Real time audio signal from microphone

Hello everyone, I am working with Matlab 2018b.

Im trying to develop a software wich aims to process in realtime a voice audio signal coming from the microphone.

This is the way I create an object and I ask to process the signal every "2*round(0.04*44100)" samples using a function called Auditive2

global f0l; f0l=[];
% global frames; frames=[];

d = daq.getDevices;
dev = d(2);
s = daq.createSession('directsound');
addAudioInputChannel(s, dev.ID, 1);
s.IsContinuous = true
procesamiento = @(src, event) Auditive2(event.Data, src.Rate);
hl = addlistener(s, 'DataAvailable', procesamiento);
s.NotifyWhenDataAvailableExceeds =2*round(0.04*s.Rate);
startBackground(s); 
pause(10)
stop(s);
s.IsContinuous = false;

The next is Auditive 2 function wich returns the f0l value

function Auditive2(frame, fs)
global f0l
% global frames

frame = frame - mean (frame );
windowframe=2*round(0.04*fs);        
[rxx,~] = xcorr(frame, frame);

% find peaks to measure the freq
rxx(rxx < 0) = 0; %eliminar la parte negativa
center_peak_width = find(rxx(windowframe:end) == 0 ,1); %find first zero after center
    
rxx(windowframe-center_peak_width : windowframe+center_peak_width  ) = min(rxx);%Eliminar el primer pico

[~, loc] = max(rxx); % encontrar el siguiente pico y su localizacion
period = abs(loc - length(frame)+1); %el periodo en muestras
clc;
f0=fs/period;
% convert pitch into MIDI numer asociated
f0=12*log2(f0/440)+69;
f0l =[f0l f0'];
% frames =[frames];
% voice =[voice frame']; 

Im trying to Run this code in MAC computer but its not posible.

Does anyone know how to process microphone signal in real time?

[Walter Roberson]

Note; for real-time audio work, do not use audiorecorder: use the Audio Processing Toolbox tools instead. See for example https://www.mathworks.com/matlabcentral/answers/569854-issue-with-trying-to-play-and-record-audio-simultaneously

Answer 1

[Walter Roberson]

https://www.mathworks.com/hardware-support/sound-card-daq.html

Sound card support in DAQ is available only on Windows. Also, directsound is Windows only.

The link I already posted shows using the Audio System Toolbox to read sound.

[Danny Lasso]

Thank you for answering, do you know if there is any toolbox able to acquire and process on realtime a microphone signal on MAC computers? I have been looking for it but without succes.

[Walter Roberson]

Demo.

The filter was put together pretty arbitrarily.

      % The following code was used to design the filter coefficients:
        % % Equiripple Bandpass filter designed using the FIRPM function.
        %
        % % All frequency values are in Hz.
        % Fs = 22100;  % Sampling Frequency
        %
        % Fstop1 = 3000;            % First Stopband Frequency
        % Fpass1 = 4000;            % First Passband Frequency
        % Fpass2 = 6000;            % Second Passband Frequency
        % Fstop2 = 7000;            % Second Stopband Frequency
        % Dstop1 = 0.31622776602;   % First Stopband Attenuation
        % Dpass  = 0.057501127785;  % Passband Ripple
        % Dstop2 = 0.0001;          % Second Stopband Attenuation
        % dens   = 20;              % Density Factor
        %
        % % Calculate the order from the parameters using FIRPMORD.
        % [N, Fo, Ao, W] = firpmord([Fstop1 Fpass1 Fpass2 Fstop2]/(Fs/2), [0 1 ...
        %                           0], [Dstop1 Dpass Dstop2]);
        %
        % % Calculate the coefficients using the FIRPM function.
        % b  = firpm(N, Fo, Ao, W, {dens});
        
        Hd = dsp.FIRFilter( ...
            'Numerator', [0.00237477459350913 0.0064614843733655 ...
            -0.00103059936849055 -0.030238372524698 -0.0585826389852291 ...
            -0.0532937837200882 -0.0240881134086325 -0.00228288514751351 ...
            0.0142103378085535 0.028903897155177 0.0118187515150954 ...
            -0.0288463509933308 -0.0321916763899556 -6.61268554359109e-05 ...
            0.00952714002962012 0.011524554854245 0.0367211697541932 ...
            0.0210377239851772 -0.0473798975515859 -0.0516403286130514 ...
            0.0104132441650136 0.0113350605525577 0.00111067355551197 ...
            0.0871711535781902 0.0841395148868027 -0.138981148118066 ...
            -0.225937282476684 0.0684005595135966 0.296849246242398 ...
            0.0684005595135966 -0.225937282476684 -0.138981148118066 ...
            0.0841395148868027 0.0871711535781902 0.00111067355551197 ...
            0.0113350605525577 0.0104132441650136 -0.0516403286130514 ...
            -0.0473798975515859 0.0210377239851772 0.0367211697541932 ...
            0.011524554854245 0.00952714002962012 -6.61268554359109e-05 ...
            -0.0321916763899556 -0.0288463509933308 0.0118187515150954 ...
            0.028903897155177 0.0142103378085535 -0.00228288514751351 ...
            -0.0240881134086325 -0.0532937837200882 -0.0585826389852291 ...
            -0.030238372524698 -0.00103059936849055 0.0064614843733655 ...
            0.00237477459350913]);


        
fs = 22100/4;
FrameSize = 512;
adr = audioDeviceReader(fs, FrameSize, 'NumChannels', 2);
adw = audioDeviceWriter(fs, 'SupportVariableSizeInput', true, 'BufferSize', FrameSize);
outbuf = zeros(FrameSize, 2);
while true
    inbuf = adr();
    outbuf = step(Hd, double(inbuf));
    adw(outbuf);
end