Abstract: An audio decoder configured to generate a high band extension of an audio signal from an envelope and an excitation. The audio decoder includes a control arrangement configured to jointly control envelope shape and excitation noisiness with a common control parameter (f).

Abstract: In a quantization method for quantizing a received excitation signal in a communication system performing the steps of re-shuffling S301 the elements of the received excitation signal to provide a re-shuffled excitation signal; coding S302 the re-shuffled excitation signal with a variable bit-rate algorithm to provide a coded excitation signal; and reassigning S303 codewords of the coded excitation signal if a number of used bits exceeds a predetermined fixed bit rate requirement to provide a quantized excitation signal.

Abstract: The present invention relates to a frequency domain based method of encoding and decoding an audio signal, wherein an adaptive spectral code book is updated with synthesized frequency domain representations of a time domain signal segment. A frequency analysis is performed of a received time domain signal segment in order to obtain a frequency domain representation, and the adaptive spectral code book is searched for a first approximation of the frequency domain representation. A fixed spectral code book is searched for an approximation of the residual frequency representation. A synthesized frequency domain representation may be generated from the two approximations.

Abstract: Audio decoder and method therein for supporting bandwidth extension (BWE) of a received signal. The method involves receiving a first signal representing the lower frequency spectrum of a segment of an original audio signal; receiving a second signal, being a BWE signal, representing a higher frequency spectrum of the segment of the original audio signal. The method further comprises determining a degree of voicing in the lower frequency spectrum of the audio signal, based on the received first signal; and selecting a spectral tilt adaptation filter, out of at least two spectral tilt adaptation filters having different spectral attenuation characteristics, based on the determined degree of voicing. The selected spectral tilt adaptation filter is then applied on the received second signal. Thus, a differentiation of spectral tilt in the higher frequency spectrum of a reconstructed audio signal, based on lower frequency spectrum characteristics of the original audio signal is enabled.

Abstract: An audio decoder configured to generate a high band extension of an audio signal from an envelope and an excitation. The audio decoder includes a control arrangement configured to jointly control envelope shape and excitation noisiness with a common control parameter (f).

Abstract: The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.

Abstract: The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.

Abstract: An object of embodiments of the present invention is to provide an improved gain-shape VQ. This is achieved by determining a number of bits to be allocated to a gain adjustment- and shape-quantizer for a plurality of combinations of a current bit rate and a first signal property. The determined allocated number of bits to the gain adjustment- and shape quantizer should provide a better result for the given bitrate and signal property than using a single fixed allocation scheme. That can be achieved by deriving the bit allocation by using an average of optimal bit allocations for a training data set. Thus by pre-calculating a number of bits to the gain adjustment and the shape quantizers for a plurality of combinations of the bit rate and a first signal property and creating a table indicating the number of bits to be allocated to the gain adjustment- and the shape-quantizers for a plurality of combinations of the bit rate and a first signal property.

Abstract: A gain adjustment apparatus (60) for use in decoding of audio that has been encoded with separate gain and shape representations includes an accuracy meter (62) configured to estimate an accuracy measure (A(b)) of the shape representation (Ñ(b)), and to determine a gain correction (gc(b)) based on the estimated accuracy measure (A(b)). It also includes an envelope adjuster (64) configured to adjust the gain representation (Ê(b)) based on the determined gain correction.

Abstract: A method for encoding of an audio signal comprises performing (214) of a transform of the audio signal. An energy offset is selected (216) for each of the first subbands. An energy measure of a first reference band within a low band of an encoding of a synthesis signal is obtained (212). The first high band is encoded (220) by providing quantization indices representing a respective scalar quantization of a spectrum envelope in the first subbands of the first high band relative to the energy measure of the first reference band by use of the selected energy offset. An encoder apparatus comprises means for carrying out the steps of the method. Corresponding decoder methods and apparatuses are also described.

Abstract: The present invention relates to a frequency domain based method of encoding and decoding an audio signal, wherein an adaptive spectral code book is updated with synthesized frequency domain representations of a time domain signal segment. A frequency analysis is performed of a received time domain signal segment in order to obtain a frequency domain representation, and the adaptive spectral code book is searched for a first approximation of the frequency domain representation. A fixed spectral code book is searched for an approximation of the residual frequency representation. A synthesized frequency domain representation may be generated from the two approximations.

Abstract: The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.

Abstract: In signal processing using digital filtering the representation of a filter is adapted depending on the filter characteristics If e.g. a digital filter is represented by filter coefficients for transform bands Nos. 0, 1, . . . , K in the frequency domain, a reduced digital filter having coefficients for combined transform bands, i.e. subsets of the transformed bands, Nos. 0, 1, . . . , L, is formed and only these coefficients are stored. When the actual filtering in the digital filter is to be performed, an actual digital filter is obtained by expanding the coefficients of the reduced digital filter according to a mapping table and then used instead of the original digital filter.

Abstract: An overall encoding procedure and associated decoding procedure are presented. The encoding procedure involves at least two signal encoding processes operating on signal representations of a set of audio input channels. Local synthesis is used in connection with a first encoding process to generate a locally decoded signal, including a representation of the encoding error of the first encoding process. This locally decoded signal is applied as input to a second encoding process. The overall encoding procedure generates at least two residual encoding error signals from at least one of said encoding processes, including at least said second encoding process. The residual error signals are then subjected to compound residual encoding in a further encoding process, preferably based on correlation between the residual error signals.

Abstract: The invention relates to the technical field of audio encoding and/or decoding technologies, and thus concerns an overall encoding procedure and associated decoding procedure. The encoding procedure involves at least two signal encoding processes (S1-S3) operating on signal representations of a set of audio input channels, as well as residual encoding (S7-S8). It also involves a dedicated process (S4-S6) to estimate and encode energies of the audio input channels. Each encoding process is associated with a corresponding decoding process. In the overall decoding procedure the decoded signals from each encoding process are preferably combined such that the output channels are close to the input channels in terms of energy and/or quality. Normally, the combination step also adapts to the possible loss of one or more signal representation in part or in whole, such that the energy and quality is optimized with the signals at hand in the decoder. In this way, the overall quality of the output channels is improved.

Abstract: An overall encoding procedure and associated decoding procedure are presented. The encoding procedure involves at least two signal encoding processes operating on signal representations of a set of audio input channels. Local synthesis is used in connection with a first encoding process to generate a locally decoded signal, including a representation of the encoding error of the first encoding process. This locally decoded signal is applied as input to a second encoding process. The overall encoding procedure generates at least two residual encoding error signals from at least one of said encoding processes, including at least said second encoding process. The residual error signals are then subjected to compound residual encoding in a further encoding process, preferably based on correlation between the residual error signals.

Abstract: In signal processing using digital filtering the representation of a filter is adapted depending on the filter characteristics. If e.g. a digital filter is represented by filter coefficients for transform bands Nos. 0, 1, . . . , K in the frequency domain, a reduced digital filter having coefficients for combined transform bands, i.e. subsets of the transformed bands, Nos. 0, 1, . . . , L, is formed and only these coefficients are stored. When the actual filtering in the digital filter is to be performed, an actual digital filter is obtained by expanding the coefficients of the reduced digital filter according to a mapping table and then used instead of the original digital filter.