using the signal processing toolbox (if necessary), how can you correct for doppler shift in a .wav file? in matlab

To correct for Doppler shift in a .wav file using the signal processing toolbox in MATLAB, you can follow these steps:

1. Read the .wav file using the audioread function.
2. Use the fft function to perform a Fast Fourier Transform on the audio data to obtain the frequency spectrum.
3. Apply the necessary frequency shift to correct for the Doppler effect. This can be done by multiplying the Fourier Transform by a phase shift term that corresponds to the frequency shift.
4. Finally, use the ifft function to perform an Inverse Fourier Transform to obtain the corrected audio signal.

Here is an example code snippet to correct for Doppler shift in a .wav file:

main.m
% Read the .wav file
[input_audio, Fs] = audioread('input_audio.wav');

% Perform FFT
input_audio_fft = fft(input_audio);

% Define the Doppler shift frequency (change this value as needed)
doppler_shift = 1000; % Hz

% Calculate phase shift
phase_shift = exp(1i*2*pi*doppler_shift/Fs*(0:length(input_audio)-1));

% Apply frequency shift
corrected_fft = input_audio_fft .* phase_shift;

% Perform IFFT
corrected_audio = ifft(corrected_fft);

% Play the corrected audio
sound(corrected_audio, Fs);
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You can adjust the doppler_shift value according to your specific requirements.