Why do I get additional frequencies using standard audiowrite BitsPerSample values?

Hello,

I want to write spectral data to wav file and read it again.

If I do some example with two sine waves, after writing+reading a wav file I get more frequencies in spectrum (aliasing/quantization noise?).

% generate sin-waves
wave = dsp.SineWave(Frequency=2.45e6,SampleRate=5e6);
wave.SamplesPerFrame = 1024;
wave2 = dsp.SineWave(Frequency=2.46e6,SampleRate=5e6);
wave2.SamplesPerFrame = 1024;

% save and visualize
scope1 = spectrumAnalyzer(SampleRate=wave.SampleRate,...
ViewType="spectrum-and-spectrogram",...
PlotAsTwoSidedSpectrum=false);
%data = [];
y = [];
for ii = 1:250
    x = wave() + 0.5*wave2()+ 0.05*randn(1024,1);
    scope1(x);
    pause(0.01)
    if scope1.isNewDataReady
        y = [y,x];
        %data = [data;getSpectrumData(scope)];
    end
end

% set
Fs = 48000;

% write wav
audiowrite(wavefile,y,Fs,'BitsPerSample',16);

% visualize output orig
scope2 = spectrumAnalyzer(SampleRate=wave.SampleRate,...
ViewType="spectrum-and-spectrogram",...
PlotAsTwoSidedSpectrum=false);
for a = 1:width(y)
    scope2(y(:,a))
    pause(0.01)
end

% read wav
[b,Fs] = audioread('test.wav');
info = audioinfo('test.wav'); 

% visualize output wav
scope3 = spectrumAnalyzer(SampleRate=wave.SampleRate,...
ViewType="spectrum-and-spectrogram",...
PlotAsTwoSidedSpectrum=false);
for a = 1:width(b)
    scope3(b(:,a))
    pause(0.01)
end

when I change BitsPerSample parameter to 32/64:

% write wav
audiowrite(wavefile,y,Fs,'BitsPerSample',64);

I get a right values/signal form.

Why it is happends? It is aliasing/quantization noise?

Answer 1

[Mathieu NOE]

hello

if you export data to wav format , you have to make sure that the amplitude do not exceed +/- 1 otherwise the signal will be clipped and there you have a distorted signal (and that's why you get all the spurious frequencies)

seems to me this condition is not met as I can see the FFT peaks are well above 0 dB => clipping !

x = wave() + 0.5*wave2()+ 0.05*randn(1024,1); 
% and normalize !!
x = x./max(abs(x));

[Mathieu NOE]

hello again

problem solved ?

[Nik Rocky]

Hello Mathieu NOE,

thank you so much and sorry for late answer. Yes, its solves my problem!

If I inspect my x variable, thereare lot of values under -1:

-1.19660966653338
1.32569250089870
-1.29899512583688
1.30270363867150
-1.31475515734044
1.23504577190159
-1.35537389624305
1.21100708613462
-1.05105824664192
1.16503633855557
-0.848889696268412
...

and in the description of audiowrite funktion we can read:

Data beyond the valid range is clipped.
If y is single or double, then audio data in y should be normalized to values in the range −1.0 and 1.0, inclusive.

So, the normalizing solves this issue. Thank you very much!

Best regards,

Nik

[Mathieu NOE]

hello Nik

ok - good to see it works now !

all the best