ++
help/en_US/impz.xml | 34 +
help/en_US/impzlength.xml | 76 ++
help/en_US/interp.xml | 64 ++
help/en_US/intfilt.xml | 23 +
help/en_US/invimpinvar.xml | 71 ++
help/en_US/is2rc.xml | 74 ++
help/en_US/isallpass.xml | 23 +
help/en_US/isfir.xml | 23 +
help/en_US/islinphase.xml | 23 +
help/en_US/ismaxphase.xml | 23 +
help/en_US/isminphase.xml | 23 +
help/en_US/isstable.xml | 23 +
help/en_US/kaiser.xml | 66 ++
help/en_US/lar2rc.xml | 60 +
help/en_US/latc2tf.xml | 89 ++
help/en_US/latcfilt.xml | 23 +
help/en_US/latcfilt1.xml | 23 +
help/en_US/levdown.xml | 23 +
help/en_US/levin.xml | 23 +
help/en_US/levinson.xml | 23 +
help/en_US/lpc.xml | 90 ++
help/en_US/lsf2poly.xml | 30 +
help/en_US/mag2db.xml | 23 +
help/en_US/marcumq.xml | 67 ++
help/en_US/master_help.xml | 407 +++++++
help/en_US/medfilt1.xml | 98 ++
help/en_US/mexihat.xml | 62 +
help/en_US/meyeraux.xml | 56 +
help/en_US/midcross.xml | 23 +
help/en_US/modulate.xml | 82 ++
help/en_US/morlet.xml | 62 +
help/en_US/movingrms.xml | 68 ++
help/en_US/musicBase.xml | 23 +
help/en_US/nnls.xml | 23 +
help/en_US/nuttallwin.xml | 62 +
help/en_US/parzenwin.xml | 59 +
help/en_US/pchip.xml | 23 +
help/en_US/pchips.xml | 23 +
help/en_US/peak2peak.xml | 23 +
help/en_US/peak2rms.xml | 140 +++
help/en_US/peig.xml | 101 ++
help/en_US/periodogram.xml | 23 +
help/en_US/phaseInputParseAs_ab.xml | 23 +
help/en_US/phaseInputParseAs_sos.xml | 23 +
help/en_US/phasedelay.xml | 23 +
help/en_US/phasez.xml | 23 +
help/en_US/pmusic.xml | 101 ++
help/en_US/poly2ac.xml | 80 ++
help/en_US/poly2lsf.xml | 23 +
help/en_US/poly2rc.xml | 31 +
help/en_US/polyscale.xml | 23 +
help/en_US/polystab.xml | 55 +
help/en_US/polyval.xml | 23 +
help/en_US/pow2db.xml | 23 +
help/en_US/primitive.xml | 60 +
help/en_US/prony.xml | 23 +
help/en_US/pulseperiod.xml | 36 +
help/en_US/pulsesep.xml | 36 +
help/en_US/pulsewidth.xml | 36 +
help/en_US/pulstran.xml | 67 ++
help/en_US/rc2ac.xml | 30 +
help/en_US/rc2is.xml | 80 ++
help/en_US/rc2lar.xml | 23 +
help/en_US/rc2poly.xml | 31 +
help/en_US/rcosdesign.xml | 23 +
help/en_US/rectpuls.xml | 60 +
help/en_US/rectwin.xml | 59 +
help/en_US/resample.xml | 63 +
help/en_US/residued.xml | 59 +
help/en_US/residuez.xml | 59 +
help/en_US/risetime.xml | 23 +
help/en_US/rlevinson.xml | 33 +
help/en_US/rms.xml | 23 +
help/en_US/rooteig.xml | 62 +
help/en_US/rootmusic.xml | 62 +
help/en_US/rssq.xml | 121 ++
help/en_US/sampled2continuous.xml | 60 +
help/en_US/sawtooth.xml | 59 +
help/en_US/schtrig.xml | 61 +
help/en_US/schurrc.xml | 23 +
help/en_US/scilab_en_US_help/JavaHelpSearch/DOCS | Bin 0 -> 10301 bytes
.../scilab_en_US_help/JavaHelpSearch/DOCS.TAB | Bin 0 -> 1385 bytes
.../en_US/scilab_en_US_help/JavaHelpSearch/OFFSETS | Bin 0 -> 666 bytes
.../scilab_en_US_help/JavaHelpSearch/POSITIONS | Bin 0 -> 43193 bytes
help/en_US/scilab_en_US_help/JavaHelpSearch/SCHEMA | 2 +
help/en_US/scilab_en_US_help/JavaHelpSearch/TMAP | Bin 0 -> 24576 bytes
help/en_US/scilab_en_US_help/ScilabCaution.png | Bin 0 -> 513 bytes
help/en_US/scilab_en_US_help/ScilabEdit.png | Bin 0 -> 414 bytes
help/en_US/scilab_en_US_help/ScilabExecute.png | Bin 0 -> 535 bytes
help/en_US/scilab_en_US_help/ScilabImportant.png | Bin 0 -> 637 bytes
help/en_US/scilab_en_US_help/ScilabNote.png | Bin 0 -> 687 bytes
help/en_US/scilab_en_US_help/ScilabTip.png | Bin 0 -> 687 bytes
help/en_US/scilab_en_US_help/ScilabWarning.png | Bin 0 -> 513 bytes
help/en_US/scilab_en_US_help/ac2poly.html | 81 ++
help/en_US/scilab_en_US_help/ac2rc.html | 64 ++
help/en_US/scilab_en_US_help/arParEst.html | 59 +
help/en_US/scilab_en_US_help/arburg.html | 90 ++
help/en_US/scilab_en_US_help/arcov.html | 59 +
help/en_US/scilab_en_US_help/armcov.html | 59 +
help/en_US/scilab_en_US_help/aryule.html | 84 ++
help/en_US/scilab_en_US_help/barthannwin.html | 81 ++
help/en_US/scilab_en_US_help/bartlett.html | 80 ++
help/en_US/scilab_en_US_help/besself.html | 87 ++
help/en_US/scilab_en_US_help/bitrevorder.html | 78 ++
help/en_US/scilab_en_US_help/blackman.html | 84 ++
help/en_US/scilab_en_US_help/blackmanharris.html | 86 ++
help/en_US/scilab_en_US_help/blackmannuttall.html | 81 ++
help/en_US/scilab_en_US_help/bohmanwin.html | 81 ++
help/en_US/scilab_en_US_help/boxcar.html | 83 ++
help/en_US/scilab_en_US_help/buffer.html | 87 ++
help/en_US/scilab_en_US_help/butter.html | 92 ++
help/en_US/scilab_en_US_help/buttord.html | 91 ++
help/en_US/scilab_en_US_help/c_code.css | 54 +
help/en_US/scilab_en_US_help/cconv.html | 59 +
help/en_US/scilab_en_US_help/cell2sos.html | 100 ++
help/en_US/scilab_en_US_help/cheb1ord.html | 88 ++
help/en_US/scilab_en_US_help/cheb2ord.html | 88 ++
help/en_US/scilab_en_US_help/chebwin.html | 88 ++
help/en_US/scilab_en_US_help/cheby1.html | 95 ++
help/en_US/scilab_en_US_help/cheby2.html | 95 ++
help/en_US/scilab_en_US_help/check.html | 59 +
help/en_US/scilab_en_US_help/chirp.html | 96 ++
help/en_US/scilab_en_US_help/clustersegment.html | 81 ++
help/en_US/scilab_en_US_help/cmorwavf.html | 59 +
help/en_US/scilab_en_US_help/convmtx.html | 59 +
help/en_US/scilab_en_US_help/corrmtx.html | 142 +++
help/en_US/scilab_en_US_help/cummax.html | 87 ++
help/en_US/scilab_en_US_help/cummin.html | 87 ++
help/en_US/scilab_en_US_help/db.html | 59 +
help/en_US/scilab_en_US_help/db2pow.html | 75 ++
help/en_US/scilab_en_US_help/dctmtx.html | 79 ++
help/en_US/scilab_en_US_help/decimate.html | 59 +
help/en_US/scilab_en_US_help/dftmtx.html | 77 ++
help/en_US/scilab_en_US_help/diric.html | 79 ++
help/en_US/scilab_en_US_help/downsample.html | 82 ++
help/en_US/scilab_en_US_help/dutycycle.html | 59 +
help/en_US/scilab_en_US_help/ellip.html | 95 ++
help/en_US/scilab_en_US_help/ellipord.html | 89 ++
help/en_US/scilab_en_US_help/enbw.html | 59 +
help/en_US/scilab_en_US_help/eqtflength.html | 64 ++
help/en_US/scilab_en_US_help/falltime.html | 101 ++
help/en_US/scilab_en_US_help/fftfilt.html | 90 ++
help/en_US/scilab_en_US_help/filternorm.html | 88 ++
help/en_US/scilab_en_US_help/filtfilt.html | 79 ++
help/en_US/scilab_en_US_help/filtic.html | 86 ++
help/en_US/scilab_en_US_help/filtord.html | 59 +
help/en_US/scilab_en_US_help/firpmord.html | 108 ++
help/en_US/scilab_en_US_help/firtype.html | 59 +
help/en_US/scilab_en_US_help/flattopwin.html | 90 ++
help/en_US/scilab_en_US_help/fracshift.html | 83 ++
help/en_US/scilab_en_US_help/fwhm.html | 72 ++
help/en_US/scilab_en_US_help/fwhmjlt.html | 59 +
help/en_US/scilab_en_US_help/gauspuls.html | 83 ++
help/en_US/scilab_en_US_help/gaussdesign.html | 59 +
help/en_US/scilab_en_US_help/gaussian.html | 86 ++
help/en_US/scilab_en_US_help/gausswin.html | 84 ++
help/en_US/scilab_en_US_help/gmonopuls.html | 76 ++
help/en_US/scilab_en_US_help/goertzel.html | 89 ++
help/en_US/scilab_en_US_help/grpdelay.html | 68 ++
help/en_US/scilab_en_US_help/hann.html | 88 ++
.../helperHarmonicDistortionAmplifier.html | 59 +
help/en_US/scilab_en_US_help/icceps.html | 59 +
help/en_US/scilab_en_US_help/iirlp2mb.html | 88 ++
help/en_US/scilab_en_US_help/impinvar.html | 89 ++
help/en_US/scilab_en_US_help/impz.html | 67 ++
help/en_US/scilab_en_US_help/impzlength.html | 92 ++
help/en_US/scilab_en_US_help/index.html | 1207 ++++++++++++++++++++
help/en_US/scilab_en_US_help/interp.html | 85 ++
help/en_US/scilab_en_US_help/intfilt.html | 59 +
help/en_US/scilab_en_US_help/invimpinvar.html | 92 ++
help/en_US/scilab_en_US_help/is2rc.html | 89 ++
help/en_US/scilab_en_US_help/isallpass.html | 59 +
help/en_US/scilab_en_US_help/isfir.html | 59 +
help/en_US/scilab_en_US_help/islinphase.html | 59 +
help/en_US/scilab_en_US_help/ismaxphase.html | 59 +
help/en_US/scilab_en_US_help/isminphase.html | 59 +
help/en_US/scilab_en_US_help/isstable.html | 59 +
help/en_US/scilab_en_US_help/jhelpidx.xml | 3 +
help/en_US/scilab_en_US_help/jhelpmap.jhm | 199 ++++
help/en_US/scilab_en_US_help/jhelpset.hs | 28 +
help/en_US/scilab_en_US_help/jhelptoc.xml | 201 ++++
help/en_US/scilab_en_US_help/kaiser.html | 87 ++
help/en_US/scilab_en_US_help/lar2rc.html | 80 ++
help/en_US/scilab_en_US_help/latc2tf.html | 103 ++
help/en_US/scilab_en_US_help/latcfilt.html | 59 +
help/en_US/scilab_en_US_help/latcfilt1.html | 59 +
help/en_US/scilab_en_US_help/levdown.html | 59 +
help/en_US/scilab_en_US_help/levin.html | 59 +
help/en_US/scilab_en_US_help/levinson.html | 59 +
help/en_US/scilab_en_US_help/lpc.html | 98 ++
help/en_US/scilab_en_US_help/lsf2poly.html | 63 +
help/en_US/scilab_en_US_help/mag2db.html | 59 +
help/en_US/scilab_en_US_help/marcumq.html | 88 ++
help/en_US/scilab_en_US_help/medfilt1.html | 110 ++
help/en_US/scilab_en_US_help/mexihat.html | 83 ++
help/en_US/scilab_en_US_help/meyeraux.html | 77 ++
help/en_US/scilab_en_US_help/midcross.html | 59 +
help/en_US/scilab_en_US_help/modulate.html | 104 ++
help/en_US/scilab_en_US_help/morlet.html | 83 ++
help/en_US/scilab_en_US_help/movingrms.html | 89 ++
help/en_US/scilab_en_US_help/musicBase.html | 59 +
help/en_US/scilab_en_US_help/nnls.html | 59 +
help/en_US/scilab_en_US_help/nuttallwin.html | 83 ++
help/en_US/scilab_en_US_help/parzenwin.html | 80 ++
help/en_US/scilab_en_US_help/pchip.html | 59 +
help/en_US/scilab_en_US_help/pchips.html | 59 +
help/en_US/scilab_en_US_help/peak2peak.html | 59 +
help/en_US/scilab_en_US_help/peak2rms.html | 127 ++
help/en_US/scilab_en_US_help/peig.html | 125 ++
help/en_US/scilab_en_US_help/periodogram.html | 59 +
.../scilab_en_US_help/phaseInputParseAs_ab.html | 59 +
.../scilab_en_US_help/phaseInputParseAs_sos.html | 59 +
help/en_US/scilab_en_US_help/phasedelay.html | 59 +
help/en_US/scilab_en_US_help/phasez.html | 59 +
help/en_US/scilab_en_US_help/pmusic.html | 125 ++
help/en_US/scilab_en_US_help/poly2ac.html | 92 ++
help/en_US/scilab_en_US_help/poly2lsf.html | 59 +
help/en_US/scilab_en_US_help/poly2rc.html | 64 ++
help/en_US/scilab_en_US_help/polyscale.html | 59 +
help/en_US/scilab_en_US_help/polystab.html | 76 ++
help/en_US/scilab_en_US_help/polyval.html | 59 +
help/en_US/scilab_en_US_help/pow2db.html | 59 +
help/en_US/scilab_en_US_help/primitive.html | 81 ++
help/en_US/scilab_en_US_help/prony.html | 59 +
help/en_US/scilab_en_US_help/pulseperiod.html | 69 ++
help/en_US/scilab_en_US_help/pulsesep.html | 69 ++
help/en_US/scilab_en_US_help/pulsewidth.html | 69 ++
help/en_US/scilab_en_US_help/pulstran.html | 88 ++
help/en_US/scilab_en_US_help/rc2ac.html | 63 +
help/en_US/scilab_en_US_help/rc2is.html | 91 ++
help/en_US/scilab_en_US_help/rc2lar.html | 59 +
help/en_US/scilab_en_US_help/rc2poly.html | 64 ++
help/en_US/scilab_en_US_help/rcosdesign.html | 59 +
help/en_US/scilab_en_US_help/rectpuls.html | 81 ++
help/en_US/scilab_en_US_help/rectwin.html | 80 ++
help/en_US/scilab_en_US_help/resample.html | 84 ++
help/en_US/scilab_en_US_help/residued.html | 80 ++
help/en_US/scilab_en_US_help/residuez.html | 80 ++
help/en_US/scilab_en_US_help/risetime.html | 59 +
help/en_US/scilab_en_US_help/rlevinson.html | 66 ++
help/en_US/scilab_en_US_help/rms.html | 59 +
help/en_US/scilab_en_US_help/rooteig.html | 86 ++
help/en_US/scilab_en_US_help/rootmusic.html | 86 ++
help/en_US/scilab_en_US_help/rssq.html | 116 ++
.../scilab_en_US_help/sampled2continuous.html | 81 ++
help/en_US/scilab_en_US_help/sawtooth.html | 80 ++
help/en_US/scilab_en_US_help/schtrig.html | 82 ++
help/en_US/scilab_en_US_help/schurrc.html | 59 +
help/en_US/scilab_en_US_help/scilab_code.css | 96 ++
.../section_be19baaac84b122b8ad501d34462f5bf.html | 1207 ++++++++++++++++++++
help/en_US/scilab_en_US_help/seqperiod.html | 105 ++
help/en_US/scilab_en_US_help/sgolay.html | 87 ++
help/en_US/scilab_en_US_help/sgolayfilt.html | 96 ++
help/en_US/scilab_en_US_help/shanwavf.html | 88 ++
help/en_US/scilab_en_US_help/shiftdata.html | 118 ++
help/en_US/scilab_en_US_help/slewrate.html | 59 +
help/en_US/scilab_en_US_help/sos2cell.html | 96 ++
help/en_US/scilab_en_US_help/sos2ss.html | 59 +
help/en_US/scilab_en_US_help/sos2tf.html | 84 ++
help/en_US/scilab_en_US_help/sos2zp.html | 92 ++
help/en_US/scilab_en_US_help/sosbreak.html | 59 +
help/en_US/scilab_en_US_help/sosfilt.html | 79 ++
help/en_US/scilab_en_US_help/specgram.html | 59 +
help/en_US/scilab_en_US_help/ss2sos.html | 59 +
help/en_US/scilab_en_US_help/statelevels.html | 97 ++
help/en_US/scilab_en_US_help/stmcb.html | 59 +
help/en_US/scilab_en_US_help/strips.html | 85 ++
help/en_US/scilab_en_US_help/style.css | 350 ++++++
.../subspaceMethodsInputParser.html | 59 +
help/en_US/scilab_en_US_help/tf2sos.html | 84 ++
help/en_US/scilab_en_US_help/tf2zp.html | 59 +
help/en_US/scilab_en_US_help/tf2zpk.html | 59 +
help/en_US/scilab_en_US_help/transpose.html | 59 +
help/en_US/scilab_en_US_help/trial_iirlp2mb.html | 59 +
help/en_US/scilab_en_US_help/triang.html | 82 ++
help/en_US/scilab_en_US_help/tripuls.html | 84 ++
help/en_US/scilab_en_US_help/truth.html | 59 +
help/en_US/scilab_en_US_help/tukeywin.html | 84 ++
help/en_US/scilab_en_US_help/udecode.html | 86 ++
help/en_US/scilab_en_US_help/uencode.html | 87 ++
help/en_US/scilab_en_US_help/ultrwin.html | 92 ++
help/en_US/scilab_en_US_help/unshiftdata.html | 122 ++
help/en_US/scilab_en_US_help/upfirdn.html | 83 ++
help/en_US/scilab_en_US_help/upsample.html | 82 ++
help/en_US/scilab_en_US_help/upsamplefill.html | 83 ++
help/en_US/scilab_en_US_help/var.html | 75 ++
help/en_US/scilab_en_US_help/vco.html | 87 ++
help/en_US/scilab_en_US_help/welchwin.html | 86 ++
help/en_US/scilab_en_US_help/window.html | 88 ++
help/en_US/scilab_en_US_help/wkeep.html | 83 ++
help/en_US/scilab_en_US_help/wrev.html | 77 ++
help/en_US/scilab_en_US_help/xcorr2.html | 65 ++
help/en_US/scilab_en_US_help/xml_code.css | 94 ++
help/en_US/scilab_en_US_help/zerocrossing.html | 78 ++
help/en_US/scilab_en_US_help/zp2sos.html | 87 ++
help/en_US/scilab_en_US_help/zplane.html | 57 +
help/en_US/seqperiod.xml | 88 ++
help/en_US/sgolay.xml | 66 ++
help/en_US/sgolayfilt.xml | 75 ++
help/en_US/shanwavf.xml | 67 ++
help/en_US/shiftdata.xml | 99 ++
help/en_US/slewrate.xml | 23 +
help/en_US/sos2cell.xml | 87 ++
help/en_US/sos2ss.xml | 23 +
help/en_US/sos2tf.xml | 63 +
help/en_US/sos2zp.xml | 71 ++
help/en_US/sosbreak.xml | 23 +
help/en_US/sosfilt.xml | 58 +
help/en_US/specgram.xml | 23 +
help/en_US/ss2sos.xml | 23 +
help/en_US/statelevels.xml | 79 ++
help/en_US/stmcb.xml | 23 +
help/en_US/strips.xml | 59 +
help/en_US/subspaceMethodsInputParser.xml | 23 +
help/en_US/tf2sos.xml | 63 +
help/en_US/tf2zp.xml | 23 +
help/en_US/tf2zpk.xml | 23 +
help/en_US/transpose.xml | 23 +
help/en_US/trial_iirlp2mb.xml | 23 +
help/en_US/triang.xml | 61 +
help/en_US/tripuls.xml | 63 +
help/en_US/truth.xml | 23 +
help/en_US/tukeywin.xml | 63 +
help/en_US/udecode.xml | 66 ++
help/en_US/uencode.xml | 67 ++
help/en_US/ultrwin.xml | 71 ++
help/en_US/unshiftdata.xml | 102 ++
help/en_US/upfirdn.xml | 62 +
help/en_US/upsample.xml | 61 +
help/en_US/upsamplefill.xml | 62 +
help/en_US/var.xml | 46 +
help/en_US/vco.xml | 65 ++
help/en_US/welchwin.xml | 65 ++
help/en_US/window.xml | 67 ++
help/en_US/wkeep.xml | 62 +
help/en_US/wrev.xml | 56 +
help/en_US/xcorr2.xml | 32 +
help/en_US/zerocrossing.xml | 56 +
help/en_US/zp2sos.xml | 66 ++
help/en_US/zplane.xml | 23 +
411 files changed, 28676 insertions(+)
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create mode 100644 help/en_US/scilab_en_US_help/sos2zp.html
create mode 100644 help/en_US/scilab_en_US_help/sosbreak.html
create mode 100644 help/en_US/scilab_en_US_help/sosfilt.html
create mode 100644 help/en_US/scilab_en_US_help/specgram.html
create mode 100644 help/en_US/scilab_en_US_help/ss2sos.html
create mode 100644 help/en_US/scilab_en_US_help/statelevels.html
create mode 100644 help/en_US/scilab_en_US_help/stmcb.html
create mode 100644 help/en_US/scilab_en_US_help/strips.html
create mode 100644 help/en_US/scilab_en_US_help/style.css
create mode 100644 help/en_US/scilab_en_US_help/subspaceMethodsInputParser.html
create mode 100644 help/en_US/scilab_en_US_help/tf2sos.html
create mode 100644 help/en_US/scilab_en_US_help/tf2zp.html
create mode 100644 help/en_US/scilab_en_US_help/tf2zpk.html
create mode 100644 help/en_US/scilab_en_US_help/transpose.html
create mode 100644 help/en_US/scilab_en_US_help/trial_iirlp2mb.html
create mode 100644 help/en_US/scilab_en_US_help/triang.html
create mode 100644 help/en_US/scilab_en_US_help/tripuls.html
create mode 100644 help/en_US/scilab_en_US_help/truth.html
create mode 100644 help/en_US/scilab_en_US_help/tukeywin.html
create mode 100644 help/en_US/scilab_en_US_help/udecode.html
create mode 100644 help/en_US/scilab_en_US_help/uencode.html
create mode 100644 help/en_US/scilab_en_US_help/ultrwin.html
create mode 100644 help/en_US/scilab_en_US_help/unshiftdata.html
create mode 100644 help/en_US/scilab_en_US_help/upfirdn.html
create mode 100644 help/en_US/scilab_en_US_help/upsample.html
create mode 100644 help/en_US/scilab_en_US_help/upsamplefill.html
create mode 100644 help/en_US/scilab_en_US_help/var.html
create mode 100644 help/en_US/scilab_en_US_help/vco.html
create mode 100644 help/en_US/scilab_en_US_help/welchwin.html
create mode 100644 help/en_US/scilab_en_US_help/window.html
create mode 100644 help/en_US/scilab_en_US_help/wkeep.html
create mode 100644 help/en_US/scilab_en_US_help/wrev.html
create mode 100644 help/en_US/scilab_en_US_help/xcorr2.html
create mode 100644 help/en_US/scilab_en_US_help/xml_code.css
create mode 100644 help/en_US/scilab_en_US_help/zerocrossing.html
create mode 100644 help/en_US/scilab_en_US_help/zp2sos.html
create mode 100644 help/en_US/scilab_en_US_help/zplane.html
create mode 100644 help/en_US/seqperiod.xml
create mode 100644 help/en_US/sgolay.xml
create mode 100644 help/en_US/sgolayfilt.xml
create mode 100644 help/en_US/shanwavf.xml
create mode 100644 help/en_US/shiftdata.xml
create mode 100644 help/en_US/slewrate.xml
create mode 100644 help/en_US/sos2cell.xml
create mode 100644 help/en_US/sos2ss.xml
create mode 100644 help/en_US/sos2tf.xml
create mode 100644 help/en_US/sos2zp.xml
create mode 100644 help/en_US/sosbreak.xml
create mode 100644 help/en_US/sosfilt.xml
create mode 100644 help/en_US/specgram.xml
create mode 100644 help/en_US/ss2sos.xml
create mode 100644 help/en_US/statelevels.xml
create mode 100644 help/en_US/stmcb.xml
create mode 100644 help/en_US/strips.xml
create mode 100644 help/en_US/subspaceMethodsInputParser.xml
create mode 100644 help/en_US/tf2sos.xml
create mode 100644 help/en_US/tf2zp.xml
create mode 100644 help/en_US/tf2zpk.xml
create mode 100644 help/en_US/transpose.xml
create mode 100644 help/en_US/trial_iirlp2mb.xml
create mode 100644 help/en_US/triang.xml
create mode 100644 help/en_US/tripuls.xml
create mode 100644 help/en_US/truth.xml
create mode 100644 help/en_US/tukeywin.xml
create mode 100644 help/en_US/udecode.xml
create mode 100644 help/en_US/uencode.xml
create mode 100644 help/en_US/ultrwin.xml
create mode 100644 help/en_US/unshiftdata.xml
create mode 100644 help/en_US/upfirdn.xml
create mode 100644 help/en_US/upsample.xml
create mode 100644 help/en_US/upsamplefill.xml
create mode 100644 help/en_US/var.xml
create mode 100644 help/en_US/vco.xml
create mode 100644 help/en_US/welchwin.xml
create mode 100644 help/en_US/window.xml
create mode 100644 help/en_US/wkeep.xml
create mode 100644 help/en_US/wrev.xml
create mode 100644 help/en_US/xcorr2.xml
create mode 100644 help/en_US/zerocrossing.xml
create mode 100644 help/en_US/zp2sos.xml
create mode 100644 help/en_US/zplane.xml
(limited to 'help/en_US')
diff --git a/help/en_US/ac2poly.xml b/help/en_US/ac2poly.xml
new file mode 100644
index 0000000..e90759d
--- /dev/null
+++ b/help/en_US/ac2poly.xml
@@ -0,0 +1,62 @@
+
+
+
+
+
+
+
+ ac2poly
+ Convert autocorrelation sequence to polynomial of prediction filter
+
+
+
+
+ Calling Sequence
+
+ a = ac2poly(r)
+ [a,e] = ac2poly(r)
+
+
+
+
+
+ Parameters
+
+ r:
+ Autocorrelation sequence to be represented with an FIR linear prediction filter
+ a:
+ Output polynomial representing the linear prediction filter e/(a(1) + a(2)z + a(3)z^2 .. a(N)z^N-1)
+ e:
+ Output scaling for the lienar prediction filter
+
+
+
+
+ Description
+
+Function ac2poly() finds the best fit polynomial for FIR linear prediction filter a, corresponding to the autocorrelation sequence r. a is the same length as r, and is normalized with the first element. So a(1) = 1.
+
+
+Author
+Parthe Pandit
+
+
+
+
+
+
+ Bibliography
+ Kay, Steven M. Modern Spectral Estimation. Englewood Cliffs, NJ: Prentice-Hall, 1988.
+
+
diff --git a/help/en_US/ac2rc.xml b/help/en_US/ac2rc.xml
new file mode 100644
index 0000000..2e8a109
--- /dev/null
+++ b/help/en_US/ac2rc.xml
@@ -0,0 +1,31 @@
+
+
+
+
+
+
+
+ ac2rc
+
+
+
+
+
+ Calling Sequence
+
+ k = ac2rc(R)
+ [k,R0] = ac2rc(R)
+
+
+
diff --git a/help/en_US/arParEst.xml b/help/en_US/arParEst.xml
new file mode 100644
index 0000000..c1e0a78
--- /dev/null
+++ b/help/en_US/arParEst.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ arParEst
+
+
+
+
diff --git a/help/en_US/arburg.xml b/help/en_US/arburg.xml
new file mode 100644
index 0000000..75648ba
--- /dev/null
+++ b/help/en_US/arburg.xml
@@ -0,0 +1,69 @@
+
+
+
+
+
+
+
+ arburg
+ This function calculates coefficients of an autoregressive (AR) model of complex data.
