Audio signal processing device, audio signal processing method, and audio signal processing program

An audio signal processing method of separating a specific sound source from an audio signal in which a plurality of sound sources are mixed and extracting or eliminating the specific sound source, the audio signal processing method comprising:

• a first step of performing a short-time fast Fourier transform on an input audio signal;
• a second step of applying a low pass filter to a signal in a frequency domain generated at the first step to smooth the signal in a frequency domain;
• a third step of determining a sharpness of a waveform of a peak portion included in a waveform of the signal in a frequency domain on a basis of an output difference between the signal in a frequency domain and a signal output at the second step;
• a fourth step of determining whether the waveform of the peak portion is a steady sound on a basis of a result of determination at the third step;
• a fifth step of dynamically calculating a filter coefficient for comb filtering on a basis of a result of determination at the fourth step;
• a sixth step of filtering the signal in a frequency domain generated at the first step using the filter coefficient calculated at the fifth step;
• and
• a seventh step of transforming an output of filtering at the sixth step into a signal in a time domain and outputting the signal in a time domain, wherein
• the second step includes, when applying the low pass filter, adjusting the filter coefficient such that the higher a frequency band is, the smoother the waveform of the signal is.

An audio signal processing method of separating a specific sound source from an audio signal in which a plurality of sound sources are mixed and extracting or eliminating the specific sound source, the audio signal processing method comprising:

• a first step of performing a short-time fast Fourier transform on an input audio signal;
• a second step of evaluating, for a waveform of a peak portion included in a waveform of a signal in a frequency domain, an amplitude drop rate that is a ratio of a drop amount from a peak value of the peak portion in a preset frequency width to the frequency width;
• a third step of determining, on a basis of a result of evaluation at the second step, whether the waveform of the peak portion is a steady sound;
• a fourth step of dynamically calculating a filter coefficient for comb filtering on a basis of a result of determination at the third step;
• a fifth step of filtering the signal in a frequency domain generated at the first step using the filter coefficient calculated at the fourth step;
• and
• a sixth step of transforming an output of filtering at the fifth step into a signal in a time domain and outputting the signal in a time domain, wherein
• the second step includes, when evaluating the amplitude drop rate, adjusting the filter coefficient such that the higher a frequency band is, the smaller an evaluated value of the amplitude drop.