Abstract: A method and apparatus for introducing a time-varying time delay or phase shift randomly into the individual reproduction channels of a sound recording, two in the case of binaural presentation. This emulates the temporal aspect of microphone and/or listener motion. The present invention may be applied as a unidirectional process. No preparation of the source material is required. It can be applied to any multichannel audio signal set. It can process analog or digital signals. The process may be used with headphones, loudspeakers, hearing aids or similar assistive hearing devices.

Abstract: A driving device for a sound system by loudspeaker signals, wherein the sound system has a wave field synthesis loudspeaker array and one or several supply loudspeakers arranged separate from the wave field synthesis array includes an audio input for receiving at least one audio signal from at least one sound source, a position input for receiving information on a position of the sound source, a wave field synthesis unit for calculating loudspeaker signals for the loudspeakers of the wave field synthesis loudspeaker array, and a provider for providing the loudspeaker signal for the one or the several supply loudspeakers. The driving device enables a sound system by means of which sound localization becomes possible for the audience and at the same time pleasant levels can be achieved also in the first rows of the audience.

Abstract: An acoustic control system for hospitality spaces includes a passive component (17) for producing a baseline reverberation time (RT) value within the space, and an active component (LN, MN, 15) for picking up sounds in the space (11) and reintroducing the sounds into the space such that the reintroduced sounds have an RT value and sound level capable of being adjusted and such that the resulting RT value can be adjusted to a relatively large value as compared to said baseline RT value.

Abstract: An apparatus includes a group of microphones and a surface compensator that is operatively coupled to switch logic and to a signal conditioner that provides a control channel to voice recognition logic. The surface compensator may detect surfaces in proximity to the apparatus as well as the surface's acoustic reflectivity or acoustic absorptivity and may accordingly configure the group of microphones including selecting appropriate signal conditioning and beamforming based on the surface acoustic reflectivity or acoustic absorptivity and the orientation of the apparatus. Voice recognition performance is thus improved when microphones are impeded or occluded by proximate surfaces. A group of sensors of the apparatus is used by the surface compensator to detect surfaces and surface type, and to determine apparatus orientation and motion.

Abstract: A surround sound system for reproducing a spatial sound field in a sound control region within a room having at least one sound reflective surface. The system uses multiple steerable loudspeakers located about the sound control region, each loudspeaker having a plurality of different individual directional response channels being controlled by respective speaker input signals to generate sound waves emanating from the loudspeaker with a desired overall directional response. A control unit connected drives each of the loudspeakers and has pre-configured filters based on measured acoustic transfer functions for the room for filtering the input spatial audio signals to generate the speaker input signals for all the loudspeakers to generate sound waves with coordinated overall directional responses that combine together at the sound control region in the form of either direct sound or reflected sound from the reflective surface(s) of the room to reproduce the spatial sound field.

Abstract: A method includes detecting a surface in proximity to a mobile device using sensor data and determining an acoustic reflectivity or acoustic absorptivity of the surface using the sensor data. The method may further compensate for the acoustic reflectivity or acoustic absorptivity by controlling a configurable group of microphones of the mobile device. Compensating for the surface acoustic reflectivity or acoustic absorptivity may include beamforming the outputs of the configurable group of microphones to obtain one of an omnidirectional beamform pattern or a directional beamform pattern. An apparatus that performs the method include a configurable group of microphones, a signal conditioner, and a surface compensator. The surface compensator is operative to detect a surface in proximity to the apparatus and determine a surface acoustic reflectivity or acoustic absorptivity.

Abstract: Visual information is used to alter or set an operating parameter of an audio signal processor, other than a beamformer. A digital camera captures visual information about a scene that includes a human speaker and/or a listener. The visual information is analyzed to ascertain information about acoustics of a room. A distance between the speaker and a microphone may be estimated, and this distance estimate may be used to adjust an overall gain of the system. Distances among, and locations of, the speaker, the listener, the microphone, a loudspeaker and/or a sound-reflecting surface may be estimated. These estimates may be used to estimate reverberations within the room and adjust aggressiveness of an anti-reverberation filter, based on an estimated ratio of direct to indirect (reverberated) sound energy expected to reach the microphone.

