Puneet Rana

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Con actividad desde 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:...

alrededor de 7 años hace

Respondida
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...

alrededor de 7 años hace | 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...

alrededor de 7 años hace | 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...

más de 7 años hace | 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...

más de 7 años hace | 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...

casi 8 años hace | 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...

más de 8 años hace | 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...

más de 8 años hace | 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...

más de 8 años hace | 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...

más de 9 años hace | 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' ...

más de 9 años hace | 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...

más de 9 años hace | 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...

más de 9 años hace | 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...

más de 9 años hace | 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...

casi 10 años hace | 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...

casi 10 años hace | 0

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

alrededor de 10 años hace | 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...

más de 10 años hace | 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

alrededor de 12 años hace | 0

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