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43 Threads found on edaboard.com: Nyquist Filter
Method 1 (simple undersampling) will mirror noise and unwanted signal above the nyquist frequency of 100 Hz into the base band. Unless the input signal is already band limited, you need to low-pass filter before down sampling. A 20 tap FIR can't achieve a cut-off frequency of 0.01 fs as required for effective implementation of method 2. In pr
Hi All, The nyquist criteria which is described in various undergraduate textbooks is for single input and single output. There exists a similar nyquist criteria for MIMO systems. Atleast a google search presents a lot of results. But I have difficulty in its application. There are opamp based circuits which have MIMO loops. Example: Two-tho
Effect of DAC resolution can be estimated by quantization noise calculation. Sampling rate will theoretically not affect THD as long it's above the nyquist rate and a sufficient reconstruction (anti-aliasing) filter is used. Existence of the filter is presu
You have to add ~8bit on top of high frequency range to avoid phase jitter, which brings 24 bits you have calculated to 32 bits. Sounds like a misunderstanding of DDS principle. You can get pretty low jitter after the DDS output filter with a carrier fulfilling the nyquist criterion, achieving low jitter is much more a problem of s
The bandwidth limitation on the output is less than Sample Rate/ n What are you expecting real-time? Bandwidth? nyquist filter BW? Input SNR Resolution required ? number of bits for lossless = ? read
nyquist theorem states 2 samples per Hz for a brick wall filter. Typically 2.5 samples per Hz for resolution bandwidth (RBW) is the minimum to prevent modulation aliasing. But often it is much higher sampling rate with enough samples to ensure peak envelope has settled usually >5xT cycles of where T=1/f_RBW Does it sweep or step the frequency
The question must be asked, what determines validity of data? If data is has redundancy and bandwidth limitation, then the sampling rate of both input and output must exceed twice the bandwidth of the signal. If there is an anti-aliasing filter or nyquist filter, the sampling rates do not need to be synchronous. There is no obvious need (...)
It's been a few decades since I worked with these types of A Law Codecs. TI user forum may be helpful too. Good performance with synchronous conversions and wide dynamic range high SNR in 8 bits with 4KHz BW and 8KHz sampling rate, which includes S&H and 5th Order nyquist filter and sin x/x correction with 5mW driver.
Sampling frequency should be as low as possible to allow handy filter order, but must at least fulfill the nyquist criterion for the input signal. You can play around with the matlab filter tool to get a feeling of meaningful filter parameters. I find IIR filters more effective for my range of applications.
The types of filters used to prevent aliasing in ADC's in theory are called nyquist filters, which define the stop bandwidth fSTOP must be <= 1/2 of the sampling rate. The implementation however is a tradeoff between distortion from noise above fSTOP , group delay distortion in the passband, amplitude ripple in the passband and degree of (...)
100 Hz pulse has spectrum that spans many harmonics, depending on accuracy desired an ADC must operate at minimum of 2x highest frequency and have an anti-aliasing nyquist filter. Consider >>10x rate or 1kHz if you want to reconstruct input signal from ADC data and find peak. Otherwise if that is all you need, just add a Cap and charge to peak val
Hi, I understand u have data with 100 samples per second ( assuming dt = 0.01s is the sampling interval) this will give your nyquist frequency as fs/2 = 50 Hz, this is important as the cut off frequency of your filter is normalized with respect to the nyquist frequency. fc is your cutoff frquency you need to ge
As mentioned, a simple downsampling mirrors spectral components above the nyquist frequency of 2.5 MHz into the base band. But the 40 MHz sampled data may already contain aliasing products if the ADC hasn't a suitable anti-alias filter in front of it. For a reasonable answer, you need to know the input signal spectrum and the applied (...)
Hi, all when doing the sampling for the audio, do we really care about the sampling rate? I mean as long as we satsify the nyquist conditoin, we will always get back the original audio? The process is done on a PC, I tired several times, it seems this is true. If so, does this mean the reconstruction filter I have build is really flexib
Condition 1+2 contradict the nyquist criterion.
You can do 1 experiment on your seat. Try to measure the harmonics frequency of RS485 isolated supply with & without comms. If you are using Tektronix (CRO), put in Math mode, in nyquist waveform can see interesting frequency related to CE test. 150kHz to 600kHz freq may be getting prominent with & without RS485 comms, this require little patienc
I think you probably need a passive LPF before your ADC. Using a digital filter (averaging or FIR) won't achieve the same function. The passive LPF is to prevent aliasing, any signal or noise will fold into the first nyquist zone (10 kHz) when the analog signal is sampled. For instance, the AD8616 claims BW> 20 MHz. So all the noise energy from
I am no DSP expert, but I think if you just run the adc at >20 MHz clock speed (2 x the nyquist rate) you can separate negative frequencies from positive frequencies. yes/no? The noise figure is not 3 dB higher, per se, than with a low IF frequency. It is just that you have both the negative and positive frequencies shown in 0 to 5 MHz baseband.
So say we have a 1st order lowpass filter at 44100khz using a pole zero method and I subtract the lowpass from the input to acheive the reciprocal highpass and want to blend it to correct for proper gain in the analog domain by using: (1/(1 + (22050/fc)^2))^(1/2) or 1/(1 + (22050/fc)^2) for power so then the highpass and lowpass will be in q
Hi folks, I have a problem in understanding nyquist theorem. When i sample a signal and receive the sampled signal in RX side,in order to reconstruct the aTX signal, should i use a practical sample and hold filter(which has a sinc shape in freq.domain to obtain a step shape signal) or use a filter that has a trapez