Abstract: The apparatus for constructing a multi-channel output signal using an input signal and parametric side information, the input signal including the first input channel and the second input channel derived from an original multi-channel signal, and the parametric side information describing interrelations between channels of the multi-channel original signal uses base channels for synthesizing first and second output channels on one side of an assumed listener position, which are different from each other. The base channels are different from each other because of a coherence measure. Coherence between the base channels (for example the left and the left surround reconstructed channel) is reduced by calculating a base channel for one of those channels by a combination of the input channels, the combination being determined by the coherence measure. Thus, a high subjective quality of the reconstruction can be obtained because of an approximated original front/back coherence.

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: Prior to embedding a watermark in an audio signal, a spectral representation of the audio signal and a spectral representation of the watermark signal are determined. The spectral representation of the watermark signal is then processed on the basis of a psychoacoustic masking threshold of the audio signal. The processed watermark signal is combined with the audio signal to obtain an audio signal bearing a watermark.

Abstract: On an encoder-side, a multi-channel input signal is analyzed for obtaining smoothing control information, which is to be used by a decoder-side multi-channel synthesis for smoothing quantized transmitted parameters or values derived from the quantized transmitted parameters for providing an improved subjective audio quality in particular for slowly moving point sources and rapidly moving point sources having tonal material such as fast moving sinusoids.

Abstract: For synthesizing at least three output channels using two stereo input channels, the stereo input channels are analyzed to detect signal components occurring in both input channels. A signal generator is operative to introduce at least a part of the detected signal components into the second channel associated with a second speaker in an intended speaker scheme, which is positioned between a first and a third speaker in the speaker scheme. When, however, feeding of the complete detected signal components would result in a clipping situation, then only a part of the detected signal components is fed into the second channel as a real center channel and the remainder is located in the first and third channels as a phantom center channel.

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: The envelope of a decorrelated signal derived from an original signal can be shaped without introducing additional distortion, when a spectral flattener is used to spectrally flatten the spectrum of the decorrelated signal and the original signal prior to using the flattened spectra for deriving a gain factor describing the energy distribution between the flattened spectra, and when the so derived gain factor is used by an envelope shaper to timely shape the envelope of the decorrelated 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: 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: A parametric representation of a multi-channel audio signal describes the spatial properties of the audio signal well with compact side information when a coherence information, describing the coherence between a first and a second channel, is derived within a hierarchical encoding process only for channel pairs including a first channel having only information of a left side with respect to a listening position and including a second channel having only information from a right side with respect to a listening position. As within the hierarchical process the multiple audio channels of the audio signal are downmixed iteratively into monophonic channels, one can pick the relevant parameters from an encoding step involving only channel pairs carrying the information needed to describe the spatial properties of the multi-channel audio signal.

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: An apparatus for analyzing an audio signal with regard to rhythm information of the audio signal by using an autocorrelation function comprises a filter bank for separating the audio signal into at least two sub-band signals. The sub-band signals are examined with regard to periodicities by an autocorrelation function, to obtain rhythm raw-information for the at least two sub-band signals. To reduce or eliminate the ambiguities of the autocorrelation function for periodical signals, the rhythm raw-information is postprocessed to obtain post-processed rhythm raw-information for the sub-band signal. The rhythm information of the audio signal is established based on the postprocessed rhythm raw-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: In order to analyze an information signal, a significant short-time spectrum is extracted from the information signal, the means for extracting being configured to extract such short-time spectra which come closer to a specific characteristic than other short-time spectra of the information signal. The short-time spectra extracted are then decomposed into component signals using ICA analysis, a component signal spectrum representing a profile spectrum of a tone source which generates a tone corresponding to the characteristic sought for. From a sequence of short-time spectra of the information signal and from the profile spectra determined, an amplitude envelope is eventually calculated for each profile spectrum, the amplitude envelope indicating how a profile spectrum of a tone source all in all changes over time.

Abstract: The apparatus for constructing a multi-channel output signal using an input signal and parametric side information, the input signal including the first input channel and the second input channel derived from an original multi-channel signal, and the parametric side information describing interrelations between channels of the multi-channel original signal uses base channels for synthesizing first and second output channels on one side of an assumed listener position, which are different from each other. The base channels are different from each other because of a coherence measure. Coherence between the base channels (for example the left and the left surround reconstructed channel) is reduced by calculating a base channel for one of those channels by a combination of the input channels, the combination being determined by the coherence measure. Thus, a high subjective quality of the reconstruction can be obtained because of an approximated original front/back coherence.

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 a method for characterizing a signal representing an audio content a measure is determined for a tonality of the signal, whereupon a statement is made about the audio content of the signal on the basis of the measure for the tonality of the signal. The measure for the tonality is derived from a quotient whose numerator is the mean of the summed values of spectral components of the signal exponentiated with a first power and whose denominator is the mean of the summed values of spectral components exponentiated with a second power, the first and second powers differing from each other. The measure for the tonality of the signal for the content analysis is robust in relation to a signal distortion, due e.g. to MP3 coding, and has a high correlation with the content of the analysed signal.

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 order to obtain an integer transform, which provides integer output values, the TDAC function of a MDCT is explicitly carried out in the time domain before the forward transform. In overlapping windows, this results in a Givens rotation which may be represented by lifting matrices, wherein time-discrete sampled values of an audio signal may at first be summed up on a pair-wise basis to build a vector so as to be sequentially provided with a lifting matrix. In accordance with the invention, after each multiplication of a vector by a lifting matrix, a rounding step is carried out such that, on the output-side, only integers will result. By transforming the windowed integer sampled value with an integer transform, a spectral representation with integer spectral values may be obtained. The inverse mapping with an inverse rotation matrix and corresponding inverse lifting matrices results in an exact reconstruction.

Abstract: Prior to embedding a watermark in an audio signal, a spectral representation of the audio signal and a spectral representation of the watermark signal are determined. The spectral representation of the watermark signal is then processed on the basis of a psychoacoustic masking threshold of the audio signal. The processed watermark signal is combined with the audio signal to obtain an audio signal bearing a watermark.