Abstract: An audio signal decoder for providing a decoded multi-channel audio signal representation on the basis of an encoded multi-channel audio signal representation has a time warp decoder configured to selectively use individual audio channel specific time warp contours or a joint multi-channel time warp contour for a reconstruction of a plurality of audio channels represented by the encoded multi-channel audio signal representation.

Abstract: An audio signal decoder configured to provide a decoded audio signal representation on the basis of an encoded audio signal representation having a time warp contour evolution information has a time warp contour calculator, a time warp contour data rescaler and a warp decoder. The time warp contour calculator is configured to generate time warp contour data repeatedly restarting from a predetermined time warp contour start value on the basis of a time warp contour evolution information describing a temporal evolution of the time warp contour. The time warp contour data rescaler is configured to rescale at least a portion of the time warp contour data such that a discontinuity at a restart is avoided, reduced or eliminated in a rescaled version of the time warp contour. The warp decoder is configured to provide the decoded audio signal representation on the basis of the encoded audio signal representation and using the rescaled version of the time warp contour.

Abstract: A processed representation of an audio signal having a sequence of frames is generated by sampling the audio signal within first and second frames of the sequence of frames, the second frame following the first frame, the sampling using information on a pitch contour of the first and second frames to derive a first sampled representation. The audio signal is sampled within the second and third frames, the third frame following the second frame in the sequence of frames. The sampling uses the information on the pitch contour of the second frame and information on a pitch contour of the third frame to derive a second sampled representation. A first scaling window is derived for the first sampled representation, and a second scaling window is derived for the second sampled representation, the scaling windows depending on the samplings applied to derive the first sampled representations or the second sampled representation.

Abstract: A filter bank device for generating a complex spectral representation of a discrete-time signal includes a generator for generating a block-wise real spectral representation, which, for example, implements an MDCT, to obtain temporally successive blocks of real spectral coefficients. The output values of this spectral conversion device are fed to a post-processor for post-processing the block-wise real spectral representation to obtain an approximated complex spectral representation having successive blocks, each block having a set of complex approximated spectral coefficients, wherein a complex approximated spectral coefficient can be represented by a first partial spectral coefficient and by a second partial spectral coefficient, wherein at least one of the first and second partial spectral coefficients is determined by combining at least two real spectral coefficients.

Abstract: A method of concealing a packet loss during video decoding is provided. An input stream having a plurality of network abstraction layer units NAL is received. A loss of a network abstraction layer unit in a group of pictures in the input stream is detected. A valid network abstraction layer unit order from the available network abstraction layer units is outputted. The network abstraction layer unit order is received by a video coding layer (VCL) and data is outputted.

Abstract: A filter bank device for generating a complex spectral representation of a discrete-time signal includes a generator for generating a block-wise real spectral representation, which, for example, implements an MDCT, to obtain temporally successive blocks of real spectral coefficients. The output values of this spectral conversion device are fed to a post-processor for post-processing the block-wise real spectral representation to obtain an approximated complex spectral representation having successive blocks, each block having a set of complex approximated spectral coefficients, wherein a complex approximated spectral coefficient can be represented by a first partial spectral coefficient and by a second partial spectral coefficient, wherein at least one of the first and second partial spectral coefficients is determined by combining at least two real spectral coefficients.

Abstract: An embodiment of an apparatus for generating audio subband values in audio subband channels includes an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function including a sequence of window coefficients to obtain windowed samples. The analysis window function includes a first number of window coefficients derived from a larger window function including a sequence of a larger second number of window coefficients, wherein the window coefficients of the window function are derived by an interpolation of window coefficients of the larger window function. The apparatus further includes a calculator for calculating the audio subband values using the windowed samples.

Abstract: In order to process a subband signal of a plurality of real subband signals which are a representation of a real discrete-time signal generated by an analysis filter bank, a weighter for weighting a subband signal by a weighting factor determined for the subband signal is provided to obtain a weighted subband signal. In addition, a correction term is calculated by a correction term determiner, the correction term determiner being implemented to calculate the correction term using at least one other subband signal and using another weighting factor provided for the other subband signal, the two weighting factors differing. The correction term is then combined with the weighted subband signal to obtain a corrected subband signal, resulting in reduced aliasing, even if subband signals are weighted to a different extent.

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: An embodiment of an apparatus for generating audio subband values in audio subband channels has an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function having a sequence of window coefficients to obtain windowed samples. The analysis window function has a first group of window coefficients and a second group of window coefficients. The first group of window coefficients is used for windowing later time-domain samples and the second group of window coefficients is used for windowing an earlier time-domain samples. The apparatus further has a calculator for calculating the audio subband values using the windowed samples.

