Abstract: The present application describes a method (400) for providing personalized audio to a user. The method (400) comprises receiving (401) a manifest file (140) for a media element from which audio is to be rendered, wherein the manifest file (140) comprises a description (141) for a plurality of different presentations (152) of audio content of the media element. In addition, the method (400) comprises selecting (402) a presentation (152) from the plurality of presentations (152) based on the manifest file (140). The method (400) further comprises receiving (403) a list of audio track objects comprised within the media element, and selecting (404) an audio track object from the list of audio track objects, in dependence of the selected presentation (152).

Abstract: The present document relates to methods and systems for audio encoding. In particular, the present document relates to methods and systems for fast audio encoding using a parallel system architecture. A frame-based audio encoder (300, 400, 500, 600) comprising K parallel transform units (303, 403) is described; wherein each of the K parallel transform units (303, 403) is configured to transform a respective one of a group of K frames (305) of an audio signal (101) into a respective one of K sets of frequency coefficients; wherein K>1; wherein each of the K frames (305) comprises a plurality of samples of the audio signal (101).

Abstract: The present document relates to methods and systems for audio encoding. In particular, the present document relates to methods and systems for fast audio encoding using a parallel system architecture. A frame-based audio encoder (300, 400, 500, 600) comprising K parallel transform units (303, 403) is described; wherein each of the K parallel transform units (303, 403) is configured to transform a respective one of a group of K frames (305) of an audio signal (101) into a respective one of K sets of frequency coefficients; wherein K>1; wherein each of the K frames (305) comprises a plurality of samples of the audio signal (101).

Abstract: The present document relates to the design of anti-aliasing and/or anti-imaging filters for resamplers using rational resampling factors. In particular, the present document relates to a method for designing such filters having a reduced number of filter coefficients or an increased perceptual performance, as well as to the filters designed using such method. A method for designing a filter (102) configured to reduce imaging and/or aliasing of an output audio signal (113) at an output sampling rate (fsout) is described. The output audio signal (113) is a resampled version of an input audio signal (110) at an input sampling rate (fsin). The ratio of the output sampling rate (fsout) and the input sampling rate (fsin) is a rational number N/M. The filter (102) operates at an upsampled sampling rate which equals N times the input sampling rate (fsin).

Abstract: An audio encoder (109) has a hierarchical encoding structure and generates a data stream comprising one or more audio channels as well as parametric audio encoding data. The encoder (109) comprises an encoding structure processor (305) which inserts decoder tree structure data into the data stream. The decoder tree structure data comprises at least one data value indicative of a channel split characteristic for an audio channel at a hierarchical layer of the hierarchical decoder structure and may specifically specify the decoder tree structures to be applied by a decoder A decoder (115) comprises a receiver (401) which receives the data stream and a decoder structure processor (405) for generating the hierarchical decoder structure in response to the decoder tree structure data. A decode processor (403) then generates output audio channels from the data stream using the hierarchical decoder structure.

Abstract: Analyzing an analysis time signal that has been generated from encoding and decoding and original time signal according to an encoding algorithm. The encoding block raster underlying the analysis time signal used by the encoding algorithm is determined. The analysis time signal is converted from its timely representation of analysis spectral coefficients to a spectral representation by using the established encoding block raster. At least two analysis spectral coefficients are grouped. The greatest common divisor of the analysis spectral coefficients are calculated, corresponding to the quantization step width used when quantizing the encoding algorithm or an integer multiple of it. In the case of an audio signal, the scale factor can easily be established for this group of spectral coefficients, i.e., for a scale factor band, from the quantization step width. All parameters used for the quantization of the original time signal are known; full iteration loops need not be performed.

Abstract: An apparatus for analyzing an analysis time signal that has been generated from encoding and decoding an original time signal according to an encoding algorithm first, wherein first the encoding block raster underlying the analysis time signal used by the encoding algorithm is determined. Thereupon, the analysis time signal will be converted from its timely representation comprising a plurality of analysis spectral coefficients, to a spectral representation by using the established encoding block raster. Then, at least two analysis spectral coefficients or at least two spectral coefficients derived from the analysis spectral coefficients by multiplication of an encoding amplification factor or by multiplication with a compression function are grouped.

Abstract: In determining a coding block raster on which a decoded signal is based, a segment of the decoded signal is picked out first, said segment beginning at a certain output sampling value of the decoded signal. Said segment is then converted into a spectral representation, whereupon said spectral representation is then evaluated in relation to a predetermined criterion in order to obtain an evaluation result for the segment. This procedure is repeated for a plurality of different segments beginning at different output sampling values each, in order to obtain a plurality of evaluation results. Finally, the plurality of the evaluation results is searched in order to establish the evaluation result that has an extreme value as compared to the other evaluation results, in such a way that it can be assumed that the segment to which this evaluation result is allocated matches the coding block raster on which the decoded signal is based.

Abstract: In determining a coding block raster on which a decoded signal is based, a segment of the decoded signal is picked out first (11), said segment beginning at a certain output sampling value of the decoded signal. Said segment is then converted into a spectral representation (12), whereupon said spectral representation is then evaluated in relation to a predetermined criterion (13) in order to obtain an evaluation result for the segment. This procedure is repeated for a plurality of different segments beginning at different output sampling values each, in order to obtain a plurality of evaluation results. Finally, the plurality of the evaluation results is searched (14) in order to establish the evaluation result that has an extreme value as compared to the other evaluation results, in such a way that it can be assumed that the segment to which this evaluation result is allocated matches the coding block raster on which the decoded signal is based.

Abstract: In a method for concealing errors in an audio data stream the occurrence of an error is detected in the audio data stream, audio data prior to the occurrence of the fault being intact audio data. Thereafter a spectral energy of a subgroup of the intact audio data is calculated. After forming a pattern for substitute data on the basis of the spectral energy calculated for the subgroup of the intact audio data, substitute data for erroneous or missing audio data which correspond to the subgroup are created on the basis of the pattern.