Abstract: This invention concerns a loudspeaker driver includes at least one actuator connected to a vibrating support to impart excitation to the latter when caused to move, wherein the loudspeaker driver further includes a plurality of sensing members arranged to move with the at least one actuator, each sensing member providing output sensing data dependent on the velocity of said at least one actuator, and means for determining the position of the at least one actuator based on at least one ratio (X/Y) of output sensing data or of linear combinations of output sensing data provided from the plurality of sensing members, said at least one ratio being independent of the velocity of the at least one actuator.

Abstract: A system for determining the detectability of a fleet of vehicles by a listener at a target site. The system receives as input at least the number of vehicles, a selection of a noise signature for each vehicle, and a distance to the target. The system may then estimate noise pressure level for the fleet along the estimated distance from the given location to the target site based on the noise signature associated with each vehicle and display a graph showing the noise pressure level vs. distance from the given location to the target site. In an embodiment, the system may estimate the background noise level at the target site and subtract this from the noise pressure level of the fleet at the target site to determine an expected noise level above background noise at the target site.

Abstract: In embodiments of the present invention improved capabilities are described for determining a multi-dimensional sound signature for a location within a hypothetical space, comparing the multi-dimensional sound signature to a known multi-dimensional sound signature, and modifying the hypothetical space such that the similarity between the multi-dimensional sound signature for the location within the modified hypothetical space and the known multi-dimensional sound signature is increased.

Abstract: Sound reproduction is controlled so as to be heard in the optimal state for the hearing function specific to elderly people. A frequency characteristic setting portion for setting the frequency characteristics of an inputted sound signal, and a sound volume setting portion for variable controlling the volume is disclosed. The frequency characteristic setting portion changes a frequency characteristic in which a sound band including a human voice band is emphasized to a frequency characteristic in which the characteristics of a gain in accordance with a frequency gradually becomes flat with an increase in the volume set by the sound volume setting portion.

Abstract: For signal processing of solid-borne or structure-borne sound signals, in particular in motor vehicles, and a corresponding occupant protection system, a first-order high-pass filter is provided as a filter, wherein the ?3 dB cut-off frequency thereof lies between the upper and lower operating frequency of the signal evaluation. The signal delay in the filter becomes significantly lower. Damping of the frequencies in the operating range becomes non-uniform, wherein components below the cut-off frequency are damped more heavily. Not only can triggering cases be detected earlier, but also non-triggering cases can be reliably detected.

Abstract: An audio beamforming apparatus includes a receiving circuit (103) which receives signals from an at least two-dimensional microphone array (101). A reference circuit (105) generates reference beams and a combining circuit (107) generates an output signal corresponding to a desired beam pattern by combining the reference beams. An estimation circuit (109) generates a direction estimate by determining angles corresponding to local minima for a power measure of the output signal in at least a first and respectively second angle interval. The direction estimate is generated by selecting one of the angles. The combining circuit (107) determines combination parameters to provide a notch in an angle corresponding to the direction estimate and a maximization of a directivity cost measure where the directivity cost measure is indicative of a ratio between a gain in the first direction and an energy averaged gain.

Abstract: In one embodiment the present invention includes a circuit for automatically adjusting an output of an audio device. The circuit includes a memory circuit, a detector circuit, a control circuit, and an output circuit. The memory circuit stores configuration information. The detector circuit detects environment information related to an environment in which the apparatus is present. The control circuit selects selected configuration information from the memory circuit according to the environment information detected by the detector circuit. The output circuit receives an input audio signal and the selected configuration information, modifies the input audio signal according to the selected configuration information, and generates an output audio signal corresponding to the input audio signal as modified according to the selected configuration information.

Abstract: A method of controlling audio recording using an electronic device and an electronic device are described. The electronic device comprises a microphone arrangement having a directivity pattern. A target direction relative to the electronic device is automatically determined in response to sensor data representing at least a portion of an area surrounding the electronic device. The microphone arrangement is automatically controlled in response to the determined target direction to adjust an angular orientation of the directivity pattern relative to the electronic device.

Abstract: A controller 103 that conducts various controls corresponding to input information selects target voltage information and target dead time information corresponding to input audio signal level information from data table information, outputs the target voltage information to a variable power supply unit 104, and outputs the target dead time information associated with the target voltage information to a dead time generator 105, to thereby set an output supply voltage and a dead time corresponding to an input audio signal level in a half bridge circuit 107. Accordingly, there can be provided a class D amplification device that can minimize the dead time and reduce distortion while suppressing the amount of heat generation caused by a through-current.

Abstract: A stereo ribbon microphone includes four ribbon microphone units disposed in a circle on a single horizontal plane. The ribbon microphone units are alternately assigned to a right channel and a left channel along the circumferential of the circle.

