Abstract: A sound reproducing device has a loudspeaker arranged to produce sound from an audio signal provided by an audio signal source. A microphone is positioned to pick up ambient noise and generate a microphone signal which comprises the noise. An ambient noise cancellation (ANC) system receives the microphone signal from the microphone and generates anti-noise corresponding to the ambient noise in the microphone signal. An automatic polarity adaptation (AAP) system monitors the ANC system and, when a decision criterion is fulfilled, causes a switch in polarity for the generated anti-noise.

Abstract: A volume control system comprises: an amplifying apparatus; and a controller connectable to the amplifying apparatus. The amplifying apparatus includes; a first receiving section for receiving a volume value from the controller, a first input section into which an instruction for changing the volume value is input by a user's operation, a first transmitting section for transmitting the volume value input into the first input section to the controller, and a volume control section for, when the first receiving section receives the volume value or the changed volume value is input into the first input section, controlling a volume value of the amplifying apparatus based on the volume value.

Abstract: There is provided an apparatus for enabling karaoke. The apparatus includes a casing for the apparatus; a microphone array incorporated within the casing, the microphone array being concealed from a user by the casing; a controller coupled to the microphone array to at least process audio signals from the user input at the microphone array; a data storage device coupled to the controller, the data storage device being for storing pre-recorded songs used for karaoke, and for storing songs as sung by the user; and an image capturing device coupled to the controller, the image capturing device being for capturing images of the user while the user is singing, with the captured images of the user being stored on the data storage device.

Abstract: A device for generating an encoded stereo signal from a multi-channel representation includes a multi-channel decoder generating three of more multi-channels from at least one basic channel and parametric information. The three or more multi-channels are subjected to headphone signal processing to generate an uncoded first stereo channel and an uncoded second stereo channel which are then supplied to a stereo encoder to generate an encoded stereo file on the output side. The encoded stereo file may be supplied to any suitable player in the form of a CD player or a hardware player such that a user of the player does not only get a normal stereo impression but a multi-channel impression.

Abstract: A method of enhancing an audio signal includes the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-

Abstract: Methods and techniques to implement a digital signal processor for avoidance of audio feedback are disclosed, in particular, audio signal processing systems that reduce the requirement for physical segregation of sound acquisition and diffusion zones. In a more general aspect, the components and techniques described herein provide a for a sound space and sound processing equipment such that sound travelling electronically in a loop through the sound processing equipment that is output into a physical sound diffusion zone, received at the input to the sound processing equipment, and then re-amplified, etc. is attenuated over that loop by frequency modification. The frequency modification is such that, at least for some signals, on each pass through the loop, the sound processing equipment will attenuate or amplify individual sub-bands of the frequency spectrum of the audio signal that is received at the input of the sound processing equipment.

Abstract: A method for simulating spatially extended sound sources comprising: panning a first input signal over a plurality of output channels to generate a first multi-channel directionally encoded signal; panning a second input signal over the plurality of output channels to generate a second multi-channel directionally encoded signal; combining the first and second multi-channel directionally encoded signals to generate a plurality of loudspeaker output channels; and applying a bank of decorrelation filters on the loudspeaker output channels.

Abstract: An audio signal compensation device includes: a signal processor configured to perform filtering on an input audio signal; a filter coefficients storage module configured to store a plurality of filter coefficients; a user interface configured to provide options for a determination of filter coefficients to a user and to obtain a selection result from the user; and a filter coefficients determining module configured to determine a set of filter coefficients among the plurality of filter coefficients based on the selection result.

Abstract: A howling suppression device that can reduce quality deterioration of processed sound includes: a delay unit delaying the input signal to output the delayed input signal as a reference signal; a signal separation unit including an adaptive filter extracting a periodic signal component from the reference signal by adaptively updating a filter coefficient; a howling detection unit detecting an occurrence of howling using at least a signal of the periodic signal component output from the adaptive filter; and a howling suppression unit. The howling suppression unit includes: a suppression filter obtaining the updated filter coefficient from the adaptive filter with timing when the howling detection unit detects the occurrence of the howling, to extract the periodic signal component from the reference signal based on the filter coefficient; and a subtractor subtracting the periodic signal component from the input signal so as to output a signal obtained by the subtraction.

Abstract: Method, user terminal and computer program product for controlling audio signals at the user device during a communication session between the user device and a remote node, in which a primary audio signal is received at audio input means of the user device for transmission to the remote node in the communication session. It is determined whether the user device is operating in (i) a first mode in which secondary audio signals output from the user device are likely to disturb the primary audio signal received at the audio input means, or (ii) a second mode in which secondary audio signals output from the user device are not likely to disturb the primary audio signal received at the audio input means.

