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: An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio 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, 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 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: 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: An apparatus and a method for generating a multi-channel synthesizer control signal, a multi-channel synthesizer, a method of generating an output signal from an input signal and a machine-readable storage medium are provided. 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: 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 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: 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: 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: 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: 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: 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: An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier for modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal.

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: A significant short-time spectrum is extracted from an information signal, the means for extracting being configured to extract such short-time spectra which come closer to a specific characteristic than others. 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. From a sequence of short-time spectra of the information signal and from the profile spectra determined, an amplitude envelope is calculated for each profile spectrum to indicate how a tone source profile spectrum changes over time. The profile spectra and all the amplitude envelopes associated therewith provide a description of the information signal which may be evaluated further, for example for transcription purposes in the case of a music signal.

Abstract: An integer transform, which provides integer output values, carries out the TDAC function of a MDCT 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. 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: 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 analyzed signal.

Abstract: For producing a fingerprint of an audio signal, use is made of information defining a plurality of predetermined fingerprint modi, all of the fingerprint modi relating to the same type of fingerprint, the fingerprint modi, however, providing different fingerprints differing from each other with regard to their data volume, on the one hand, and to their characterizing strength for characterizing the audio signal, on the other hand, the fingerprint modi being pre-determined such that a fingerprint in accordance with a fingerprint modus having a first characterizing strength is convertible to a fingerprint in accordance with a fingerprint modus having a second characterizing strength, without using the audio signal. A predetermined fingerprint modus of the plurality of predetermined fingerprint modi is set and subsequently used for computing a fingerprint using the audio 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.