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Feeds
Ability to change audioPlugin reported latency dynamically?
There are some limitations, for example latency cannot be changed during playback is active. But I'm not sure that's the case he...
etwa ein Monat vor | 0
acousticLoudness() function incorrect?
No need to modify your "signal". You can use calibrateMicrophone with a 1kHz tone that you specify the SPL level, and that will ...
3 Monate vor | 0
How to detect which audio output device is being used by Windows
The best practice for Audio Toolbox users would be to use WASAPI mode (see audioDeviceWriter). It is WASAPI exclusive mode, so t...
9 Monate vor | 1
Real-time ANC Simulink model based design simulation
Hi, the easiest way to get started might be to pre-record some audio and replace the noise source with that data. With an audio...
12 Monate vor | 0
Acoustic Loudness - Is Scaling Required?
Are you using audioIn and FS (the audio signal itself) or SPLin (28 one-third-octave bands between 25 Hz and 12.5 kHz) ? If you...
12 Monate vor | 1
| akzeptiert
Audio generation in Simulink Desktop Real-Time
If you are using the Speedgoat, you would want to use their device-specific output blocks instead of the Audio Device Writer. So...
mehr als ein Jahr vor | 0
Simultaneously play and record a signal with audioPlayerRecorder
To specify IN/OUT channels, use channel mapping. For example, to play on channels 1+2, and record on 4: audioPlayerRecorder("Pl...
fast 2 Jahre vor | 0
Error when recording multi-channel audio with loopback
Hi Charlotte, When using the reader and writer at the same time, some settings like frame size must match. Here, fileReader.Sa...
fast 2 Jahre vor | 0
| akzeptiert
Can I use two unique ASIO audio devices at once?
Unfortunately, Windows does not allow loading multiple ASIO drivers at the same time, so MATLAB cannot do so. You might be able ...
etwa 2 Jahre vor | 0
| akzeptiert
Sound Pressure level returning -inf
Leq, Lpeak and Lmax are only computed every TimeInterval, so they correspond to the SPL of zero for the first 0.125 seconds. And...
etwa 2 Jahre vor | 1
| akzeptiert
how to compute the RMS dB(A) value from an acoustic signal in a spectrum ?
It might just be a matter of scaling. For example, if there is a microphone involved, you may need to calibrate for that if the ...
etwa 2 Jahre vor | 0
What is dBFS in the Impulse Response Measurer App?
Hi Yuqing, If you use the Export button, you will get a table of your measurements including the recording. You can get an exc...
etwa 2 Jahre vor | 0
| akzeptiert
Will MATLAB VST plugin generation support ARA (Audio Random Access) API?
There is no support at this time, but users with Coder can use the "juceproject" option to produce a JUCE project, which support...
mehr als 2 Jahre vor | 1
Real-Time Audio Processing: Varying Delay Depending on Initial Under/Overruns
Some updates since these issues were raised: Starting 21b, Audio Toolbox I/O has a new buffering mechanism to prevent the laten...
mehr als 2 Jahre vor | 0
audio codec performance measurement DAQ
Hi, I think the Spectrum Analyzer could help. Here is an example: https://www.mathworks.com/help/dsp/ug/spectrum-analyzer-measu...
mehr als 2 Jahre vor | 0
| akzeptiert
Only one channel from two-channel Behringer UCA202
1. Try specifying the name of the device as well, in case you are picking up another (mono) device instead of you UCA202. >> a...
mehr als 2 Jahre vor | 0
Microsoft Visual Studio 2022 set up as default C and C++ compiler in Matlab 2021a
Plugin generation is part of the Audio Toolbox, which does not support MinGW. For plugins: VS 2019 was added to R2019b (appears...
mehr als 2 Jahre vor | 0
| akzeptiert
(solved) Why can't I generate an audio plugin from the StereoWidth example?
I see you found a solution. Just for reference, here is a list of supported compilers for Audio and other toolboxes (there is a ...
mehr als 2 Jahre vor | 1
audio input that is not recognize by matlab
Hi Fanny, First is to make sure the OS recognized the input devices. If on Windows, check that your devices show up in "Sound s...
mehr als 2 Jahre vor | 1
| akzeptiert
Impulse Response Measurer: Amplitude of recorded signal is almost half compared to recording in audacity
Playback level in the impulseResponseMeasurer is controlled by the "Excitation Level (dBFS)" setting in the toolbar. Does settin...
fast 3 Jahre vor | 1
| akzeptiert
Impulse Response Measurer in audio toolbox
Yes, sound is output on the selected device and channel. If there is a loudspeaker connected to that device, it will play. But i...
fast 3 Jahre vor | 0
How to access 'non-automatable' parameters in a AU/VST in MATLAB with `loadAudioPlugin()`?
Hi Jamie, The ability to display the plugin's custom GUI is not currently available. If you'd like to share what's your affilia...
fast 3 Jahre vor | 0
How can i fix this error in the Impulse Response Measurer?
Hi Mehmet, could you try increasing the “Duration per run” relative to the “sweep duration”. The sweep methods requires this per...
fast 3 Jahre vor | 1
| akzeptiert
Parameters for room impulse response measurements using swept sine
Hi Jon, Since you do not have access to MATLAB on site, you will want to use sweeptone to create the exponential swept sine tha...
mehr als 3 Jahre vor | 1
| akzeptiert
Sending a midi message from MATLAB to an external source audio plug in
If you want to use MIDI (as opposed to loading a plugin), have a look at midimsg. Just note that we don't support loading VSTi p...
mehr als 3 Jahre vor | 0
How to use synthesizer VST plugin after loading it using loadAudioPlugin function?
Unfortunately we do not support VSTi (MIDI instruments) at this time.
mehr als 3 Jahre vor | 0
| akzeptiert
How do I interrupt a sound file that is currently being played?
You should write your audio samples by smaller frames (such as 1024 samples per frame). That way, you can stop writing at any ti...
mehr als 3 Jahre vor | 0
How do I play more than one sound file concurrently with audioDeviceWriter?
If you use audioDeviceWriter (or audioPlayerRecorder), you need to manage each frame yourself, including adding any signals that...
mehr als 3 Jahre vor | 0
Audio toolbox: using acousticLoudness() in reverse for "normalizing" audio signal
If you want average perceived loudness, consider omitting the time-varying option even though the signal isn't stationary. Othe...
mehr als 3 Jahre vor | 2
| akzeptiert
How do I implement this code to the example called Active Noise Control Using a Filtered-X LMS FIR Adaptive Filter?
Hi Sean, I'll suggest that you also have a look at the Real-Time Example (linked here). The beginning of that example gives deta...
fast 4 Jahre vor | 0
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