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: Described herein is a method for processing of media data for playback, wherein the method includes the steps of: (a) fetching, by a web proxy, from two or more media servers media data and a media manifest file including metadata information relating to the fetched media data, and merging, by the web proxy, the media data; (b) modifying, by said web proxy, the content of the media manifest file and/or the content of the media data; and (c) providing, by said web proxy, the media manifest file and the media data as modified in step (b) to a media retrieval element for receiving and processing the media manifest file and the media data for decoding or playback, wherein a localhost address is assigned to the web proxy and the web proxy acts as a server for said media retrieval element. Described are further a respective device and computer program product.

Abstract: A method of playing out media from a media engine run on a receiving apparatus, the method comprising: at the receiving apparatus, receiving a media data structure comprising audio or video content formatted in a plurality of layers, including at least a first layer comprising the audio or video content encoded according to an audio or video encoding scheme respectively, and a second layer encapsulating the encoded content in one or more media containers according to a media container format; determining that at least one of the media containers further encapsulates runnable code for processing at least some of the formatting of the media data structure in order to support playout of the audio or video content by the media engine; running the code on a code engine of the receiving apparatus in order to perform the processing of the media data structure for input to the media engine.

Abstract: A method of playing out media from a media engine run on a receiving apparatus, the method comprising: at the receiving apparatus, receiving a media data structure comprising audio or video content formatted in a plurality of layers, including at least a first layer comprising the audio or video content encoded according to an audio or video encoding scheme respectively, and a second layer encapsulating the encoded content in one or more media containers according to a media container format; determining that at least one of the media containers further encapsulates runnable code for processing at least some of the formatting of the media data structure in order to support playout of the audio or video content by the media engine; running the code on a code engine of the receiving apparatus in order to perform the processing of the media data structure for input to the media engine.

Abstract: Described herein is a method for processing of media data for playback, wherein the method includes the steps of: (a) fetching, by a web proxy, from two or more media servers media data and a media manifest file including metadata information relating to the fetched media data, and merging, by the web proxy, the media data; (b) modifying, by said web proxy, the content of the media manifest file and/or the content of the media data; and (c) providing, by said web proxy, the media manifest file and the media data as modified in step (b) to a media retrieval element for receiving and processing the media manifest file and the media data for decoding or playback, wherein a localhost address is assigned to the web proxy and the web proxy acts as a server for said media retrieval element. Described are further a respective device and computer program product.

Abstract: A method for inserting a first audio signal into a bitstream which comprises a downmix signal and associated bitstream metadata is described. The downmix signal and associated bitstream metadata are indicative of an audio program comprising a plurality of spatially diverse audio signals. The downmix signal comprises at least one audio channel and the bitstream metadata comprise upmix metadata for reproducing the plurality of spatially diverse audio signals from the at least one channel. The method comprises mixing the first audio signal with the at least one audio channel to generate a modified downmix signal. The method further comprises generating an output bitstream comprising the modified downmix signal and the associated modified bitstream metadata indicative of a modified audio program comprising a plurality of modified spatially diverse audio signals.

Abstract: A method (500) for resampling an audio signal (110) is described. The method (500) comprising providing (501) a set of input subband signals (210) which is representative of a time domain audio signal. Furthermore, the method (500) comprises applying (502) a first ripple pre-emphasis gain (323) to a first input subband signal (210) of the set of input subband signals (210) to determine a corresponding first output subband signal (213) of a set of output subband signals (213). In addition, the method (500) comprises determining (503) a time domain input audio signal (110) from the set of output subband signals (213).

Abstract: A method for inserting a first audio signal into a bitstream which comprises a downmix signal and associated bitstream metadata is described. The downmix signal and associated bitstream metadata are indicative of an audio program comprising a plurality of spatially diverse audio signals. The downmix signal comprises at least one audio channel and the bitstream metadata comprise upmix metadata for reproducing the plurality of spatially diverse audio signals from the at least one channel. The method comprises mixing the first audio signal with the at least one audio channel to generate a modified downmix signal. The method further comprises generating an output bitstream comprising the modified downmix signal and the associated modified bitstream metadata indicative of a modified audio program comprising a plurality of modified spatially diverse audio signals.

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: A method and apparatus for preserving matrix-surround information in encoded audio/video includes a receiver operative to receive matrix-surround encoded audio signals via a modem, separate the audio signals into a frequency spectrum having discrete audio frequencies, and determine a cutoff threshold used to encode the matrix-surround encoded audio signals. The method and apparatus further includes a decoder operative to decode a first set of the audio frequencies below the determined cutoff threshold using a first matrix-surround preserving audio encoding method and to decode a second set of audio frequencies above the cutoff threshold using a second non matrix-surround preserving audio encoding method.

