Abstract: An acoustic conversion device for an active noise control is provided. The acoustic conversion device can effectively suppress or remove noise by controlling variation in a lowest resonance frequency. A volume of a closed space is smaller than an equivalent compliance air volume of a speaker unit. Therefore, motion of a vibration plate, an edge and a damper is suppressed, it is possible to restrain the edge and the damper from being deteriorated with time, variation in lowest resonance frequency can be suppressed, phase of emitted sound can be less prone to be deviated from opposite phase of noise, and it is possible to effectively suppress or remove noise.

Abstract: The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency Q?+r?0 is generated on the basis of existing components at ? and ?+?0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.

Abstract: The present invention discloses an implementation method and a device of a multi-bit ?-? modulation-based digital speaker system. The method comprises, 1) digital format converting; 2) oversampling interpolation filtering; 3) multi-bit ?-? modulating; 4) thermometer coding; 5) multi-channel mismatch shaping; 6) coding format converting; 7) multi-channel digital power-amplifying; 8) driving a speaker array or a multiple voice coil speaker to sound. The device comprises: a digital input interface, an oversampling interpolation filter, a multi-bit ?-? modulator, a thermometer coder, a multi-channel mismatch shaper, a coding format converter, a multi-channel digital power-amplifier, and a speaker array or a multiple voice coil speaker; each portion being connected in proper order.

Abstract: A method for modifying sound in a motor vehicle that has an audio system. The audio system has at least one electro-acoustic transducer that creates sound from audio signals. An initial audio signal that represents a particular vehicle sound is provided, where the initial audio signal has a plurality of signal properties. A modified audio signal is then provided to the audio system, based on at least one current motor vehicle operating condition. The modified audio signal includes at least one modified signal property that differs from that of the initial audio signal. The at least one signal property is modified by a modification process that relates to a vehicle operating condition.

Abstract: A hearing assistance system having an audio signal wireless transmission unit for transmitting a stereo audio signal with left and right ear channels, a first ear unit for being worn at a user's ear and having a hearing instrument and a wireless audio signal receiver unit, the hearing instrument providing the first receiver unit with information as to which ear of the user the hearing instrument is being worn and information as to whether, or not, there is a like second ear unit. The first receiver unit is adapted to decide, depending on the received information as to which ear the first hearing instrument is fitted and whether a second ear unit is worn at the other ear, to receive one of the right and left ear channels, and, in absence of a stereo channel, to supply the respective mono channel of the audio signal to the first hearing instrument.

Abstract: A system, article, and method of audio quality and latency adjustment for audio processing by using audio feedback.

Abstract: The present disclosure concerns a method for determining a mold shape quality (Q) of an ear mold (2) customized to an ear canal (31) of a user by means of a mold testing system (100). The mold testing system (100) comprises an acoustic generator (3) and a microphone (4) arranged in a test housing (1) with an open end (1c) that is connectable to an inner passage (2c) of the custom ear mold (2). The method comprises inserting the ear mold (2) into the ear canal (31); connecting the test housing (1) to the ear mold (2); generating an acoustic signal (Sg) into the ear canal (31) via the inner passage (2c); measuring the acoustic signal (Sr) from the ear canal (31); comparing the measured signal (Sm) with a reference signal (Se); and providing a feedback signal (Sf) based on the said comparison. The feedback signal (Sf) is a measure for the mold shape quality (Q) of the custom ear mold (2).

Abstract: This disclosure relates to speakers and more specifically to an array speaker for distributing music uniformly across a room. A number of audio drivers can be radially distributed within a speaker housing so that an output of the drivers is distributed evenly throughout the room. In some embodiments, the exit geometry of the audio drivers can be configured to bounce off a surface supporting the array speaker to improve the distribution of music throughout the room. The array speaker can include a number of vibration isolation elements distributed within a housing of the array speaker. The vibration isolation elements can be configured reduce the strength of forces generated by a subwoofer of the array speaker.

Abstract: A method comprises converting an audio frequency domain signal into one or more voltage signals. Then the characteristics of the one or more voltage signals are determined. Afterwards the characteristics of the one or more voltage signals are compared with one or more characteristics of an audio trigger command. Activation of an audio user interface is then activated on the basis of the comparison.

Abstract: A playback request processing method and apparatus belongs to the field of computer technologies. The method includes: receiving a playback request sent by a sound source; determining whether a sound diffusion range of the sound source is within a current receiving range of a recipient; and processing the playback request if the sound diffusion range of the sound source is within the current receiving range of the recipient. In the present disclosure, it is determined according to the received playback request sent by the sound source that the sound diffusion range of the sound source is within the current receiving range of the recipient, and whether to process the playback request sent by the sound source is determined according to a determination result.

Abstract: A connector for providing audio signals to reduce noise during transmissions. A multi-pin connector receives audio signals and a ground, power, and audio signal connector transfers the audio signals to a noise-reduction device. An audio output transfers audio signals from the multi-pin connector to an in-ear system and at, least one ear-protection device. A multi-pin signal separator in communication with the multi-pin connector, the ground, power, and audio signal connector, and the audio output can receive all or a portion of the audio signals from the multi-pin connector, the ground, power, and audio signal connector, and the audio output and can selectively pass the audio signals to at least one of: the in-ear system and the at least one ear-protection device, while accepting a plurality of pin configurations.

