How can I add a low pass filter after each delay line in my multi channel feedback delay network

Im about to program this structure in Matlab in a for loop

So far so good. And I have already done the for loop. I also implemented a Hadamard Mixing Matrix. Here is the code:

in = [ 1; 0 ]; % Dirac Impulse
Fs = 44100;
in = [in; zeros(3*Fs,1)]; % Space for reverb
% Define delay parameters
       %Hadamard Matrix as feedback matrix
H = 0.75*hadamard(16);




feedbackGain =[0.1,-0.15,-0.2,0.25,-0.3,0.35,0.4,-0.45,-0.5,0.55,0.6,-0.65,-0.7,-0.75,-0.8,0.85]; %0.1,0.15,0.2,0.25,0.3,0.35,0.4,0.45,0.5,0.55,0.6,0.65,0.7,0.75,0.8,0.85
wetGain = [0.7,0.7,0.7,0.7,0.7,0.7,0.7,0.7,0.7,0.7,0.7,0.7,0.7,0.7,0.7,0.7]; % adjust to control wet signal level
% Convert delay time to samples
delaySamples = [101,233, 439,613, 853,1163, 1453,1667 , 1801,2053, 2269,2647, 3001,3607, 4153,4591]; %0.02s, 0.03, 0.04, 0.05 881, 1009, 1321,1511, 1777, 1999, 2203, 2351
                                                               %101, 439, 853, 1453 , 1801, 2269, 3001, 4153

%Summe delays = 30644


% delaySamples = [1200, 1800, 2200,2500];
% Create an empty array to store delayed audio
delayedAudio = zeros(size(in));
delayedAudio2 = zeros(size(in));
delayedAudio3 = zeros(size(in));
delayedAudio4 = zeros(size(in));
delayedAudio5 = zeros(size(in));
delayedAudio6 = zeros(size(in));
delayedAudio7 = zeros(size(in));
delayedAudio8 = zeros(size(in));
delayedAudio9 = zeros(size(in));
delayedAudio10 = zeros(size(in));
delayedAudio11 = zeros(size(in));
delayedAudio12 = zeros(size(in));
delayedAudio13 = zeros(size(in));
delayedAudio14 = zeros(size(in));
delayedAudio15 = zeros(size(in));
delayedAudio16 = zeros(size(in));




delayedAudios =[delayedAudio,delayedAudio2,delayedAudio3,delayedAudio4,delayedAudio5,delayedAudio6,delayedAudio7,delayedAudio8,delayedAudio9,delayedAudio10,delayedAudio11,delayedAudio12,delayedAudio13,delayedAudio14,delayedAudio15,delayedAudio16];
delayedAudios(1,:) = 1;


% Apply delay effect

for i = 1:length(in)-2*max(delaySamples)
     for k = 1:16
        if delayedAudios(i+delaySamples(k),k) == 0
        delayedAudios(i+delaySamples(k),k) = feedbackGain(k)*delayedAudios(i,k);
        end
        
     end      
 
     for k = 1:16
        for j = 1:16
           if delayedAudios(i+delaySamples(j)+delaySamples(k),k) == 0 && k~= j 
            delayedAudios(i+delaySamples(j)+delaySamples(k),k) = H(j,k)*delayedAudios(i+delaySamples(j),j);
           end
        end
     end     
end


%Original and Mix together

for k = 1:16
    delayedAudios(:,k) = wetGain(k)*delayedAudios(:,k);
end

%outputAudioM = delayedAudios(:,16) + delayedAudios(:,15) + delayedAudios(:,14) + delayedAudios(:,13) + delayedAudios(:,12) +delayedAudios(:,11) +delayedAudios(:,10) +delayedAudios(:,9)+delayedAudios(:,8) +delayedAudios(:,7) +delayedAudios(:,6) +delayedAudios(:,5) +delayedAudios(:,4) +delayedAudios(:,3) +delayedAudios(:,2 )+delayedAudios(:,1);
outputAudioM = sum(delayedAudios,2);
outputAudioM(1) = outputAudioM(1)*0.02;

%Low pass filtering + phase adjusting
d = designfilt('lowpassfir','FilterOrder',4 ,'CutoffFrequency',2000,'SampleRate', Fs);
y = filtfilt(d,outputAudioM);


% %Write the output audio to a new file and plot
audiowrite('output_audio_with_delay.wav', y, Fs);
plot(y);

% Play the output audio
sound(y,Fs);

As you can see in the code I implemented a lpf AFTER the sound was computed. That of course is no problem but I have no idea how I can do that in the for loop like in the picture I showed. Can someone explain how it is to be done or give me hints?

Answer 1

[jibrahim]

Hi Muhsin,

In general, in order to perform filtering and listen to the output inside your for loop, you use filter System objects to perform filtering, and audioDeviceWriter to play the output.

The application you are working with is somewhat similar to a reverberation object that ships with Audio Toolbox: audioexample.FreeverbReverberator. That objects internally uses a bank of allpass filters implemented using dsp.IIRFilter objects.

Here is a simple sample code of how to use the object to filter input data and listen ot the output in the loop.

% Process samples from an audio file
afr = dsp.AudioFileReader("FunkyDrums-44p1-stereo-25secs.mp3",PlayCount=Inf);
% The filter
r =  audioexample.FreeverbReverberator;
% Play the result
adw = audioDeviceWriter(SampleRate=afr.SampleRate);

while ~isDone(afr)
    x = afr(); % read an input frame
    y = r(x);  % filter the frame. the filter retains states between consecutive calls
    adw(y);    % listen ot the output frame
end

[Muhsin Zerey]

Thanks for your answer and your help! But I need to filter each sample in the loop with a low pass filter. what exactly is afr() and how big is it in your example? I still dont understand how I can use your code as an example for my problem.

Really appreciate your help tho

[jibrahim]

Hi Muhsin,

afr is an object that reads an audio file one frame at a time. The length of the frame is determined by the SamplesPerFrame property. You can set it to one for sample-by-sample processing. You can also rearrange your code to process your samples one by one.

Here is another simple example where you process sample-by-sample:

afr = dsp.AudioFileReader("FunkyDrums-44p1-stereo-25secs.mp3",...
                          PlayCount=Inf,...
                          SamplesPerFrame=512);

% To listen to audio
adw = audioDeviceWriter(SampleRate=afr.SampleRate);

% Define a second-order section filter.
% Replace this by your own processing eventually
filt = dsp.SOSFilter(Numerator=[1 0 0],Denominator=[1 0 0]);

% You process sample-by-sample, and you listen to each output frame

y = zeros(afr.SamplesPerFrame,1);
while ~isDone(afr)
    x = afr(); % read an input frame
    % process the frame sample-by-sample if you want
    for index=1:size(x,1)
        y(index) = filt(x(index));  
    end
    adw(y);    % listen ot the output frame
end

[Muhsin Zerey]

Thanks for the answer. In my case I have to filter at the same time while I am doing a feedback loop. Also I dont have a finished mp3 file, in my case its a WAV File. When I filter by a low pass filter that value needs to get fed back. Does your code support this?

Sorry for all these questions. Im fairly new to this and there are some things I dont quite understand.

[jibrahim]

You are free to do whatever you want in the loop, including feedback. In case you want to read from a wav file instead of an mp3 file, dsp.AudioFileReader supports that.

These examples might be a good introduction to streaming dignal processing:

https://www.mathworks.com/help/dsp/ug/introduction-to-streaming-signal-processing-in-matlab.html

https://www.mathworks.com/help/audio/gs/audio-input-and-audio-output.html