How do I perform an FFT on a .wav file?

Question

I've been trying all day to figure out how to do this. I have been able to import the data without any problem and have been able to also plot just the data. However, I keep getting error messages whenever I am trying to do anything with the FFT and I don't know why nothing is happening. All I am trying to do is plot the Fourier Transform so that I can see which frequencies are playing at different amplitudes.

This is what I've done to get the file in.

signal = Import["C:\\Users\\Joshua\\Downloads\\Sound Samples\\INSTRUMENTS single \samples\\horns and brass\\bachtrumpet_d4.wav", "Data"];

I can plot it just fine

ListPlot[signal, Joined -> False, PlotRange -> All]

and then I can get the Fourier Transform without any problem (and I have tried multiple methods) This is the easy method

dft = Fourier[signal]

But then there's a problem with attempting to plot it

In[9]:= ListLinePlot[Part[dft, 1 ;; 20000], PlotRange -> Full]

During evaluation of In[9]:= Part::take: Cannot take positions 1 through 
20000 in {{-7.03996+0. I,0.255089 +0.109254 I,0.153395 +0.137607 I,0.101421 
+0.108852 I,<<44>>,0.00897083 -0.0093402 I,-0.0446355+0.0773417 I,
<<127950>>},{-1.57903+0. I,<<49>>,<<127950>>}}. >>

During evaluation of In[9]:= ListLinePlot::lpn: {{-7.03996+0. I,0.255089 
+0.109254 I,0.153395 +0.137607 I,0.101421 +0.108852 I,<<44>>,0.00897083 
-0.0093402 I,-0.0446355+0.0773417 I,<<127950>>},{-1.57903+0. I,<<49>>,
<<127950>>}}[[<<1>>]] is not a list of numbers or pairs of numbers. >>

During evaluation of In[9]:= ListLinePlot::lpn: {{-7.03996+0. I,0.255089 
+0.109254 I,0.153395 +0.137607 I,0.101421 +0.108852 I,<<44>>,0.00897083 
-0.0093402 I,-0.0446355+0.0773417 I,<<127950>>},{-1.57903+0. I,<<49>>,
<<127950>>}}[[<<1>>]] is not a list of numbers or pairs of numbers. >>

Out[9]= Out[9]

Then I tried this more complicated method that I've used for previous examples and was able to get good results with it.

n = 5000;
DiscreteFourier[data_] := 
Module[{ft = Abs[Fourier[data[[All, 2]]]]}, 
datanew = RotateRight[ft, Quotient[Length[ft], 2]]; 
freq[t_] := 
With[{tau = t[[2]] - t[[1]], n = Length[t]}, 
Table[-tau/2 + i/(n tau), {i, 0, n - 1}]]; 
Take[Transpose[{freq[data[[All, 1]]], 
 RotateRight[ft, Quotient[Length[data], 1]]}], 
Round[Length[ft]/2]]]

But then when I even attempt to just put the wav file into this program, I get an endless amount of error messages.

One thing I have also tried doing to no avail is shortening the amount of data being analyzed

localizedtimedata = Table[signal[[j]], {j, 10000, 10100}];

I can do the same thing with this data set as I can do with the other data set, meaning plot just the data, but not anything with the Transform unfortunately.

If anybody could help me with this I would be extremely grateful. I need to do this kind of analysis for a lot of files.

Answer 1

Assume you use Mathematica 11.

a = Audio[File["ExampleData/car.mp3"]]
Column[{AudioPlot[a, ImageSize -> Medium], Spectrogram[a, ImageSize -> Medium]}]

Verify this use Adobe Audition cc. First one is samples of Audio,and the second is the STFT of audio.

And Periodogram is useful when you want to look at some position in audio.

For example you want to see the fourier data in 3.8s.

You can use this code to generate a audio that center in 3.8s

(*length of data:1024*)
aNew = AudioTrim[a, {3.8 - 512/QuantityMagnitude@AudioSampleRate[a], 
                     3.8 + 512/QuantityMagnitude@AudioSampleRate[a]}]

And use this to perform a 1-dim fourier on this 1024-len data.

Periodogram[AudioData[aNew], 1024, 1024, BlackmanHarrisWindow, 
            SampleRate -> QuantityMagnitude@AudioSampleRate[aNew], 
            Ticks -> {Range[0, 22000, 1000], Range[0, -160, -20]}, TicksStyle -> Italic, 
            AspectRatio -> 1/5, AxesLabel -> {"Hz", "dB"}, 
            ImageSize -> 900, Background -> Black, 
            PlotStyle -> Green, AxesStyle -> White]

And you can see the two plot are exactly the same.