This operation in digital signal processing involves converting a sampled signal from one sampling frequency to another without changing the period of the sample as would occur if the sampled audio was played at the new rate directly. Clicking Resampling menu item the following form shows up. 

Enter the sampling rate you need to convert the current audio stream and click ok button. 

Sampling rate 

The sampling rate, sample rate, or sampling frequency (fs) defines the number of samples per unit of time (usually seconds) taken from a continuous signal to make a discrete signal. For time-domain signals, the unit for sampling rate is hertz, (samples per second). The reciprocal of the sampling frequency is the sampling period or sampling interval, which is the time between samples. 

Sampling theorem 

The Nyquist–Shannon sampling theorem states that perfect reconstruction of a signal is possible when the sampling frequency is greater than twice the maximum frequency of the signal being sampled, or equivalently, when the Nyquist frequency (half the sample rate) exceeds the highest frequency of the signal being sampled. If lower sampling rates are used, the original signal's information may not be completely recoverable from the sampled signal. For example, if a signal has an upper band limit of 100 Hz, a sampling frequency greater than 200 Hz will avoid aliasing and would theoretically allow perfect reconstruction. 

The full range of human hearing is between 20 Hz and 20 kHz. The minimum sampling rate that satisfies the sampling theorem for this full bandwidth is 40 kHz. The 44.1 kHz sampling rate used for Compact Disc was chosen for this and other technical reasons. 

Oversampling 

In some cases it is desirable to have a sampling frequency more than twice the desired system bandwidth so that a steep digital filter and a less steep analog anti-aliasing filter can be used in exchange for a steep analog anti-aliasing filter. The reason for wanting a less steep analog anti-aliasing filter is that the digital filter is not subject to any component variations thus always giving the filter response (filtering function) that the designer has chosen. This process is known as oversampling 

Undersampling 

Conversely, one may sample below the Nyquist rate. For a baseband signal (one that has components from 0 to the band limit), this introduces aliasing, but for a passband signal (one that does not have low frequency components), there are no low frequency signals for the aliases of high frequency signals to collide with, and thus one can sample a high frequency (but narrow bandwidth) signal at a much lower sample rate than the Nyquist 

A list of common audio sample rates