Abstract: A decoder for block-based decoding of picture data of a picture is configured to decode a current block of the picture by determining first and second sets of luma in a neighborhood of the current block, the first set of luma samples having smaller luma values than the second set of luma samples, fitting a straight-line having a linear function through first and second pairs of luma-chroma samples, wherein the first and second pairs of luma-chroma samples are determined based on the first and second sets of luma samples, respectively, and on first and second corresponding chroma samples, respectively, in said neighborhood of the current block, and predicting chroma samples of the current block using luma samples of the current block and said linear function. The present invention further concerns a respective encoder and corresponding methods for encoding and decoding.

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: A picture in a data stream is subdivided into different size transform bocks. For a transform block, a transformation is selected by checking whether the block's size exceeds a threshold. If it does, a default technique is used to select the transformation. If it does not, a transformation is used from a list of transformations as identified by an index, the list including an identity transformation and non-identity transformations. A coefficients block in the data stream is decoded, and if the selected transformation is the identity transformation, then the coefficients block is used as a residual sample array, and if the selected transformation is not the identity transformation, the coefficients block is subjected to an inverse transformation corresponding to the selected transformation to obtain a residual sample array used to correct a prediction for the transform block.

Abstract: An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm is provided. The apparatus 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 is configured to determine 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 apparatus decodes an encoded audio signal. The apparatus includes a spectral domain audio decoder that generates a decoded representation of a set of spectral portions, the decoded representation being spectral prediction residual values. A frequency regenerator generates a reconstructed spectral portion using a portion of the same set spectral portions. The reconstructed spectral portion also includes spectral prediction residual values. An inverse prediction filter is configured using prediction filter information included in the encoded audio signal and performs an inverse prediction over frequency using the spectral prediction residual values.

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: A decoder for block-based decoding of picture data of a picture is configured to decode a current block of the picture by determining first and second sets of luma in a neighborhood of the current block, the first set of luma samples having smaller luma values than the second set of luma samples, fitting a straight-line having a linear function through first and second pairs of luma-chroma samples, wherein the first and second pairs of luma-chroma samples are determined based on the first and second sets of luma samples, respectively, and on first and second corresponding chroma samples, respectively, in said neighborhood of the current block, and predicting chroma samples of the current block using luma samples of the current block and said linear function. The present proposal further concerns a respective encoder and corresponding methods for encoding and decoding.

Abstract: A frequency-domain audio codec is provided with the ability to additionally support a certain transform length in a backward-compatible manner, by the following: the frequency-domain coefficients of a respective frame are transmitted in an interleaved manner irrespective of the signalization signaling for the frames as to which transform length actually applies, and additionally the frequency-domain coefficient extraction and the scale factor extraction operate independent from the signalization. By this measure, old-fashioned frequency-domain audio coders/decoders, insensitive for the signalization, would be able to nevertheless operate without faults and with reproducing a reasonable quality. Concurrently, frequency-domain audio coders/decoders able to support the additional transform length would offer even better quality despite the backward compatibility.

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: The invention provides an audio encoder including a combination of a linear predictive coding filter having a plurality of linear predictive coding coefficients and a time-frequency converter, wherein the combination is configured to filter and to convert a frame of the audio signal into a frequency domain in order to output a spectrum based on the frame and on the linear predictive coding coefficients; a low frequency emphasizer configured to calculate a processed spectrum based on the spectrum, wherein spectral lines of the processed spectrum representing a lower frequency than a reference spectral line are emphasized; and a control device configured to control the calculation of the processed spectrum by the low frequency emphasizer depending on the linear predictive coding coefficients of the linear predictive coding filter.

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: An apparatus for decoding an encoded audio signal, includes a spectral domain audio decoder for generating a first decoded representation of a first set of first spectral portions, the decoded representation having a first spectral resolution; a parametric decoder for generating a second decoded representation of a second set of second spectral portions having a second spectral resolution being lower than the first spectral resolution; a frequency regenerator for regenerating every constructed second spectral portion having the first spectral resolution using a first spectral portion and spectral envelope information for the second spectral portion; and a spectrum time converter for converting the first decoded representation and the reconstructed second spectral portion into a time representation.