Abstract: In a harmonic adding device, an input/output function part converts a digital sound signal having an input signal level into a digital sound signal having an output signal level determined according to a input/output function and the input signal level. A parameter storage unit stores at least one inflection point. A controller sets the inflection point to the input/output function to enable the input/output function part to add harmonics to the digital sound signal. The input/output function is defined between a positive maximum point and a negative maximum point. The inflection point is set between the positive maximum point and the negative maximum point. The positive maximum point, the inflection point and the negative maximum point are sequentially connected by linear interpolation to formulate the input/output function.

Abstract: A voltage generation circuit includes a control circuit which outputs a first digital signal, a DAC which outputs a first analog signal corresponding to the first digital signal, and an attenuator which is connected to an output terminal of the DAC and is configured to output a voltage signal obtained by attenuating the first analog signal input from the DAC.

Abstract: A test system for evaluates acoustic feedback characteristics of a listening device by changing the transfer function of the signal path from an acoustic output to an acoustic input of the listening device. A variable filter comprises an acoustic input, an acoustic output, and a control unit for changing a transfer function of the filter. First and/or second acoustic propagation elements propagate acoustic sound, the first acoustic propagation element acoustically connecting the acoustic output of the listening device with the acoustic input of the variable filter, the second acoustic propagation element acoustically connecting the acoustic output of the variable filter with the acoustic input of the listening device; wherein the system is configured to allow a determination of acoustic feedback or a level of acoustic feedback of the listening device at different frequencies.

Abstract: A method of generating left (Ls) and right (Rs) sound signals known as “surround” signals from a stereo sound signal composed of a left sound signal and of a right sound signal. One of the stereo signals is subtracted from the other signal with the aid of a subtraction module to obtain a single subtraction signal (LR) in which the correlated and in-phase components of the stereo signals (L, R) have been removed. A right sound signal (Rs) and a left sound signal (Ls) are generated from the subtraction signal (L?R), these signals being decorrelated with respect to one another and corresponding respectively to the right and left sound “surround” signals. These “surround” signals (Ls, Rs) intend for broadcast on the rear channels of an acoustic system.

Abstract: Disclosed are an apparatus and method for creating an acoustic radiation pattern, which may improve sound focusing performance to a target position. The acoustic radiation pattern creating apparatus may process an input signal which may be output through a speaker array in a room such that at least one vertical-directional reflected wave component of the input signal may overlap a direct sound of the input signal at the target position. Then, the acoustic radiation pattern creating apparatus may replicate the processed input signal by the number of output channels of the speaker array, and may process the replicated input signals such that the replicated input signals may form a predetermined radiation pattern in a horizontal direction in the room when the replicated input signals are output through the speaker array.

Abstract: There is provided a system to provide a sequence of motion samples to a plurality of motion platforms, such as the plurality of seats in a movie theatre, such that the motion is synchronized with a sequence of images, such as a movie. In order to reduce the cost of the system, synchronization of a motion data signal with the sequence of images is centralized at a unique synchronizing unit. The synchronized sequence of motion samples is distributed to a plurality of motion platform hubs, each controlling and managing a reduced number of motion platforms.

Abstract: 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.

Abstract: To provide a speaker device that can form a substantially uniform sound field over a range from a long distance to a short distance without significantly increasing a calculation load. A plurality of FIR filters 21 to 2n perform delay control of respective speakers so as to increase a delay time difference between adjacent speakers 51 to 5n in a line array speaker 5 toward one end of the line array speaker 5, and thereby over a wide range from a long distance to a short distance, a sound filed 12 is formed. Also, by adding a common shift delay time Dc to filter coefficients for the FIR filters 21 to 2n, the delay time difference between adjacent speakers 51 to 5n is made less than a sampling period of a sound signal to form a wide and uniform sound field 12.

