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?
Kshitij Singh answered .
2025-11-20
% 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