Abstract: An upmixer for upmixing a downmix audio signal into an upmixed audio signal describing one or more upmixed audio channels includes a parameter applier configured to apply upmixing parameters to upmix the downmix audio signal in order to obtain the upmixed audio signal. The parameter applier is configured to apply a phase shift to the downmix audio signal to obtain a phase-shifted version of the downmix audio signal, while leaving a decorrelated signal unmodified by the phase shift. The parameter applier is further configured to combine the phase-shifted version of the downmix audio signal with the decorrelated signal to obtain the upmixed audio signal.

Abstract: An apparatus for upmixing a downmix audio signal describing one or more downmix audio channels into an upmixed audio signal describing a plurality of upmixed audio channels includes an upmixer configured to apply temporally variable upmixing parameters to upmix the downmix audio signal in order to obtain the upmixed audio signal. The apparatus also includes a parameter interpolator, wherein the parameter interpolator is configured to obtain one or more temporally interpolated upmix parameters to be used by the upmixer on the basis of a first complex-valued upmix parameter and a subsequent second complex-valued upmix parameter.

Abstract: An apparatus for generating a synthesis audio signal using a patching control signal has a first converter, a spectral domain patch generator, a high frequency reconstruction manipulator and a combiner. The first converter is configured for converting a time portion of an audio signal into a spectral representation. The spectral domain patch generator is configured for performing a plurality of different spectral domain patching algorithms, wherein each patching algorithm generates a modified spectral representation having spectral components in an upper frequency band derived from corresponding spectral components in a core frequency band of the audio signal.

Abstract: Parameters being a measure for a characteristic of a channel or of a pair of channels, wherein the parameter is a measure for a characteristic of the channel or of the pair of channels with respect to another channel of a multi-channel signal can be quantized more efficiently using a quantization rule that is generated based on a relation of an energy measure of the channel or the pair of channels and an energy measure of the multi-channel signal. With generation of the quantization rule taking into account a psycho acoustic approach, the size of an encoded representation of the multi-channel signal can be decreased by coarser quantization without significantly disturbing the perceptual quality of the multi-channel signal when reconstructed from the encoded representation.

Abstract: Binaural rendering a multi-channel audio signal into a binaural output signal is described. The multi-channel audio signal has a stereo downmix signal into which a plurality of audio signals are downmixed, and side information having a downmix information, as well as object level information of the plurality of audio signals and inter-object cross correlation information. Based on a first rendering prescription, a preliminary binaural signal is computed from the first and second channels of the stereo downmix signal. A decorrelated signal is generated as an perceptual equivalent to a mono downmix of the first and second channels of the stereo downmix signal being, however, decorrelated to the mono downmix. Depending on a second rendering prescription, a corrective binaural signal is computed from the decorrelated signal and the preliminary binaural signal is mixed with the corrective binaural signal to obtain the binaural output signal.

Abstract: An apparatus for upmixing a downmix audio signal describing one or more downmix audio channels into an upmixed audio signal describing a plurality of upmixed audio channels includes an upmixer and a parameter determinator. The upmixer is configured to apply temporally variable upmix parameters to upmix the downmix audio signal in order to obtain the upmixed audio signal, wherein the temporally variable upmix parameters include temporally variable smoothened phase values. The parameter determinator is configured to obtain one or more temporally smoothened upmix parameters for usage by the upmixer on the basis of a quantized upmix parameter input information. The parameter determinator is configured to combine a scaled version of a previous smoothened phase value with a scaled version of an input phase information using a phase change limitation algorithm, to determine a current smoothened phase value on the basis of the previous smoothened phase value and the phase input information.

Abstract: In the transition into the logarithmic range, not the entire bit width of the result linearly dependent upon the square of the value must be considered. Rather, it is possible to scale the result of a value with x bits such that a representation with less than x bits of the result is sufficient to receive the logarithmic representation based thereon. The effect of the scaling factor on the resulting logarithmic representation may be compensated for by adding or subtracting a correction value received by the logarithm function applied to the scaling factor to or from the scaled logarithmic representation without any loss of dynamics. This way, a method and an apparatus for creating a representation of a result linearly dependent upon a square of a value are provided so that the calculation is simple and/or possible with little hardware expenditure.

Abstract: A device for generating a binaural signal based on a multi-channel signal representing a plurality of channels and intended for reproduction by a speaker configuration having a virtual sound source position associated to each channel, is described. It includes a correlation reducer for differently processing, and thereby reducing a correlation between, at least one of a left and a right channel of the plurality of channels, a front and a rear channel of the plurality of channels, and a center and a non-center channel of the plurality of channels, in order to obtain an inter-similarity reduced set of channels; a plurality of directional filters, a first mixer for mixing outputs of the directional filters modeling the acoustic transmission to the first ear canal of the listener, and a second mixer for mixing outputs of the directional filters modeling the acoustic transmission to the second ear canal of the listener.

