Abstract: In processing a multi-channel audio signal having at least three original channels, a first downmix channel and a second downmix channel are provided, which are derived from the original channels. For a selected original channel of the original channels, channel side information are calculated such that a downmix channel or a combined downmix channel including the first and the second downmix channels, when weighted using the channel side information, results in an approximation of the selected original channel. The channel side information and the first and second downmix channels form output data to be transmitted to a decoder, which, in case of a low level decoder only decodes the first and second downmix channels or, in case of a high level decoder provides a full multi-channel audio signal based on the downmix channels and the channel side information.

Abstract: In processing a multi-channel audio signal having at least three original channels, a first downmix channel and a second downmix channel are provided, which are derived from the original channels. For a selected original channel of the original channels, channel side information are calculated such that a downmix channel or a combined downmix channel including the first and the second downmix channels, when weighted using the channel side information, results in an approximation of the selected original channel. The channel side information and the first and second downmix channels form output data to be transmitted to a decoder, which, in case of a low level decoder only decodes the first and second downmix channels or, in case of a high level decoder provides a full multi-channel audio signal based on the downmix channels and the channel side information.

Abstract: An apparatus for generating a multi-channel output signal performs a center channel cancellation to obtain improved base channels for reconstructing left-side output channels or right-side output channels. In particular, the apparatus includes a cancellation channel calculator for calculating a cancellation channel using information related to the original center channel available at the decoder. The device furthermore includes a combiner for combining a transmission channel with the cancellation channel. Finally, the apparatus includes a reconstructor for generating the multi-channel output signal. Due to the center channel cancellation, the channel reconstructor not only uses a different base channel for reconstructing the center channel but also uses base channels different from the transmission channels for reconstructing left and right output channels which have a reduced or even completely cancelled influence of the original center channel.

Abstract: According to the present invention, multiple parametrically encoded audio signals can be efficiently combined using an audio signal generator, which generates an audio output signal by combining the down-mix channels and the associated parameters of the audio signals directly within the parameter domain, i.e. without reconstructing or decoding the individual input audio signals prior to the generation of the audio output signal. This is achieved by direct mixing of the associated down-mix channels of the individual input signals. It is one key feature of the present invention that the combination of the down-mix channels is achieved by simple, computationally inexpensive arithmetic operations.

Abstract: For determining an estimate of a need for information units for encoding a signal, a measure for the distribution of the energy in the frequency band is taken into account in addition to the admissible interference for a frequency band and an energy of the frequency band. With this, a better estimate of the need for information units is obtained, so that coding can be done more efficiently and more accurately.

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: In a multi-channel encoder generating several different parameter sets for reconstructing a multi-channel output signal using at least one transmission channel, the data stream is written such that the two parameter sets are decodable independently of each other. Thus, a multi-channel decoder is enabled to skip a parameter set which is marked as optional and/or has a higher version number when reading the data stream and still to perform a valid multi-channel reconstruction using a data set marked as mandatory or a data set having a sufficiently low version number. This achieves a flexible encoder/decoder concept suitable for future updates characterized by backward compatibility and reliability.

Abstract: For flexibly signaling a synchronous mode or an asynchronous mode in the multi-channel parameter reconstruction, a parameter configuration cue is inserted in the data stream, which is used by a configurator on the side of a multi-channel decoder to configure a multi-channel reconstructor. If the parameter configuration cue has a first meaning, the configurator will look for further configuration information in its input data, while, when the parameter configuration cue has another meaning, the configurator performs a configuration setting of the multi-channel reconstructor based on information on a coding algorithm with which transmission channel data have been coded, so that it is ensured efficiently on the one hand and flexibly on the other hand that there will always be obtained a correct association between parameter data and decoded transmission channel data.

Abstract: For determining an estimate of a need for information units for encoding a signal, a measure for the distribution of the energy in the frequency band is taken into account in addition to the admissible interference for a frequency band and an energy of the frequency band. With this, a better estimate of the need for information units is obtained, so that coding can be done more efficiently and more accurately.

Abstract: The purpose of the invention is to bridge the gap between parametric multi-channel audio coding and matrixed-surround multi-channel coding by gradually improving the sound of an up-mix signal while raising the bit-rate consumed by the side-information starting from 0 up to the bit-rates of the parametric methods. More specifically, it provides a method of flexibly choosing an “operating point” somewhere between matrixed-surround (no side-information, limited audio quality) and fully parametric reconstruction (full side-information rate required, good quality). This operating point can be chosen dynamically (i.e. varying over time) and in response to the permissible side-information rate, as it is dictated by the individual application.