Abstract: An audio information processing apparatus for processing input audio information is adapted to input an audio signal composed of front area audio information that should be input to a front speaker and rear area audio information that should be input to a plurality of speakers. A mixer is adapted to mix the rear area audio information in accordance with an instruction from an instruction circuit.

Abstract: A sound processor includes a conversion unit converts a reference sound signal corresponding to a base of sound to be output and an observation sound signal based on each of sound signals output by a plurality of sound receiving units into frequency components, an echo suppression unit estimates echo derived from sound based on a converted reference sound signal and suppressing the estimated echo in a converted observation sound signal, a noise suppression unit estimates noise based on an arrival direction of sound and suppressing the estimated noise in the converted observation sound signal and an integrating process unit suppresses, with respect to each frequency component, echo and noise in the converted sound signal based on a observation sound signal obtained after echo suppression and a observation sound signal obtained after noise suppression.

Abstract: The present invention discloses a circuit with three-stage of power-on sequence used for suppressing the pop noise in audio system. It mainly comprises a first resistor (R1); a capacitor (Cout); a first switch (SW1); a second switch (SW2); a soft start device; a first feedback amplifier; and a second feedback amplifier. By using the three-stage of power-on sequence, the present invention can effectively suppress the pop noise when the audio driver is power on.

Abstract: The invention provides a noise removal device and method capable of more proper interpolation on an input signal. The noise removal device is for removing noise in an input signal and includes: a noise detector detecting noise in an IF signal and outputting a noise detection signal; an interpolation controller determining a period and amount of interpolation for noise correction processing, based on the IF signal and the noise detection signal; and a noise gate processor performing the noise correction processing on the IF signal, based on the interpolation period and amount supplied from the interpolation controller. The interpolation controller sets a predetermined first interpolation period, based on a first noise detection signal inputted from the noise detector, and redefines a second interpolation period longer than the first interpolation period when a second noise detection signal is detected within the first interpolation period.

Abstract: A spatial audio system for implementing a head-related transfer function (HRTF). A first stage implements a lateral HRTF that reproduces the median frequency response for a sound source located at a particular lateral distance from a listener, and second stage implements a vertical HRTF that reproduces the spectral changes when the vertical distance of a sound source changes relative to the listener. The system improves the vertical localization accuracy provided by an arbitrary measured HRTF by introducing an enhancement factor into the second processing stage. The enhancement factor increases the spectral differentiation between simulated sound sources located at different positions within the same “cone of confusion.

Abstract: A signal processing apparatus includes: clip detector means that detects presence/absence of a clipped part with a deformed waveform in each of N audio signals output from N microphones (where N is an integer equal to or greater than 2) based on a dynamic range of a circuit; and interpolation means that treats an audio signal in the N audio signals which has the clipped part detected by the clip detector means as an interpolation target, and other audio signals as non-interpolation targets, and interpolates the waveform of the clipped part of the interpolation target using the waveform of at least one audio signal in the non-interpolation targets.

Abstract: Implementations related to echo cancellation are depicted and described herein.

Abstract: A sound processing apparatus according to the present invention acquires a test signal for measuring a standing wave state emitted in a listening room, and determines a peak position or a dip position due to a standing wave based on frequency characteristics of the test signal. Next, the sound processing apparatus emits a burst signal corresponding to the frequency of the peak position or the dip position, and acquires this signal. The sound processing apparatus calculates an increment ?P of the acquired signal, which indicates an amount of increase of a peak in the trailing edge portion corresponding to the end position of the burst signal relative to a peak in the portion corresponding to the stationary portion of the burst signal, and attenuates the frequency of the above peak position or dip position of a sound signal to be output by an attenuation depending on ?P.

Abstract: Presented herein is a system for transmitting sounds to an in-utero infant, that is worn by a pregnant woman, said system comprising a flexible carrier with two arms and adapted to accommodate the continuously changing shape of the abdomen of a pregnant woman; an array of speakers disposed on the arms of the carrier and substantially surrounding the in-utero infant with the sounds emitted from said speakers; an amplifier operatively associated with the array of speakers that enables tailoring the frequencies and tonal qualities of the sounds emitted from the array of speakers to levels suitable for an in-utero infant; and an audio device, operatively connected to the array of speakers and the amplifier, that provides a source from which sounds are played back.

Abstract: An active noise reduction system is provided for receiving an audio input signal and a noise interference signal and calculating an audio broadcasting signal according to a Feedback Filtered-X Least-Mean-Square (FFXLMS) algorithm, wherein the FFXLMS algorithm optimizes a (convergence factor) ? so as to decrease the numbers of divisions operated by the active noise reduction system and increase the operation speed of the active noise reduction system.