Abstract: An embodiment of an analysis filterbank for filtering a plurality of time domain input frames, wherein an input frame comprises a number of ordered input samples, comprises a windower configured to generating a plurality of windowed frames, wherein a windowed frame comprises a plurality of windowed samples, wherein the windower is configured to process the plurality of input frames in an overlapping manner using a sample advance value, wherein the sample advance value is less than the number of ordered input samples of an input frame divided by two, and a time/frequency converter configured to providing an output frame comprising a number of output values, wherein an output frame is a spectral representation of a windowed frame.

Abstract: An apparatus for scalable encoding a spectrum of a signal including audio and/or video information, with the spectrum comprising binary spectral values, includes a means for generating a first sub-scaling layer and a second sub-scaling layer in addition to a means for forming the encoded signal, with the means for forming being implemented so as to include the first sub-scaling layer and the second sub-scaling layer into the encoded signal that the first and the second sub-scaling layer are separately decodable from each other. In contrast to a full-scaling layer, a sub-scaling layer includes only the bits of a certain order of a part of the binary spectral values in the band, so that, by additionally decoding a sub-scaling layer, a more finely controllable and a more finely scalable precision gain may be achieved.

Abstract: An audio encoder, an audio decoder or an audio processor includes a filter for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal, the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.

Abstract: For the reduction of the rounding error, a first and a second non-integer input value are provided and combined, for example by addition, in non-integer state to obtain a non-integer result value which is rounded and added to a third input value. Thus, the rounding error may be reduced at an interface between two rotations divided into lifting steps or between a first rotation divided into lifting steps and a first lifting step of a subsequent multi-dimensional lifting sequence.

Abstract: Quantizing an information signal of a sequence of information values includes frequency-selective filtering the sequence of information values to obtain a sequence of filtered information values and quantizing the filtered information values to obtain a sequence of quantized information values by means of a quantizing step function which maps the filtered information values to the quantized information values and the course of which is steeper below a threshold information value than above the threshold information value.

Abstract: The central idea of the present invention is that the prior procedure, namely interpolation relative to the filter coefficients and the amplification value, for obtaining interpolated values for the intermediate audio values starting from the nodes has to be dismissed. Coding containing less audible artifacts can be obtained by not interpolating the amplification value, but rather taking the power limit derived from the masking threshold, preferably as the area below the square of the magnitude of the masking threshold, for each node, i.e. for each parameterization to be transferred, and then performing the interpolation between these power limits of neighboring nodes, such as, for example, a linear interpolation.

Abstract: If an adaptive prediction algorithm controllable by a speed coefficient is started from to operate with a first adaption speed and a first adaption precision and an accompanying first prediction precision in the case that the speed coefficient has a first value and to operate with a second, compared to the first one, lower adaption speed and a second, but compared to the first one, higher precision in the case that the speed parameter has a second value, the adaption durations occurring after the reset times where the prediction errors are at first increased due to the, not yet, adapted prediction coefficients may be decreased by at first setting the speed parameter to the first value and, after a while, to a second value. After the speed parameter has again been set to the second value after a predetermined duration after the reset times, the prediction errors and thus the residuals to be transmitted are more optimized or smaller than would be possible with the first speed parameter value.

Abstract: Coding an audio signal of a sequence of audio values into a coded signal includes determining a first listening threshold for a first block of audio values of the sequence of audio values and a second listening threshold for a second block of audio values of the sequence of audio values; calculating a version of a first parameterization of a parameterizable filter such that the transfer function thereof roughly corresponds to the inverse of the magnitude of the first listening threshold and a version of a second parameterization of the parameterizable filter such that the transfer function thereof roughly corresponds to the inverse of the magnitude of the second listening threshold; filtering a predetermined block of audio values of the sequence of audio values with the parameterizable filter using a predetermined parameterization which in a predetermined manner depends on the version of the second parameterization to obtain a block of filtered audio values corresponding to the predetermined block; quantizing

Abstract: When processing a signal having a sequence of discrete values, wherein there is a first frequency range, in which the signal has a high energy, and wherein there is a second frequency range, in which the signal has a low energy, the sequence of discrete values is first manipulated to obtain a sequence of manipulated values, so that at least one of the manipulated values is non-integer. Then the sequence of manipulated values is rounded to obtain a sequence of manipulated values. The rounding is formed to effect a spectral shaping of a generated rounding error so that a spectrally shaped rounding error has a higher energy in the first frequency range than in the second frequency range. By spectrally shaping the rounding error so that the rounding error does not have any energy either in the storage areas where there is no signal energy, an especially efficient coding is obtained particularly in connection with a lossless coding context.

Abstract: A perceptual coder is disclosed for encoding image signals, such as speech or music, with different spectral and temporal resolutions for redundancy reduction and irrelevancy reduction. The image signal is initially spectrally shaped using a prefilter. The prefilter output samples are thereafter quantized and coded to minimize the mean square error (MSE) across the spectrum. The disclosed perceptual image coder can use fixed quantizer step-sizes, since spectral shaping is performed by the pre-filter prior to quantization and coding. The disclosed pre-filter and post-filter support the appropriate frequency dependent temporal and spectral resolution for irrelevancy reduction. A filter structure based on a frequency-warping technique is used that allows filter design based on a non-linear frequency scale.

Abstract: For converting first and second blocks of discrete values into a transformed representation, the first block is transformed according to a first transformation rule and then rounded. Then, the rounded transformed values are summed with the second block of original discrete values, to then process the summation result according to a second transformation rule. The output values of the transformation via the second transformation rule are again rounded and then subtracted from the original discrete values of the first block of discrete values to obtain a block of integer output values of the transformed representation. By this multi-dimensional lifting scheme, a lossless integer transformation is obtained, which can be reversed by applying the same transformation rule, but with different signs in summation and subtraction, respectively, so that an inverse integer transformation can also be obtained.

Abstract: For embedding binary payload in a carrier signal, which, for example, is an audio signal, a sequence of time-discrete values of the carrier signal is converted to the frequency domain by means of an integer transform algorithm to obtain binary spectral representation values. Bits of the binary spectral representation values with a valency less than signal limit valency are determined and set according to the payload. The signal limit valency for a spectral representation value is less than the valency of the leading bit of this spectral representation value, so that, with adequate distance, a psychoacoustic transparent insertion of information is achieved. Thus a modified spectral representation with inserted information is generated which is finally converted back to the time domain using an integer back transform algorithm. For extracting the payload, the time-discrete signal with the inserted information is again converted to a spectral representation with the integer forward transform algorithm.

Abstract: An apparatus for scalable encoding a spectrum of a signal including audio and/or video information, with the spectrum comprising binary spectral values, includes a means (102) for generating a first sub-scaling layer and a second sub-scaling layer in addition to a means (106) for forming the encoded signal, with the means (106) for forming being implemented so as to include the first sub-scaling layer and the second sub-scaling layer into the encoded signal that the first and the second sub-scaling layer are separately decodable from each other. In contrast to a full-scaling layer, a sub-scaling layer includes only the bits of a certain order of a part of the binary spectral values in the band, so that, by additionally decoding a sub-scaling layer, a more finely controllable and a more finely scalable precision gain may be achieved.