1248results about "Pseudo-stereo systems" patented technology

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.

A listener can experience the sound of virtual loudspeakers over headphones with a level of realism that is difficult to distinguish from the real loudspeaker experience. Sets of personalized room impulse responses (PRIRs) are acquired for the loudspeaker sound sources over a limited number of listener head positions. The PRIRs are then used to transform an audio signal for the loudspeakers into a virtualized output for the headphones. Basing the transformation on the listener's head position, the system can adjust the transformation so that the virtual loudspeakers appear not to move as the listener moves the head.

A spatial processing stereo system (“SPSS”) that receives audio signals and a limited number of user input parameters associated with the spatial attributes of a room, such as “room size”, “stage distance”, and “stage width”. The input parameters are used to define a listening room and generate coefficients, room impulse responses, and scaling factors that are used generate additional surround signals.

Separating a source in a stereo signal having a left channel and a right channel includes transforming the signal into a short-time transform domain; computing a short-time similarity measure between the left channel and the right channel; classifying portions of the signals having similar panning coefficients according to the short-time similarity measure; segregating a selected one of the classified portions of the signals corresponding to the source; and reconstructing the source from the selected portions of the signals.

A scheme for stereo and multi-channel synthesis of inter-channel correlation (ICC) (normalized cross-correlation) cues for parametric stereo and multi-channel coding. The scheme synthesizes ICC cues such that they approximate those of the original. For that purpose, diffuse audio channels are generated and mixed with the transmitted combined (e.g., sum) signal(s). The diffuse audio channels are preferably generated using relatively long filters with exponentially decaying Gaussian impulse responses. Such impulse responses generate diffuse sound similar to late reverberation. An alternative implementation for reduced computational complexity is proposed, where inter-channel level difference (ICLD), inter-channel time difference (ICTD), and ICC synthesis are all carried out in the domain of a single short-time Fourier transform (STFT), including the filtering for diffuse sound generation.

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.

The present invention relates to a method for selecting auditory signal components for reproduction by means of one or more supplementary sound reproducing transducers, such as loudspeakers, placed between a pair of primary sound reproducing transducers, such as left and right loudspeakers in a stereophonic loudspeaker setup or adjacent loudspeakers in a surround sound loudspeaker setup, the method comprising the steps of (i) specifying azimuth angle range within which one of said supplementary sound reproducing transducers is located or is to be located and a listening direction; (Ii) based on said azimuth angle range and said listening direction, determining left and right interaural level difference limits and left and right interaural time difference limits, respectively; (iii) providing a pair of input signals for said pair of primary sound reproducing transducers; (iv) pre-processing each of said input signals, thereby providing a pair of pre-processed input signals; (v) determining interaural level difference and interaural time difference as a function of frequency between said pre-processed signals; and (vi) providing those signal components of said input signals that have interauial level differences and interaural time differences in the interval between said left and right interaural level difference limits, and left and right interaural time difference limits, respectively, to the corresponding supplementary sound reproducing transducer. The invention also relates to a device for carrying out the above method and systems of such devices.

A multi-dimensional audio processor receives as an input either a single channel signal or a two channel signal from an audio signal source; for example a musical instrument or an audio mixer. The processor is programmable to divide the input among at least 3 output channels in a user-defined manner. The processor is also user programmable to provide a variety of effect and mixing functions for the output channel signals.

The present invention describes techniques that can be used to provide novel methods of spatial audio rendering using adapted M-S matrix shuffler topologies. Such techniques include headphone and loudspeaker-based binaural signal simulation and rendering, stereo expansion, multichannel upmix and pseudo multichannel surround rendering.

The envelope of a decorrelated signal derived from an original signal can be shaped without introducing additional distortion, when a spectral flattener is used to spectrally flatten the spectrum of the decorrelated signal and the original signal prior to using the flattened spectra for deriving a gain factor describing the energy distribution between the flattened spectra, and when the so derived gain factor is used by an envelope shaper to timely shape the envelope of the decorrelated signal.

Methods and apparatus are disclosed for processing an audio sound source to create four-dimensional spatialized sound. A virtual sound source may be moved along a path in three-dimensional space over a specified time period to achieve four-dimensional sound localization. The various embodiments described herein provide methods and systems for converting existing mono, 2-channel and / or multi-channel audio signals into spatialized audio signals have two or more audio channels. The incoming audio signals may be down-mixed, up-mixed or otherwise translated into fewer, greater or the same number of audio channels. The various embodiments also describe methods, systems and apparatus for generating low frequency effect and center channel signals from incoming audio signals having one or more channels.

In some embodiments, spatially realistic audio may be provided for a conference call. Voices from participants on the left side of a display, in a conference call, may be directed through audio on the left side of the display at the other conferencing system in the conference call (similarly for voices from the center and right side of the display). In some embodiments, two speakers may be used in the system to create synthesized stereo sound at a location specified by directional information received as side information along with the existing audio channel. The location may be determined by using beamforming with integrated microphones on a camera or speakerphone. In some embodiments, the audio signal and directional information may be sent in the form of a left audio channel and a right audio channel.

Techniques and systems for enhancing delivered audio signals are disclosed which may be employed in a delivery system at a server side, a client side, or both. The techniques include forming a processed audio signal by processing audio signals through multiple pathways which operate on different frequency bands using dynamic processing and other elements, and thereafter providing recording or listening environment enhancements and other sound enhancements to the processed audio signal. Also disclosed are techniques and systems for implementing the multi-pathway processing and environmental and sound enhancements.

