How To Convert Digital To Analog In Matlab

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84 Threads found on edaboard.com: How To Convert Digital To Analog In Matlab
zedman, ok I will do it at home tonight (within 10 hours from now...) the icd2 is at home and i'm at work for now. Added after 1 minutes: well, this usbmon is a nice piece of software.. do you know how long it works in demo mode ? can't find it anywhere..
I use TLC5540 for my home DSO,but TLC5540' min sample rate is if i need low sample rate(example 100KHZ),how can I do with it. my systerm:TLC5540+XC9572+MCU Skip some samples, sampling on 5MHz and use each 50 sample ? Or use signal averaging by software ?
Hi, I'm really interested in this class I'm master student, however i'm not good at matlab
how did you perform the integration operation using VHDL? M
Hi pals, Currently, i haf this problem. I wanted to down convert some IF signal.I have done quite an amount of my main program, left the last part which is my problem i'm facing. how should i implement the Direct digital Synthesizer(DDS)? It works like that, my input,"Xi2 & Xr2", with a 120MHz will be compared with signal generated by DDS (...)
Added after 11 seconds: The oscillator is a clapp oscillator that is meant to be frequency modulated by the two varactor diodes. The oscillator is designed to transmitt on the FM band (88 MHz to 108 MHz). The LM 386 op amp output is coupled by the
I am going to do two antennas, one for transmitting and the other for receiving. I will be transmitting a simple sine wave at around 600MHz and for the receiving side, I will be using an equivalent antenna, same as the one used for transmitting along with filters and amplifiers to capture the signal and amplify it so i can analyze it using a softwa
Do you have SPICE model for photodiode ? how to simulate it ?
as I know , delta-sigma A/D have 2 block , 1 is analopg modulator it is easy implement by comparator -> use 1/2/3 order over sample and 2 is digital filter , I think many analog design book already talk about how to make delta-sigma modulator , but really deisgn flow is (1) use C or matlab or saber simulation (...)
how can you compute fourier transform form Z-transform ? Why IIR filters doesnt have Linear phase? Can IIR filters be Linear phase? how to make it linear Phase? What is the advantage of a Direct form II FIR over fom I? Tell some thing about Interpolation and decimation? What is Interpolation and decimation filters and
HERE IN THE US THERE IS NOTHING HOT .. NO DSP ,NO FPGA .. NO WIRELESS SO DON"T BE FOOLED ! ..EVERY THING IS IN STANDBY .. IF ANY ONE SAYS THE CONTRARY WELL HE BETTER GIVES SOME PROOFS . OTHERWISE IS NOT GOOD TO GIVE FALSE HOPES ! Somebody called me with "A NICE FPGA CONSULTING JOB " a day ago .. I responded inquisitive and asked .. how much ar
useful. 8) If you are interested how @ltera does this: @ltera QPSK Modem Referance Design an281.pdf No need point if download from here
Hi , I have a system with an ADC that recieves an analog signal that may have values in the 0 - 5 V range. The signal must be filtered in the digital domain. My question is how should the 8 bit samples be converted to match the format of the filter coefficients? Thanks.
@ bepobalote: You mentioned "if your Line-in input is stereo: if yes it will be OK for your requirements. " Does this mean that there should be two input jacks (line in)as i have to input 2 signals (I and Q).... No, you will use the Left/Right channel as I/Q, so you will need a stereo jack. here there are some link
As an electrical engineering undergraduate student, I think Calculus , Differential equations, Complex veriables, Probability are very important. Learning matlab is a good idea , however, I do not think that it is really necessary. You should try to learn thos topics related to digital signal processing, and signals and systems very (...)
I am just curious how to implement algorithm into hardware. Let say, I have written a convolutional in C++ language for simulation purpose. Then, what should I do to implement in hardware?Is it using FPGA or VHDL?Any good informations about it? Thanks.
hello, i want to ask about the excess loop delay in continuous time delta sigma , i know how can see it's effect on the stability of the modulator by using matlab, but what's the model for it ? thanks This paper discusses how to model excess loop delay effects in continuous time sigma delta modulators. Excess loop
but,my hydrophones can only receive and transmit analog signal... What model of the hydrophones are you using? i'm using hydrophones type 8104(transmitter) and hydrophones type8105(receiver).. Thanks! Do you know what company makes these hydrophones or where they can be bought?
