Abstract: A method is shown for supporting a multichannel audio extension at an encoding end of a multichannel audio coding system. In order to improve the audio quality over a large frequency range, the method comprises transforming each channel of a multichannel audio signal into the frequency domain and dividing a bandwidth of the frequency domain signals into a first region of lower frequencies and at least one further region of higher frequencies. Then, the frequency domain signals are encoded in each of the frequency regions with another type of coding to obtain parametric multichannel extension information for the respective frequency region. The invention relates equally to a method for supporting in a corresponding manner a multichannel audio extension at a decoding end. Also shown are a corresponding encoder, a corresponding decoder, and corresponding devices, systems and software program products.

Abstract: A decoder (e.g., an AAC-LTP decoder) receives a stream containing coded audio data and prediction data. The coded data is upsampled or downsampled during decoding. Portions of the decoded data are stored in a buffer for use in decoding subsequent coded data. The buffer into which the decoded data is placed has different dimensions than a buffer used in a coder when generating the coded data. A portion of the data in the decoder buffer is identified and modified with interleaved zero values so as to correspond to the dimensions of the prediction coding buffer in the coder.

Abstract: Coding an audio signal includes selecting a reference sequence that has the smallest error relative to a sequence of the audio signal, calculating pitch predictor coefficients for the selected reference sequence using one of a set of pitch predictor orders, producing a predicted sequence from the selected reference sequence using the calculated pitch predictor coefficients, and calculating a coding error by comparing the predicted sequence to the sequence to be coded. Coding also includes calculating pitch predictor coefficients for the selected reference sequence, producing a predicted sequence from the selected reference sequence, and calculating a coding error by comparing the predicted sequence to the sequence to be coded, for each of the remaining orders of the set of pitch predictor orders, and using an order from the set of pitch predictor orders that results in the smallest coding error to select a coding method for the sequence to be coded.

Abstract: The invention relates to a method in the decompression of a compressed audio signal. In the decompression, a predicting coding has been used, wherein samples taken of the audio signal have been formed into frames, and the samples of the frames have been compared with past samples to find out the prediction error. In the method, frames (frn, frn?1, frn?2) of the compressed audio signal are stored, and a predicting decoding is used to decompress the audio signal compressed with the predicting coding, on the basis of said stored frames (frn, frn?1, frn?2). In the method, at least one memory pointer is used to indicate the storage location of the frames (frn, frn?1, frn?2). Said memory pointers (P1, P2) are used to point to the storage location of the frame (frn?1) preceding the frame (frn) being processed at the time, and to the storage location of the frame (frn?2) preceding said past frame (frn?1).

Abstract: An error concealment method and device for recovering lost data in the AAC bitstream in the compressed domain. The bitstream are partitioned into frames each having a plurality of data parts including the header/global gain, scale factors and QMDCT coefficients. The data parts are stored in a plurality of buffers, so that if one or more data parts of a current frame is corrupted or lost, the corresponding data part in the neighboring frames is used to conceal the errors in the current frame.