Is there any Matlab built-in function for 4ASK and 8ASK modulation?

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Faheem Ur Rehman 2021-6-3

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https://ww2.mathworks.cn/matlabcentral/answers/846735-is-there-any-matlab-built-in-function-for-4ask-and-8ask-modulation

评论： Walter Roberson 2022-1-21

i have found below function but i don't know where to change for 4 and 8Ask.

format long;
% Clear all variables and close all figures
clear all;
close all;
% The number of bits to send - Frame Length
N = 8;
% Generate a random bit stream
bit_stream = round(rand(1,N));
% Enter the two Amplitudes
% Amplitude for 0 bit
A1 = 3; 
% Amplitude for 1 bit
A2 = 5;
% Frequency of Modulating Signal
f = 3;
% Sampling rate - This will define the resoultion
fs = 100;
% Time for one bit
t = 0: 1/fs : 1;
% This time variable is just for plot
time = [];
ASK_signal = [];
Digital_signal = [];
for ii = 1: 1: length(bit_stream)
    
    % The FSK Signal
    ASK_signal = [ASK_signal (bit_stream(ii)==0)*A1*sin(2*pi*f*t)+...
        (bit_stream(ii)==1)*A2*sin(2*pi*f*t)];
    
    % The Original Digital Signal
    Digital_signal = [Digital_signal (bit_stream(ii)==0)*...
        zeros(1,length(t)) + (bit_stream(ii)==1)*ones(1,length(t))];
    
    time = [time t];
    t =  t + 1;
   
end
% Plot the ASK Signal
subplot(2,1,1);
plot(time,ASK_signal,'LineWidth',2);
xlabel('Time (bit period)');
ylabel('Amplitude');
title('ASK Signal with two Amplitudes');
%axis([0 time(end) 1.5 1.5]);
grid  on;
% Plot the Original Digital Signal
subplot(2,1,2);
plot(time,Digital_signal,'r','LineWidth',2);
xlabel('Time (bit period)');
ylabel('Amplitude');
title('Original Digital Signal');
axis([0 time(end) -0.5 1.5]);
grid on;

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Answer 1

Walter Roberson 2021-6-10

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No, there is no built-in function for that. See however the File Exchange https://www.mathworks.com/matlabcentral/fileexchange/47375-ber-of-m-ask-modulation and use different M= values.

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Faheem Ur Rehman 2021-6-14

在 MATLAB Online 中打开

SNR = 0:1:22;   % Signal to Noise Ratio[dB]
data = audioread('audio_mix_441.wav', 'native');
binary_signal = de2bi(data(:, 1), 8);
fulllength = ceil(length(binary_signal)/spf) * spf;
padsignal = binary_signal;
padsignal(end+1:fulllength) = 0;
buffered_signal = reshape(padsignal, spf, []);
St   = buffered_signal;      % total Number of symbols
N = length(buffered_signal);
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
M = 16;         % Number of Symbols
k = log2(M);    % Number of bits per Symbol
alphabet = [-(M-1) : 2 : (M-1)];
Eavg = (1/M)*(sum(alphabet.^2));  % average power of transmitted signal
Eb_N0_dB = SNR;                   % signal to noise ratio
Eb_N0 = 10.^(Eb_N0_dB/10);
Es_N0 = Eb_N0*k;                  % symbols energy to noise ratio
% M-ASK transmitted signal
St = randsrc(1,N,alphabet);
St_norm = St/sqrt(Eavg);                % Normalization of transmitted signal power to one
% Channel Model and received signal
sigma2 = 1./(Es_N0);                % because signal energy is normalized to 1
N0 = sigma2;
for i = 1:length(SNR)
    n = sqrt(sigma2(i)/2)*(randn(1,length(St))+j*randn(1,length(St)));
    Sr_norm(i,:) = St_norm + n;
end

I have tried this code following your instruction but i got the following error.

Input must contain only finite real nonnegative integers.

Walter Roberson 2021-6-14

在 MATLAB Online 中打开

sps = 8; % Samples per symbol

spf = 1024; % Samples per frame

if mod(spf, sps) ~= 0

error('Samples per frame must be divisible by Samples per symbol');

end

symbolsPerFrame = spf / sps;

fs = 200e3; % Sample rate

data = audioread('audio_mix_441.wav', 'native');

udata = typecast(data(:,1), 'uint8');

binary_signal = reshape(de2bi(udata, 8).', 1, []);

full_length = ceil(length(binary_signal)/spf) * spf;

padsignal = binary_signal;

padsignal(end+1:full_length) = 0;

buffered_signal = reshape(padsignal, spf, []);

St = buffered_signal;

St_by_sample = reshape(St, sps, symbolsPerFrame, []);

symbol_index = reshape(sum(double(St_by_sample) .* 2.^((sps-1:-1:0).'),1), symbolsPerFrame, []) + 1;

%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%

k = sps; % Number of bits per Symbol

M = 2.^k;

alphabet = -(M-1) : 2 : (M-1);

Eavg = (1/M)*(sum(alphabet.^2)); % average power of transmitted signal

St_symbols = alphabet(symbol_index);

St_norm = St_symbols/sqrt(Eavg); % Normalization of transmitted signal power to one

t = (0:symbolsPerFrame-1)/fs;

plot(t, St_norm(:,1:3)); xlabel('time relative to frame'); ylabel('amplitude');

legend("frame " + (1:3))

I think the parts about SNR were just for modeling to see how robust it is.

I would recommend that you add error detection and correction bits to each frame. You would do either by making each frame longer (so 1024 samples plus the number of samples used for detection and correction), or else by making each frame shorter (so that after adding error codes, it came out as 1024 samples).

After the assignment to binary_signal then binary_signal is just a vector of bits. In this particular case it happens to be derived from int16 sound values, 16 bits per sound value, but one binary_signal exists it does not matter how it was created.

sps is not required to be 8. When it is 8, then the effect is that the bytes in udata are decomposed into bits and then later reconstructed into the same bytes (the symbol_index), but it would be valid to go down to (say) sps = 4, which would have the effect of making the transmission more robust.