Abstract: An audio system comprises a receiver (301) for receiving an audio signal, such as an audio object or a signal of a channel of a spatial multi-channel signal. A binaural circuit (303) generates a binaural output signal by processing the audio signal. The processing is representative of a binaural transfer function providing a virtual sound source position for the audio signal. A measurement circuit (307) generating measurement data indicative of a characteristic of the acoustic environment and a determining circuit (311) determines an acoustic environment parameter in response to the measurement data. The acoustic environment parameter may typically be a reverberation parameter, such as a reverberation time. An adaptation circuit (313) adapts the binaural transfer function in response to the acoustic environment parameter. For example, the adaptation may modify a reverberation parameter to more closely resemble the reverberation characteristics of the acoustic environment.

Abstract: A stereo audio decoder generates a set of stereo output channels in response to a parametric audio input including signal parameters and stereo related parameters. A parameter processor generates two different set of parameters based on the input signal parameters thus up-mixing the signal parameters by altering or manipulating the signal parameters corresponding to the stereo related parameters. The two different parameters are synthesized by separate signal synthesizers to form respective stereo output channels. The signal synthesizers may be sinusoidal synthesizers, and the decoder also includes transient and noise synthesizers to generate transient and noise signal portions to be applied to the stereo output channels. Further, different transient and noise signal portions to the output channels may be provided by applying different gains based on the stereo related parameter. The two different parameters may be determined from current and previous signal parameter inputs using an input delay line.

Abstract: Multi-channel audio signals are coded into a monaural audio signal and information allowing to recover the multi-channel audio signal from the monaural audio signal and the information. The information is generated by determining a first portion of the information for a first frequency region of the multi-channel audio signal, and by determining a second portion of the information for a second frequency region of the multi-channel audio signal. The second frequency region is a portion of the first frequency region and thus is a sub-range of the first frequency region. The information is multi-layered enabling a scaling of the decoding quality versus bit rate.

Abstract: An audio encoder adapted to encode a multi-channel audio signal. The encoder comprises an encoder combination module (ECM) for generating a dominant signal part and a residual signal part being a combined representation of first and second audio signals, the dominant and residual signal parts being obtained by applying a mathematical procedure to the first and second audio signals. The mathematical procedure involves a spatial parameter comprising a description of spatial properties of the first and second audio signals. Embodiments include a plurality of interconnected encoder combination module, so that e.g. six independent 5.1 format audio signals can be encoded to a single or two dominant signal parts and a number of parameter sets and residual signal parts.

Abstract: A multi-channel audio encoder (10) for encoding a multi-channel audio signal (101), e.g. a 5.1 channel audio signal, into a spatial down-mix (102), e.g. a stereo signal, and associated parameters (104, 105). The encoder (10) comprises first and second units (110, 120). The first unit (110) encodes the multi-channel audio signal (101) into the spatial down-mix (102) and parameters (104). These parameters (104) enable a multi-channel decoder (20) to reconstruct the multi-channel audio signal (203) from the spatial down-mix (102). The second unit (120) generates, from the spatial down-mix (102), parameters (105) that enable the decoder to reconstruct the spatial down-mix (202) from an alternative down-mix (103), e.g. a so-called artistic down-mix that has been manually mixed in a sound studio. In this way, the decoder (20) can efficiently deal with a situation in which an alternative down-mix (103) is received instead of the regular spatial, down-mix (102).

Abstract: Synthesizing an output audio signal is provided on the basis of an input audio signal, the input audio signal comprising a plurality of input sub-band signals, wherein at least one input sub-band signal is transformed (T) from the sub-band domain to the frequency domain to obtain at least one respective transformed signal, wherein the at least one input sub-band signal is delayed and transformed (D, T) to obtain at least one respective transformed delayed signal, wherein at least two processed signals are derived (P)from the at least one transformed signal and the at least one transformed delayed signal, wherein the processed signals are inverse transformed (T?1) from the frequency domain to the sub-band domain to obtain respective processed sub-band signals, and wherein the output audio signal is synthesized from the processed sub-band signals.

Abstract: An encoder for a multi-channel audio signal which comprises a down-mixer (201, 203, 205) for generating a down-mix as a combination of at least a first and second channel signal weighted by respectively a first and second weight with different amplitudes for at least some time-frequency intervals. Furthermore, a circuit (201, 203, 209) generates up-mix parametric data characterizing a relationship between the channel signals as well as characterizing the weights. A circuit generates weight estimates for the encoder weights from the up-mix parametric data; and comprises an up-mixer (407) which recreates the multi channel audio signal by up-mixing the down-mix in response to the up-mix parametric data, the first weight estimate and the second weight estimate. The up-mixing is dependent on the amplitude of at least one of the weight estimate(s).

