Abstract: A device for transmitting engine sound into an interior of a motor vehicle includes a sound transmission line having a first end in acoustic communication with an air intake tract and a second end arranged to direct engine sound towards the motor vehicle interior. A sound transmitting porous media is arranged in-line with said sound transmission line and restricts airflow wherein the porous media such that airflow in the sound transmission line is required to pass through said porous media.

Abstract: A method for reducing howling in a communication system containing collocated mobile devices is presented. In a transmitter, an audio signal is received at a microphone. Acoustic feedback is removed from the audio signal and the resulting signal is encoded and transmitted either using direct or trunked mode operation to a receiver. The encoded signal is decoded at the transmitter, in addition to at the receiver, and fed back to an echo canceller with sufficient delay to account for substantially the entirety of a loop delay from encoding of the audio signal to reception of the acoustic feedback at the microphone to enable removal of the acoustic feedback. An estimate of the acoustic feedback is used to initially remove the acoustic feedback, the error being fed back to the processor to adaptively change the signal being subtracted from the audio signal to better reduce the acoustic feedback.

Abstract: The present invention disclosed an all-digital speaker system device based on multi-bit ?-? Modulation, comprising an A/D converter, an interpolation filter, a ?-? modulator, a dynamic mismatch regulator, a differential buffer and a speaker array. The present invention takes advantage of the ?-? modulation technique to effectively reduce the cost and complexity of the hardware realization of the speaker system device, thus realizing the full digitalization of the whole sound transmission link and the integration of the system device with low power dissipation and volume. Furthermore, the system device according to the present invention is provided with a better control ability on the local sound field, thus providing a better practicable method for the voice secret transmission.

Abstract: A method of suppressing wind noise in a voice signal determines an upper frequency limit that lies within the frequency spectrum of the voice signal, and for each of a plurality of frequency bands below the upper frequency limit, compares the average power of signal components in a first portion of the signal to the average power of signal components in a second portion of the signal, where the second portion is successive to the first portion. Signal components are identified in at least one of the plurality of frequency bands as containing impulsive wind noise in dependence on the comparison, and the identified signal components are attenuated.

Abstract: A stereo speaker sound capturing and guiding system. The system includes devices that connected on each one of the stereo speakers as a cap. The device covers the speaker cones in hermetic way that captures the sound of the specific speaker and don't let the sound mixed with the sound of the other speaker that located in the same closed box or very near to the first speaker. The sound in the cap device guided directly from the speaker through connector facility between the speakers and the user's ears—such as air tubes—and transmitted exclusively to the user's matched ear—the sound from the right speaker gets only to right ear and sound from the left speaker gets only to the left ear—as at least two independent audio channels.

Abstract: An earpiece of an ANR device incorporates one or more of feedforward-based ANR; feedback-based ANR; passive noise reduction (PNR) of environmental noise sounds in the environment external to the casing of the earpiece in higher audible frequencies; a controlled leak acoustically coupling the front cavity to the environment external to the casing of the ANR device where the coupling may be through another cavity that is closable to the environment external to the casing with a leaky cover; a combination of an acoustically resistive port and a mass port coupling a rear cavity to the environment external to the casing where the coupling may be through another cavity that is closable to the environment external to the casing with a leaky cover; a feedforward microphone given acoustic access to the environment external to the casing through an aperture that is overlain with a leaky cover or that is enclosed within a cavity that is acoustically coupled to the environment external to the casing with a leak.

Abstract: A method and an apparatus for removing white noise in a portable terminal are provided. The method for removing the white noise in the portable terminal includes measuring a volume variation of a voice signal output from a power amplifier; detecting a frequency band including white noise using the measured volume variation; and removing signals of the detected frequency band in the voice signal before output to speaker.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; receiving user gain input; generating a linear gain factor and a non-linear gain factor using the user gain input; modifying the non-linear gain factor using absolute threshold of hearing and power of the input signal to generate a modified non-linear gain factor; and, applying the linear gain factor and the modified non-linear gain factor to the audio signal.

Abstract: An electronic device includes an enclosure and a speaker set received in the enclosure. The speaker set includes a shell, a pair of first and second speakers received in the shell. The first speakers provide high frequency acoustic output, and the second speakers provide low and middle frequency acoustic output. The shell includes a fixing plate. The fixing plate defines two tweet outlets aligning respectively with the first speakers, and two middle woof outlets aligning respectively with the second speakers. A central point of each middle woof outlet is located nearer to an imaginary central line of the fixing plate than a central point of each tweet outlet. A portion of each of the middle woof outlets is shaded by a portion of the electronic device. The tweet outlets are located beside the portion of the electronic device but not shaded by the portion.

Abstract: In embodiments of the present invention improved capabilities are described for digitally transmitting audio that is converted from analog audio received from analog media pickup devices in a live performance venue by a stage box to a base unit over off-the-shelf twisted pair cable while sending pre-amplification control signals and power over the cable to the stage box. Audio for the performance venue is remotely managed from a virtual audio engineering mixing board that wirelessly communicates audio control commands to the stage box from a handheld computing device.