+
+
+
+
+ Calling Sequence
+
+ a = arburg(x, poles)
+ a = arburg(x, poles, criterion)
+ [a, v] = arburg(...)
+ [a, v, k] = arburg(...)
+
+
+
+
+ Parameters
+
+ x:
+ vector of real or complex numbers, of length > 2
+ poles:
+ positive integer value < length(x) - 2
+ criterion:
+ string value, takes in "AKICc", "KIC", "AICc", "AIC" and "FPE", default it not using a model-selection criterion
+ a, v, k:
+ Output variables
+
+
+
+
+ Description
+
+This is an Octave function.
+This function calculates coefficients of an autoregressive (AR) model of complex data x using the whitening lattice-filter method of Burg.
+The first argument is the data sampled. The second argument is the number of poles in the model (or limit in case a criterion is supplied).
+The third parameter takes in the criterion to limit the number of poles. The acceptable values are "AIC", "AKICc", "KIC", "AICc" which are based on information theory.
+Output variable a is a list of P+1 autoregression coefficients.
+Output variable v is the mean square of residual noise from the whitening operation of the Burg lattice filter.
+Output variable k corresponds to the reflection coefficients defining the lattice-filter embodiment of the model.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/arcov.xml b/help/en_US/arcov.xml
new file mode 100644
index 0000000..64a5364
--- /dev/null
+++ b/help/en_US/arcov.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ arcov
+
+
+
+
diff --git a/help/en_US/armcov.xml b/help/en_US/armcov.xml
new file mode 100644
index 0000000..62e33b7
--- /dev/null
+++ b/help/en_US/armcov.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ armcov
+
+
+
+
diff --git a/help/en_US/aryule.xml b/help/en_US/aryule.xml
new file mode 100644
index 0000000..9948a0f
--- /dev/null
+++ b/help/en_US/aryule.xml
@@ -0,0 +1,63 @@
+
+
+
+
+
+
+
+ aryule
+ This function fits an AR (p)-model with Yule-Walker estimates.
+
+
+
+
+ Calling Sequence
+
+ a = aryule (x, p)
+ [a, v] = aryule (x, p)
+ [a, v, k] = aryule (x, p)
+
+
+
+
+ Parameters
+
+ x:
+ vector of real or complex numbers, length > 2
+ p:
+ positive integer value < length(x) - 1
+ a, v, k:
+ Output variables
+
+
+
+
+ Description
+
+This is an Octave function.
+This function fits an AR (p)-model with Yule-Walker estimates.
+The first argument is the data vector which is to be estimated.
+Output variable a gives the AR coefficients, v gives the variance of the white noise and k gives the reflection coefficients to be used in the lattice filter.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/barthannwin.xml b/help/en_US/barthannwin.xml
new file mode 100644
index 0000000..ad1139a
--- /dev/null
+++ b/help/en_US/barthannwin.xml
@@ -0,0 +1,60 @@
+
+
+
+
+
+
+
+ barthannwin
+ This function returns the filter coefficients of a modified Bartlett-Hann window.
+
+
+
+
+ Calling Sequence
+
+ y = barthannwin(m)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ y:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a modified Bartlett Hann window of length m supplied as input, to the output vector y.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/bartlett.xml b/help/en_US/bartlett.xml
new file mode 100644
index 0000000..41ef318
--- /dev/null
+++ b/help/en_US/bartlett.xml
@@ -0,0 +1,56 @@
+
+
+
+
+
+
+
+ bartlett
+ Generates a Bartlett window
+
+
+
+
+ Calling Sequence
+
+ w=bartlett(L)
+
+
+
+
+ Parameters
+
+
+
+
+
+ Description
+
+w=bartlett(L) returns an L-point Bartlett window in a column vector w
+Example
+w=bartlett(4)
+w =
+
+
+0.
+0.6666667
+0.6666667
+0.
+Author
+Ankur Mallick
+References
+[1] Oppenheim, Alan V., Ronald W. Schafer, and John R. Buck. Discrete-Time Signal Processing. Upper Saddle River, NJ: Prentice Hall, 1999.
+
+
+
diff --git a/help/en_US/besself.xml b/help/en_US/besself.xml
new file mode 100644
index 0000000..b792eaf
--- /dev/null
+++ b/help/en_US/besself.xml
@@ -0,0 +1,66 @@
+
+
+
+
+
+
+
+ besself
+ This function generates a Bessel filter.
+
+
+
+
+ Calling Sequence
+
+ [a, b] = besself(n, w)
+ [a, b] = besself (n, w, "high")
+ [a, b, c] = besself (…)
+ [a, b, c, d] = besself (…)
+ […] = besself (…, "z")
+
+
+
+
+ Parameters
+
+ n:
+ positive integer value
+ w:
+ positive real value
+
+
+
+
+ Description
+
+This is an Octave function.
+This function generates a Bessel filter. The default is a Laplace space (s) filter.
+The third parameter takes in high or low, the default value being low. The cutoff is pi*Wc radians.
+[z,p,g] = besself(...) returns filter as zero-pole-gain rather than coefficients of the numerator and denominator polynomials.
+[...] = besself(...,’z’) returns a discrete space (Z) filter. w must be less than 1.
+[a,b,c,d] = besself(...) returns state-space matrices.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/bitrevorder.xml b/help/en_US/bitrevorder.xml
new file mode 100644
index 0000000..996ed26
--- /dev/null
+++ b/help/en_US/bitrevorder.xml
@@ -0,0 +1,57 @@
+
+
+
+
+
+
+
+ bitrevorder
+
+
+
+
+
+ Calling Sequence
+
+ [y,i]=bitrevorder(x)
+
+
+
+
+ Parameters
+
+ x:
+ Vector of real or complex values
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the input data after reversing the bits of the indices and reordering the elements of the input array.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/blackman.xml b/help/en_US/blackman.xml
new file mode 100644
index 0000000..83bc83d
--- /dev/null
+++ b/help/en_US/blackman.xml
@@ -0,0 +1,60 @@
+
+
+
+
+
+
+
+ blackman
+ Generates a Blackman window
+
+
+
+
+ Calling Sequence
+
+ w=blackman(N)
+ w=blackman(N,sflag)
+
+
+
+
+ Parameters
+
+
+
+
+
+ Description
+
+w=blackman(N) returns an N-point symmetric Blackman window in a column vector w
+w=blackman(N,sflag)
+Returns an N point Blackman window using the type of sampling specified by sflag
+sflag can be either 'symmetric' (default) or 'periodic' (used in spectral analysis)
+Example
+w=blackman(4)
+w =
+
+
+- 1.388D-17
+0.63
+0.63
+- 1.388D-17
+Author
+Ankur Mallick
+References
+[1] Oppenheim, Alan V., Ronald W. Schafer, and John R. Buck. Discrete-Time Signal Processing. Upper Saddle River, NJ: Prentice Hall, 1999.
+
+
+
diff --git a/help/en_US/blackmanharris.xml b/help/en_US/blackmanharris.xml
new file mode 100644
index 0000000..25e09e2
--- /dev/null
+++ b/help/en_US/blackmanharris.xml
@@ -0,0 +1,65 @@
+
+
+
+
+
+
+
+ blackmanharris
+ This function returns the filter coefficients of a Blackman-Harris window.
+
+
+
+
+ Calling Sequence
+
+ w = blackmanharris (m)
+ w = blackmanharris (m, opt)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ opt:
+ string value, takes "periodic" or "symmetric"
+ w:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a Blackman-Harris window of length m supplied as input, to the output vector w.
+The second parameter can take the values "periodic" or "symmetric", depending on which the corresponding form of window is returned. The default is symmetric.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/blackmannuttall.xml b/help/en_US/blackmannuttall.xml
new file mode 100644
index 0000000..fc4fd9a
--- /dev/null
+++ b/help/en_US/blackmannuttall.xml
@@ -0,0 +1,56 @@
+
+
+
+
+
+
+
+ blackmannuttall
+ This function returns the filter coefficients of a Blackman-Nuttall window.
+
+
+
+
+ Calling Sequence
+
+ w = blackmannuttall (m)
+ w = blackmannuttall (m, opt)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ opt:
+ string value, takes "periodic" or "symmetric"
+ w:
+ output variable, vector of real numbers
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/bohmanwin.xml b/help/en_US/bohmanwin.xml
new file mode 100644
index 0000000..082342d
--- /dev/null
+++ b/help/en_US/bohmanwin.xml
@@ -0,0 +1,60 @@
+
+
+
+
+
+
+
+ bohmanwin
+ This function returns the filter coefficients of a Bohman window.
+
+
+
+
+ Calling Sequence
+
+ y = bohmanwin (m)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ y:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a Bohman window of length m supplied as input, to the output vector y.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/boxcar.xml b/help/en_US/boxcar.xml
new file mode 100644
index 0000000..ad5da0a
--- /dev/null
+++ b/help/en_US/boxcar.xml
@@ -0,0 +1,62 @@
+
+
+
+
+
+
+
+ boxcar
+ This function returns the filter coefficients of a rectangular window.
+
+
+
+
+ Calling Sequence
+
+ y = boxcar (m)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ y:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a rectangular window of length m supplied as input, to the output vector y.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/buffer.xml b/help/en_US/buffer.xml
new file mode 100644
index 0000000..24b5214
--- /dev/null
+++ b/help/en_US/buffer.xml
@@ -0,0 +1,66 @@
+
+
+
+
+
+
+
+ buffer
+ This function buffers the given data into a matrix of signal frames
+
+
+
+
+ Calling Sequence
+
+ [y] = buffer (x, n)
+ [y] = buffer (x, n, p)
+ [y] = buffer (x, n, p)
+ [y, z, opt] = buffer (...)
+
+
+
+
+ Parameters
+
+ x:
+ Data to be buffered
+ n:
+ Positive integer equal to number of rows in the produced data buffer
+ p:
+ Integer less than n, default value 0
+ opt:
+ In case of overlap, it can be a vector of length p or the string "nodelay", In case of underlap, it is an integer between 0 and p
+
+
+
+
+ Description
+
+This function buffers the given data into a matrix of signal frames
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/build_help.sce b/help/en_US/build_help.sce
new file mode 100644
index 0000000..5d051c3
--- /dev/null
+++ b/help/en_US/build_help.sce
@@ -0,0 +1,16 @@
+// Copyright (C) 2017 - IIT Bombay - FOSSEE
+//
+// Author: Shamika Mohanan
+// Organization: FOSSEE, IIT Bombay
+// Email: toolbox@scilab.in
+// This file must be used under the terms of the CeCILL.
+// This source file is licensed as described in the file COPYING, which
+// you should have received as part of this distribution. The terms
+// are also available at
+// http://www.cecill.info/licences/Licence_CeCILL_V2-en.txt
+
+help_lang_dir = get_absolute_file_path('build_help.sce');
+
+tbx_build_help(TOOLBOX_TITLE, help_lang_dir);
+
+clear help_lang_dir;
diff --git a/help/en_US/butter.xml b/help/en_US/butter.xml
new file mode 100644
index 0000000..bacacaf
--- /dev/null
+++ b/help/en_US/butter.xml
@@ -0,0 +1,71 @@
+
+
+
+
+
+
+
+ butter
+ This function generates a Butterworth filter.
+
+
+
+
+ Calling Sequence
+
+ [a, b] = butter (n, w)
+ [a, b] = butter (n, w, "high")
+ [a, b] = butter (n, [wl, wh])
+ [b, a] = butter (n, [wl, wh], "stop")
+ [a, b, c] = butter (…)
+ [a, b, c, d] = butter (…)
+ […] = butter (…, "s")
+
+
+
+
+ Parameters
+
+ n:
+ positive integer value
+ w:
+ positive real value, w in the range [0,1]
+
+
+
+
+ Description
+
+This is an Octave function.
+This function generates a Butterworth filter. Default is a discrete space (Z) filter.
+The third parameter takes in low or high, default value is low. The cutoff is pi*Wc radians.
+[b,a] = butter(n, [Wl, Wh]) indicates a band pass filter with edges pi*Wl and pi*Wh radians.
+[b,a] = butter(n, [Wl, Wh], ’stop’) indicates a band reject filter with edges pi*Wl and pi*Wh radians.
+[z,p,g] = butter(...) returns filter as zero-pole-gain rather than coefficients of the numerator and denominator polynomials.
+[...] = butter(...,’s’) returns a Laplace space filter, w can be larger than 1.
+[a,b,c,d] = butter(...) returns state-space matrices.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/buttord.xml b/help/en_US/buttord.xml
new file mode 100644
index 0000000..da92712
--- /dev/null
+++ b/help/en_US/buttord.xml
@@ -0,0 +1,70 @@
+
+
+
+
+
+
+
+ buttord
+ /This function computes the minimum filter order of a Butterworth filter with the desired response characteristics.
+
+
+
+
+ Calling Sequence
+
+ n = buttord(Wp, Ws, Rp, Rs)
+ [n, Wc] = buttord(Wp, Ws, Rp, Rs)
+
+
+
+
+ Parameters
+
+ Wp:
+ scalar or vector of length 2
+ Ws:
+ scalar or vector of length 2, elements must be in the range [0,1]
+ Rp:
+ real or complex value
+ Rs:
+ real or complex value
+
+
+
+
+ Description
+
+This is an Octave function.
+This function computes the minimum filter order of a Butterworth filter with the desired response characteristics.
+The filter frequency band edges are specified by the passband frequency wp and stopband frequency ws.
+Frequencies are normalized to the Nyquist frequency in the range [0,1].
+Rp is measured in decibels and is the allowable passband ripple, and Rs is also in decibels and is the minimum attenuation in the stop band.
+If ws>wp, the filter is a low pass filter. If wp>ws, the filter is a high pass filter.
+If wp and ws are vectors of length 2, then the passband interval is defined by wp the stopband interval is defined by ws.
+If wp is contained within the lower and upper limits of ws, the filter is a band-pass filter. If ws is contained within the lower and upper limits of wp the filter is a band-stop or band-reject filter.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/cconv.xml b/help/en_US/cconv.xml
new file mode 100644
index 0000000..5c9a665
--- /dev/null
+++ b/help/en_US/cconv.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ cconv
+ [nargout,nargin]=argn();
+
+
+
diff --git a/help/en_US/cell2sos.xml b/help/en_US/cell2sos.xml
new file mode 100644
index 0000000..90df61d
--- /dev/null
+++ b/help/en_US/cell2sos.xml
@@ -0,0 +1,100 @@
+
+
+
+
+
+
+
+ cell2sos
+ Converts a cell array to a second order section matrix
+
+
+
+
+ Parameters
+
+
+
+
+
+ Description
+
+s=cell2sos(c) converts a a cell array c = { {B1},{A1}, {B2},{A2}, ... {BL},{AL}}
+to an L-by-6 second-order-section matrix s given by:
+s = [B1 A1
+B2 A2
+...
+BL AL]
+numerator vector Bi and denominator vector Ai contains the coefficients of a
+linear or quadratic polynomial. If the polynomial is linear, the coefficients
+zero-padded on the right.
+[s,g]=cell2sos(c) estimates the gain from the leading term of the cell array
+c={ {[g1,g2]},{B1},{A1}, {B2},{A2}, ... {BL},{AL}} to give g=g1/g2 as the gain
+Example
+c=cell(1,5);
+
+
+c(1,1).entries=[2, 1];
+
+
+c(1,2).entries=rand(1,3);
+
+
+c(1,3).entries=rand(1,3);
+
+
+c(1,4).entries=rand(1,3);
+
+
+c(1,5).entries=rand(1,3);
+
+
+c =
+column 1 to 3
+
+
+![2,1] [0.2113249,0.7560439,0.0002211] [0.3303271,0.6653811,0.6283918] !
+
+
+column 4 to 5
+
+
+![0.8497452,0.6857310,0.8782165] [0.0683740,0.5608486,0.6623569] !
+[s,g]=cell2sos(c);
+s =
+
+
+column 1 to 5
+
+
+0.2113249 0.7560439 0.0002211 0.3303271 0.6653811
+0.8497452 0.6857310 0.8782165 0.0683740 0.5608486
+
+
+column 6
+
+
+0.6283918
+0.6623569
+
+
+g =
+
+
+2.
+Author
+Ankur Mallick
+
+
+
diff --git a/help/en_US/cheb1ord.xml b/help/en_US/cheb1ord.xml
new file mode 100644
index 0000000..e60a114
--- /dev/null
+++ b/help/en_US/cheb1ord.xml
@@ -0,0 +1,67 @@
+
+
+
+
+
+
+
+ cheb1ord
+ This function computes the minimum filter order of a Chebyshev type I filter with the desired response characteristics.
+
+
+
+
+ Calling Sequence
+
+ n = cheb1ord(Wp, Ws, Rp, Rs)
+ [n, Wc] = cheb1ord(Wp, Ws, Rp, Rs)
+
+
+
+
+ Parameters
+
+ Wp:
+ scalar or vector of length 2, all elements must be in the range [0,1]
+ Ws:
+ scalar or vector of length 2, all elements must be in the range [0,1]
+ Rp:
+ real value
+ Rs:
+ real value
+
+
+
+
+ Description
+
+This is an Octave function.
+This function computes the minimum filter order of a Chebyshev type I filter with the desired response characteristics.
+Stopband frequency ws and passband frequency wp specify the the filter frequency band edges.
+Frequencies are normalized to the Nyquist frequency in the range [0,1].
+Rp is measured in decibels and is the allowable passband ripple and Rs is also measured in decibels and is the minimum attenuation in the stop band.
+If ws>wp then the filter is a low pass filter. If wp>ws, then the filter is a high pass filter.
+If wp and ws are vectors of length 2, then the passband interval is defined by wp and the stopband interval is defined by ws.
+If wp is contained within the lower and upper limits of ws, the filter is a band-pass filter. If ws is contained within the lower and upper limits of wp, the filter is a band-stop or band-reject filter.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/cheb2ord.xml b/help/en_US/cheb2ord.xml
new file mode 100644
index 0000000..199c595
--- /dev/null
+++ b/help/en_US/cheb2ord.xml
@@ -0,0 +1,67 @@
+
+
+
+
+
+
+
+ cheb2ord
+ This function computes the minimum filter order of a Chebyshev type II filter with the desired response characteristics.
+
+
+
+
+ Calling Sequence
+
+ n = cheb2ord(Wp, Ws, Rp, Rs)
+ [n, Wc] = cheb2ord(Wp, Ws, Rp, Rs)
+
+
+
+
+ Parameters
+
+ Wp:
+ scalar or vector of length 2, all elements must be in the range [0,1]
+ Ws:
+ scalar or vector of length 2, all elements must be in the range [0,1]
+ Rp:
+ real value
+ Rs:
+ real value
+
+
+
+
+ Description
+
+This is an Octave function.
+This function computes the minimum filter order of a Chebyshev type II filter with the desired response characteristics.
+Stopband frequency ws and passband frequency wp specify the the filter frequency band edges.
+Frequencies are normalized to the Nyquist frequency in the range [0,1].
+Rp is measured in decibels and is the allowable passband ripple and Rs is also measured in decibels and is the minimum attenuation in the stop band.
+If ws>wp then the filter is a low pass filter. If wp>ws, then the filter is a high pass filter.
+If wp and ws are vectors of length 2, then the passband interval is defined by wp and the stopband interval is defined by ws.
+If wp is contained within the lower and upper limits of ws, the filter is a band-pass filter. If ws is contained within the lower and upper limits of wp, the filter is a band-stop or band-reject filter.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/chebwin.xml b/help/en_US/chebwin.xml
new file mode 100644
index 0000000..e901bef
--- /dev/null
+++ b/help/en_US/chebwin.xml
@@ -0,0 +1,67 @@
+
+
+
+
+
+
+
+ chebwin
+ This function returns the filter coefficients of a Dolph-Chebyshev window.
+
+
+
+
+ Calling Sequence
+
+ w = chebwin (m)
+ w = chebwin (m, at)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ at:
+ real scalar value
+ w:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a Dolph-Chebyshev window of length m supplied as input, to the output vector w.
+The second parameter is the stop band attenuation of the Fourier transform in dB. The default value is 100 dB.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/cheby1.xml b/help/en_US/cheby1.xml
new file mode 100644
index 0000000..92a7848
--- /dev/null
+++ b/help/en_US/cheby1.xml
@@ -0,0 +1,74 @@
+
+
+
+
+
+
+
+ cheby1
+ This function generates a Chebyshev type I filter with rp dB of passband ripple.
+
+
+
+
+ Calling Sequence
+
+ [a, b] = cheby1 (n, rp, w)
+ [a, b] = cheby1 (n, rp, w, "high")
+ [a, b] = cheby1 (n, rp, [wl, wh])
+ [a, b] = cheby1 (n, rp, [wl, wh], "stop")
+ [a, b, c] = cheby1 (…)
+ [a, b, c, d] = cheby1 (…)
+ […] = cheby1 (…, "s")
+
+
+
+
+ Parameters
+
+ n:
+ positive integer value
+ rp:
+ non negative scalar value
+ w:
+ vector, all elements must be in the range [0,1]
+
+
+
+
+ Description
+
+This is an Octave function.
+This function generates a Chebyshev type I filter with rp dB of passband ripple.
+The fourth parameter takes in high or low, default value is low. The cutoff is pi*Wc radians.
+[b, a] = cheby1(n, Rp, [Wl, Wh]) indicates a band pass filter with edges pi*Wl and pi*Wh radians.
+[b, a] = cheby1(n, Rp, [Wl, Wh], ’stop’) indicates a band reject filter with edges pi*Wl and pi*Wh radians.
+[z, p, g] = cheby1(...) returns filter as zero-pole-gain rather than coefficients of the numerator and denominator polynomials.
+[...] = cheby1(...,’s’) returns a Laplace space filter, w can be larger than 1.
+[a,b,c,d] = cheby1(...) returns state-space matrices.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/cheby2.xml b/help/en_US/cheby2.xml
new file mode 100644
index 0000000..24cef7d
--- /dev/null
+++ b/help/en_US/cheby2.xml
@@ -0,0 +1,74 @@
+
+
+
+
+
+
+
+ cheby2
+ This function generates a Chebyshev type II filter with rs dB of stopband attenuation.
+
+
+
+
+ Calling Sequence
+
+ [a, b] = cheby2 (n, rs, wc)
+ [a, b] = cheby2 (n, rs, wc, "high")
+ [a, b] = cheby2 (n, rs, [wl, wh])
+ [a, b] = cheby2 (n, rs, [wl, wh], "stop")
+ [a, b, c] = cheby2 (…)
+ [a, b, c, d] = cheby2 (…)
+ […] = cheby2 (…, "s")
+
+
+
+
+ Parameters
+
+ n:
+ positive integer value
+ rp:
+ non negative scalar value
+ w:
+ vector, all elements must be in the range [0,1]
+
+
+
+
+ Description
+
+This is an Octave function.
+This function generates a Chebyshev type II filter with rs dB of stopband attenuation.
+The fourth parameter takes in high or low, default value is low. The cutoff is pi*Wc radians.
+[b, a] = cheby2(n, Rp, [Wl, Wh]) indicates a band pass filter with edges pi*Wl and pi*Wh radians.
+[b, a] = cheby2(n, Rp, [Wl, Wh], ’stop’) indicates a band reject filter with edges pi*Wl and pi*Wh radians.
+[z, p, g] = cheby2(...) returns filter as zero-pole-gain rather than coefficients of the numerator and denominator polynomials.
+[...] = cheby2(...,’s’) returns a Laplace space filter, w can be larger than 1.
+[a,b,c,d] = cheby2(...) returns state-space matrices.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/check.xml b/help/en_US/check.xml
new file mode 100644
index 0000000..f8956dc
--- /dev/null
+++ b/help/en_US/check.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ check
+ funcprot(0);
+
+
+
diff --git a/help/en_US/chirp.xml b/help/en_US/chirp.xml
new file mode 100644
index 0000000..31ccb71
--- /dev/null
+++ b/help/en_US/chirp.xml
@@ -0,0 +1,75 @@
+
+
+
+
+
+
+
+ chirp
+ This function evaluates a chirp signal at time t.
+
+
+
+
+ Calling Sequence
+
+ y = chirp(t)
+ y = chirp(t, f0)
+ y = chirp(t, f0, t1)
+ y = chirp(t, f0, t1, f1)
+ y = chirp(t, f0, t1, f1, frm)
+ y = chirp(t, f0, t1, f1, frm, phse)
+
+
+
+
+ Parameters
+
+ t:
+ vector
+ f0:
+
+ t1:
+
+ f1:
+
+ frm:
+ string value, takes in "linear", "quadratic", "logarithmic"
+ phse:
+
+
+
+
+
+ Description
+
+This is an Octave function.
+This function evaluates a chirp signal at time t. A chirp signal is a frequency swept cosine wave.
+The first argument is a vector of times to evaluate the chirp signal, second argument is the frequency at t=0, third argument is time t1 and fourth argument is frequency at t1.
+The fifth argument is the form which takes in values "linear", "quadratic" and "logarithmic", the sixth argument gives the phase shift at t=0.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/clustersegment.xml b/help/en_US/clustersegment.xml
new file mode 100644
index 0000000..d3ab0ee
--- /dev/null
+++ b/help/en_US/clustersegment.xml
@@ -0,0 +1,60 @@
+
+
+
+
+
+
+
+ clustersegment
+ This function calculates boundary indexes of clusters of 1’s.
+
+
+
+
+ Calling Sequence
+
+ c = clustersegment(s)
+
+
+
+
+ Parameters
+
+ s:
+ scalar, vector or matrix of real numbers (clusters of 1s)
+ c:
+ output variable, cell array of size 1 by N, where N is the number of rows in s
+
+
+
+
+ Description
+
+This is an Octave function.
+This function calculates boundary indexes of clusters of 1’s.
+This function calculates the initial and end indices of the sequences of 1's present in the input argument.
+The output variable c is a cell array of size 1 by N, where N is the number of rows in s and each element has two rows indicating the initial index and end index of the cluster of 1's respectively. The indexing starts from 1.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/cmorwavf.xml b/help/en_US/cmorwavf.xml
new file mode 100644
index 0000000..aa03ba8
--- /dev/null
+++ b/help/en_US/cmorwavf.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ cmorwavf
+ funcprot(0);
+
+
+
diff --git a/help/en_US/convmtx.xml b/help/en_US/convmtx.xml
new file mode 100644
index 0000000..80ff071
--- /dev/null
+++ b/help/en_US/convmtx.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ convmtx
+ n=double(n);
+
+
+
diff --git a/help/en_US/corrmtx.xml b/help/en_US/corrmtx.xml
new file mode 100644
index 0000000..2d07fcb
--- /dev/null
+++ b/help/en_US/corrmtx.xml
@@ -0,0 +1,129 @@
+
+
+
+
+
+
+
+ corrmtx
+ Generate data matrix for autocorrelation matrix estimation
+
+
+
+
+ Calling Sequence
+
+ X = corrmtx(x,m)
+ [X,R] = corrmtx(x,m)
+ X = corrmtx(x,m,s)
+ [X,R] = corrmtx(x,m,s)
+
+
+
+
+
+ Parameters
+
+ x:
+ input vector of size N for which correlation matrix of size m is to be calculated
+ m:
+ size of correlation matrix to be computed. Positive integer strictly smaller than the length of the input x
+ X:
+ data matrix as specified according to the input 'method'
+ s:
+ method for type of output matrix X
+ 'autocorrelation':
+ (default) X is the (n + m)-by-(m + 1) rectangular Toeplitz matrix that generates an autocorrelation estimate for the leng th-n data vector x, derived using prewindowed and postwindowed data, based on an mth-order prediction error model.
+ 'prewindowed':
+ X is the n-by-(m + 1) rectangular Toeplitz matrix that generates an autocorrelation estimate for the length-n data vector x, derived using prewindowed data, based on an mth-order prediction error model.
+ 'postwindowed':
+ X is the n-by-(m + 1) rectangular Toeplitz matrix that generates an autocorrelation estimate for the length-n data vector x , derived using postwindowed data, based on an mth-order prediction error model.
+ 'covariance':
+ X is the (n – m)-by-(m + 1) rectangular Toeplitz matrix that generates an autocorrelation estimate for the length-n data vect or x, derived using nonwindowed data, based on an mth-order prediction error model.
