Provided is pitch enhancement processing having little unnaturalness even in time segments for consonants, and having little unnaturalness to listeners caused by discontinuities even when time segments for consonants and other time segments switch frequently. A pitch emphasis apparatus carries out the following as the pitch enhancement processing: for a time segment in which a spectral envelope of a signal has been determined to be flat, obtaining an output signal for each of times in the time segment, the output signal being a signal including a signal obtained by adding (1) a signal obtained by multiplying the signal of a time, further in the past than the time by a number of samples T.sub.0 corresponding to a pitch period of the time segment, a pitch gain σ.sub.0 of the time segment, a predetermined constant B.sub.0, and a value greater than 0 and less than 1, to (2) the signal of the time.

A method of equalising an audio signal derived from a microphone, the method comprising: receiving the audio signal; applying an order-statistic filter to the audio signal in the frequency domain to generate a statistically filtered audio signal; equalising the received audio signal based on the statistically filtered audio signal to generate an equalised audio signal.

A system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of electronic device audible cues. Audible cues are stored as sound files. Ambient noise is detected, and the output of the audible cues is altered based on the ambient noise. Various embodiments include processed sound files that are more robust in noisy environments.

This application discloses an audio signal processing method, a model training method, and related apparatus. The audio signal processing method includes inputting a first audio input signal into a machine learning model to obtain a first howling point, and obtaining a first gain value according to the first howling point, the first howling point corresponding to an effective audio signal in the first audio input signal, and the first gain value indicating a suppression parameter of the first howling point; processing the first audio input signal according to the first gain value to obtain a second audio input signal; detecting a second howling point, and obtaining a second gain value according to the second howling point, the second howling point corresponding to a frequency band of an ineffective audio signal in the second audio input signal; and processing the second audio input signal to obtain an audio output signal.

This application discloses an audio signal processing method, a model training method, and related apparatus. The audio signal processing method includes inputting a first audio input signal into a machine learning model to obtain a first howling point, and obtaining a first gain value according to the first howling point, the first howling point corresponding to an effective audio signal in the first audio input signal, and the first gain value indicating a suppression parameter of the first howling point; processing the first audio input signal according to the first gain value to obtain a second audio input signal; detecting a second howling point, and obtaining a second gain value according to the second howling point, the second howling point corresponding to a frequency band of an ineffective audio signal in the second audio input signal; and processing the second audio input signal to obtain an audio output signal.

A method, computer program product, and computing system for receiving one or more inputs indicative of at least one of: a relative location of a speaker and a microphone array, and a relative orientation of the speaker and the microphone array. One or more reference signals may be received. A speech processing system may be trained using the one or more inputs and the one or more reference signals.

A method, computer program product, and computing system for receiving one or more inputs indicative of at least one of: a relative location of a speaker and a microphone array, and a relative orientation of the speaker and the microphone array. One or more reference signals may be received. A speech processing system may be trained using the one or more inputs and the one or more reference signals.

A mixing device of a first signal and a second signal on a time-frequency plane, includes a control signal generation unit to generate a control signal indicating whether to perform prioritized mixing that includes amplification of the first signal and attenuation of the second signal; and a gain derivation unit to derive a first gain for amplifying the first signal and a second gain for attenuating the second signal based on the control signal. The control signal takes at least a first value and a second value different from the first value, wherein the first value is not continuous beyond a predetermined bandwidth on a frequency axis. The mixing device applies the prioritized mixing to the first and second signals when the control signal indicates the first value, and applies simple addition to the first and second signals when the control signal indicates the second value.

A mixing device of a first signal and a second signal on a time-frequency plane, includes a control signal generation unit to generate a control signal indicating whether to perform prioritized mixing that includes amplification of the first signal and attenuation of the second signal; and a gain derivation unit to derive a first gain for amplifying the first signal and a second gain for attenuating the second signal based on the control signal. The control signal takes at least a first value and a second value different from the first value, wherein the first value is not continuous beyond a predetermined bandwidth on a frequency axis. The mixing device applies the prioritized mixing to the first and second signals when the control signal indicates the first value, and applies simple addition to the first and second signals when the control signal indicates the second value.

A method of equalising an audio signal derived from a microphone, the method comprising: receiving the audio signal; applying an order-statistic filter to the audio signal in the frequency domain to generate a statistically filtered audio signal; equalising the received audio signal based on the statistically filtered audio signal to generate an equalised audio signal.