Abstract: A noise filler for providing a noise-filled spectral representation of an audio signal on the basis of an input spectral representation of the audio signal has a spectral region identifier configured to identify spectral regions of the input spectral representation spaced from non-zero spectral regions of the input spectral representation by at least one intermediate spectral region, to obtain identified spectral regions, and a noise inserter configured to selectively introduce noise into the identified spectral regions to obtain the noise-filled spectral representation of the audio signal. A noise filling parameter calculator for providing a noise filling parameter on the basis of a quantized spectral representation of an audio signal has a spectral region identifier, as mentioned above, and a noise value calculator configured to selectively consider quantization errors of the identified spectral regions for a calculation of the noise filling parameter.

Abstract: An audio encoder adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame includes a number of time domain audio samples. The audio encoder includes a predictive coding analysis stage for determining information on coefficients of a synthesis filter and a prediction domain frame based on a frame of audio samples. The audio encoder further includes a time-aliasing introducing transformer for transforming overlapping prediction domain frames to the frequency domain to obtain prediction domain frame spectra, wherein the time-aliasing introducing transformer is adapted for transforming the overlapping prediction domain frames in a critically-sampled way. Moreover, the audio encoder includes a redundancy reducing encoder for encoding the prediction domain frame spectra to obtain the encoded frames based on the coefficients and the encoded prediction domain frame spectra.

Abstract: An apparatus for encoding an audio signal includes the windower for windowing a first block of the audio signal using an analysis window having an aliasing portion and a further portion. The apparatus furthermore includes a processor for processing the first sub-block of the audio signal associated with the aliasing portion by transforming the sub-block from a domain into a different domain subsequent to windowing the first sub-block to obtain the processed first sub-block, and for processing a second sub-block of the audio signal associated with the further portion by transforming the second sub-block from the domain into the different domain before windowing the second sub-block to obtain a processed second sub-block.

Abstract: An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal includes a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also includes an audio stream provider for providing the audio stream such that the audio stream includes information describing an audio content of the frequency bands and information describing the multi-band quantization error. A decoder for providing a decoded representation of an audio signal on the basis of an encoded audio stream representing spectral components of frequency bands of the audio signal includes a noise filler for introducing noise into spectral components of a plurality of frequency bands to which separate frequency band gain information is associated on the basis of a common multi-band noise intensity value.

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: 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: 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: An audio encoder, an audio decoder or an audio processor includes a filter (12) 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 (16), the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller (18) 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 (22) 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: 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: 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, 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. On both the coder and the decoder side, an amplification value can then be calculated from the intermediate power limit determined such that the quantizing noise caused by quantization, which has a constant frequency before post-filtering on the decoder side, is below the power limit or corresponds thereto after post-filtering.

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: For generating a signal to be transmitted original information is encoded into a main channel and a side channel, wherein the side channel is more robust against channel influences than the main channel. On the receiver side, when the receive quality is above a threshold, which is necessitated to execute a successful decoding of the main channel, the main channel is reproduced. If the receive quality falls below this threshold, however, the side channel is reproduced which may have less bits than the main channel and which is a correspondingly lower quality representation of the original information than the main channel.

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: 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: 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: 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.