Abstract: A sound system for capturing and reproducing sounds produced by a plurality of sound sources. The system comprises a device for receiving sounds produced by the plurality of sound sources and converting the separately received sounds to a plurality of separate audio signals without mixing the audio signals. The system may further comprise a device for separately storing the plurality of separate audio signals on a recording medium without mixing the audio signals and a device for reading the stored audio signals from the recording medium. The system further includes a reproduction system for recreating the plurality of separate audio signals. Also, the system comprises an amplification network which comprises a plurality of amplifier systems, with one or more separate amplifiers in each amplifier system for separately amplifying each of the separate audio signals.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; estimating indicator function using a signal power of the input signal; obtaining an adapted filter using the indicator function and an equalization filter; and, generating an output signal by applying the adapted filter to the input signal.

Abstract: There is provided a loudspeaker unit, in particular a ceiling loudspeaker unit, which has a housing (50) having a front housing portion (60). Arranged in the front housing portion (60) are a woofer unit (10), a baffle panel (70) and a tweeter unit (30) in a 2-way coaxial system. The tweeter unit (30) has a tweeter horn having a plurality of legs (33) and a plurality of openings (31) at the edge of the tweeter horn (30).

Abstract: An energy saving PWM sound device forms a frequency-dependent impedance device implemented on the PWM output of the voice-ICs in series to lower the power consumption and to enhance sound quality of the voice-IC products. The device includes a hollow core body and an elongated conductive element. The core body has a top rim, a bottom rim, and two side rims to define a through channel therewithin. The conductive element winds around the top rim of the core body, wherein two ends of the conductive element are adapted for connecting with the PWM output, such that when the conductive element is electrically conducted, the core body acts as an inductor at lower frequency and a frequency-dependent resistor in series with the inductor at higher frequency for blocking most of PWM carrier frequency so as to reduce the power consumption of the voice-ICs product.

Abstract: An apparatus for processing a signal and method thereof are disclosed. The present invention includes receiving a downmix signal generated from a plural-channel signal, mix information and phase shift information on the plural-channel signal, upmixing the downmix signal into the plural-channel signal by applying the mix information to the downmix signal, and generating an original plural-channel signal by shifting a phase of at least one channel of the plural-channel signal based on the phase shift information. According to the present invention compensate a reconstructed original plural-channel signal using compensation information (phase shift information), thereby compensating a phase or a gain lost in the plural-channel signal reconstructed by upmixing using mix information.

Abstract: Provided is an encoding/decoding apparatus and method of multi-channel signals. The encoding apparatus and method of multi-channel signals may encode phase information of the multi-channel signals using a quantization scheme and a lossless encoding scheme, and the decoding apparatus and method of multi-channel signals may decode the phase information using an inverse-quantization scheme and a lossless decoding scheme.

Abstract: The present invention is related to an improved power amplifier and a method for restraining power of the improved power amplifier. The improved power amplifier has an output power restraint unit, and the output power restraint unit is capable of restraining output power of the improved power amplifier when the output power is exceedingly large. A method for restraining power of a power amplifier, the method comprises the steps of: determining whether power of output powers signal are exceedingly large through a power signal transformation unit, if yes, adjusting two variable resistor of an input amplifier unit for adjusting the power of the power signals, and outputting the adjusted power signals for driving a load via output terminals of the power amplifier.

Abstract: A micro-electromechanical (MEMS) based directional sound sensor includes a two sensor wings attached to a surrounding support structure by two legs. The support structure is hollow beneath the sensor wings allowing the sensor wings to vibrate in response to sound excitation. In one embodiment, interdigitated comb finger capacitors attached on the sensor wing edges and the support structure enable an electrostatic (capacitive) readout related to the vibrations of the sensor which allows determination of the sound direction.

Abstract: One or a combination of two or more specific loudness model functions selected from a group of two or more of such functions are employed in calculating the perceptual loudness of an audio signal. The function or functions may be selected, for example, by a measure of the degree to which the audio signal is narrowband or wideband. Alternatively or with such a selection from a group of functions, a gain value G[t] is calculated, which gain, when applied to the audio signal, results in a perceived loudness substantially the same as a reference loudness. The gain calculating employs an iterative processing loop that includes the perceptual loudness calculation.

Abstract: Speaker control systems are implemented using a variety of systems and methods. In one such system, an audio signal is distributed to at least one speaker selected from a set of speakers. The system includes a set of audio-distribution circuits for distributing the audio signal to the set of speakers. Logic circuitry is used to automatically determine an impedance-based parameter for use in distributing the audio signal to the at least one selected speaker based at least partly on a load-tolerance parameter for an audio source and on an impedance of the at least one selected speaker. The logic circuitry controls distribution of the audio signal to the selected speakers based at least partly on the impedance-based parameter. One or more of the set of audio-distribution circuits are used by the logic circuitry to distribute the audio signal to the at least one selected speaker.