Adaptive Filter | ADPCM | ADSP | ADSP-2181 | Aliasing | AMR | Anti-Aliasing | ARMA | Autocorrelation | AutoCovariance | Beamforming | Bessel | Blackfin | Butterworth | C6713 | CCS | Chebyshev | CIC Filter | Circular Convolution | Code Composer Studio | Comb Filter | Compression | Convolution | Cross Correlation | DCT | Decimation | Deconvolution | Demodulation | DM642 | DSP Boards | DSP/BIOS | DTMF | Echo Cancellation | Equalization | Equalizer | ETSI | EZLITE (Ez-kit Lite) | FFT | FFTW | FIR Filter | Fixed Point | FSK | G.711 | G.723 | G.729 | Gaussian Noise | Goertzel | GPIO | Hilbert Transform | IFFT | IIR Filter | Interpolation | Invariance | JTAG | Kalman | Laplace Transform | Levinson | LPC | McBSP | MIPS | Modulation | MPEG | Multirate | Notch Filter | Nyquist | OFDM | Oversampling | Pink Noise | Pitch | PLL | Polyphase | QAM | QDMA | Quantization | Quantizer | Radar | Random Noise | Reed Solomon | Remez | Resampling | RTDX | Sampling | Sharc | TI C6711 | Undersampling | Viterbi | Wavelets | White Noise | Wiener Filter | Windowing | XDS510PP | Z Transform

Re: Does anyone have any reference to sampling distortion?

In comp.dsp, Steve wrote: > To all you DSP mavens out there. > > I am looking for some references to distortions introduced by sampling = > errors, > especially sampling close to the Nyquist criterion. In all the = > literature I > have seen, reconstruction of a waveform is guaranteed ...

Delta Sigma ADC + channel switching

Hi ! I have the following question: If I use an AD7738 (Analog Devices) or ADS1256 (TI) 24bit delta/sigma ADC with "fast channel switching" (I think it's some mechanism to flush the digital filter so that you don't have to throw away 6+ samples before you get a valid one), what effect on the a...

Re: PCM 3003

On 10 Nov 2003 07:53:55 -0800, Hamid wrote: > Anybody knows how to change the sampling rate of PCM3003(Daughter Card > for C6711 DSK card), it is 48Ksps and I need to work with 8ksps, > apparently it has internally anti aliasing and post processing filter > for 48Ksps. PCM3003 is all ha...

Real-time octave analysis

Hello ! I would like to perform ***real-time*** octave analysis with my C54x (100 Mips, audio signal @ 48kHz). Althought i already implement FFT, I will do it with IIR filtering. (I know the avantages of FIR decimator, but I dont think it will be usefull in this particular case). The "method"...

Re: New Digital Audio Standard?

Hi all . . . The point about filtering definately brings up some issues . . . While my thoughts aren't fully formed on the subject, I feel that it's likely that the perceived "improvements" in higher-bandwidth PCM formats are strictly a result of the higher rates applied to "common practice"...

Re: IIR design and stability

"Sam (rép. sans -no-sp-am)" wrote in message news:3ff754cd$0$726$5402220f@news.sunrise.ch... > Hi all ! > > Thanks for reading ! I would like to design a 1st order digital high pass > filter with a 50 us time constant. The analog transfer function is > Ha(s)=50E-6*s/(1+50E-6*s). > >...

Resampling Questions - Newbie requests your help

I've got a single channel of 16-bit/44.1kHz audio that I'd like to resample to 16-bit/88.1kHz. I'll start with this since I believe that it might be easier than dealing with uneven mutiples if I chose another rate (like 48kHz for example). Please let me know if my understanding is correct: ...

Re: equivalence between Z-transform and Laplace-Transform

AG wrote: > Hi Tim, -- snip -- > > > The term "damping ratio" is much more slippery when you're talking > > about discrete-time systems. Assuming that I'm not messing up the > > math, if you find the pole locations of your transfer function, z_0 = > > e^{jw + q) then the "damping...

Re: Pitch shifting question

H wrote: > Hi. > > I want to do some pitch shifting of some samples I have. I'm trying to > avoid the whole sample rate conversion problem. I can hack up the > hardware some. > > Q: Can I just alter the clock and rate that I send data to my DAC > *WITHOUT* changing the anti-a...

Creating brick-wall anti-aliasing filters?

How about using a Cauer with it's poor phase reponse,but the software compensating for it? Consider the following:- 1. An analogue filter is cascaded with an N-bit ADC feeding an N bit DAC. 2. An analogue filter is cascaded with an N-bit digital filter designed to create a response equal...

