2082results about "Public address systems" patented technology

Techniques for controlling zone group and zone group characteristics such as audio volume in a multi-zone system are disclosed. The multi-zone system includes a number of multimedia players, each preferably located in a zone. A controller may control the operations of all of the zone players remotely from any one of the zones. Two or more zone players may be dynamically grouped as a zone group for synchronized operations. According to one aspect of the techniques, a zone group configuration can be managed, updated, modified via an interactive user interface provided in a controlling device. The zone group configuration may be saved in one of zone players. According to another aspect of the techniques, the audio volume control of a zone group can be performed individually or synchronously as a group.

Described herein are various methods and systems relating to the unification of media devices, and more specifically to the provision of wireless audio systems. In overview, two or more wireless speaker subsystem units substantially autonomously form a single wireless audio system having its own control interface. This control interface is used to apply operational changes across the wireless audio system, such as volume adjustment. That is, an operational change may be applied to the system as a whole, and this change is subsequently implemented by each of the individual wireless speaker subsystem units.

An audio distribution network system (20) allowing an audio distribution system to be created that is integrated with the home automation system into a home network that permits vocal feedback, status and even control with the audio through network speakers (100).

Embodiments are described for a system of rendering object-based audio content through a system that includes individually addressable drivers, including at least one driver that is configured to project sound waves toward one or more surfaces within a listening environment for reflection to a listening area within the listening environment; a renderer configured to receive and process audio streams and one or more metadata sets associated with each of the audio streams and specifying a playback location of a respective audio stream; and a playback system coupled to the renderer and configured to render the audio streams to a plurality of audio feeds corresponding to the array of audio drivers in accordance with the one or more metadata sets.

In a multi-speaker audio system for, e.g., a home entertainment system or other entertainment system, each networked-speaker (wired or wireless) is activated or deactivated as appropriate such that audio play follows a user as the user moves around the home.

A user interface 103 for an electronic device 101 (and corresponding method) is arranged and constructed for intuitive control of interface functionality. The user interface includes: a user interface component, e.g. speaker, microphone, display backlighting, etc., that is one of a plurality of user interface components 105; interface circuitry 117 coupled to the user interface components 105; and a sensor 129 located in a position that is logically associated with, e.g. proximate to or co-located with, the user interface component and configured to provide an output signal when the sensor 129 is triggered, e.g. by proximity to a user, where the output signal facilitates, via for example a controller 143, a change in an operating mode of one or more of the user interface components 105.

A processor operates on samples of a digital output signal to determine samples of a digital correction signal. The output signal samples are directed to an output channel for transmission from a speaker. The digital correction signal samples are supplied to a first digital-to-analog converter for conversion into an analog correction signal. The subtraction circuit generates a difference between a first analog signal provided by a microphone and the analog correction signal. The analog correction signal is an estimate of a contribution to the first analog signal due to a direct path transmission between the speaker and the microphone. The processor also receives a digital input signal derived from the difference signal, and, performs acoustic echo cancellation on the digital input signal to obtain a resultant signal. The acoustic echo cancellation is configured to remove contributions to the digital input signal due to reflected path transmissions.

This invention relates to a wireless digital transmission system for loudspeakers comprising:compression means for the file representing the digital audio signal of the “compact disc” type, a transmission device comprising means of converting this compressed signal into a series signal moving by packets going to a modulator circuit with phase quadrature and means of transmitting the signals exiting the modulator circuit with phase quadrature to the domestic network for feeding electricity;a receiving device comprising means of connecting to this domestic network and of extracting from the feed electrical signal, by a demodulator with phase quadrature, data packets moving the digital audio signal to convert it into a parallelized digital signal sent to a decompression circuit;means of converting the decompressed digital signals into an analog signal intended to feed a loudspeaker after adequate amplification.

Electronic apparatus and method are provided for management of communications between channel master (e.g. audio player), responding device (e.g. remote control / headphones) and, optionally, passive devices (e.g. headphones) wirelessly connected in a wireless audio network. The channel master transmits, and the responding device receives, audio signals and network management information. Profile negotiation means communicates application profiles for the devices, establishing at least one application profile which is common to them and designating a selected, common application profile for use in associating the devices. Channel selection means selects a channel for the communications using an ordered selection of channels (PCS) obtained from scanning the channels for idle channels and ordering the scanned channels according to priority based on channel quality. Normalized control and status interfaces establish interoperability between devices which use different data streams to communicate in the network.

Output control system and method includes calculating gain for an input signal, detecting a predetermined condition, and modifying the gain calculation in accordance with one or more kneepoints to provide a varying output signal. A companding ratio is then established in accordance with the kneepoints. System gain is then adjusted, and the companding ratio may be further adjusted by adjusting one of the kneepoints relative to another.

A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, filters, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions. The system also provides a methodology for selecting loudspeaker locations, the number of loudspeakers, the types of loudspeakers, correction factors, listening positions, crossover filters or a combination of these schemes in an audio system that has a single listening position or multiple listening positions.

Intelligibility of a human voice projected by a loudspeaker in an environment of high ambient noise is enhanced by processing a voice signal in accordance with the frequency response characteristics of the human hearing system. Intelligibility of the human voice is derived largely from the pattern of frequency distribution of voice sounds, such as formants, as perceived by the human hearing system. Intelligibility of speech in a voice signal is enhanced by filtering and expanding the voice signal with a transfer function that approximates an inverse of equal loudness contours for tones in a frontal sound field for humans of average hearing acuity.