Abstract: Encoding an audio signal is provided wherein the audio signal includes a first audio channel and a second audio channel, the encoding comprising subband filtering each of the first audio channel and the second audio channel in a complex modulated filterbank to provide a first plurality of subband signals for the first audio channel and a second plurality of subband signals for the second audio channel, downsampling each of the subband signals to provide a first plurality of downsampled subband signals and a second plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, deriving spatial parameters from the sub-subband signals and from those downsampled subband signals that are not further subband filtered, and deriving a single channel audio signal comprising derived subband signals derived from the first plurality of downsampled subband signals and the second plurality of

Abstract: An encoder (109) comprises a receiver (201) which receives a time domain audio signal. A filter bank (203) generates a first subband signal from the time domain audio signal where the first subband signal corresponds to a non-critically sampled complex subband domain representation of the time domain signal. A conversion processor (205) generates a second subband signal from the first subband signal by subband processing. The second subband signal corresponds to a critically sampled complex subband domain representation of the time domain audio signals. An encode processor (207) then generates a waveform encoded data stream by encoding data values of the second subband signal. The conversion processor (205) generates the second subband signal by direct subband conversion without converting back to the time domain. The invention allows an oversampled subband signal typically generated in parametric encoding to be waveform encoded with reduced complexity. A decoder performs the inverse operation.

Abstract: A method of encoding input signals (l, r) to generate encoded data (100) is provided. The method involves processing the input signals (l, r) to determine first parameters (?1, ?2) describing relative phase difference and temporal difference between the signals (l, r), and applying these first parameters (?1, ?2) to process the input signals to generate intermediate signals. The method involves processing the intermediate signals to determine second parameters (?; IID, ?) describing angular rotation of the first intermediate signals to generate a dominant signal (m) and a residual signal (s), the dominant signal (m) having a magnitude or energy greater than that of the residual signal (s). These second parameters are applicable to process the intermediate signals to generate the dominant (m) and residual (s) signals.

Abstract: An encoding device (1) and method convert a set of signals (l, r) into a dominant signal (m) containing most signal energy, a residual signal (s) containing a remainder of the signal energy, and signal parameters (IID, ICC) associated with the conversion. The dominant signal (m) and selected parts of the residual signal (s) are encoded. Selecting parts of the residual signal involves a residual signal (s?) passing perceptually relevant parts of the residual signal (s), attenuating perceptually less relevant parts of the residual signal and suppressing least relevant parts of the residual signal. An associated decoding device (2) and method decode the encoded dominant signal and the encoded residual signal so as to produce a decoded dominant signal (m?u) and a decoded residual signal (s?mod) respectively. A synthetic residual signal (s?syn) is derived from the decoded dominant signal (m?u) and is attenuated so as to produce an attenuated synthetic residual signal (s?syn,mod).

Abstract: Techniques are described for combining parametric multi-channel audio coding with matrixing, reconstructing a full-quality multi-channel, independent of the decoder. A stereo signal is obtained from encoding an N-channel audio signal into spatial parameters and a stereo down-mix signal having first and second stereo signals, including adding a first signal and a third signal to obtain a first output signal, the first signal having the first stereo signal modified by a first complex function, the third signal having the second stereo signal modified by a third complex function. A second signal and fourth signal are similarly added to obtain a second output signal. Complex functions are chosen such that an energy value of the difference between first signal and the second signals (fourth signal and third signals) is larger than or equal to the energy value of the sum of the first and the second signal (fourth signal and third signal).

Abstract: A method and a device are described for processing a stereo signal obtained from an encoder, which encodes an N-channel audio signal into spatial parameters (P) and a stereo down-mix comprising first and second stereo signals (LO, RO). A first signal and a third signal are added in order to obtain a first output signal (L0w), wherein the first signal (L0wL) comprises the first stereo signal (LO) modified by a first complex function (g1), and the third signal (L0wR) comprises the second stereo signal (RO) modified by a third complex function (g3). A second signal and a fourth signal are added to obtain a second output signal (R0w). The fourth signal (R0wR) comprises the second stereo signal (RO) modified by a fourth complex function (g4), and the second signal (R0wL) comprises the first stereo signal (LO) modified by a second complex function (g2).

