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High latency for audio use on a Raspberry Pi using Simulink

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Amadeus
Amadeus 2014 年 11 月 24 日
回答済み: Dong-uk Hyon 2018 年 3 月 1 日
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

回答 (3 件)

Tom Haelvoet
Tom Haelvoet 2016 年 10 月 8 日
編集済み: Tom Haelvoet 2016 年 10 月 8 日
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
  1 件のコメント
Amadeus
Amadeus 2016 年 12 月 13 日
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

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Murat Belge
Murat Belge 2014 年 12 月 10 日
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.

Dong-uk Hyon
Dong-uk Hyon 2018 年 3 月 1 日
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

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