Abstract: There is proposed a signal processing apparatus for enhancing a target signal and suppressing the remaining components without increasing an array size or the number of sensors. This signal processing apparatus includes a first array processor that generates a first array processing signal by enhancing a predetermined signal among signals received from a plurality of sensors, and a decorrelator that generates a decorrelated signal by removing, from the first array processing signal, a signal component correlated with a signal received from an auxiliary sensor different from the plurality of sensors.

Abstract: An audio system that maintains an identical or similar direct-to-reverberant ratio for sound produced from a first speaker array and sound produced by a second speaker array at the location of a listener is described. The audio system may determine characteristics of the first and second speaker arrays, including the distance between the first speaker array and the listener and the second speaker array and the listener. Based on these characteristics, beam patterns are selected for one or more of the speaker arrays such that sound produced by each of the speaker arrays maintains a preferred direct-to-reverberant ratio at the location of the listener.

Abstract: There are provided decoding and encoding methods for encoding and decoding of multichannel audio content for playback on a speaker configuration with N channels. The decoding method comprises decoding, in a first decoding module, M input audio signals into M mid signals which are suitable for playback on a speaker configuration with M channels; and for each of the N channels in excess of M channels, receiving an additional input audio signal corresponding to one of the M mid signals and decoding the input audio signal and its corresponding mid signal so as to generate a stereo signal including a first and a second audio signal which are suitable for playback on two of the N channels of the speaker configuration.

Abstract: A computing device includes a first housing and a second housing attached by a hinge. The first housing includes a first microphone and the second housing includes a second microphone. After determining that an angle between the first and second housing has changed to a current angle, the computing device may determine a distance between the first microphone and the second microphone based on the current angle. A first audio signal from the first microphone and a second audio signal from the second microphone may each be modified (e.g., using a beamforming algorithm) to create first and second modified audio signals. The first and second modified audio signals may include less noise than the first and second audio signals. The first and second modified audio signals may be sent to an output jack or to an audio application.

Abstract: A communications system and method is configured to provide multimedia services utilizing a signaling protocol such as a session initiated protocol (SIP), via a local access network. The method includes providing a local proxy having an internet protocol (IP) address, wherein the local proxy is integrated with the local access network. The method further includes providing a client device having a signaling protocol client, wherein the client device is coupled to a remote access network that is external to the local access network. Additionally, the method includes initiating a session by the client device accessing the IP address of the local proxy.

Abstract: Embodiments are directed processing adaptive audio content by determining an audio type as one of channel-based audio and object-based audio for each audio segment of an adaptive audio bitstream, tagging the each audio segment with a metadata definition indicating the audio type of the corresponding audio segment, processing audio segments tagged as channel-based audio in a channel audio renderer component, and processing audio segments tagged as object-based audio in an object audio renderer component that is distinct from the channel audio renderer component. Object-based audio is rendered through an object audio renderer interface that dynamically adjusts processing block sizes of the object audio segments based on timing and alignment of metadata updates and maximum/minimum block size parameters.

Abstract: The present invention pertains to an encoding device and method, a decoding device and method, and to a program, with which sound of an appropriate volume level can be obtained with a smaller quantity of codes. A first gain calculation circuit calculates a first gain for volume level correction of an input time series signal, and a second gain calculation circuit calculates a second gain for volume level correction of a downmixed signal obtained by downmixing of the input time series signal. A gain encoding circuit computes the gain differential between the first gain and the second gain, the gain differential between time frames, and the gain differential within time frames, and encodes the first gain and the second gain. The present invention can be applied in encoding devices and decoding devices.

Abstract: A mobile device can be assigned to an extension configured as a Queue Agent (“QA”) in a call-queuing enabled virtual private branch exchange (vPBX). This and other mobile or fixed devices can be interconnected to form a virtual call center. Each extension can be assigned various rules, which specify how the incoming calls are answered at the extension and how calls in the call queue are selected for answering by the QA. As part of the rules, the vPBX system forwards calls from a call queue to the QA based on QA's availability status. If a location-aware mobile device is associated with the QA, the vPBX system can determine the QA's availability status based on a geographic location of the mobile device. A most geographically proximate agent can be selected for answering the call.

Abstract: Systems and methods for determining the trustworthiness of calling party information, such as caller ID and ANI information, contained in a call request are provided. The method includes receiving a call request at a service provider network element, such as a telecommunication carrier switch. A decision is made as to whether the call request should be verified by reviewing a database of called telephone numbers for monitoring. When the call request is to be verified, a determination is made whether a discrepancy exists between the calling party information contained within the call request and authenticated stored calling party information. For example, the caller ID information in a call request is compared to service provider caller ID information for the calling party to determine if they match. When a discrepancy exists, a discrepancy report is transmitted to the called party.