3171results about "Loudspeaker signals distribution" patented technology

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.

Display apparatus including a display and an orientation sensitive interface mechanism is disclosed. In an exemplary embodiment, the orientation sensitive interface includes first and second loudspeaker pairs. The first loudspeaker pair includes first and second loudspeakers and the second loudspeaker pair includes the second and third loudspeaker. The first and second loudspeaker pairs are disposed along transverse directions to each other. The display apparatus comprises a switch which switches between the first loudspeaker pair and the second loudspeaker pair. By providing the respective loudspeaker pairs, and switching between them, it is possible to orient the display apparatus in transverse directions corresponding to respective loudspeaker pairs, yet maintain a substantially stereophonic reproduction for each orientation.

Described herein are systems and methods for providing media playback in a networked environment. In one embodiment, a networked media playback device is configured to provide a web server for delivering data indicative of a browser-renderable control interface for the networked media playback device, or for a related networked media playback system. In overview, the user of a networked device, such as a wireless web enabled device, is able to render the control interface in a web browser, and in this manner control playback of digital media via the networked media playback device or networked media playback system.

A wireless speaker system configured to receive stereo audio information wirelessly transmitted by an audio source including first and second loudspeakers. The first loudspeaker establishes a bidirectional secondary wireless link with the audio source for receiving and acknowledging receipt of the stereo audio information. The first and second loudspeakers communicate with each other via a primary wireless link, and the first and second loudspeakers are configured to extract first and second audio channels, respectively, from the stereo audio information. A wireless audio system including an audio source and first and second loudspeakers, each having a wireless transceiver. The first and second loudspeakers communicate via a primary wireless link. The audio source communicates audio information to the first loudspeaker via a secondary wireless link which is configured according to a standard wireless protocol. The first loudspeaker is configured to acknowledge successful reception of audio information via the secondary wireless link.

A sound produced at the location of a listener is captured by a microphone in each of a plurality of speaker devices. A sever apparatus receives an audio signal of the captured sound from all speaker devices, and calculates a distance difference between the distance of the location of the listener to the speaker device closest to the listener and the distance of the listener to each of the plurality of speaker devices. When one of the speaker devices emits a sound, the server apparatus receives an audio signal of the sound captured by and transmitted from each of the other speaker devices. The server apparatus calculates a speaker-to-speaker distance between the speaker device that has emitted the sound and each of the other speaker devices. The server apparatus calculates a layout configuration of the plurality of speaker devices based on the distance difference and the speaker-to-speaker distance.

One or more acoustic transducers of a device may be adjusted based on automatic detection of device proximity to the user. In a mobile telephone, when the user is using the receiver and holding the telephone against his / her ear, if the telephone detects that the user has moved the telephone further from his / her ear, the telephone will raise the receiver volume. Similarly, if the user is using the speaker, the telephone will adjust the speaker volume as user distance from the telephone changes. In another embodiment the telephone may fade between the receiver and the speaker. Volume is not the only acoustic property that could be adjusted according to user proximity. Frequency response is another property that could be adjusted, such as using appropriate electronic filtering, or by turning on another transducer that is not otherwise being used.

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.

An audio content playback method for a portable terminal The audio content playback method includes checking a channel that is supportable by audio content that is currently engaged in group's simultaneous playback, in group's simultaneous playback of the audio content. The method includes allocating a channel to each of devices included in a group based on position information of each device included in the group or based on an input state in a user interface environment that is preset for channel allocation for each device included in the group, and transmitting the allocated channel information to each device included in the group to allow the device to select its allocated channel and play the audio content.

A sound produced at the location of a listener is captured by a microphone in each of a plurality of speaker devices. A sever apparatus receives an audio signal of the captured sound from all speaker devices, and calculates a distance difference between the distance of the location of the listener to the speaker device closest to the listener and the distance of the listener to each of the plurality of speaker devices. When one of the speaker devices emits a sound, the server apparatus receives an audio signal of the sound captured by and transmitted from each of the other speaker devices. The server apparatus calculates a speaker-to-speaker distance between the speaker device that has emitted the sound and each of the other speaker devices. The server apparatus calculates a layout configuration of the plurality of speaker devices based on the distance difference and the speaker-to-speaker distance.

