Puneet Rana

MathWorks

Attivo dal 2012

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Times 2 - START HERE
Try out this test problem first. Given the variable x as your input, multiply it by two and put the result in y. Examples:...

circa 7 anni fa

Risposto
High Latency with ASIO driver (Behringer UCA222, Audio System Toolbox)
What is the frame size of the input that goes into your Audio Device Writer block? What is the SamplesPerFrame used in Audio Dev...

oltre 7 anni fa | 0

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Recording and reading audio in realtime
Hi Michael, What do you want the audioDeviceWriter to play? The recorded audio? The audio from a file? In either case, the se...

oltre 7 anni fa | 0

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How do I call the dsp toolbox "peak finder" from code?
Hi Tianqi, You can run findpeaks on the result of <http://www.mathworks.com/help/dsp/ref/dsp.spectrumestimator-class.html dsp...

oltre 7 anni fa | 0

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Am I doing something wrong, or does fdesign.octave not actually work?
Hi Jerome, The red color on the mask does not mean that the ANSI compliance is not met. When fdesign.octave is used with fvto...

oltre 7 anni fa | 0

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ASIO audio driver with MATLAB 2016a
Axel, You can continue using ASIO with DSP System Toolbox in R2016a by changing the driver using MATLAB command-line as descr...

circa 8 anni fa | 2

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Computing and Ploting continuous Fourier transform in simulink
You can look at the Spectrum Estimator block in DSP System Toolbox: <http://www.mathworks.com/help/dsp/ref/spectrumestimator.htm...

oltre 8 anni fa | 0

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real time audio processing and algebraic loop issues
The algebraic loop is in SD2 subsystem. This will help you identify it: <http://www.mathworks.com/help/simulink/ug/algebraic-lo...

quasi 9 anni fa | 0

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Matlab Upsample Filter Object
Hi Morgan, You can do this by setting Numerator of interpolator to [1,0]. For example, using the dsp.FIRInterpolator System o...

quasi 9 anni fa | 0

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Carrying filter state from adaptive to non-adaptive
Instead of adaptfilt.nlms, try the recent dsp.LMSFilter object: <http://www.mathworks.com/help/dsp/ref/dsp.lmsfilter-class.ht...

oltre 9 anni fa | 0

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How can I process audio to get frequencial information in real time?
Try the Spectrum Analyzer block: <http://www.mathworks.com/help/dsp/ref/spectrumanalyzer.html> It also has a 'spectrogram' ...

oltre 9 anni fa | 0

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How to preserve base band signal's bandwidth
For up-conversion you typically need to first increase the sample rate of your input through (multistage) interpolation and then...

oltre 9 anni fa | 0

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sample rate of a DUC object
The concept of sample rate is a bit different in MATLAB, since it isn't directly attached to a signal. In the case of System obj...

quasi 10 anni fa | 0

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Plot adaptive filter coefficients
There are a lot of ways to achieve this: * Use XY Graph for convergence: <http://www.mathworks.com/help/dsp/examples/adaptive...

quasi 10 anni fa | 0

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Signal Processing vs DSP System Toolbox -- Which One?
To add to Star Strider's answer above: * DSP System Toolbox (DST) has more specialized filter design algorithms (e.g., multir...

quasi 10 anni fa | 0

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implementing multistage multirate filters
A more detailed example is here: <http://www.mathworks.com/help/dsp/examples/multistage-design-of-decimators-interpolators.ht...

quasi 10 anni fa | 0

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Simulink CIC show output in scope
You can use the 'Unbuffer' block before the scope block.

oltre 10 anni fa | 0

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How can i set block parameters from m-file?
Hello Sergio, 'DeviceName' is the property of the 'To Audio Device' and 'From Audio Device' that contains the currently selec...

quasi 11 anni fa | 0

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Computing the inverse of a matrix without using the 'backslash' command
You can use the Moore-Penrose pseudoinverse as follows: solver=pinv(I-A)*d

circa 12 anni fa | 0

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