Re: Sound Card Question

"HelpmaBoab" wrote in message news:ph6Pf.4437$JZ1.96604@news.xtra.co.nz... > Are you saying that a SB has a fixed anti-aliasing filter at 20kHz even > you > you sample at say 22050Hz? Surely not. > I understood they used over-sampling and down-sampling. If I record from, say, an SB ...

Re: CAUTION! was "What is the advantage on high-sampling rate ?"

Jerry Avins wrote: > Tim Wescott wrote: > > ... > > > > > I am showing what happens when you you sample without antialiasing > > > > filter, and when you do so, you get high frequency images. I do so > > > > INTENTIONALY to show the need for anti aliasing filters ahead of the > ...

Re: inv filter deconvolution

Dan Shorb wrote: > > In the paper, he claims the inverse filter of the log-swept sine for > > deconvolving the speaker's output is simply the time-reversed input > > with an exponentially-decreasing envelope applied. > > The author also claims that envelope applied to the inverse filter h...

Re: Advice for non "DSP Guy"

Jack Klein wrote in message > My issue is this: I have an ECG that sends 256 samples per second > over a serial port. I have a medical device that needs to receive > exactly 100 samples per second over a serial port. The baud rates and > format of the data (integer in both formats) are ...

Re: Designing Filters with non-uniform sampling

"dnb" wrote in message news:fa0deac8.0407280822.5f8a3cf4@posting.google.com... > The data input is inherently non-uniformly sampled (I dont have access to the > source). I can interpolate if that is what you mean. Yes. > If so, can you > kindly mention type of interpolation you w...

Re: Interpolation

Eric Jacobsen wrote: > > On Tue, 01 Apr 2008 19:37:07 -0600, jim > wrote: > > > > > > > Eric Jacobsen wrote: > > > > > > If the process requires an AA filter to reduce bandwidth they > > > > "necessarily" will. > > > > > > Back to my previous example with a so...

Re: Problem in Sampling

Sai Kumar wrote: > Hi Tim, > Thank you, > But i am unlucky as u said. > > "Sai Kumar" wrote in message > news:cfheji$cv6$1@ns1.fe.internet.bosch.com... > > > Dear All, > > > > > > I have one problem in real time signal processing. My signal frequency is...

Re: Questions about optimizing Sensor outputs to dsp/microcontroller A/D inputs

Bernhard Holzmayer wrote in message news: ... > steve wrote: > > > Ok, when hooking up a sensor output to a/d input we want to > > minimize the ratio of RMS noise to the LSB value. > ... > > If I had to connect such a sensor to an ADC, the first step in my > design would cer...

Re: A Basic Question or two on CD Emphasis/Deemphasis

Randy Yates writes: > [...] > Allan Herriman writes: > > > I guess it seemed like a good idea at the time. (Remember that the > > red book CD format was thought up in the 1970s, with 1x oversampling > > and analog reconstruction filters in mind. > > > The extra lowpass...

Re: Real time FFT voice processing...why wont it work?

Jerry Avins wrote in message news: ... > Robert Lacoste wrote: > > > Of course this should work, as IFFT(FFT(x))=x ... > > That's true only if all the data are dealt with at once, and if they > represent one cycle of something that repeats. For non-repetitive > inputs, one mus...

Re: Decimation vs. Collapsing and their spectral effects

Sorry, if I distract a little from the OPs main issue, but it's very close to my problem, therefore I would like to challenge you a little more... Eric Jacobsen wrote: > This is a pretty easy case to analyze from a DSP perspective. > You're applying an averaging filter that is essenti...

Re: Difference between Digital and Discrete Signal

Jerry Avins wrote: > I imagine that you mean "you can generate signals that can not always be > a sampling (if they were analog)", but I can't imagine such a signal. > Please enlighten me. Well, in certain condition, for example a square wave. Assuming you have a "generic" sampling sys...

what are the appearance of aliasing in image and audio processing?

Hi all, I've heard a lot of aliasing. What do they look like in image and audio/speech/music with aliasing? I did not see/hear any real aliasing stuff, so the concept of aliasing looks abstract to me, although I know how it occurs in terms of mathematics... I am also wondering about the ...

Re: Building a (cheap) software radio

Jerry Avins wrote: > Michel Rouzic wrote: > > Bryan Hackney wrote: > > > > > No such thing exists as an off the shelf product. If it did, it would > > > probably sell, but that's what the USRP is all about - it's just too > > > expensive for your bugdet. > > > > > > PCs are not fast ...

Re: Filter design.

Jerry Avins wrote: > To sample at 8KHz, you need an analog filter that removes /everything/ > above 4KHz from the analog signal. It's easier to start rolling off > below that, so you will sensibly do all the filtering with an analog > filter (Switched capacitor chip?). Your other option i...

Re: Maximum useable oversampling rate10x analog signal bandwidth?