Abstract: A binaural decoder for an MPEG surround stream, which decodes an MPEG surround stream into a stereo 3D signal, and a decoding method thereof. The method includes dividing a compressed audio stream and head related transfer function (HRTF) data into subbands, selecting predetermined subbands of the HRTF data divided into subbands and filtering the HRTF data to obtain the selected subbands, decoding the audio stream divided into subbands into a stream of multi-channel audio data with respect to subbands according to spatial additional information, and binaural-synthesizing the HRTF data of the selected subbands with the multi-channel audio data of corresponding subbands.

Abstract: An audio amplifier circuit and a method of the same are provided. The audio amplifier circuit includes a processor, a pre-processing module, and a post-processing module. The processor controls a pre-gain value of the pre-processing module and a post-gain value of the post-processing module in accordance with a volume adjustment signal so as to process an audio input signal and generate an audio output signal. The pre-gain value and the post-gain value are dynamically adjusted by the audio amplifier circuit and a method thereof by according to the volume adjustment signal so as to generate low distortion, low noise audio output signals and improve audio playback quality of an audio output device.

Abstract: A method and system for removing noise due to bias electric power (MIC_Bias) or Time Division Multiple Access (TDMA) noise are provided. The system includes an interface with a microphone contact, for connecting to an accessory, an Analog-to-Digital (AD) converter for converting an analog voltage level, input via the microphone contact, into a digital voltage level, an audio processing unit for processing an audio signal input via the microphone contact, a switch for at least one of connecting and disconnecting the microphone contact and the AD converter, and a controller. The controller compares the digital voltage level with voltage levels stored in a look up table. The controller identifies a type of accessory connected to the interface. The controller also switches off the switch to prevent noise from being introduced into the audio signal via a wire connecting the microphone contact and the AD converter.

Abstract: An apparatus for driving speakers includes user-configurable amplifier cells for driving selected speakers, a synchronous rectifier circuit, and a control system. Each cell connects between power rails. The rectifier circuit provides current on the rails for consumption by the cells. The current depends on the configuration. The control system implements a model of the rectifier circuit and cells and uses it to control an audio input signal in response to information concerning electrical outputs of the amplifier cells. This indirectly limits current drawn from the synchronous rectifier circuit.

Abstract: An apparatus and a method for efficiently converting, by means of mixdown, audio signals recorded in multiple tracks into stereo signals. A plurality of audio signals input from an input terminal are recorded into a plurality of tracks of a recording medium. A mixer reads the audio signals recorded in the multiple tracks, subjects the audio signals to mixdown according to a mixer parameter set by an operation section, and records the processed signals as audio signals in the recording medium without reproducing the processed signal in real time. A plurality of mixer parameters are set, and a user can select a desired mixer parameter.

Abstract: Audio signals are processed for use in a communication event. A data store may be queried to obtain an indication of an echo direction, which relates to a direction from which audio signals output from the audio output are likely to be received at a microphone array (plurality of microphones) of a device. Beamformer coefficients of an adaptive beamformer of the device are determined in dependence upon the received indication of the echo direction. Audio signals are received at the microphone array. The adaptive beamformer applies the determined beamformer coefficients to the received audio signals, thereby generating a beamformer output for use in the communication event. The beamformer coefficients are determined such that echo suppression is applied to audio signals received at the microphone array from the indicated echo direction.

Abstract: Beamformer coefficients may include a plurality of sets of theoretical statistical data for theoretical signals. Each theoretical signal may have its own particular attributes. The statistical data may be used in computing beamformer coefficients for application by a beamformer to signals received at a device. Signals are received at an input of the device. A respective plurality of weights is determined, for the theoretical statistical data sets, based on an analysis of the extent to which the signals have the particular attributes of the theoretical signals. The theoretical are retrieved, and a statistical data set is calculated for the signals by performing a weighted sum of the theoretical statistical data sets using the determined respective plurality of weights. Beamformer coefficients are computed based on the calculated statistical data set for the signals, which are used by a beamformer to the signals for generating a beamformer output.

Abstract: A method for reducing the noise emission of a transformer, the transformer tank of which is filled with liquid and the tank wall of which vibrates during operation, is provided. The method is characterized by the sequence of the following method steps: detecting natural frequency values of the tank wall for at least one excitation frequency; determining at least one eigenmode for which the vibration of the tank wall is composed at an excitation frequency, from the natural frequency values, wherein areas of large curvature are determined on the tank wall; arranging at least one vibration loading device in at least one of said areas; and controlling the at least one vibration loading device by means of a control device in order to counteract the vibration of the tank wall.

Abstract: A system for headphone equalization includes a stored set of predetermined tone burst reference signals and a stored set of predetermined tone burst test signals that form a range of frequencies used in a user specific audio test to develop a headphone correction filter. A predetermined tone burst reference signal and a predetermined tone burst test signal may intermittently and sequentially drive a transducer included in the headphone. A loudness of the predetermined tone burst reference signal may be fixed and a loudness of the predetermined tone burst test signal may be variable with a gain setting. The gain setting may be used to generate the headphone correction filter.

Abstract: This document discusses, among other things, systems and methods to at least partially compensate for temperature sensitivity in a digital microphone system including, for example, a temperature sensitive membrane, such as an electrets condenser microphone (ECM).