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...

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 anti- aliasing filter requirements and transfer function does the channel switching have ? It wo...

Dear All 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...

Dear All 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. Thanks: Hamid

How about using a Cauer with it's poor phase reponse,but the software compensating for it? Consider the following:- 1. An analogue...

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 to that above. Will there be a difference between the two responses? Suppose that for the abo...

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...

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 anti-aliasing filter before downsampling... By filtering, you lose information, right? It is har...

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...

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 either as we need analogue filters jus before sampling. Shytot

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...

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 Quantisation level is delta = dynamic range/2^16=20/65536=0.000305176 volts. (or 0.305mV) Now...

Is it possible to recover any data and/or information (e.g. trends) about a signal below the HP cutoff frequency of a system? System: Data is...

Is it possible to recover any data and/or information (e.g. trends) about a signal below the HP cutoff frequency of a system? System: Data is band limited by hardware, at the low end by an AC coupled amplifier (around 1 Hz) and at the high end by a hardware anti-aliasing filter (around 64 Hz). Is it possible in software to ascertain any information about the signal at frequencies below 1 ...

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...

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-aliasing low-pass and accomplish my goal? Thank you. Henry.

I have a large bunch of WAV files sampled at 44.1kHz and at 48kHz, which I would like to convert to 8kHz sampling rate, efficiently/quickly but...

I have a large bunch of WAV files sampled at 44.1kHz and at 48kHz, which I would like to convert to 8kHz sampling rate, efficiently/quickly but professionally & accurately (i.e. no aliasing etc.) Could anyone recommend me a tool for doing that (free software or Matlab)? (For example I know GoldWave is bad for this purpose since its anti-aliasing filter is unprofessional.)

Let's say I have a signal of

Let's say I have a signal of

Hello everyone, I want to design a real time DSP system.Band of interest extends from o to 4kHz.I will use 12 bit ADC and 3rd order butterworth...

Hello everyone, I want to design a real time DSP system.Band of interest extends from o to 4kHz.I will use 12 bit ADC and 3rd order butterworth lowpass filter. What I want to know is how I can estimate the minimum stopband attenuation for the anti-aliasing filter, minimum sampling frequency and The level of the aliasing error relative to signal level in the passband for the estimated Amin an...

Looks like Treo and maybe other Windows Mobile based devices have huge (~15% THD) wideband aliasing products when you play any non 44.1 file...

Looks like Treo and maybe other Windows Mobile based devices have huge (~15% THD) wideband aliasing products when you play any non 44.1 file ( MP3 or .Wav) back. Windows resamples all to 44.1 k and seems to have poor anti-aliasing filters. Whats up with that? My Laptop (XP) AC97 is OK on any sample rate. Kevin T

I want to change sampling rate of a narrow band signal by a fractional factor of 15/13. The original signal is -60dB or lower after f=0.08...

I want to change sampling rate of a narrow band signal by a fractional factor of 15/13. The original signal is -60dB or lower after f=0.08 (1 corresponds to half the sampling rate). I use matlab upfirdn function and design the anti-imaging/anti-aliasing filter using remez. The problem is that i get small droop in the passband no matter how sharp a filter i have got. I know this sounds a bit vague ...

hello, i'm trying to program a wavetable-based synth in c++. I have some waveforms and different envelopes for them and want to switch...

hello, i'm trying to program a wavetable-based synth in c++. I have some waveforms and different envelopes for them and want to switch between settings while playing. I use a 4096 samples buffer. My question is, how do i now implement anti-aliasing? I thought of multiplying envelope & waveform, and then lowpass and the resample with linear interpolation at a lower sample rate. Is it right, tha...

If you are estimating time-delay bewteen two microphones would it be better to use a Bessel filter for anti-aliasing rather than a Butterworth...

If you are estimating time-delay bewteen two microphones would it be better to use a Bessel filter for anti-aliasing rather than a Butterworth due to the approximate linear phase. Wang King

Dear all, Thanks for all your replies but I am still a bit confused. Let me try rephrasing my question so that everyone gets a better picture...

Dear all, Thanks for all your replies but I am still a bit confused. Let me try rephrasing my question so that everyone gets a better picture of what I am trying to do. I have a large number (order of 10^13) of discretized data samples. I have to obtain the frequency spectrum of this data so that I can predict minima in the discretized data. I am using the FFTW C library to obtain the Frequency...

Hi all, I'm looking for a good textbook that serves as an introduction to anti- aliasing filters, Nyquist frequency, sampling rates, SNR, low...

Hi all, I'm looking for a good textbook that serves as an introduction to anti- aliasing filters, Nyquist frequency, sampling rates, SNR, low pass filter design, etc. Can someone please suggest a reference? Thanks in advance, -weg

Hello guys and girls, my first posting here. On some other, hardware design-related forum there was a question raised by someone whose goal is to...

Hello guys and girls, my first posting here. On some other, hardware design-related forum there was a question raised by someone whose goal is to built a sampling system to achieve 500 MHz sampling rate by means of two 250 Msps ADCs while driving them with samlping closk featured by 180 deg. phase shift. So that in time domain would be sampling by two channels in interleaved manner. Then the enti...

On Mar 31, 12:36 am, dbd wrote: > > When the sinc function is used as a prototype filter in windowed > filter design, the width of the...

On Mar 31, 12:36 am, dbd wrote: > > When the sinc function is used as a prototype filter in windowed > filter design, the width of the sinc is scaled wider in time for > lowpass responses narrower than the Nyquist band. that's a choice that someone might make. > This would be done > for any lowpass anti-aliasing filter designed for sample rate > reduction with this me

Suppose I take samples at once per minute for a data logger. Do I still need an anti-aliasing filter? The sampling freq would be 1/T where T=60...

Suppose I take samples at once per minute for a data logger. Do I still need an anti-aliasing filter? The sampling freq would be 1/T where T=60 secs...so how is this possible? The values of the capacitors etc would be hugh. Of course I could over-sample1000s of times...is this the norm? K.