Abstract: An audio encoder has a first information sink oriented encoding branch such as a spectral domain encoding branch, a second information source or SNR oriented encoding branch such as an LPC-domain encoding branch, and a switch for switching between the first encoding branch and the second encoding branch, wherein the second encoding branch has a converter into a specific domain different from the spectral domain such as an LPC analysis stage generating an excitation signal, and wherein the second encoding branch furthermore has a specific domain coding branch such as LPC domain processing branch, and a specific spectral domain coding branch such as LPC spectral domain processing branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch.

Abstract: An apparatus for encoding includes a first domain converter, a switchable bypass, a second domain converter, a first processor and a second processor to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms. Corresponding decoding stages in the decoder together with a bypass for bypassing a domain converter allow the generation of a decoded audio signal with high quality and low bit rate.

Abstract: An audio encoder has a common preprocessing stage, an information sink based encoding branch such as spectral domain encoding branch, a information source based encoding branch such as an LPC-domain encoding branch and a switch for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage. An audio decoder has a spectral domain decoding branch, an LPC-domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.

Abstract: The present invention is based on the finding that parameters including a first set of parameters of a representation of a first portion of an original signal and including a second set of parameters of a representation of a second portion of the original signal can be efficiently encoded, when the parameters are arranged in a first sequence of tuples and in a second sequence of tuples, wherein the first sequence of tuples comprises tuples of parameters having two parameters from a single portion of the original signal and wherein the second sequence of tuples comprises tuples of parameters having one parameter from the first portion and one parameter from the second portion of the original signal. An efficient encoding can be achieved using a bit estimator to estimate the number of necessary bits to encode the first and the second sequence of tuples, wherein only the sequence of tuples is encoded, that results in the lower number of bits.

Abstract: An efficient encoded representation of a first and a second input audio signal can be derived using correlation information indicating a correlation between the first and the second input audio signals, when a signal characterization information, indicating at least a first or a second, different characteristic of the input audio signal is additionally considered. Phase information indicating a phase relation between the first and the second input audio signals is derived, when the input audio signals have the first characteristic. The phase information and a correlation measure are included into the encoded representation when the input audio signals have the first characteristic, and only the correlation information is included into the encoded representation when the input audio signals have the second characteristic.

Abstract: The present invention is based on the finding that parameters including: a first set of parameters of a representation of a first portion of an original signal and a second set of parameters of a representation of a second portion of the original signal can be efficiently encoded when the parameters are arranged in a first sequence of tuples and a second sequence of tuples. The first sequence of tuples includes tuples of parameters having two parameters from a single portion of the original signal and the second sequence of tuples includes tuples of parameters having one parameter from the first portion and one parameter from the second portion of the original signal. A bit estimator estimates the number of necessary bits to encode the first and the second sequence of tuples. Only the sequence of tuples, which results in the lower number of bits, is encoded.

Abstract: A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeaker configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters have an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.

Abstract: The present invention is based on the finding that an efficient code for encoding information values can be derived, when two or more information values are grouped in a tuple in a tuple order and when an encoding rule is used, that assigns the same code word to tuples having identical information values in different orders and that does derive an order information, indicating the tuple order, and when the code word is output in association with the order information.

Abstract: An apparatus for synthesizing a rendered output signal having a first audio channel and a second audio channel includes a decorrelator stage for generating a decorrelator signal based on a downmix signal, and a combiner for performing a weighted combination of the downmix signal and a decorrelated signal based on parametric audio object information, downmix information and target rendering information. The combiner solves the problem of optimally combining matrixing with decorrelation for a high quality stereo scene reproduction of a number of individual audio objects using a multichannel downmix.

Abstract: For postprocessing spectral values which are based on a first transformation algorithm for converting the audio signal into a spectral representation, first a sequence of blocks of the spectral values representing a sequence of blocks of samples of the audio signal are provided.

Abstract: A method for decoding a multi-audio-object signal having audio signals of first and second types encoded therein, the multi-audio-object signal having a downmix signal and side information having level information of the audio signals of the first and second types in a first predetermined time/frequency resolution, the method including computing a prediction coefficient matrix C based on the level information; and up-mixing the downmix signal based on the prediction coefficients to obtain a first and/or a second up-mix audio signal approximating the audio signals of the first and second types, respectively, wherein up-mixing yields the first and/or second up-mix signals S1 and S2 from the downmix signal d according to a computation representable by ( S 1 S 2 ) = D - 1 ? { ( 1 C ) ? d + H } , with “1” denoting—depending on the number of channels of d—a scalar, or an identity matrix, and D?1 being a matrix uniquely determined by a downmix prescription according

Abstract: An audio decoder for decoding a multi-audio-object signal having an audio signal of a first type and an audio signal of a second type encoded therein is described, the multi-audio-object signal having a downmix signal and side information, the side information having level information of the audio signals of the first and second types in a first predetermined time/frequency resolution, and a residual signal specifying residual level values in a second predetermined time/frequency resolution, the audio decoder having a processor for computing prediction coefficients based on the level information; and an up-mixer for up-mixing the downmix signal based on the prediction coefficients and the residual signal to obtain a first up-mix audio signal approximating the audio signal of the first type and/or a second up-mix audio signal approximating the audio signal of the second type.