An audio signal processing system is configured to separate an audio signal into a dry signal component and one or more reverberant signal components. The dry signal component and the reverberant signal components can be separately modified and then recombined to form a processed audio signal. Alternatively, the dry signal component may be combined with an artificial reverberation component to form the processed audio signal. Modification of the reverberation signal component and generation of the artificial reverberation component may be performed in order to modify the acoustic characteristics of an acoustic space in which the audio signal is driving loudspeakers. The audio signal may be a pre-recorded audio signal or a live audio signal generated inside or outside the acoustic space.

Disclosed is a method and system for synchronizing environmental modifiers and actuators with the video and audio components of television programs to produce sensual representations or simulations of suggested environment or actions. The system includes a central device in the viewing household that receives the control data for any and / or all of the environmental modifiers (actuators). This device may be built into the television or associated components or may be entirely independent. The device receives the control signals from any available source and sends the proper control data or signals to the associated peripherals or components.

An automatic sound field correcting device executes a signal process to the plurality of audio signals on respective correspondent signal transmission paths, and outputs them to a plurality of correspondent speakers to correct sound characteristics on the respective signal transmission paths. Namely, a measurement signal is supplied to each signal transmission path, and a measurement sound corresponding to it is outputted from the speaker to a sound space. The outputted measurement sound is detected as a detecting signal. The frequency characteristic of the audio signal on each signal transmission path is corrected by an equalizer, and a gain value of the equalizer is determined by a correction amount determining unit. A frequency characteristics correction is performed predetermined times. At a first correction, the correction amount determining unit determines the correction amount by performing a frequency analysis, based on the detecting signal, i.e. base on the detecting signal corresponding to the measurement sound actually outputted to the sound space. On the contrary, at and after a second correction, the correction amount determining unit determines the correction amount based on the detecting signal or an output signal of the equalizer. Namely, at and after the second correction, the output signal of the equalizer is supplied to the correction amount determining unit in a signal processing circuit as the need arises, and the frequency characteristics correction is performed without actually outputting the measurement sound to the sound space.

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.

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 device and an audio processing method are provided for adjusting the position of a virtual speaker. The audio device comprises a decoder which has audio data provided thereto, the audio data including an audio component for a center speaker and a plurality of audio components corresponding to other speakers disposed with the center speaker interposed therewith, and which decodes these audio components to separate them from the audio data, a center delay processor for delaying the audio component for the center speaker received from the decoder, and a downmixing processor for distributing the delayed center speaker audio component between the other speakers and for merging the audio component distributed to each of the other speakers and the original audio component for each other speaker. Audio sounds corresponding to the downmixed audio components are produced from the other speakers.

A method and an apparatus to reproduce a multi-channel audio signal, in which mixing of a center channel signal is performed with a center channel signal in a home theater system. The method of reproducing a multi-channel audio signal includes calculating a delay value of a center channel signal according to location relationships of a listener, a center channel speaker and other channel speakers, regulating a time delay of the center channel signal according to the calculated delay value, and mixing the time-delay regulated center channel signal with other channel signals.

An augmented reality sound systems is disclosed. An augmented reality sound system includes a at least one microphone for receiving ambient sound and a memory storing one or more augmented reality sound profiles and a respective set of processing parameters. The system further includes a processor coupled to the memory and configured to generate augmented ambient sound from the ambient sound by reproducing the ambient sound in conjunction with processed sound superimposed over the ambient sound as directed by one or more of the set of processing parameters retrieved from the memory based on a selected augmented reality sound profile.

A method for assigning audio channels of audio data content to speakers, each of which is associated with a different one of N speaker nodes of a wireless network. The method is performed by a manager node of the wireless network. The method includes, for each of the N speaker nodes, obtaining a piece of orientation information for each of N−1 other speaker nodes different from the given speaker node, regarding an orientation of a reception antenna of the given speaker node when receiving data sent by each of the N−1 other speaker nodes through the wireless network. The method also includes assigning a different one of the audio channels to a different one of the speaker nodes, as a function of N−1 pieces of orientation information obtained for each of the N speaker nodes.

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.

A system and method for producing a binaural and localized audio signal to a user is provided. A signal processing method is provided for delivering spatialized sound in various ways 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 a single user to create a virtual surround sound 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. In a privacy mode, a privacy zone could be created in which a primary audio beam would deliver a signal of interest to the user while secondary beams would be aimed at different angles to provide a masking noise.

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.

Embodiments of the disclosure can include systems, methods, and apparatus for assigning three-dimensional spatial data to sounds and audio files. In one embodiment, a method can include receiving at least one audio signal, receiving sonic spatial data, associating the sonic spatial data with the at least one audio signal, associating the at least one audio signal and sonic spatial data with a time code, and storing the sonic spatial data, the at least one audio signal, and time code in an encoded sound file.

An auditory scene is synthesized from a mono audio signal by modifying, for each critical band, an auditory scene parameter (e.g., an inter-aural level difference (ILD) and / or an inter-aural time difference (ITD)) for each sub-band within the critical band, where the modification is based on an average estimated coherence for the critical band. The coherence-based modification produces auditory scenes having objects whose widths more accurately match the widths of the objects in the original input auditory scene.