search for: nyquist

...of about 2 months right in the middle of it. I want to do the acf, pacf, ccf, and a cross-spectral analysis of the two series. I realise that I could just break each series into two segments and cross-correlate with the shorter series, but I'd rather deal with the whole series to increase the nyquist frequency. I think the its function in the irregular time series package will create a class its object with the right time stamps, but can this then be used in the same was as a class ts object for the correlation and spectral anayses? Sam ---- Sam McClatchie, Sub-program leader, Pelagic Fisher...

Hi List, I have a time series of 122 values, actualy it is a time series of daily indian monsoon rainfall. now i want to filter this time series for a particular oscilation say 10 to 20days oscilation. i want to find out what amount of variance is explained by this mode. Which package is available in R for this purpose. and how to calculate nquest frequancy of this series. any help is much

.... ;-) Silvia. On Tue, Mar 19, 2013 at 3:22 PM, Benjamin Schwartz <ben at bemasc.net> wrote: > Presuming that you are asking regarding the Ogg Vorbis audio format, the > correct answer is: there is no minimum or maximum cutoff frequency. Vorbis > can code all frequencies from DC to Nyquist. What Vorbis will actually do > is extremely complex, extremely nonlinear, and highly dependent on > bitrate. If you are in the mentality of linear time-invariant filters, you > will never be able to understand it. > > > On Mon, Mar 18, 2013 at 11:08 AM, Fernando Alberto Marengo...

...n't there yet. That's OK. My primary reason for writing, is a really mindblowing lecture I attended a few weeks ago, given by Professor Jaan Pelt of Tartu Observatory in Estonia. Professor Pelt argued "don't sample regularily!" If you sample regularily, you are limited by the Nyquist theorem, but you're not if you don't, so given a finite number of sampling points, you should sample here and there instead of regularily. Research has also been done to figure out an optimal sampling strategy. I have very little training in signal processing, but it occured to me that this...

I've looked around and not found details on the expected frequency range the Speex codec can be expected to carry. Is there any documentation available or a table of some sort that has been compiled which would give an indication of the frequency range based on the various compression options in speex? Best regards, Baldvin Hansson Reykjavik, Iceland baldvin@baldvin.com -------------- next

Small update. This makes --without-gpl-fftw3 do the expected, and also allow --with-gpl-fftw3=/path/to/fftw3 for alternate installs. It also zeroes the imaginary of the nyquist element; it works without it, but it's recommended to do. -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: speex-fftw3-fix.diff Url: http://lists.xiph.org/pipermail/speex-dev/attachments/20080527/f46859ed/attachment.txt

...simsalabim package recommens that L=N/2 I ran the command getSignal and felt uncomfortable at picking values for parameters C0 and r0. I would appreciate some suggestion/guidelines because these values seem to influence the trend extraction. I chose C0=0.0005, r0=0.0005 which are both below the Nyquist frequency 1/Amps.dc$L = 0.0007223346 As a consequence, getSignal finds two Eigenvectors capturing the trend. >From my experience with toolkit SSA/MTM I know this output can be misleading. I am trying to figure out whether trend has really been found by double-checking the print-out data: >...

Had a question about Wine ignoring the X11 SendEvent method, specifically for something using "xdotool" as follows: "xdotool type --window $wineWindow test" Basically, I would like to know if it is possible to configure or recompile wine to accept and act upon events received in the above manner; and if it is possible, how can I do it? Currently, any input in such a manner

...o plot( Sxy $ freq, Sxy $ phase[ ,1], type = 'l' ) # Make y a quadrature component of x y = ts( cos( w * t ), deltat = deltaT ) Sxy = spectrum( cbind( x, y ), plot = F ) # The phase should be either +pi or -pi # since exp(i pi) = exp(-i pi) = -1 + 0i # Why isn't it? Sampling issues? Nyquist has been satisfied. plot( Sxy $ freq, Sxy $ phase[ ,1], type = 'l' ) # The real question (limited to a 2-D input): # How to combine the amplitude/phase into a single # complex valued cross spectrum? mySxy = complex( real = Sxy $ spec[,1] * Sxy$ spec[,2], imaginary = Sxy...

> Presuming that you are asking regarding the Ogg Vorbis audio format, the > correct answer is: there is no minimum or maximum cutoff frequency. Vorbis > can code all frequencies from DC to Nyquist. What Vorbis will actually do > is extremely complex, extremely nonlinear, and highly dependent on bitrate. > If you are in the mentality of linear time-invariant filters, you will never > be able to understand it. Well, it's not quite that complex, but it is adaptive. There are no...

