Single bit ADC question

Started by Tom in comp.dsp17 years ago 9 replies

Somebody asked me this question and I was unsure of the answer. It is well known that if you quadruple the sampling rate that you reduce the...

Somebody asked me this question and I was unsure of the answer. It is well known that if you quadruple the sampling rate that you reduce the quentisation noise by 6dB. Normally this means you can drop off one bit.So if you have 12 bits you can make do with 11 bits by sampling 4 times faster and maintain the same quantisation noise level. Taking this to its logical conclusion if we have 16 bi...


Processing Time in TI eZdsp F2812

Started by vickykulkarni in comp.dsp14 years ago 1 reply

Hi I have a question regarding the flow of my code. I am oversampling my input signal using the ADC at a rate 3.125 MHz. I have 2 inputs & am...

Hi I have a question regarding the flow of my code. I am oversampling my input signal using the ADC at a rate 3.125 MHz. I have 2 inputs & am using simultaneous sampling. I am using the adcseqovdtest code of TI as a framework. So I have 16 samples from the ADC Result registers. My question is with sampling rate of 3.125 and clock frequency of 150 MHz, I have (150/3.125) 48 instructions per samp...


Oversampling and quantization...

Started by kl31n in comp.dsp14 years ago 18 replies

Hopefully some of you can point me to the solution of this problem. If I oversample a signal, I quantize it and then I reduce the sampling...

Hopefully some of you can point me to the solution of this problem. If I oversample a signal, I quantize it and then I reduce the sampling rate with a decimator by a factor M, I can improve the QSNR by a factor of log2(M). Nice and easy. But what if I lower the sampling rate using a fitting technique? I mean if I know how the signal is done and I retrieve some parameters out of it using a f...


Want to sampel 4 channels at the same time

Started by Ben in comp.dsp14 years ago 5 replies

Hello I must sample 4 analog signals, and I must ideally sample them at the same time. And I'm going to down sample the data rate later....

Hello I must sample 4 analog signals, and I must ideally sample them at the same time. And I'm going to down sample the data rate later. The sampling frequency at the ADC is 10 kHz and afterwards the signal gets LP filtered to 50 Hz bandwidth before down sampling to 200 Hz rate. But what happens if I use a muxed ADC. The last channel is always sampled later than the first cha...


Sampling question

Started by Anonymous in comp.dsp12 years ago 11 replies

The sampling theorem tells us that we must sample at least twice the bandwidth of a signal. When the signal is centered around dc to some max...

The sampling theorem tells us that we must sample at least twice the bandwidth of a signal. When the signal is centered around dc to some max freq say Fmax than we can say Fs> =2Fmax. However, when the signal is centred around some other higher frequency (say Fc), how is bandwidth defined? is it from Fc (the center freq) to the max value or from Fc-minfreq to Fc + maxfreq ? If you apply te


[OT?] faulty irregularity of data sampling

Started by Richard Owlett in comp.dsp15 years ago 6 replies

I've sortof been following several threads concerned with irregular sampling of data and various noise effects. I think I might benefit from...

I've sortof been following several threads concerned with irregular sampling of data and various noise effects. I think I might benefit from a *QUALITATIVE* discussion of how to approach a problem which APPEARS to be outside realm of DSP. [The problem is "real" but ... ] QUESTION: What is fuel mileage of a class of vehicles. AVAILABLE DATA (and possible error sources) [source is cr...


effect of clock jitter on ADC

Started by praveen in comp.dsp17 years ago 2 replies

Hello, I am designed a DSP based which measures the rotation rate (sagnac phase). But the problem i am facing is that jitter in the clock...

Hello, I am designed a DSP based which measures the rotation rate (sagnac phase). But the problem i am facing is that jitter in the clock which is used as sampling input to the ADC will spoil by algorithm. The sampling point is very crucial. How can i solved this problem. suggest we a solution waiting for reply with regards praveen


DSD frequency response and sampling rate

Started by Carey Carlan in comp.dsp15 years ago 17 replies

Pardon this intrusion from rec.audio.pro, but some said I might find an answer here. Here's what I understand: Fact: DSD has a sampling...

Pardon this intrusion from rec.audio.pro, but some said I might find an answer here. Here's what I understand: Fact: DSD has a sampling rate of 1 bit * 64 * 44100 per second. Assumption: Each sample raises or lowers the volume one "bit", one unit. Conclusion: A 22.05 kHz tone gets 128 samples and has a peak to peak height of -32 to +31 to -32 again (6 bits) and a 44.1 kHz tone only -1...


Following peer review - the derivation of the representation of sampling by Diracian Delta functions

Started by gareth in comp.dsp5 years ago 3 replies

Revised after peer review, approached with a little less haste So... Sampling with a period of T is given by (after asciification) as...

Revised after peer review, approached with a little less haste So... Sampling with a period of T is given by (after asciification) as .. (1/T)sum (n : 0, inf)(d(t-nT) * f(nT) ) ... with * representing multiplication and not convolution as we are still in the time domain. However, (and this is where my protest came in having previously fully revised Fourier, Laplace, Butterworth...


multi sampling frequency filtering

Started by Fred Nach in comp.dsp15 years ago 1 reply

Hi all ! I'm trying to design a digital graphical equalizer (8 bands), which can deal with many samlping frequency... so it should be able to...

