Abstract: A system and method for routing an emergency data message to a PSAP may include receiving an emergency data message and cell code identifier indicative of a location of a wireless communications device of a user. A PSAP local to the user may be selected, and the emergency data message is sent to the selected PSAP. The location may be a centralized network location associated with an emergency network address, where the emergency network address is an easy address for users to remember.

Abstract: An active noise control system generates an anti-noise signal to drive a speaker to produce sound waves to destructively interfere with an undesired sound in a quiet zone. The anti-noise signal is generated with an adaptive filter having filter coefficients. The coefficients of the adaptive filter may be adjusted based on a first filter adjustment from a first listening region, and a second filter adjustment from a second listening region. A first weighting factor may be applied to the first filter adjustment, and a second weighting factor may be applied to the second filter adjustment. The first and second weighting factors may dictate the location and size of the quiet zone as being outside or partially within at least one of the first listening region and the second listening region.

Abstract: Level compression applied to the acoustic signals (18) received by a binaural hearing aid system (1) counteracts the preservation of inter-aural level differences (ILD) and thereby reduces the user's ability to locate the sound source and consequently his or her ability to understand speech in noisy environments. It is therefore known to increase the gain (57) in the hearing aid (2) receiving the louder signal and/or decreasing the gain (58) in the hearing aid (3) receiving the quieter signal, which at least in part allows for preserving the ILDs. However, in some situations this instead reduces the user's ability to understand speech, e.g. when acoustic noise is received at one ear (4) at a higher level (53, 54) than simultaneous speech at the other ear (5).

Abstract: The present invention has: a condenser microphone unit which performs electroacoustic conversion according to a change in an electrostatic capacitance between a diaphragm and a fixed pole; a non-inverting amplifier which is connected to one of the diaphragm and the fixed pole, and which has an impedance converter which converts an output impedance of the microphone unit into a low impedance; an inverting amplifier which receives an input of an output signal of the non-inverting amplifier through an input resistance, and which has a feedback resistance; and a variable resistor which is connected between an output of the non-inverting amplifier and an output of the inverting amplifier, and in which a wiper is connected to the diaphragm or the fixed pole, whichever is not connected to the non-inverting amplifier, and the sensitivity changes according to the position of the wiper of the variable resistor.

Abstract: There is provided a sound processing apparatus including a sound separation unit that separates an input sound into a plurality of sounds caused by a plurality of sound sources, a sound type estimation unit that estimates sound types of the plurality of sounds separated by the sound separation unit, a mixing ratio calculation unit that calculates a mixing ratio of each sound in accordance with the sound type estimated by the sound type estimation unit, and a sound mixing unit that mixes the plurality of sounds separated by the sound separation unit in the mixing ratio calculated by the mixing ratio calculation unit.

Abstract: The embodiments described herein relate to amplifier circuits with power rails. The power rails are generated in response to an input signal, which will typically be an audio signal, and with a sufficient magnitude to power the amplifier and to generate an output signal corresponding to the input signal.

Abstract: A decoding device includes a decoder configured to separate a first signal obtained by performing down-mix on original signals of a plurality of channels, a residual signal representing a component of a difference between the original signals and the first signal, and spatial information representing the relationship among the original signals of the plurality of channels from an input signal which is obtained by multiplexing the first signal, the residual signal, and the spatial information and decode the separated encoded first signal, the encoded residual signal, and the encoded spatial information; a decorrelation signal generation unit configured to generate a decorrelation signal as decorrelation of the first signal decoded by the decoder; a residual signal determination unit configured to determine whether a level of the residual signal decoded by the decoder is equal to or smaller than a predetermined residual threshold value; a second-signal generation unit.

Abstract: An audio apparatus is suitable for generating crowd sounds from an audio signal is disclosed in which the apparatus comprises modulation means operable to modulate a noise signal in response to the audio signal to generate a modulated noise signal, and diffusion delay means. The diffusion delay means is operable to apply a series of two or more delay operations, the input signal to a first such delay operation in the series being the modulated noise signal, and input to each subsequent delay operation in the series being the output signal generated by a preceding delay operation. Each delay operation comprises modifying that operation's input signal by the addition of a delayed version of that operation's input signal.

Abstract: Multiple independent sound images are formed by integrally performing uncorrelation processing and sound image localization processing on an input audio signal with signal processors that use output functions hl(x) and hr(x) obtained by integrating an uncorrelation function that generate multiple audio signals with low mutual correlation from an input audio signal. The signal processors have a sound image localization function for localizing the sound image of each of the multiple audio signals at a given sound source position.

Abstract: A display apparatus having an object-oriented 3D sound coordinate indication is disclosed, in which sounds are outputted from a speaker positioned matching with a coordinate of an object in a matrix speaker apparatus formed of a plurality of small size speakers disposed behind a display apparatus in cooperation with an object content displayed on a display apparatus screen.

