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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...
mer än 8 år ago | 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...
mer än 8 år ago | 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...
mer än 8 år ago | 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...
mer än 8 år ago | 0
| accepted
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...
mer än 9 år ago | 2
| accepted
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...
mer än 9 år ago | 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...
nästan 10 år ago | 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...
nästan 10 år ago | 0
| accepted
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...
mer än 10 år ago | 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' ...
nästan 11 år ago | 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...
nästan 11 år ago | 0
| accepted
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...
ungefär 11 år ago | 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...
ungefär 11 år ago | 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...
ungefär 11 år ago | 0
implementing multistage multirate filters
A more detailed example is here: <http://www.mathworks.com/help/dsp/examples/multistage-design-of-decimators-interpolators.ht...
ungefär 11 år ago | 0
Simulink CIC show output in scope
You can use the 'Unbuffer' block before the scope block.
mer än 11 år ago | 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...
nästan 12 år ago | 0
| accepted
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
mer än 13 år ago | 0
| accepted