A method and apparatus are disclosed to permit real time, distributed acoustic performance by multiple musicians at remote locations. The latency of the communication channel is reflected in the audio monitor used by the performer. This allows a natural accommodation to be made by the musician. Simultaneous remote acoustic performances are played together at each location, though not necessarily simultaneously at all locations. This allows locations having low latency connections to retain some of their advantage. The amount of induced latency can be overridden by each musician. The method preferably employs a CODEC able to aesthetically synthesize packets missing from the audio stream in real time.

The present invention extends to methods, devices, systems, and computer program products for managing audio output through an intermediary. In some embodiments, an audio controller emulates a direct connection between an audio source device and an audio output device. In other embodiments, audio content local to an audio controller is combined with other audio content passing through the audio controller on its way from an audio source device to an audio output device. In additional embodiments, an audio output device is locked to an audio controller.

A control device and a corresponding control method are provided for establishing an information-output ranking of a plurality of information sources, in particular audio sources. In order to output the information of the information sources to a common information-output device, the information-output ranking is established in pairs for the audio sources which is in the form of an audio-output matrix having a nonlinear order with respect to the matrix elements, and wherein each matrix element of the information-output matrix is used to determine the priority of a corresponding information source with respect to another information source. A conflict among competing information outputs is solved in this manner.

A sound masking system for shaping the ambient noise level in a physical environment. The sound masking system comprises a networked and distributed system having a number of master units coupled together and to a control unit. One or more of the master units may include satellite sound masking units which function to reproduce the sound masking signal generated by the master sound masking unit. Each of the master units is addressable over the network by the control unit enabling the control unit to program the contour, spectral band, and gain characteristics of the sound masking output signal. The system may also include a remote control unit which provides the capability to tune and adjust each master sound masking unit in situ without requiring physical access through the ceiling installation.

The present publication describes a data transfer method and system in a digital sound reproduction system. The method comprises method steps for generating a digital audio stream for multiple channels in a host data source, e.g. a computer, the audio stream is formed by multiple consecutive samples, receiving the digital audio stream sent by the host data source through a digital data transmission network by several digital receivers each of which including a microcontroller with a clock, the receivers further including means for generating an audio signal. In accordance with the invention the host data source sends repeatedly a synchronization sample to at least one receiver, the receiver replies to the synchronization sample by a return sample, the host calculates a latency (T) for each receiver based on the sending time (Th1) of the synchronization sample and the reception time (Th2) of the return sample and the processing time (Tt1-Tt2) of the receiver, the host sends to the receiver information of the calculated latency (T) in combination with the time stamp the measurement time, based on this information the receiver adjusts the function of its clock, and the above synchronization steps are repeated continuously.

A pointing element enhanced speaker system addresses the need for consistent sound. Despite wide variations in the design and architecture of different venues, the system helps performers ensure that they deliver the desired sound for their audiences. Performers and their technicians, though faced with grueling schedules that impose severe time constraints on equipment setup and tuning as the performers move between venues, may turn to the system to provide the sound desired at each new venue.

A system for providing a listener with an augmented audio reality in a geographical environment said system comprising a position locating system for determining a current position and orientation of a listener in. said geographical environment; an audio track creation system for creating an audio track having a predetermined spatialization component dependent on an apparent location of an apparent source associated with said audio track in said geographical environment; an audio track rendering system adapted to render an audio signal based on said audio track to a series of speakers surrounding said listener such that said listener experiences an apparent preservation of said spatialization component; and an audio track playback system interconnected to said position locating system and said audio track creation system and adapted to forward a predetermined audio track to said audio rendering system for rendering depending on said current position and orientation of said listener in said geographical environment.

Method and apparatus for producing, combining, and customizing virtual sound environments. A binaural sound system (400) includes a transceiver (492) configured for receiving a signal (600) containing at least a first type of information and a second type of information. The first type of information includes enunciated data (602). The enunciated data specifies certain information intended to be audibly enunciated to a user. The second type of information comprises first type of metadata (604-1) and a second type of metadata (604-2). The first type of metadata includes information which identifies a characteristic of the enunciated data exclusive of spatial position information. The second type of metadata identifies a spatial position information associated with the enunciated data.

An audio control system may receive a plurality of audio input signals. A switch or multiplexer may switch one of the plurality of audio input signals from respective audio sources to a power amplifier. A controller in communication with the multiplexer may control switching of the multiplexer. A display may provide the user with information regarding the switching of the audio signal inputs. A user interface may permit the user to manually switch the audio source. A signal detector in communication with an audio input signal may detect a signal from a priority source. When the signal from the priority source is detected, the controller may direct the multiplexer to route the signal from the priority source to the power amplifier.

A sound masking system and method for providing sound masking in networked workstations or offices. The sound masking system comprises a communication network, a plurality of sound masking devices adapted to emit a sound masking signal, and a remote controller. One or more of the sound masking devices includes a controller, and the controller includes an interface for receiving information from the communication network and a component for controlling or adjusting output characteristics associated with the sound masking signal, such as volume. The controller includes a component responsive to a user input for setting or varying output characteristics associated with the sound masking signal.

An apparatus and method for providing a centralized speaker system that allows multiple speakers connected to a central amplifier speaker line to be monitored and controlled from a central location via a master / slave protocol. The centralized speaker system comprises a central station for selectively communicating at least one of a command and an information signal to a destination device. A tone generator is adapted to communicate an activation tone to the destination device. An amplifier, which is colocated with the central station, is adapted to amplify the signals to the destination device.

A repetitive motion pacing system for pacing a user that comprises a user profile database that contains a plurality of user-defined parameters that include at least a pre-selected interval type, a pre-selected interval profile, and a target tempo value for a repetitive motion activity. A data storage and playback device is programmed to identify tempo information of one or more data files and matching or closely matching the tempo of the one or more data files with the target tempo value. A communications device for transferring the matched or closely matched one or more data files to the user.