Abstract: Stereophonic sound is reproduced by acquiring image depth information indicating a distance between at least one object in an image signal and a reference location, acquiring sound depth information indicating a distance between at least one sound object in a sound signal and a reference location based on the image depth information, and providing sound perspective to the at least one sound object based on the sound depth information.

Abstract: An audio encoder receives multi-channel audio data comprising a group of plural source channels and performs channel extension coding, which comprises encoding a combined channel for the group and determining plural parameters for representing individual source channels of the group as modified versions of the encoded combined channel. The encoder also performs frequency extension coding. The frequency extension coding can comprise, for example, partitioning frequency bands in the multi-channel audio data into a baseband group and an extended band group, and coding audio coefficients in the extended band group based on audio coefficients in the baseband group. The encoder also can perform other kinds of transforms. An audio decoder performs corresponding decoding and/or additional processing tasks, such as a forward complex transform.

Abstract: In embodiments of the present invention improved capabilities are described for determining a multi-dimensional sound signature for a location within a hypothetical space, comparing the multi-dimensional sound signature to a known multi-dimensional sound signature, and modifying the hypothetical space such that the similarity between the multi-dimensional sound signature for the location within the modified hypothetical space and the known multi-dimensional sound signature is increased.

Abstract: An audio apparatus includes a decay pattern generator configured to generate a decay pattern to be applied in an early reflection region, and generate an audio signal to which a decay pattern is applied by convoluting the generated decay pattern with PCM raw data of a sound source, a reverberation generator configured to generate reverberation from the audio signal to which the decay pattern is applied, and an adder configured to generate an output signal having a sound enhancement effect by adding the PCM raw data to an output of the reverberation generator.

Abstract: Provided are an apparatus and method for reducing digital noise. The digital noise reducing apparatus includes: a clarified signal generator configured to generate a clarity improvement pattern for increasing an energy ratio of an early reflection region with respect to all reverberations for a received audio source signal, to convolve the clarity improvement pattern with the audio source signal, and to output an audio source signal convolved the audio source signal with the clarity improvement pattern; an early reflection generator configured to output an early reflection signal convolved the audio source signal with an early reflection pattern; a late reverberation generator configured to receive the audio source signal, and to generate a late reverberation signal for attenuating digital noise of the audio source signal; and a noise attenuator configured to generate an audio signal added the early reflection signal and the late reverberation signal to the audio source signal.

Abstract: A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.

Abstract: An ultrasonic noise cancelation system can include a communication module configured to receive a noise signal detected by a noise detection module, the noise signal representing a noise sound in a listener environment; a noise cancelation module configured to invert the received noise signal thereby creating an inverse noise signal representing an inverse of the noise sound; and a modulator configured to modulate the inverse noise signal onto an ultrasonic carrier to generate an ultrasonic signal.

Abstract: A reverberation suppressing apparatus, includes: a sound acquiring unit which acquires a sound signal; a reverberation data computing unit which computes reverberation data from the acquired sound signal; a reverberation characteristics estimating unit which estimates reverberation characteristics based on the computed reverberation data; a filter length estimating unit which estimates a filter length of a filter which is used to suppress a reverberation based on the estimated reverberation characteristics; and a reverberation suppressing unit which suppresses the reverberation based on the estimated filter length.

Abstract: Methods, systems, and computer program products for simulating propagation of sound in a static scene can be operated for pre-computing a transfer operator for simulating results of sound reflection within a modeled scene, simulating distribution of sound energy from a sound source positioned within the scene, applying the transfer operator to the distribution of sound energy from the sound source to compute simulated sound at a plurality of points within the scene, and simulating collection of sound energy from the plurality of points within the scene to a listener positioned within the scene.

Abstract: A method of and a system for generating an acoustic wave representing reverberations from a desired acoustic environment are described including having a recording surface (11) defined by a spatial distribution of recording transducers (?) and an emitting surface defined by a spatial distribution of emitting transducers (x), wherein the emitting surface (12) defines a volume within which the recording surface (11) is located, recording an acoustic wave (14) originating from within a volume defined by the recording surface (11) using the recording transducers (x), extrapolating the recorded wave (14) to the emitting surface using wavefield propagator system (IS) representing the desired acoustic environment, and emitting the extrapolated wave from the emitting transducers (?).

