Abstract: In accordance with an example embodiment of the present invention, disclosed is a method and an apparatus thereof for assisting a selection of an encoding mode for a multi-channel audio signal encoding where different encoding modes may be chosen for the different channels. The method is performed in an audio encoder and comprises obtaining a plurality of audio signal channels and coordinating or synchronizing the selection of an encoding mode for a plurality of the obtained channels, wherein the coordination is based on an encoding mode selected for one of the obtained channels or for a group of the obtained channels.

Abstract: Detecting voiced speech in an audio signal. A method comprises calculating an autocorrelation function (ACF) of a portion of an input audio signal and detecting a highest peak of said autocorrelation function within a determined range. A peak width and a peak height of said detected highest peak are determined and based on the peak width and the peak height it is decided whether a segment of an input audio signal comprises voiced speech.

Abstract: An encoder for encoding frequency transform coefficients of a harmonic audio signal include the following elements: A peak locator configured to locate spectral peaks having magnitudes exceeding a predetermined frequency dependent threshold. A peak region encoder configured to encode peak regions including and surrounding the located peaks. A low-frequency set encoder configured to encode at least one low-frequency set of coefficients outside the peak regions and below a crossover frequency that depends on the number of bits used to encode the peak regions. A noise-floor gain encoder configured to encode a noise-floor gain of at least one high-frequency set of not yet encoded coefficients outside the peak regions.

Abstract: An encoder (20) for encoding frequency transform coefficients (Y(k)) of a harmonic audio signal include the following elements: A peak locator (22) configured to locate spectral peaks having magnitudes exceeding a predetermined frequency dependent threshold. A peak region encoder (24) configured to encode peak regions including and surrounding the located peaks. A low-frequency set encoder (26) configured to encode at least one low-frequency set of coefficients outside the peak regions and below a crossover frequency that depends on the number of bits used to encode the peak regions. A noise-floor gain encoder (28) configured to encode a noise-floor gain of at least one high-frequency set of not yet encoded coefficients outside the peak regions.

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: An encoder (20) for encoding frequency transform coefficients (Y(k)) of a harmonic audio signal include the following elements: A peak locator (22) configured to locate spectral peaks having magnitudes exceeding a predetermined frequency dependent threshold. A peak region encoder (24) configured to encode peak regions including and surrounding the located peaks. A low-frequency set encoder (26) configured to encode at least one low-frequency set of coefficients outside the peak regions and below a crossover frequency that depends on the number of bits used to encode the peak regions. A noise-floor gain encoder (28) configured to encode a noise-floor gain of at least one high-frequency set of not yet encoded coefficients outside the peak regions.

Abstract: Estimation of a high band extension of a low band audio signal includes the following steps: extracting (S1) a set of features of the low band audio signal; mapping (S2) extracted features to at least one high band parameter with generalized additive modeling; frequency shifting (S3) a copy of the low band audio signal into the high band; controlling (S4) the envelope of the frequency shifted copy of the low band audio signal by said at least one high band parameter.

Abstract: The present invention relates to an improved scheme for coding of audio. In particular, the present invention relates to an encoder device and a method for coding an input signal in an encoder system. The method comprises applying a first mode to the input signal to form a first output and applying a second mode to the input signal to form a second output. A first processed output is then formed from at least a part of the first output, and a second processed output is formed from at least a part of the second output. Forming a second processed output comprises estimating a part of the input signal from at least a part of the second output. Then, an optimum mode is determined based on the first processed output and the second processed output, and the output according to the optimum mode is selected.

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: Estimation of a high band extension of a low band audio signal includes the following steps: extracting (S1) a set of features of the low band audio signal; mapping (S2) extracted features to at least one high band parameter with generalized additive modeling; frequency shifting (S3) a copy of the low band audio signal into the high band; controlling (S4) the envelope of the frequency shifted copy of the low band audio signal by said at least one high band parameter.

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: The present invention relates to an improved scheme for coding of audio. In particular, the present invention relates to an encoder device and a method for coding an input signal in an encoder system. The method comprises applying a first mode to the input signal to form a first output and applying a second mode to the input signal to form a second output. A first processed output is then formed from at least a part of the first output, and a second processed output is formed from at least a part of the second output. Forming a second processed output comprises estimating a part of the input signal from at least a part of the second output. Then, an optimum mode is determined based on the firstprocessedoutput and the secondprocessedoutput, and the output according to the optimum mode is selected.

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.