# Tag Info

47

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 ...

46

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. ...

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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 ...

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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; ...

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 ...

24

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

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

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 ...

21

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 ...

21

The only way that a signal that is "buried in the noise" can be detected is if you can run the signal + noise through some filter that attenuates the noise more than it attenuates the signal. At which point the signal is no longer buried in the noise, so "buried in the noise" was just a hasty assumption. In radio carrying an audio (or Morse code) signal in ...

21

What you're missing is the bandwidth, both of signal and noise. If you look at, let's say, a 1 V rms sinewave signal, together with 10 V rms noise on an oscilloscope, you'll see only noise. However, if the noise occupies a 1 MHz bandwidth, and is flat with frequency, and you pass the signal + noise through a 1 kHz bandwidth filter centred on the signal, then ...

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 ...

19

Compression is all about finding the redundancies in the data and removing them. Since you don't seem to be able to tell us much about your actual data sets, this answer will have to be very generic. I gather that "potentiostat" data is continuous, varies slowly in general, but might have small deviations from sample to sample. One good way to ...

19

The system takes advantage of a couple of mathematical properties of the PRNs: First, the cross-correlation of a PRN with itself (i.e. the auto-correlation) looks somewhat like a single impulse with some noise, and second, the cross-correlation of a PRN with a different PRN looks like noise without an impulse. We also note that cross-correlating an unknown ...

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

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.

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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 ...

18

Have there been any projects trying to explore that? Ben North attempted to play video from a standard audio cassette tape using QAM modulation. First he tested the signal through a loopback cable. The resulting constellation looked pretty good:- Then he tried recording to an actual cassette tape in an NAD 6220, and playing it back with a Sony WM-EX194. ...

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 ...

16

It is a result of calculating the number of bits after the imperfections such as noise and distortion are taken into account. An ENOB of 6.8 basically tells you that for example an 8-bit ADC has real world performance that is better than an ideal 6-bit ADC but worse than an ideal 7-bit ADC. You can also think of it as not having 2^6=64 or 2^7=128 discrete ...

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 ...

14

The KTC noise is actually the total RMS noise power at the output of the low pass RC filter. The noise power is the product of the noise power spectral density (PSD) and the bandwidth (more accurately "effective bandwidth") of the system. So, actually the total noise power does depend on the bandwidth. Interestingly, for the case of RC filter although the ...

14

Many texts prove that a signal cannot be both time limited and bandlimited. It is quite a deep result and depends on complex analysis, but the shortest proof I know starts with a bandlimited signal $f(t)$. It is straightforward to show from the fourier transform that being bandlimited means that $f(t)$ is analytic over the entire complex plane, hence if ...

14

A fixed DC signal has information - the DC offset - that would be lost if it were blocked (that is, reset to 0V potential.) The bandwidth of that information is nevertheless 0Hz. This idea comes up in information coding, such as the DCT used in JPEG. Each block is assigned a block-average DC offset at the beginning, and subsequent components are coded ...

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

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 ...

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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 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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