Arnfinn
  • Member for 5 years, 8 months
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  • Newcastle, New South Wales, Australia
Can you turn a square wave into a sine wave using a low pass filter?
6 votes

In principle you can, in practice you almost can. The square wave consits of sinusoids with frequencies that are at multiples of the fundamental one (the inverse of the length of one high and one low)....

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How many taps does an FIR filter need?
4 votes

I don't think there really is some definitive minimum number of taps. It it pretty common to generate FIR filters by convex optimization, e.g. $$ \text{minimize} \ \text{max}\left(\left| H(\omega) \...

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Do FIR filter have any application in control theory?
4 votes

FIR filters are fairly common in some areas of control theory. As they usually incur a lot of added phase/time-delay, they are not really usable in the feedback path of regular control systems, but ...

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Method of reconstructing a band-limited signal from discrete samples
3 votes

The general mathematical framework for interpolation is approximation theory. I guess the most important result is that for signals with bandwidth limitation, you can have perfect reconstruction via $...

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Modelling the amplitude and phase through an amplifier
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2 votes

I apologize, the following is a bit rough, but I've put together an example in MATLAB illustrating all the different elements I think you need. That is: a fist-order model for an amplifier (a common ...

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RMS averaging in spectrum analyzers
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2 votes

You can check out the manual for the HP3582A spectrum analyzer or the manual for the SR785 dynamic signal analyzer, which is basically a copy of the former instrument. My guess is that the definition ...

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Is there a way to derive an FIR filter using an IIR filter?
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2 votes

Approximating the frequency response of an IIR filter or physical process using an FIR filter is useful in learning control. It is quite common to do FIR filter design based on frequency response ...

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FIR analog low-pass filter?
2 votes

You can use a bucket-brigade device to introduce a delay, or a piezoelectric or other type of delay line.

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getting frequency response from input and output signal
2 votes

What you are looking for is probably an empirical transfer-function estimate (ETFE). Several methods can be used, but Welch's averaged periodogram [1] is arguably the most used [2]. The plant output ...

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How is a PCM byte converted to a voltage level?
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2 votes

There are quite a few different DAC topologies with different properties. The typical tradeoff is between resolution and speed and power usage -- higher speed means lower resolution, and higher ...

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One-bit depth audio
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2 votes

You could try PWM or Delta-Sigma modulation. The result will depend on the sample-rate of your music player, and the frequency content of the sound you are trying to play back. You can get good ...

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Difference between $\tt rand()$ and $\tt randn()$ in MATLAB
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2 votes

The reason for the difference is that rand() has an output range of $[0,1]$, and therefore you have a constant in both $a$ and $b$ in the first example. The other function randn() generates an output ...

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Measurement resolution for an accelerometer
1 votes

In measurement science there is less talk of resolution and more talk of accuracy and precision. Accuracy is how close to the true value your are; and precision is how much spread (variance) you ...

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Remove multiplied white noise from periodic signal
1 votes

I suppose you can use synchronous averaging in this case due to the signal being periodic and the nature of the noise (as long as it is sufficiently uncorrelated). See MATLAB example: ts = 1e-4; % ...

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Proper rounding when converting between digitized values and floating point
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1 votes

I think you don't have to worry too much about so-called 'transparency' (ability to represent accurately the integer value as a floating point value) unless you do normalization. See the libsndfile ...

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Is it normal that a DAC outputs spikes?
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1 votes

The TI DAC8532 uses a Kelvin resistor-string (string of resistors that divide the reference voltage, with individual switches from the nodes to the output), so it is guaranteed to be monotonic. What ...

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PWM samplerate to equal PCM accuracy
1 votes

Short answer: For a direct implementation of PWM in discrete-time, yes, for 48 kHz, 16-bit audio, a clock of 3.2 GHz is required. Direct Stream Digital (DSD) format uses 1-bit $\Delta$-$\Sigma$ ...

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Resampling an audio signal
1 votes

There are some excellent papers on this topic you might want to read. Sampling-50 years after Shannon, M Unser - Proceedings of the IEEE, 2000 Splines: A perfect fit for signal and image processing, ...

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Does filter length correspond to "spread" of frequency points?
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1 votes

I assume that by "spread" you mean "number of samples". The filter length, or number of taps, is not necessarily related to the number of frequency samples. If you generate the filter by frequency ...

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Kalman Filter with single input?
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1 votes

The Kalman filter is the least-squares optimal solution for the states of a set of linear dynamic equations under a Gaussian additive noise assumption. Hence, if you have an unforced linear first-...

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Is there a strategy for discrete control of a system with dynamics near sample rate?
1 votes

Depending on your plant you might be able to do something. If the plant is unstable, but dominantly linear, you should be able to get a very accurate model using exact discretization (zero-order hold)....

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What are the advantages of higher order Kalman Filters like EKF, UKF?
1 votes

They should improve the convergence properties of the state estimate, i.e. if you have an initial value far from the trajectory, they could improve the region of attraction. They should also reduce ...

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How to find PSD (Power spectral density) of spatial data
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1 votes

A spectrum estimation method that works well in many cases is Welch's method. The MATLAB implementation is pretty straight forward to use, even when not knowing about the math behind it. Following ...

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Basic tools for digital filter design?
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1 votes

There are lots and lots of software that can aid you in designing digital filters. MATLAB is probably the most used software, at least in the university sector. The DSP toolbox and the signal ...

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Measuring the open loop transfer function in closed loop: what's the better approach?
1 votes

You have two signals to use, the reference and the measurements. You generate the error from these two signals. There is no additional information in that signal. Closed-loop identification is more ...

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How to tune PID parameters to track a family of reference signals?
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1 votes

As already mentioned, a linear system that tracks a step reference, will track it for arbitrary values and time of application. Due to the internal model principle, in a feedback control system, if ...

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Continuous low-pass filter (LPF) on a periodic signal
1 votes

It looks like you want a filter that is linear time invariant. I don't think it is possible to meet all your criteria with an LTI filter, but I think it you can get close with the filter $$ H(s) = \...

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What criteria should be used for smoothing
1 votes

You want to look at the innovation sequence; the difference between the filtered value (the estimate or forecast) and the measured value. If you have a perfect filter that removes the noise, the ...

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Active noise cancellation using kalman filter
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1 votes

I guess you use the LMS and RLS filters because they are adaptive, and you can do the same thing with the Kalman filter, if you augment the state-vector with the parameters of the model. You then get ...

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How to determine brown noise variance from samples?
1 votes

If the white noise has a Gaussian probability distribution, then it should be possible to find the variance of both noise sources using a power spectral density estimate. $S_{1}(j\omega) = {\sigma_{1}...

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