Dear list, Could you please tell me the values of the minimum and maximum cutoff frequencies for each coding version of the 44.1 kHz sampled data? For instance, are the values fmin=100 Hz and fmax=12 kHz valid? Thank you very much in advance. Kind regards, ? Fernando A. Marengo Rodriguez, PhD Post-doctoral fellow on Acoustics and Beamforming -- Laboratory of Noise and Vibration (LVA) Federal

...but some books are not careful about stating whether one- or two-sided is to be assumed" (2002, p. 504). As a result, I infer that spec.pgram() is returning what Press, et al. would call a two-sided PSD. Thus, the power spectrum returned by spec.pgram() should be multiplied by 2 between (DC, nyquist) non-inclusive, and the mean can be resolved separately as the DC component is not returned by spec.pgram(), as: ## N/2 used here because the length of PSD$spec is N/2 rather than N/2 + 1 # as it does not return a DC value. >mean(a) + PSD$spec[floor(N/2)] + 2 * sum(PSD$spec[2:floor(N/2)-1])...

...the 0->4khz range (which for 48khz output means the lower 1/6th of the power spectrum). The second is the 0->8khz range (full original signal), and the last is the full range. The reason I split it is that the filter-based resampler has cutoff filter, so it zeroes out frequencies near the nyquist. So the SNR is unfair for the 0->8 range. Anyway, on to the results. First, a 16=>16 resampling. Filt Q10: Diff 0.883327, SNR 3.12531e-07 / 0.472589 FFT 320: Diff 0.00292969, SNR 2.57974e-07 / 4.77473e-05 Both resamplers will recreate the original samples. The filter based does limit the...

Hi I used R for the first time yesterday. I wanted to plot the aliasing effect of sampling a 5.5KHz sinusoid at only 8KHz (below the Nyquist limit). So I wrote a small R script that a) plots 1msec worth of a 5.5KHz sin wave b) plots 1msec of the resulting 2.5KHz alias and c) plots the 8 sampling points on the 5.5KHz source wave. I think I have found a bug. The script is as follows: #truesamplingfreq <- 1000*5.5 freqin1msec = 5.5 #al...

...rning: non-expert opinion). In the definition I learned, a periodogram is considered "normalized" in the sense that the sum across the frequency range has something to do with some quantity (Sum Squared, Mean Squared, etc.) of a signal in the time domain. Let N = Number of samples Fc = Nyquist critical frequency C = Normalization parameter P2(f) = C * Mod(fft(signal))^2 from -fc to fc: Press et al call a "two-sided" estimate, even if just the power at positive frequencies is returned. P1(f) = C * Mod(fft(signal))^2 from DC to fc, but all frequencies excluding DC and Nyquist a...

...onents are those, >> > in a sweep from 0 to 20kHz? >> >> The frequency component at any time should be more or less exactly one >> (THE frequency). > > Exactly. So what frequencies above 48kHz are there > to be sampled at 96kHz? None. Do we want to discuss the Nyquist theorem? (AFAIR it's for pure sinus waves and is says "at least"). > >> >> With higher sampling frequencies in the original, >> >> I was expecting to reduce the aliasing effects for higher frequencies. >> > >> > What higher frequencies?...

>> Yes, I plan to use it in a VoIP environment if I can get latency reduced to >> an acceptable level :) >> The latency depends directly on the overlap parameter, which also controls >> the quality. Higher quality => higher latency. You could set the overlap to >> 0, but that would give you some nasty artifacts. >> You can also resample with smaller block

John, Thanks for the reply! You mentioned output sample rates should be 44100 or 48000, should I worry about input (Mic) Sample rates as well? (Currently I was requesting the sample rate on both ends to be 16000 samplesPerSecond, for ease of passing into the codec) Also, do you recommend any particular resampler that I should use, or are any of the ones out there probably okay, or should

Hi! Excuse the delay, but I had to deal with a corrupted NTFS file system that ate many important files on an USB stick... The FLAC version of the original is almost 6MB and it can be downloaded slowly from this time-limited link: https://sbr5vjid0jgmce4q.myfritz.net:40262/nas/filelink.lua?id=0ba5a10529a6fe7b On the meaning of a logarithmic sweep: If you use foobar2000 and the

...he current input frame. > > Careful when looking at W because of how the real and imaginary parts > are packed in the array. Err. Ok, as I got it, 'bin 0' has it's amplitude in W[0], bin 1 to N-1 has it's real part in W[i*2-1] and it's imag in W[i*2], and finally the nyquist amplitude is in W[N-1] I took this from how power_spectrum() computes, so I might be off :) >> Anyway, I did some proper testing. I took my headset, bent the microphone >> arm so it's resting inside the .. uh.. whatever you call that large >> muffler thing that goes around y...