+ 'modified':
+ X is the 2(n – m)-by-(m + 1) modified rectangular Toeplitz matrix that generates an autocorrelation estimate for the length-n d ata vector x, derived using forward and backward prediction error estimates, based on an mth-order prediction error model.
+ R:
+ (m + 1)-by-(m + 1) autocorrelation matrix estimate calculated as X'*X
+
+
+
+
+ Description
+
+Consider the generic matrix X below
+_ _
+| x(1) ..........0 |
+| : . : |
+| : . : |
+| x(m+1).......x(1)|
+| : . : |
+| : . : |
+X = | x(n-m).....x(m+1)|
+| : . : |
+| : . : |
+| x(n).......x(n-m)|
+| : . : |
+| : . : |
+|_0 ..........x(n)_|
+--
+For different inputs of string s the output would vary ass described below
+'autocorrelation' — (default) X = X, above.
+'prewindowed' — X is the n-by-(m + 1) submatrix of X whose first row is [x(1) … 0] and whose last row is [x(n) … x(n – m)]
+'postwindowed' — X is the n-by-(m + 1) submatrix of X whose first row is [x(m + 1) … x(1)] and whose last row is [0 … x(n)]
+'covariance' — X is the (n – m)-by-(m + 1) submatrix of X whose first row is [x(m + 1) … x(1)] and whose last row is [x(n) … x(n – m)]
+'modified' — X is the 2(n – m)-by-(m + 1) matrix X_mod shown below
+_ _
+| x(m+1) ......x(1)|
+| : . : |
+| : . : |
+| x(n-m).....x(m+1)|
+| : . : |
+| : . : |
+| x(n).......x(n-m)|
+X_mod= | x*(1).....x*(m+1)|
+| : . : |
+| : . : |
+| x*(m+1)...x*(n-m)|
+| : . : |
+| : . : |
+|_x*(n-m) ...x*(n)_|
+
+
+
+
+
+
+ Examples
+
+
+
+
+ See also
+
+ peig
+ pmusic
+ rooteig
+ rootmusic
+ xcorr
+ Author:
+ Pandit
+ Bilbligraphy
+ S. Lawrence. Digital Spectral Analysis. Englewood Cliffs, NJ: Prentice-Hall, 1987.
+
+
+
diff --git a/help/en_US/cummax.xml b/help/en_US/cummax.xml
new file mode 100644
index 0000000..76c863a
--- /dev/null
+++ b/help/en_US/cummax.xml
@@ -0,0 +1,70 @@
+
+
+
+
+
+
+
+ cummax
+ Cumulative maximum
+
+
+
+
+ Calling Sequence
+
+ M = cummax(A)
+ returns the cumulative maximum of the arguments of A. The dimension
+ of M is same as the dimension of A. If A is a 2D matrix, the operation
+ is performed along the columns. For a hypermatrix, the operation is
+ performed along the first non-zero dimension
+ M = cummax(A,dim)
+ The operation is performed along the dimension specified by dim
+ M = cummax(_,direction)
+ direction specifies as the direction of operation
+
+
+
+
+
+ Parameters
+
+
+
+
+
+ Examples
+
+
+
+
+ Authors
+
+ Ayush Baid
+
+
+
+
+ See also
+
+
+
+
diff --git a/help/en_US/cummin.xml b/help/en_US/cummin.xml
new file mode 100644
index 0000000..e586d3d
--- /dev/null
+++ b/help/en_US/cummin.xml
@@ -0,0 +1,70 @@
+
+
+
+
+
+
+
+ cummin
+ Cumulative minimum
+
+
+
+
+ Calling Sequence
+
+ M = cummin(A)
+ returns the cumulative minimum of the arguments of A. The dimension
+ of M is same as the dimension of A. If A is a 2D matrix, the operation
+ is performed along the columns. For a hypermatrix, the operation is
+ performed along the first non-zero dimension
+ M = cummin(A,dim)
+ The operation is performed along the dimension specified by dim
+ M = cummin(_,direction)
+ direction specifies as the direction of operation
+
+
+
+
+
+ Parameters
+
+
+
+
+
+ Examples
+
+
+
+
+ Authors
+
+ Ayush Baid
+
+
+
+
+ See also
+
+
+
+
diff --git a/help/en_US/db.xml b/help/en_US/db.xml
new file mode 100644
index 0000000..5066562
--- /dev/null
+++ b/help/en_US/db.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ db
+
+
+
+
diff --git a/help/en_US/db2pow.xml b/help/en_US/db2pow.xml
new file mode 100644
index 0000000..2f126fa
--- /dev/null
+++ b/help/en_US/db2pow.xml
@@ -0,0 +1,53 @@
+
+
+
+
+
+
+
+ db2pow
+
+
+
+
+
+ Calling Sequence
+
+
+
+
+
+ Parameters
+
+ ydb :
+ scalar or vector or matrix or N-D array
+
+
+
+
+ Examples
+
+
+
+
+ Authors
+
+ Ishita Bedi
+ Modified to handle char i/p by Debdeep Dey
+
+
+
diff --git a/help/en_US/dctmtx.xml b/help/en_US/dctmtx.xml
new file mode 100644
index 0000000..0b05453
--- /dev/null
+++ b/help/en_US/dctmtx.xml
@@ -0,0 +1,58 @@
+
+
+
+
+
+
+
+ dctmtx
+
+
+
+
+
+ Calling Sequence
+
+ [y]=dctmtx(n)
+
+
+
+
+ Parameters
+
+ n:
+ Real scalar integer greater than or equal to 1
+
+
+
+
+ Description
+
+This is an Octave function
+dctmtx(n) returns a Discrete cosine transform matrix of order n-by-n. It is useful for jpeg image compression. D*A is the DCT of the columns of A and D'*A is the inverse DCT of the columns of A (when A is n-by-n).
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/decimate.xml b/help/en_US/decimate.xml
new file mode 100644
index 0000000..bc0efe7
--- /dev/null
+++ b/help/en_US/decimate.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ decimate
+ rhs = argn(2)
+
+
+
diff --git a/help/en_US/dftmtx.xml b/help/en_US/dftmtx.xml
new file mode 100644
index 0000000..ed6ecf7
--- /dev/null
+++ b/help/en_US/dftmtx.xml
@@ -0,0 +1,56 @@
+
+
+
+
+
+
+
+ dftmtx
+
+
+
+
+
+ Calling Sequence
+
+ [d]=dftmtx(n)
+
+
+
+
+ Parameters
+
+ n:
+ Real positive scalar number
+
+
+
+
+ Description
+
+This is an Octave function
+This fuction gives a complex matrix of values whose product with a vector produces the discrete Fourier transform. This can also be achieved by directly using the fft function i.e. y=fft(x) is same as y=A*x where A=dftmtx(n).
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/diric.xml b/help/en_US/diric.xml
new file mode 100644
index 0000000..fdbc3fa
--- /dev/null
+++ b/help/en_US/diric.xml
@@ -0,0 +1,58 @@
+
+
+
+
+
+
+
+ diric
+
+
+
+
+
+ Calling Sequence
+
+ [y]=diric(x,n)
+
+
+
+
+ Parameters
+
+ x:
+ Real valued vector or matrix
+ n:
+ Real positive integer or complex integer
+
+
+
+
+ Description
+
+This is an Octave function
+y=diric(x,n) returns the dirichlet function values of parameter x.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/downsample.xml b/help/en_US/downsample.xml
new file mode 100644
index 0000000..bbdfcd0
--- /dev/null
+++ b/help/en_US/downsample.xml
@@ -0,0 +1,61 @@
+
+
+
+
+
+
+
+ downsample
+ This function downsamples the signal by selecting every nth element.
+
+
+
+
+ Calling Sequence
+
+ y = downsample (x, n)
+ y = downsample (x, n, phase)
+
+
+
+
+ Parameters
+
+ x:
+ scalar, vector or matrix of real or complex numbers
+ n:
+ real number or vector
+ phase:
+ integer value, 0 <= phase <= (n - 1), default value 0, or logical
+
+
+
+
+ Description
+
+This is an Octave function.
+This function downsamples the signal by selecting every nth element supplied as parameter 2. If x is a matrix, the function downsamples every column.
+If the phase is specified, every nth element is selected starting from the sample phase. The default phase is 0.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/dutycycle.xml b/help/en_US/dutycycle.xml
new file mode 100644
index 0000000..0fb782a
--- /dev/null
+++ b/help/en_US/dutycycle.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ dutycycle
+
+
+
+
diff --git a/help/en_US/ellip.xml b/help/en_US/ellip.xml
new file mode 100644
index 0000000..847f91e
--- /dev/null
+++ b/help/en_US/ellip.xml
@@ -0,0 +1,74 @@
+
+
+
+
+
+
+
+ ellip
+ This function generates an elliptic or Cauer filter with rp dB of passband ripple and rs dB of stopband attenuation.
+
+
+
+
+ Calling Sequence
+
+ [a, b] = ellip (n, rp, rs, wp)
+ [a, b] = ellip (n, rp, rs, wp, "high")
+ [a, b] = ellip (n, rp, rs, [wl, wh])
+ [a, b] = ellip (n, rp, rs, [wl, wh], "stop")
+ [a, b, c] = ellip (…)
+ [a, b, c, d] = ellip (…)
+ […] = ellip (…, "s")
+
+
+
+
+ Parameters
+
+ n:
+ positive integer value
+ rp:
+ non negative scalar value
+ rs:
+ non negative scalar value
+ w:
+ scalar or vector, all elements should be in the range [0,1]
+
+
+
+
+ Description
+
+This is an Octave function.
+This function generates an elliptic or Cauer filter with rp dB of passband ripple and rs dB of stopband attenuation.
+[b, a] = ellip(n, Rp, Rs, Wp) indicates low pass filter with order n, Rp decibels of ripple in the passband and a stopband Rs decibels down and cutoff of pi*Wp radians. If the fifth argument is high, then the filter is a high pass filter.
+[b, a] = ellip(n, Rp, Rs, [Wl, Wh]) indictaes band pass filter with band pass edges pi*Wl and pi*Wh. If the fifth argument is stop, the filter is a band reject filter.
+[z, p, g] = ellip(...) returns filter as zero-pole-gain.
+[...] = ellip(...,’s’) returns a Laplace space filter, w can be larger than 1.
+[a, b, c, d] = ellip(...) returns state-space matrices.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/ellipord.xml b/help/en_US/ellipord.xml
new file mode 100644
index 0000000..0f562c8
--- /dev/null
+++ b/help/en_US/ellipord.xml
@@ -0,0 +1,68 @@
+
+
+
+
+
+
+
+ ellipord
+ This function computes the minimum filter order of an elliptic filter with the desired response characteristics.
+
+
+
+
+ Calling Sequence
+
+ [n] = ellipord(Wp, Ws, Rp, Rs)
+ [n, Wp] = ellipord(Wp, Ws, Rp, Rs)
+
+
+
+
+ Parameters
+
+ Wp:
+ scalar or vector of length 2, all elements must be in the range [0,1]
+ Ws:
+ scalar or vector of length 2, all elements must be in the range [0,1]
+ Rp:
+ real or complex value
+ Rs:
+ real or complex value
+
+
+
+
+ Description
+
+This is an Octave function.
+This function computes the minimum filter order of an elliptic filter with the desired response characteristics.
+Stopband frequency ws and passband frequency wp specify the the filter frequency band edges.
+Frequencies are normalized to the Nyquist frequency in the range [0,1].
+Rp is measured in decibels and is the allowable passband ripple and Rs is also measured in decibels and is the minimum attenuation in the stop band.
+If ws>wp then the filter is a low pass filter. If wp>ws, then the filter is a high pass filter.
+If wp and ws are vectors of length 2, then the passband interval is defined by wp and the stopband interval is defined by ws.
+If wp is contained within the lower and upper limits of ws, the filter is a band-pass filter. If ws is contained within the lower and upper limits of wp, the filter is a band-stop or band-reject filter.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/enbw.xml b/help/en_US/enbw.xml
new file mode 100644
index 0000000..d7c2972
--- /dev/null
+++ b/help/en_US/enbw.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ enbw
+
+
+
+
diff --git a/help/en_US/eqtflength.xml b/help/en_US/eqtflength.xml
new file mode 100644
index 0000000..cfb2823
--- /dev/null
+++ b/help/en_US/eqtflength.xml
@@ -0,0 +1,31 @@
+
+
+
+
+
+
+
+ eqtflength
+ Modifies the input vector to give output vectors of the same length
+
+
+
+
+ Calling Sequence
+
+ [b,a] = eqtflength(b,a)
+ [b,a,N,M] = eqtflength(b,a)
+
+
+
diff --git a/help/en_US/falltime.xml b/help/en_US/falltime.xml
new file mode 100644
index 0000000..eb398e4
--- /dev/null
+++ b/help/en_US/falltime.xml
@@ -0,0 +1,73 @@
+
+
+
+
+
+
+
+ falltime
+
+
+
+
+
+ Calling Sequence
+
+ r=falltime(x)
+ r=falltime(x, t)
+ r=falltime(x, Fs)
+ r=falltime(x, t, 'PercentReferenceLevels', N )
+ r=falltime(x, t, 'Tolerance', M)
+ r=falltime(x, t,'StateLevels', O)
+ [r lowercrossvalue uppercrossvalue lowerreference upperreference]=falltime(x)
+ [r lowercrossvalue uppercrossvalue lowerreference upperreference]=falltime(x, Fs)
+ [r lowercrossvalue uppercrossvalue lowerreference upperreference]=falltime(x, t)
+ [r lowercrossvalue uppercrossvalue lowerreference upperreference]=falltime(x, t, 'PercentReferenceLevels', N )
+ [r lowercrossvalue uppercrossvalue lowerreference upperreference]= falltime(x, t, 'Tolerance', M)
+ [r lowercrossvalue uppercrossvalue lowerreference upperreference]= falltime(x, t,'StateLevels', O)
+ [r lowercrossvalue uppercrossvalue lowerreference upperreference]= falltime(x, t,'StateLevels', O, 'fig', on or off)
+
+
+
+
+
+ Parameters
+
+ x:
+ real vector.
+ Fs:
+ specifies the sample rate, Fs, as a positive scalar, where the first sample instant corresponds to a time of zero.
+ t:
+ defiene instant sample time t as vector with same length of x, or specifies the sample rate, t, as a positive scalar.
+ PercentReferenceLevels:
+ specify the percentreferenceleves as a percentage, default value of N is [10 90].
+ Tolerance:
+ define the tolerance value as real scaler value, where default value of M is 2.0.
+ StateLevels:
+ define the lower and upper state levels as two element real vector.
+ fig:
+ specify the logical input value to display figure as one of 'on' or 'off', where the default input in 'off'.
+ f:
+ return fall time of negative-going bilevel waveform transitions X.
+ lowercrossvalue:
+ return the lowerc cross value.
+ uppercrossvalue:
+ return the upper cross value.
+ lowerreference:
+ return lower reference value corrosponding to lower percent reference value.
+ upperreference:
+ return upper reference value corrosponding to upper percent reference value.
+
+
+
diff --git a/help/en_US/fftfilt.xml b/help/en_US/fftfilt.xml
new file mode 100644
index 0000000..868244f
--- /dev/null
+++ b/help/en_US/fftfilt.xml
@@ -0,0 +1,74 @@
+
+
+
+
+
+
+
+ fftfilt
+ Performs FFT-based FIR filtering using overlap-add method
+
+
+
+
+ Calling Sequence
+
+
+
+
+
+ Parameters
+
+ x:
+ real|complex numbers - vector|matrix
+ b:
+ real|complex numbers - vector|matrix
+ n:
+ positive integer
+
+
+
+
+ Description
+
+y = fftfilt(b,x) filters the data in vector x with the filter described
+by coefficient vector b.
+y = fftfilt(b,x,n) uses n to determine the length of the FFT.
+
+
+
+
+
+
+ Examples
+
+
+
+
+ Authors
+
+ Ayush Baid
+
+
+
diff --git a/help/en_US/filternorm.xml b/help/en_US/filternorm.xml
new file mode 100644
index 0000000..7e897ba
--- /dev/null
+++ b/help/en_US/filternorm.xml
@@ -0,0 +1,75 @@
+
+
+
+
+
+
+
+ filternorm
+ Calculates the L-2 norm or L-infinity norm of a digital filter
+
+
+
+
+ Calling Sequence
+
+ L = filternorm(b,a)
+ L = filternorm(b,a,pnorm)
+ L = filternorm(b,a,2,tol)
+
+
+
+
+
+
+ Parameters
+
+ b:
+ The filter numerator coefficients.
+ a:
+ The filter denominator coefficients.
+ pnorm:
+ The L-norm to be calculated. The values accepted are 2 (L2 norm) or %inf (L-infinity norm). Default value is 2.
+ tol:
+ The tolerance of the L-2 norm to be calculated. If tol not specified, it defaults to 10^(-8). tol must be a positive scalar
+
+
+
+
+ Examples
+
+
+
+
+ See also
+
+ norm
+ zp2sos
+
+
+
+
+ Authors
+
+ Ayush Baid
+
+
+
diff --git a/help/en_US/filtfilt.xml b/help/en_US/filtfilt.xml
new file mode 100644
index 0000000..baf8dfe
--- /dev/null
+++ b/help/en_US/filtfilt.xml
@@ -0,0 +1,58 @@
+
+
+
+
+
+
+
+ filtfilt
+
+
+
+
+
+ Calling Sequence
+
+ [y]=filtfilt(b,a,x)
+
+
+
+
+ Parameters
+
+ b:
+ Real or complex valued vector or matrix
+ a:
+ Real or complex valued vector or matrix
+ x:
+ Real or complex valued vector or matrix
+
+
+
+
+ Description
+
+This is an Octave function
+In theory, it forwards and reverse filters the signal and corrects phase distortion upto an extent by a one-pass filter but squares the magnitude response in the process. Practically though, the correction isn't perfect and magnitude response, particularly the stop band is distorted.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/filtic.xml b/help/en_US/filtic.xml
new file mode 100644
index 0000000..ec5e08e
--- /dev/null
+++ b/help/en_US/filtic.xml
@@ -0,0 +1,65 @@
+
+
+
+
+
+
+
+ filtic
+
+
+
+
+
+ Calling Sequence
+
+ zf = filtic (b, a, y)
+ zf = filtic (b, a, y, x)
+
+
+
+
+ Parameters
+
+ b:
+ vector of real or complex numbers
+ a:
+ vector of real or complex numbers
+ y:
+ vector of real or complex numbers
+ x:
+ vector of real or complex numbers
+
+
+
+
+ Description
+
+This function finds the initial conditions for the delays in the transposed direct-form II filter implementation.
+The vectors b and a represent the numerator and denominator coefficients of the filter's transfer function.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/filtord.xml b/help/en_US/filtord.xml
new file mode 100644
index 0000000..aa8037c
--- /dev/null
+++ b/help/en_US/filtord.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ filtord
+ and denominator coefficients, a.
+
+
+
diff --git a/help/en_US/firpmord.xml b/help/en_US/firpmord.xml
new file mode 100644
index 0000000..f4bd81a
--- /dev/null
+++ b/help/en_US/firpmord.xml
@@ -0,0 +1,91 @@
+
+
+
+
+
+
+
+ firpmord
+ Parks-McClennan optimal FIR filter order estimation
+
+
+
+
+ Calling Sequence
+
+
+
+
+
+ Parameters
+
+ f:
+ double - positive - vector
+ a:
+ double - positive - vector
+ dev:
+ double - positive - vector
+ n:
+ int - scalar
+ fo:
+ double - positive - vector
+ ao:
+ double - positive - vector
+ w:
+ double - vector
+
+
+
+
+ Examples
+
+
+
+
+ Authors
+
+ Ayush Baid
+
+
+
+
+ See also
+
+
+
+
diff --git a/help/en_US/firtype.xml b/help/en_US/firtype.xml
new file mode 100644
index 0000000..8050565
--- /dev/null
+++ b/help/en_US/firtype.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ firtype
+ if (type(b)~=1) then
+
+
+
diff --git a/help/en_US/flattopwin.xml b/help/en_US/flattopwin.xml
new file mode 100644
index 0000000..d990123
--- /dev/null
+++ b/help/en_US/flattopwin.xml
@@ -0,0 +1,69 @@
+
+
+
+
+
+
+
+ flattopwin
+ This function returns the filter coefficients of a Flat Top window.
+
+
+
+
+ Calling Sequence
+
+ w = flattopwin (m)
+ w = flattopwin (m, opt)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ opt:
+ string value, takes in "periodic" or "symmetric"
+ w:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a Flat Top window of length m supplied as input, to the output vector w.
+The second parameter can take the values "periodic" or "symmetric", depending on which the corresponding form of window is returned. The default is symmetric.
+This window has low pass-band ripple but a high bandwidth.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/fracshift.xml b/help/en_US/fracshift.xml
new file mode 100644
index 0000000..8825f71
--- /dev/null
+++ b/help/en_US/fracshift.xml
@@ -0,0 +1,62 @@
+
+
+
+
+
+
+
+ fracshift
+ This function shifts the series x supplied as input argument by a number of samples d.
+
+
+
+
+ Calling Sequence
+
+ y = fracshift(x, d)
+ y = fracshift(x, d, h)
+ [y, h] = fracshift(...)
+
+
+
+
+ Parameters
+
+ x:
+
+ d:
+
+ h:
+
+
+
+
+
+ Description
+
+This is an Octave function.
+This function shifts the series x supplied as input argument by a number of samples d.
+The third parameter is the interpolator, which is designed with a Kaiser-windowed sinecard by default, if not supplied.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/fwhm.xml b/help/en_US/fwhm.xml
new file mode 100644
index 0000000..dad3e8a
--- /dev/null
+++ b/help/en_US/fwhm.xml
@@ -0,0 +1,43 @@
+
+
+
+
+
+
+
+ fwhm
+ This function computes peak full width at half minimum or at another level of peak minimum for vector or matrix data y supplied as input.
+
+
+
+
+ Calling Sequence
+
+ f = fwhm (y)
+ f = fwhm (x, y)
+ f = fwhm (…, "zero")
+ f = fwhm (…, "min")
+ f = fwhm (…, "alevel", level)
+ f = fwhm (…, "rlevel", level)
+
+
+
+
+ Parameters
+
+ y:
+ vector or matrix
+
+
+
diff --git a/help/en_US/fwhmjlt.xml b/help/en_US/fwhmjlt.xml
new file mode 100644
index 0000000..4663d1a
--- /dev/null
+++ b/help/en_US/fwhmjlt.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ fwhmjlt
+ rhs = argn(2)
+
+
+
diff --git a/help/en_US/gauspuls.xml b/help/en_US/gauspuls.xml
new file mode 100644
index 0000000..e24d9e4
--- /dev/null
+++ b/help/en_US/gauspuls.xml
@@ -0,0 +1,62 @@
+
+
+
+
+
+
+
+ gauspuls
+
+
+
+
+
+ Calling Sequence
+
+ [y]=gauspuls(t,fc,bw)
+ [y]=gauspuls(t,fc)
+ [y]=gauspuls(t)
+
+
+
+
+ Parameters
+
+ t:
+ Real or complex valued vector or matrix
+ fc:
+ Real non negative number or complex number
+ bw:
+ Real positive number or complex number
+
+
+
+
+ Description
+
+This is an Octave function
+This function returns a Gaussian RF pulse of unity amplitude at the times indicated in array t.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/gaussdesign.xml b/help/en_US/gaussdesign.xml
new file mode 100644
index 0000000..2b81543
--- /dev/null
+++ b/help/en_US/gaussdesign.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ gaussdesign
+ GAUSSDESIGN designs a Gaussian pulse-shaping filter which is a low pass FIR
+
+
+
diff --git a/help/en_US/gaussian.xml b/help/en_US/gaussian.xml
new file mode 100644
index 0000000..5a0ab2e
--- /dev/null
+++ b/help/en_US/gaussian.xml
@@ -0,0 +1,65 @@
+
+
+
+
+
+
+
+ gaussian
+ This function returns a Gaussian convolution window.
+
+
+
+
+ Calling Sequence
+
+ w = gaussian (m)
+ w = gaussian (m, a)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ a:
+
+ w:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns a Gaussian convolution window of length m supplied as input, to the output vector w.
+The second parameter is the width measured in sample rate/number of samples and should be f for time domain and 1/f for frequency domain. The width is inversely proportional to a.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/gausswin.xml b/help/en_US/gausswin.xml
new file mode 100644
index 0000000..9d1af4a
--- /dev/null
+++ b/help/en_US/gausswin.xml
@@ -0,0 +1,63 @@
+
+
+
+
+
+
+
+ gausswin
+ This function returns the filter coefficients of a Gaussian window.
+
+
+
+
+ Calling Sequence
+
+ w = gausswin (m)
+ w = gausswin (m, a)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ a:
+
+ w:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a Gaussian window of length m supplied as input, to the output vector w.
+The second parameter is the width measured in sample rate/number of samples and should be f for time domain and 1/f for frequency domain. The width is inversely proportional to a.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/gmonopuls.xml b/help/en_US/gmonopuls.xml
new file mode 100644
index 0000000..72436d9
--- /dev/null
+++ b/help/en_US/gmonopuls.xml
@@ -0,0 +1,51 @@
+
+
+
+
+
+
+
+ gmonopuls
+
+
+
+
+
+ Parameters
+
+ t:
+ Real or complex valued vector or matrix
+ fc:
+ Real non-negative value or complex value or a vector or matrix with not all real values negative.
+
+
+
+
+ Description
+
+This is an Octave function
+This function returns samples of the Gaussian monopulse of amplitude unity.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/goertzel.xml b/help/en_US/goertzel.xml
new file mode 100644
index 0000000..95772a3
--- /dev/null
+++ b/help/en_US/goertzel.xml
@@ -0,0 +1,67 @@
+
+
+
+
+
+
+
+ goertzel
+ Computes DFT using the second order Goertzel Algorithm
+
+
+
+
+ Calling Sequence
+
+ Y = goertzel(X,INDVEC,DIM)
+
+
+
+
+ Parameters
+
+
+
+
+
+ Description
+
+goertzel(X,INDVEC)
+Computes the DFT of X at indices INDVEC using the second order algorithm along
+the first non-singleton dimension. Elements of INDVEC must be positive integers
+less than the length of the first non-singleton dimension. If INDVEC is empty
+the DFT is computed at all indices along the first non-singleton dimension
+goertzel(X,INDVEC,DIM)
+Implements the algorithm along dimension DIM
+In general goertzel is slower than fft when computing the DFT for all indices
+along a particular dimension. However it is computationally more efficient when
+the DFT at only a subset of indices is desired
+Example
+x=rand(1,5)
+x =
+
+
+0.6283918 0.8497452 0.6857310 0.8782165 0.0683740
+y=goertzel(x,2);
+y =
+
+
+- 0.3531539 - 0.6299881i
+Author
+Ankur Mallick
+References
+Goertzel, G. (January 1958), "An Algorithm for the Evaluation of Finite Trigonometric Series", American Mathematical Monthly 65 (1): 34–35, doi:10.2307/2310304
+
+
+
diff --git a/help/en_US/grpdelay.xml b/help/en_US/grpdelay.xml
new file mode 100644
index 0000000..dba4587
--- /dev/null
+++ b/help/en_US/grpdelay.xml
@@ -0,0 +1,35 @@
+
+
+
+
+
+
+
+ grpdelay
+ This function computes the group delay of a filter.
+
+
+
+
+ Calling Sequence
+
+ gd = grpdelay(b)
+ gd = grpdelay(b, a)
+ gd = grpdelay(b, a, nfft)
+ gd = grpdelay(b, a, nfft, whole)
+ gd = grpdelay(b, a, nfft, whole, Fs)
+ [gd, w] = grpdelay(...)
+
+
+
diff --git a/help/en_US/hann.xml b/help/en_US/hann.xml
new file mode 100644
index 0000000..2387cce
--- /dev/null
+++ b/help/en_US/hann.xml
@@ -0,0 +1,67 @@
+
+
+
+
+
+
+
+ hann
+ This function returns the filter coefficients of a Hanning window.
+
+
+
+
+ Calling Sequence
+
+ w = hann(m)
+ w = hann(m, "symmteric")
+ w = hann(m, "periodic")
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ opt:
+ string value, takes in "periodic" or "symmetric"
+ w:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a Hanning window of length m supplied as input, to the output vector w.
