Abstract: An audio processing system for processing a single channel audio signal includes a a processor configured to derive a synthetic difference component from the single channel audio input signal a filtering module configured to apply a first filter to the sum signal represented by the single channel signal and to apply a second filter to the synthetic difference signal; and a control module configured to crossfade to control the amount of the resulting audio signal effect by respectively scaling the sum signal and the difference signal.

Abstract: The application relates to audio encoder and decoder systems. An embodiment of the encoder system comprises a downmix stage for generating a downmix signal and a residual signal based on a stereo signal. In addition, the encoder system comprises a parameter determining stage for determining parametric stereo parameters such as an inter-channel intensity difference and an inter-channel cross-correlation. Preferably, the parametric stereo parameters are time- and frequency-variant. Moreover, the encoder system comprises a transform stage. The transform stage generates a pseudo left/right stereo signal by performing a transform based on the downmix signal and the residual signal. The pseudo stereo signal is processed by a perceptual stereo encoder. For stereo encoding, left/right encoding or mid/side encoding is selectable. Preferably, the selection between left/right stereo encoding and mid/side stereo encoding is time- and frequency-variant.

Abstract: An apparatus for microphone positioning includes a spatial power distribution determiner and a spatial information estimator. The spatial power distribution determiner is adapted to determine a spatial power density indicating power values for a plurality of locations of an environment based on sound source information indicating one or more power values and one or more position values of one or more sound sources located in the environment. The spatial information estimator is adapted to estimate acoustic spatial information based on the spatial power density.

Abstract: A hearing aid having an electrical assembly, the electrical assembly includes a printed circuit board having a first board surface and a second board surface, the printed circuit board having a first pad in a first pad region on the first board surface, the first pad region having a first board normal; a flexible printed circuit board comprising an antenna, the antenna having a first terminal in a first terminal region on a first flexfilm surface of the flexible printed circuit board, the first terminal region having a first terminal normal; and first electrically conductive material connecting the first pad and the first terminal, and wherein the first terminal normal and the first board normal forms a first angle that is anywhere from 30° to 150°.

Abstract: A speaker unit according to an embodiment includes a diaphragm, a frame, and a current plate. The frame supports an outer circumferential portion of the diaphragm. The current plate is provided in the frame so as to be located at a backside of the diaphragm, and its principal surface is arranged in a direction along an amplitude direction of the diaphragm.

Abstract: The invention is suited for improving a low bit rate compressed and decompressed Higher Order Ambisonics HOA signal representation of a sound field, wherein the decompression provides a spatially sparse decoded HOA representation and a set of indices of coefficient sequences of this representation. From reconstructed signals of the original HOA representation a number of modified phase spectra signals are created using de-correlation filters, which modified phase spectra signals are uncorrelated with the signals of said original representation. The modified phase spectra signals are mixed with each other using predetermined mixing parameters, in order to provide a replicated ambient HOA component. Finally the spatially sparse decoded HOA representation is enhanced with the replicated time domain HOA representation.

Abstract: A microphone pivot device of a headphone comprises at least one magnet in communication with a movable microphone, the at least one magnet having a plurality of segments, each segment having a magnet characteristic; a sensor that detects a magnet characteristic of a magnet segment when the segment is moved in response to a movement of the microphone so that the segment is in proximity of the sensor; and a controller that receives a signal from the sensor in response to the detected magnet characteristic, and that controls an operation of the microphone in response to the received signal.

Abstract: A system, including an adaptive noise cancellation sub-system, wherein the system is configured to adjust operation of the sub-system from a first operating state to a second operating state upon a determination that operation of the adaptive noise cancellation sub-system will be effectively affected by an own voice body conducted noise phenomenon.

Abstract: The present invention proposes a method for providing at least one indication regarding whether an input signal contains a tonal signal component. The proposed method comprises decomposing the input signal into a plurality of band-limited input signals and then for each of one or more of the plurality of band-limited input signals performing the following step: estimating parameters of a parametric tonal signal model adapted to model and predict a tonal-only signal component of a selected one of the one or more of the plurality of band-limited input signals; synthesizing based on the estimated parameters a corresponding predicted band-limited input signal; and determining a measure of mismatch between the selected band-limited input signal and the corresponding predicted band-limited input signal. Subsequently, the at least one indication is provided based on the measure of mismatch determined for at least one of the one or more of the plurality of band-limited input signals.

Abstract: A device for processing audio data obtains data representing quantized versions of a set of one or more spatial vectors. Each respective spatial vector of the set of spatial vectors corresponds to a respective audio signal of the set of audio signals. Each of the spatial vectors is in a Higher-Order Ambisonics (HOA) domain and is computed based on a set of loudspeaker locations. The device inverse quantizes the quantized versions of the spatial vectors.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone is provided proximate the speaker to measure the output of the transducer in order to control the adaptation of the anti-noise signal and to estimate an electro-acoustical path from the noise canceling circuit through the transducer. The anti-noise signal is adaptively generated to minimize the ambient audio sounds at the error microphone. A processing circuit that performs the adaptive noise canceling (ANC) function also filters one or both of the reference and/or error microphone signals, to bias the adaptation of the adaptive filter in one or more frequency regions to alter a degree of the minimization of the ambient audio sounds at the error microphone.

