How to reduce Audio Latency by MATLAB DSP System Toolbox?

Question

I have been working on my ANC project. For this I have two microphone inputs and one loud speaker output, but initially I am using single microphone and dspStreamingPassthrough to pass microphone input to loud speaker. Here is my code

% Initialization
numIterations = 500;
% Construct sources (for all inputs)
src1 = dsp.AudioRecorder('DeviceName','Mikrofon (USB-Audiogerät)','NumChannels',1);
% Construct sinks (for all outputs)
sink1_1 = dsp.SpectrumAnalyzer('SampleRate',44100, ...
'PlotAsTwoSidedSpectrum',false, ...
'ShowLegend',true);
sink1_2 = dsp.TimeScope('BufferLength',44100, ...
'SampleRate',44100, ...
'TimeSpan',1, ...
'ShowLegend',true, ...
'ShowGrid',true, ...
'YLimits',[-0.5 0.5]);
sink1_3 =
dsp.AudioPlayer('BufferSizeSource','Property','BufferSize',1024,...
'QueueDuration',0,'OutputNumUnderrunSamples',true);
sink1_3.DeviceName = 'Lautsprecher (USB-Audiogerät)';
 % Stream processing loop
clear dspStreamingPassthrough;
for i = 1:numIterations
% Sources
in1 = step(src1);
% User Algorithm
out1 = dspStreamingPassthrough(in1);
% Sinks
step(sink1_3,out1);
step(sink1_1,out1);
step(sink1_2,out1);
nUnderrun=step(sink1_3,out1);
end
% Clean up
release(src1);
release(sink1_1);
release(sink1_2);

I am using Windows DirectSound Audio driver ( I cannot use ASIO driver as I cannot access individual audio devices names. ! ) Now I have the audio latency of 1.2 seconds i.e if I say ''hello'' in microphone now, after 1.2 seconds speaker is saying ''hello''(this is absolutely with out any audio input data processing just 'dspStreamingPassthrough'). How to reduce this incredible delay ?

For my project of 1 meter length pipe(air duct), I should be able to process the data in 1.7 msec or less !! I have tried with lowest 'BufferSize' and lowest 'QueueDuration' possible !!

What other parameters can influence to speed up this process? Is it possible with MATLAB or not ?

PS: -sorry for whole code. -I am using a cheap quality Sound card (7 euros)

Answer 1

DirectSound has way higher latency than ASIO because DirectSound is not suited for low-latency apps. DSP System Toolbox does not support WASAPI yet.

The latency performance of these objects was greatly improved starting in 15a. I am not sure which version you are running but try and upgrade to 15a or higher.

As for tuning your latency, try the following: * Set the Queue Duration property to 0 seconds for both player and recorder. * For the recorder, match the SamplesPerFrame and BufferSize property. * For the player, ensure that the size of the data matches the BuferSize property.

The BufferSize property is the size at which the sound card operates.

If you get drops, increase the BufferSize value. There can be many reasons for the drops: * The algorithm you are running is not faster than BufferSize/SampleRate * The soundcard is not able to operate at this BufferSize. Some sound-cards allow you to modify this when using ASIO. * Limitation of the player/recorder objects.

Hope this helps.

Dinesh