Audio signal processing apparatus and method thereof

A method of processing a waveform of a digital audio signal to be replayed, comprising steps of:

• expanding a frequency bandwidth of the digital audio signal through conversion of a sampling frequency at which the digital audio signal is sampled;
• performing low-pass filtering on the digital audio signal expanded in the frequency bandwidth, the low-pass filtering involving a cut-off frequency corresponding to the converted sampling frequency;
• detecting an interval of time between two adjacent waveform peaks of the low-pass-filtered digital audio signal, a polarity of a gradient of the waveform changing at each of the two adjacent waveform peaks and the interval of time being detected by measuring the number of times of sampling based on the converted sampling frequency;
• calculating data of a difference between current data of the low-pass-filtered digital audio signal and past data of the low-pass-filtered digital audio signal;
• weighting the data of the difference depending on the interval of time between the two adjacent waveform peaks;
• and
• producing output data based on both the low-pass-filtered digital audio signal and the weighted data of the difference.

A computer-readable program stored in a computer readable medium and executed by a computer for processing a waveform of an inputted digital audio signal to be replayed, the program allowing the computer to functionally realize steps of:

• expanding a frequency bandwidth of the digital audio signal through conversion of a sampling frequency at which the digital audio signal is sampled;
• performing low-pass filtering on the digital audio signal expanded in the frequency bandwidth, the low-pass filtering involving a cut-off frequency corresponding to the converted sampling frequency;
• detecting an interval of time between two adjacent waveform peaks of the low-pass-filtered digital audio signal, a polarity of a gradient of the waveform changing at each of the two adjacent waveform peaks and the interval of time being detected by measuring the number of times of sampling based on the converted sampling frequency;
• calculating data of a difference between current data of the low-pass-filtered digital audio signal and past data of the low-pass-filtered digital audio signal;
• weighting the data of the difference depending on the interval of time between the two adjacent waveform peaks;
• and
• producing output data based on both the low-pass-filtered digital audio signal and the weighted data of the difference.