Abstract: This invention relates to the field of multichannel audio. More particularly, the invention relates to a method for the provision of audio channels suitable for application to loudspeakers located above conventional front loudspeakers.

Abstract: A calculation unit for a sound system includes an input terminal, a processor and an output terminal. The input terminal has the purpose to receive an audio stream to be reproduced using the sound system. The output terminal has the purpose to control the sound system based on a first and a second plurality of individual audio signals. The processor is configured to calculate the first plurality of audio signals such that beamforming is performed by the array and to calculate the second plurality of individual audio signals to perform, using the sound system, direct sound suppression such that sound is canceled towards a listening direction. Furthermore, the processor filters at least the second plurality using a second passband characteristic including a second portion of the entire frequency range.

Abstract: A method for analysing and decomposing a stereo audio signal including an audio signal for a left reproduction device and an audio signal for a right reproduction device by extracting panning coefficients that contain direction information about the sound sources from which the stereo audio signal originates based on the approximation that one sound source can be regarded as dominant for each frequency. This approximation allows the panning coefficients to be obtained, by solving a system of equations, with lower computation complexity than in the prior art. The sound quality that is obtained after re-panning the signal enhanced in this manner for a configuration with more than two loudspeakers is constant or better. Advantageously, following determination of the panning coefficients, the direct signal and two ambient signals that are not correlated with the direct sound source are extracted from the stereo audio signal.

Abstract: This invention relates to a circuit which enables even a single loudspeaker or audio channel to appear spatiotemporally as real to life. A discrete signal source with at least two discrete in-phase reference signals is split to form a first and second pair of signals. The second pair of signals is mixed to form one signal which is gain adjusted and then phase reversed before separately mixing with each of the first pair of reference signals to produce a pair of output signals. A monophonic signal source can be split to form the at least two discrete in-phase reference signals.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

Abstract: A system for rotating sound provides for the ability of the apparent direction of sound sources in a listening environment to remain in consistent orientations in space despite rotations of the microphones used to capture the sound and despite rotations of the head of the listener, even when wearing headphones. Modules are provided in the system to distinguish the sound sources and their apparent directions, as well as to rotate the sound sources in response to detected rotations of the listener's head and/or detected rotations of the microphones.

Abstract: A method and apparatus of audio processing is described. An audio processor may receive a multi-channel audio signal. In a first mode of operation, an audio processor may output a first audio signal on a first multi-channel audio output and a second audio signal on a second multi-channel audio output. In a second mode of operation, the audio processor may output a third audio signal on the first multi-channel audio output and a sub-woofer audio signal on at least one channel of the second multi-channel output.

Abstract: An acoustic output device and a control method thereof are provided.

Abstract: A microphone arrangement and a method using the microphone arrangement for recording surround sound in a mobile device, where the microphone arrangement comprises a first and a second microphone and arranged at a first distance to each other and configured to obtain a stereo signal, and comprises a third microphone configured to obtain a steering signal together with at least one of the first and second microphone or with a fourth microphone. The microphone arrangement also comprises a processor configured to separate the stereo signal into a front stereo signal and a back stereo signal based on the steering signal.

Abstract: An example playback device receives left and right channels of audio content and generates a center channel of the audio content by combining at least a portion of the left right channels. The playback device generates first and second side channels of the audio content by combining the center channel and a difference of the left channel and the right channel and combining the center channel and an inverse of the difference of the left channel and the right channel, respectively. The first and second side channels are attenuated by a filter with a given cutoff frequency. When volume is adjusted, the cutoff frequency is adjusted based on the volume level of the playback device and the spectral characteristics of the audio content. The cutoff frequency is positively related to the volume level such that a volume increase causes a corresponding increase in the cutoff frequency and vice versa.

Abstract: Techniques of rendering audio involve applying a balanced-realization state space model to each head-related transfer function (HRTF) to reduce the order of an effective FIR or even an infinite impulse response (IIR) filter. Along these lines, each HRTF G(z) is derived from a head-related impulse response filter (HRIR) via, e.g., a z-transform. The data of the HRIR may be used to construct a first state space representation [A, B, C, D] of the HRTF via the relation G(z)=C(zI?A)?1B+D This first state space representation is not unique and so for an FIR filter, A and B may be set to simple, binary-valued arrays, while C and D contain the HRIR data. This representation leads to a simple form of a Gramian Q whose eigenvectors provide system states that maximize the system gain as measured by a Hankel norm.