Abstract: A method for mixing of audio signals that allows maintaining of a consistent perceived sound level for the mixed signal by holding the sound level of the dominant signal in the mix constant by adjusting the sound level of the non-dominant signal(s) in relation to the dominant signal. It further includes receiving of a mixing balance input, which denotes the adjustable balance between the main and associated signals. It further includes identification of the dominant signal from the mixing balance input and mixing metadata, from which an appropriate scale factor for the non-dominant signal may also be determined directly from the scaling information, without the need for any analysis or measurement of the audio signals to be mixed. It further includes scaling the non-dominant signal in relation to the dominant signal and combining the scaled non-dominant signal with the dominant signal into a mixed signal.

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 application describes a method and an apparatus to prevent clipping of an audio signal when protection against signal clipping by received audio metadata is not guaranteed. The method may be used to prevent clipping for the case of downmixing a multichannel signal to a stereo audio signal. According to the method, it is determined whether first gain values (4) based on received audio metadata are sufficient for protection against clipping of the audio signal. The audio metadata is embedded in a first audio stream (1). In case a first gain value (4) is not sufficient for protection, the respective first gain value (4) is replaced with a gain value sufficient for protection against clipping of the audio signal. Preferably, in case no metadata related to dynamic range control is present in the first audio stream (1), the method may add gain values sufficient for protection against signal clipping.

Abstract: The present document relates to techniques for authentication of data streams. Specifically, the present document relates to the insertion of identifiers into a data stream, such as a Dolby Pulse, AAC or HE AAC bitstream, and the authentication and verification of the data stream based on such identifiers. A method and system for encoding a data stream comprising a plurality of data frames is described. The method comprises the step of generating a cryptographic value of a number N of successive data frames and configuration information, wherein the configuration information comprises information for rendering the data stream. The method then inserts the cryptographic value into the data stream subsequent to the N successive data frames.

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: A method for mixing of audio signals that allows maintaining of a consistent perceived sound level for the mixed signal by holding the sound level of the dominant signal in the mix constant by adjusting the sound level of the non-dominant signal(s) in relation to the dominant signal. It further includes receiving of a mixing balance input, which denotes the adjustable balance between the main and associated signals. It further includes identification of the dominant signal from the mixing balance input and mixing metadata, from which an appropriate scale factor for the non-dominant signal may also be determined directly from the scaling information, without the need for any analysis or measurement of the audio signals to be mixed. It further includes scaling the non-dominant signal in relation to the dominant signal and combining the scaled non-dominant signal with the dominant signal into a mixed signal.

Abstract: A method and apparatus for preserving matrix-surround information in encoded audio/video includes a receiver operative to receive matrix-surround encoded audio signals via a modem, separate the audio signals into a frequency spectrum having discrete audio frequencies, and determine a cutoff threshold used to encode the matrix-surround encoded audio signals. The method and apparatus further includes a decoder operative to decode a first set of the audio frequencies below the determined cutoff threshold using a first matrix-surround preserving audio encoding method and to decode a second set of audio frequencies above the cutoff threshold using a second non matrix-surround preserving audio encoding method.

Abstract: The present invention relates to techniques for authentication of data streams. Specifically, the invention relates to the insertion of identifiers into a data stream, such as a Dolby Pulse, AAC or HE AAC bitstream, and the authentication and verification of the data stream based on such identifiers. A method and system for encoding a data stream comprising a plurality of data frames is described. The method comprises the step of generating a cryptographic value of a number N of successive data frames and configuration information, wherein the configuration information comprises information for rendering the data stream. The method then inserts the cryptographic value into the data stream subsequent to the N successive data frames.

Abstract: A method and apparatus for preserving matrix-surround information in encoded audio/video includes a receiver operative to receive matrix-surround encoded audio signals, separate the audio signals into a frequency spectrum having discrete audio frequencies, and determine a cutoff threshold used to encode the matrix-surround encoded audio signals. The method and apparatus further includes a decoder operative to decode a first set of the audio frequencies below the determined cutoff threshold using a first matrix-surround preserving audio encoding method and to decode a second set of audio frequencies above the cutoff threshold using a second non matrix-surround preserving audio encoding method.

Abstract: The application describes a method and an apparatus to prevent clipping of an audio signal when protection against signal clipping by received audio metadata is not guaranteed. The method may be used to prevent clipping for the case of downmixing a multichannel signal to a stereo audio signal. According to the method, it is determined whether first gain values (4) based on received audio metadata are sufficient for protection against clipping of the audio signal. The audio metadata is embedded in a first audio stream (1). In case a first gain value (4) is not sufficient for protection, the respective first gain value (4) is replaced with a gain value sufficient for protection against clipping of the audio signal. Preferably, in case no metadata related to dynamic range control is present in the first audio stream (1), the method may add gain values sufficient for protection against signal clipping.