Abstract: A method and system for applying reverb to an M-channel downmixed audio input signal indicative of X individual audio channels, where X is greater than M. Typically, the method includes steps of: in response to spatial cue parameters indicative of spatial image of the downmixed input signal, generating Y discrete reverb channel signals, where each of the reverb channel signals at a time, t, is a linear combination of at least a subset of values of the individual audio channels at the time, t, and individually applying reverb to each of at least two of the reverb channel signals, thereby generating Y reverbed channel signals.

Abstract: In some embodiments, a method applying reverberation to audio from at least one client of a set of clients which share a virtual environment, including by asserting position data and at least one input audio stream to a server, selecting (in the server) a reverberation filter for each input audio stream in response to the position data and generating wet audio by applying to the input audio an early reverberation part of the selected reverberation filter. Typically, a client applies a late reverberation filter to the wet audio using metadata from the server. In other embodiments, a server selects a reverberation filter for application to audio in response to position data, asserts the audio and metadata indicative of the filter, and a client applies the filter to the audio using the metadata. Other aspects are systems, servers, and client devices configured to perform any embodiment of the method.

Abstract: An audio processing system includes an audio processing device and an external device, the audio processing device having a signal processing unit adapted for running an algorithm for processing an input signal representing an acoustic signal from the environment of the user and providing a processed output signal. The external device includes in an information signal a measure, characteristic of the reverberation of the room or location where the audio processing device is located, and the audio processing device is adapted to extract the measure of the reverberation from the information signal and use the measure as an input to an algorithm that includes a directional algorithm for providing a directional characteristic of the input signal.

Abstract: A method and apparatus for introducing a time-varying time delay randomly into the individual reproduction channels of a sound recording, two in the case of binaural presentation. This emulates the temporal aspect of microphone and/or listener motion. The present invention may be applied as a unidirectional process. No preparation of the source material is required. It can be applied to any multichannel audio signal set. It can process analog or digital signals. The process may be used with headphones, loudspeakers, hearing aids or similar assistive hearing devices.

Abstract: Disclosed is an audio reverberation apparatus. The audio reverberation apparatus interacts with or may include a mechanical or acoustic reverberation element. An audio signal is pitch transposing upward and applied to the mechanical or acoustic reverberation element. The resulting audio reverberant signal retrieved from the mechanical or acoustic reverberation element is pitch-transposed downward by the same factor as the upward pitch transposing. This results in the mechanical or acoustic reverberation element requiring smaller dimensions in comparison to a mechanical or acoustic reverberation element where a non-pitch-transposed audio signal is applied.

Abstract: A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.

Abstract: Electronic devices having digital reverberators are disclosed, together with a method of reproducing sound for a user with the digital reverberator. The digital reverberator uses digital surface absorption filters positioned in the reverberator to simulate absorption of energy as digital audio data samples are reflected from virtual surfaces. The position of the digital surface absorption filters enables known frequency-dependent surface absorption characteristics of real materials to be directly implemented using the filter coefficients of each digital surface absorption filter. This enables virtual acoustic spaces to be designed quickly without the need for the digital reverberator to be manually tuned for each space.

Abstract: Systems, methods, and other embodiments associated with selective notch filtering for howling suppression are described. According to one embodiment, an apparatus includes a howling detector that detects howling by performing a time domain analysis of speech signals to identify a speech signal that may be exhibiting howling and performing frequency domain analysis to confirm that the speech signal is exhibiting howling. The apparatus also includes a suppression selector configured to select a suppression technique, a signal processor configured to process the speech signal according to a selected suppression technique. The apparatus outputs, without suppression-related processing, speech signals that are not identified based on the time domain analysis.

Abstract: The present invention relates to a method for determining an acoustic property of an environment. According to the method a sound signal and at least two reflection signals of the sound signal (are received, and for each of the reflection signals a time difference between a time of reception of the sound signal and a time of reception of the respective reflection signal is determined. Furthermore, for each of the reflection signals, an angle between a reception direction in which the sound signal was received and a reception direction in which the respective reflection signal was received is determined. Based on the determined time differences, the angles and predetermined constraints defining arrangement constraints of the sound reflecting surfaces an arrangement of the sound-reflecting surfaces is determined.