An audio system installed in a listening space may include a signal processor and a plurality of loudspeakers. The audio system may be tuned with an automated audio tuning system to optimize the sound output of the loudspeakers within the listening space. The automated audio tuning system may provide automated processing to determine at least one of a plurality of settings, such as channel equalization settings, delay settings, gain settings, crossover settings, bass optimization settings and group equalization settings. The settings may be generated by the automated audio tuning system based on an audio response produced by the loudspeakers in the audio system. The automated tuning system may generate simulations of the application of settings to the audio response to optimize tuning.

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.

A system for configuring an audio / video (AV) system includes a computing device having a processor to enable a user, with a user interface of the computing device, to configure the AV system. The processor may enable the user to draw output zones to create a representation of a geographical layout of a venue in which the AV system is located. The processor may enable the user to place transmitting and receiving AV devices within the output zones that substantially represent physical locations thereof within the venue. The processor may enable the user to logically associate receiving AV device channels of one or more receiving AV devices within an output zone. The processor may also enable the user to select which of a number of source AV signals from transmitting AV devices the user wants to route to the associated output zone. The processor routes the source AV signals from one or more transmitting AV devices over a network to the one or more receiving AV device channels, such that the one or more receiving AV device channels are processed according to the output zone as configured by the user in the user interface.

A neural network is provided for recognition and enhancement of multi-channel sound signals received by multiple microphones, which need not be aligned in a linear array in a given environment. Directions and distances of sound sources may also be detected by the neural network without the need for a beamformer connected to the microphones. The neural network may be trained by knowledge gained from free-field array impulse responses obtained in an anechoic chamber, array impulse responses that model simulated environments of different reverberation times, and array impulse responses obtained in actual environments.

A speaker array and microphone arrays positioned on both sides of the speaker array are provided. A plurality of focal points each serving as a position of a talker are set in front of the microphone arrays respectively symmetrically with respect to a centerline of the speaker array, and a bundle of sound collecting beams is output toward the focal points. Difference values between sound collecting beams directed toward the focal points that are symmetrical with respect to the centerline are calculated to cancel sound components that detour from the speaker array to microphones. Then, it is estimated based on totals of squares of peak values of the difference values for a particular time period that the position of the talker is close to which one of the focal points, and the position of the talker is decided by comparing the totals of the squares of the peak values of the sound collecting beams directed to the focal points that are symmetrical mutually.

An electronic apparatus is provided. The electronic apparatus includes a sensor to sense an orientation of the electronic apparatus, an audio processor to divide a stereo audio signal into a left audio signal and a right audio signal, a plurality of speakers disposed at separate locations on a main body of the electronic apparatus to output the left audio signal and the right audio signal, a switching unit to output the left audio signal and the right audio signal from the audio processor to the plurality of speakers, and a controller to control the switching unit to switch the output of the left audio signal and the right audio signal between the plurality of speakers based on a change in the orientation of the electronic apparatus.

An acoustic testing system includes at least four control microphones, at least four acoustic transducers, an acoustic enclosure with pre-determined reverberant characteristics which contains the at least four control microphones and the at least four acoustic transducers, a control system configured to produce a predetermined acoustic field as measured by the at least one control microphone. A unit under test is also disposed within the acoustic enclosure. Using an acoustic enclosure with pre-determined reverberant characteristics results of the increased proportion of reflected sounds in the area proximate to the unit under test such that less power is required to achieve a given acoustic test level than in a purely direct field acoustic test.

An electronic apparatus includes at least two internal speakers; a communicator configured to communicate with at least one external speaker; and a processor configured to, in response to a multi-channel audio signal being received, respectively assign different channel signals included in the multi-channel audio signal to the at least two internal speakers and the at least one external speaker so that the different channel signals are output through the at least two internal speakers and the at least one external speaker.