Abstract: Method and apparatus are disclosed for determining and accounting for distortions to an audio signal due to the geometric properties of the interior cabin of a vehicle. An example vehicle includes a microphone, a seat having a plurality of seat positions, and a processor. The processor is configured to determine a first seat position corresponding to a point in time at which an audio signal is received, determine a cabin impulse response corresponding to the first seat position, and determine a filtered audio signal based on the cabin impulse response and the audio signal.

Abstract: According to the invention, a method for providing audio to a user is disclosed. The method may include determining, with an eye tracking device, a gaze point of a user on a display. The method may also include causing, with a computer system, an audio device to produce audio to the user, where content of the audio may be based at least in part on the gaze point of the user on the display.

Abstract: Embodiments of a virtual surround-sound system and methods for simulating surround-sound are generally described herein. Other embodiments may be described and claimed. In some embodiments, a processing module may include spatial processor spatially processes surround-left and surround-right channel signals and front-left and front-right channel signals and combines the spatially-processed signals for providing to drivers of center speaker after crosstalk cancellation and combining with a center-channel signal. In some embodiments, the processing module may include circuitry to cause the spatial processor to refrain from spatially processing either the front-left and front-right channel signals when front-left and/or front-right speakers are connected.

Abstract: Multiple virtual source locations may be defined for a volume within which audio objects can move. A set-up process for rendering audio data may involve receiving reproduction speaker location data and pre-computing gain values for each of the virtual sources according to the reproduction speaker location data and each virtual source location. The gain values may be stored and used during “run time,” during which audio reproduction data are rendered for the speakers of the reproduction environment. During run time, for each audio object, contributions from virtual source locations within an area or volume defined by the audio object position data and the audio object size data may be computed. A set of gain values for each output channel of the reproduction environment may be computed based, at least in part, on the computed contributions. Each output channel may correspond to at least one reproduction speaker of the reproduction environment.

Abstract: An audio surround processing system receives an audio source signal having at least two audio channels and generates a number of additional surround sound signals in which an amount of artificially generated ambient energy is controlled in real-time at least in part by an estimate of ambient energy that is contained in the audio source signal. The system may divide the audio source signal into two sets of components; a first set of components and a second set of components. The first set of components may be in a range of frequency that is less than a range of frequency of the second set of components. An ambience estimate control coefficient may be generated using the transformed first set of components. An overall gain may be determined using the ambience estimate control coefficient. The overall gain may be used in generation of the additional surround sound signals.

Abstract: Techniques for personalizing audio levels, in accordance with embodiments of the present technology, provide different audio volumes to different locations in a room allowing for two of more users to enjoy the same audio content at different volumes. Differential level and delay compensation filtering based on a position of each of a plurality of speakers, the location of each user and the preferred relative audio volume of each user are utilized to produce different effective audio levels in localized regions of a room.

Abstract: A playback request processing method and apparatus belongs to the field of computer technologies. The method includes: receiving a playback request sent by a sound source; determining whether a sound diffusion range of the sound source is within a current receiving range of a recipient; and processing the playback request if the sound diffusion range of the sound source is within the current receiving range of the recipient. In the present disclosure, it is determined according to the received playback request sent by the sound source that the sound diffusion range of the sound source is within the current receiving range of the recipient, and whether to process the playback request sent by the sound source is determined according to a determination result.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

Abstract: An apparatus comprises at least one processor, and at least one memory including computer program code, the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus to: provide an audio capture event wherein one or more microphone configurations are configured to provide one or more audio signals based on at least one acoustic signal from at least one acoustic source, at least one of the one or more microphone configurations being defined by a first position of a first microphone configuration on a first portion and a second position of a second microphone configuration on a second portion, the second portion being movable relative to the first portion.

