similar to: OSC code

Dear collegues, I´ ve worked with near infrared (NIR) spectroscopy to assess chemical, physical, mechanical and anatomical properties of wood. I use "The Unscrambler" software to correlate the matrix of dependent variables (Y) with the matrix of spectral data (X) and I would like to migrate to R. The matrix of spectral variables is very large (2345 columns and n lines, where n =

>> There is also a IEEE paper, Adaptive Sampling Rate Correction for >> Acoustic Echo Control in Voice-Over-IP, which introduced a complex >> method to estimate the frequency offset and resynchronize the signals >> using arbitrary sampling rate conversion. I wonder if it can provide >> enough performance. Because I have also designed a sampling rate >>

On 01/20/2011 04:26 AM, Steve Underwood wrote: > On 01/19/2011 06:44 PM, LiMaoquan2000 wrote: >> Hi all, >> >> We have discussed so many about sampling rate asynchronous (or offset) >> between rendering (D/A converter) and capturing (A/D converter) of >> most PC soundcards. It seems all acoustic echo cancellers, include AEC >> in speex, can not deal with this

Dear Signal Processing Expert, I would like to generate a random stationary signal of gaussian probability density function to simulate narrow band noise at the output of an IF amplifier. I know the receiver's system temperature (Ts) and IF bandwidth (B) therefore I assume that my narrow band noise mean power equals KTsB watts and therefore the power spectral density No=KTs per Hz. Do you

Hi, I'd like to contribute to Vorbis and I think this may be of some interest for low bitrate coding. I have been experimenting with low bit-rate coding for the high-band (11 kHz to 22 kHz) and, though I haven't yet started quantizing my coefficients (a gain and an LPC filter), I expect to be able to approximate the whole 11-22 kHz band with around 1000 bits/s per channel (maybe even 500

Does anyone have any experience getting LIRC 0.9.0 up and running on CentOS 6.6? I have the packages from EPEL installed: lirc.x86_64 0.9.0-8.el6 @epel lirc-doc.x86_64 0.9.0-8.el6 @epel lirc-libs.x86_64 0.9.0-8.el6 @epel lirc-remotes.x86_64 0.9.0-8.el6 @epel And my IR

On 10-06-04 08:31 AM, Henry Gomersall wrote: > I hope people don't resent me making a point on this, as I don't > actually know how the noise suppression in speex works, but I have some > knowledge in the area of noise suppression. > > In general, if you have information about the nature of the noise you > are trying to suppress, you can do much better than just assuming

Thread-split from the vorbis-mailing list ("Vorbis determined to be as good as MPC at 128 kbps!") <p>On Sun, 30 May 2004, Segher Boessenkool wrote: [Steven So] SS>> If iTunes AAC can encode castanets with much less pre-echo at SS>> ABR 128 kbps, then hopefully there will be an imaginative SS>> (and non-patented) way of doing this in Vorbis without the SS>>

Dear list, i would like to build an own package using: R CMD INSTALL --html --build --auto-zip -l folder spectral.methods it seems to succeed as I get the following messages: * installing *source* package 'spectral.methods' ... ** R ** data Warning: empty 'data' directory ** preparing package for lazy loading ** help *** installing help indices converting help for package

I can only evaluate this with my subjective point of view. I had a special test scenario doing chat with cheap webcam microphones and loudspeakers. Fraunhofers solution was the only one that could eliminate the echo. In double talk the quality gets lower but is still very good. You might want to ask Fraunhofer for a demo version to test for yourself. I have no details on the algorithms being

Thank you, Andreas Engel. I downloaded the white paper of the Fraunhofer Acoustic Echo Control. http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Acoustic_Echo_Control-wp.pdf It said > "In the Fraunhofer Acoustic Echo Control, the frequency spectrum of the microphone signal is > modified so that the undesired echo components are removed from the signal transmitted to > the

(This is almost out of topic but anyway...) It is surprising how little research effort have been put into psy-acou models for CELP. The basic problem lies in that it is not easy to alter the LP model without distroying the minimum-phase property (ie. the stability of the predictor). That leaves us with psy-acou modelling of the noise-part only. However, my own research is in constrained

I've made block diagram of the encoder because I tried to find out, how it works http://stoffke.freeshell.65535.net/ogg/block.html Although there are specifiation docs, that give very detailed information about single aspects of the encoding (or decoding) , I'm missing documenations that give a more general overview, about how the encoder works. (Vorbis Illuminated seems a bit outdated,

Dear R Community, I am currently student at the Vienna University of Technology writing my Diploma thesis on causality in time series and doing some analyses of time series in R. I have the following questions: (1) Is there a function in R to estimate the PARTIAL spectral coherence of a multivariate time series? If yes, how does this work? Is there an test in R if the partial spectral

I have done a fair bit of spectral analysis, and hadn't finished collecting my thoughts for a reply, so hadn't replied yet. What exactly do you mean by normalize? I have not used the functons periodogram or spectrum, however from the description for periodogram it appears that it returns the spectral density, which is already normalized by frequency, so you don't have to worry about

Dear Tom, Short answer, if your using spec.pgram(), use the smoothing kernel to get a better estimate at the frequency centered in the bandwidth. If your frequency bin of interest is wider than the bandwidth of the kernel, average across frequencies (I think). The estimate appears to be normalized already. If you are calculating your PSD independently, then oversample (e.g. 2, perhaps 4 or more

I found this sentence in the Ogg format specs: http://www.xiph.org/ogg/vorbis/doc/vorbis-spec-res.html "A residue vector may represent spectral lines, spectral magnitude, spectral phase or hybrids as mixed by channel coupling." But where does the spectral phase information come from ? AFAIK MDCT doesn't provide any phase information. And in OGG-encoding, MDCT is taking place a few

Hi, I'm Pina and I'm a student in geology. I'm working with spectral profile of sand and I have to find the similarity between one spectral profile selected by hyperspectral image anche one that I created to mix different percentage of 4 mineral component. I have to find the best mix of percentage of this 4 mineral in order to have the best likeness with the spectral profile chose by

I've made a block diagram of the encoder because I tried to find out, how it works http://stoffke.freeshell.65535.net/ogg/block.html Although there are specifiation docs, that give very detailed information about single aspects of the encoding (or decoding) , I'm missing documenations that give a more general overview, about how the encoder works. (Vorbis Illuminated seems a bit

Hi, I've been using Speex in my voice-over-IP program on Win32, in wideband (16kHz) mode. I just starting using VBR recently and have run into something that might be a problem within Speex: If someone hasn't spoken for a little while, and the bitrate drops to very low, sometimes the high half of the spectrum becomes frozen with a looping sound. The bottom half of the spectrum is