Why if fsampling is integer multiple of fin data won't represent the signal? Where can I read something about this? As far as I know from Nyquist-Shannon sampling theorem it is not excluded that sampling frequency could be an integer multiple of the signal.2. from similar to previous reason it can happen you sample the same value points of a signal, and data won't represent the real signal when the fsampling is integer multiple of fin.
I am simulating transient noise analysis with Noise fmax=5*amplifier bandwidth4. just a guess, but are you simulating transient or PSS analysis? PSS is used when the fsampling is integer multiple of fin.
This load capacitance is just the capacitance seen at the output of the amplifier during the amplification phase, see figure below (only difference is mine is a fully differential implementation):1. it is a bit simplified equation for the ... output noise maybe. It is true for an RC... but flicker or thermal noise is the problem? Where is this load capacitor exactly?
The amplifier is a 2 stage folded cascode in figure below (260 MHz bandwidth, DC gain 74 dB, Vdd=800 mV)2. pass. I should see your whole circuit, I don't know.
Use correct terminology.For the simulation I am using spectrum in Cadence virtuoso,
It is very natural, since there are only two points during one period.1. first not only the ENOB I get is lower than 9 but it also depends on the initial phase of my input sinusoid,
so it is higher if the sampling occurs at the peaks of the waveform, why is that?
(1) fin=50MHz, fs=100MHz2. It also depends on the frequency of input signal (I do not understand why since for each input signal anyway has a frequency equal or less than half of the sampling frequency so the DFT should "see" the correct signal regardless of its frequency).
I can not understand what you want to mean at all.3. I know that one should choose a number of samples N=2^n (a power of 2) so the algorithm is faster,
but it is still correct even though I do not choose a power of two, isn't it?
It should just be slower?
Simply, your simulation can not reach to steady state.4. I get the following warning: "The function values at from and to are not equal." but since the waveform is noisy I suppose it is not possible that first and last values are equal, is there a way to fix this warning?
Can you understand your EDA Tool Play correctly ?Also as far as I know, the only ways now to improve ENOB would be to reduce noise or distortion:
Correct.1. To reduce noise which has root mean square vn, in the feedback amplfier I have vn^2=γkT/C (where C=load capacitance and γ a constant parameter) so I cannot do much if I cannot increase C, is it correct?
Apply large supply voltage.2. To reduce distortion I should improve linearity so I could inject higher signals and thus get better SINAD, but how can this be done?
With spectrum I was not meaning the simulator (which is instead spectre) but the tool used to get DFT in ADEL (measurements menu> spectrum)Use correct terminology. Such simulator never exist.
(1) also or not? If not what is the differnce between (1) and (2)?(1) fin=50MHz, fs=100MHz
(2) fin=100MHz, fs=200MHz
Generally, (2) results in bad ENOB.
When using a DFT one has to choose how many samples to use, and usually it is suggested to use an integer number N which is a power of 2, I was just asking if still the result is correct if N is not a power of 2.I can not understand what you want to mean at all.
Describe correctly.
Sorry I am quite new with using the simulator what do you mean by EDA tool play?Can you understand your EDA Tool Play correctly ?
Do you invoke Transient-Noise Analysis ?
If not, device noises are never reflected in your EDA Tool Play.
My supply voltage is 800 mV and cannot be increased.Apply large supply voltage.
Increase bias current of device to improve linearity.
If you don't care noise, use resitive feedback or degeneration.
I got you mean spectrum() function of Cadence Skill Language.With spectrum I was not meaning the simulator (which is instead spectre)
but the tool used to get DFT in ADEL (measurements menu> spectrum)
You can not understand sampling theorem and fourier series correctly.Why if fsampling is integer multiple of fin data won't represent the signal?
Where can I read something about this?
As far as I know from Nyquist-Shannon sampling theorem it is not excluded that sampling frequency could be an integer multiple of the signal.
No.Anyway I have tried simulating with fin=40 MHz so that fsampling=100 MHz is not its integer multiple
Ok.Anyway yes I have activated transient noise analysis with noise fmax=5*amplifier bandwidth
I can not judge.(I cannot take a number of samples which is a power of 2 in this case though since fsampling/fin is not a power of 2) and ENOB decreases
but now I get SINAD < SNR thus I guess the DFT is "seeing" the distortion,
so I guess distortion should be the main contribution to focus on to improve ENOB.
I can not understand what you want to mean at all.(1) also or not? If not what is the differnce between (1) and (2)?
Correct.I was just asking if still the result is correct if N is not a power of 2.
Sorry, I didn't think enough on it, actually it is not a problem if sampling frequency is integer number multiple of the input frequency, only the fsampling=2*finput is the problem. My mistake.Why if fsampling is integer multiple of fin data won't represent the signal? Where can I read something about this? As far as I know from Nyquist-Shannon sampling theorem it is not excluded that sampling frequency could be an integer multiple of the signal.
Reevaluate by setting followings.
I assume you use Cadence Spectre.
Transient Analysis Option:maxstep=1/(2*8*fin), strobeperiod=1/(8*fin), errpreset=conservative
Start=0.0 Stop=10/fin+8192/(8*fin)
Spectre Option:++aps
DFT Start=10/fin
DFT Stop=10/fin+8192/(8*fin)
DFT Sample Numbers=8192
(1) Transient Analysis without Noise
(2) Transient Analysis with Noise
Compare SINAD, SNR for both cases.
I am not sure whether flicker noise dominates, I am trying to google which kind of simulation I should do to figure it out, do you perhaps have any suggestion about it?To reduce flicker-noise you should try to increase diff-pair area, use PMOS differential pair instead of NMOS, you still didn't say flicker is the problem. For a switched circuit maybe not.
It looks a lot to read, I will start to study it and then will come back to say whether that gives some improvement to my simulation or not, thanks!If fs=8*fin use PSS(Periodic Steady State) analysis, then you can add the PNOISE easily. And PSS gives much more accurate results for the fourier transform, even it does the frequency domain analysis automatically.
Setting: https://www2.ece.ohio-state.edu/~bibyk/ece822/SpectreRF_0728.pdf
Hi,
did you use some analog antialias filter?
If not then please do a new measurement with filter.
Klaus
I don't think LPF at output is required, since your target SNR is 56dB at most.No I am not using such filter,
should I just put an ideal low pass filter at the output of my amplifier
However, if you would like to place LPF at output, use "analogLib/vcvs of gain=2", "analogLib/res as terminator", "rfLib/butterworth_lp of fc=fs/4" and "analogLib/res as load".or is there any more proper way to do this?
SINAD=51.7dB, SNR=156.6dBfirst noiseless trace
results that even without noise I have ENOB=8.3 while I would need 9
I think so.If so means that all that huge distortion is due to amplifier non linearities?
Correct.So what I wonder is,
since I am taking my samples only during end of phi1 clock phase,
I never sample when the circuit is on hold mode,
thus I shouldn't get distortion due to hold operation,
is this idea correct?
I don't think flicker noise is dominant.I am not sure whether flicker noise dominates,
I am trying to google which kind of simulation I should do to figure it out,
do you perhaps have any suggestion about it?
//+ noiseoff=[ R1p R1m R2p R2m R3p R3m R4p R4m ]
//+ noiseon=[ I0.I0.R0 I0.I0.R1 ]
//+ noiseon=[ I0.I0.R0 I0.I0.R1 I0.I0.IDAC1 I0.I0.I1 ]
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