+The second parameter can take the values "periodic" or "symmetric", depending on which the corresponding form of window is returned. The default is symmetric.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/helperHarmonicDistortionAmplifier.xml b/help/en_US/helperHarmonicDistortionAmplifier.xml
new file mode 100644
index 0000000..c58cc51
--- /dev/null
+++ b/help/en_US/helperHarmonicDistortionAmplifier.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ helperHarmonicDistortionAmplifier
+ helperHarmonicDistortionADC Helper function for HarmonicDistortionExample.m
+
+
+
diff --git a/help/en_US/icceps.xml b/help/en_US/icceps.xml
new file mode 100644
index 0000000..bd10adc
--- /dev/null
+++ b/help/en_US/icceps.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ icceps
+ ICCEPS computes the inverse cepstrum of a real-valued input. This spectrum
+
+
+
diff --git a/help/en_US/iirlp2mb.xml b/help/en_US/iirlp2mb.xml
new file mode 100644
index 0000000..5cf6819
--- /dev/null
+++ b/help/en_US/iirlp2mb.xml
@@ -0,0 +1,67 @@
+
+
+
+
+
+
+
+ iirlp2mb
+ This function does IIR Low Pass Filter to Multiband Filter Transformation.
+
+
+
+
+ Calling Sequence
+
+ [Num, Den, AllpassNum, AllpassDen] = iirlp2mb(B, A, Wo, Wt)
+ [Num, Den, AllpassNum, AllpassDen] = iirlp2mb(B, A, Wo, Wt, Pass)
+
+
+
+
+ Parameters
+
+ B:
+ real or complex value
+ A:
+ real or complex value
+ Wo:
+ scalar or vector
+ Wt:
+ scalar or vector, elements must be monotonically increasing and >= 0 and <= 1.
+
+
+
+
+ Description
+
+This is an Octave function.
+This function does IIR Low Pass Filter to Multiband Filter Transformation.
+The first two parameters give the numerator and denominator of the prototype low pass filter.
+The third parameter is the normalized angular frequency/pi to be transformed.
+The fourth parameter is the normalized angular frequency/pi target vector.
+The first two output variables are the numerator and denominator of the transformed filter.
+The third and fourth output variables are the numerator and denominator of the allpass transform.
+The fifth parameter can have values pass or stop, default value is pass.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/impinvar.xml b/help/en_US/impinvar.xml
new file mode 100644
index 0000000..446586c
--- /dev/null
+++ b/help/en_US/impinvar.xml
@@ -0,0 +1,68 @@
+
+
+
+
+
+
+
+ impinvar
+ This function converts analog filter with coefficients b and a to digital, conserving impulse response.
+
+
+
+
+ Calling Sequence
+
+ [b, a] = impinvar (b, a)
+ [b, a] = impinvar (b, a, fs)
+ [b, a] = impinvar (b, a, fs, tol)
+
+
+
+
+ Parameters
+
+ b:
+ real or complex valued scalar or vector
+ a:
+ real or complex valued scalar or vector, order should be greater than b
+ fs:
+ real or complex value, default value 1Hz
+ tol:
+ real or complex value, default value 0.0001
+
+
+
+
+ Description
+
+This is an Octave function.
+This function converts analog filter with coefficients b and a to digital, conserving impulse response.
+This function does the inverse of impinvar.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/impz.xml b/help/en_US/impz.xml
new file mode 100644
index 0000000..3cde46f
--- /dev/null
+++ b/help/en_US/impz.xml
@@ -0,0 +1,34 @@
+
+
+
+
+
+
+
+ impz
+
+
+
+
+
+ Calling Sequence
+
+ x_r = impz(b)
+ x_r = impz(b, a)
+ x_r = impz(b, a, n)
+ x_r = impz(b, a, n, fs)
+ [x_r, t_r] = impz(b, a, n, fs)
+
+
+
diff --git a/help/en_US/impzlength.xml b/help/en_US/impzlength.xml
new file mode 100644
index 0000000..1b5adc8
--- /dev/null
+++ b/help/en_US/impzlength.xml
@@ -0,0 +1,76 @@
+
+
+
+
+
+
+
+ impzlength
+ Impulse response length
+
+
+
+
+ Calling Sequence
+
+ len = impzlength(b, a, tol)
+ returns the impulse response length for the causal discrete-time filter
+ with the transfer function coefficients for numerator and denominator in
+ a and b respectively. For stable IIR filters, len is the effective length
+ impulse response length, i.e. the length after which the response is
+ essentially zero
+ len = impzlength(sos)
+ returns the impulse response length of the filter specified by second
+ order sections matrix. sos is a Kx6 matrix, where K is the number of
+ sections. Each row of the sos matrix corresponds to a second order
+ biquad filter
+ len = impzlength(__, tol)
+ specifies a tolerance for estimating the effective impulse response
+ length in case of an IIR filter. By default, tol is 5e-5. Increasing tol
+ leads to shorter len and vice-versa
+
+
+
+
+
+ Parameters
+
+
+
+
+
+ Examples
+
+
+
+
+ See also
+
+ | digitalFilter | impz | zp2sos
+
+
+
+
+ Authors
+
+ Ayush Baid
+
+
+
diff --git a/help/en_US/interp.xml b/help/en_US/interp.xml
new file mode 100644
index 0000000..ec22cbc
--- /dev/null
+++ b/help/en_US/interp.xml
@@ -0,0 +1,64 @@
+
+
+
+
+
+
+
+ interp
+ function y = interp(x, q, n, Wc)
+
+
+
+
+ Calling Sequence
+
+ y = interp(x, q)
+ y = interp(x, q, n)
+ y = interp(x, q, n, Wc)
+
+
+
+
+ Parameters
+
+ x:
+ scalar or vector of complex or real numbers
+ q:
+ positive integer value, or logical
+ n:
+ positive integer, default value 4
+ Wc:
+ non decreasing vector or scalar, starting from 0 uptill 1, default value 0.5
+
+
+
+
+ Description
+
+This is an Octave function.
+This function upsamples the signal x by a factor of q, using an order 2*q*n+1 FIR filter.
+The second argument q must be an integer. The default values of the third and fourth arguments (n, Wc) are 4 and 0.5 respectively.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/intfilt.xml b/help/en_US/intfilt.xml
new file mode 100644
index 0000000..a674813
--- /dev/null
+++ b/help/en_US/intfilt.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ intfilt
+
+
+
+
diff --git a/help/en_US/invimpinvar.xml b/help/en_US/invimpinvar.xml
new file mode 100644
index 0000000..5517be5
--- /dev/null
+++ b/help/en_US/invimpinvar.xml
@@ -0,0 +1,71 @@
+
+
+
+
+
+
+
+ invimpinvar
+ This function converts digital filter with coefficients b and a to analog, conserving impulse response.
+
+
+
+
+ Calling Sequence
+
+ [b, a] = impinvar (b, a)
+ [b, a] = impinvar (b, a, fs)
+ [b, a] = impinvar (b, a, fs, tol)
+
+
+
+
+ Parameters
+
+ b:
+ real or complex valued scalar or vector
+ a:
+ real or complex valued scalar or vector, order should be greater than b
+ fs:
+ real or complex value, default value 1Hz
+ tol:
+ real or complex value, default value 0.0001
+
+
+
+
+ Description
+
+This is an Octave function.
+This function converts digital filter with coefficients b and a to analog, conserving impulse response.
+This function does the inverse of impinvar.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/is2rc.xml b/help/en_US/is2rc.xml
new file mode 100644
index 0000000..8f1bd14
--- /dev/null
+++ b/help/en_US/is2rc.xml
@@ -0,0 +1,74 @@
+
+
+
+
+
+
+
+ is2rc
+ Convert inverse sine parameters to reflection coefficients
+
+
+
+
+ Calling Sequence
+
+ K = is2rc(isin)
+
+
+
+
+
+ Parameters
+
+ isin:
+ input inverse sine parameters. Needs to be an array real numbers
+ k:
+ output reflection coefficients corresponding to the reflection coefficients in input
+
+
+
+
+ Description
+
+This function returns a vector of reflection coefficients from a vector of inverse sine parameters
+output array has k(i) = sin(pi/2*isin(i))
+
+
+Example
+k = [0.3090 0.9801 0.0031 0.0082 -0.0082];
+isin = rc2is(k) //Gives inverse sine parameters
+k_dash = is2rc(isin)
+
+
+
+
+
+
+ See also
+
+ rc2is
+ rc2poly
+ rc2ac
+ rc2lar
+ Author
+ Pandit
+
+
+
+
+ Bibliography
+ J.R. Deller, J.G. Proakis, J.H.L. Hansen, "Discrete-Time Processing of Speech Signals", Prentice Hall, Section 7.4.5
+
+
diff --git a/help/en_US/isallpass.xml b/help/en_US/isallpass.xml
new file mode 100644
index 0000000..b519d07
--- /dev/null
+++ b/help/en_US/isallpass.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ isallpass
+ [nargout,nargin]=argn();
+
+
+
diff --git a/help/en_US/isfir.xml b/help/en_US/isfir.xml
new file mode 100644
index 0000000..7aa7e0d
--- /dev/null
+++ b/help/en_US/isfir.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ isfir
+ [nargout,nargin]=argn();
+
+
+
diff --git a/help/en_US/islinphase.xml b/help/en_US/islinphase.xml
new file mode 100644
index 0000000..54ba2d7
--- /dev/null
+++ b/help/en_US/islinphase.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ islinphase
+ [nargout,nargin]=argn();
+
+
+
diff --git a/help/en_US/ismaxphase.xml b/help/en_US/ismaxphase.xml
new file mode 100644
index 0000000..67b480d
--- /dev/null
+++ b/help/en_US/ismaxphase.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ ismaxphase
+ [nargout,nargin]=argn();
+
+
+
diff --git a/help/en_US/isminphase.xml b/help/en_US/isminphase.xml
new file mode 100644
index 0000000..f395fbe
--- /dev/null
+++ b/help/en_US/isminphase.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ isminphase
+ [nargout,nargin]=argn();
+
+
+
diff --git a/help/en_US/isstable.xml b/help/en_US/isstable.xml
new file mode 100644
index 0000000..295c1d4
--- /dev/null
+++ b/help/en_US/isstable.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ isstable
+ SOS matrix corresponds to [bi(1) bi(2) bi(3) ai(1) ai(2) ai(3)].
+
+
+
diff --git a/help/en_US/kaiser.xml b/help/en_US/kaiser.xml
new file mode 100644
index 0000000..394119b
--- /dev/null
+++ b/help/en_US/kaiser.xml
@@ -0,0 +1,66 @@
+
+
+
+
+
+
+
+ kaiser
+ This function returns the filter coefficients of a Kaiser window.
+
+
+
+
+ Calling Sequence
+
+ w = kaiser (m)
+ w = kaiser (m, beta)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ beta:
+ real scalar value
+ w:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a Kaiser window of length m supplied as input, to the output vector w.
+The second parameter gives the stop band attenuation of the Fourier transform of the window on derivation.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/lar2rc.xml b/help/en_US/lar2rc.xml
new file mode 100644
index 0000000..5f07b8d
--- /dev/null
+++ b/help/en_US/lar2rc.xml
@@ -0,0 +1,60 @@
+
+
+
+
+
+
+
+ lar2rc
+
+
+
+
+
+ Calling Sequence
+
+ k = lar2rc(g)
+
+
+
+
+ Parameters
+
+ g:
+ define log area ratios.
+ k:
+ returns the reflection coefficients.
+
+
+
+
+ Examples
+
+
+
+
+ See also
+
+ Author
+ Singh
+ to match MATLAB o/p when i/p is of type char and is a string by Debdeep Dey
+
+
+
diff --git a/help/en_US/latc2tf.xml b/help/en_US/latc2tf.xml
new file mode 100644
index 0000000..8e90565
--- /dev/null
+++ b/help/en_US/latc2tf.xml
@@ -0,0 +1,89 @@
+
+
+
+
+
+
+
+ latc2tf
+ Convert lattice filter parameters to transfer function coefficients
+
+
+
+
+ Calling Sequence
+
+
+
+
+
+ Description
+
+[num,den] = latc2tf(k,v)
+Finds the transfer function of the IIR filter from the lattice
+coefficients k and ladder coefficients v.
+[num,den] = latc2tf(k,'iiroption')
+Finds the transfer function of the allpass or allpole (specified by
+the iiroption flag) IIR filter.
+num = latc2tf(k,'firoption')
+Finds the transfer function of the FIR filter from the lattice
+coefficients k. The firoption flag specifies the type of the FIR
+filter (can be 'min, 'max', or 'FIR')
+
+
+Parameters:
+k - double - vector
+Lattice coefficients
+Lattice coefficients for FIR/IIR filter. Can be real or complex.
+v - double - vector
+Ladder coefficients
+Ladder coefficients for IIR filters. Can be real or complex.
+iiroption - string flag - 'allpole', or 'allpass'
+Specification of the type if IIR filter
+firoption - string flag - 'min', 'max', or 'FIR' (default)
+Speficication of the type of FIR filter
+
+
+
+
+
+
+ Examples
+
+
+
+
+ See also
+
+ | tf2latc
+ References
+ J.G. Proakis, D.G. Manolakis, Digital Signal Processing,
+ ed., Prentice Hall, N.J., 1996, Chapter 7.
+ S. K. Mitra, Digital Signal Processing, A Computer
+ Approach, McGraw-Hill, N.Y., 1998, Chapter 6.
+
+
+
+
+ Authors
+
+ Ayush Baid
+
+
+
diff --git a/help/en_US/latcfilt.xml b/help/en_US/latcfilt.xml
new file mode 100644
index 0000000..d1a7327
--- /dev/null
+++ b/help/en_US/latcfilt.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ latcfilt
+ function [f,g,zo]=latcfilt(k,x,v,zi,dim)
+
+
+
diff --git a/help/en_US/latcfilt1.xml b/help/en_US/latcfilt1.xml
new file mode 100644
index 0000000..75cd7b1
--- /dev/null
+++ b/help/en_US/latcfilt1.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ latcfilt1
+ function [f,g,zo]=latcfilt1(k,v,x,zi)
+
+
+
diff --git a/help/en_US/levdown.xml b/help/en_US/levdown.xml
new file mode 100644
index 0000000..35cafba
--- /dev/null
+++ b/help/en_US/levdown.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ levdown
+
+
+
+
diff --git a/help/en_US/levin.xml b/help/en_US/levin.xml
new file mode 100644
index 0000000..e8c3a2f
--- /dev/null
+++ b/help/en_US/levin.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ levin
+ [ar,sigma2,rc]=lev(r)
+
+
+
diff --git a/help/en_US/levinson.xml b/help/en_US/levinson.xml
new file mode 100644
index 0000000..056795f
--- /dev/null
+++ b/help/en_US/levinson.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ levinson
+
+
+
+
diff --git a/help/en_US/lpc.xml b/help/en_US/lpc.xml
new file mode 100644
index 0000000..1ebe531
--- /dev/null
+++ b/help/en_US/lpc.xml
@@ -0,0 +1,90 @@
+
+
+
+
+
+
+
+ lpc
+ Linear prediction filter coefficients
+
+
+
+
+ Calling Sequence
+
+ [a,g] = lpc(x)
+ [a,g] = lpc(x,p)
+
+
+
+
+
+
+ Description
+
+[a,g] = lpc(x,p)
+Determines the coefficients of a pth order forward linear predictor
+filter by minimizing the squared error. If p is unspecified, a
+default value of length(x)-1 is used.
+
+
+
+
+
+
+ Parameters
+
+ x:
+ double
+ p:
+ int, natural number, scalar
+ a:
+ double
+ g:
+ double
+
+
+
+
+ Examples
+
+
+
+
+ See also
+
+ | levinson | prony | pyulear | stmcb
+
+
+
+
+ Authors
+
+ Ayush Baid
+
+
+
diff --git a/help/en_US/lsf2poly.xml b/help/en_US/lsf2poly.xml
new file mode 100644
index 0000000..799c953
--- /dev/null
+++ b/help/en_US/lsf2poly.xml
@@ -0,0 +1,30 @@
+
+
+
+
+
+
+
+ lsf2poly
+ lsf2poly function convert line spectral frequencies to prediction polynomial.
+
+
+
+
+ Calling Sequence
+
+ a = lsf2poly(lsf)
+
+
+
diff --git a/help/en_US/mag2db.xml b/help/en_US/mag2db.xml
new file mode 100644
index 0000000..79c952d
--- /dev/null
+++ b/help/en_US/mag2db.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ mag2db
+ funcprot(0);
+
+
+
diff --git a/help/en_US/marcumq.xml b/help/en_US/marcumq.xml
new file mode 100644
index 0000000..dfcf658
--- /dev/null
+++ b/help/en_US/marcumq.xml
@@ -0,0 +1,67 @@
+
+
+
+
+
+
+
+ marcumq
+ This function computes the generalized Marcum Q function of order m with noncentrality parameter a and argument b.
+
+
+
+
+ Calling Sequence
+
+ q = marcumq (a, b)
+ q = marcumq (a, b, m)
+ q = marcumq (a, b, m, tol)
+
+
+
+
+ Parameters
+
+ a:
+
+ b:
+
+ m:
+ default value 1
+ tol:
+ default value eps
+
+
+
+
+ Description
+
+This is an Octave function.
+This function computes the generalized Marcum Q function of order m with noncentrality parameter a and argument b.
+The third argument m is the order, which by default is 1.
+The fourth argument tol is the tolerance, which by default is eps.
+If input arguments are vectors which correspond in size and degree, the output is a table of values.
+This function calculates Marcum’s Q function using the infinite Bessel series, which is truncated when the relative error is less than the specified tolerance.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/master_help.xml b/help/en_US/master_help.xml
new file mode 100644
index 0000000..106caf6
--- /dev/null
+++ b/help/en_US/master_help.xml
@@ -0,0 +1,407 @@
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
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+
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+
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+
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+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+]>
+
+
+ FOSSEE Signal Processing Toolbox
+
+
+
+FOSSEE Signal Processing Toolbox
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+
diff --git a/help/en_US/medfilt1.xml b/help/en_US/medfilt1.xml
new file mode 100644
index 0000000..2766e88
--- /dev/null
+++ b/help/en_US/medfilt1.xml
@@ -0,0 +1,98 @@
+
+
+
+
+
+
+
+ medfilt1
+ 1D median filtering
+
+
+
+
+ Calling Sequence
+
+
+
+
+
+ Description
+
+y = medfilt1(x)
+Applies a 3rd order 1-dimensional median filter to input x along the
+first non-zero dimension. The function appropriately pads the signal
+with zeros at the endings. For a segment, a median is calculated as
+the middle value (average of two middle values) for odd number
+number (even number) of data points.
+y = medfilt1(x,n)
+Applies a nth order 1-dimensional median filter.
+y = medfilt1(x,n,dim)
+Applies the median filter along the n-th dimension
+y = medfilt1(__, nanflag, padding)
+nanflag specifies how NaN values are treated. padding specifies the
+type of filtering to be performed at the signal edges.
+
+
+
+
+
+
+ Parameters
+
+ x:
+ int | double
+ n:
+ positive integer scalar
+ dim:
+ positive integer scalar
+ nanflag:
+ 'includenan' (default) | 'omitnan'
+ * includenan:
+ Filtering such that the median of any segment
+ * omitnan:
+ Filtering with NaNs omitted in each segment. If a segment
+ y:
+ int | double
+
+
+
+
+ Examples
+
+
+
+
+ See also
+
+ | hampel | median | sgolayfilt
+
+
+
+
+ Authors
+
+ Ayush Baid
+
+
+
diff --git a/help/en_US/mexihat.xml b/help/en_US/mexihat.xml
new file mode 100644
index 0000000..3306e6f
--- /dev/null
+++ b/help/en_US/mexihat.xml
@@ -0,0 +1,62 @@
+
+
+
+
+
+
+
+ mexihat
+
+
+
+
+
+ Calling Sequence
+
+ [psi,x]=mexihat(lb,ub,n)
+
+
+
+
+ Parameters
+
+ lb:
+ Real or complex valued vector or matrix
+ ub:
+ Real or complex valued vector or matrix
+ n:
+ Real strictly positive scalar number
+
+
+
+
+ Description
+
+This is an Octave function
+This function returns values of the Mexican hat wavelet in the specified interval at all the sample points.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/meyeraux.xml b/help/en_US/meyeraux.xml
new file mode 100644
index 0000000..897a8fa
--- /dev/null
+++ b/help/en_US/meyeraux.xml
@@ -0,0 +1,56 @@
+
+
+
+
+
+
+
+ meyeraux
+
+
+
+
+
+ Calling Sequence
+
+ [y]=meyeraux(x)
+
+
+
+
+ Parameters
+
+ x:
+ Real or complex valued vector or matrix
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns values of the auxiliary function used for Meyer wavelet generation.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/midcross.xml b/help/en_US/midcross.xml
new file mode 100644
index 0000000..b8fe623
--- /dev/null
+++ b/help/en_US/midcross.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ midcross
+
+
+
+
diff --git a/help/en_US/modulate.xml b/help/en_US/modulate.xml
new file mode 100644
index 0000000..f8a4859
--- /dev/null
+++ b/help/en_US/modulate.xml
@@ -0,0 +1,82 @@
+
+
+
+
+
+
+
+ modulate
+ Modulates signal according to the modulation method
+
+
+
+
+ Calling Sequence
+
+ y=modulate(x,fc,fs,method,opt)
+ [y,t]=modulate(x,fc,fs,method,opt)
+
+
+
+
+ Parameters
+
+
+
+
+
+ Description
+
+[y,t]=modulate(x,fc,fs,method,opt)
+Returns the modulated vector y and the time vector 't'
+Modulation is performed according to the following table
+METHOD MODULATION SCHEME
+'am', Amplitude modulation, double side-band, suppressed carrier
+'amdsb-sc' opt not used. This is the default method.
+'amdsb-tc' Amplitude modulation, double side-band, transmitted carrier
+opt is a scalar subtracted from x before multiplying x
+with the carrier wave. It defaults to min(min(x)) so that
+the input signal after offset is always non-negative
+'amssb' Amplitude modulation, single side-band
+OPT not used.
+'fm' Frequency modulation
+opt is the constant of frequency modulation.
+opt = (fc/fs)*2*pi/max(max(abs(x))) by default
+'pm' Phase modulation
+OPT is the constant of phase modulation.
+opt = pi/max(max(abs(x))) by default
+Phase lies between -pi and +pi
+'pwm' Pulse width modulation
+opt='left' corresponds to left justified pulses.
+opt='centered' correspondes to centered pulses.
+The default value of opt is 'left'.
+'ppm' Pulse position modulation
+opt is a scalar between 0 and 1 which specifies the pulse
+width in fractions of the carrier period with default value 0.1.
+'qam' Quadrature amplitude modulation
+opt is a matrix of the same size as X which is modulated in
+quadrature with x.
+
+
+If x is a matrix, its columns are modulated.
+Example
+y =
+
+
+1. 1. 0. 0.
+Author
+Ankur Mallick
+
+
+
diff --git a/help/en_US/morlet.xml b/help/en_US/morlet.xml
new file mode 100644
index 0000000..612568d
--- /dev/null
+++ b/help/en_US/morlet.xml
@@ -0,0 +1,62 @@
+
+
+
+
+
+
+
+ morlet
+
+
+
+
+
+ Calling Sequence
+
+ [psi,x]= morlet(lb,ub,n)
+
+
+
+
+ Parameters
+
+ lb:
+ Real or complex valued vector or matrix
+ ub:
+ Real or complex valued vector or matrix
+ n:
+ Real strictly positive scalar number
+
+
+
+
+ Description
+
+This is an Octave function
+This function returns values of the Morlet wavelet in the specified interval for all the sample points.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/movingrms.xml b/help/en_US/movingrms.xml
new file mode 100644
index 0000000..c40672d
--- /dev/null
+++ b/help/en_US/movingrms.xml
@@ -0,0 +1,68 @@
+
+
+
+
+
+
+
+ movingrms
+
+
+
+
+
+ Calling Sequence
+
+ [rmsx,w]=movingrms(x,w,rc,Fs=1)
+
+
+
+
+ Parameters
+
+ x:
+ Real or complex valued vector or matrix
+ w:
+ Real or complex scalar value
+ rc:
+ Real or complex scalar value
+ Fs:
+ Real or complex scalar value
+
+
+
+
+ Description
+
+This is an Octave function.
+The signal is convoluted against a sigmoid window of width w and risetime rc with the units of these parameters relative to the value of the sampling frequency given in Fs (Default value=1).
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/musicBase.xml b/help/en_US/musicBase.xml
new file mode 100644
index 0000000..e95d3ff
--- /dev/null
+++ b/help/en_US/musicBase.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ musicBase
+ Implements the core of the MUSIC algorithm
+
+
+
diff --git a/help/en_US/nnls.xml b/help/en_US/nnls.xml
new file mode 100644
index 0000000..ca50d34
--- /dev/null
+++ b/help/en_US/nnls.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ nnls
+ Non Negative Least Squares (nnls) for Ex=f with the constraint x>=0
+
+
+
diff --git a/help/en_US/nuttallwin.xml b/help/en_US/nuttallwin.xml
new file mode 100644
index 0000000..5732027
--- /dev/null
+++ b/help/en_US/nuttallwin.xml
@@ -0,0 +1,62 @@
+
+
+
+
+
+
+
+ nuttallwin
+ This function returns the filter coefficients of a Blackman-Harris window.
+
+
+
+
+ Calling Sequence
+
+ w = nuttallwin (m)
+ w = nuttallwin (m, opt)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ opt:
+ string value, takes in "periodic" or "symmetric"
+ w:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a Blackman-Harris window defined by Nuttall of length m supplied as input, to the output vector w.
+The second parameter can take the values "periodic" or "symmetric", depending on which the corresponding form of window is returned. The default is symmetric.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/parzenwin.xml b/help/en_US/parzenwin.xml
new file mode 100644
index 0000000..e3bbbee
--- /dev/null
+++ b/help/en_US/parzenwin.xml
@@ -0,0 +1,59 @@
+
+
+
+
+
+
+
+ parzenwin
+ This function returns the filter coefficients of a Parzen window.
+
+
+
+
+ Calling Sequence
+
+ y = parzenwin (m)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ y:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a Parzen window of length m supplied as input, to the output vector y.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/pchip.xml b/help/en_US/pchip.xml
new file mode 100644
index 0000000..c9aaa34
--- /dev/null
+++ b/help/en_US/pchip.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ pchip
+
+
+
+
diff --git a/help/en_US/pchips.xml b/help/en_US/pchips.xml
new file mode 100644
index 0000000..8f356f5
--- /dev/null
+++ b/help/en_US/pchips.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ pchips
+
+
+
+
diff --git a/help/en_US/peak2peak.xml b/help/en_US/peak2peak.xml
new file mode 100644
index 0000000..cdc3221
--- /dev/null
+++ b/help/en_US/peak2peak.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ peak2peak
+ funcprot(0);
+
+
+
diff --git a/help/en_US/peak2rms.xml b/help/en_US/peak2rms.xml
new file mode 100644
index 0000000..dd21dae
--- /dev/null
+++ b/help/en_US/peak2rms.xml
@@ -0,0 +1,140 @@
+
+
+
+
+
+
+
+ peak2rms
+ This function calculates the ratio of peak magnitude to the Root Mean Square(RMS) value.
+
+
+
+
+ Calling Sequence
+
+ OUT=peak2rms(IN)
+ OUT=peak2rms(IN,orientation)
+
+
+
+
+
+ Parameters
+
+ in:
+ Vector or Matrix of real or complex elements.
+ orientation:
+ A string with possible values "r", "c" or "m" giving the dimension along which the peak2rms value is to be calculated.
+ out:
+ A scalar with real value when input is a vector.When input is a matrix, out is the peak magnitude to RMS value along the orientation specified or the default one when not specified.
+
+
+
+
+ Description
+
+For vector as input, the output is the ratio of peak value to the RMS value. The RMS value can be calculated by taking the square root of mean value of the squared sum of the elements.
+
+
+When a matrix is given as input the output is peak to RMS ratio in the orientation specified.
+The orientation can be given as string with values "r","c" or "m".
+
+
+peak2rms(in, 1) calculates the values of ratio of peak to RMS of columns of matrix. The output in this case is a row vector with peak2rms value of each column of in.
+
+
+peak2rms(in, 2) calculates the values of ratio of peak to RMS of rows of matrix, where the output would be a column vector having peak2rms value of each row of in.
+
+
+The default orientation is chosen to be the index of first dimension of input greater than 1.Hence peak2rms(in) is equivalent to peak2rms(in, "m").
+
+
+For an N dimensional array the orientation is the index of first non singleton dimension of the array.
+
+
+If the elements of matrix are complex the absolute values are considered in the calculation of RMS value.
