Abstract: An apparatus for processing an encoded signal, the encoded signal having an encoded audio signal having information on a pitch delay or a pitch gain, and a bass post-filter control parameter, has: an audio signal decoder for decoding the encoded audio signal using the information on the pitch delay or the pitch gain to obtain a decoded audio signal; a controllable bass post-filter for filtering the decoded audio signal to obtain a processed signal, wherein the controllable bass post-filter has the variable bass post-filter characteristic controllable by the bass post-filter control parameter; and a controller for setting the variable bass post-filter characteristic in accordance with the bass post-filter control parameter included in the encoded signal.

Abstract: An encoder for predictively encoding a signal having a sequence of signal values has a predictor for performing an adaptive prediction in dependence on the signal, and in dependence on one or more weighting values, to obtain predicted signal values, wherein the predictor is configured to reset the weighting values at times which are dependent on the signal, and wherein the predictor is configured to adapt the weighting values to the signal between subsequent resets.

Abstract: A basic idea of the present application is that in case of determining a position of the object on the surface, it is possible to also use transport mechanisms for the transport of the object on the surface which leads to less reproducible transport movements as the regulation may be executed directly on the basis of the observed movement as compared to the desired movement. Embodiments using compressed air, magnetism and/or bending waves are described.

Abstract: An audio codec supporting both, time-domain and frequency-domain coding modes, having low-delay and an increased coding efficiency in terms of iterate/distortion ratio, is obtained by configuring the audio encoder such that same operates in different operating modes such that if the active operative mode is a first operating mode, a mode dependent set of available frame coding modes is disjoined to a first subset of time-domain coding modes, and overlaps with a second subset of frequency-domain coding modes, whereas if the active operating mode is a second operating mode, the mode dependent set of available frame coding modes overlaps with both subsets, i.e. the subset of time-domain coding modes as well as the subset of frequency-domain coding modes.

Abstract: A basic idea of the present application is that in case of determining a position of the object on the surface, it is possible to also use transport mechanisms for the transport of the object on the surface which leads to less reproducible transport movements as the regulation may be executed directly on the basis of the observed movement as compared to the desired movement. Embodiments using compressed air, magnetism and/or bending waves are described.

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: An audio codec supporting both, time-domain and frequency-domain coding modes, having low-delay and an increased coding efficiency in terms of iterate/distortion ratio, is obtained by configuring the audio encoder such that same operates in different operating modes such that if the active operative mode is a first operating mode, a mode dependent set of available frame coding modes is disjoined to a first subset of time-domain coding modes, and overlaps with a second subset of frequency-domain coding modes, whereas if the active operating mode is a second operating mode, the mode dependent set of available frame coding modes overlaps with both subsets, i.e. the subset of time-domain coding modes as well as the subset of frequency-domain coding modes.

Abstract: An encoder for predictively encoding a signal having a sequence of signal values has a predictor for performing an adaptive prediction in dependence on the signal, and in dependence on one or more weighting values, to obtain predicted signal values, wherein the predictor is configured to reset the weighting values at times which are dependent on the signal, and wherein the predictor is configured to adapt the weighting values to the signal between subsequent resets.

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.

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: 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: The transient problem may be sufficiently addressed, and for this purpose, a further delay on the side of the decoding may be reduced if a new SBR frame class is used wherein the frame boundaries are not shifted, i.e. the grid boundaries are still synchronized with the frame boundaries, but wherein a transient position indication is additionally used as a syntax element so as to be used, on the encoder and/or decoder sides, within the frames of these new frame class for determining the grid boundaries within these frames.

Abstract: In the context of packet-oriented data transmission via a network, an apparatus for transmitting a sequence of data packets checks whether a packet is missing or faulty. In this case, a replacement packet is provided which is a valid packet with regard to a packet syntax which, however, has a predetermined contents characteristic. On the decoder side, a basic decoder recognizes the packet as a valid packet and decodes same, whereas an extension decoder can enable an error concealing measure on the basis of an indication in the replacement packet to the fact that this is a replacement packet, in order to provide a higher-quality audio reproduction.

Abstract: An apparatus according to an embodiment of the present invention for mixing a first frame of a first input data stream and a second frame of a second input data stream has a processing unit adapted to generate an output frame, wherein the output frame has output spectral data describing a lower part of an output spectrum up to an output cross-over frequency, and wherein the output frame further has output SBR-data describing a higher part of the output spectrum above the output cross-over frequency by way of energy-related values in an output time/frequency grid resolution.

Abstract: The transient problem may be sufficiently addressed, and for this purpose, a further delay on the side of the decoding may be reduced if a new SBR frame class is used wherein the frame boundaries are not shifted, i.e. the grid boundaries are still synchronized with the frame boundaries, but wherein a transient position indication is additionally used as a syntax element so as to be used, on the encoder and/or decoder sides, within the frames of these new frame class for determining the grid boundaries within these frames.

Abstract: An apparatus for mixing a plurality of input data streams is described, which has a processing unit adapted to compare the frames of the plurality of input data streams, and determine, based on the comparison, for a spectral component of an output frame of an output data stream, exactly one input data stream of the plurality of input data streams. The output data stream is generated by copying at least a part of an information of a corresponding spectral component of the frame of the determined data stream. Further or alternatively, the control values of the frames of the first and second input data streams are compared, and, if so, the control value is adopted.

Abstract: A basic idea of the present application is that in case of determining a position of the object on the surface, it is possible to also use transport mechanisms for the transport of the object on the surface which leads to less reproducible transport movements as the regulation may be executed directly on the basis of the observed movement as compared to the desired movement. Embodiments using compressed air, magnetism and/or bending waves are described.

Abstract: The transient problem may be sufficiently addressed, and for this purpose, a further delay on the side of the decoding may be reduced if a new SBR frame class is used wherein the frame boundaries are not shifted, i.e. the grid boundaries are still synchronized with the frame boundaries, but wherein a transient position indication is additionally used as a syntax element so as to be used, on the encoder and/or decoder sides, within the frames of these new frame class for determining the grid boundaries within these frames.

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: A conference terminal for a digital voice conference system comprises a first sound transducer means, which is implemented to generate a microphone signal from an acoustic signal, a second sound transducer means, which is implemented to generate an acoustic signal from a loudspeaker signal, a connecting means, which is implemented to enable a connection between conference terminal and a conference central unit, to receive a composite conference signal from the conference central unit, and a means for echo suppression, which is implemented to combine the microphone signal or a signal derived therefrom in a noise suppression means with a composite conference signal in a listening operation of the conference terminal, such that a loudspeaker signal results, in which the acoustic signal, on which the microphone signal is based, is reduced.