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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

PLD, SPLD, GAL, CPLD, FPGA Design :: 02-10-2017 13:56 :: FvM :: Replies: **12** :: Views: **1292**

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

Analog Circuit Design :: 12-07-2016 12:07 :: deba_fire :: Replies: **9** :: Views: **1991**

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

Analog Circuit Design :: 09-17-2016 14:34 :: FvM :: Replies: **11** :: Views: **1794**

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

Microcontrollers :: 02-05-2016 09:38 :: FvM :: Replies: **18** :: Views: **1152**

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

Digital Signal Processing :: 12-19-2015 16:14 :: SunnySkyguy :: Replies: **4** :: Views: **1060**

Electromagnetic Design and Simulation :: 07-04-2015 02:48 :: SunnySkyguy :: Replies: **6** :: Views: **2415**

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 (...)

ASIC Design Methodologies and Tools (Digital) :: 02-05-2015 15:47 :: SunnySkyguy :: Replies: **15** :: Views: **1396**

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.

Hobby Circuits and Small Projects Problems :: 12-17-2014 19:19 :: SunnySkyguy :: Replies: **1** :: Views: **706**

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 **filter**s more effective for my range of applications.

PLD, SPLD, GAL, CPLD, FPGA Design :: 11-24-2014 22:54 :: FvM :: Replies: **1** :: Views: **1100**

The types of **filter**s used to prevent aliasing in ADC's in theory are called **nyquist** **filter**s, 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 (...)

Analog Circuit Design :: 06-16-2014 12:24 :: SunnySkyguy :: Replies: **12** :: Views: **2968**

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

Microcontrollers :: 04-11-2014 10:41 :: SunnySkyguy :: Replies: **7** :: Views: **1047**

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

Digital Signal Processing :: 03-18-2014 08:45 :: deboleena18 :: Replies: **3** :: Views: **963**

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 (...)

PLD, SPLD, GAL, CPLD, FPGA Design :: 01-21-2014 07:33 :: FvM :: Replies: **5** :: Views: **2885**

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

Digital Signal Processing :: 04-15-2013 04:39 :: W_Heisenberg :: Replies: **1** :: Views: **567**

Condition 1+2 contradict the **nyquist** criterion.

Digital Signal Processing :: 01-25-2013 08:18 :: FvM :: Replies: **3** :: Views: **608**

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

Professional Hardware and Electronics Design :: 12-31-2012 13:04 :: pasaajan :: Replies: **3** :: Views: **1689**

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

Digital Signal Processing :: 05-09-2012 16:43 :: pstuckey :: Replies: **2** :: Views: **1046**

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.

RF, Microwave, Antennas and Optics :: 01-25-2012 01:21 :: biff44 :: Replies: **3** :: Views: **1652**

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

Digital Signal Processing :: 01-22-2012 19:55 :: cbbuntz :: Replies: **0** :: Views: **747**

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

Analog Circuit Design :: 12-19-2011 20:42 :: hfss_newbie :: Replies: **0** :: Views: **786**

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