Recovering data below a high pass cutoff frequency

Started by Marc in comp.dsp14 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 ...


using IIR or FIR to implement the lowpass filter for downsampling?

Started by Nasser M. Abbasi in comp.dsp7 years ago 12 replies

For downsampling (decimation) one normally uses low pass filter before downsampling (anti-aliasing filter). I have thought that FIR is the...

For downsampling (decimation) one normally uses low pass filter before downsampling (anti-aliasing filter). I have thought that FIR is the best choice here for the low pass filter implementation for this case. But I was reading this page: http://www.dspguru.com/dsp/faqs/multirate/decimation and it said in 2.3.1 under "How do I implement decimation?" "To implement the filtering part...


sample rate change of a narrow band signal

Started by fahim in comp.dsp13 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.dsp12 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...


Anti Aliasing a NI 6259 BNC Daq System

Started by fsou...@yahoo.co in comp.dsp8 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...


need FFT Overlap

Started by Prasad in comp.dsp13 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...