Dear echo47, Thanks for your help. Let me make things specific. If I want to design a system, which can make a image inverse. That is, if the pix of the point in the image is 0, then after this system, the pix of the point will be 1. how do we implement this functionality by using FPGA?
OK well i wanted to test something like the following using SPICE simulation. I have some VHDL which is just a PWM generator. I would like to add this VHDL in a A/D circuit in PSPICE and test it with some different value resistors and capacitors for a low-pass filter for the output... Basically to create a 1-bit DAC. But i have no idea how to do
First of all I would think that you don't have enough resolution from your temperature reading. With 20mV you are reading temperature in 2? increments and you could expect that your system will not be abe to stabilize and at very minimum would oscilate between two values 2? appart. Second, if you are using opto isolator with triac output you
In the filters if you use high value resistors then you can use low value small accurate metallized poly capacitors. The circuit from the university is very old. It uses very low noise opamps that have a very low input offset voltage. The lousy old 741 opamps that you use have the noise that shows on your 'scope. The university circuit has an
Hi all, I want to simulate the phase noise of the whole PLL. It is easy to simulate the phase noise of the individual blocks in the PLL, but how to simulate the phase noise of the whole PLL? Should we use Mablab or Cadence to build the noise model for the PLL? Any suggestion or links are welcome. Lunren
Sampling --------------------------- converting an analog signal to a digital one is a necessary step for a computer to analyze a signal: modern computers are digital machines, and can store only digital values. In the continuous function x(t), we replace t, the continuous variable, with nTs, a discre
Q1 : sir i want to ask you that if i am using an external ADC which have 400ksps t0 sample the frequency of 100khz. if i applied input of 100khz to ADC which converts into bits and these bits can serialy transmit to matlab through pic16fxxx, so it is possible for pic to work as fast to transmit all the bits before taking
Hello, I'm totally new in FPGA field, so i hope i can get as much possible guidance here if possible...Thank you everyone for your time and support. I am currently trying to process video using FPGA Board. It is just a simple mapping of pixels. my process will be something like: Export a continuous video (taken using webcam), into FPGA, th
Hi Anybody know any material which contains how to model differential equations to electronic circuit ? Thank You
how to simulation SNDR on hspice ?? someone said use .Four Fin and will find THD , SNDR=THD
how to use RTL code implement it ?? some filter like sinc filter use matlab simulation is ok , but how convert it to RTL code ?? some equation --> ( 1+ a X + b X^2) / (1+ c X + d X^2 ).. like this equation how to implement ?? use "shift register " for divide ?