Abstract: Described herein is a retrofit digital speaker system comprising two or more retrofitted speaker enclosures, each of the two or more retrofitted speaker enclosures (enclosures) comprising: at least one speaker; and an analog-and-digital interface adapted to receive digitally encoded audio signals, electrical power, and digital command signals, and wherein the digitally encoded audio signals, electrical power, and digital command signals are transmitted over existing two wire analog audio cables, and further wherein the two or more retrofitted speaker enclosures are wired in a daisy chain fashion, via the existing two wire analog audio cables.

Abstract: A method for altering an audio signal of interest in a multi-channel soundfield representation of an audio environment, the method including the steps of: (a) extracting the signal of interest from the soundfield representation; (b) determining a residual soundfield signal; (c) inputting a further associated audio signal, which is associated with the signal of interest; (d) transforming the associated audio signal into a corresponding associated soundfield signal compatable with the residual soundfield; and (e) combining the residual soundfield signal with the associated soundfield signal to produce an output soundfield signal.

Abstract: Apparatus for linear and nonlinear acoustic echo control includes loudspeaker, first, second, and third microphone, beamformer, and first echo canceller. The loudspeaker outputs a loudspeaker signal that includes reference signal. The first microphone and the second microphone are collocated with the loudspeaker, receive at least one of: a near-end speaker signal from a near-end speaker and the loudspeaker signal, and generate first and second microphone uplink signals, respectively. The third microphone receives the near-end speaker signal and generates a third microphone uplink signal. The beamformer receives the first and second microphone uplink signals, directs a beam towards the loudspeaker and drives a null towards the near-end speaker, and generates a beamformer output. The first echo canceler receives the third microphone uplink signal and the beamformer output, and cancels echoes in the third microphone uplink signal based on the beamformer output to generate an echo cancelled signal.

Abstract: An assembly configured to prevent phase error includes an input unit, a filter unit, a slicer unit, a subtraction unit and an addition unit. The input unit provides an original audio signal. The filter unit selects one part of an original audio signal according to a preset bandwidth setting on the filter unit to form a first audio signal, and outputs said audio signal from a second output terminal. The slicer unit analyzes the amplitude magnitude according to a preset threshold amplitude to form a second audio signal. The subtraction unit subtracts the first audio signal from the original audio signal to form a third audio signal. The second audio signal and the third audio signal have the same phase retardation. The addition unit adds the second audio signal and the third audio signal to form a fourth audio signal.

Abstract: 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.

Abstract: A spatial audio processing system and method including the steps of: dividing the series of virtual speakers into a series of horizontal planes around the expected listener; rendering the audio source to an intermediate spatial format for playback over a series of virtual speakers arranged in each of the series of planes around the listener, the rendering including: an initial panning of the spatialized virtual audio source to each of the horizontal planes to produce a plane rendered audio emission; a subsequent panning of each of the plane rendered audio emissions to a series of virtual speaker locations within each plane, with the subsequent panning utilizing a series of panning curves which are spatially smoothed to can include spatial frequency components which are less than the Nyquist sampling rate of the audio source.

Abstract: An audio processor may receive audio input channels including stereo channels and one or more surround channels. The audio processor may downmix the audio input channels into stereo output channels; developing the stereo output channels into upmixed audio channels including at least one additional surround channel not present in the audio input channels; delay the audio input channels into delayed audio channels that are time-aligned with the upmixed audio channels; and mix the delayed audio channels and the upmixed audio channels into audio output channels.

Abstract: An audio emission device and an audio capture device that may respectively emit and capture sound within a listening area is described. The audio emission device may produce one or more primary audio beams in the listening area. Each of the primary audio beams may be formed by weighting a set of modal beam patterns. Separate orthogonal test signals may be injected into each modal beam pattern. Based on these separate orthogonal test signals, the individual modal beam patterns may be extracted from a detected sound signal, produced by the audio capture device, such that the contribution from each of these modal patterns in the detected sound signal may be determined. Utilizing the contributions from each modal beam pattern in the detected sound signal, the spatial relationship (e.g., distance and/or orientation/angle) between the audio emission device and the audio capture device may be determined.

