Running into error when using pwelch for FFT

I am trying to conduct a FFT on a wav file in order to then calculate SPL within a bandwidth (120-450hz). I am using code from another answered question and I get hung up on an error:

Error using signal.internal.spectral.welchparse>segment_info
The length of the segments cannot be greater than the length of the input signal.

Error in signal.internal.spectral.welchparse (line 34)
[L,noverlap,win] = segment_info(M,win1,noverlap1);

Error in welch (line 55)
    signal.internal.spectral.welchparse(x,esttype,args{:});

Error in pwelch (line 170)
[welchOut{1:nargout}] = welch(x,funcName,inputArgs{:});

Error in snip_extractor (line 16)
[sensor_spectrum, freq] = pwelch(samples,w,NOVERLAP,NFFT,Fs);

My code follows. I am loading a 30sec wav file and then selecting a 5sec period with the file for analysis. Any help here would be most appreciated:

[y,Fs]=audioread('67649542.060120201700.wav') %Load fullfile

snips = y([163840:327680], [1]) %Extract data from period of interest

%%%Borrowed code from https://www.mathworks.com/matlabcentral/answers/636155-plotting-fft-for-audio-wav-file
NFFT = (Fs/2);    
NOVERLAP = round(0.75*NFFT);
w = hanning(NFFT);


samples = length(snips)
dt = 1/Fs
t = (0:dt:(samples-1)*dt)

[sensor_spectrum, freq] = pwelch(samples,w,NOVERLAP,NFFT,Fs);

sensor_spectrum_dB = 20*log10(sensor_spectrum);

figure(1),semilogx(freq,sensor_spectrum_dB);grid
title(['Averaged FFT Spectrum  / Fs = ' num2str(Fs) ' Hz / Delta f = ' num2str(freq(2)-freq(1)) ' Hz ']);
xlabel('Frequency (Hz)');ylabel(my_ylabel);

[Jonas]

maybe it would also be easier to use a bandpass filter or a combined low- and highpass on your time signal plus an splMeter object without additional frequency weighting (Z-weighting).

[Benjamin Colbert]

Would it? I'm very new at coding this. I have previously used prebuilt tools (e.g., PAMGuide) but I have a need now to write something specific for my code. Eventually I will run this to extract RMS dB re 1uPa between 120hz and 450hz for 30sec samples of about 300 wav files. I need to do some type of FFT in order to extract the sequency component of the signal.

[Jonas]

is there a reason for re 1*10^-6 Pa and not 20*10^-6 Pa?

do you want to calculate one SPL value for one file in your frequency range or do you want to extract multiple SPL values over time for each file?

[George]

Referencing to 1*10^-6 Pa is used for underwater signals.

That is SPL re 1 micro Pascal @ 1m @ 1W

Whereas yoru second reference, that is, 20 micro Pascals is used for referencing signal thresholds in air where it is generally acknowldeged that a normal average listener's threshold of hearing is 20 micro Pascals.

Answer 1

[Jonas]

[

sensor_spectrum, freq] = pwelch(samples,w,NOVERLAP,NFFT,Fs);

should be

[sensor_spectrum, freq] = pwelch(snips,w,NOVERLAP,NFFT,Fs);

since samples is only a scalar and not your data vector

the displayed error says, that your window of length fs/2 is lager than the given data vector, which is 1x1 by mistake