Simple FM Receiver: Overview Details Name: simple_fm_receiver Created: 03-Jan-2005 08:33:23 Updated: 29-Mar-2005 07:03:20 CVS: browse Other project properties Category :: Other Language :: VHDL Phaze :: FPGA proven Development status :: Production/Stable
The process has some variation. E.g. if Vth nominal is 0.5V (so called "typical value"), actually when you fabricate it might be anywhere between 0.45 and 0.55 with some distrbution. There are lots of parameters and each of them has some variation or uncertainty. Corners take the extremes, like most unfavorable case and most favorable
Hi HanGu, I have checked that path. Actually the I have searched by "find" sitting in the mother directory. But, I don't find it. It would really be helpful if you give me the document, if u have it. looking for ur help... sankudey I would be also interested in getting the file. BTW, I really need to find out how to
hi all, does anyone knows how to convert actual freq to normalized freq from ? Eg: Actual freq=500hz. Sampling freq=5khz. thanks regards scdoro
ok if i want to give equivalent digital signal of analog .....how i can get those digital signals.. thanks in advance
Hi Robin, Although I understand academicaly your description, I can't really follow it as I am new in signal processing ( ver, very new). So, I will be more detailed in want I want and what I understand: I have a video signal. Two cases exist there: 1. Signal is allready separeted in R, G, B. 2. Signal is raw video. I guess this is
Not in the way we've been discussing here. An ordinary sampling oscilloscope doesn't have the oscillator, mixer, and filters to convert the sampled data back up to 200 MHz. however, a sampling oscilloscope can store thousands of those 60 ksps points and plot them at the correct places on the screen to gradually build up the original waveform, as
Your ECG signal has very low amplitude at the fundamental frequency, so a plain FFT would give you poor info. It would be better to apply some sort of non-linear filter to it first, such as computing fft(ECG_1 > 0.5) instead of fft(ECG_1). This example shows the first strong spectral peak at about 1.23 Hz. Zoom in to see it: clear; lo
Hi all, Anyone knows how to build a DAC in matlab or simulink? Just a basic and simple one is good enough. What I want to experiment is to convert a digital signal to analog signal, and filter it by an analog filter. Thanks, cfy30
Have you tried to have a look at the CYFI kit (CY3271) of Cypress? It is the first touch kit for Cypress's Wireless protocol CyFi. Read from the following link: It comes with a PC dongle with a usb interface that can be used as a hub, an RF node and a multifunction sensor card. It basically has a
I'm want to implement Band Stop digital filter for audio wave using VC++. For this Which FFT is good? how can i manipulate audio samples? What is the nature of samples? When i tried ;the samples are of int type.For calculations using radix-2 fft these int values are converted to decimal-point values.Here is my doubt...While casting (...)
two tone IMD can be a tricky measurement,because the additional equipment %required(a power combiner to combine two input frequencies) can contribute %unwanted intermodulation products that falsify the ADC's intermodualtion %distortion. Can a power combiner produce intermodulation???? Power combiners are passive systems! how can a power
m working on speech recognition project i want to convert my audio into .mat file using matlab code plzz help me out thanx can u plzz tell me how to convert an audio signal in .mat file . ie to convert an analog to digital it's very urgent
Hello everybody, a need to produce a digital FM demodulation in matlab but i am a bit confuss, because i have always studied that fm is an analog modulation so why digital? how can implement the digital fm demodulation? where can i find the code? thank you very much
@erikl: the method you mention is totally new to me and could potentially cut simulation time by a huge amount. Do you have any reference I could use to learn how to choose the number of sine waves wrt the number of bits, for instance (your choice of 8 seems somewhat arbitrary). Thank you! @zvilupu: Not having known erikl's method,
You might create a grid of 32 columns and 32 rows. Each column is a wire. Each row is a wire. Where they intersect, you have one component. It is connected across the two wires. You poll an image by turning on the column where it is located, and sensing it at its row. Such a layout is similar to how computer keyboards are read. They use a grid
how about digitizing it at higher than the Nyquist rate and than slowing down the readout rate of the signal to lower its frequency. Reading out at 1/8th speed would lower 80kHz to 10kHz and 20kHz to 2.5kHz . You could also dynamically adjust the amplitude upon playback to adjust the dynamic range. Of course that would also lengthen the s
I think we maker sure , first , delta-sigma A/D or D/A use "over sample" & "noise shaping" we use over sample by modulator (a comparator) and delta-signma modulator have 1/2/3 ..order ( hi order will unstable but over sample is small , ex. if you use 2 order , you need x256 sample , if you input signal=1M , you need a 256M c
Hi how to use matlab write a Low_pass filter .. I can use simulink for simulation , but I want to know how to write a matlab code.. simulink file can novert to matlab code or not ?? a low pass filter = 1/ (s+1) = e^(-t) but how to descript expt(-t) in (...)
If you can generate a PWM signal you can make a simple D/A converter. Just low pass filter the PWM signal. This link explains how it is done.
in iir filter design, we design the equivalent prototype analog filter, and use the bilinear transformation to convert H(s) to H(z). My question is: how do we select the bandstop & bandpass frequency for the equivalent low-pass analog filter in the case of designing digital bandstop, or bandpass?