Abstract: A method of controlling a system which includes the steps of obtaining at least one signal representative of information communicated by a user via an input device in an environment of the user, wherein a signal from a first source is available in a perceptible form in the environment; estimating at least a point in time when a transition between information flowing from the first source and information flowing from the user is expected to occur; and timing the performance of a function by the system in relation to the estimated time.

Abstract: An audio signal processor receives a plurality of encoded multi-channel audio signals. A multi-channel decoder (105) decodes a first encoded multi-channel signal to generate a first decoded multi-channel signal. A generator (109) generates an encoded further audio signal by selecting audio encoding data from at least a second encoded multi-channel audio signal such that a number of channels of the encoded further audio signal comprising audio encoding data from the second encoded multi-channel audio signal is less than a number of channels in the second encoded multi-channel signal. Thus, a channel reduction is performed in the encoded data domain. A further decoder (111) generates a further decoded signal by decoding the further encoded audio signal. A combiner (107) combines the first decoded multi-channel signal and the further decoded signal to generate a multi-channel output signal. An exciting user experience can be provided while maintaining low complexity and resource usage.

Abstract: An audio system comprises a receiver which receives an input audio signal. A decomposer (103) decomposes the audio signal into at least a transient component signal and a non-transient component signal. An output circuit (105, 107, 109) then generates a first output audio signal in response to a weighted combination of the transient component signal and the non-transient component signal. In the combination the weighting of the transient component signal is different than the weighting of the non-transient component signal. A new signal with different emphasis of specific sound characteristics can be achieved. The approach may be particularly suited to generation of new spatial audio channels from an existing spatial audio channel, such as in particular the generation of an elevated channel from audio signals of a lower channel.

Abstract: An audio reproduction system includes an arrangement of audio speakers of a first kind having a first degree of directivity in combination with at least one audio speaker of a second kind having a second degree of directivity. In order to create a virtual sound source at a desired distance to a listener's position, the second degree of directivity is substantially larger than the first degree of directivity.

Abstract: A sound processing system receives a stereo signal which, by a segmenter (109) is divided into stereo time-frequency signal segments, each of which may correspond to a frequency domain sample in a given time segment. A decomposer (111) decomposes the time-frequency signal segments by for each pair of stereo time-frequency signal segments performing the steps of: determining a similarity measure indicative of a degree of similarity of the stereo time frequency signal segments; generating a sum time-frequency signal segment as a sum of the stereo time-frequency signal segments; and generating a centre time-frequency signal segment from the sum time-frequency signal segment and a pair of side stereo time-frequency segments from the pair of stereo time-frequency signal segments in response to the similarity measure.

Abstract: An apparatus for spatial sound reproduction comprises a receiver (101) for receiving a multi-channel audio signal. An analyzer (107) determines a spatial property of the multi-channel audio signal, such as a spatial complexity or organization. A selection processor (109) then selects a reproduction mode from a plurality of sound reproduction modes where the multi-channel sound reproduction modes employ different spatial rendering techniques. A reproduction circuit (103) then drives a set of loud-speakers (105) to reproduce the multi-channel audio signal using the selected reproduction mode. The switching between the reproduction modes may be fast (e.g. in the order of 100 ms to 10 secs) thereby allowing a short term adaptation of the reproduction mode to the signal characteristics. The approach may in particular provide an improved spatial experience to a listener.

Abstract: An audio apparatus comprises a processor (101) for providing a set of audio channels. A prediction circuit (103) generates a predicted signal for a first channel by adaptive filtering of a second channel by an adaptive filter. An adaptation processor (105) adapts the adaptive filter to minimize a cost function indicative of a difference between the predicted signal and the first channel. A compensation processor (107) then generates a non-predicted signal by compensating the first signal for the predicted signal and a distribution processor (109) generates an output set of audio channels by distributing at least the predicted signal and the non-predicted signal over the output set of audio signals where the distribution is different for the predicted signal and the non-predicted signal.

Abstract: A surround sound system comprises a receiver (301) for receiving a multichannel spatial signal that comprises at least one surround channel. A directional ultrasound transducer (305) is used for emitting ultrasound towards a surface to reach a listening position (111) via a reflection of the surface. The ultrasound signal may specifically reach the listening position from the side, above or behind of a nominal listener. A first drive unit (303) generates a drive signal for the directional ultrasound transducer (301) from the surround channel. The use of an ultrasound transducer for providing the surround sound signal provides an improved spatial experience while allowing the speaker to be located e.g. to the front of the user. In particular, an ultrasound beam is much narrower and well defined than conventional audio beams and can accordingly better be directed to provide the desired reflections. In some scenarios, the ultrasound transducer (305) may be supplemented by an audio range loudspeaker (309).