Abstract: The purpose of the invention is to bridge the gap between parametric multi-channel audio coding and matrixed-surround multi-channel coding by gradually improving the sound of an up-mix signal while raising the bit-rate consumed by the side-information starting from 0 up to the bit-rates of the parametric methods. More specifically, it provides a method of flexibly choosing an “operating point” somewhere between matrixed-surround (no side-information, limited audio quality) and fully parametric reconstruction (full side-information rate required, good quality). This operating point can be chosen dynamically (i.e. varying over time) and in response to the permissible side-information rate, as it is dictated by the individual application.

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: A compact encoded representation of information values not exceeding a predefined size can be derived when a first encoding rule generating an encoded representation of the information values of variable length is compared to a second encoding rule generating an encoded representation of the information values of fixed length and when the encoding rule resulting in the encoded representation requiring the lower number of information units is chosen. Thus, the maximum bit rate can be guaranteed to be at least the maximum bit rate of the second encoding rule deriving the second encoded representation. Signaling the choice of the encoding rule by some rule information together with the encoded representation of the information values, the correct information values can later on be derived on a decoder side, using a decoding rule fitting the encoding rule used during the encoding.

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: 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 generating a multi-channel output signal performs a center channel cancellation to obtain improved base channels for reconstructing left-side output channels or right-side output channels. In particular, the apparatus includes a cancellation channel calculator for calculating a cancellation channel using information related to the original center channel available at the decoder. The device furthermore includes a combiner for combining a transmission channel with the cancellation channel. Finally, the apparatus includes a reconstructor for generating the multi-channel output signal. Due to the center channel cancellation, the channel reconstructor not only uses a different base channel for reconstructing the center channel but also uses base channels different from the transmission channels for reconstructing left and right output channels which have a reduced or even completely cancelled influence of the original center channel.

Abstract: A multi-channel synthesizer includes a post processor for determining post processed reconstruction parameters or quantities derived from the reconstruction parameter for an actual time portion of the input signal so that the post processed reconstruction parameter or the post processed quantity is different from the corresponding quantized and inversely quantized reconstruction parameter in that the value of the post processed reconstruction parameter or the derived quantity is not bound by the quantization step size. A multi-channel reconstructor uses the post-processed reconstruction parameter for reconstructing the multi-channel output signal.

Abstract: In processing a multi-channel audio signal having at least three original channels, a first downmix channel and a second downmix channel are provided, which are derived from the original channels. For a selected original channel of the original channels, channel side information are calculated such that a downmix channel or a combined downmix channel including the first and the second downmix channels, when weighted using the channel side information, results in an approximation of the selected original channel. The channel side information and the first and second downmix channels form output data to be transmitted to a decoder, which, in case of a low level decoder only decodes the first and second downmix channels or, in case of a high level decoder provides a full multi-channel audio signal based on the downmix channels and the channel side information.

Abstract: A method for detecting a transient in a discrete-time audio signal is performed completely in the time domain and includes the step of segmenting the discrete-time audio signal so as to generate consecutive segments of the same length with unfiltered discrete-time audio signals xs(T−1). The discrete-time audio signal in a current segment is subsequently filtered. Then either the energy of the filtered discrete-time audio signal in the current segment can be compared with the energy of the filtered discrete-time audio signal in a preceding segment or a current relationship between the energy of the filtered discrete-time audio signal in the current segment and the energy of the unfiltered discrete-time audio signal in the current segment can be formed and this current relationship compared with a preceding corresponding relationship. On the basis of the one and/or the other of these comparisons it is detected whether a transient is present in the discrete-time audio signal.

Abstract: In an inventive method for encrypting a discrete signal consisting of successive samples the successive samples are subdivided into successive time blocks, and the successive time blocks are then encoded into encoded data blocks having a predetermined order. Subsequently, the predetermined order of the encoded data blocks is altered in accordance with a predetermined interchange specification. The underlying findings are that a very high level of security of the encryption may be achieved by introducing temporal discontinuity, and that the occurrence of errors in unauthorized processing of signals encoded in such a manner maybe prevented, and the compatibility with standard codings may be ensured by performing the alteration of the chronological order in accordance with a coding of the discrete signal, i.e. with regard to encoded data blocks into which an encoder encodes the discrete signal.