Hi all ! I'm trying to design a digital graphical equalizer (8 bands), which can deal with many samlping frequency... so it should be able to equalize let say either in 48k, 44.1k, 22k, 32k... in the same way. So my question is ... do I have to design an equalizer for each sampling frequency or is there a good (and memory coefficent table saving) trick ? I'm think of maybe designing t...


Software modulation timing issue

Started by kalden in comp.dsp16 years ago 1 reply

Hello, I am developing a software modulator and am currently working on the qpsk portion of it. This problem will apply to pretty much...

Hello, I am developing a software modulator and am currently working on the qpsk portion of it. This problem will apply to pretty much all modulation schemes however. Let me first explain my algorithm: This program needs to be very fast (capable 100 mhz sampling rates) so I am pregenerating all of my samples. The user inputs the center frequency and the sampling rate and I fill a seri...


random sampling theory

Started by amara vati in comp.dsp16 years ago 2 replies

hi, could anybody give me references on random sampling theory, i.e theory for signals which are sampled at random intervals of...

hi, could anybody give me references on random sampling theory, i.e theory for signals which are sampled at random intervals of time. regards amar


Singer Acceleration Model and Kalman

Started by nicosd in comp.dsp15 years ago 1 reply

Hi, I'm looking for someone familiar with the Singer acceleration model. Could such a person explain to me why Singer makes no assumption...

Hi, I'm looking for someone familiar with the Singer acceleration model. Could such a person explain to me why Singer makes no assumption on the sampling rate and the nyquist theorem? How can he say things such as the correlation coefficient goes to infinity, when it is actually bounded by 1/2T where T is the sampling period? This is driving me bananas!! thnaks This message was se...


Interpolation and decimation

Started by seb in comp.dsp17 years ago 44 replies

Hello, i am looking for decimation and interpolation technique in order to, given a sampling rate fs, obtain a new sampling rate like...

Hello, i am looking for decimation and interpolation technique in order to, given a sampling rate fs, obtain a new sampling rate like (a/b)*fs. A way to to do is to decimate and then use linear interpolation... Is there some other ways (documents) to do this ? If so, have you got some book or url ? Thanks


frequency detection Maximum sampling rate

Started by kazino in comp.dsp12 years ago 8 replies

Hello everyone, I am new to dsp and working on a project where I need to take 1000 8-bit samples per second. Each sample will be at 1 of 11...

Hello everyone, I am new to dsp and working on a project where I need to take 1000 8-bit samples per second. Each sample will be at 1 of 11 predefined frequencies. My question is, is it possible to detect frequencies at a sampling rate of 1 per millisecond. (1000 samples per second). I would like to use a computer to do this rather than any hardware. Any help would be greatly appreciate...


Where did Shannon introduce the Sampling Theorem?

Started by Funky in comp.dsp16 years ago 1 reply

Which publication did Shannon prove the Sampling Theorem in? Thanks

Which publication did Shannon prove the Sampling Theorem in? Thanks


Question about spectrum analysis and sampling theory

Started by Krellan in comp.dsp9 years ago 15 replies

This is probably a FAQ, but I looked for a while and couldn't find an answer. Apologies in advance. What is the "sweet spot" in a spectrum...

This is probably a FAQ, but I looked for a while and couldn't find an answer. Apologies in advance. What is the "sweet spot" in a spectrum analysis waterfall display, for generating or decoding a narrower signal that's embedded within it? Let's say I'm sampling at 48 kHz, and there's a signal in there that's 10 kHz wide. Where shall I place this signal? If it's too far left (lower fr...


Audio FFT problem - PLEASE HELP

Started by cppt...@yahoo.com in comp.dsp12 years ago 1 reply

Could some DSP guru please help me a bit ? I am creating a FFT based application that uses an audio input. The sampling details are...

Could some DSP guru please help me a bit ? I am creating a FFT based application that uses an audio input. The sampling details are as follows: encoding : PCM sampling frequency : 16000 Hz resolution : 16 bits channel : mono signed : true endianness : little (as this runs on an Intel processor) The raw bytes get collected in a byte array, and I take two bytes at a time (resolution is 1...


Audio FFT problem - PLEASE HELP

Started by cppt...@yahoo.com in comp.dsp12 years ago 5 replies

Could some DSP guru please help me a bit ? I am creating a FFT based application that uses an audio input. The sampling details are...

Could some DSP guru please help me a bit ? I am creating a FFT based application that uses an audio input. The sampling details are as follows: encoding : PCM sampling frequency : 16000 Hz resolution : 16 bits channel : mono signed : true endianness : little (as this runs on an Intel processor) The raw bytes get collected in a byte array, and I take two bytes at a time (resolution is 1...


60Hz Notch filter at 25kHz sampling rate?

Started by abradley1984 in comp.dsp12 years ago 22 replies

Hello, I have an ENG signal, sampled at 25kHz that has high mains interference that I want to remove. The problem is I need the surrounding...

Hello, I have an ENG signal, sampled at 25kHz that has high mains interference that I want to remove. The problem is I need the surrounding frequencies, maybe not 50-70, but I can't afford to sacrifice much more than this. The filter designs I've been looking at don't seem to be able to do this at such a high sampling rate. Any suggestions? Would downsampling help? Thank you in adva...