Recovering data below a high pass cutoff frequency

Started by Marc in comp.dsp15 years ago 8 replies

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 ...


Nyquist rate for sampling complex-valued data?

Started by kiki in comp.dsp16 years ago 14 replies

I (vaguely) heard that sampling complex-valued data does not abide by the Nyquist rate criteria, i.e., the sampling rate fs can go lower than...

I (vaguely) heard that sampling complex-valued data does not abide by the Nyquist rate criteria, i.e., the sampling rate fs can go lower than Nyquist rate and it still can avoid aliasing and reconstruct perfectly... Is that true? Any theory behind it? Thanks a lot


Anti Aliasing a NI 6259 BNC Daq System

Started by fsou...@yahoo.co in comp.dsp9 years ago 6 replies

Hello, I am trying to record 8 channels of analog data using an NI 6259 BNC Data acquisition system. The Daq system has a built in lowpass anti...

Hello, I am trying to record 8 channels of analog data using an NI 6259 BNC Data acquisition system. The Daq system has a built in lowpass anti alias filter, but it has a much higher cut off frequency than what i need (i want a 500Hz cutoff). Does anyone know of a good 8 channel programmable/non programmable anti alias filter. I have also been suggested to use a software filter, but i though...


FM carrier offset and aliasing effects

Started by jacobfenton in comp.dsp10 years ago 5 replies

I am doing digital FM demodulation using the common ATAN2 and first difference method. When the carrier is offset from center, there is...

I am doing digital FM demodulation using the common ATAN2 and first difference method. When the carrier is offset from center, there is a non-linear region that appears and seems to be tied to the FM demodulation sample rate. Without be able to change the FM demodulation sample rate, are there any work arounds from this non-linear region? This non-linear region is just outside my bandwidth of ...


need FFT Overlap

Started by Prasad in comp.dsp14 years ago 1 reply

Hello All, After Block based FFT an overlap is done between blocks to remove block artifacts. But my query is if i perform FFT on 512...

Hello All, After Block based FFT an overlap is done between blocks to remove block artifacts. But my query is if i perform FFT on 512 block of data, i get 512 freq samples output. If i continue this for 10 blocks i will get 512 * 10 samples in freq samples then how can perform overlab of the block? if i do i will be loosing data also and aliasing will be thr. Can any one explain...


sample rate change of a narrow band signal

Started by fahim in comp.dsp14 years ago 2 replies

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 ...


Downshifted pitches and sample-rates to decrease bandwidth-usage of audio files.

Started by Green Xenon [Radium] in comp.dsp13 years ago 6 replies

Hi: Decreasing the pitch of the audio in an audio linear-PCM [wave] file means the sample-rate of the wave file can be decreased without...

Hi: Decreasing the pitch of the audio in an audio linear-PCM [wave] file means the sample-rate of the wave file can be decreased without causing aliasing. If the bit-resolution and # of channels [1 in mono, 2 in stereo] of the file are kept constant, then decreasing the sample-rate will decrease the file size. Adobe Audition allows the alteration of pitch without changing speed. ht...