Abstract: Decoder for decoding a picture from a data stream, configured to check whether a plurality of coding parameters, which are contained in the data stream, relate to a predetermined portion of the picture and control a prediction residual transform mode and a quantization accuracy with respect to the predetermined portion, are indicative of a coding parameter setting corresponding to a lossless prediction residual coding. Responsive to the plurality of coding parameters being indicative of the coding parameter setting corresponding to the lossless prediction residual coding, the decoder is configured to set one or more predetermined coding options relating to one or more tools of the decoder for processing a prediction residual corrected predictive reconstruction with respect to the predetermined portion so that the one or more tools are disabled with respect to the predetermined portion.

Abstract: An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.

Abstract: An apparatus processing an encoded signal to acquire first and second frames comprising spectral values and an aliasing portion; applying a transform to the first frame using a first window function to acquire a first block of samples, applying another transform to a first portion of the second frame using a second window function, and applying another one or more transforms to a second portion of the second frame using one or more third window functions to acquire a second block of samples; and post-processing the second block of samples using a folding-out operation to acquire a post-processed second block of samples comprising a portion of the second block of samples overlapping with the first block of samples in a multi-overlap region, windowing the post-processed second block of samples using an auxiliary window function, and overlap-adding a windowed post-processed second block of samples and the first block of samples.

Abstract: In multichannel audio coding, an improved coding efficiency is achieved by the following measure: the noise filling of zero-quantized scale factor bands is performed using noise filling sources other than artificially generated noise or spectral replica. In particular, the coding efficiency in multichannel audio coding may be rendered more efficient by performing the noise filling based on noise generated using spectral lines from a previous frame of, or a different channel of the current frame of, the multichannel audio signal.

Abstract: A schematic block diagram of a decoder for decoding an encoded audio signal is shown. The decoder includes an adaptive spectrum-time converter and an overlap-add-processor. The adaptive spectrum-time converter converts successive blocks of spectral values into successive blocks of time values, e.g. via a frequency-to-time transform. Furthermore, the adaptive spectrum-time converter receives a control information and switches, in response to the control information, between transform kernels of a first group of transform kernels including one or more transform kernels having different symmetries at sides of a kernel, and a second group of transform kernels including one or more transform kernels having the same symmetries at sides of a transform kernel. Moreover, the overlap-add-processor overlaps and adds the successive blocks of time values to obtain decoded audio values, which may be a decoded audio signal.

Abstract: Decoder for decoding a picture from a data stream, configured to check whether a plurality of coding parameters, which are contained in the data stream, relate to a predetermined portion of the picture and control a prediction residual transform mode and a quantization accuracy with respect to the predetermined portion, are indicative of a coding parameter setting corresponding to a lossless prediction residual coding. Responsive to the plurality of coding parameters being indicative of the coding parameter setting corresponding to the lossless prediction residual coding, the decoder is configured to set one or more predetermined coding options relating to one or more tools of the decoder for processing a prediction residual corrected predictive reconstruction with respect to the predetermined portion so that the one or more tools are disabled with respect to the predetermined portion.

Abstract: An encoder for encoding an audio signal is configured to encode the audio signal in a transform domain or filter-bank domain, is configured to determine spectral coefficients of the audio signal for a current frame and at least one previous frame, and is configured to selectively apply predictive encoding to a plurality of individual spectral coefficients or groups of spectral coefficients which are separated by at least one spectral coefficient.

Abstract: An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.

Abstract: Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using a filtering. Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using different noise filling methods. Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using a tilt.

Abstract: Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using a filtering. Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using different noise filling methods. Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using a tilt.

Abstract: Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using a filtering. Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using different noise filling methods. Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using a tilt.

