46

Baud rate is the rate of individual bit times or slots for symbols. Not all slots necessarily carry data bits, and in some protocols, a slot can carry multiple bits. Imagine, for example, four voltage levels used to indicate two bits at a time. Bit rate is the rate at which the actual data bits get transferred. This can be less than the baud rate because ...


45

Fundamentally, all circuits are analog. The problem with performing calculations with analog voltages or currents is a combination of noise and distortion. Analog circuits are subject to noise and it is very hard to make analog circuits linear over huge orders of magnitude. Each stage of an analog circuit will add noise and/or distortion to the signal. ...


30

This is a windowing artifact. The linked code pads out a 10,000 sample signal with zeroes so that the length is a power of two. %% Author :- Embedded Laboratory %%This Project shows how to apply FFT on a signal and its physical % significance. fSampling = 10000; %Sampling Frequency tSampling = 1/fSampling; %Sampling Time L = 10000; ...


30

The line bit rate is the number of bits per second being moved. The data bit rate is the number of information bits being moved per second. The baud rate is the number of symbols per second (Baud is named after Emile Baudot) The line rate and information rate can be different due to line coding An example of line coding is QAM; QAM64 encodes 6 bits per ...


25

Am I missing some important reason why you would do the signal conditioning in this order? Yes you are... The front-end differential amplifier will be chosen such that it has a common-mode rejection level of many tens of dB, quite possibly in the region of 80 dB. This diff amp converts a differential signal into a single-ended signal and any common-mode ...


23

The concept is good, but you will find it is not so simple in practice. Pitch is not simply the predominant tone, so there's problem number 1. The FFT frequency bins can't hit all (or even multiple) tones of the musical scale simultaneously. I would suggest playing with an audio program (for example, Audacity) that includes an FFT analyser and tone ...


23

If the compiler writers put some effort into optimizing it for that target, it will at least make some use of the special DSP instructions / architecture. But for ultimate performance it will never be as good as hand-tuned assembly. It might be plenty good enough, though - depends on your application. Other alternatives include: Write the majority of ...


23

in the sense that two telegraph signals (in other words, two electric currents) can travel in opposite directions on the same wire, at the same time, without interfering with each other This is the point. A telegraph wire is not just a wire that transports a direct current. It transports a signal, which in essence is a voltage or current wave. The current ...


23

Forget sampling rate for a few seconds... Think about sampling period for a second, which is the time interval between two consecutive samples. This time can be an integer or any real number (as long as it’s positive, of course). Sampling rate is simply the inverse of sampling period. Does it make more sense this way?


21

Premature optimization is the root of all evil. - Donald Knuth When you find that you don't get enough performance from your code, profile your program first, find the bottlenecks, analyze your performance requirements, and only then start doing optimizations. Writing assembly code is last resort. My question is if I just program in C, wouldn't the ...


20

I've attended an IEEE talk last month titled “Back to the Future: Analog Signal Processing”. The talk was arranged by IEEE Solid State Circuit Society. It was proposed that an analog MAC (multiply and accumulate) could consume less power than digital one. One issue, however, is that an analog MAC is a subject to analog noise. So, if you present it with ...


20

It's always better to have your algorithm implemented in a higher-level language (which C is compared to assembly), even if you plan to implement everything in assembly in the end. chances are, you won't even need assembly. If the code generated by your compiler meets your design goals, your job is done. if not, you won't be starting your assembly coding ...


18

What you're talking about is called an Analog Computer, and was fairly widespread in the early days of computers. By about the end of the '60s they had essentially disappeared. The problem is that not only is precision much worse than for digital, but accuracy is as well. And speed of digital computation is much faster than even modest analog circuits. ...


18

Your DSP will be advertised with a maximum sustained MACs, assuming all the pipes are filled. That is obviously an upper limit to what can be achieved. You know how many MACs your filters and other processing will take, from your analysis. Aim to have the first at least twice the second, as you will not be able to keep the DSP core running at maximum. Just ...


17

Ask yourself which cable is likely to have the lowest loop area: A large loop area has greater inductance and can emit more EM interference. It can also receive more EM interference. If each forward conductor has its own return wire then this potentially minimizes each circuits loop area.