Abstract: A home use sound reproduction system for Hi-fi, with digital signal transfer from a playback unit via a control unit to one or more active loudspeakers, each including or arranged beside an amplifier unit. The control unit is arranged to control sound parameters and send both a digital sound information signal and a power signal for powering the amplifier units, and to superimpose the single ended or differential digital signal together with the power signal on at least one common lead in a cable.

Abstract: An audio signal processing device includes: head related transfer function convolution processing units convoluting head related transfer functions with audio signals of respective channels of plural channels, which allow the listener to listen to sound so that sound images are localized at assumed virtual sound image localization positions concerning respective channels of the plural channels of two or more channels when sound is reproduced by electro-acoustic transducer means; and 2-channel signal generation means for generating 2-channel audio signals to be supplied to the electro-acoustic transducer means from audio signals of plural channels from the head related transfer function convolution processing units, wherein, in the head related transfer function convolution processing units, at least a head related transfer function concerning direct waves from the assumed virtual image localization positions concerning a left channel and a right channel in the plural channels to both ears of the listener is n

Abstract: The invention relates to a system for simulating stereophonic sound (1), comprising the following: a core module (10) having a space simulation module (11), an echo module and an interface module (15), a control module (20), a digital audio delay matrix module (21), and a digital audio/network system (30). The invention is characterized in that said system provides an echo and/or directional acoustic irradiation on the basis of a system latency less than 2.5 ms by means of the core module (10), the control module (20), and the digital audio network system. A system for simulating stereophonic sound is thus provided, which system operates with a reduced number of loudspeakers and without dedicated components and/or proprietary hardware and provides a plurality of different functions, such as extension of the echo time and directional acoustic irradiation.

Abstract: Disclosed is an audio reverberation apparatus. The audio reverberation apparatus interacts with or may include a mechanical or acoustic reverberation element. An audio signal is pitch transposing upward and applied to the mechanical or acoustic reverberation element. The resulting audio reverberant signal retrieved from the mechanical or acoustic reverberation element is pitch-transposed downward by the same factor as the upward pitch transposing. This results in the mechanical or acoustic reverberation element requiring smaller dimensions in comparison to a mechanical or acoustic reverberation element where a non-pitch-transposed audio signal is applied.

Abstract: A stereophonic sound output apparatus and an early reflection generation method thereof. The stereophonic sound output apparatus includes an early reflection generator to implement an early reflection when a 5.1 channel audio signal is down-mixed to a 2-channel audio signal to play back a 5.1 channel audio signal through a 2-channel headphone. The early reflection generator generates early reflections in pairs in which there is an appropriate time difference between the left side reflections and the right side reflections by generating an interaural time difference between two input audio signals and filtering. It is possible to copy the characteristics of early reflections in a real listening room. It is also possible to implement an early reflection similar to a real reflection measured in an apparatus for playing back the 5.1 channel audio signal through 2-channel headphone. A natural 5.1 channel effect may also be obtained using little computation.

Abstract: A controller device is provided whereby players can experience realistic sounds. The controller device is connected to the main body of an information processing apparatus, wherein the controller device comprises a first speaker for emitting a sound signal in a low frequency range, and a second speaker for emitting a sound signal in a frequency range that is higher in comparison with the first speaker.

Abstract: An apparatus for reverberation time extension is provided. The apparatus includes a module for calculating wave field synthesis information, and a signal processor for generating a plurality of audio output signals for a plurality of loudspeakers based on the audio input signals that have been recorded by a plurality of microphones, and based on the wave field synthesis information. Further, the apparatus includes an operating unit for determining a virtual position of one or several virtual walls. The module for calculating wave field synthesis information is implemented to calculate the wave field synthesis information based on the virtual position of the one or several virtual walls. Further, the virtual position is adjustable by the operating unit for at least one of the virtual walls.