Abstract: An example playback device (i) provides a center channel of audio content to one or more first audio drivers and one or more second audio drivers so that the center channel is reproduced according to a first radiation pattern and (ii) provides a side channel of audio content to the one or more first audio drivers so that the side channel is reproduced according to a second radiation pattern. The first and second radiation patterns may combine to form a response lobe that has a maximum between the respective maxima of the first and second radiation patterns. An example non-transitory computer-readable medium and an example method, both related to the example playback device, are also disclosed herein.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

Abstract: Audio spatial effects are provided using a spherical array of ultrasonic speakers, with the azimuth angle and, if desired, elevation angle demanded by a control signal from, for example, a game console being matched to the sonic axis of one of the speakers in the array to activate the matching speaker.

Abstract: Embodiments of the present invention relate to adaptive audio content generation. Specifically, a method for generating adaptive audio content is provided. The method comprises extracting at least one audio object from channel-based source audio content, and generating the adaptive audio content at least partially based on the at least one audio object. Corresponding system and computer program product are also disclosed.

Abstract: Devices and methods are adapted to characterize a multi-channel loudspeaker configuration, to correct loudspeaker/room delay, gain and frequency response or to configure sub-band domain correction filters.

Abstract: Embodiments are described for a system for virtual rendering of object based audio through binaural rendering of each object followed by panning of the resulting stereo binaural signal between a plurality of cross-talk cancelation circuits feeding a corresponding plurality of speaker pairs. In comparison to prior art virtual rendering utilizing a single pair of speakers, the described embodiments improve the spatial impression for both listeners inside and outside of the cross-talk canceller sweet spot. Also described is an improved equalization technique for a crosstalk canceller that is computed from both the crosstalk canceller filters and the binaural filters and applied to a monophonic audio signal being virtualized. The described techniques improve timbre for listeners outside of the sweet-spot as well as a smaller timbre shift when switching from standard rendering to virtual rendering.

Abstract: The communication device comprising a voice communication implementer, a ringing mode implementer, a silent mode implementer, an answering machine implementer, and a stereo audio data output implementer.

Abstract: Systems and methods of audio reproduction are described. In one embodiment, one or more processors can be configured, for example, to receive an L channel signal and an R channel signal, to combine the L channel signal and the R channel signal, to average the combined signal, to detect an impulse signal in the averaged signal, to multiply the impulse signal, to scale the multiplied impulse signal, and to add the scaled impulse signal to the averaged signal to form a resultant signal. Sound can be generated, for example, by speakers based on at least the resultant signal.

Abstract: An information processing apparatus including: an attack detection unit that detects an attack; and a strength adjustment unit that incrementally raises the strength of a security measure every time that an attack is detected by the attack detection unit.

Abstract: A musical performance apparatus is provided with a tone generation circuit 15 and a sound system 16 for emitting musical tones of musical instruments and control tones corresponding to musical score data SD which controls a musical score display apparatus 20. The tone generation circuit 15 has a tone volume adjustment circuit 15a3 which changes tone volume of only the musical tones in accordance with player's instruction and maintains tone volume of the control tones at a certain tone volume. The tone generation circuit 15 also has a pan adjustment circuit 15a4 which localizes the control tones so that the control tones will be emitted only from a certain speaker. Furthermore, the musical performance apparatus is designed such that the control tones are formed only of frequency components included in a certain high frequency band.

Abstract: In general, techniques are described for obtaining decomposed versions of spherical harmonic coefficients. In accordance with these techniques, a device comprising one or more processors may be configured to determine a first non-zero set of coefficients of a vector that represent a distinct component of a sound field, the vector having been decomposed from a plurality of spherical harmonic coefficients that describe the sound field.

Abstract: In order to realize a correction for changes in the reproduction loudness at low frequencies in a downmix-arrangement, a mixing arrangement is proposed for mixing at least two audio signals, which mixing arrangement is provided with a first unit (104) for deriving a first power signal, which is a measure for the power of the first audio signal, a second unit (105) for deriving a second power signal, which is a measure for the power of the second audio signal, a cross-correlation unit (103) for deriving a cross-correlation signal, which is a measure for a cross-correlation between the first and the second audio signal, a unit (106) for deriving multiplication parameters from the first and second power signals and the cross-correlation signal, and a multiplication and combination unit (107) for carrying out a signal processing on the first and second audio signals and combining them.

