Abstract: A noise reduction is provided for a hearing apparatus, with which both stationary and also non-stationary interference noises can be attenuated in an input signal. An output signal is in this way to convey a quite sound impression. A signal processing is provided, which effects a noise reduction on the basis of two different methods. Provision is made on the one hand for a noise reduction for stationary interference noises and on the other hand for a noise reduction for spatially oriented interference noises. A selection facility selects between the two noise reductions.

Abstract: There is provided a sound processing apparatus and a sound processing method which are capable of reproducing discrete data with a high-quality sound matching users' preferences. In a sound processing means 2, since an interpolation value reflecting a value of a variable parameter ? by which the value of a control sampling function c0(t) is multiplied can be calculated, an analog signal obtained through the interpolation performed in a sampling function sN(t) can be regulated in accordance with the variable parameter ? by changing the value of the variable parameter ?. In this way, by allowing the user to appropriately change the variable parameter ? in accordance with various conditions including music reproduction environments, sound sources, musical tones and so on, it becomes possible to reproduce high-quality-sound music in which its frequency characteristics of the analog signal have changed and a high quality desired by the user is obtained.

Abstract: Methods and apparatus are disclosed for dynamically selecting the number of audio channels to mix per listener in an audio processor, such as a server-based three-dimensional audio mixer, as a function of any reasonable measurement of the load on the audio processor.

Abstract: The present invention relates to an apparatus and method for redeeming otherwise closed and concealed information contained in audio signals. An active circuit balances the ratio between in-phase and out-of-phase signals through the application of sum and difference signals and adjusts the ratio of gain in stereophonic signals as well as in monophonic and multichannel signal applications. This includes both the primary reference signal, and a plurality of redundant duplicate signals, substantially identical in all respects to the primary reference signal except in relation to magnitude and phase, for the purpose of unfolding, or opening the audio signal content. A pair of output signal levels approximates the golden ratio where the golden ratio is one plus the square root of five divided by two which gives an irrational number 1.618.

Abstract: A coefficient calculation unit calculates a coefficient (p) according to input signals (x[n]). A balance setting unit decides, according to this coefficient (p), a coefficient used to control the level of a signal in a digital level correction unit and a coefficient used to adjust the characteristics of a signal in a characteristic correction unit, thus allowing the digital level correction unit and characteristic correction unit to execute adaptive processing.

Abstract: The teachings are directed to a power amplifier expansion card (“amplifier card”) for a computer. The amplifier card receives audio data through an input port and amplifies the audio data to high-level passive speaker power for transmissions through an output port. The amplifier card can comprise a circuit board having (i) at least two audio channels; (ii) an audio power amplification circuit for amplifying audio power to at least 20 W RMS per channel; and (iii) an onboard cooling system. The amplifier card can be used, for example, in a multimedia system having at least a studio controller operably connected to the amplifier card for receiving and processing input audio data. Examples of such multimedia systems include, but are not limited to, a television home entertainment system, an audio home entertainment system, a music production studio system, a gaming system, a personal computing system, or any combination thereof.

Abstract: The present invention relates to direct voice and instrument monitoring systems and methods of use thereof.

Abstract: A portable computing device having a substantially non-conducting outer housing and alternative electrical grounding and audio system architectures is disclosed. The device can be a laptop computer having a main logic board, a keyboard assembly, an audio source positioned below the keyboard assembly, and an equalizer electrically coupled to the audio source, with each of these components being electrically coupled to a universal grounding structure. The audio source emits sound waves that are propagated through the keyboard assembly and between gaps between keyboard keys and the outer housing. Settings for the equalizer can be selected to account for sound absorption and amplification characteristics of the sound waves along these sound transmission paths.

Abstract: An exemplary audio processing system includes a gain control unit, a sampling unit, and a triggering unit. The gain control unit is configured for amplifying an audio signal. The sampling unit is configured for sampling the audio signal. The triggering unit is configured for generating a gain reduction unit if the amplitude exceeds a predetermined value over a predetermined time period. The first predetermined value is set so that if the amplitude of the audio signal exceeds the predetermined value and the gain of the gain control unit is not reduced, the amplitude of the amplified audio signal exceeds a predetermined acceptable range. The gain control unit is also configured for reducing the gain of the gain control unit responding to the gain reduction signal to limit the amplitude of the amplified audio signal within the predetermined acceptable range.

Abstract: Systems and methods described herein provide for low latency active noise cancellation, which alleviates the problems associated with analog filter circuitry. The present technology utilizes low latency digital signal processing techniques that overcome the high latency conventionally associated with conversion between the analog and digital domains. As a result, low latency active noise cancellation is performed utilizing digital filter circuitry which is not subject to the inaccuracies and drift of analog filter components. In doing so, the present technology provides robust, high quality active noise cancellation.