Abstract: An apparatus for generating a decoded two-channel signal, comprising: a parametric decoder for providing parametric data for a second set of second spectral portions and a two-channel identification identifying for a second spectral portion of the second set of second spectral portions either a first two-channel representation for the second spectral portion of the second set of second spectral portions or a second two-channel representation for the second spectral portion of the second set of second spectral portions, the second two-channel representation being different from the first two-channel representation; and a frequency regenerator for regenerating the second spectral portion of the second set of second spectral portions depending on a first spectral portion of a first set of first spectral portions, the parametric data for the second spectral portion of the second set of second spectral portions and the two-channel identification for the second spectral portion of the second set of second spectral

Abstract: An apparatus for determining visual activity information for a predetermined picture block of a video sequence including a plurality of video frames, the plurality of video frames including a current video frame and one or more timely-preceding video frames, wherein the one or more timely-preceding video frames precede the current video frame in time, is provided. The apparatus is configured to receive the predetermined picture block of each of the one or more timely-preceding video frames and the predetermined picture block of the current video frame. Moreover, the apparatus is configured to determine the visual activity information depending on the predetermined picture block of the current video frame and depending on the predetermined picture block of each of the one or more timely-preceding video frames and depending on a temporal high-pass filter.

Abstract: An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided. The apparatus includes a receiving interface, a delay buffer and a sample processor for processing the selected audio signal samples to obtain reconstructed audio signal samples of the reconstructed audio signal. The sample selector is configured to select, if a current frame is received by the receiving interface and if the current frame being received by the receiving interface is not corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being included by the current frame.

Abstract: An encoder for encoding an audio signal is configured to encode the audio signal in a transform domain or filter-bank domain, is configured to determine spectral coefficients of the audio signal for a current frame and at least one previous frame, and is configured to selectively apply predictive encoding to a plurality of individual spectral coefficients or groups of spectral coefficients which are separated by at least one spectral coefficient.

Abstract: An apparatus for encoding a speech signal that determines a codebook vector of a speech coding algorithm includes a matrix determiner for determining an autocorrelation matrix R, and a codebook vector determiner for determining the codebook vector depending on the autocorrelation matrix R. The matrix determiner determines the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R includes a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein R(i , j)=r(|i?j|), wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R.

Abstract: An audio decoder provides a decoded audio information on the basis of an encoded audio information including linear prediction coefficients (LPC) and includes a tilt adjuster to adjust a tilt of a noise using linear prediction coefficients of a current frame to acquire a tilt information and a noise inserter configured to add the noise to the current frame in dependence on the tilt information. Another audio decoder includes a noise level estimator to estimate a noise level for a current frame using a linear prediction coefficient of at least one previous frame to acquire a noise level information; and a noise inserter to add a noise to the current frame in dependence on the noise level information provided by the noise level estimator. Thus, side information about a background noise in the bit-stream may be omitted. Methods and computer programs serve a similar purpose.

Abstract: A device and method for manipulating an audio signal includes a windower for generating a plurality of consecutive blocks of audio samples, the plurality of consecutive blocks including at least one padded block of audio samples, the padded block having padded values and audio signal values, a first converter for converting the padded block into a spectral representation having spectral values, a phase modifier for modifying phases of the spectral values to obtain a modified spectral representation and a second converter for converting the modified spectral representation into a modified time domain audio signal.

Abstract: A device and method for manipulating an audio signal includes a windower for generating a plurality of consecutive blocks of audio samples, the plurality of consecutive blocks including at least one padded block of audio samples, the padded block having padded values and audio signal values, a first converter for converting the padded block into a spectral representation having spectral values, a phase modifier for modifying phases of the spectral values to obtain a modified spectral representation and a second converter for converting the modified spectral representation into a modified time domain audio signal.

Abstract: A device and method for manipulating an audio signal includes a windower for generating a plurality of consecutive blocks of audio samples, the plurality of consecutive blocks including at least one padded block of audio samples, the padded block having padded values and audio signal values, a first converter for converting the padded block into a spectral representation having spectral values, a phase modifier for modifying phases of the spectral values to obtain a modified spectral representation and a second converter for converting the modified spectral representation into a modified time domain audio signal.