Abstract: An apparatus for encoding a first channel and a second channel of an audio input signal including two or more channels to obtain an encoded audio signal according to an embodiment includes a normalizer configured to determine a normalization value for the audio input signal depending on the first channel of the audio input signal and depending on the second channel of the audio input signal. Moreover, the apparatus includes an encoding unit configured to generate a processed audio signal having a first channel and a second channel. The encoding unit is configured to encode the processed audio signal to obtain the encoded audio signal.

Abstract: An apparatus for selecting one of a first encoding algorithm having a first characteristic and a second encoding algorithm having a second characteristic for encoding a portion of an audio signal to obtain an encoded version of the portion of the audio signal has a first estimator for estimating a first quality measure for the portion of the audio signal, which is associated with the first encoding algorithm, without actually encoding and decoding the portion of the audio signal using the first encoding algorithm. A second estimator is provided for estimating a second quality measure for the portion of the audio signal, which is associated with the second encoding algorithm, without actually encoding and decoding the portion of the audio signal using the second encoding algorithm. The apparatus has a controller for selecting the first or second encoding algorithms based on a comparison between the first and second quality measures.

Abstract: Apparatus for encoding a video having a sequence of frames using rate control, configured to determine a global quantization parameter for the sequence of frames based on a target bit-rate, and to perform a coding pass, coding the sequence of frames, using the global quantization parameter by determining a frame quantization parameter per frame of the sequence of frames on the basis of the global quantization parameter, and subjecting the sequence of frames to R/D optimizing encoding by using, for each frame, the frame quantization parameter determined for the respective frame so as to obtain an encoded version of an associated coding size for the respective frame.

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: An audio decoder for providing at least four audio channel signals on the basis of an encoded representation is configured to provide a first residual signal and a second residual signal on the basis of a jointly encoded representation of the first residual signal and of the second residual signal using a multi-channel decoding. The audio decoder is configured to provide a first audio channel signal and a second audio channel signal on the basis of a first downmix signal and the first residual signal using a residual-signal-assisted multi-channel decoding. The audio decoder is configured to provide a third audio channel signal and a fourth audio channel signal on the basis of a second downmix signal and the second residual signal using a residual-signal-assisted multi-channel decoding. An audio encoder is based on corresponding considerations.

Abstract: An apparatus for decoding an audio signal includes a receiving interface, wherein the receiving interface is configured to receive a first frame and a second frame. Moreover, the apparatus includes a noise level tracing unit for determining noise level information being represented in a tracing domain. Furthermore, the apparatus includes a first reconstruction unit for reconstructing a third audio signal portion of the audio signal depending on the noise level information and a second reconstruction unit for reconstructing a fourth audio signal portion depending on noise level information being represented in the second reconstruction domain.

Abstract: A frequency-domain audio codec is provided with the ability to additionally support a certain transform length in a backward-compatible manner, by the following: the frequency-domain coefficients of a respective frame are transmitted in an interleaved manner irrespective of the signalization signaling for the frames as to which transform length actually applies, and additionally the frequency-domain coefficient extraction and the scale factor extraction operate independent from the signalization. By this measure, old-fashioned frequency-domain audio coders/decoders, insensitive for the signalization, would be able to nevertheless operate without faults and with reproducing a reasonable quality. Concurrently, frequency-domain audio coders/decoders able to support the additional transform length would offer even better quality despite the backward compatibility.

Abstract: The invention provides an audio encoder including a combination of a linear predictive coding filter having a plurality of linear predictive coding coefficients and a time-frequency converter, wherein the combination is configured to filter and to convert a frame of the audio signal into a frequency domain in order to output a spectrum based on the frame and on the linear predictive coding coefficients; a low frequency emphasizer configured to calculate a processed spectrum based on the spectrum, wherein spectral lines of the processed spectrum representing a lower frequency than a reference spectral line are emphasized; and a control device configured to control the calculation of the processed spectrum by the low frequency emphasizer depending on the linear predictive coding coefficients of the linear predictive coding filter.