Abstract: An audio encoder adapted to encode a multi-channel audio signal. The encoder comprises an encoder combination module (ECM) for generating a dominant signal part and a residual signal part being a combined representation of first and second audio signals, the dominant and residual signal parts being obtained by applying a mathematical procedure to the first and second audio signals. The mathematical procedure involves a spatial parameter comprising a description of spatial properties of the first and second audio signals. Embodiments include a plurality of interconnected encoder combination module, so that e.g. six independent 5.1 format audio signals can be encoded to a single or two dominant signal parts and a number of parameter sets and residual signal parts.

Abstract: An audio synthesizing apparatus receives an encoded signal comprising a downmix signal and parametric extension data for expanding the downmix signal to a multi-sound source signal. A decomposition processor (205) performs a signal decomposition of the downmix signal to generate at least a first signal component and a second signal component, where the second signal component is at least partially decorrelated with the first signal component. A position processor (207) determines a first spatial position indication for the first signal component in response to the parametric extension data and a binaural processor (211) synthesizes the first signal component based on the first spatial position indication and the second signal component to originate from a different direction. The invention may provide improved spatial experience from e.g. headphones by using a direct synthesis of a main directional signal from the appropriate position rather than as a combination of signals from virtual loudspeaker positions.

Abstract: The invention proposes a method for embedding an ancillary data into a compressed audio signal. This is achieved by replacing Least Significant Bits (LSBs) in at least one frequency subband of the compressed audio signal by the ancillary data. When replacing LSB bits of compressed subband signals with the ancillary data, the subband signal is effectively modified, resulting in a different decoded output. The replaced LSB bits corresponding to the ancillary data are conveyed as part of the bitstream and can be easily extracted at the decoder. In such a way the decoder obtains the ancillary data that can be used for more advanced audio reproduction at the decoder. The compressed audio itself maintains a good audio quality despite the replacement of the LSB bits of the frequency subband, because the LSB bits do not contribute to the audible artefacts.

Abstract: Encoding an audio signal is provided wherein the audio signal includes a first audio channel and a second audio channel, the encoding comprising subband filtering each of the first audio channel and the second audio channel in a complex modulated filterbank to provide a first plurality of subband signals for the first audio channel and a second plurality of subband signals for the second audio channel, downsampling each of the subband signals to provide a first plurality of downsampled subband signals and a second plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, deriving spatial parameters from the sub-subband signals and from those downsampled subband signals that are not further subband filtered, and deriving a single channel audio signal comprising derived subband signals derived from the first plurality of downsampled subband signals and the second plurality of

Abstract: A head tracking system (400) is proposed in the present invention that determines a rotation angle (300) of a head (100b) of a user (100) with respect to a reference direction (310), which is dependent on a movement of a user (100). Here the movement of a user should be understood as an act or process of moving including e.g. changes of place, position, or posture, such as e.g. lying down or sitting in a relaxation chair. The head tracking system according to the invention comprises a sensing device (410) for measuring a head movement to provide a measure (401) representing the head movement, and a processing circuit (420) for deriving the rotation angle of the head of the user with respect to the reference direction from the measure. The reference direction (310) used in the processing circuit (420) is dependent on the movement of the user.

Abstract: An output signal is generated from an input signal by applying a send effect processing to the input signal. The input signal comprises a weighted sum of component signals. Dependencies between the weighted component signals are represented by parameters. In accordance with the present invention, the output signal is generated in dependence of the parameters to compensate for an unequal weighting of component signals comprised in the input signal. Due to this compensation the strength of the send effect corresponding to the separate component signals is (nearly) proportional to the strength of each of the component signals, which results in more realistic surround experience.

Abstract: An audio encoder for encoding a multi-channel audio signal includes an encoder combination module (ECM) for generating a dominant signal part (m) and a residual signal part (s) being a combined representation of first and second audio signals (x1, x2), the dominant and residual signal parts (m, s) being obtained by applying a mathematical procedure to the first and second audio signals (x1, x2), wherein the mathematical procedure involves a first spatial parameter (SP1) including a description of spatial properties of the first and second audio signals (x1, x2), a parameter generator (PG) for generating a first parameter (PS1) set including a second spatial parameter (SP2), and a second parameter (PS2) set including a third spatial parameter (SP3), and an output generator for generating an encoded output signal having a first output part (OP1) including the dominant signal part (m) and the first parameter set (PS1), and a second output part (OP2) including the residual signal part (s) and the second parameter