+
+
+
+
+
+
+ Examples
+
+
+
+
+ Examples
+
+
+
+
+ Examples
+
+
+
+
+ See also
+
+ abs
+ mean
+ max
+ sqrt
+ isempty
+
+
+
+
+ Authors
+
+ Indira Askaukar
+
+
+
+
+ Bibliography
+ Matlab help document.
+ Modified to accept char i/p
+ MOdified function to match MATLAB input arguments
+ Now for calculating the values of ratio of peak to RMS of columns of matrix use peak2rms(in,1)
+ And for calculates the values of ratio of peak to RMS of rows of matrix. use peak2rms(in,2)
+ Updated help comments accordingly
+ MOdifications done by by Debdeep Dey
+
+
diff --git a/help/en_US/peig.xml b/help/en_US/peig.xml
new file mode 100644
index 0000000..8c85698
--- /dev/null
+++ b/help/en_US/peig.xml
@@ -0,0 +1,101 @@
+
+
+
+
+
+
+
+ peig
+ Psuedospectrum using the eigenvector method.
+
+
+
+
+ Calling Sequence
+
+ [S,w] = peig(x,p)
+ [S,w] = peig(x,p,w)
+ [S,w] = peig(x,p,nfft)
+ [S,w] = peig(x,p,nfft,fs)
+ [S,w] = peig(x,p,f,fs)
+ [S,f] = peig(...,'corr')
+ [S,f] = peig(x,p,nfft,fs,nwin,noverlap)
+ [...] = peig(...,freqrange)
+ [...,v,e] = peig(...)
+
+ Parameters:
+ x - int|double - vector|matrix
+ Input signal. In case of a matrix, each row of x represents a
+ seperate observation of the signal. If 'corr' flag is specified,
+ then x is the correlation matrix.
+ If w is not specified in the input, it is determined by the
+ algorithm. If x is real valued, then range of w is [0, pi].
+ Otherwise, the range of w is [0, 2pi)
+ p - int|double - scalar|vector
+ p(1) is the dimension of the signal subspace
+ p(2), if specified, represents a threshold that is multiplied by
+ the smallest estimated eigenvalue of the signal's correlation matrix.
+ w - int|double - vector
+ w is the vector of normalized frequencies over which the
+ pseuspectrogram is to be computed.
+ nfft - int - scalar (Default = 256)
+ Length of the fft used to compute pseudospectrum. The length of S
+ (and hence w/f) depends on the type of values in x and nfft.
+ If x is real, length of s is (nfft/2 + 1) {Range of w = [0, pi]} if
+ nfft is even and (nfft+1)/2 {Range of w = [0, pi)} otherwise.
+ If x is complex, length of s is nfft.
+ fs - int|double - scalar (Default = 1)
+ Sampling rate. Used to convert the normalized frequencies (w) to
+ actual values (f) and vice-versa.
+ nwin - int|double - scalar (int only)|vector (Default = 2*p(1))
+ If nwin is scalar, it is the length of the rectangular window.
+ Otherwise, the vector input is considered as the window coefficients.
+ Not used if 'corr' flag present.
+ If x is a vector, windowing not done in nwin in scalar. If x is a
+ matrix,
+ noverlap - int - scalar (Default = nwin-1)
+ number of points by which successive windows overlap. noverlap not
+ used if x is a matrix
+ freqrange - string
+ The range of frequencies over which the pseudospetrogram is
+ computed. Three possible values - 'onesided', 'twosided', 'centered'
+ 'corr' flag
+ Presence indicates that the primary input x is actually a
+ correlation matrix
+
+ Examples:
+ TODO:
+
+
+
+
+
+ See also
+
+ | pmusic | pmtm | pcov | pmcov | pburg | pyulear | pwelch | corrmtx
+
+
+
+
+ Authors
+
+ Ayush Baid
+ References
+ [1] Petre Stoica and Randolph Moses, Introduction To Spectral
+ Analysis, Prentice-Hall, 1997, pg. 15
+ [2] S. J. Orfanidis, Optimum Signal Processing. An Introduction.
+ 2nd Ed., Macmillan, 1988.
+
+
+
diff --git a/help/en_US/periodogram.xml b/help/en_US/periodogram.xml
new file mode 100644
index 0000000..8b5b2ed
--- /dev/null
+++ b/help/en_US/periodogram.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ periodogram
+ funcprot(0);
+
+
+
diff --git a/help/en_US/phaseInputParseAs_ab.xml b/help/en_US/phaseInputParseAs_ab.xml
new file mode 100644
index 0000000..e396476
--- /dev/null
+++ b/help/en_US/phaseInputParseAs_ab.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ phaseInputParseAs_ab
+ fs=0;
+
+
+
diff --git a/help/en_US/phaseInputParseAs_sos.xml b/help/en_US/phaseInputParseAs_sos.xml
new file mode 100644
index 0000000..30d79d3
--- /dev/null
+++ b/help/en_US/phaseInputParseAs_sos.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ phaseInputParseAs_sos
+ fs=0;
+
+
+
diff --git a/help/en_US/phasedelay.xml b/help/en_US/phasedelay.xml
new file mode 100644
index 0000000..5790eb1
--- /dev/null
+++ b/help/en_US/phasedelay.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ phasedelay
+ cas variable is 2 if sos form is involved and 1 if direct rational form is given
+
+
+
diff --git a/help/en_US/phasez.xml b/help/en_US/phasez.xml
new file mode 100644
index 0000000..7127566
--- /dev/null
+++ b/help/en_US/phasez.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ phasez
+ cas variable is 2 if sos form is involved and 1 if direct rational form is given
+
+
+
diff --git a/help/en_US/pmusic.xml b/help/en_US/pmusic.xml
new file mode 100644
index 0000000..d299b3a
--- /dev/null
+++ b/help/en_US/pmusic.xml
@@ -0,0 +1,101 @@
+
+
+
+
+
+
+
+ pmusic
+ Psuedospectrum using MUSIC algorithm
+
+
+
+
+ Calling Sequence
+
+ [S,w] = pmusic(x,p)
+ [S,w] = pmusic(x,p,w)
+ [S,w] = pmusic(x,p,nfft)
+ [S,w] = pmusic(x,p,nfft,fs)
+ [S,w] = pmusic(x,p,f,fs)
+ [S,f] = pmusic(...,'corr')
+ [S,f] = pmusic(x,p,nfft,fs,nwin,noverlap)
+ [...] = pmusic(...,freqrange)
+ [...,v,e] = pmusic(...)
+
+ Parameters:
+ x - int|double - vector|matrix
+ Input signal. In case of a matrix, each row of x represents a
+ seperate observation of the signal. If 'corr' flag is specified,
+ then x is the correlation matrix.
+ If w is not specified in the input, it is determined by the
+ algorithm. If x is real valued, then range of w is [0, pi].
+ Otherwise, the range of w is [0, 2pi)
+ p - int|double - scalar|vector
+ p(1) is the dimension of the signal subspace
+ p(2), if specified, represents a threshold that is multiplied by
+ the smallest estimated eigenvalue of the signal's correlation matrix.
+ w - int|double - vector
+ w is the vector of normalized frequencies over which the
+ pseuspectrogram is to be computed.
+ nfft - int - scalar (Default = 256)
+ Length of the fft used to compute pseudospectrum. The length of S
+ (and hence w/f) depends on the type of values in x and nfft.
+ If x is real, length of s is (nfft/2 + 1) {Range of w = [0, pi]} if
+ nfft is even and (nfft+1)/2 {Range of w = [0, pi)} otherwise.
+ If x is complex, length of s is nfft.
+ fs - int|double - scalar (Default = 1)
+ Sampling rate. Used to convert the normalized frequencies (w) to
+ actual values (f) and vice-versa.
+ nwin - int|double - scalar (int only)|vector (Default = 2*p(1))
+ If nwin is scalar, it is the length of the rectangular window.
+ Otherwise, the vector input is considered as the window coefficients.
+ Not used if 'corr' flag present.
+ If x is a vector, windowing not done in nwin in scalar. If x is a
+ matrix,
+ noverlap - int - scalar (Default = nwin-1)
+ number of points by which successive windows overlap. noverlap not
+ used if x is a matrix
+ freqrange - string
+ The range of frequencies over which the pseudospetrogram is
+ computed. Three possible values - 'onesided', 'twosided', 'centered'
+ 'corr' flag
+ Presence indicates that the primary input x is actually a
+ correlation matrix
+
+ Examples:
+ TODO:
+
+
+
+
+
+ See also
+
+ | peig | periodogram | pmtm | prony | pwelch | rooteig | rootmusic
+
+
+
+
+ Authors
+
+ Ayush Baid
+ References
+ [1] Petre Stoica and Randolph Moses, Introduction To Spectral
+ Analysis, Prentice-Hall, 1997, pg. 15
+ [2] S. J. Orfanidis, Optimum Signal Processing. An Introduction.
+ 2nd Ed., Macmillan, 1988.
+
+
+
diff --git a/help/en_US/poly2ac.xml b/help/en_US/poly2ac.xml
new file mode 100644
index 0000000..f7ac3e0
--- /dev/null
+++ b/help/en_US/poly2ac.xml
@@ -0,0 +1,80 @@
+
+
+
+
+
+
+
+ poly2ac
+ Convert prediction polynomial to autocorrelation sequence.
+
+
+
+
+ Calling Sequence
+
+ R = poly2ac(a,efinal)
+
+
+
+
+
+ Parameters
+
+ a:
+ input prediction polynomial with 1st element 1 (if not, poly2ac normalizes it to 1 before proceeding).
+ efinal:
+ input prediction error
+ r:
+ output autocorrelation sequence
+
+
+
+
+ Description
+
+This function obtains the underlying autocorrelation sequence that would best fit a linear prediction filter described by the
+denominator polynomial and the numerator scaling. The filter is H(z) = efinal/(a(1) + a(2) x z a(3) x z^2 ... a(n) x z^n-1)
+
+
+
+
+
+
+ Examples
+
+
+
+
+ See also
+
+ ac2poly
+ poly2rc
+ rc2poly
+ rc2ac
+ ac2rc
+ Parthe Pandit
+
+
+
+
+ Bibliography
+ S. Kay, Modern Spectral Estimation, Prentice Hall, N.J., 1987, Chapter 6.
+
+
diff --git a/help/en_US/poly2lsf.xml b/help/en_US/poly2lsf.xml
new file mode 100644
index 0000000..27ca6a2
--- /dev/null
+++ b/help/en_US/poly2lsf.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ poly2lsf
+
+
+
+
diff --git a/help/en_US/poly2rc.xml b/help/en_US/poly2rc.xml
new file mode 100644
index 0000000..ad61ebc
--- /dev/null
+++ b/help/en_US/poly2rc.xml
@@ -0,0 +1,31 @@
+
+
+
+
+
+
+
+ poly2rc
+
+
+
+
+
+ Calling Sequence
+
+ kr = poly2rc(a)
+ [kr, R0] = rc2poly(a, efinal)
+
+
+
diff --git a/help/en_US/polyscale.xml b/help/en_US/polyscale.xml
new file mode 100644
index 0000000..f689e36
--- /dev/null
+++ b/help/en_US/polyscale.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ polyscale
+ errcheck1
+
+
+
diff --git a/help/en_US/polystab.xml b/help/en_US/polystab.xml
new file mode 100644
index 0000000..7c9010b
--- /dev/null
+++ b/help/en_US/polystab.xml
@@ -0,0 +1,55 @@
+
+
+
+
+
+
+
+ polystab
+ This function stabilizes the polynomial transfer function.
+
+
+
+
+ Calling Sequence
+
+ b = polystab(a)
+
+
+
+
+ Parameters
+
+ a:
+
+
+
+
+
+ Description
+
+This is an Octave function.
+This function stabilizes the polynomial transfer function by replacing all roots outside the unit circle with their reflection inside the unit circle.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/polyval.xml b/help/en_US/polyval.xml
new file mode 100644
index 0000000..f041227
--- /dev/null
+++ b/help/en_US/polyval.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ polyval
+
+
+
+
diff --git a/help/en_US/pow2db.xml b/help/en_US/pow2db.xml
new file mode 100644
index 0000000..cc6a49b
--- /dev/null
+++ b/help/en_US/pow2db.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ pow2db
+ rhs = argn(2)
+
+
+
diff --git a/help/en_US/primitive.xml b/help/en_US/primitive.xml
new file mode 100644
index 0000000..c31f555
--- /dev/null
+++ b/help/en_US/primitive.xml
@@ -0,0 +1,60 @@
+
+
+
+
+
+
+
+ primitive
+ This function calculates the primitive of a given function supplied as input.
+
+
+
+
+ Calling Sequence
+
+ y = primitive(f, t)
+ y = primitive(f, t, x)
+
+
+
+
+ Parameters
+
+ f:
+
+ t:
+
+
+
+
+
+ Description
+
+This is an Octave function.
+This function calculates the primitive of a given function supplied as input.
+The second parameter t is a vector at which the output is evaluated (at the points t). This vector should be ascending and ordered.
+The function approximates the primitive (indefinite integral) of the univariate function handle f with constant of integration x.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/prony.xml b/help/en_US/prony.xml
new file mode 100644
index 0000000..25734a1
--- /dev/null
+++ b/help/en_US/prony.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ prony
+
+
+
+
diff --git a/help/en_US/pulseperiod.xml b/help/en_US/pulseperiod.xml
new file mode 100644
index 0000000..7fc379f
--- /dev/null
+++ b/help/en_US/pulseperiod.xml
@@ -0,0 +1,36 @@
+
+
+
+
+
+
+
+ pulseperiod
+
+
+
+
+
+ Calling Sequence
+
+ p=pulseperiod(x)
+ p=pulseperiod(x, Fs)
+ p=pulseperiod(x, t)
+ p=pulseperiod (x, t, 'Polarity', pol)
+ p=pulseperiod(x, t, 'MidPercentReferenceLevel', N )
+ p=pulseperiod(x, t, 'Tolerance', M)
+ p=pulseperiod(x, t,'StateLevels', O)
+
+
+
diff --git a/help/en_US/pulsesep.xml b/help/en_US/pulsesep.xml
new file mode 100644
index 0000000..4c28fd4
--- /dev/null
+++ b/help/en_US/pulsesep.xml
@@ -0,0 +1,36 @@
+
+
+
+
+
+
+
+ pulsesep
+
+
+
+
+
+ Calling Sequence
+
+ s=pulsesep(x)
+ s=pulsesep(x, Fs)
+ s=pulsesep(x, t)
+ s=pulsesep (x, t, 'Polarity', pol)
+ s=pulsesep(x, t, 'MidPercentReferenceLevel', N )
+ s=pulsesep(x, t, 'Tolerance', M)
+ s=pulsesep(x, t,'StateLevels', [O 1])
+
+
+
diff --git a/help/en_US/pulsewidth.xml b/help/en_US/pulsewidth.xml
new file mode 100644
index 0000000..ee08682
--- /dev/null
+++ b/help/en_US/pulsewidth.xml
@@ -0,0 +1,36 @@
+
+
+
+
+
+
+
+ pulsewidth
+
+
+
+
+
+ Calling Sequence
+
+ w=pulsewidth(x)
+ w=pulsewidth(x, Fs)
+ w=pulsewidth(x, t)
+ w=pulsewidth (x, t, 'Polarity', pol)
+ w=pulsewidth(x, t, 'MidPercentReferenceLevel', N )
+ w=pulsewidth(x, t, 'Tolerance', M)
+ w=pulsewidth(x, t,'StateLevels', O)
+
+
+
diff --git a/help/en_US/pulstran.xml b/help/en_US/pulstran.xml
new file mode 100644
index 0000000..96ef0c0
--- /dev/null
+++ b/help/en_US/pulstran.xml
@@ -0,0 +1,67 @@
+
+
+
+
+
+
+
+ pulstran
+ This function generates the signal y = sum(func(t+d,...)) for each d.
+
+
+
+
+ Calling Sequence
+
+ y = pulstran (t, d, func, ...)
+ y = pulstran (t, d, p)
+ y = pulstran (t, d, p, fs)
+ y = pulstran (t, d, p, Ffs, meth)
+
+
+
+
+ Parameters
+
+ t:
+
+ d:
+ vector or matrix
+ p:
+
+ fs:
+ default value 1Hz
+ func:
+ function which accepts vector (of times)
+
+
+
+
+ Description
+
+This is an Octave function.
+This function generates the signal y = sum(func(t+d,...)) for each d. If d is a matrix of two columns, the first column is the delay d and the second column is the amplitude a, and y = sum(a*func(t+d)) for each d, a. Here, func is a function which accepts a vector of times.
+If a pulse shape sampled at frequency Fs (default 1 Hz) is supplied instead of a function name, an interpolated version of the pulse is added at each delay d.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/rc2ac.xml b/help/en_US/rc2ac.xml
new file mode 100644
index 0000000..acfe121
--- /dev/null
+++ b/help/en_US/rc2ac.xml
@@ -0,0 +1,30 @@
+
+
+
+
+
+
+
+ rc2ac
+
+
+
+
+
+ Calling Sequence
+
+ a = rc2ac(k, R0)
+
+
+
diff --git a/help/en_US/rc2is.xml b/help/en_US/rc2is.xml
new file mode 100644
index 0000000..1761ce6
--- /dev/null
+++ b/help/en_US/rc2is.xml
@@ -0,0 +1,80 @@
+
+
+
+
+
+
+
+ rc2is
+ Convert reflection coefficients to inverse sine parameters
+
+
+
+
+ Calling Sequence
+
+ isin = rc2is(K)
+
+
+
+
+
+ Parameters
+
+ k:
+ input reflection coefficients. Needs to be an array of real numbers between -1 and 1
+ isin:
+ inverse sine parameters corresponding to the reflection coefficients in input
+
+
+
+
+ Description
+
+This function returns the inverse sine parameters corresponding to the input reflection coefficients K.
+output array has isin(i) = 2/pi*asin(k(i))
+
+
+Example
+k = [0.3090 0.9801 0.0031 0.0082 -0.0082];
+isin = rc2is(k) //Gives inverse sine parameters
+
+
+
+
+
+
+ See also
+
+ is2rc
+ rc2poly
+ rc2ac
+ rc2lar
+
+
+
+
+ Authors
+
+ Parthe Pandit
+
+
+
+
+ Bibliography
+ J.R. Deller, J.G. Proakis, J.H.L. Hansen, "Discrete-Time Processing of Speech Signals", Prentice Hall, Section 7.4.5
+ modified function to handle char i/p and also changed error statements to match those of MATLAB by Debdeep Dey
+ convert char i/p to their respective ascii values
+
+
diff --git a/help/en_US/rc2lar.xml b/help/en_US/rc2lar.xml
new file mode 100644
index 0000000..3dd7bcf
--- /dev/null
+++ b/help/en_US/rc2lar.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ rc2lar
+
+
+
+
diff --git a/help/en_US/rc2poly.xml b/help/en_US/rc2poly.xml
new file mode 100644
index 0000000..f8c66a4
--- /dev/null
+++ b/help/en_US/rc2poly.xml
@@ -0,0 +1,31 @@
+
+
+
+
+
+
+
+ rc2poly
+
+
+
+
+
+ Calling Sequence
+
+ a = rc2poly(kr)
+ [a, efinal] = rc2poly(kr,R0)
+
+
+
diff --git a/help/en_US/rcosdesign.xml b/help/en_US/rcosdesign.xml
new file mode 100644
index 0000000..e586d0d
--- /dev/null
+++ b/help/en_US/rcosdesign.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ rcosdesign
+ RCOSDESIGN computes the raised cosine FIR filter
+
+
+
diff --git a/help/en_US/rectpuls.xml b/help/en_US/rectpuls.xml
new file mode 100644
index 0000000..261d203
--- /dev/null
+++ b/help/en_US/rectpuls.xml
@@ -0,0 +1,60 @@
+
+
+
+
+
+
+
+ rectpuls
+
+
+
+
+
+ Calling Sequence
+
+ [y]=rectpuls(t)
+ [y]=rectpuls(t,w)
+
+
+
+
+ Parameters
+
+ t:
+ Real or complex valued vector or matrix
+ w:
+ Real or complex valued vector or matrix
+
+
+
+
+ Description
+
+This is an Octave function
+y = rectpuls(t) returns a continuous, aperiodic, unity-height rectangular pulse depending upon input t, centered about t=0 and having default width of 1.
+y = rectpuls(t,w) generates a rectangle of width w.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/rectwin.xml b/help/en_US/rectwin.xml
new file mode 100644
index 0000000..c91a49b
--- /dev/null
+++ b/help/en_US/rectwin.xml
@@ -0,0 +1,59 @@
+
+
+
+
+
+
+
+ rectwin
+ This function returns the filter coefficients of a rectangular window.
+
+
+
+
+ Calling Sequence
+
+ y = rectwin (m)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ y:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a rectangular window of length m supplied as input, to the output vector y.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/resample.xml b/help/en_US/resample.xml
new file mode 100644
index 0000000..aab50da
--- /dev/null
+++ b/help/en_US/resample.xml
@@ -0,0 +1,63 @@
+
+
+
+
+
+
+
+ resample
+ This function resamples in the input sequence x supplied by a factor of p/q.
+
+
+
+
+ Calling Sequence
+
+ y = resample(x, p, q)
+ y = resample(x, p, q, h)
+ [y, h] = resample(...)
+
+
+
+
+ Parameters
+
+ x:
+ scalar, vector or matrix of real or complex numbers
+ p:
+ positive integer value
+ q:
+ positive integer value
+ h:
+ scalar, vector or matrix of real or complex numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function resamples in the input sequence x supplied by a factor of p/q. If x is a matrix, then every column is resampled.hange the sample rate of x by a factor of p/q.
+This is performed using a polyphase algorithm. The impulse response h, given as parameter 4, of the antialiasing filter is either specified or designed with a Kaiser-windowed sinecard.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/residued.xml b/help/en_US/residued.xml
new file mode 100644
index 0000000..8bdecc5
--- /dev/null
+++ b/help/en_US/residued.xml
@@ -0,0 +1,59 @@
+
+
+
+
+
+
+
+ residued
+
+
+
+
+
+ Calling Sequence
+
+ [r,p,f,m]=residued(b,a)
+
+
+
+
+ Parameters
+
+ b:
+ Real or complex valued vector or matrix
+ a:
+ Real or complex valued vector or matrix
+
+
+
+
+ Description
+
+This is an Octave function.
+Similar to the "residuez" function. The difference being in the function "residuez", the IIR part (poles p and residues r) is driven in parallel with the FIR part(f) whereas in the function "residued", the IIR part is driven by the output of the FIR part. In signal modeling applications, this structure can be more accurate.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/residuez.xml b/help/en_US/residuez.xml
new file mode 100644
index 0000000..4655c59
--- /dev/null
+++ b/help/en_US/residuez.xml
@@ -0,0 +1,59 @@
+
+
+
+
+
+
+
+ residuez
+
+
+
+
+
+ Calling Sequence
+
+ [r,p,f,m]=residuez(b,a)
+
+
+
+
+ Parameters
+
+ b:
+ Real or complex valued vector or matrix
+ a:
+ Real or complex valued vector or matrix
+
+
+
+
+ Description
+
+This is an Octave function
+It compute the PFE of filter H(z)= B(z)/A(z) where inputs b and a are vectors specifying the digital filter.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/risetime.xml b/help/en_US/risetime.xml
new file mode 100644
index 0000000..e973c45
--- /dev/null
+++ b/help/en_US/risetime.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ risetime
+
+
+
+
diff --git a/help/en_US/rlevinson.xml b/help/en_US/rlevinson.xml
new file mode 100644
index 0000000..799fcb1
--- /dev/null
+++ b/help/en_US/rlevinson.xml
@@ -0,0 +1,33 @@
+
+
+
+
+
+
+
+ rlevinson
+
+
+
+
+
+ Calling Sequence
+
+ a = rlevinson(a, efinal)
+ [a, U] = rlevinson(a, efinal)
+ [a, U, kr] = rlevinson(a, efinal)
+ [a, U, kr, e] = rlevinson(a, efinal)
+
+
+
diff --git a/help/en_US/rms.xml b/help/en_US/rms.xml
new file mode 100644
index 0000000..4bac697
--- /dev/null
+++ b/help/en_US/rms.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ rms
+ convert i/p values to their ascii values if they are of type char
+
+
+
diff --git a/help/en_US/rooteig.xml b/help/en_US/rooteig.xml
new file mode 100644
index 0000000..9b2abe2
--- /dev/null
+++ b/help/en_US/rooteig.xml
@@ -0,0 +1,62 @@
+
+
+
+
+
+
+
+ rooteig
+ Frequencies and power of sinusoids using eigenvector algorithm
+
+
+
+
+ Calling Sequence
+
+ w = rooteig(x,p)
+ [w,pow] = rooteig(x,p)
+ [f,pow] = rooteig(...,fs)
+ [w,pow] = rooteig(...,'corr')
+
+
+
+
+
+ Parameters
+
+ Examples:
+
+ 1) 3 complex exponentials:
+
+ n=0:
+ 99;
+
+
+
+
+ See also
+
+ | peig | pmusic | rootmusic
+ References
+ Stoica, P. and R. Moses, INTRODUCTION TO SPECTRAL ANALYSIS,
+ Prentice-Hall
+ arguments
+ - double - vector
+ frequencies of the complex sinusoids
+ - double - vector
+ absolute value squared amplitudes of the sinusoids at
+ frequencies w
+
+
+
diff --git a/help/en_US/rootmusic.xml b/help/en_US/rootmusic.xml
new file mode 100644
index 0000000..82178e1
--- /dev/null
+++ b/help/en_US/rootmusic.xml
@@ -0,0 +1,62 @@
+
+
+
+
+
+
+
+ rootmusic
+ Frequencies and power of sinusoids using the root MUSIC algorithm
+
+
+
+
+ Calling Sequence
+
+ w = rootmusic(x,p)
+ [w,pow] = rootmusic(x,p)
+ [f,pow] = rootmusc(...,fs)
+ [w,pow] = rootmusic(...,'corr')
+
+
+
+
+
+ Parameters
+
+ Examples:
+
+ 1) 3 complex exponentials:
+
+ n=0:
+ 99;
+
+
+
+
+ See also
+
+ | peig | pmusic | rooteig
+ References
+ Monson H. Hayes, Statistical Digital Signal Processing And Modeling,
+ & Sons, Inc, [Section 8.6.3]
+ arguments
+ - double - vector
+ frequencies of the complex sinusoids
+ - double - vector
+ absolute value squared amplitudes of the sinusoids at
+ frequencies w
+
+
+
diff --git a/help/en_US/rssq.xml b/help/en_US/rssq.xml
new file mode 100644
index 0000000..251eb88
--- /dev/null
+++ b/help/en_US/rssq.xml
@@ -0,0 +1,121 @@
+
+
+
+
+
+
+
+ rssq
+ This function calculates the square root of the sum of values of input vector IN.
+
+
+
+
+ Calling Sequence
+
+ OUT=rssq(IN)
+ OUT=rssq(IN,orientation)
+
+
+
+
+
+ Parameters
+
+ in:
+ Vector or Matrix of real or complex elements.
+ orientation:
+ A string with possible values "r", "c" or "m" or numericals such as '1' or '2',giving the dimension along which the rssq value is to be calculated.
+ out:
+ A scalar with real value when input is a vector.When input is a matrix, out is the root sum squared value along the orientation specified or the default one when not specified.
+
+
+
+
+ Description
+
+For vector as input, the output is real valued scalar containing the rssq value. The rssq value can be calculated by taking the square root of the squared sum of the elements.
+If the input IN is a matrix, the output of function is rssq value of each column stored in a row vector OUT.
+
+
+When the elements of IN are COMPLEX, the absolute value of the element is used to calculate the output.
+When the orientation is not specified for N dimensional array, it is taken as the index of the first dimension of IN that is greater than 1 and calculation is done along that orientation.
+
+
+When the orientation is specified the output is calculated along that dimension.
+The orientation can be specified as 1 for rssq value of columns of matrix IN or as r.
+For rssq value of rows of matrix orientation should be 2 or c.
+
+
+
+
+
+
+ Examples
+
+
+
+
+ Examples
+
+
+
+
+ Examples
+
+
+
+
+ See also
+
+ abs
+ mean
+ sqrt
+ isempty
+
+
+
+
+ Authors
+
+ Indira Askaukar
+
+
+
+
+ Bibliography
+ Matlab help document.
+
+
diff --git a/help/en_US/sampled2continuous.xml b/help/en_US/sampled2continuous.xml
new file mode 100644
index 0000000..a08d04a
--- /dev/null
+++ b/help/en_US/sampled2continuous.xml
@@ -0,0 +1,60 @@
+
+
+
+
+
+
+
+ sampled2continuous
+ This function calculates the output reconstructed from the samples n supplied as input, at a rate of 1/s samples per unit time.