Abstract: A cochlear implant for processing signal parameters which are adapted for controlling the cochlear implant, which are based on the audio signal and which enable generating a representation of the audio signal by the cochlear implant includes a receive interface which is implemented to receive the signal parameters and a nerve stimulator for processing the signal parameters to generate nerve cell stimulation signals based on the signal parameters. A device for generating a control signal for a cochlear implant on the basis of an audio signal includes a cochlear parameter extractor for analyzing the audio signal which is implemented to generate signal parameters as input information for the cochlear implant based on an analysis of the audio signal using a human hearing simulation model, and a transmit interface for transmitting the signal parameters to the cochlear implant.

Abstract: Standard video compression techniques apply motion-compensated prediction combined with transform coding of the prediction error. In the context of prediction with fractional-pel motion vector resolution it was shown, that aliasing components contained in an image signal are limiting the prediction efficiency obtained by motion compensation. In order to consider aliasing, quantization and motion estimation errors, camera noise, etc., we analytically developed a two dimensional (2D) non-separable interpolation filter, which is independently calculated for each frame by minimizing the prediction error energy. For every fractional-pel position to be interpolated, an individual set of 2D filter coefficients is determined. Since transmitting filter coefficients as side information results in an additional bit rate, which is almost constant for different image resolutions and total bit rates, the loss in coding gain increases when total bit rates sink.

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: A filter bank device for generating a complex spectral representation of a discrete-time signal includes a generator for generating a block-wise real spectral representation, which, for example, implements an MDCT, to obtain temporally successive blocks of real spectral coefficients. The output values of this spectral conversion device are fed to a post-processor for post-processing the block-wise real spectral representation to obtain an approximated complex spectral representation having successive blocks, each block having a set of complex approximated spectral coefficients, wherein a complex approximated spectral coefficient can be represented by a first partial spectral coefficient and by a second partial spectral coefficient, wherein at least one of the first and second partial spectral coefficients is determined by combining at least two real spectral coefficients.

Abstract: In a method of coding discrete time signals (X1) sampled with a first sampling rate, second time signals (x2) are generated using the first time signals having a bandwidth corresponding to a second sampling rate, with the second sampling rate being lower than the first sampling rate. The second time signals are coded in accordance with a first coding algorithm. The coded second signals (X2c) are decoded again in order to obtain coded/decoded second time signals (X2cd) having a bandwidth corresponding to the second sampling frequency. The first time signals, by frequency domain transformation, become first spectral values (X1). Second spectral values (X2cd) are generated from the coded/decoded second time signals, the second spectral values being a representation of the coded/decoded time signals in the frequency domain.

Abstract: A redundance reduction method used in coding multichannel signals is proposed, where signals available in digitized form are predicted. A prediction error is computed, which is subsequently quantized and loaded for transmission over a transmission path. In the method, prediction is performed linearly in a backward-adaptive fashion for at least two channels simultaneously, and the statistical relationships of the signals within a channel and between at least two channels is taken into account. A device for decoding redundance-reduced multichannel signals is proposed. It comprises a linear backward-adaptive predictor for at least two channels.

Abstract: A method for reducing frequency crosstalk in the transmission of digitized audio signals. Signals from partial bands in which spectral components of specific frequencies occur as signal components and signals from partial bands in which spectral components occur as crosstalk components in the stop range, undergo a weighted summation. Following transmission, the partial band signals undergo an inverse operation to weighted summation. The method operates independently of the selected encoding process, and is consequently universally usable.

Abstract: In a method for reducing data in a multi-channel data transmission, existing cross-relations between the signal channels are determined with due consideration of the transit-time difference. Upon the data transmission, a signal channel, the predictor coefficients and transit-time differences obtained from the cross-relations ascertained, and the prediction error are transmitted. At the receiver, the signal of the second channel is reconstructed from the transmitted signal channel, the predictor coefficients, the transit-time difference, and the prediction error. In this way, the amount of data to be transmitted in the case of multi-channel video or audio signals is reduced.

Abstract: Transmitting a signal wherein the signals is segmented by means of windows into successive overlapping blocks, the partial signals contained in the blocks are converted by transformation into a spectrum, with the spectra then being coded, transmitted, decoded after transmission and converted back into partial signals by retransformation. Finally, the blocks containing the partial signals are joined in an overlapping manner, with the overlapping regions of the blocks being weighted such that the resultant of the window functions in the respective overlapped regions equals one. To2.2. In order to avoid interferences in adjacent blocks upon changes in the signal amplitude, the length of the window functions is selected as a function of signal amplitude changes. The method is suitable for the treatment of audio and video signals which are subjected to data reduction during transmission.