Abstract: A circuit and method is disclosed for filtering an audio signal. The circuit has a first quadrature source and multipliers for multiplying the input signal by the I and Q outputs of the quadrature source. The multiplied inputs are then passed through a pair of low pass filters, which may have an adjustable Q factor. The outputs of the low pass filters are then multiplied in a second pair of multipliers by the I and Q outputs, respectively, of a second quadrature source, which will typically be of the same frequency, but different amplitude and phase, of the first quadrature source. The twice-multiplied signals are then summed by an adder to provide an output signal. The circuit may be modified to include a companding circuit between the low pass filters and the second pair of multipliers that determines the amplitude of the input signal, filters it, and compands the signal in a compandor. The compandor may have adjustable parameters.

Abstract: An audio apparatus is configured to switch, when there exists a first audio interface between the audio apparatus and a computer apparatus, to using a second audio interface between the audio apparatus and the computer apparatus, the second audio interface being different from the first audio interface. The switching comprises: receiving, via the first audio interface, combined audio data and non-audio data, the non-audio data comprising a request to switch to using the second audio interface; obtaining the request from the data; and, in response to obtaining the request, transmitting to the computer apparatus a confirmation of switching to using the second audio interface. The audio apparatus and the computer apparatus are described and claimed.

Abstract: The MPEG2 Advanced Audio Coder (AAC) standard limits the number of filters used to either one filter for a “short” block or three filters for a “long” block. In cases where the need for additional filters is present but the limit of permissible filters has been reached, the remaining frequency spectra are simply not covered by TNS. Two solutions are proposed to deploy TNS filters in order to get the entire spectrum of the signal into TNS. The first method involves a filter bridging technique and complies with the current AAC standard. The second method involves a filter clustering technique. Although the second method is both more efficient and accurate in capturing the temporal structure of the time signal, it is not AAC standard compliant. Thus, a new syntax for packing filter information derived using the second method for transmission to a receiver is also outlined.

Abstract: A hearing device has a microphone for detecting an acoustic signal from the environment of the hearing device, and a housing in which the microphone is arranged. The housing is formed with a microphone opening through which the microphone communicates with the environment of the hearing device for detecting the acoustic signal. The hearing device further has a damping element, by way of which the microphone is supported against the housing. Moreover, the hearing device has a barrier element for sealing the microphone opening against moisture in a sound-transmitting manner. The damping element has a pocket in which the barrier element is received reversibly. Furthermore, the barrier element is sealed off from the housing by way of the damping element all around the microphone opening.

Abstract: A transducer module, comprising: a supporting substrate, having a first side and a second side; a cap, which extends over the first side of the supporting substrate and defines therewith a first chamber and a second chamber internally isolated from one another; a first transducer in the first chamber; a second transducer in the second chamber; and a control chip, which extends at least partially in the first chamber and/or in the second chamber and is functionally coupled to the first and second transducers for receiving, in use, the signals transduced by the first and second transducers.

Abstract: The invention relates to an audio processing device (1) including: at least one connector (2, 7), suitable for receiving data defining an input sound signal; a processing module (5, 6) suitable for determining at least one signal intended for a loudspeaker (12, 13) as a function of said defining data, a communication module (8, 9) for communicating with at least one other audio processing device, according to claim 1; said device being characterized in that it furthermore includes a detection module (4) suitable for detecting the proximity of another device according to claim 1 and in that the processing module is suitable for determining said signal intended for the loudspeaker as a function furthermore of a number of other detected device(s).

Abstract: We disclose a medical toilet that includes at least one stethoscope probe in the toilet lid. The stethoscope probes may be mounted on a compliant extension which extends the stethoscope probes toward a user seated on the toilet. The medical toilet may include a device which adjusts the height of the stethoscope probes to adjust for different size users and to collect measurements from different parts of a user. The medical toilet may include a device which records the sounds collected by the stethoscope and may also include a controller that proposes a diagnosis based on the sounds collected by the stethoscope. The medical toilet may also transmit recordings and other information to an external machine.

Abstract: According to an embodiment, a method includes amplifying a signal provided by a capacitive signal source to form an amplified signal, detecting a peak voltage of the amplified signal, and adjusting a controllable impedance coupled to an output of the capacitive signal source in response to detecting the peak voltage. The controllable impedance is adjusted to a value inversely proportional to the detected peak voltage.

Abstract: A display device having a speaker module includes a display, an ultrasonic speaker and an adjustment module. The speaker module includes at least one first speaker module and at least one second speaker module. The first speaker module is disposed at the display and adapted to emit a first ultrasonic wave. The second speaker module is disposed at the display and adapted to emit a second ultrasonic wave. The first ultrasonic wave and the second ultrasonic wave are intersected to form an audible region. The adjustment module is adapted to control the first speaker module and the second speaker module to rotate relative to the display, so as to drive the audible region to target a user.