Abstract: The present research relates to controlling rendering of multi-object or multi-channel audio signals. The present research provides a method and apparatus for controlling rendering of multi-object or multi-channel audio signals based on spatial cues in a process of decoding the multi-object or multi-channel audio signals. To achieve the purpose, the method suggested in the research controls rendering in a spatial cue domain in the process of decoding the multi-object or multi-channel audio signals.

Abstract: Various embodiments related to providing an audio notification to a listener via a dual-mode speaker system are provided. In one embodiment, a computing device and associated dual-mode speaker system are configured to be selectively operable in a non-directional broadcast mode and a directional broadcast mode. Based on an audio notification condition, it is determined that the audio notification is to be delivered using the directional broadcast mode. The dual-mode speaker system is then used to broadcast the audio notification to the listener using the directional broadcast mode.

Abstract: A method for encoding audio signals, for later reproduction in arbitrary three-dimensional loudspeaker layouts, based on the generation of an intermediate channel-independent representation, which enables the creation, manipulation and reproduction of sounds with complex apparent size and shape, including multiple disconnected shapes.

Abstract: There is provided a method for enlarging a location with optimal three-dimensional audio perception. Optimal three-dimensional audio perception may relate to a fully spatial sound effect. The method includes deriving three-dimensional encoded localization cues from an audio input signal having a first channel signal and a second channel signal; decoding the first channel signal and the second channel signal into a plurality of decoded channel signals, the plurality of decoded channel signals being equal to a number of speaker units; performing crosstalk cancellation on the plurality of decoded channel signals to eliminate crosstalk between the plurality of decoded channel signals; and outputting the plurality of decoded channel signals which have been subjected to crosstalk cancellation to each of the number of speaker units. It is advantageous that the crosstalk cancellation includes further processing to generate a smoothed frequency envelope.

Abstract: A device may store a subset of a plurality of head-related transfer functions (HRTFs) for emulating stereo sound from a source in three-dimensional (3D) space, each of the HRTFs corresponding to a direction from which the stereo sound is perceived to arrive, by a user hearing the stereo sound. The device may also obtain a first direction from which first stereo sound is perceived to arrive, by the user and determine whether the subset of the plurality of HRTFs includes a first HRTF corresponding to the first direction, wherein the plurality of HRTFs include the first HRTF. Further, the device may select two HRTFs in the subset of the HRTFs, wherein directions that are associated with the two HRTFs are closer to the first direction than directions of other HRTFs in the subset of the HRTFs.

Abstract: This application relates to a system for fitting a hearing aid by testing the hearing aid patient with a three-dimensional sound field having one or more localized sound sources. In one embodiment, a signal processing system employing head-related transfer functions is used to produce audio signals that simulate a three-dimensional sound field when a sound source driven by such audio signals is coupled directly to one or both ears. By transmitting the audio signals produced by the signal processing system to the hearing aid by means of a wired or wireless connection, the hearing aid itself may be used as the sound source.

Abstract: An apparatus for enhancing a multichannel audio signal comprising at least two channels configured to: estimate a value representing a direction of arrival associated with a first audio signal from at least a first channel and a second audio signal from at least a second channel of at least two channels of a multichannel audio signal; determine a scaling factor dependent on the direction of arrival associated with the first audio signal and the second audio signal; and apply the scaling factor to a parameter associated with a difference in audio signal levels between the first audio signal and the second audio sign.

Abstract: Methods and systems for processing an audio signal are provided. The method includes generating a pseudorandom sequence and generating at least one reciprocal of the pseudorandom sequence such that the at least one reciprocal is substantially decorrelated with the pseudorandom sequence. The pseudorandom sequence and the at least one reciprocal form a set of sequences. The method further includes convolving the audio signal with the set of sequences to generate a corresponding number of output signals and providing the number of output signals to a corresponding number of loudspeakers.

Abstract: This invention relates to the field of multichannel audio. More particularly, the invention relates to a method for the provision of audio channels suitable for application to loudspeakers located above conventional front loudspeakers.

Abstract: An audio signal is processed in the frequency domain to convert an input signal format to an output signal format. That is, a multichannel audio signal intended for playback over a predefined speaker layout can be formatted to achieve spatial reproduction over a different layout comprising a different number of speakers.