similar to: project "Sphinx" kicked off

> I had the idea of implementing a lot of the operations in FFTs. ( for > example, it is possible to do auto-correlation and FIR filtering using > FFTs.) There are two advantages to this. > 1. It's almost always faster > 2. By swapping fft implementations, it could be easy to recompile for > fixed or floating point versions. No. FFT's require higher precision than

> I've just kicked off project "Sphinx". Which is supposed to > sound like "Speex" merged with "INT". ;) Meaning I am working > on an integer encoder and decoder. Good news! Although you might want to choose another name since "Sphinx" is already used by CMU's open-source speech recognition engine (http://fife.speech.cs.cmu.edu/sphinx/).

> > <with Prof. Farnsworth voice> "Good News, everyone". > > I've just kicked off project "Sphinx". Which is supposed to > sound like "Speex" merged with "INT". ;) Meaning I am working > on an integer encoder and decoder. > Great. I looked into converting speex to fixed point a while ago, but my job has gotten much busier

>> I had the idea of implementing a lot of the operations in FFTs. ( for >> example, it is possible to do auto-correlation and FIR filtering using >> FFTs.) There are two advantages to this. >> 1. It's almost always faster >> 2. By swapping fft implementations, it could be easy to recompile for >> fixed or floating point versions. > > No. FFT's

Hi, I'm currently working on a paper describing Ogg Vorbis. It is not finished yet. Mostly the decoder is being described. If you have any comments, please send them to me. See: http://www.turbocat.net/~hselasky/math/vorbis/files/ Then I have a question about the function "bark_noise_hybridmp()" which is used in the encoder. Can someone describe what the function does in

I have an irregular time series in a Zoo object, and I've been unable to find any way to do an FFT on it. More precisely, I'd like to do an NFFT (non-equispaced / non-uniform time FFT) on the data. The data is timestamped samples from a cheap self-logging accelerometer. The data is weakly regular, with the following characteristics: - short gaps every ~20ms - large gaps every ~200ms

This is not the same as the recent thread on a waterfall graph. I'm thinking about the rolling FFT display used in acoustics and other spectrum analysis tasks. Here's an example of a very fancy 3-D waterfall display: http://www.ultimaserial.com/UltimaWaterfall.html I was just wondering if there are any simple hidden-line tools in R that I could use to draw simple waterfall displays.

I'm trying to get these cleaned up and landed, but I'm running into some trouble with this patch. Using commit a08b29d88e3c (July 21) of Ne10, I'm seeing test failures for 60-point FFTs: nfft=60 inverse=0,snr = -3.312408 ** poor snr: -3.312408 ** nfft=60 inverse=1,snr = -16.079597 ** poor snr: -16.079597 ** All other sizes tested appear to work fine (84 to 140 dB of SNR). This

Analog Devices sponsored a fixed point version for AEC which was posted early this year, but it seems from the source code that the fft routines are still in floating point. Is the port still not complete or am I missing something? Has anyone out there ported speex AEC on RISC architecture? Please let me know. Thanks in advance, -Deepa -------------- next part -------------- An HTML

Thanks Jean, I missed that detail. I have a few more compile errors if you could help me with it => In function speex_echo_state_init in mdf.c I am getting error: "undefined reference to exp". I don't see a fixed point implementation of "exp". Can you point me to the correct files. I am using the following files from speex 1.2 beta1 => mdf.c,

And here's a patch for Intel Math Kernel Library. This allows commercial users of Speex to use a high-speed FFT library that isn't GPL'd. (You do need to pay for it though). This is 3 times faster than the default FFT in speex, and also faster than FFTW3 since MKL has native support for the complex packing Speex uses. Since Intel hasn't supplied any pkg-config files, and which

I write a work over Vorbis. ? ?Now I have a few questions to the function. ? ?For what is the MDCT used? ? ?For what is the FFT used? ? ?How is the expiration with the encodieren? ?In the reference to the MDCT,FFT and the hearing model . ? ?Is there a volume fragmentation in Subband as MP3? ? ?If it is possible, the answers in the detail ?Thanks ? -------------- next

Dear list, I apologies first for my English, hope you will understand well my question. I am working on 1/2 hour piezometric data, time unit is second. They present daily oscillation when using the spectrum() function. What I am really interested in, is to find the amplitude corresponding to this oscillation. I work with a college using Matlab, and although we apply the same methodology, our

Greetings, I have several sets of oscillation data and would like to estimate the parameters of a sine function to each set (and hopefully automate this). A colleague provided an excel sheet that uses solver to minimize the RSS after fitting the sine function to each data set, but this cumbersome and difficult to automate. Is there a method in R for fitting a given sine function to a

Alexander Chemeris wrote: > Hi, > > On 5/27/08, Thorvald Natvig <thorvald at natvig.com> wrote: > >> And here's a patch for Intel Math Kernel Library. This allows commercial >> users of Speex to use a high-speed FFT library that isn't GPL'd. (You do >> need to pay for it though). This is 3 times faster than the default FFT in >> speex, and

Hi, I made a small error in the impulse reponse function; when doing the inverse fft in fixed point, it will overflow unless it's properly scaled. This patch uses the same scaling as that used when updating the filters, and the outputs now have the same shape and the same scale in fixed and floating point. Best regards, Thorvald -------------- next part -------------- diff -ubBwr

I'm not an engineer so I hope I'm using the correct terminology here. I have a recorded waveform that I want to apply low and high pass filters too, are tehre already R functions existing to do this or am I going to have to program my own? thanks for any pointers tom

On 6/7/14, 1:55 AM, Jean-Marc Valin wrote: > Actually... no! 24-bit can indeed be useful as extra margin and Opus > can actually represent even more dynamic range than 24-bit PCM. That's > not the case for 192 kHz. There's no "margin" that 192 kHz buys you > over 48 kHz. You can do as much linear filtering as you like, the > stuff above 20 kHz isn't going to

We are having trouble with an installation that is getting a lot of echo on some calls. The installation is all SIP phones and they have a VoIP provider. When we call through the voip provider and into another of their customers (voip throughout) there is no echo problem. If we call in their landline, through the TDM400's FXO to one of the SIP phones, there is no echo problem. Sometimes

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