Abstract: A digital audio signal can be processed using continuously variable time-frequency resolution by selecting a portion of an input digital audio signal, resampling the selected portion of the input digital audio signal, generating a plurality of spectral characteristics associated with the resampled portion of the input digital audio signal, generating a portion of an output digital audio signal from the plurality of spectral characteristics, and resampling the portion of the output digital audio signal. Further, resampling the selected portion of the input digital audio signal can comprise determining a sampling ratio and resampling the selected portion of the input digital audio signal in accordance with the determined sampling ratio. Additionally, the portion of the output digital audio signal can be resampled in accordance with the inverse of the determined sampling ratio. The sampling ratio can be determined based on a time-frequency resolution requirement associated with an audio processing algorithm.

Abstract: A method and apparatus for mixed dimensionality encoding and decoding are provided in embodiments of the present invention. The method includes: obtaining at least one variable collection through calculation according to a processed spectral coefficient, determining a processing dimension for a spectral coefficient to be processed, according to a relationship between the at least one variable collection and a corresponding threshold collection, and performing, according to a selected dimension, encoding or decoding under the dimension on the spectral coefficient to be processed. Through the preceding technical means, different processing dimensions are used for different spectral coefficients, improving the encoding and decoding efficiency.

Abstract: The invention relates to a method for supporting an encoding of an audio signal, wherein at least one section of the audio signal is to be encoded with a coding model that allows the use of different coding frame lengths. In order to enable a simple selection of the respectively best suited coding frame length, it is proposed that at least one control parameter is determined based on signal characteristics of the audio signal. The control parameter is then used for limiting the options of possible coding frame lengths for the at least one section. The invention relates equally to a module 10,11 in which this method is implemented, to a device 1 and a system comprising such a module 10,11, and to a software program product including a software code for realizing the proposed method.

Abstract: A filter bank device for generating a complex spectral representation of a discrete-time signal includes a generator for generating a block-wise real spectral representation, which, for example, implements an MDCT, to obtain temporally successive blocks of real spectral coefficients. The output values of this spectral conversion device are fed to a post-processor for post-processing the block-wise real spectral representation to obtain an approximated complex spectral representation having successive blocks, each block having a set of complex approximated spectral coefficients, wherein a complex approximated spectral coefficient can be represented by a first partial spectral coefficient and by a second partial spectral coefficient, wherein at least one of the first and second partial spectral coefficients is determined by combining at least two real spectral coefficients.

Abstract: Provided are a method and apparatus for encoding or decoding an audio signal or a speech signal. In the encoding method, encoding is performed by performing domain transformation on a received signal in units of frequency bands by applying a psychoacoustic model, encoding the transformation result with respect to predetermined one or more frequency bands by using a high temporal resolution coding tool, and then quantizing the encoding result. In the decoding method, decoding is performed by inversely quantizing signals obtained by encoding in units of frequency bands, decoding one or more signals from among the inversely quantized signals, which are allocated to one or more frequency bands which have a predetermined domain resolution, determined by applying the psychoacoustic model, that is greater than a predetermined value, according to a predetermined method, and then inversely transforming either the inversely quantized or the one or more decoded signals.

Abstract: A method and apparatus for frame classification and rate determination in voice transcoders. The apparatus includes a classifier input parameter preparation module that unpacks the bitstream from the source codec and selects the codec parameters to be used for classification, parameter buffers that store previous input and output parameters of previous frames, and a frame classification and rate decision module that uses the source codec parameters from the current frame and zero or more frames to determine the frame class, rate, and classification feature parameters for the destination codec. The classifier input parameter preparation module separates the bitstream code and unquantizes the sub-codes into the codec parameters. The frame classification and rate decision module comprises M sub-classifiers and a final decision module.