16

There IS one analog technology that can be used to do the job ... the CCD "bucket brigade" delay line. It IS analog, but it has a lot in common with digital techniques in that it's a sampled-data system. A typical CCD delay line has 512 or 1024 capacitors in a line, and a network of CMOS switches to interconnect them. It works roughly as follows: Charge ...


14

Yes, the sampling rate can be any number you want. But you obviously would not get partial samples in the end, you just have to round down. In your example the first sample is taken at \$ \frac{1}{15.5}s \$ = 64.5 ms and then at every multiple from that. This means you get your last sample at 6,966 s. That is the 108's sample. So at 7 s you still have taken ...


13

I assume for "high speed" you mean a small delay from data collection to the resultant FFT. With a low sample rate, your computational ability isn't the limiting factor, given modern computers. The delay problem lies in having enough data for analysis. If you want your 1Hz bin to be different from DC/0Hz, you have to accumulate enough signal data to capture ...


13

Some things are always an integer. Samples are always integer. You can take 108 or 109 samples. Sample rate can be a floating point number, or more generally a rational, or even a real. You calculate the sample rate by dividing the number of samples (less one to get the number of periods between samples) by the time it takes to obtain those samples. ...


12

It's not (phase) (shift keying), for example. It's (phase shift) (keying). Another answer has explained why the word keying is used --- it dates to the days when modulation was controlled by a human operator using a telegraph key. The phase shift (or amplitude shift or whatever) is because something is changing (shifting) when keyed. If the system were ...


12

The physics explanation is that waveguides (including the free space) have orthogonal modes for the two propagation directions. This means that the two signals traveling in opposite directions will not interfere. (This is not an approximation, there will be no interference). The device which separates "transmitted" and "received" signal is a circulator. It ...


12

Capacitors block DC and pass AC. You can use a series capacitor into an opamp with whatever gain you need. Even better might be a simple RC high-pass filter...One capacitor (series) and one resistor (to ground) in front of your amplifier. Like this: simulate this circuit – Schematic created using CircuitLab R2 and R3 set your gain. C1 and R1 set ...


11

There are basically two types of attenuator I would consider and these can be combined in a couple of ways: - (A) is used because it offers simplicity with the ability to design it to suit the driving source and what it interfaces to (centre tap). (B) is used when you want to "ratio" down an AC voltage whilst not being concerned with the DC levels but for ...


11

Is analog signal division possible (as FPU multiplication often takes one CPU cycle anyway)? If you have an analog multiplier, an analog divider is "easy" to make: simulate this circuit – Schematic created using CircuitLab Assuming X1 and X2 are positive, this solves Y = X1 / X2. Analog multipliers do exist, so this circuit is possible in principle....


11

I'd suggest recording the signal on a tape and playing it back at half the speed. I cannot follow the reason why that does not satisfy you. Of course you could use other media (e.g. wires, disks etc.); the basic principle is the same. If none of that is good for you, you have to specify the requirements further.


11

There's your problem: telegraph signals are not electric currents. (We might as well say that telegraph signals are voltage instead.) Which is correct? Neither. To solve this, give up on electronics and instead fall back to the physics behind it. Actually the telegraph signals (and even all electrical signals everywhere) are actually electrical ...


10

I'm not sure what you're looking at, but you need to understand the exmples you link. None of them use the truth-value within the actual filtering. It's there so you have something to compare to with regard to the filter output. Here is the simple script: import random # intial parameters iteration_count = 500 actual_values = [-0.37727 + j * j * 0.00001 ...


10

Yes. Evaluate the function as s approaches zero and as s approaches infinity. That will give you a very quick look at low and high pass filters. Band pass can be a little trickier, and may require some factoring first to get it to a form that makes sense to apply the aforementioned process.


10

One usually needs to acquire multiple samples per waveform period to get good results from an FFT. The Nyquist limit of 2 samples per period is a lower bound but usually 10 samples per period or more is what is practically used. So to analyze a 64Hz signal you probably want to acquire samples at a rate of 640Hz or more. Also (up to a point) you will get ...


10

It's worth having a USB cable with ferrite 'stoppers' on it, to attenuate conducted emissions along the cable. This will reduce the chance of hash from the switch mode power supply and other PC generated interference 'getting into' the audio circuits of the ADC that's doing the recording. With an electrically noisy PC, and a cheap ADC, a cable without ...


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