High latency for audio use on a Raspberry Pi using Simulink

Hi everybody!

For a student project we decided to build test hardware for musical instrument effects. As base we decided to go for a Raspberry Pi with some peripheral equipment installed. We've programed the hardware just using Simulink with their tools and libraries. Everything works fine but we do have a huge latency with our audio signal.

Even when we try to put the signal through without effects, the latency is half a second long.

Do you have any ideas why?

Does the Raspberry Pi compute on its own or is all data just sent via ethernet?

How do I reduce buffer sizes?

Thanks for your help!

Regards, Amadeus

Answer 1

[Tom Haelvoet]

Hi all,

The latency can be reduced from within Simulink after performing the following steps:

1. Drag & drop an ALSA Audio Playback block into your model.

2. Jump to the Library (CTRL+L) and unlock it.

3. Edit the mask (CTRL+M) of the ALSA Audio Playback block.

4. Navigate to Parameters & Dialog, Select Que duration & check the Visible flag.

5. Lock & save the library.

The Queue duration should now be visible and ready to be modified.

FYI: Tested with a Focusrite 2i2 (Device name = 'plughw:1.0') and a Queue duration of 50ms

[Amadeus]

Thank you very much!

That helped indeed.

It is better to get some help lately than have none. It works so much better now!

Regards, Amadeus

Answer 2

[Murat Belge]

The ALSA Audio Playback block uses and audio buffer size of 0.5 seconds of data. You can not change this parameter unless you modify the source file (MW_alsa_audio.c). This parameter determines the latency.

It is very hard to achieve real-time, low-latency audio in Linux. If you reduce the audio buffer size, you might observe stutter / interruptions in the audio.

Answer 3

[Dong-uk Hyon]

Hi,

On top of Tom's answer above, I am also carrying out audio experiments using Raspberry Pi and have found lowering the ALSA frame sample size down will significantly reduce the latency.

I found these settings by double clicking on the ALSA audio capture block. The default frame size for me was 4410 (N), by reducing this value you will find a better response of your system. I managed to go down to 10, and the delay is just about noticeable by audio.

Info found here: https://uk.mathworks.com/help/dsp/ug/sample-and-frame-based-concepts.html#bso3wxr-1

Kind regards, Don