Implementing the Chorus Effect

Question

Vincent Abraham 2020-11-18

1
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此问题的直接链接

https://ww2.mathworks.cn/matlabcentral/answers/652453-implementing-the-chorus-effect

评论： Mathieu NOE 2020-11-23

I have implemented a code for flanging. Flanging uses a delay of 0-10ms (in this case 0-3ms) to produce the desired output. Now, I want to implement this same code for creating a chorus effect, which uses a delay of 25-30 or 30-50ms. Please suggest what I should do

clear all;
close all;
infile='acoustic.wav';
outfile='out_chorus.wav';
% current sample is 11kHz so 0-3 ms is 0 - 33 samples
% read the sample waveform
[x,Fs] = audioread(infile);
% parameters to vary the effect %
min_time_delay=0.010; % 10ms min delay in seconds
max_time_delay=0.025; % 25ms max delay in seconds
rate=1; %rate of flange in Hz
index=1:length(x);
% sin reference to create oscillating delay
sin_ref = (sin(2*pi*index*(rate/Fs)))';    % sin(2pi*fa/fs);
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
min_samp_delay=round(min_time_delay*Fs); %convert delay in ms to min delay in samples
max_samp_delay=round(max_time_delay*Fs); %convert delay in ms to max delay in samples
y = zeros(length(x),1);       % create empty out vector
y(1:max_samp_delay)=x(1:max_samp_delay); % to avoid referencing of negative samples
amp=0.7; % suggested coefficient from page 71 DAFX
% for each sample
for i = (max_samp_delay+1):length(x)
    cur_sin=abs(sin_ref(i));    %abs of current sin val 0-1
    cur_delay=ceil(cur_sin*max_samp_delay);  % generate delay from 1-max_samp_delay and ensure whole number  
    y(i) = (amp*x(i)) + amp*(x(i-cur_delay));   % add delayed sample
end
% write output
audiowrite(outfile, y, Fs);
figure(1)
hold on
plot(x,'r');
plot(y,'b');
title('Flanger and original Signal');

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Mathieu NOE 2020-11-19

and you don't have Simulink ?

Vincent Abraham 2020-11-19

I do. But i need to implement this in code

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Answer 1

Mathieu NOE 2020-11-19

0
链接

此回答的直接链接

https://ww2.mathworks.cn/matlabcentral/answers/652453-implementing-the-chorus-effect#answer_549823

在 MATLAB Online 中打开

hello again

so this could be considered as a vectorized version of the original code. You can add multiple delayed versions of the original signal

just make sure the dimensions of rate and amp are consistent

hope it matches your musical expectations...

infile='DirectGuitar.wav';
outfile='out_chorus.wav';
% current sample is 11kHz so 0-3 ms is 0 - 33 samples
% read the sample waveform
[x,Fs] = audioread(infile);
% normalize x to +/- 1 amplitude
x = x ./ (max(abs(x)));
samples = length(x);
% parameters to vary the effect %
min_time_delay=0.010; % 10ms min delay in seconds
max_time_delay=0.025; % 25ms max delay in seconds
% nb_effects = 4; % assume we do 4 delayed versions of the signal that will be summed 
rate=[1 0.9 0.8 0.7] ; %rate of flange in Hz (as many as chorus effects)
nb_effects = length(rate); % assume we do 4 delayed versions of the signal that will be summed in the final section
% amplitude is now a vector
amp=[0.7 0.6 0.5 0.4 0.3];; % suggested coefficient from page 71 DAFX; 
% Nb the first value applies to x and the remaining values apply to the 4
% delayed versions of x) so make sure this vector has length = nb_effects  +1
index=(1:samples)';
% sin reference to create oscillating delay
sin_ref = (sin(2*pi*index*(rate/Fs)))';    % sin(2pi*fa/fs);  % make sure this is a matrix dimensins =  (nb_effects , samples)
% sin_ref = 0.5*(1+sin(2*pi*index*(rate/Fs)))';    % sin(2pi*fa/fs);  % make sure this is a matrix dimensins =  (nb_effects , samples)
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
min_samp_delay=round(min_time_delay*Fs); %convert delay in ms to min delay in samples
max_samp_delay=round(max_time_delay*Fs); %convert delay in ms to max delay in samples
y = zeros(samples,1);       % create empty out vector
y(1:max_samp_delay)=x(1:max_samp_delay); % to avoid referencing of negative samples
% amp=0.7; % suggested coefficient from page 71 DAFX
% for each sample
for i = (max_samp_delay+1):length(x)
    cur_sin=abs(sin_ref(:,i));    %abs of current sin val 0-1
    cur_delay=ceil(cur_sin*max_samp_delay);  % generate delay from 1-max_samp_delay and ensure whole number  
    % add the amp weigthed multiple delayed x values
    y(i) = amp(1)*x(i) + sum(amp(2:nb_effects+1)'.*x(i-cur_delay));   % add all 4 delayed sample (in one shot !)
end
% write output
% normalize y to +/- 1 amplitude
y = y ./ (max(abs(y)));
audiowrite(outfile, y, Fs);
figure(1)
hold on
plot(x,'r');
plot(y,'b');
title('Chorus and original Signal');