Abstract: Noise reduction circuitry for a communication apparatus can apply different noise reduction transfer functions, depending on whether a listening device is connected to the apparatus. If no listening device is connected, the noise reduction transfer function can be adapted for use with microphones (12) and speakers (28) that form an integral part of the communication apparatus, which may be a cellular telephone. If a listening device is connected, the noise reduction transfer function can be adapted for use with microphones (12) and speakers (28) that form a part of the listening device. This allows the noise reduction circuitry to provide improved noise reduction performance.

Abstract: Methods and systems for power conservation in mobile devices are presented. In one example, a mobile communication device includes a first microphone and a second microphone. The mobile communication monitors an ambient noise level and responsive to the ambient noise level operates the mobile communication device in a normal operation mode or a power conservation mode. In power conservation mode, use of the second microphone is discretionary.

Abstract: There is provided a control system (1) comprising an earphone (2) and a processing device (3). The control system (1) has a processing device (3) for the evaluation of head movement information and/or head position information. In that case the processing device is adapted to output at least one control signal dependent on the evaluation, wherein the earphone has a sensor device (9a, 9b, 9c) for detecting the movement information and/or position information and a transmitting device (10) for transmitting the movement information and/or the position information to the processing device.

Abstract: In one embodiment, a directional microphone array having (at least) two microphones generates forward and backward cardioid signals from two (e.g., omnidirectional) microphone signals. An adaptation factor is applied to the backward cardioid signal, and the resulting adjusted backward cardioid signal is subtracted from the forward cardioid signal to generate a (first-order) output audio signal corresponding to a beampattern having no nulls for negative values of the adaptation factor. After low-pass filtering, spatial noise suppression can be applied to the output audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order output audio signals.

Abstract: Systems and methods for reconstructing decomposed audio signals are presented. In exemplary embodiments, a decomposed audio signal is received. The decomposed audio signal may include a plurality of frequency sub-band signals having successively shifted group delays as a function of frequency from a filter bank. The plurality of frequency sub-band signals may then be grouped into two or more groups. A delay function may be applied to at least one of the two or more groups. Subsequently, the groups may be combined to reconstruct the audio signal, which may be outputted accordingly.

Abstract: A sound zooming technique synchronized with a moving picture zooming function is disclosed. A signal to control ambient noise and a signal to control a long distance sound are extracted from an audio signal input through a plurality of microphones. A front directivity-emphasized signal is masked using the two signals and a weighted value to control volume.

Abstract: The present invention relates to an acoustic signal processing apparatus that suppresses auditory noise caused in a difference signal generated by acoustic signals of a plurality of channels, a processing method therefor, and a program. A difference spectrum calculation unit 320 calculates, as a difference spectrum, the absolute difference value between frequency spectra of the left and right channels sent from frequency spectrum generation units 311 and 312. A low-level band determination unit 330 determines a difference spectrum corresponding to a low-level band among difference spectra of all the frequency bands. A substitution spectrum generation unit 350 generates a substitution spectrum for which a difference spectrum is substituted, on the basis of the frequency spectrum of the left channel. A spectrum substitution unit 360 substitutes the difference spectrum corresponding to the low-level band for a substitution spectrum corresponding to the difference spectrum.

Abstract: A decoding method, medium, and system decoding an input compressed multi-channel signal, as a mono or stereo signal, into 2-channel binaural signals. Channel signals making up the multi-channel signals may be reconstructed from the input compressed signal in the quadrature mirror filter (QMF) domain, and head related transfer functions (HRTFs) for localizing channel signals in the frequency domain, represented as values in the time domain, may be transformed into spatial parameters in the QMF domain. Accordingly, channel signals may be localized in the QMF domain in directions corresponding to the channels, thereby decoding the input compressed signal as 2-channel binaural signals.

Abstract: A method for audio signal enhancement comprising obtaining (222) a first audio signal from a first physical microphone element and obtaining a second audio signal from a second physical microphone element. The audio signals are array processed (226) to generate a virtual linear first order element and a virtual non-linear even order element. The array processing (226) includes combining the virtual linear first order element and the virtual non-linear even order element to generate a directional audio signal having a primary audio beam. An apparatus is disclosed for implementing the method.

Abstract: Systems and methods for utilizing inter-microphone level differences to attenuate noise and enhance speech are provided. In exemplary embodiments, energy estimates of acoustic signals received by a primary microphone and a secondary microphone are determined in order to determine an inter-microphone level difference (ILD). This ILD in combination with a noise estimate based only on a primary microphone acoustic signal allow a filter estimate to be derived. In some embodiments, the derived filter estimate may be smoothed. The filter estimate is then applied to the acoustic signal from the primary microphone to generate a speech estimate.