+
+
+
+
+ Calling Sequence
+
+ x = sampled2continuous (n, s, t)
+
+
+
+
+ Parameters
+
+ n:
+
+ s:
+
+ t:
+
+
+
+
+
+ Description
+
+This is an Octave function.
+This function calculates the output reconstructed from the samples n supplied as input, at a rate of 1/s samples per unit time.
+The third parameter t is all the instants where output x is needed from intput n and this time is relative to x(0).
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/sawtooth.xml b/help/en_US/sawtooth.xml
new file mode 100644
index 0000000..aee4088
--- /dev/null
+++ b/help/en_US/sawtooth.xml
@@ -0,0 +1,59 @@
+
+
+
+
+
+
+
+ sawtooth
+
+
+
+
+
+ Calling Sequence
+
+ [y]=sawtooth(t)
+ [y]=sawtooth(t,width)
+
+
+
+
+ Parameters
+
+ t:
+ Real valued vector or matrix
+ width:
+ Real number between 0 and 1
+
+
+
+
+ Description
+
+This is an Octave function
+This function returns a sawtooth wave with period 2*pi with +1/-1 as the maximum and minimum values for elements of t. If width is specified, it determines where the maximum is in the interval [0,2*pi].
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/schtrig.xml b/help/en_US/schtrig.xml
new file mode 100644
index 0000000..9e7b08b
--- /dev/null
+++ b/help/en_US/schtrig.xml
@@ -0,0 +1,61 @@
+
+
+
+
+
+
+
+ schtrig
+ This function implements a multisignal Schmitt triggers with lev levels supplied as input.
+
+
+
+
+ Calling Sequence
+
+ v = schtrig (x, lev)
+ v = schtrig (x, lev, rs)
+
+
+
+
+ Parameters
+
+ x:
+ vector or matrix of real numbers
+ lev:
+ real number
+ rs:
+ default value 1
+
+
+
+
+ Description
+
+This is an Octave function.
+This function implements a multisignal Schmitt triggers with lev levels supplied as input.
+The argument 1 is a matrix (or a vector) and this trigger works along its first dimension.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/schurrc.xml b/help/en_US/schurrc.xml
new file mode 100644
index 0000000..b269840
--- /dev/null
+++ b/help/en_US/schurrc.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ schurrc
+ narginchk(1,1,argn(2));
+
+
+
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new file mode 100644
index 0000000..677736a
--- /dev/null
+++ b/help/en_US/scilab_en_US_help/JavaHelpSearch/SCHEMA
@@ -0,0 +1,2 @@
+JavaSearch 1.0
+TMAP bs=2048 rt=1 fl=-1 id1=1967 id2=1
diff --git a/help/en_US/scilab_en_US_help/JavaHelpSearch/TMAP b/help/en_US/scilab_en_US_help/JavaHelpSearch/TMAP
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diff --git a/help/en_US/scilab_en_US_help/ac2poly.html b/help/en_US/scilab_en_US_help/ac2poly.html
new file mode 100644
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+++ b/help/en_US/scilab_en_US_help/ac2poly.html
@@ -0,0 +1,81 @@
+
Convert autocorrelation sequence to polynomial of prediction filter
+
+
+
Calling Sequence
+
a = ac2poly(r)
+[a,e] = ac2poly(r)
+
+
Parameters
+
r:
+
Autocorrelation sequence to be represented with an FIR linear prediction filter
+
a:
+
Output polynomial representing the linear prediction filter e/(a(1) + a(2)z + a(3)z^2 .. a(N)z^N-1)
+
e:
+
Output scaling for the lienar prediction filter
+
+
Description
+
Function ac2poly() finds the best fit polynomial for FIR linear prediction filter a, corresponding to the autocorrelation sequence r. a is the same length as r, and is normalized with the first element. So a(1) = 1.
+Author:
+Parthe Pandit
+
+
+
Bibliography
+
Kay, Steven M. Modern Spectral Estimation. Englewood Cliffs, NJ: Prentice-Hall, 1988.
string value, takes in "AKICc", "KIC", "AICc", "AIC" and "FPE", default it not using a model-selection criterion
+
a, v, k:
+
Output variables
+
+
Description
+
This is an Octave function.
+This function calculates coefficients of an autoregressive (AR) model of complex data x using the whitening lattice-filter method of Burg.
+The first argument is the data sampled. The second argument is the number of poles in the model (or limit in case a criterion is supplied).
+The third parameter takes in the criterion to limit the number of poles. The acceptable values are "AIC", "AKICc", "KIC", "AICc" which are based on information theory.
+Output variable a is a list of P+1 autoregression coefficients.
+Output variable v is the mean square of residual noise from the whitening operation of the Burg lattice filter.
+Output variable k corresponds to the reflection coefficients defining the lattice-filter embodiment of the model.
This is an Octave function.
+This function fits an AR (p)-model with Yule-Walker estimates.
+The first argument is the data vector which is to be estimated.
+Output variable a gives the AR coefficients, v gives the variance of the white noise and k gives the reflection coefficients to be used in the lattice filter.
This function returns the filter coefficients of a modified Bartlett-Hann window.
+
+
+
Calling Sequence
+
y = barthannwin(m)
+
+
Parameters
+
m:
+
positive integer value
+
y:
+
output variable, vector of real numbers
+
+
Description
+
This is an Octave function.
+This function returns the filter coefficients of a modified Bartlett Hann window of length m supplied as input, to the output vector y.
w=bartlett(L) returns an L-point Bartlett window in a column vector w
+Example
+w=bartlett(4)
+w =
+
0.
+0.6666667
+0.6666667
+0.
+Author
+Ankur Mallick
+References
+[1] Oppenheim, Alan V., Ronald W. Schafer, and John R. Buck. Discrete-Time Signal Processing. Upper Saddle River, NJ: Prentice Hall, 1999.
This is an Octave function.
+This function generates a Bessel filter. The default is a Laplace space (s) filter.
+The third parameter takes in high or low, the default value being low. The cutoff is pi*Wc radians.
+[z,p,g] = besself(...) returns filter as zero-pole-gain rather than coefficients of the numerator and denominator polynomials.
+[...] = besself(...,’z’) returns a discrete space (Z) filter. w must be less than 1.
+[a,b,c,d] = besself(...) returns state-space matrices.
This is an Octave function.
+This function returns the input data after reversing the bits of the indices and reordering the elements of the input array.
w=blackman(N) returns an N-point symmetric Blackman window in a column vector w
+w=blackman(N,sflag)
+Returns an N point Blackman window using the type of sampling specified by sflag
+sflag can be either 'symmetric' (default) or 'periodic' (used in spectral analysis)
+Example
+w=blackman(4)
+w =
+
- 1.388D-17
+0.63
+0.63
+- 1.388D-17
+Author
+Ankur Mallick
+References
+[1] Oppenheim, Alan V., Ronald W. Schafer, and John R. Buck. Discrete-Time Signal Processing. Upper Saddle River, NJ: Prentice Hall, 1999.
This function returns the filter coefficients of a Blackman-Harris window.
+
+
+
Calling Sequence
+
w = blackmanharris (m)
+w = blackmanharris (m, opt)
+
+
Parameters
+
m:
+
positive integer value
+
opt:
+
string value, takes "periodic" or "symmetric"
+
w:
+
output variable, vector of real numbers
+
+
Description
+
This is an Octave function.
+This function returns the filter coefficients of a Blackman-Harris window of length m supplied as input, to the output vector w.
+The second parameter can take the values "periodic" or "symmetric", depending on which the corresponding form of window is returned. The default is symmetric.
This function returns the filter coefficients of a rectangular window.
+
+
+
Calling Sequence
+
y = boxcar (m)
+
+
Parameters
+
m:
+
positive integer value
+
y:
+
output variable, vector of real numbers
+
+
Description
+
This is an Octave function.
+This function returns the filter coefficients of a rectangular window of length m supplied as input, to the output vector y.
This is an Octave function.
+This function generates a Butterworth filter. Default is a discrete space (Z) filter.
+The third parameter takes in low or high, default value is low. The cutoff is pi*Wc radians.
+[b,a] = butter(n, [Wl, Wh]) indicates a band pass filter with edges pi*Wl and pi*Wh radians.
+[b,a] = butter(n, [Wl, Wh], ’stop’) indicates a band reject filter with edges pi*Wl and pi*Wh radians.
+[z,p,g] = butter(...) returns filter as zero-pole-gain rather than coefficients of the numerator and denominator polynomials.
+[...] = butter(...,’s’) returns a Laplace space filter, w can be larger than 1.
+[a,b,c,d] = butter(...) returns state-space matrices.
scalar or vector of length 2, elements must be in the range [0,1]
+
Rp:
+
real or complex value
+
Rs:
+
real or complex value
+
+
Description
+
This is an Octave function.
+This function computes the minimum filter order of a Butterworth filter with the desired response characteristics.
+The filter frequency band edges are specified by the passband frequency wp and stopband frequency ws.
+Frequencies are normalized to the Nyquist frequency in the range [0,1].
+Rp is measured in decibels and is the allowable passband ripple, and Rs is also in decibels and is the minimum attenuation in the stop band.
+If ws>wp, the filter is a low pass filter. If wp>ws, the filter is a high pass filter.
+If wp and ws are vectors of length 2, then the passband interval is defined by wp the stopband interval is defined by ws.
+If wp is contained within the lower and upper limits of ws, the filter is a band-pass filter. If ws is contained within the lower and upper limits of wp the filter is a band-stop or band-reject filter.
Converts a cell array to a second order section matrix
+
+
+
Parameters
+
+
+
Description
+
s=cell2sos(c) converts a a cell array c = { {B1},{A1}, {B2},{A2}, ... {BL},{AL}}
+to an L-by-6 second-order-section matrix s given by:
+s = [B1 A1
+B2 A2
+...
+BL AL]
+numerator vector Bi and denominator vector Ai contains the coefficients of a
+linear or quadratic polynomial. If the polynomial is linear, the coefficients
+zero-padded on the right.
+[s,g]=cell2sos(c) estimates the gain from the leading term of the cell array
+c={ {[g1,g2]},{B1},{A1}, {B2},{A2}, ... {BL},{AL}} to give g=g1/g2 as the gain
+Example
+c=cell(1,5);
scalar or vector of length 2, all elements must be in the range [0,1]
+
Ws:
+
scalar or vector of length 2, all elements must be in the range [0,1]
+
Rp:
+
real value
+
Rs:
+
real value
+
+
Description
+
This is an Octave function.
+This function computes the minimum filter order of a Chebyshev type I filter with the desired response characteristics.
+Stopband frequency ws and passband frequency wp specify the the filter frequency band edges.
+Frequencies are normalized to the Nyquist frequency in the range [0,1].
+Rp is measured in decibels and is the allowable passband ripple and Rs is also measured in decibels and is the minimum attenuation in the stop band.
+If ws>wp then the filter is a low pass filter. If wp>ws, then the filter is a high pass filter.
+If wp and ws are vectors of length 2, then the passband interval is defined by wp and the stopband interval is defined by ws.
+If wp is contained within the lower and upper limits of ws, the filter is a band-pass filter. If ws is contained within the lower and upper limits of wp, the filter is a band-stop or band-reject filter.
scalar or vector of length 2, all elements must be in the range [0,1]
+
Ws:
+
scalar or vector of length 2, all elements must be in the range [0,1]
+
Rp:
+
real value
+
Rs:
+
real value
+
+
Description
+
This is an Octave function.
+This function computes the minimum filter order of a Chebyshev type II filter with the desired response characteristics.
+Stopband frequency ws and passband frequency wp specify the the filter frequency band edges.
+Frequencies are normalized to the Nyquist frequency in the range [0,1].
+Rp is measured in decibels and is the allowable passband ripple and Rs is also measured in decibels and is the minimum attenuation in the stop band.
+If ws>wp then the filter is a low pass filter. If wp>ws, then the filter is a high pass filter.
+If wp and ws are vectors of length 2, then the passband interval is defined by wp and the stopband interval is defined by ws.
+If wp is contained within the lower and upper limits of ws, the filter is a band-pass filter. If ws is contained within the lower and upper limits of wp, the filter is a band-stop or band-reject filter.
This function returns the filter coefficients of a Dolph-Chebyshev window.
+
+
+
Calling Sequence
+
w = chebwin (m)
+w = chebwin (m, at)
+
+
Parameters
+
m:
+
positive integer value
+
at:
+
real scalar value
+
w:
+
output variable, vector of real numbers
+
+
Description
+
This is an Octave function.
+This function returns the filter coefficients of a Dolph-Chebyshev window of length m supplied as input, to the output vector w.
+The second parameter is the stop band attenuation of the Fourier transform in dB. The default value is 100 dB.
This is an Octave function.
+This function generates a Chebyshev type I filter with rp dB of passband ripple.
+The fourth parameter takes in high or low, default value is low. The cutoff is pi*Wc radians.
+[b, a] = cheby1(n, Rp, [Wl, Wh]) indicates a band pass filter with edges pi*Wl and pi*Wh radians.
+[b, a] = cheby1(n, Rp, [Wl, Wh], ’stop’) indicates a band reject filter with edges pi*Wl and pi*Wh radians.
+[z, p, g] = cheby1(...) returns filter as zero-pole-gain rather than coefficients of the numerator and denominator polynomials.
+[...] = cheby1(...,’s’) returns a Laplace space filter, w can be larger than 1.
+[a,b,c,d] = cheby1(...) returns state-space matrices.
This is an Octave function.
+This function generates a Chebyshev type II filter with rs dB of stopband attenuation.
+The fourth parameter takes in high or low, default value is low. The cutoff is pi*Wc radians.
+[b, a] = cheby2(n, Rp, [Wl, Wh]) indicates a band pass filter with edges pi*Wl and pi*Wh radians.
+[b, a] = cheby2(n, Rp, [Wl, Wh], ’stop’) indicates a band reject filter with edges pi*Wl and pi*Wh radians.
+[z, p, g] = cheby2(...) returns filter as zero-pole-gain rather than coefficients of the numerator and denominator polynomials.
+[...] = cheby2(...,’s’) returns a Laplace space filter, w can be larger than 1.
+[a,b,c,d] = cheby2(...) returns state-space matrices.
string value, takes in "linear", "quadratic", "logarithmic"
+
phse:
+
+
+
Description
+
This is an Octave function.
+This function evaluates a chirp signal at time t. A chirp signal is a frequency swept cosine wave.
+The first argument is a vector of times to evaluate the chirp signal, second argument is the frequency at t=0, third argument is time t1 and fourth argument is frequency at t1.
+The fifth argument is the form which takes in values "linear", "quadratic" and "logarithmic", the sixth argument gives the phase shift at t=0.
This function calculates boundary indexes of clusters of 1’s.
+
+
+
Calling Sequence
+
c = clustersegment(s)
+
+
Parameters
+
s:
+
scalar, vector or matrix of real numbers (clusters of 1s)
+
c:
+
output variable, cell array of size 1 by N, where N is the number of rows in s
+
+
Description
+
This is an Octave function.
+This function calculates boundary indexes of clusters of 1’s.
+This function calculates the initial and end indices of the sequences of 1's present in the input argument.
+The output variable c is a cell array of size 1 by N, where N is the number of rows in s and each element has two rows indicating the initial index and end index of the cluster of 1's respectively. The indexing starts from 1.
input vector of size N for which correlation matrix of size m is to be calculated
+
m:
+
size of correlation matrix to be computed. Positive integer strictly smaller than the length of the input x
+
X:
+
data matrix as specified according to the input 'method'
+
s:
+
method for type of output matrix X
+
'autocorrelation':
+
(default) X is the (n + m)-by-(m + 1) rectangular Toeplitz matrix that generates an autocorrelation estimate for the leng th-n data vector x, derived using prewindowed and postwindowed data, based on an mth-order prediction error model.
+
'prewindowed':
+
X is the n-by-(m + 1) rectangular Toeplitz matrix that generates an autocorrelation estimate for the length-n data vector x, derived using prewindowed data, based on an mth-order prediction error model.
+
'postwindowed':
+
X is the n-by-(m + 1) rectangular Toeplitz matrix that generates an autocorrelation estimate for the length-n data vector x , derived using postwindowed data, based on an mth-order prediction error model.
+
'covariance':
+
X is the (n – m)-by-(m + 1) rectangular Toeplitz matrix that generates an autocorrelation estimate for the length-n data vect or x, derived using nonwindowed data, based on an mth-order prediction error model.
+
'modified':
+
X is the 2(n – m)-by-(m + 1) modified rectangular Toeplitz matrix that generates an autocorrelation estimate for the length-n d ata vector x, derived using forward and backward prediction error estimates, based on an mth-order prediction error model.
This is an Octave function
+dctmtx(n) returns a Discrete cosine transform matrix of order n-by-n. It is useful for jpeg image compression. D*A is the DCT of the columns of A and D'*A is the inverse DCT of the columns of A (when A is n-by-n).
This is an Octave function
+This fuction gives a complex matrix of values whose product with a vector produces the discrete Fourier transform. This can also be achieved by directly using the fft function i.e. y=fft(x) is same as y=A*x where A=dftmtx(n).
This function downsamples the signal by selecting every nth element.
+
+
+
Calling Sequence
+
y = downsample (x, n)
+y = downsample (x, n, phase)
+
+
Parameters
+
x:
+
scalar, vector or matrix of real or complex numbers
+
n:
+
real number or vector
+
phase:
+
integer value, 0 <= phase <= (n - 1), default value 0, or logical
+
+
Description
+
This is an Octave function.
+This function downsamples the signal by selecting every nth element supplied as parameter 2. If x is a matrix, the function downsamples every column.
+If the phase is specified, every nth element is selected starting from the sample phase. The default phase is 0.
scalar or vector, all elements should be in the range [0,1]
+
+
Description
+
This is an Octave function.
+This function generates an elliptic or Cauer filter with rp dB of passband ripple and rs dB of stopband attenuation.
+[b, a] = ellip(n, Rp, Rs, Wp) indicates low pass filter with order n, Rp decibels of ripple in the passband and a stopband Rs decibels down and cutoff of pi*Wp radians. If the fifth argument is high, then the filter is a high pass filter.
+[b, a] = ellip(n, Rp, Rs, [Wl, Wh]) indictaes band pass filter with band pass edges pi*Wl and pi*Wh. If the fifth argument is stop, the filter is a band reject filter.
+[z, p, g] = ellip(...) returns filter as zero-pole-gain.
+[...] = ellip(...,’s’) returns a Laplace space filter, w can be larger than 1.
+[a, b, c, d] = ellip(...) returns state-space matrices.
scalar or vector of length 2, all elements must be in the range [0,1]
+
Ws:
+
scalar or vector of length 2, all elements must be in the range [0,1]
+
Rp:
+
real or complex value
+
Rs:
+
real or complex value
+
+
Description
+
This is an Octave function.
+This function computes the minimum filter order of an elliptic filter with the desired response characteristics.
+Stopband frequency ws and passband frequency wp specify the the filter frequency band edges.
+Frequencies are normalized to the Nyquist frequency in the range [0,1].
+Rp is measured in decibels and is the allowable passband ripple and Rs is also measured in decibels and is the minimum attenuation in the stop band.
+If ws>wp then the filter is a low pass filter. If wp>ws, then the filter is a high pass filter.
+If wp and ws are vectors of length 2, then the passband interval is defined by wp and the stopband interval is defined by ws.
+If wp is contained within the lower and upper limits of ws, the filter is a band-pass filter. If ws is contained within the lower and upper limits of wp, the filter is a band-stop or band-reject filter.
Performs FFT-based FIR filtering using overlap-add method
+
+
+
Calling Sequence
+
+
+
Parameters
+
x:
+
real|complex numbers - vector|matrix
+
b:
+
real|complex numbers - vector|matrix
+
n:
+
positive integer
+
+
Description
+
y = fftfilt(b,x) filters the data in vector x with the filter described
+by coefficient vector b.
+y = fftfilt(b,x,n) uses n to determine the length of the FFT.
This is an Octave function
+In theory, it forwards and reverse filters the signal and corrects phase distortion upto an extent by a one-pass filter but squares the magnitude response in the process. Practically though, the correction isn't perfect and magnitude response, particularly the stop band is distorted.
This function finds the initial conditions for the delays in the transposed direct-form II filter implementation.
+The vectors b and a represent the numerator and denominator coefficients of the filter's transfer function.
Parks-McClennan optimal FIR filter order estimation
+
+
+
Calling Sequence
+
+
+
Parameters
+
f:
+
double - positive - vector
+
a:
+
double - positive - vector
+
dev:
+
double - positive - vector
+
n:
+
int - scalar
+
fo:
+
double - positive - vector
+
ao:
+
double - positive - vector
+
w:
+
double - vector
+
+
Examples
+
[1]Alow-passfilter
+f=[15002000];// frequency edges for bands
+a=[10];// desired amplitude for each band
+dev=[0.010.1];// Acceptable deviation for each band
+fs=8000;// Sampling frequency
+[n,fo,ao,w]=firpmord(f,a,dev,fs);
+
+[2]Abandstopfilter
+f=[1000180024003000];
+a=[100.5];
+dev=[0.010.10.03];
+fs=8000;
+[n,fo,ao,w]=firpmord(f,a,dev,fs);
+
+References
+[1]Rabiner,LawrenceR.,andBernardGold."Theoryandapplicationof
+digitalsignalprocessing." Englewood Cliffs, NJ, Prentice-Hall, Inc.,
+1975.777p.156-7(1975).
+[2]Rabiner,LawrenceR.,andOttoHerrmann."Thepredictabilityofcertain
+optimumfinite-impulse-responsedigitalfilters." Circuit Theory,
+IEEETransactionson20.4(1973):401-408.
This function returns the filter coefficients of a Flat Top window.
+
+
+
Calling Sequence
+
w = flattopwin (m)
+w = flattopwin (m, opt)
+
+
Parameters
+
m:
+
positive integer value
+
opt:
+
string value, takes in "periodic" or "symmetric"
+
w:
+
output variable, vector of real numbers
+
+
Description
+
This is an Octave function.
+This function returns the filter coefficients of a Flat Top window of length m supplied as input, to the output vector w.
+The second parameter can take the values "periodic" or "symmetric", depending on which the corresponding form of window is returned. The default is symmetric.
+This window has low pass-band ripple but a high bandwidth.
This is an Octave function.
+This function shifts the series x supplied as input argument by a number of samples d.
+The third parameter is the interpolator, which is designed with a Kaiser-windowed sinecard by default, if not supplied.
This function returns a Gaussian convolution window.
+
+
+
Calling Sequence
+
w = gaussian (m)
+w = gaussian (m, a)
+
+
Parameters
+
m:
+
positive integer value
+
a:
+
+
w:
+
output variable, vector of real numbers
+
+
Description
+
This is an Octave function.
+This function returns a Gaussian convolution window of length m supplied as input, to the output vector w.
+The second parameter is the width measured in sample rate/number of samples and should be f for time domain and 1/f for frequency domain. The width is inversely proportional to a.
This function returns the filter coefficients of a Gaussian window.
+
+
+
Calling Sequence
+
w = gausswin (m)
+w = gausswin (m, a)
+
+
Parameters
+
m:
+
positive integer value
+
a:
+
+
w:
+
output variable, vector of real numbers
+
+
Description
+
This is an Octave function.
+This function returns the filter coefficients of a Gaussian window of length m supplied as input, to the output vector w.
+The second parameter is the width measured in sample rate/number of samples and should be f for time domain and 1/f for frequency domain. The width is inversely proportional to a.
Computes DFT using the second order Goertzel Algorithm
+
+
+
Calling Sequence
+
Y = goertzel(X,INDVEC,DIM)
+
+
Parameters
+
+
+
Description
+
goertzel(X,INDVEC)
+Computes the DFT of X at indices INDVEC using the second order algorithm along
+the first non-singleton dimension. Elements of INDVEC must be positive integers
+less than the length of the first non-singleton dimension. If INDVEC is empty
+the DFT is computed at all indices along the first non-singleton dimension
+goertzel(X,INDVEC,DIM)
+Implements the algorithm along dimension DIM
+In general goertzel is slower than fft when computing the DFT for all indices
+along a particular dimension. However it is computationally more efficient when
+the DFT at only a subset of indices is desired
+Example
+x=rand(1,5)
+x =
This is an Octave function.
+This function returns the filter coefficients of a Hanning window of length m supplied as input, to the output vector w.
+The second parameter can take the values "periodic" or "symmetric", depending on which the corresponding form of window is returned. The default is symmetric.
This function does IIR Low Pass Filter to Multiband Filter Transformation.
+
+
+
Calling Sequence
+
[Num, Den, AllpassNum, AllpassDen] = iirlp2mb(B, A, Wo, Wt)
+[Num, Den, AllpassNum, AllpassDen] = iirlp2mb(B, A, Wo, Wt, Pass)
+
+
Parameters
+
B:
+
real or complex value
+
A:
+
real or complex value
+
Wo:
+
scalar or vector
+
Wt:
+
scalar or vector, elements must be monotonically increasing and >= 0 and <= 1.
+
+
Description
+
This is an Octave function.
+This function does IIR Low Pass Filter to Multiband Filter Transformation.
+The first two parameters give the numerator and denominator of the prototype low pass filter.
+The third parameter is the normalized angular frequency/pi to be transformed.
+The fourth parameter is the normalized angular frequency/pi target vector.
+The first two output variables are the numerator and denominator of the transformed filter.
+The third and fourth output variables are the numerator and denominator of the allpass transform.
+The fifth parameter can have values pass or stop, default value is pass.
This function converts analog filter with coefficients b and a to digital, conserving impulse response.
+
+
+
Calling Sequence
+
[b, a] = impinvar (b, a)
+[b, a] = impinvar (b, a, fs)
+[b, a] = impinvar (b, a, fs, tol)
+
+
Parameters
+
b:
+
real or complex valued scalar or vector
+
a:
+
real or complex valued scalar or vector, order should be greater than b
+
fs:
+
real or complex value, default value 1Hz
+
tol:
+
real or complex value, default value 0.0001
+
+
Description
+
This is an Octave function.
+This function converts analog filter with coefficients b and a to digital, conserving impulse response.
+This function does the inverse of impinvar.
y = interp(x, q)
+y = interp(x, q, n)
+y = interp(x, q, n, Wc)
+
+
Parameters
+
x:
+
scalar or vector of complex or real numbers
+
q:
+
positive integer value, or logical
+
n:
+
positive integer, default value 4
+
Wc:
+
non decreasing vector or scalar, starting from 0 uptill 1, default value 0.5
+
+
Description
+
This is an Octave function.
+This function upsamples the signal x by a factor of q, using an order 2*q*n+1 FIR filter.
+The second argument q must be an integer. The default values of the third and fourth arguments (n, Wc) are 4 and 0.5 respectively.
This function converts digital filter with coefficients b and a to analog, conserving impulse response.
+
+
+
Calling Sequence
+
[b, a] = impinvar (b, a)
+[b, a] = impinvar (b, a, fs)
+[b, a] = impinvar (b, a, fs, tol)
+
+
Parameters
+
b:
+
real or complex valued scalar or vector
+
a:
+
real or complex valued scalar or vector, order should be greater than b
+
fs:
+
real or complex value, default value 1Hz
+
tol:
+
real or complex value, default value 0.0001
+
+
Description
+
This is an Octave function.
+This function converts digital filter with coefficients b and a to analog, conserving impulse response.
+This function does the inverse of impinvar.
This function returns the filter coefficients of a Kaiser window.
+
+
+
Calling Sequence
+
w = kaiser (m)
+w = kaiser (m, beta)
+
+
Parameters
+
m:
+
positive integer value
+
beta:
+
real scalar value
+
w:
+
output variable, vector of real numbers
+
+
Description
+
This is an Octave function.
+This function returns the filter coefficients of a Kaiser window of length m supplied as input, to the output vector w.
+The second parameter gives the stop band attenuation of the Fourier transform of the window on derivation.
Convert lattice filter parameters to transfer function coefficients
+
+
+
Calling Sequence
+
+
+
Description
+
[num,den] = latc2tf(k,v)
+Finds the transfer function of the IIR filter from the lattice
+coefficients k and ladder coefficients v.
+[num,den] = latc2tf(k,'iiroption')
+Finds the transfer function of the allpass or allpole (specified by
+the iiroption flag) IIR filter.
+num = latc2tf(k,'firoption')
+Finds the transfer function of the FIR filter from the lattice
+coefficients k. The firoption flag specifies the type of the FIR
+filter (can be 'min, 'max', or 'FIR')
+
Parameters:
+k - double - vector
+Lattice coefficients
+Lattice coefficients for FIR/IIR filter. Can be real or complex.
