Abstract: The manipulation of audio data can be simplified, such as, for example, with regard to the combination of individual audio data streams to multi-channel audio data streams or the general manipulation of an audio data stream, by modifying a data block in an audio data stream divided into data blocks with determination block and data block audio data, such as by completing or adding or replacing part of the same, so that the same includes a length indicator indicating an amount or length of data, respectively, of the data block audio data or an amount or length of data, respectively, of the data block to obtain a second audio data stream with modified data blocks. Alternatively, an audio data stream with pointers in determination blocks, which point to determination block audio data associated to this determination blocks, but distributed among different data blocks, is converted into an audio data stream, wherein the determination block audio data are combined to contiguous determination block audio data.

Abstract: Coding an audio signal of a sequence of audio values into a coded signal includes determining first and second listening thresholds for first and second blocks of audio values of the sequence of audio values; calculating versions of first second parameterizations of the parameterizable filter such that the transfer function thereof roughly corresponds to the inverse of the magnitude of the first and second listening thresholds, respectively; filtering a predetermined block of audio values of the sequence of audio values with the parameterizable filter using a predetermined parameterization which depends on the version of the second parameterization to obtain a block of filtered audio values corresponding to the predetermined block which is quantized; forming a difference between the versions of the first and second parameterizations; integrating information on, inter alias, the difference into the coded signal.

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: 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 very coarse quantization exceeding the measure determined by the masking threshold without or only very little quality losses is enabled by quantizing not immediately the prefiltered signal, but a prediction error obtained by forward-adaptive prediction of the prefiltered signal. Due to the forward adaptivity, the quantizing error has no negative effect on the prediction on the decoder side.

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: 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: In a method for generating a scalable data stream, when a block of output data of a first encoder is present, this block of output data is written into the scalable data stream. If output data of a second encoder is present for a preceding period of time, this output data for the preceding section is written in transmission direction behind the block of output data of the first encoder into the data stream. When the output data of the scalable encoder for the current section is present, the output data of the second encoder is written into the bit stream subsequent to the output data of the first encoder. A determining data block is generated and written into the bit stream delayed by a period of time which corresponds to the size of the bit savings bank of the second encoder.

Abstract: In a method for generating a scalable data stream from one or several blocks of output data of a first encoder and from one or several blocks of output data of a second encoder a determining data block for a current section of an input signal is written. In addition, output data of the second encoder representing a preceding section of the input signal are written in transmission direction from an encoder to a decoder after the determining data block. When the output data of the second encoder are written for a preceding section of the input signal, the output data of the second encoder are written representing the current section of the input signal. In order to signalize where the output data of the second encoder for the preceding section end and where the output data of the second encoder for the current section begin, buffer information is written into the scalable data stream.

Abstract: In a method of producing a scalable data stream of at least two blocks of output data of a first coder and a block of output data of a second coder, wherein the at least two blocks of output data of the first coder together represent a current section of an input signal in the first coder, and wherein the block of output data of the second coder represents the same current section of the input signal, a determination data block for the current section of the input signal is written. In addition, the block of output data of the second coder, in the direction of transfer from a coding device to a decoding device, is written after the determination data block for the current section of the input signal.

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: 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: 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: 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.

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: 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: The manipulation of audio data can be simplified, such as, for example, with regard to the combination of individual audio data streams to multi-channel audio data streams or the general manipulation of an audio data stream, by modifying a data block in an audio data stream divided into data blocks with determination block and data block audio data, such as by completing or adding or replacing part of the same, so that the same includes a length indicator indicating an amount or length of data, respectively, of the data block audio data or an amount or length of data, respectively, of the data block to obtain a second audio data stream with modified data blocks. Alternatively, an audio data stream with pointers in determination blocks, which point to determination block audio data associated to this determination blocks, but distributed among different data blocks, is converted into an audio data stream, wherein the determination block audio data are combined to contiguous determination block audio data.

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.