Abstract: The present technology provides robust, high quality dereverberation of an acoustic signal which can overcome or substantially alleviate the problems associated with the diverse and dynamic nature of the surrounding acoustic environment. The present technology utilizes acoustic signals received from a plurality of microphones to carry out a multi-faceted analysis which accurately identifies reverberation based on the correlation between the acoustic signals. Due to the spatial distance between the microphones and the variation in reflection paths present in the surrounding acoustic environment, the correlation between the acoustic signals can be used to accurately determine whether portions of one or more of the acoustic signals contain desired speech or undesired reverberation. These correlation characteristics are then used to generate signal modifications applied to one or more of the received acoustic signals to preserve speech and reduce reverberation.

Abstract: A method for continuously estimating reverberation decay comprising receiving a sequence of audio data samples. Determining whether a plateau is present in the sequence of audio data samples. Generating one or more reverberation parameters from the sequence of audio data samples if it is determined that the plateau is present.

Abstract: Analysis of the audio component in multimedia data is disclosed. Rooms can be described through room impulse responses (RIR), the “fingerprint” of a specific room. The method uses machine learning techniques to identify rooms from ordinary audio recordings.

Abstract: An audio signal encoding apparatus includes a processor to quantize an audio signal, to obtain a characteristic of reverberation masking that is exerted on a sound represented by the audio signal by reverberation of the sound generated in a reproduction environment by reproducing the sound, and to control a quantization step size of the audio signal that quantized based on the characteristic of the reverberation masking.

Abstract: The subject matter described herein includes a method for simulating directional sound reverberation. The method includes performing ray tracing from a listener position in a scene to surface as visible from a listener position. The method further includes determining a directional local visibility representing a distance from a listener position to nearer surface in the scene alone each ray. The method further includes determining directional reverberation at the listener position based on the directional local visibility. The method further includes rendering a simulated sound indicative of the directional reverberation at the listener position.

Abstract: A reverberant characteristic of an acoustic space is superimposed on an audio signal that is received by an apparatus. The apparatus decomposes the audio signal into an estimated original dry signal component and an estimated reverberant characteristic of the acoustic space. Estimation of the original dry signal component and the reverberant characteristic of the acoustic space is based on determination of an estimated impulse response of the acoustic space from the received audio signal. Once the audio signal is decomposed, the estimated original dry signal component and the estimated reverberant characteristic of the acoustic space may be independently modified by the apparatus. The modified or unmodified estimated original dry signal component and estimated reverberant characteristic of the acoustic space may be combined by the apparatus to produce one or more adjusted frequency spectra.

Abstract: A signal processing apparatus is provided. The signal processing apparatus comprises: an inputting section for inputting audio signals on a plurality of channels; an acoustic type acquiring section which is adapted to acquire an acoustic type of an audio signal on at least one channel of the audio signals; and a process controlling section which is adapted to control a characteristic of sound-field effect applied to the audio signals based on the acquired acoustic type.

Abstract: One example embodiment increases perceived signal strength of a sound signal based on persistence of hearing. When the ear perceives a signal it takes a finite length of time to process it. During that time period the ear does not recognize input and is, in effect, “Turned Off”. In accordance with one aspect of the technology, these periods of “unheard” audio input are utilized so that they supplement the information which the ear sends to the brain. An exemplary system includes a switch, configured to route the audio alternatively to two signal paths—one which includes a delay circuit that delays that audio by an amount equal to the ear's persistence of hearing interval. The system also includes a signal combiner configured to combine outputs from the two signal paths, so as to provide the brain with a signal that it will perceive as being twice as loud as the original.

Abstract: An echo canceller system includes a first echo canceller having a first voltage divider and an adaptable second voltage divider that is configured to generate a first replica of an echo. A second echo canceller is configured to generate a second replica of an echo and has tap values that are generated in response to an error signal. A controller is coupled to the first and second echo cancellers and includes a selection algorithm that responds to the tap values of the second echo canceller and selects a voltage divider value for the adaptable second voltage divider.