Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder.

An apparatus receives a first audio signal captured by a first microphone of a device and at least a second audio signal captured by at least a second microphone of the device. The apparatus estimates a diffuseness of sound based on the received first and at least second audio signals. The apparatus may then form at least one final audio signal based on at least one of the received first audio signal and the received at least second audio signal by adjusting an audibility of diffuse sound for the final audio signal in response to the estimated diffuseness, in order to enable an enhanced perception of sound with respect to at least one criterion with the at least one final audio signal.

An audio system is proposed, dynamically playing optimized audio signals based on user position. A sensor circuits dynamically senses a target space to generate field context information. First speaker and second speaker are arranged for audio playback. A host device recognizes a user from the field context information, determines the user position corresponding to the target space, and adaptively assigns the user position as a target listening spot. A sensor circuit contains a camera capturing an ambient image out of the target space. A control circuit utilizes a user interface circuit to perform a configuration procedure which determines location, size and acoustic attribute information of an ambient object, allowing the control circuit to accordingly perform an object-based compensation operation on the target listening spot to generate optimized first channel audio signal and second channel audio signal.

Some audio processing methods may involve receiving audio data corresponding to a plurality of audio channels and determining audio characteristics of the audio data, which may include transient information. An amount of decorrelation for the audio data may be based, at least in part, on the audio characteristics. If a definite transient event is determined, a decorrelation process may be temporarily halted or slowed. Determining transient information may involve evaluating the likelihood and/or the severity of a transient event. In some implementations, determining transient information may involve evaluating a temporal power variation in the audio data. Explicit transient information may or may not be received with the audio data, depending on the implementation. Explicit transient information may include a transient control value corresponding to a definite transient event, a definite non-transient event or an intermediate transient control value.

The present invention relates to a method for generating a filter for an audio signal and a parameterization device for the same, and more particularly, to a method for generating a filter for an audio signal, to implement filtering of an input audio signal with a low computational complexity, and a parameterization device therefor. To this end, provided are a method for generating a filter for an audio signal, including: receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of an input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining average reverberation time information of a corresponding subband by using reverberation time information extracted from the subband filter coefficients; obtaining at least one coefficient for curve fitting of the obtained average reverberation time information; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; obtaining filter order information for determining a truncation length of the subband filter coefficients, the filter order information being obtained by using the average reverberation time information or the at least one coefficient according to the obtained flag information and the filter order information of at least one subband being different from filter order information of another subband; and truncating the subband filter coefficient by using the obtained filter order information and a parameterization device therefor.

There is inter alia disclosed an apparatus for spatial audio encoding comprising: means for analysing a plurality of spatial audio parameter sets associated with a frame of one or more audio signals, wherein the plurality of spatial audio parameter sets are associated with a plurality of subframes, a plurality of frequency sub bands and a plurality of sound source directions for the frame of the one or more audio signals; and means for determining from the analysis of the plurality of spatial audio parameter sets at least one spatial audio parameter set for subframes of the frame of the one or more audio signals.

An apparatus, for enabling adaptive playback, comprising means configured to: obtain, for a first point of view, a first audio signal for at least a first channel and a second channel; obtain, for a second point of view, a second audio signal for at least the first channel and the second channel; determine a single-channel difference audio signal, for the second point of view, based on at least a difference between the first audio signal and the second audio signal; and enable estimation of both the first channel and the second channel of the second audio signal for the second point of view in dependence on the single-channel difference audio signal and the first audio signal.

In one embodiment, a system for providing three-dimensional (3D) immersive sound is provided. The system includes a loudspeaker and at least one controller. The loudspeaker transmits an audio output signal in a listening environment. The at least one controller is programmed to store a plurality of directional bands with each directional band being defined by a narrowband frequency interval and to store at least psychoacoustic scale including a sub-band for each directional band. The at least one controller is further programmed to determine an energy for the sub-band and generate a loudspeaker driving signal based at least on the energy for the sub-band to drive the loudspeaker to transmit the audio output signal.

A device for generating a multitude of loudspeaker signals based on a virtual source object which has a source signal and a meta information determining a position or type of the virtual source object. The device has a modifier configured to time-varyingly modify the meta information. In addition, the device has a renderer configured to transfer the virtual source object and the modified meta information to form a multitude of loudspeaker signals.

A signal processor includes an input unit that receives a first audio signal and a second audio signal including mutually correlated components, a delay unit that delays the first audio signal received at the input unit by a prescribed delay time, a synthesis unit that synthesizes the first audio signal having been delayed by the delay unit with the second audio signal received at the input unit, and outputs a third audio signal resulting from synthesis, and a frequency band restriction unit that restricts a level of the first audio signal before the synthesis in a prescribed frequency band including a frequency of a dip occurring at a lowest frequency among a plurality of dips occurring in a frequency characteristic of the third audio signal as a result of the synthesis performed by the synthesis unit.