changing sampling rate of an audio

How to read a MP3 or WAV file and reduce it's sampling rate and save the new audio into a new file

Answer 1

[Ameer Hamza]

Something like this

[y, Fs] = audioread('audio_in.wav');
audiowrite('audio_out.wav', y, round(Fs/2));

This divides the sampling frequency by 2.

[Sebastien Dupertuis]

In the documentation, it is written that: audiowrite supports only samples rates of 44100 and 48000.

So doing:

audiowrite('audio_out.wav', y, round(Fs/2));

will not be accepted by audiowrite (except if the result of round(Fs/2) is 44100 and 48000).

Answer 2

[Star Strider]

Import the file, then use the Signal Processing Toolbox resample function to resample it to a different sampling frequency.