Abstract: An earplug connector for simultaneously providing an audio signal from a vehicle or an extra-vehicular source to a pilot while providing power to a noise reduction device while operating the vehicle. The earplug connector has a multi-pin connector for receiving audio signal and power from a pilot interface assembly connected to the vehicle. A ground, power, and audio signal connector transfers power to a noise reduction device worn by a pilot. An audio output continuously transfers audio signals from the multi-pin connector to earplugs and a multi-pin signal separator is in electrical and audio communication between the multi-pin connector and the ground, power, and audio signal connector. The audio output receives audio signals from the multi-pin connector and selectively passes the audio signals to the earplugs while transferring power to a noise reduction device while accepting a plurality of pin configurations.

Abstract: An earphone wire control circuit and wire control method for a mobile terminal. The wire control circuit includes a circuit in the earphone and a circuit in the mobile terminal which are coupled via a signal wire. The circuit in the earphone includes a micro controller unit and an attenuation and microphone circuit coupled to the micro controller unit of the earphone. The circuit in the mobile terminal includes an audio analog input channel and an amplifier which are coupled in parallel. The amplifier is coupled to a micro controller unit of the mobile terminal via a wire. The micro controller unit of the mobile terminal is coupled to a central processing unit of the mobile terminal via a wire. The micro controller unit of the earphone includes a button detection module and a pulse-width modulation module.

Abstract: Provided is a loudspeaker system capable of easily determining control parameters for controlling the directional characteristics of a speaker set including two or more speaker units. The loudspeaker system is configured to include: a speaker set 11 including two or more speaker units 2; a camera 15 adapted to photograph an acoustic space of the speaker set 11; a monitor 22 adapted to display a camera image 30 photographed by the camera 15; and a directional control parameter generating part 24 adapted to generate directional control parameters providing the directional characteristics of the speaker set 11.

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: An acoustic noise canceling microphone arrangement and processor that uses a principal microphone and other microphones that may be incidentally or deliberately located in the vicinity of the principal microphone in order to derive an audio signal of enhanced signal-to-background noise ratio. In one implementation, the principal and incidental microphones comprise the microphone built into a mobile phone and the microphone built into a Bluetooth headset.

Abstract: An audio reproduction method and apparatus, in which an audio volume is automatically controlled based on audio energy and human auditory characteristics. The audio reproduction method includes splitting a reproduction audio signal into audio signals corresponding to a plurality of frequency bands, extracting audio energy for each of the frequency bands, and comparing the audio energy for each of the frequency bands with a predetermined threshold and controlling the volume of the audio signal corresponding to each of the frequency bands.

Abstract: An exemplary audio signal processing system includes a modal decomposer and an adaptive modal beamformer. The modal decomposer generates a plurality of zeroth-order eigenbeams from audio signals from an (e.g., spherical) array of audio sensors. The adaptive modal beamformer (i) steers the zeroth-order eigenbeams to a specified direction, (ii) adaptively generates a plurality of weighting coefficients for the plurality of zeroth-order eigenbeams, where the plurality of weighting coefficients satisfy a constraint of having only non-negative values, (iii) respectively applies the plurality of adaptively generated weighting coefficients to the plurality of steered, zeroth-order eigenbeams to generate a plurality of weighted, steered, zeroth-order eigenbeams, and (iv) combines the plurality of weighted, steered, zeroth-order eigenbeams to generate an output audio signal. Some embodiments have a further constraint that the weighting coefficients sum to a specified value (e.g., one).

Abstract: The processing efficiency of a process used to decode frames of an enhanced AC-3 bit stream is improved by processing each audio block in a frame only once. Audio blocks of encoded data are decoded in block order rather than in channel order. Exemplary decoding processes for enhanced bit stream coding features such as adaptive hybrid transform processing and spectral extension are disclosed.

Abstract: Disclosed is a sound generating device for vehicle, more particularly, in case of fastening a sound generating device for outputting artificial engine sound (for example, start-up, or accelerating etc.) onto the automobile vehicle's surface, for example the bonnet, the sound generating device's sound transfer face is pressed to the bonnet, and the sound or vibration of the sound generator is dispersed and output outwardly through the bonnet without attenuation.

Abstract: In a method for use in Interactive Voice Response (IVR) systems, an IVR is configured such that voice prompts provided to a caller vary systematically and caller hang-up activity is attributed to the particular version of a prompt played to the caller. Voice prompt modifications are chosen based on how long the caller willingly listens to the voice prompt. A relatively short time listening to a prompt before hang-up, indicates dissatisfaction with that prompt. The system compares caller hang-up rates for each of the phrases in a prompt and chooses the optimum solution, which is the variant of the prompt with the longest caller listening time. Optionally, the system compares hang up rates to a threshold and contingent on that comparison chooses an alternative version of the prompt.