+v - double - vector
+Ladder coefficients
+Ladder coefficients for IIR filters. Can be real or complex.
+iiroption - string flag - 'allpole', or 'allpass'
+Specification of the type if IIR filter
+firoption - string flag - 'min', 'max', or 'FIR' (default)
+Speficication of the type of FIR filter
[a,g] = lpc(x,p)
+Determines the coefficients of a pth order forward linear predictor
+filter by minimizing the squared error. If p is unspecified, a
+default value of length(x)-1 is used.
This function computes the generalized Marcum Q function of order m with noncentrality parameter a and argument b.
+
+
+
Calling Sequence
+
q = marcumq (a, b)
+q = marcumq (a, b, m)
+q = marcumq (a, b, m, tol)
+
+
Parameters
+
a:
+
+
b:
+
+
m:
+
default value 1
+
tol:
+
default value eps
+
+
Description
+
This is an Octave function.
+This function computes the generalized Marcum Q function of order m with noncentrality parameter a and argument b.
+The third argument m is the order, which by default is 1.
+The fourth argument tol is the tolerance, which by default is eps.
+If input arguments are vectors which correspond in size and degree, the output is a table of values.
+This function calculates Marcum’s Q function using the infinite Bessel series, which is truncated when the relative error is less than the specified tolerance.
y = medfilt1(x)
+Applies a 3rd order 1-dimensional median filter to input x along the
+first non-zero dimension. The function appropriately pads the signal
+with zeros at the endings. For a segment, a median is calculated as
+the middle value (average of two middle values) for odd number
+number (even number) of data points.
+y = medfilt1(x,n)
+Applies a nth order 1-dimensional median filter.
+y = medfilt1(x,n,dim)
+Applies the median filter along the n-th dimension
+y = medfilt1(__, nanflag, padding)
+nanflag specifies how NaN values are treated. padding specifies the
+type of filtering to be performed at the signal edges.
+
+
+
Parameters
+
x:
+
int | double
+
n:
+
positive integer scalar
+
dim:
+
positive integer scalar
+
nanflag:
+
'includenan' (default) | 'omitnan'
+
* includenan:
+
Filtering such that the median of any segment
+
* omitnan:
+
Filtering with NaNs omitted in each segment. If a segment
[y,t]=modulate(x,fc,fs,method,opt)
+Returns the modulated vector y and the time vector 't'
+Modulation is performed according to the following table
+METHOD MODULATION SCHEME
+'am', Amplitude modulation, double side-band, suppressed carrier
+'amdsb-sc' opt not used. This is the default method.
+'amdsb-tc' Amplitude modulation, double side-band, transmitted carrier
+opt is a scalar subtracted from x before multiplying x
+with the carrier wave. It defaults to min(min(x)) so that
+the input signal after offset is always non-negative
+'amssb' Amplitude modulation, single side-band
+OPT not used.
+'fm' Frequency modulation
+opt is the constant of frequency modulation.
+opt = (fc/fs)*2*pi/max(max(abs(x))) by default
+'pm' Phase modulation
+OPT is the constant of phase modulation.
+opt = pi/max(max(abs(x))) by default
+Phase lies between -pi and +pi
+'pwm' Pulse width modulation
+opt='left' corresponds to left justified pulses.
+opt='centered' correspondes to centered pulses.
+The default value of opt is 'left'.
+'ppm' Pulse position modulation
+opt is a scalar between 0 and 1 which specifies the pulse
+width in fractions of the carrier period with default value 0.1.
+'qam' Quadrature amplitude modulation
+opt is a matrix of the same size as X which is modulated in
+quadrature with x.
+
If x is a matrix, its columns are modulated.
+Example
+y =
This is an Octave function.
+The signal is convoluted against a sigmoid window of width w and risetime rc with the units of these parameters relative to the value of the sampling frequency given in Fs (Default value=1).
This function returns the filter coefficients of a Blackman-Harris window.
+
+
+
Calling Sequence
+
w = nuttallwin (m)
+w = nuttallwin (m, opt)
+
+
Parameters
+
m:
+
positive integer value
+
opt:
+
string value, takes in "periodic" or "symmetric"
+
w:
+
output variable, vector of real numbers
+
+
Description
+
This is an Octave function.
+This function returns the filter coefficients of a Blackman-Harris window defined by Nuttall of length m supplied as input, to the output vector w.
+The second parameter can take the values "periodic" or "symmetric", depending on which the corresponding form of window is returned. The default is symmetric.
This function calculates the ratio of peak magnitude to the Root Mean Square(RMS) value.
+
+
+
Calling Sequence
+
OUT=peak2rms(IN)
+OUT=peak2rms(IN,orientation)
+
+
Parameters
+
in:
+
Vector or Matrix of real or complex elements.
+
orientation:
+
A string with possible values "r", "c" or "m" giving the dimension along which the peak2rms value is to be calculated.
+
out:
+
A scalar with real value when input is a vector.When input is a matrix, out is the peak magnitude to RMS value along the orientation specified or the default one when not specified.
+
+
Description
+
For vector as input, the output is the ratio of peak value to the RMS value. The RMS value can be calculated by taking the square root of mean value of the squared sum of the elements.
+
When a matrix is given as input the output is peak to RMS ratio in the orientation specified.
+The orientation can be given as string with values "r","c" or "m".
+
peak2rms(in, 1) calculates the values of ratio of peak to RMS of columns of matrix. The output in this case is a row vector with peak2rms value of each column of in.
+
peak2rms(in, 2) calculates the values of ratio of peak to RMS of rows of matrix, where the output would be a column vector having peak2rms value of each row of in.
+
The default orientation is chosen to be the index of first dimension of input greater than 1.Hence peak2rms(in) is equivalent to peak2rms(in, "m").
+
For an N dimensional array the orientation is the index of first non singleton dimension of the array.
+
If the elements of matrix are complex the absolute values are considered in the calculation of RMS value.
Convert prediction polynomial to autocorrelation sequence.
+
+
+
Calling Sequence
+
R = poly2ac(a,efinal)
+
+
Parameters
+
a:
+
input prediction polynomial with 1st element 1 (if not, poly2ac normalizes it to 1 before proceeding).
+
efinal:
+
input prediction error
+
r:
+
output autocorrelation sequence
+
+
Description
+
This function obtains the underlying autocorrelation sequence that would best fit a linear prediction filter described by the
+denominator polynomial and the numerator scaling. The filter is H(z) = efinal/(a(1) + a(2) x z a(3) x z^2 ... a(n) x z^n-1)
This function stabilizes the polynomial transfer function.
+
+
+
Calling Sequence
+
b = polystab(a)
+
+
Parameters
+
a:
+
+
+
Description
+
This is an Octave function.
+This function stabilizes the polynomial transfer function by replacing all roots outside the unit circle with their reflection inside the unit circle.
This function calculates the primitive of a given function supplied as input.
+
+
+
Calling Sequence
+
y = primitive(f, t)
+y = primitive(f, t, x)
+
+
Parameters
+
f:
+
+
t:
+
+
+
Description
+
This is an Octave function.
+This function calculates the primitive of a given function supplied as input.
+The second parameter t is a vector at which the output is evaluated (at the points t). This vector should be ascending and ordered.
+The function approximates the primitive (indefinite integral) of the univariate function handle f with constant of integration x.
This is an Octave function.
+This function generates the signal y = sum(func(t+d,...)) for each d. If d is a matrix of two columns, the first column is the delay d and the second column is the amplitude a, and y = sum(a*func(t+d)) for each d, a. Here, func is a function which accepts a vector of times.
+If a pulse shape sampled at frequency Fs (default 1 Hz) is supplied instead of a function name, an interpolated version of the pulse is added at each delay d.
This is an Octave function
+y = rectpuls(t) returns a continuous, aperiodic, unity-height rectangular pulse depending upon input t, centered about t=0 and having default width of 1.
+y = rectpuls(t,w) generates a rectangle of width w.
This function returns the filter coefficients of a rectangular window.
+
+
+
Calling Sequence
+
y = rectwin (m)
+
+
Parameters
+
m:
+
positive integer value
+
y:
+
output variable, vector of real numbers
+
+
Description
+
This is an Octave function.
+This function returns the filter coefficients of a rectangular window of length m supplied as input, to the output vector y.
This function resamples in the input sequence x supplied by a factor of p/q.
+
+
+
Calling Sequence
+
y = resample(x, p, q)
+y = resample(x, p, q, h)
+[y, h] = resample(...)
+
+
Parameters
+
x:
+
scalar, vector or matrix of real or complex numbers
+
p:
+
positive integer value
+
q:
+
positive integer value
+
h:
+
scalar, vector or matrix of real or complex numbers
+
+
Description
+
This is an Octave function.
+This function resamples in the input sequence x supplied by a factor of p/q. If x is a matrix, then every column is resampled.hange the sample rate of x by a factor of p/q.
+This is performed using a polyphase algorithm. The impulse response h, given as parameter 4, of the antialiasing filter is either specified or designed with a Kaiser-windowed sinecard.
This is an Octave function.
+Similar to the "residuez" function. The difference being in the function "residuez", the IIR part (poles p and residues r) is driven in parallel with the FIR part(f) whereas in the function "residued", the IIR part is driven by the output of the FIR part. In signal modeling applications, this structure can be more accurate.
This function calculates the square root of the sum of values of input vector IN.
+
+
+
Calling Sequence
+
OUT=rssq(IN)
+OUT=rssq(IN,orientation)
+
+
Parameters
+
in:
+
Vector or Matrix of real or complex elements.
+
orientation:
+
A string with possible values "r", "c" or "m" or numericals such as '1' or '2',giving the dimension along which the rssq value is to be calculated.
+
out:
+
A scalar with real value when input is a vector.When input is a matrix, out is the root sum squared value along the orientation specified or the default one when not specified.
+
+
Description
+
For vector as input, the output is real valued scalar containing the rssq value. The rssq value can be calculated by taking the square root of the squared sum of the elements.
+If the input IN is a matrix, the output of function is rssq value of each column stored in a row vector OUT.
+
When the elements of IN are COMPLEX, the absolute value of the element is used to calculate the output.
+When the orientation is not specified for N dimensional array, it is taken as the index of the first dimension of IN that is greater than 1 and calculation is done along that orientation.
+
When the orientation is specified the output is calculated along that dimension.
+The orientation can be specified as 1 for rssq value of columns of matrix IN or as r.
+For rssq value of rows of matrix orientation should be 2 or c.
This function calculates the output reconstructed from the samples n supplied as input, at a rate of 1/s samples per unit time.
+
+
+
Calling Sequence
+
x = sampled2continuous (n, s, t)
+
+
Parameters
+
n:
+
+
s:
+
+
t:
+
+
+
Description
+
This is an Octave function.
+This function calculates the output reconstructed from the samples n supplied as input, at a rate of 1/s samples per unit time.
+The third parameter t is all the instants where output x is needed from intput n and this time is relative to x(0).
This is an Octave function
+This function returns a sawtooth wave with period 2*pi with +1/-1 as the maximum and minimum values for elements of t. If width is specified, it determines where the maximum is in the interval [0,2*pi].
This function implements a multisignal Schmitt triggers with lev levels supplied as input.
+
+
+
Calling Sequence
+
v = schtrig (x, lev)
+v = schtrig (x, lev, rs)
+
+
Parameters
+
x:
+
vector or matrix of real numbers
+
lev:
+
real number
+
rs:
+
default value 1
+
+
Description
+
This is an Octave function.
+This function implements a multisignal Schmitt triggers with lev levels supplied as input.
+The argument 1 is a matrix (or a vector) and this trigger works along its first dimension.
[p,num]=seqperiod(x)
+Returns an integer p such that x(1:p) is the smallest subsequence that repeats in x
+The number of times the subsequence repeats is returned in num (may not be an integer)
+Repetitions may be incomplete at the end of the sequence but no breaks are permitted between repetitions
+If there is no subsequence that repeats in x then p=length(x)
+If x is a matrix or n-dimesnional array, the function operates along the first non-singleton dimension of x
This function computes the filter coefficients for all Savitzsky-Golay smoothing filters.
+
+
+
Calling Sequence
+
F = sgolay (p, n)
+F = sgolay (p, n, m)
+F = sgolay (p, n, m, ts)
+
+
Parameters
+
p:
+
polynomial
+
n:
+
odd integer value, larger than polynomial p
+
m:
+
positive integer less than 2^31 or logical
+
ts:
+
real or complex value
+
+
Description
+
This is an Octave function.
+This function computes the filter coefficients for all Savitzsky-Golay smoothing filters of order p for length n (odd).
+m can be used in order to get directly the mth derivative; ts is a scaling factor.
y = sgolayfilt (x)
+y = sgolayfilt (x, p)
+y = sgolayfilt (x, p, n)
+y = sgolayfilt (x, p, n, m)
+y = sgolayfilt (x, p, n, m, ts)
+
+
Parameters
+
x:
+
vector or matrix of real or complex numbers
+
p:
+
polynomial order, real number less than n, default value 3
+
n:
+
integer, odd number greater than p
+
m:
+
vector of real positive valued numbers, length n
+
ts:
+
real number, default value 1
+
+
Description
+
This function applies a Savitzky-Golay FIR smoothing filter to the data given in the vector x; if x is a matrix, this function operates
+on each column.
+The polynomial order p should be real, less than the size of the frame given by n.
+m is a weighting vector with default value identity matrix.
+ts is the dimenstion along which the filter operates. If not specified, the function operates along the first non singleton dimension.
Real or complex valued vector or matrix, strictly positive value for scalar input
+
fc:
+
Real or complex valued vector or matrix, strictly positive value for scalar input
+
+
Description
+
This is an Octave function
+This function implements the complex Shannon wavelet function and returns the value obtained. The complex Shannon wavelet is defined by a bandwidth parameter FB, a wavelet center frequency FC on an N point regular grid in the interval [LB,UB].
[y,perm,nshifts]=shiftdata(x,dim)
+Shifts the entries along dimension dim in x to the first column and returns the permutation vector in perm
+[y,perm,nshifts]=shiftdata(x)
+Shifts the entries along dimension dim in x to the first column and returns the number of shifts in nshifts
+
+
Examples
+
//When dim is specified:
+x=testmatrix('magi',3)
+x=
+
+8.1.6.
+3.5.7.
+4.9.2.
+[y,perm,nshifts]=shiftdata(x,2)
+nshifts=
+
+[]
+perm=
+
+2.1.
+y=
+
+8.3.4.
+1.5.9.
+6.7.2.
+//When dim is not specified:
+x=1:5
+x=
+
+1.2.3.4.5.
+[y,perm,nshifts]=shiftdata(x)
+nshifts=
+
+1.
+perm=
+
+[]
+y=
+
+1.
+2.
+3.
+4.
+5.
Converts a second order section matrix to a cell array
+
+
+
Parameters
+
+
+
Description
+
c=sos2cell(s) converts an L-by-6 second-order-section matrix s given by:
+s = [B1 A1
+B2 A2
+...
+BL AL]
+to a cell array c = { {B1},{A1}, {B2},{A2}, ... {BL},{AL}} where each
+numerator vector Bi and denominator vector Ai contains the coefficients of a
+linear or quadratic polynomial. If the polynomial is linear, the coefficients
+zero-padded on the right
+c=sos2cell(s,g) adds a leading gain term to the start of the cell array as:
+c={ {[g,1]},{B1},{A1}, {B2},{A2}, ... {BL},{AL}}
+Example
+s=rand(2,6)
+s =
This is an Octave function.
+This function converts series second-order sections to direct H(z) = B(z)/A(z) form.
+The input is the sos matrix and the second parameter is the overall gain, default value of which is 1.
+The output is a vector.
This is an Octave function.
+This function converts series second-order sections to zeros, poles, and gains (pole residues).
+The input is the sos matrix and the second parameter is the overall gain, default value of which is 1.
+The outputs are z, p, k. z and p are column vectors containing zeros and poles respectively, and k is the overall gain.
This is an Octave function
+Second order section digital filter sos is applied to the input vector and the output vector obtained is of the same length.
strips(x)
+Plots a vector x in horizontal strips of length 250
+If x is a matrix, it plots each column of x on a separate strip with the leftmost
+column as the topmost strip
+strips(x,sd)
+Plots x in strips of length sd samples each
+strips(x,sd,fs)
+Plots x in strips of duration sd seconds with sampling frequency fs (in Hz)
+strips(x,sd,fs,scale)
+Plots x in strips as above, and scales the vertical axis by scale
+If x is a matrix, strips uses a column vector of all the elements of x for the strip plot
+If x has complex entries, only the real part of those entries are considered
+Author
+Ankur Mallick
This function converts direct-form filter coefficients to series second-order sections.
+
+
+
Calling Sequence
+
[sos] = tf2sos (b, a)
+[sos, g] = tf2sos (b, a)
+
+
Parameters
+
b:
+
matrix of real numbers
+
a:
+
matrix of real numbers
+
+
Description
+
This is an Octave function.
+This function converts direct-form filter coefficients to series second-order sections.
+The input parameters b and a are vectors specifying the digital filter H(z) = B(z)/A(z).
+The output is the sos matrix and the overall gain.
+If there is only one output argument, the overall filter gain is applied to the first second-order section in the sos matrix.
This function returns the filter coefficients of a triangular window.
+
+
+
Calling Sequence
+
y = triang (m)
+
+
Parameters
+
m:
+
positive integer value
+
y:
+
output variable, vector of real numbers
+
+
Description
+
This is an Octave function.
+This function returns the filter coefficients of a triangular window of length m supplied as input, to the output vector y.
This function generates a triangular pulse which is sampled at times t over the interval [-w/2,w/2]. The value of skew lies between -1
+and 1.
+The value of skew represents the relative placement of the peak in the given width.
This function returns the filter coefficients of a Tukey window.
+
+
+
Calling Sequence
+
w = tukeywin (m)
+w = tukeywin (m, r)
+
+
Parameters
+
m:
+
positive integer
+
r:
+
positive real number, between 0 and 1
+
+
Description
+
This is an Octave function.
+This function returns the filter coefficients of a Tukey window of length m supplied as input, to the output vector w.
+The second parameter r defines the ratio between the constant and cosine section and its value has to be between 0 and 1, with default value 0.5.
Uniformly decodes the input vector or n-dimensional array of integers u with peak values +/- v
+If u has only positive values, the range of integers is assumed to be [0,2^n-1]
+If u has positive and negative values the range of integers is assumed to be [-2^(n-1),2^(n-1)-1]
+If v is not specified, its default value is 1
+If saturatemode='wrap' the output is wrapped using modulo arithmetic if overflow occurs
+If saturatemode='saturate' the output is saturated if overflow accors
+Example
+u = int8([-1 1 2 -5]);
+ysat = udecode(u,3)
+ysat =
+
- 0.25 0.25 0.5 - 1.
+Author
+Ankur Mallick
+[1] International Telecommunication Union. General Aspects of Digital Transmission Systems: Vocabulary of Digital Transmission and Multiplexing, and Pulse Code Modulation (PCM) Terms. ITU-T Recommendation G.701. March, 1993.
Performs uniform quantization of the input into 2^n levels
+
+
+
Calling Sequence
+
y=uencode(u,n,v,'signflag')
+
+
Parameters
+
+
+
Description
+
Uniformly quantizes the input vector or n-dimensional array u into 2^n levels in the interval [-v,v]
+If v is not specified, its default value is 1
+'signflag' is a string that determines the nature of the quantization
+If signflag='unsigned' then y contains unsigned integers in the range [0,2^n-1] corresponding to the 2^n levels
+If signflag='unsigned' then y contains signed integers in the range [-2^(n-1),2^(n-1)-1] corresponding to the 2^n levels
+The size of the integers in y in bits(8,16, or 32) depends on the value of n
+If the input lies beyond +/- v it is saturated
+Example
+y=uencode(-1:0.5:1,3)
+y =
+
0 2 4 6 7
+Author
+Ankur Mallick
+References
+[1] International Telecommunication Union. General Aspects of Digital Transmission Systems: Vocabulary of Digital Transmission and Multiplexing, and Pulse Code Modulation (PCM) Terms. ITU-T Recommendation G.701. March, 1993.
This is an Octave function.
+This function returns the coefficients of an Ultraspherical window of length m supplied as input, to the output vector w.
+The second parameter controls the ratio between side lobe to side lobe of the window's Fourier transform.
+The third parameter controls the ratio between main lobe width to side lobe. The default value is beta.
+The value of xmu is also returned for given beta, att or latt.
This function upsamples the input data, applies the FIR filter and then downsamples it.
+
+
+
Calling Sequence
+
y = upfirdn (x, h, p, q)
+
+
Parameters
+
x:
+
+
h:
+
+
p:
+
+
q:
+
+
+
Description
+
This is an Octave function.
+This function upsamples the input data in the matrix by a factor of n. Then the upsampled data is FIR filtered. After this, the resultant is downsampled.
This function upsamples the signal, inserting n-1 zeros between every element.
+
+
+
Calling Sequence
+
y = upsample (x, n)
+y = upsample (x, n, phase)
+
+
Parameters
+
x:
+
scalar, vector or matrix of real or complex numbers
+
n:
+
real number or vector
+
phase:
+
integer value, 0 <= phase <= (n - 1 ), default value 0, or logical
+
+
Description
+
This is an Octave function.
+This function upsamples the signal, inserting n-1 zeros between every element. If x is a matrix, every column is upsampled.
+The phase determines the position of the inserted sample in the block of zeros. The default value is 0.
scalar, vector or matrix of real or complex numbers
+
w:
+
scalar or vector of real or complex values
+
cpy:
+
can take in "true" or "false", default is false
+
+
Description
+
This is an Octave function.
+This function upsamples a vector interleaving given values or copies of the vector elements.
+The second argument has the values in the vector w that are placed in between the elements of x.
+The third argument, if true, means that w should be scalar and that each value in x repeated w times.
y=vco(x,fc,fs)
+Creates a frequency modulated cosine wave y whose frequency varies as the magnitude of x
+x lies in [-1,1]. x=-1 corresponds to a frequency of 0, x=0 corresponds to a frequency of fc
+and x=1 corresponds to a frequency of 2fc.
+y=vco(x,[fmin fmax],fs)
+Scales the frequency range so that x=-1 corresponds to a frequency of fmin and
+x=1 corresponds to a frequency of fmax
+If x is a matrix the same operation is performed on the columns on x
+Size of y is the same as the size of x
+Example
+x=rand()
+x =
This function returns the filter coefficients of a Welch window.
+
+
+
Calling Sequence
+
w = welchwin (m)
+w = welchwin (m, opt)
+
+
Parameters
+
m:
+
positive integer value
+
opt:
+
string value, takes "periodic" or "symmetric"
+
w:
+
output variable, vector of real numbers
+
+
Description
+
This is an Octave function.
+This function returns the filter coefficients of a Welch window of length m supplied as input, to the output vector w.
+The second parameter can take the values "periodic" or "symmetric", depending on which the corresponding form of window is returned. The default is symmetric.
+For symmetric, the length should be an integer>2. For periodic, the length should be an integer>1.
This function creates an m-point window from the function f given as input.
+
+
+
Calling Sequence
+
w = window(f, m)
+w = window(f, m, opts)
+
+
Parameters
+
f:
+
string value
+
m:
+
positive integer value
+
opts:
+
string value, takes in "periodic" or "symmetric"
+
w:
+
output variable, vector of real numbers
+
+
Description
+
This is an Octave function.
+This function creates an m-point window from the function f given as input, in the output vector w.
+f can take any valid function as a string, for example "blackmanharris".
Real, complex or string type input vector or matrix
+
l:
+
Length of matrix required
+
opt:
+
Character input to determine which side to extract from
+
+
Description
+
This is an Octave function
+[y]=wkeep(x,l) extracts a vector of length l from the centre of input vector x.
+[y]=wkeep(x,l,opt) extracts vector based on opt which could be 'l','r' or 'c' (left, right or central).
This is an Octave function.
+This function converts filter poles and zeros to second-order sections.
+The first and second parameters are column vectors containing zeros and poles. The third parameter is the overall filter gain, the default value of which is 1.
+The output is the sos matrix and the overall gain.
+If there is only one output argument, the overall filter gain is applied to the first second-order section in the sos matrix.
+
+
diff --git a/help/en_US/seqperiod.xml b/help/en_US/seqperiod.xml
new file mode 100644
index 0000000..f2f7e33
--- /dev/null
+++ b/help/en_US/seqperiod.xml
@@ -0,0 +1,88 @@
+
+
+
+
+
+
+
+ seqperiod
+ Calculates the period of a sequence
+
+
+
+
+ Calling Sequence
+
+ [p,num]=seqperiod(x)
+
+
+
+
+ Parameters
+
+ x:
+ A vector matrix or n-dimensional array
+
+
+
+
+ Description
+
+[p,num]=seqperiod(x)
+Returns an integer p such that x(1:p) is the smallest subsequence that repeats in x
+The number of times the subsequence repeats is returned in num (may not be an integer)
+Repetitions may be incomplete at the end of the sequence but no breaks are permitted between repetitions
+If there is no subsequence that repeats in x then p=length(x)
+If x is a matrix or n-dimesnional array, the function operates along the first non-singleton dimension of x
+
+
+
+
+ Examples
+
+
+
+
+ Authors
+
+ Ankur Mallick
+
+
+
diff --git a/help/en_US/sgolay.xml b/help/en_US/sgolay.xml
new file mode 100644
index 0000000..676b931
--- /dev/null
+++ b/help/en_US/sgolay.xml
@@ -0,0 +1,66 @@
+
+
+
+
+
+
+
+ sgolay
+ This function computes the filter coefficients for all Savitzsky-Golay smoothing filters.
+
+
+
+
+ Calling Sequence
+
+ F = sgolay (p, n)
+ F = sgolay (p, n, m)
+ F = sgolay (p, n, m, ts)
+
+
+
+
+ Parameters
+
+ p:
+ polynomial
+ n:
+ odd integer value, larger than polynomial p
+ m:
+ positive integer less than 2^31 or logical
+ ts:
+ real or complex value
+
+
+
+
+ Description
+
+This is an Octave function.
+This function computes the filter coefficients for all Savitzsky-Golay smoothing filters of order p for length n (odd).
+m can be used in order to get directly the mth derivative; ts is a scaling factor.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/sgolayfilt.xml b/help/en_US/sgolayfilt.xml
new file mode 100644
index 0000000..23ca7d5
--- /dev/null
+++ b/help/en_US/sgolayfilt.xml
@@ -0,0 +1,75 @@
+
+
+
+
+
+
+
+ sgolayfilt
+
+
+
+
+
+ Calling Sequence
+
+ y = sgolayfilt (x)
+ y = sgolayfilt (x, p)
+ y = sgolayfilt (x, p, n)
+ y = sgolayfilt (x, p, n, m)
+ y = sgolayfilt (x, p, n, m, ts)
+
+
+
+
+ Parameters
+
+ x:
+ vector or matrix of real or complex numbers
+ p:
+ polynomial order, real number less than n, default value 3
+ n:
+ integer, odd number greater than p
+ m:
+ vector of real positive valued numbers, length n
+ ts:
+ real number, default value 1
+
+
+
+
+ Description
+
+This function applies a Savitzky-Golay FIR smoothing filter to the data given in the vector x; if x is a matrix, this function operates
+on each column.
+The polynomial order p should be real, less than the size of the frame given by n.
+m is a weighting vector with default value identity matrix.
+ts is the dimenstion along which the filter operates. If not specified, the function operates along the first non singleton dimension.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/shanwavf.xml b/help/en_US/shanwavf.xml
new file mode 100644
index 0000000..3a633c3
--- /dev/null
+++ b/help/en_US/shanwavf.xml
@@ -0,0 +1,67 @@
+
+
+
+
+
+
+
+ shanwavf
+
+
+
+
+
+ Calling Sequence
+
+ [psi,x]=shanwavf(lb,ub,n,fb,fc)
+
+
+
+
+ Parameters
+
+ lb:
+ Real or complex valued vector or matrix
+ ub:
+ Real or complex valued vector or matrix
+ n:
+ Real valued integer strictly positive
+ fb:
+ Real or complex valued vector or matrix, strictly positive value for scalar input
+ fc:
+ Real or complex valued vector or matrix, strictly positive value for scalar input
+
+
+
+
+ Description
+
+This is an Octave function
+This function implements the complex Shannon wavelet function and returns the value obtained. The complex Shannon wavelet is defined by a bandwidth parameter FB, a wavelet center frequency FC on an N point regular grid in the interval [LB,UB].