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 multi-channel audio system is constituted using at least one line array speaker unit, in which plural speakers are arrayed in line, wherein the same audio signal is supplied with a prescribed delay time to each of the speakers, thus forming plural sound beams. The plural sound beams are reflected on a wall surface and a ceiling of a room so as to form plural virtual sound sources surrounding a listening position, and emission directions and intensities of the sound beams are controlled so as to localize a phantom at a prescribed position based on the plural virtual sound sources. By appropriately arranging plural line array speaker units horizontally, vertically, and slantingly in such a way that each line array speaker unit forms sound beams distributed and spread in a sectorial form, it is possible to realize a surround audio system having a high degree of freedom with regard to setup positions for forming virtual sound sources.

An audio reproduction system includes a plurality of mobile devices, such as mobiles phones. Each of the mobile devices includes a loudspeaker, memory for storing audio data, and instructions, stored in memory, for prompting the plurality of mobile devices to each output at least a portion of an audio data file to the loudspeaker of the respective mobile device. A processor is configured for executing the instructions, thereby enabling contemporaneous play of the audio data file by the plurality of mobile devices.

A user is guided through various setup routines to optimize speaker parameters and / or positions and / or frequency assignations for the particular space in which the speaker system is located and intended to be used. This can be done using an application downloaded from a cloud server to a smart phone or tablet computer, which is then employed by the user to optimize speaker configurations for various speaker locations in the room.

A loudspeaker system is described including an array of electro-acoustic transducers capable of generating steerable beams of sound and additional transducers adapted to reproduce low frequency sound being placed at the perimeter of said array.

An apparatus and method of reproducing a 2-channel virtual sound while dynamically controlling a sweet spot and crosstalk cancellation are disclosed. The method includes: receiving broadband signals, setting compensation filter coefficients according to response characteristics of bands and setting stereophonic transfer functions according to spectrum analysis; down mixing an input multi-channel signal into two channel signals by adding head related transfer functions (HRTFs) measured in a near-field and a far-field to the input multi-channel signal, canceling crosstalk of the down mixed signals on the basis of compensation filter coefficients calculated using the set stereophonic transfer functions, and compensating levels and phases of the crosstalk cancelled signals on the basis of the set compensation filter coefficients for each of the bands.

An audio apparatus includes: an audio device arranged to supply an audio signal to a speaker to play an audio; an external adapter connected to the audio device; and a mobile phone device detachably mounted to the external adapter, in which the audio device plays, via the external adapter, an audio signal from the mobile phone device, and the external adapter includes a control unit and in conjunction with mounting and detachment of the mobile phone device with respect to the external adapter, the control unit changes a state of the audio device.

A system and method for audio telepresence. The system includes a user station and a telepresence unit. The telepresence unit includes a directional microphone for capturing sounds at the remote location, and means for converting the captured sounds into a stream of data to be communicated to the user station. The user station includes means for receiving the stream of data and a plurality of speakers for recreating the sounds of the remote location. The user station and the speakers are located within an anechoic chamber where sound reflections are substantially absorbed by anechoic linings of the chamber walls. Because of the substantial lack of sound reflection within the anechoic chamber, a user within the anechoic chamber will be able to experience an aural ambience that closely resembles the sounds captured at the remote location. The user station may include microphones for capturing the user's voice, and the telepresence unit may include speakers for projecting the user's voice at the remote location. Feedback suppression, audio direction steering, and head-coding techniques may also be used to enhance the user's sense of remote presence.

An audio apparatus, an audio signal transmission method, and an audio system are provided. The audio signal transmission method includes: wirelessly receiving an audio signal from a first external device; converting the received audio signal into audio signals of multi-channels; and wirelessly transmitting an audio signal of at least one of the multi-channels to at least one of second external devices. Therefore, the audio apparatus wirelessly communicates with a plurality of external devices, and thus a user can connect an audio device to an external device without using wired cables.

Systems and methods for providing object-oriented audio are described. Audio objects can be created by associating sound sources with attributes of those sound sources, such as location, velocity, directivity, and the like. Audio objects can be used in place of or in addition to channels to distribute sound, for example, by streaming the audio objects over a network to a client device. The objects can define their locations in space with associated two or three dimensional coordinates. The objects can be adaptively streamed to the client device based on available network or client device resources. A renderer on the client device can use the attributes of the objects to determine how to render the objects. The renderer can further adapt the playback of the objects based on information about a rendering environment of the client device. Various examples of audio object creation techniques are also described.