A signal processing technique the noise suppressing performance of which is more improved than conventional one is provided. A first component extraction unit 14 extracts a non-stationary component ^φ.sub.S.sup.(A)(ω, τ) derived from a sound coming from a target area and a stationary component ^φ.sub.S.sup.(B)(ω, τ) derived from an incoherent noise from a power spectrum density ^φ.sub.S(ω, τ) of the target area through processing of time average. A second component extraction unit 15 extracts a non-stationary component ^φ.sub.N.sup.(A)(ω, τ) derived from an interference noise and a stationary component ^φ.sub.N.sup.(B)(ω, τ) derived from an incoherent noise from a power spectrum density ^φ.sub.N(ω, τ) of a noise area.

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

A filtering method includes: receiving a first audio signal and a second audio signal that include sound emitted from a same sound source at different volumes; generating a filter signal by convoluting adaptive filter coefficients into the second audio signal; removing components of the filter signal from the first audio signal; and limiting a gain of the adaptive filter coefficients to 1.0 or less.

A filtering method includes: receiving a first audio signal and a second audio signal that include sound emitted from a same sound source at different volumes; generating a filter signal by convoluting adaptive filter coefficients into the second audio signal; removing components of the filter signal from the first audio signal; and limiting a gain of the adaptive filter coefficients to 1.0 or less.

Disclosed herein are method, system, and computer program product embodiments for performing the continuous tuning of received audio input from an earpiece or microphone especially customized for karaoke singing, wherein the audio input may be mixed with user-selected song input, and the joint mixed input is independently altered in the frequency domain for output to an earpiece worn by a user as well as separately for an additional audio output to an external connected speaker, for an optimal karaoke experience.

A media system and a method of using the media system to accommodate hearing loss of a user, are described. The method includes selecting a personal level-and-frequency dependent audio filter that corresponds to a hearing loss profile of the user. The personal level-and-frequency dependent audio filter can be one of several level-and-frequency-dependent audio filters having respective average gain levels and respective gain contours. An accommodative audio output signal can be generated by applying the personal level-and-frequency dependent audio filter to an audio input signal to enhance the audio input signal based on an input level and an input frequency of the audio input signal. The audio output signal can be played by an audio output device to deliver speech or music that the user perceives clearly, despite the hearing loss of the user. Other aspects are also described and claimed.

A voice processing device includes a noise-originating coefficient calculation section that calculates a noise-originating coefficient that gradually decreases as a target value of stationary noise for each frequency increases, the target value being calculated based on an amplitude value of a frequency spectrum obtained by time-frequency transforming a voice signal for a predetermined period of time, and a suppression signal generation section that generates, when the frequency spectrum is determined as being stationary on the basis of the amplitude value, a suppression signal by multiplying a suppression coefficient based on the noise-originating coefficient by the amplitude value, the suppression signal being frequency-time transformed to be output.

An audio transmitter processor for generating an error protected frame using encoded audio data of an audio frame, the encoded audio data for the audio frame having a first amount of information units and a second amount of information units, has: a frame builder for building a codeword frame having a codeword raster, wherein the frame builder is configured to determine a border between a first amount of information units and a second amount of information units so that a starting information unit of the second amount of information units coincides with a codeword border; and an error protection coder to obtain a plurality of processed codewords representing the error protected frame.

A device includes one or more processors configured to obtain audio signals representing sound captured by at least three microphones and determine spatial audio data based on the audio signals. The one or more processors are further configured to determine a metric indicative of wind noise in the audio signals. The metric is based on a comparison of a first value and a second value. The first value corresponds to an aggregate signal based on the spatial audio data, and the second value corresponds to a differential signal based on the spatial audio data.