Abstract: Methods are disclosed for improving sound localization of the human ear. In some embodiments, the method may include creating virtual movement of a plurality of localized sources by applying a periodic function to one or more location parameters of a head related transfer function (HRTF).

Abstract: A passive sound pressure level (SPL) limiter is provided that can be used with audio sources of varying drive levels and headphones or earphones of varying sensitivity. The SPL limiter includes a control circuit and left and right channel limiting circuits, each of which includes a pair of FETS. The SPL limiter may include a balancing circuit that separates the control signal output by the control circuit into left and right channel control signals. The SPL limiter may also include a microphone, for example integrated into an earpiece cable.

Abstract: There is provided a capacitor microphone comprising a capacitor transducer (KW), a high frequency bridge (HFB) coupled to the capacitor transducer (KW), a high frequency coil (HFS) coupled to the high frequency bridge (HFB), an HF transformer (HFT), a synchronous demodulator (SD), a low frequency output (NFA) and a high frequency stabilizer unit (SE). The high frequency stabilizer unit (SE) is coupled between the synchronous demodulator (SD) and the low frequency output (NFA) and serves to stabilize the HF voltage.

Abstract: The instant development relates to utilizing the sonic motion generated by a speaker to move objects in various directions in response to the variation in the frequency and amplitude of the sonic vibrations. This can be used to move objects in a linear and/or rotating manner. This is accomplished by the utilization of directionality oriented members.

Abstract: A microphone signal compensation apparatus includes a plurality of audio input units to respectively receive a target signal, each audio input unit of the plurality of audio input units including a microphone; a constant filter unit to selectively apply a constant filtering calibration scheme to signals output by the plurality of audio input units to compensate for a difference in at least one characteristic among the audio input units, the constant filtering calibration scheme being estimated from an average value of a ratio of a desired signal to a reference signal among the signals output by the plurality of audio input units; and a noise remover to remove noise from the signals processed by the constant filter unit, and to separate the target signal from the signals from which the noise has been removed.

Abstract: A mobile communication device that may be used by an investigator to secretly record a suspect and track the investigator's location includes a microphone for detecting sounds, a communication component for transmitting signals or data representative of the sounds to one of the portable computers via the wireless communication network, a display for displaying information related to operation of the device, a computing device, and a computer program implemented by the computing device. The computer program is operable to place the device in a stealth mode in which the display is turned off or otherwise deactivated to make the entire device appear to be turned off while the microphone continues to detect sounds and the communication component continues to transmit signals or data representative of the sounds to a surveillance or support team operating one of the portable computers.

Abstract: For headphone subsystems that employ common ground switches for speaker outputs (for example), there can be a significant issue with cross-talk and ground noise. Here, configurations for an amplifier and switch network are provided, which generally cancel noise from the “ground switch,” so as to provide an improvement over conventional configurations with little overhead. Additionally, the cross-talk for these configurations are not generally dependent on the “ground switch” or speaker impedance.

Abstract: An audio system including an inverter, a first amplifier, a first mixer, a main speaker, and a first audio output port is provided. The inverter receives a first audio signal and inverts the first audio signal into an inverted audio signal. The first amplifier receives the inverted audio signal from the inverter and multiplies the inverted audio signal by a first gain to generate a first compensating audio signal. The first mixer receives a second audio signal and mixes the first compensating audio signal with the second audio signal to generate a first low-interference audio signal. The main speaker receives and plays the first audio signal. The first audio output port receives the first low-interference audio signal from the first mixer and transmits the first low-interference audio signal to a first external speaker. An electronic device and a car audio system capable of generating the low-interference audio signal are provided.

Abstract: A stethoscope chest piece has a chest piece housing containing a piezoelectric (piezo) element mounted within a housing. The piezo element converts body signals to an electrical representation that is processed by circuitry on a circuit board. A back stop is affixed to the circuit board to prevent excess excursion by the piezo element and may be one of the circuit elements on the circuit board. This abstract is not to be considered limiting, since other embodiments may deviate from the features described in this abstract.

Abstract: A surround sound system acquires a set of original channel signals as picked up or modeled by a set of microphones, located in a real or virtual acoustic space. Two microphones, mounted on or near the front of a head that may be real or artificial, are oriented to pick up the sounds emanating from the front. Two additional microphones are located near the ear canals of the head configured to emulate a human head in a manner that encodes the surround sound signals with a head-related transfer function (HRTF) that, in reproduction, enhances the accuracy and realism with which a listener perceives the various source locations. The head can be augmented, e.g. with small baffles strategically configured and located to modify the HRTF for overall accuracy and realism. Surround sound channels, so originated, may be transmitted for real-time reproduction or may be recorded, filtered, delayed, or otherwise processed and stored in memory for later reproduction. In an exemplary 4.