Abstract: A loudspeaker port may include tunable physical components to tune the port to different frequencies to improve speaker efficiency at those frequencies. The ports may be activated by at least partly opening associated shutters, or disabled by closing the associated shutters. Activated ports may enhance speaker efficiency in a frequency range. However, activated ports may also introduce sound artifacts, thereby reducing sound quality. Therefore, the ports may be disabled when appropriate to reduce their negative impact to sound quality. A Digital Signal Processor (DSP) may determine the frequency components of a played sound to determine when to open the ports and how to tune the ports. Accordingly, a loudspeaker may benefit from the improved efficiency facilitated by the ports while also avoiding typical drawbacks created by the ports.

Abstract: An optical sensor may be integrated into headphones and feedback from the sensor used to adjust an audio output from the headphones. For example, an emergency vehicle traffic preemption signal may be detected by the optical sensor. Optical signals may be processed in a pattern discriminator, which may be integrated with an audio controller integrated circuit (IC). When the signal is detected, the playback of music through the headphones may be muted and/or a noise cancellation function turned off. The optical sensor may be integrated in a music player, a smart phone, a tablet, a cord-mounted module, or the earpieces of the headphones.

Abstract: A hands-free audio system for a vehicle and method of using the system is described. The method includes controlling the directionality of a vehicle microphone. The steps of the method may include: (a) receiving a sensor value from at least one of a vehicle seat position sensor, a vehicle seat orientation sensor, or a vehicle mirror orientation sensor; (b) based on the received sensor value(s) of step (a), determining an origin of a vehicle user's speech; and (c) controlling the microphone sensitivity directionality based on the determined origin.

Abstract: Surrounding awareness is provided using sound recognition by detecting sound, determining frequency components of the detected sound and comparing the frequency components of the detected sound to the frequency components of known sound sources. If the components of a known sound source are detected, the source of the sound is determined to be present. An appropriate warning or enunciation of the sound source is provided.

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: A signal processing method and system are provided for delivering spatialized sound using highly optimized inverse filters to deliver narrow localized beams of sound from the included speaker array. The inventive method can be used to provide private listening areas in a public space and provide spatialization of source material for individual users to create a virtual 3D audio effect. In a binaural mode, a speaker array provides two targeted beams aimed towards the primary user's ears—one discrete beam for the left ear and one discrete beam for the right ear.

Abstract: A waterproof sound-transmitting membrane (1) of the present invention includes a porous membrane (11) containing polytetrafluoroethylene as a main component and a supporting member (12) that supports the porous membrane (11). The supporting member (12) is joined to the porous membrane (11) placed thereon and is not subjected to stretching after the porous membrane (11) is placed thereon. The supporting member (12) is a mesh or a net having a fiber diameter of 45 to 60 ?m and an opening size of 700 to 1100 ?m across the supporting member (12). The waterproof sound-transmitting membrane (1) has a water entry pressure of 20 kPa or more as measured according to the method B (high hydraulic pressure method) of JIS L 1092.

Abstract: Methods and systems to facilitate real time communications via a telephonic apparatuses that support data communications. In one embodiment, a telephonic apparatus includes: a network interface to submit a query over a data communication network and to receive a listing with embedded information about the query; and a call module coupled to the network interface, the call module to embed the information about the query in a call signaling message for a telephonic connection according to the listing.

Abstract: A modified end user network access point includes a switching module and a controller to track usage of the end user network access point by wirelessly connected end user devices. The tracked usage information is transmitted from the end user network access point to a remotely coupled data center. The data center collects usage information from a plurality of different end user network access points. The collected usage information is in turn transmitted from the data center to a network carrier. The network carrier uses the received usage information to bill users for accessing the end user network access points and using any services provided by the end user network access points. An owner of each end user network access point is compensated by the network carrier each instance that an end user device utilizes the end user network access point to access the network carrier's communications network.

Abstract: A method is provided for supplementing existing interaction routines in a contact center with added capability including acts for (a) creating at least one rule having at least one rule attribute; (b) creating one or more processes, the processes integral to the rule; (c) defining the at least one rule and integral processes as a strategy; and (d) linking the strategy to the interaction routine, the link serving to cause execution of the strategy during an interaction between an entity and the routine, execution thereof promoting further interaction defined in the strategy.