John E. Hadstate wrote: > "steve" wrote in message > news:1106449193.210985.208270@z14g2000cwz.googlegroups.com... > > Analog Devices comments on their web site that for > oversampling to work > > the sample rate cannot be over 10 times the analog signal > bandwidth. So > > thi...

Re: Please suggest a good filter

Ajay wrote: > Can you provide me more information about decimation? Decimation is the practice of reducing the effective sample rate. This is done quite simply by discarding samples. Say for example, you sample something at 100Hz and only save every 10th sample. The effective sample rat...

Re: Do Nyquist/filtering requirments hold for raster video digitizing?

glen herrmannsfeldt wrote: > > jim wrote: > > (snip) > > > Well, yes and no. If you are raster scanning then each scan line is > > effectively a digital sampling. If I'm understanding correctly, in the > > vertical direction the data you will be collecting is essentially ju...

Re: Delta-Sigma: bad choice for image digitizing? Horrible settlingtime! Not phase linear?

Tim Wescott wrote: > > > The converter will only work well if it has good phase linearity -- i.e. > the output should be a delayed version of the input and not be otherwise > smeared out. If your ADC fits this then the pure delay, as Jerry said, > doesn't matter much. > In th...

Re: who understands signal detection?

"glen herrmannsfeldt" wrote in message news:d0ni53$2o4$1@gnus01.u.washington.edu... > Jerry Avins wrote: > > > Fred Marshall wrote: > > > > if fs ==1Hz then Nyquist is < 0.5Hz ... > > > I think too much can be made of that truth. That condition assures that > > the original ...

Re: ringing: minimum vs linear phase

Eric, OK thanks. Now lets put this into an audio context. It would seem (and this is subjective now) that the ringing due to min phase filter would be less objectionable because it is all post edge and would sound similar to reverberation. The linear phase filter has pre ringing and I ...

Re: Interchanging FIR and Decimation operation

> That is correct, you can't. I would question whether a length = 3 > filter is going to do you much good for decimating data by a factor of 8 > -- you would at least need a length = 8 filter and probably more. I forgot to mention one more thing abt the filter. It is not normal anti-aliasi...

Re: IIR filter Implementation

Ronald H. Nicholson Jr. wrote: > In article , > Erik de Castro Lopo wrote: > > The resolution of the FFT is limited by the length of the data. Zero > > padding the data to extend its length does not provide more resolution. > Not true. If the noise level is low enough then the n...

Re: dynamic range per bit

"glen herrmannsfeldt" wrote in message news:V_idnbhVGae6OAPfRVn-2w@comcast.com... > Rimmer wrote: > (snip) > > > It comes from the expression for quantisation noise variance which is (I > > think) delta/sqrt(12) or some such. > > This assumes a Uniform PDF from delta/2 to -delta...

Re: PLESIOCHROUNOUS re-sampling of audio using D/A > A/D

Mark wrote: > I have a question about PLESIOCHROUNOUS re-sampling of audio. > > > It was mentioned in another thread that simply deleting or repeating a > sample now and then is a poor solution because it adds clicks. > Correct. > > It was mentioned that linear interpolation is ...

Re: Sampling Problems

Might want to discuss anti-aliasing filtering topics including, avoiding aliasing, phase effects, and possibility of compromises between filtering, signal bandwidth, and sample rate. Might want to look at/discuss sigma-delta/delta-sigma converters. Dirk Tim Wescott wrote: > I'm writin...

Re: causal system, ??? plz hlp

"Tim Wescott" wrote in message news:11g1o5rt9eisv77@corp.supernews.com... Forgetting that _real_ systems are _always_ non-linear is a > _bad thing_ because then you forget how to do real things with all your > knowledge. > I think thats a bit strong - there are many examples of syst...

Sound Card Question

How does a sound card set its anti-aliasing filters? After all, you can program a sound card to read at say 44.1kHz or 22,050Hz or half of that again so how do the ani-aliasing filters change? Switched cap filters are sampled filters so they would not be good and digital filters are no good eith...

Anti-Aliasing filter

Ok I am sampling with an ordinary A/D (not sigma-delta) at 11025kHz which mean I need any noise to be attenuated at half sampling which is at 5512.5Hz. I have read that the nosie level needs to be less that the R.M.S Quantisation Level of my A/D. Now I swing +or - 10 volts with 16 bits. So my Qu...

Re: IQ Demod related questions

On Thu, 25 Aug 2005 22:25:10 +1000, "Paul Solomon" wrote: > Hi, > > Sorry for all the questions all the time, however I am still working on this > FM demod design and My shiny new stratix II dev board arrived the other day > as well as a long needed copy of Rick Lyons Understanding DSP...