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Mathieu NOE 2020-11-19

在 MATLAB Online 中打开

so this is it

I must admit the listening of the effect is strange

Now you have plenty of parameters to play with :

nb of effects : 4 (now)
amplitudes of delayed signals : amp = [0.7 0.3 0.3 0.3 0.3];

initial randon signal amplitude (0.2) : rand_ref = 0.2*rand(nb_effects , samples);

you can apply a low pass filter effct on this "brute" random values to get a LP filtered random. This is driven by : b = 0.2; % coefficient for 1st order IIR low pass filter
you can see the effect of this LP filter on figure (2). at the beginning there is no filtering (no worry as it is not applyed yet on the signal), then when i = max_samp_delay+1 at the beginning of the for loop , we have the LP effect coming into play

infile='DirectGuitar.wav';
outfile='out_chorus.wav';
% current sample is 11kHz so 0-3 ms is 0 - 33 samples
% read the sample waveform
[x,Fs] = audioread(infile);
% normalize x to +/- 1 amplitude
x = x ./ (max(abs(x)));
samples = length(x);
% parameters to vary the effect %
min_time_delay=0.010; % 10ms min delay in seconds
max_time_delay=0.025; % 25ms max delay in seconds
nb_effects = 4; % assume we do 4 delayed versions of the signal that will be summed 
% amplitude is now a vector
amp = [0.7 0.3 0.3 0.3 0.3]; % suggested coefficient from page 71 DAFX; 
% Nb the first value applies to x and the remaining values apply to the 4
% delayed versions of x) so make sure this vector has length = nb_effects  +1
index=(1:samples)';
% rand reference to create delay
rand_ref = 0.2*rand(nb_effects , samples);    % rand returns a single uniformly distributed random number in the interval (0,1).
                                            % make sure this is a matrix dimensins =  (nb_effects , samples)
b = 0.2;     % coefficient for 1st order IIR low pass filtering or rand modulation amplitude
             % b = 1 => no LP filter, b = 0.01 very slow filtering 
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
rand_ref_filtered = rand_ref; % init
min_samp_delay=round(min_time_delay*Fs); %convert delay in ms to min delay in samples
max_samp_delay=round(max_time_delay*Fs); %convert delay in ms to max delay in samples
y = zeros(samples,1);       % create empty out vector
y(1:max_samp_delay)=x(1:max_samp_delay); % to avoid referencing of negative samples
% amp=0.7; % suggested coefficient from page 71 DAFX
% for each sample
for i = (max_samp_delay+1):samples
    
    rand_ref_filtered(:,i)  = (1-b)*rand_ref_filtered(:,i-1) + b*rand_ref(:,i);    % IIR low pass filtering of rand_ref
    cur_delay=ceil(rand_ref_filtered(:,i)*max_samp_delay);  % generate delay from 1-max_samp_delay and ensure whole number  
%     cur_delay=ceil(rand_ref(:,i)*max_samp_delay);  % generate delay from 1-max_samp_delay and ensure whole number  
    % add the amp weigthed multiple delayed x values
    y(i) = amp(1)*x(i) + sum(amp(2:nb_effects+1)'.*x(i-cur_delay));   % add all 4 delayed sample (in one shot !)
end
% write output
% normalize y to +/- 1 amplitude
y = y ./ (max(abs(y)));
audiowrite(outfile, y, Fs);
figure(1)
hold on
plot(x,'r');
plot(y,'b');
title('Chorus and original Signal');
figure(2),plot(1:1000,rand_ref_filtered(:,1:1000));grid
sound(y,Fs);

Mathieu NOE 2020-11-23

chorus.m

hello

yes , for the fixed dealy (4 effects) the sound is good

as soon as we make the delay variable, it's horrible... don't know what we are doing wrong here

Mathieu NOE 2020-11-23

so what do we do next ??

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Implementing the Chorus Effect

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