Abstract: Apparatus and methods for generating an encoded audio bitstream, including by including substream structure metadata (SSM) and/or program information metadata (PIM) and audio data in the bitstream. Other aspects are apparatus and methods for decoding such a bitstream, and an audio processing unit (e.g., an encoder, decoder, or post-processor) configured (e.g., programmed) to perform any embodiment of the method or which includes a buffer memory which stores at least one frame of an audio bitstream generated in accordance with any embodiment of the method.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (a) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: An audio processing unit (APU) is disclosed. The APU includes a buffer memory configured to store at least one frame of an encoded audio bitstream, where the encoded audio bitstream includes audio data and a metadata container. The metadata container includes a header and one or more metadata payloads after the header. The one or more metadata payloads include dynamic range compression (DRC) metadata, and the DRC metadata is or includes profile metadata indicative of whether the DRC metadata includes dynamic range compression (DRC) control values for use in performing dynamic range compression in accordance with at least one compression profile on audio content indicated by at least one block of the audio data.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: Disclosed is a non-transitory computer readable storage medium which receives, by an audio decoder (operating in a specific playback environment different from a reference channel configuration), an audio signal for the reference channel configuration. The audio signal includes audio sample data and encoder-generated loudness metadata which includes a plurality of portions of loudness metadata for a plurality of playback environments. The plurality of portions of loudness metadata includes one or more respective portions of loudness metadata for each playback environment in the plurality of playback environments. The medium also selects one or more portions of specific loudness metadata (based on the specific playback environment), from among the plurality of portions of loudness metadata for the plurality of playback environments.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: Essence of a media program can be encoded such that it carries essence identification information and a set of time points. Auxiliary content such as companion content and other auxiliary data can be associated with the essence of the media program by way of the set of time points. The set of time points and/or fingerprints used to retrieve/recover the set of time points can be delivered to one or more of a media device, a companion device/application, etc., in any of a variety of different methods. Through the set of time points and/or fingerprints, and user preferences, the auxiliary content can be retrieved by the one or more of the media device, the companion device/application, etc., presented and/or interacted with a user. In some embodiments, the auxiliary content as described herein can be customized for a specific sender and/or a specific recipient.

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 implementing such anti-aliasing and/or anti-imaging filters with reduced computational complexity. In addition, the present document relates to further aspects of an audio encoding and decoding system, such as the phase relation between the channels of a multi-channel audio signal and/or the structure of the bitstream of an encoded audio signal.

Abstract: A reference fingerprint is generated from essence of each corresponding time segment of a media program. A watermark request with the reference fingerprint is sent to a media content identification server. An individual watermark value for the corresponding time segment of the media program is received from the media content identification server in a response to the watermark request. A watermarked version of the media program is generated to comprise one or more individual watermark values embedded in essence of the media program that include the individual watermark value for the corresponding time segment of the media program. Watermark data rates and update rates can be chosen dynamically based on underlying media content. Existing watermark values for media content in a content identification data repository may be updated dynamically when new media content is added.

Abstract: Apparatus and methods for generating an encoded audio bitstream, including by including substream structure metadata (SSM) and/or program information metadata (PIM) and audio data in the bitstream. Other aspects are apparatus and methods for decoding such a bitstream, and an audio processing unit (e.g., an encoder, decoder, or post-processor) configured (e.g., programmed) to perform any embodiment of the method or which includes a buffer memory which stores at least one frame of an audio bitstream generated in accordance with any embodiment of the method.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: Techniques for synthesizing a parameterized haptic track from a multichannel audio signal and embedding the haptic track in a multichannel audio codec bit stream. The haptic track synthesis can occur while encoding the multichannel audio signal as the multichannel audio codec bit stream. The haptic track can be synthesized in a way that allows select parameters of the haptic track to be extracted from the audio signal. The parameters can be adjusted by a user during the encoding process. For example, the parameter adjustments can be made using a authoring/monitoring tool. The parameter adjustments can be recorded as metadata that, along with the haptic track, is included in the codec bit stream. In some aspects the present technology, the adjustable parameters include center frequency, gain, decay rate, and other parameters that allow for modulation of the haptic track during decoding of the codec bit stream.

Abstract: A media signal is accessed, which has been generated with one or more first processing operations. The media signal includes one or more sets of artifacts, which respectively result from the one or more processing operations. One or more features are extracted from the accessed media signal. The extracted features each respectively correspond to the one or more artifact sets. Based on the extracted features, a conditional probability score and/or a heuristically based score is computed, which relates to the one or more first processing operations.

Abstract: Methods for generating an object based audio program, renderable in a personalizable manner, and including a bed of speaker channels renderable in the absence of selection of other program content (e.g., to provide a default full range audio experience). Other embodiments include steps of delivering, decoding, and/or rendering such a program. Rendering of content of the bed, or of a selected mix of other content of the program, may provide an immersive experience. The program may include multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects), the bed of speaker channels, and other speaker channels. Another aspect is an audio processing unit (e.g., encoder or decoder) configured to perform, or which includes a buffer memory which stores at least one frame (or other segment) of an object based audio program (or bitstream thereof) generated in accordance with, any embodiment of the method.

Abstract: The present document relates to transcoding of metadata, and in particular to a method and system for transcoding metadata with reduced computational complexity. A transcoder configured to transcode an inbound bitstream comprising an inbound content frame and an associated inbound metadata frame into an outbound bitstream comprising an outbound content frame and an associated outbound metadata frame is described. The inbound content frame is indicative of a signal encoded according to a first codec system and the outbound content frame is indicative of the signal encoded according to a second codec system. The transcoder is configured to identify an inbound block of metadata from the inbound metadata frame, the inbound block of metadata associated with an inbound descriptor indicative of one or more properties of metadata comprised within the inbound block of metadata, and to generate the outbound metadata frame from the inbound metadata frame based on the inbound descriptor.

Abstract: Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.

Abstract: Methods for generating or decoding an encoded audio bitstream including audio data and supplementary data (e.g., metadata and/or unrelated audio data), where at least some of the supplementary data is included as LSBs of audio segments, and/or at least some of the supplementary data is included in guard bands. Typical embodiments provide a scalable and video synchronous format compatible with real-time and file-based infrastructure components that support the SMPTE 337 format for carrying data in AES3 serial bitstreams, and/or provide a framework for extending distribution codecs to scale beyond an 8-channel limit to support multiples of 8 channels synchronously across multiple AES3 interfaces. Another aspect is an audio processing unit configured to perform any embodiment of the method or including a buffer memory storing at least one segment of an audio bitstream generated in accordance with any embodiment of the method.