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DSP basics,fundamentals,software

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walters

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what software is easy to write DSP algorithms for phase shifting and short time delays ?

What is the basics and fundamentals of DSP? what kind of software can do all the math algorithms for u and has sample templetes already done like a library? what else is the basics and fundamentals of DSP in a block diagram the terms of DSP?

where Is there a glossary or dictionary about DSP terms please?

How can i make different waveforms using DSP what would i need to know to do this please?
 

Your questions are vague, showing a leak of documentation on the web before asking.

The software tools are specific for the DSP core you indend to use. For fast algorithms you need multicore faster DSP. Some of them have C compilers. Others need low level programming. At this moment there are on the market DSPs running at 2.4Ghz with four core inside. Analog Devices have DSPs at 750MHz with two core inside. Take a look what software they use.

On the two and third question please google, there are a lot of info about this.

On the fourth question you didn't specified if you want analogic or digital signals and I don't want to quess. Presuming you need analogic signals you need a fast DAC connected with a GPIO port of an DSP.
 

yes i did find a glossary or dictionary about DSP terms on google

My C++ classes didn't have DSP instructions just plain office math programs never did algorithms or math waveforms

When u buy a DSP chip it comes with a special C++ with DSP instructions added library right?

DSP would have to have A/D and D/A converts for analog right?

I never heard of a DSP thats pure analog it has to have a D/A converters

The DSP chips come with built in tremolo,phase shifters,chorus,delay lines,flangers sound effect waveforms and LFO oscillator waveform shapes,envelopes that u can change with DSP code with the manual of the DSP chip?
 

DSP Phase shifter
1.) notches that are non periodical
2.) IIR comb filters

DSP Flanging
1.) notches of the comb filter which are periodical in frequency
2.) FIR comb filters

DSP vibrato
1.) IIR comb filter

Reverbs
1.) IIR comb filter
2.) Acoustical Cylinder
3.) frequency dependant feedback coefficient

DSP filters
1.) sequence of multiplication's and additions
2.) utterworth, Chebyshev, Inverse Chebyshev, Elliptic, Kaiser Window, Dolph-Chebyshev, and Equi-Ripple
3.) bi-quad digital filter structure
4.) "Windowed" filters
5.) filter complexity number of taps
6.) vector of polynomial coefficients
7.) Recursive and non-recursive filters
8.) transfer function of a digital filter this sets the frequency response
9.) output sequence for the filter
10.) Coefficients of recursive (IIR) digital filters
11.) filter coefficients-values of these coefficients determine the characteristics of a particular filter

FIR (Finite Impulse Response) filters
1.) can create transfers functions
2.) Calculation coefficients
3.) symmetric FIR filters
4.) NON-Recursive filters
5.) Fixed point math processing
6.) FIR comb filter feedback

IIR (or Infinite Impulse Response) filter
1.) Recursive filters
2.) Floating-Point math processing
3.) IIR comb filter feedback

Software programs to make DSP IIR or FIR digital filters:
1.) Matlab graphical user interface (GUI)
2.) LabVIEW Digital Filter
A recursive filter is one which in addition to input values also uses previous output values. These, like the previous input values, are stored in the processor's memory.recursive literally means "running back", and refers to the fact that previously-calculated output values go back into the calculation of the latest output. The expression for a recursive filter therefore contains not only terms involving the input values
non-recursive filter is known as an FIR (or Finite Impulse Response) filter, and a recursive filter as an IIR (or Infinite Impulse Response) filter. These terms refer to the differing "impulse responses" of the two types of filter
The impulse response of a digital filter is the output sequence from the filter when a unit impulse is applied at its input. (A unit impulse is a very simple input sequence consisting of a single value of 1 at time t = 0, followed by zeros at all subsequent sampling instants). An FIR filter is one whose impulse response is of finite duration. An IIR filter is one whose impulse response (theoretically) continues for ever, because the recursive (previous output) terms feed back energy into the filter input and keep it going. The term IIR is not very accurate, because the actual impulse responses of nearly all IIR filters reduce virtually to zero in a finite time. Nevertheless, these two terms are widely used

What else do i need to know please about IIR and FIR filters

What about for LFO oscilloscope waveforms what does DSP use for this please?
 

better how and why?
 

matlab is common known with a lot support , but labview is easier!
 

labview to do DSP programming how?

How can i programm IIR or FIR filters alogorthms coeffients and parameters in Labview?
 

Bro, I would say C or CPP is more efficient for embedded system, except ASM.
 

But Labview has a GUI so i don't have to do all the C++ coding i just use the GUI interface its more easier
 

dsp book can be a correct answer for u i think
naveen
i can help u out
clearly ask me ur questions
 

Hi walters
Nothing can beat matlab in signal processing.It has all these built in functions wat u have specified.generating coefficients and filters is very easy with the matlab FDAtool.
 

two tool available. labview and matlab. but matlab is the best option, as it is very much user friendly. also many of the codes are available on the matlab central website. its h**t//www.mathworks.com/central
 

Hi,
Transfer functions of IIR filters are rational (eg a(z)/b(z)).

There are four types of FIR filters (type 1 ,type 2 , type 3, type 4) depending on the symmetry and the length (odd and symmetrical, even and anti-symmetrical..etc)

Filter design is the process of finding filter coefficients given the desired frequency response characteristics.

stablity analysis is required when IIR filters are designed especially when fixed point arithmetic is used.

Relationship between the pole,zero locations on the z-plane and the corresponding frequency response.

Fixed-point analysis , scaling of coefficients, finite wordlength effect . (overflow, precession loss)

MATLAB filter design and fixed point toolboxes

Simulink.

Filter Structures.
 

Is Matlab easy to write code instruction IIR and FIR filters?

Labview i heard has a (GUI matlab) its matlab but its a GUI interface so its really easy to program IIR and FIR filters right?
 

walters said:
Is Matlab easy to write code instruction IIR and FIR filters?

Labview i heard has a (GUI matlab) its matlab but its a GUI interface so its really easy to program IIR and FIR filters right?

You can access the GUI-based filter design tool in MATLAB by this command :

fdatool

Which will start the Filter Design and Analysis Tool.
 

thanks for the help

So u don't have to program code its a GUI with interface parameters to adjust by the mouse? and u can save the presets from adjusting the parameters/knobs from the GUI interface without having the program code at all in Matlab?
 

walters said:
thanks for the help

So u don't have to program code its a GUI with interface parameters to adjust by the mouse? and u can save the presets from adjusting the parameters/knobs from the GUI interface without having the program code at all in Matlab?

Yes. And after the filter is designed, the filter coeffecients generated can be imported to a vector variable that you can use in the workspace. You can also create your own GUI if you want but that's a different story.
 

aren't the filters IIR and FIR already designed from the GUI all u do it adjust the GUI's parameters from the GUI interface

The IIR and FIR filters are already done for us and designed already right? all the hardwork is done for us the code has been writen?

The GUI interface can changed or modify the filter coeffecients without programming in code just using the mouse and GUI interface?

What does the vector variables do to the filters coeffecient?
 

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