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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:...
mehr als 8 Jahre vor
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...
mehr als 8 Jahre vor | 0
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...
mehr als 8 Jahre vor | 0
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...
mehr als 8 Jahre vor | 0
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...
mehr als 8 Jahre vor | 0
| akzeptiert
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...
etwa 9 Jahre vor | 2
| akzeptiert
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...
mehr als 9 Jahre vor | 0
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...
fast 10 Jahre vor | 0
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...
fast 10 Jahre vor | 0
| akzeptiert
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...
mehr als 10 Jahre vor | 0
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' ...
fast 11 Jahre vor | 0
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...
fast 11 Jahre vor | 0
| akzeptiert
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...
fast 11 Jahre vor | 0
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...
fast 11 Jahre vor | 0
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...
etwa 11 Jahre vor | 0
implementing multistage multirate filters
A more detailed example is here: <http://www.mathworks.com/help/dsp/examples/multistage-design-of-decimators-interpolators.ht...
etwa 11 Jahre vor | 0
Simulink CIC show output in scope
You can use the 'Unbuffer' block before the scope block.
mehr als 11 Jahre vor | 0
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...
fast 12 Jahre vor | 0
| akzeptiert
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
mehr als 13 Jahre vor | 0
| akzeptiert