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/shiftdata.xml b/help/en_US/shiftdata.xml
new file mode 100644
index 0000000..84bdb79
--- /dev/null
+++ b/help/en_US/shiftdata.xml
@@ -0,0 +1,99 @@
+
+
+
+
+
+
+
+ shiftdata
+ Shifts data by rearranging dimensions
+
+
+
+
+ Calling Sequence
+
+
+
+
+
+ Parameters
+
+
+
+
+
+ Description
+
+[y,perm,nshifts]=shiftdata(x,dim)
+Shifts the entries along dimension dim in x to the first column and returns the permutation vector in perm
+[y,perm,nshifts]=shiftdata(x)
+Shifts the entries along dimension dim in x to the first column and returns the number of shifts in nshifts
+
+
+
+
+ Examples
+
+
+
+
+ See also
+
+ permute
+ unshiftdata
+ Author
+ Mallick
+
+
+
diff --git a/help/en_US/slewrate.xml b/help/en_US/slewrate.xml
new file mode 100644
index 0000000..97afbfb
--- /dev/null
+++ b/help/en_US/slewrate.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ slewrate
+
+
+
+
diff --git a/help/en_US/sos2cell.xml b/help/en_US/sos2cell.xml
new file mode 100644
index 0000000..6e5da15
--- /dev/null
+++ b/help/en_US/sos2cell.xml
@@ -0,0 +1,87 @@
+
+
+
+
+
+
+
+ sos2cell
+ Converts a second order section matrix to a cell array
+
+
+
+
+ Parameters
+
+
+
+
+
+ Description
+
+c=sos2cell(s) converts an L-by-6 second-order-section matrix s given by:
+s = [B1 A1
+B2 A2
+...
+BL AL]
+to a cell array c = { {B1},{A1}, {B2},{A2}, ... {BL},{AL}} where each
+numerator vector Bi and denominator vector Ai contains the coefficients of a
+linear or quadratic polynomial. If the polynomial is linear, the coefficients
+zero-padded on the right
+c=sos2cell(s,g) adds a leading gain term to the start of the cell array as:
+c={ {[g,1]},{B1},{A1}, {B2},{A2}, ... {BL},{AL}}
+Example
+s=rand(2,6)
+s =
+
+
+
+
+column 1 to 5
+
+
+0.0437334 0.2639556 0.2806498 0.7783129 0.1121355
+0.4818509 0.4148104 0.1280058 0.2119030 0.6856896
+
+
+column 6
+
+
+0.1531217
+0.6970851
+
+
+sos2cell(s,2)
+ans =
+
+
+
+
+
+
+column 1 to 3
+
+
+![2,1] [0.0437334,0.2639556,0.2806498] [0.7783129,0.1121355,0.1531217] !
+
+
+column 4 to 5
+
+
+![0.4818509,0.4148104,0.1280058] [0.2119030,0.6856896,0.6970851] !
+Author
+Ankur Mallick
+
+
+
diff --git a/help/en_US/sos2ss.xml b/help/en_US/sos2ss.xml
new file mode 100644
index 0000000..99a5812
--- /dev/null
+++ b/help/en_US/sos2ss.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ sos2ss
+ [nargout,nargin]=argn();
+
+
+
diff --git a/help/en_US/sos2tf.xml b/help/en_US/sos2tf.xml
new file mode 100644
index 0000000..4d4e48e
--- /dev/null
+++ b/help/en_US/sos2tf.xml
@@ -0,0 +1,63 @@
+
+
+
+
+
+
+
+ sos2tf
+ This function converts series second-order sections to direct H(z) = B(z)/A(z) form.
+
+
+
+
+ Calling Sequence
+
+ [B] = sos2tf(sos)
+ [B] = sos2tf(sos, g)
+ [B,A] = sos2tf(...)
+
+
+
+
+ Parameters
+
+ sos:
+ matrix of real or complex numbers
+ g:
+ real or complex value, default value is 1
+
+
+
+
+ Description
+
+This is an Octave function.
+This function converts series second-order sections to direct H(z) = B(z)/A(z) form.
+The input is the sos matrix and the second parameter is the overall gain, default value of which is 1.
+The output is a vector.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/sos2zp.xml b/help/en_US/sos2zp.xml
new file mode 100644
index 0000000..efabf07
--- /dev/null
+++ b/help/en_US/sos2zp.xml
@@ -0,0 +1,71 @@
+
+
+
+
+
+
+
+ sos2zp
+ This function converts series second-order sections to zeros, poles, and gains (pole residues).
+
+
+
+
+ Calling Sequence
+
+ z = sos2zp (sos)
+ z = sos2zp (sos, g)
+ [z, p] = sos2zp (...)
+ [z, p, k] = sos2zp (...)
+
+
+
+
+ Parameters
+
+ sos:
+ matrix of real or complex numbers
+ g:
+ real or complex value, default value is 1
+ z:
+ column vector
+ p:
+ column vector
+
+
+
+
+ Description
+
+This is an Octave function.
+This function converts series second-order sections to zeros, poles, and gains (pole residues).
+The input is the sos matrix and the second parameter is the overall gain, default value of which is 1.
+The outputs are z, p, k. z and p are column vectors containing zeros and poles respectively, and k is the overall gain.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/sosbreak.xml b/help/en_US/sosbreak.xml
new file mode 100644
index 0000000..3957d94
--- /dev/null
+++ b/help/en_US/sosbreak.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ sosbreak
+ function for breaking a polynomial in second order polynomials (and an extra linear)
+
+
+
diff --git a/help/en_US/sosfilt.xml b/help/en_US/sosfilt.xml
new file mode 100644
index 0000000..25ce8aa
--- /dev/null
+++ b/help/en_US/sosfilt.xml
@@ -0,0 +1,58 @@
+
+
+
+
+
+
+
+ sosfilt
+
+
+
+
+
+ Calling Sequence
+
+ [y]=sosfilt(sos,x)
+
+
+
+
+ Parameters
+
+ sos:
+ Real or complex valued Lx6 vector or matrix
+ x:
+ Real or complex valued vector or matrix
+
+
+
+
+ Description
+
+This is an Octave function
+Second order section digital filter sos is applied to the input vector and the output vector obtained is of the same length.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/specgram.xml b/help/en_US/specgram.xml
new file mode 100644
index 0000000..a0215bb
--- /dev/null
+++ b/help/en_US/specgram.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ specgram
+
+
+
+
diff --git a/help/en_US/ss2sos.xml b/help/en_US/ss2sos.xml
new file mode 100644
index 0000000..b0cda76
--- /dev/null
+++ b/help/en_US/ss2sos.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ ss2sos
+ not taking if, order and scale as input since they do not seem useful
+
+
+
diff --git a/help/en_US/statelevels.xml b/help/en_US/statelevels.xml
new file mode 100644
index 0000000..70d7cfb
--- /dev/null
+++ b/help/en_US/statelevels.xml
@@ -0,0 +1,79 @@
+
+
+
+
+
+
+
+ statelevels
+
+
+
+
+
+ Calling Sequence
+
+ levels=statelevels(x, nbins, method, bounds)
+ [levels histogram]=statelevels(x, nbins, method, bounds)
+ [levels histogram bins]=statelevels(x, nbins, method, bounds)
+ [levels histogram bins]=statelevels(x, nbins, method, bounds, 'fig', On or Off)
+
+
+
+
+ Parameters
+
+ x:
+ real vector
+ nbins:
+ number of histogram bins to use in the histogram as a positive scalar, where the default value is 100
+ method:
+ method to estimate the statelevels using specified METHOD as one of 'mean' or 'mode', where the default value is 'mode'
+ bounds:
+ specify the lower and upper bound for the histogram as a two-element row vector
+ fig:
+ specify the logical input value to display figure as one of 'on' or 'off', where the default input in 'off'.
+ levels:
+ return lower and upper level values
+ histogram:
+ return histogram values
+ bins:
+ return binlevels values
+
+
+
+
+ Examples
+
+
+
+
+ See also
+
+
+
+
+
+ Authors
+
+ Jitendra Singh
+
+
+
diff --git a/help/en_US/stmcb.xml b/help/en_US/stmcb.xml
new file mode 100644
index 0000000..2490a4b
--- /dev/null
+++ b/help/en_US/stmcb.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ stmcb
+ function [b,a] = stmcb( x, u_in, q, p, niter, a_in )
+
+
+
diff --git a/help/en_US/strips.xml b/help/en_US/strips.xml
new file mode 100644
index 0000000..f31e366
--- /dev/null
+++ b/help/en_US/strips.xml
@@ -0,0 +1,59 @@
+
+
+
+
+
+
+
+ strips
+ Plots vector or matrix in strips
+
+
+
+
+ Calling Sequence
+
+ strips(x);
+ strips(x,sd);
+ strips(x,sd,fs);
+ strips(x,sd,fs,scale);
+
+
+
+
+ Parameters
+
+
+
+
+
+ Description
+
+strips(x)
+Plots a vector x in horizontal strips of length 250
+If x is a matrix, it plots each column of x on a separate strip with the leftmost
+column as the topmost strip
+strips(x,sd)
+Plots x in strips of length sd samples each
+strips(x,sd,fs)
+Plots x in strips of duration sd seconds with sampling frequency fs (in Hz)
+strips(x,sd,fs,scale)
+Plots x in strips as above, and scales the vertical axis by scale
+If x is a matrix, strips uses a column vector of all the elements of x for the strip plot
+If x has complex entries, only the real part of those entries are considered
+Author
+Ankur Mallick
+
+
+
diff --git a/help/en_US/subspaceMethodsInputParser.xml b/help/en_US/subspaceMethodsInputParser.xml
new file mode 100644
index 0000000..908ead3
--- /dev/null
+++ b/help/en_US/subspaceMethodsInputParser.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ subspaceMethodsInputParser
+ Input parser to be used by pmusic and peig
+
+
+
diff --git a/help/en_US/tf2sos.xml b/help/en_US/tf2sos.xml
new file mode 100644
index 0000000..0c63687
--- /dev/null
+++ b/help/en_US/tf2sos.xml
@@ -0,0 +1,63 @@
+
+
+
+
+
+
+
+ tf2sos
+ This function converts direct-form filter coefficients to series second-order sections.
+
+
+
+
+ Calling Sequence
+
+ [sos] = tf2sos (b, a)
+ [sos, g] = tf2sos (b, a)
+
+
+
+
+ Parameters
+
+ b:
+ matrix of real numbers
+ a:
+ matrix of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function converts direct-form filter coefficients to series second-order sections.
+The input parameters b and a are vectors specifying the digital filter H(z) = B(z)/A(z).
+The output is the sos matrix and the overall gain.
+If there is only one output argument, the overall filter gain is applied to the first second-order section in the sos matrix.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/tf2zp.xml b/help/en_US/tf2zp.xml
new file mode 100644
index 0000000..176e203
--- /dev/null
+++ b/help/en_US/tf2zp.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ tf2zp
+ [z,p,k]= tf2zp(b,a);
+
+
+
diff --git a/help/en_US/tf2zpk.xml b/help/en_US/tf2zpk.xml
new file mode 100644
index 0000000..3abb26c
--- /dev/null
+++ b/help/en_US/tf2zpk.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ tf2zpk
+ form
+
+
+
diff --git a/help/en_US/transpose.xml b/help/en_US/transpose.xml
new file mode 100644
index 0000000..a8d7108
--- /dev/null
+++ b/help/en_US/transpose.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ transpose
+ funcprot(0);
+
+
+
diff --git a/help/en_US/trial_iirlp2mb.xml b/help/en_US/trial_iirlp2mb.xml
new file mode 100644
index 0000000..b7364ca
--- /dev/null
+++ b/help/en_US/trial_iirlp2mb.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ trial_iirlp2mb
+ B = varargin(1)
+
+
+
diff --git a/help/en_US/triang.xml b/help/en_US/triang.xml
new file mode 100644
index 0000000..f351107
--- /dev/null
+++ b/help/en_US/triang.xml
@@ -0,0 +1,61 @@
+
+
+
+
+
+
+
+ triang
+ This function returns the filter coefficients of a triangular window.
+
+
+
+
+ Calling Sequence
+
+ y = triang (m)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ y:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a triangular window of length m supplied as input, to the output vector y.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/tripuls.xml b/help/en_US/tripuls.xml
new file mode 100644
index 0000000..99282a0
--- /dev/null
+++ b/help/en_US/tripuls.xml
@@ -0,0 +1,63 @@
+
+
+
+
+
+
+
+ tripuls
+
+
+
+
+
+ Calling Sequence
+
+ [y] = tripuls(t)
+ [y] = tripuls(t,w)
+ [y] = tripuls(t,w,skew)
+
+
+
+
+ Parameters
+
+ t:
+ vector of real or complex numbers
+ w:
+ real or complex number
+ skew:
+ real number, -1 <= s <= 1
+
+
+
+
+ Description
+
+This function generates a triangular pulse which is sampled at times t over the interval [-w/2,w/2]. The value of skew lies between -1
+and 1.
+The value of skew represents the relative placement of the peak in the given width.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/truth.xml b/help/en_US/truth.xml
new file mode 100644
index 0000000..d6ede61
--- /dev/null
+++ b/help/en_US/truth.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ truth
+ y = %t
+
+
+
diff --git a/help/en_US/tukeywin.xml b/help/en_US/tukeywin.xml
new file mode 100644
index 0000000..c4983d3
--- /dev/null
+++ b/help/en_US/tukeywin.xml
@@ -0,0 +1,63 @@
+
+
+
+
+
+
+
+ tukeywin
+ This function returns the filter coefficients of a Tukey window.
+
+
+
+
+ Calling Sequence
+
+ w = tukeywin (m)
+ w = tukeywin (m, r)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer
+ r:
+ positive real number, between 0 and 1
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a Tukey window of length m supplied as input, to the output vector w.
+The second parameter r defines the ratio between the constant and cosine section and its value has to be between 0 and 1, with default value 0.5.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/udecode.xml b/help/en_US/udecode.xml
new file mode 100644
index 0000000..2a0e567
--- /dev/null
+++ b/help/en_US/udecode.xml
@@ -0,0 +1,66 @@
+
+
+
+
+
+
+
+ udecode
+ Decodes the input uniformly quantized values
+
+
+
+
+ Calling Sequence
+
+ y=uencode(u,n,v,'saturatemode')
+
+
+
+
+ Parameters
+
+
+
+
+
+ Description
+
+Uniformly decodes the input vector or n-dimensional array of integers u with peak values +/- v
+If u has only positive values, the range of integers is assumed to be [0,2^n-1]
+If u has positive and negative values the range of integers is assumed to be [-2^(n-1),2^(n-1)-1]
+If v is not specified, its default value is 1
+If saturatemode='wrap' the output is wrapped using modulo arithmetic if overflow occurs
+If saturatemode='saturate' the output is saturated if overflow accors
+Example
+u = int8([-1 1 2 -5]);
+ysat = udecode(u,3)
+ysat =
+
+
+- 0.25 0.25 0.5 - 1.
+Author
+Ankur Mallick
+[1] International Telecommunication Union. General Aspects of Digital Transmission Systems: Vocabulary of Digital Transmission and Multiplexing, and Pulse Code Modulation (PCM) Terms. ITU-T Recommendation G.701. March, 1993.
+
+
+
+
+ See also
+
+ uencode
+ floor
+
+
+
diff --git a/help/en_US/uencode.xml b/help/en_US/uencode.xml
new file mode 100644
index 0000000..b5cede1
--- /dev/null
+++ b/help/en_US/uencode.xml
@@ -0,0 +1,67 @@
+
+
+
+
+
+
+
+ uencode
+ Performs uniform quantization of the input into 2^n levels
+
+
+
+
+ Calling Sequence
+
+ y=uencode(u,n,v,'signflag')
+
+
+
+
+ Parameters
+
+
+
+
+
+ Description
+
+Uniformly quantizes the input vector or n-dimensional array u into 2^n levels in the interval [-v,v]
+If v is not specified, its default value is 1
+'signflag' is a string that determines the nature of the quantization
+If signflag='unsigned' then y contains unsigned integers in the range [0,2^n-1] corresponding to the 2^n levels
+If signflag='unsigned' then y contains signed integers in the range [-2^(n-1),2^(n-1)-1] corresponding to the 2^n levels
+The size of the integers in y in bits(8,16, or 32) depends on the value of n
+If the input lies beyond +/- v it is saturated
+Example
+y=uencode(-1:0.5:1,3)
+y =
+
+
+0 2 4 6 7
+Author
+Ankur Mallick
+References
+[1] International Telecommunication Union. General Aspects of Digital Transmission Systems: Vocabulary of Digital Transmission and Multiplexing, and Pulse Code Modulation (PCM) Terms. ITU-T Recommendation G.701. March, 1993.
+
+
+
+
+ See also
+
+ udecode
+ floor
+
+
+
diff --git a/help/en_US/ultrwin.xml b/help/en_US/ultrwin.xml
new file mode 100644
index 0000000..16c6400
--- /dev/null
+++ b/help/en_US/ultrwin.xml
@@ -0,0 +1,71 @@
+
+
+
+
+
+
+
+ ultrwin
+ This function returns the coefficients of an Ultraspherical window.
+
+
+
+
+ Calling Sequence
+
+ w = ultrwin (m, mu, par)
+ w = ultrwin (m, mu, par, key)
+ w = ultrwin (m, mu, par, key, norm)
+ [w, xmu] = ultrwin (...)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ mu:
+
+ par:
+
+ key:
+
+ norm:
+
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the coefficients of an Ultraspherical window of length m supplied as input, to the output vector w.
+The second parameter controls the ratio between side lobe to side lobe of the window's Fourier transform.
+The third parameter controls the ratio between main lobe width to side lobe. The default value is beta.
+The value of xmu is also returned for given beta, att or latt.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/unshiftdata.xml b/help/en_US/unshiftdata.xml
new file mode 100644
index 0000000..57e0408
--- /dev/null
+++ b/help/en_US/unshiftdata.xml
@@ -0,0 +1,102 @@
+
+
+
+
+
+
+
+ unshiftdata
+ Inverts the effect of shiftdata
+
+
+
+
+ Calling Sequence
+
+
+
+
+
+ Parameters
+
+
+
+
+
+ Description
+
+y=unshiftdata(x,perm,nshifts)
+Applies the permutation perm or number of shifts nshifts on x to invert shiftdata
+
+
+
+
+ Examples
+
+
+
+
+ See also
+
+
+
+
diff --git a/help/en_US/upfirdn.xml b/help/en_US/upfirdn.xml
new file mode 100644
index 0000000..6688531
--- /dev/null
+++ b/help/en_US/upfirdn.xml
@@ -0,0 +1,62 @@
+
+
+
+
+
+
+
+ upfirdn
+ This function upsamples the input data, applies the FIR filter and then downsamples it.
+
+
+
+
+ Calling Sequence
+
+ y = upfirdn (x, h, p, q)
+
+
+
+
+ Parameters
+
+ x:
+
+ h:
+
+ p:
+
+ q:
+
+
+
+
+
+ Description
+
+This is an Octave function.
+This function upsamples the input data in the matrix by a factor of n. Then the upsampled data is FIR filtered. After this, the resultant is downsampled.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/upsample.xml b/help/en_US/upsample.xml
new file mode 100644
index 0000000..3d73473
--- /dev/null
+++ b/help/en_US/upsample.xml
@@ -0,0 +1,61 @@
+
+
+
+
+
+
+
+ upsample
+ This function upsamples the signal, inserting n-1 zeros between every element.
+
+
+
+
+ Calling Sequence
+
+ y = upsample (x, n)
+ y = upsample (x, n, phase)
+
+
+
+
+ Parameters
+
+ x:
+ scalar, vector or matrix of real or complex numbers
+ n:
+ real number or vector
+ phase:
+ integer value, 0 <= phase <= (n - 1 ), default value 0, or logical
+
+
+
+
+ Description
+
+This is an Octave function.
+This function upsamples the signal, inserting n-1 zeros between every element. If x is a matrix, every column is upsampled.
+The phase determines the position of the inserted sample in the block of zeros. The default value is 0.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/upsamplefill.xml b/help/en_US/upsamplefill.xml
new file mode 100644
index 0000000..8a6e352
--- /dev/null
+++ b/help/en_US/upsamplefill.xml
@@ -0,0 +1,62 @@
+
+
+
+
+
+
+
+ upsamplefill
+ This function upsamples a vector interleaving given values or copies of the vector elements.
+
+
+
+
+ Calling Sequence
+
+ y = upsamplefill (x, w)
+ y = upsamplefill (x, w, cpy)
+
+
+
+
+ Parameters
+
+ x:
+ scalar, vector or matrix of real or complex numbers
+ w:
+ scalar or vector of real or complex values
+ cpy:
+ can take in "true" or "false", default is false
+
+
+
+
+ Description
+
+This is an Octave function.
+This function upsamples a vector interleaving given values or copies of the vector elements.
+The second argument has the values in the vector w that are placed in between the elements of x.
+The third argument, if true, means that w should be scalar and that each value in x repeated w times.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/var.xml b/help/en_US/var.xml
new file mode 100644
index 0000000..53bc9bc
--- /dev/null
+++ b/help/en_US/var.xml
@@ -0,0 +1,46 @@
+
+
+
+
+
+
+
+ var
+
+
+
+
+
+ Calling Sequence
+
+ y=var(x)
+ y=var(x,w)
+ y=var(x,w,dim)
+
+
+
+
+ Parameters
+
+ x:
+ a vector or matrix.
+ w:
+ weight vector W of length X, or may take the value of 0 and 1. The default value is 0. Consider only non-negative values.
+ dim:
+ consider the variance along the dimension of X. 1 for clumun wise variamce and 2 for row wise variance.
+ y:
+ returns the variance of the values in X.
+
+
+
diff --git a/help/en_US/vco.xml b/help/en_US/vco.xml
new file mode 100644
index 0000000..44310c1
--- /dev/null
+++ b/help/en_US/vco.xml
@@ -0,0 +1,65 @@
+
+
+
+
+
+
+
+ vco
+ Voltage Controlled Oscillator
+
+
+
+
+ Calling Sequence
+
+ y=vco(x,fc,fs)
+ y=vco(x,[fmin fmax],fs)
+
+
+
+
+ Parameters
+
+
+
+
+
+ Description
+
+y=vco(x,fc,fs)
+Creates a frequency modulated cosine wave y whose frequency varies as the magnitude of x
+x lies in [-1,1]. x=-1 corresponds to a frequency of 0, x=0 corresponds to a frequency of fc
+and x=1 corresponds to a frequency of 2fc.
+y=vco(x,[fmin fmax],fs)
+Scales the frequency range so that x=-1 corresponds to a frequency of fmin and
+x=1 corresponds to a frequency of fmax
+If x is a matrix the same operation is performed on the columns on x
+Size of y is the same as the size of x
+Example
+x=rand()
+x =
+
+
+0.2113249
+y=vco(x,2000,8000)
+y =
+
+
+0.9454092
+Author
+Ankur Mallick
+
+
+
diff --git a/help/en_US/welchwin.xml b/help/en_US/welchwin.xml
new file mode 100644
index 0000000..aad08db
--- /dev/null
+++ b/help/en_US/welchwin.xml
@@ -0,0 +1,65 @@
+
+
+
+
+
+
+
+ welchwin
+ This function returns the filter coefficients of a Welch window.
+
+
+
+
+ Calling Sequence
+
+ w = welchwin (m)
+ w = welchwin (m, opt)
+
+
+
+
+ Parameters
+
+ m:
+ positive integer value
+ opt:
+ string value, takes "periodic" or "symmetric"
+ w:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function returns the filter coefficients of a Welch window of length m supplied as input, to the output vector w.
+The second parameter can take the values "periodic" or "symmetric", depending on which the corresponding form of window is returned. The default is symmetric.
+For symmetric, the length should be an integer>2. For periodic, the length should be an integer>1.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/window.xml b/help/en_US/window.xml
new file mode 100644
index 0000000..19d4312
--- /dev/null
+++ b/help/en_US/window.xml
@@ -0,0 +1,67 @@
+
+
+
+
+
+
+
+ window
+ This function creates an m-point window from the function f given as input.
+
+
+
+
+ Calling Sequence
+
+ w = window(f, m)
+ w = window(f, m, opts)
+
+
+
+
+ Parameters
+
+ f:
+ string value
+ m:
+ positive integer value
+ opts:
+ string value, takes in "periodic" or "symmetric"
+ w:
+ output variable, vector of real numbers
+
+
+
+
+ Description
+
+This is an Octave function.
+This function creates an m-point window from the function f given as input, in the output vector w.
+f can take any valid function as a string, for example "blackmanharris".
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/wkeep.xml b/help/en_US/wkeep.xml
new file mode 100644
index 0000000..3bac5e2
--- /dev/null
+++ b/help/en_US/wkeep.xml
@@ -0,0 +1,62 @@
+
+
+
+
+
+
+
+ wkeep
+
+
+
+
+
+ Calling Sequence
+
+ [y]=wkeep(x,l)
+ [y]=wkeep(x,l,opt)
+
+
+
+
+ Parameters
+
+ x:
+ Real, complex or string type input vector or matrix
+ l:
+ Length of matrix required
+ opt:
+ Character input to determine which side to extract from
+
+
+
+
+ Description
+
+This is an Octave function
+[y]=wkeep(x,l) extracts a vector of length l from the centre of input vector x.
+[y]=wkeep(x,l,opt) extracts vector based on opt which could be 'l','r' or 'c' (left, right or central).
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/wrev.xml b/help/en_US/wrev.xml
new file mode 100644
index 0000000..2068027
--- /dev/null
+++ b/help/en_US/wrev.xml
@@ -0,0 +1,56 @@
+
+
+
+
+
+
+
+ wrev
+
+
+
+
+
+ Calling Sequence
+
+ [y]=wrev(x)
+
+
+
+
+ Parameters
+
+ x:
+ Input vector of string, real or complex values
+
+
+
+
+ Description
+
+This is an Octave function.
+This function reverses the order of elements of the input vector x.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/xcorr2.xml b/help/en_US/xcorr2.xml
new file mode 100644
index 0000000..3768635
--- /dev/null
+++ b/help/en_US/xcorr2.xml
@@ -0,0 +1,32 @@
+
+
+
+
+
+
+
+ xcorr2
+
+
+
+
+
+ Calling Sequence
+
+ c = xcorr2 (a)
+ c = xcorr2 (a, b)
+ c = xcorr2 (a, b, biasflag)
+
+
+
diff --git a/help/en_US/zerocrossing.xml b/help/en_US/zerocrossing.xml
new file mode 100644
index 0000000..5109f99
--- /dev/null
+++ b/help/en_US/zerocrossing.xml
@@ -0,0 +1,56 @@
+
+
+
+
+
+
+
+ zerocrossing
+ This function estimates the points at which a given waveform crosses the x-axis.
+
+
+
+
+ Calling Sequence
+
+ x = zerocrossing (w, y)
+
+
+
+
+ Parameters
+
+ w:
+
+ y:
+
+ x:
+
+
+
+
+
+ Description
+
+This is an Octave function.
+This function estimates the points at which a given waveform y = y(w) crosses the x-axis. It uses linear interpolation.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/zp2sos.xml b/help/en_US/zp2sos.xml
new file mode 100644
index 0000000..1f57c0b
--- /dev/null
+++ b/help/en_US/zp2sos.xml
@@ -0,0 +1,66 @@
+
+
+
+
+
+
+
+ zp2sos
+ This function converts filter poles and zeros to second-order sections.
+
+
+
+
+ Calling Sequence
+
+ [sos] = zp2sos(z)
+ [sos] = zp2sos(z, p)
+ [sos] = zp2sos(z, p, k)
+ [sos, g] = zp2sos(...)
+
+
+
+
+ Parameters
+
+ z:
+ column vector
+ p:
+ column vector
+ k:
+ real or complex value, default value is 1
+
+
+
+
+ Description
+
+This is an Octave function.
+This function converts filter poles and zeros to second-order sections.
+The first and second parameters are column vectors containing zeros and poles. The third parameter is the overall filter gain, the default value of which is 1.
+The output is the sos matrix and the overall gain.
+If there is only one output argument, the overall filter gain is applied to the first second-order section in the sos matrix.
+
+
+
+
+ Examples
+
+
+
diff --git a/help/en_US/zplane.xml b/help/en_US/zplane.xml
new file mode 100644
index 0000000..64a7172
--- /dev/null
+++ b/help/en_US/zplane.xml
@@ -0,0 +1,23 @@
+
+
+
+
+
+
+
+ zplane
+ funcprot(0);
+
+
+
--
cgit