Abstract: An ambient telecommunication system (10) is provided. The system (10) comprises at least one transmitter (11) and a corresponding receiver (12) for transmitting and receiving an electronically or digitally encoded information signal between two or more conversing parties. The system further comprises a movement member (14) being configured to perform a movement or assume a certain physical posture, based on the result of content analysis of the information signal.

Abstract: A method of controlling a system which includes the steps of obtaining at least one signal representative of information communicated by a user via an input device in an environment of the user, wherein a signal from a first source is available in a perceptible form in the environment; estimating at least a point in time when a transition between information flowing from the first source and information flowing from the user is expected to occur; and timing the performance of a function by the system in relation to the estimated time.

Abstract: A method of controlling at least one device (8,12,13,20-25) using a telecommunications system, includes: obtaining signals communicating information from at least one remote telecommunication terminal (1,2) and from at least one local input device (14-16) for forming at least one local telecommunication terminal with at least one local output device (17-19), a connection being open between the remote and local telecommunications terminals; analyzing information (40,41) communicated from at least two of the telecommunication terminals (1,2,8,12,13) over time in relation to each other, using at least one criterion for detecting interaction of a type corresponding to conversational turn-taking between users of the respective telecommunication terminals (1,2,8,12,13); and adjusting at least one output signal for controlling a device (8,12,13,20-25) upon determining whether the at least one criterion is met.

Abstract: An audio reproduction system comprises an arrangement of audio speakers (1-5) of a first kind having a first degree of directivity in combination with at least one audio speaker (7) of a second kind having a second degree of directivity. In order to create a virtual sound source at a desired distance to a listeners position (LP) the second degree of directivity is substantially larger than the first degree of directivity.

Abstract: The method of the invention enables selection of an item (21) from a plurality of items (21, 23, 25). The method comprises the steps of visually representing a selected item (21) and reproducing at least part of an audio segment representing the selected item (21). The method further comprises applying a visual spatial effect to the visual representation of the selected item (21) and applying a spatial audio effect corresponding to the visual spatial effect to the reproduction of the audio segment. The device of the invention comprises electronic circuitry which is operative to perform the method of the invention. The computer program product of the invention enables a programmable device to perform the method of the invention.

Abstract: A communication system includes at least a sound re-production system (13-16,18-20) for audibly reproducing sound communicated by one user to another. A method of controlling communications between at least one first and at least one second user of the communication system includes adjusting the sound reproduction system (13-16,18-20) so as to cause an apparent distance between the other user and a location of an origin of the reproduced sound as perceived by the other user to be adjusted. Data (23,25) representative of at least one indicator of at least an interpersonal relation of the at least one first user and the at least one second user is obtained. The apparent distance is determined at least in part according to a pre-determined functional relationship between an indicator of at least an interpersonal relation and a desired interpersonal distance.

Abstract: Method for differentiation between voices including 1) analyzing perceptually relevant signal properties of the voices, e.g. average pitch and pitch variance, 2) determining sets of parameters representing the signal properties of the voices, and finally 3) extracting voice modification parameters representing modified signal properties of at least some of the voices. Hereby it is possible to increase a mutual parameter distance between the voices, and thereby the perceptual difference between the voices, when the voices have been modified according to the voice modification parameters. Preferably most of or all voices are modified in order to limit the amount of modification of one parameter.

Abstract: According to an exemplary embodiment of the invention, a device (100) for processing audio data (101, 102) is provided, wherein the device (100) comprises a manipulation unit (103) (particularly a resampling unit) adapted for manipulating (particularly for resampling) selectively a transition portion of a first audio item (104) in a manner that a time-related audio property of the transition portion is modified (particularly, it is possible to simulate also the temporal delay effects of movement in a realistic manner).

Abstract: A method of modifying an audio signal comprises the steps of analyzing the input audio signal (x) so as to produce a set of filter parameters (p) and a residual signal (r), modifying the set of filter parameters (p) so as to produce a modified set of filter parameters (p?), and synthesizing an output audio signal (y) using the modified set of filter parameters (p?) and the residual signal (r). The set of filter parameters (p) comprises poles (?A) and coefficients (a; c). The step of modifying the filter parameters (p) involves interpolating lattice filter reflection coefficients (c) so as to scale the spectral envelope of the audio signal.