Dolby International Ab

Dolby Laboratories Licensing Corporation

Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V.

Dobly International Ab

Alexander Groeschel

Jeffrey Charles Riedmiller

Holger Hoerich

Michael C Ward

Kristofer Kjoerling

Scott Gregory Norcross

Alan Jeffrey Seefeldt

Jeroen Koppens

Wolfgang Alexander Schildbach

Jens Popp

Heiko Purnhagen

Karl Jonas Roeden

Michael Schug

Martin Wolters

Juergen Herre

Christof Fersch

Karsten Linzmeier

Thomas Ziegler

Tobias Friedrich

Daniel Homm

Phillip Williams

Malte Schmidt

Kurt Krauss

Reinhold Boehm

Marc Roessler

Kristopher Kjoerling

Oliver Watzke

Jarret A Cooper

Wolfgang A Schildbach

Jenns Popp

An audio signal (X) is represented by a bitstream (B) segmented into frames. An audio processing system () comprises a buffer () and a decoding section (). The buffer joins sets of audio data (D; D, . . . , D) carried by N respective frames (F, F, . . . , F) into one decodable set of audio data (D) corresponding to a first frame rate and to a first number of samples of the audio signal per frame. The frames have a second frame rate corresponding to a second number of samples of the audio signal per frame. The first number of samples is N times the second number of samples. The decoding section decodes the decodable set of audio data into a segment of the audio signal by at least employing signal synthesis, based on the decodable set of audio data, with a stride corresponding to the first number of samples of the audio signal.

Encoding and decoding of audio signals

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.

Dynamic range control for a wide variety of playback environments

A method () for generating a bitstream () indicative of an object based audio program is described. The bitstream () comprises a sequence of containers (). A first container () of the sequence of containers () comprises a plurality of substream entities () for a plurality of substreams () of the object based audio program and a presentation section (). The method () comprises determining () a set of object channels (). The method () further comprises providing () a set of object related metadata () for the set of object channels (). In addition, the method () comprises inserting () a first set of object channel frames and a first set of object related metadata frames into a respective set of substream entities () of the first container (). Furthermore, the method () comprises inserting () presentation data into the presentation section ().

Smart access to personalized audio

The present document relates to the determination of a 200 bitrate related to an encoded bitstream, and describes a method for determining an estimate of a bitrate of a bitstream comprising a sequence of frames comprising a varying number of bitsand corresponding to excerpts of an audio and/or video signal. At least two frames of the sequence of frames comprise a parameter indicative of a processing delay for the corresponding frame. The method comprises determining: a total number of bits for a subsequence of frames from the bitstream; a corrected number 201 of frames based on a number of frames comprised within the subsequence and the parameters of at least two frames of the subsequence; and a lower bitrate bound and an upper bitrate bound of the bitrate based on the total number of bits, the corrected number of frames and a frame rate of the bitstream.

Method for bitrate signaling and bitstream format enabling such method

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.

Loudness adjustment for downmixed audio content

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.

System aspects of an audio codec

An audio processing system () accepts an audio bitstream having one of a plurality of predefined audio frame rates. The system comprises a front-end component (), which receives a variable number of quantized spectral components, corresponding to one audio frame in any of the predefined audio frame rates, and performs an inverse quantization according to predetermined, frequency-dependent quantization levels. The front-end component may be agnostic of the audio frame rate. The audio processing system further comprises a frequency-domain processing stage () and a sample rate converter (), which provide a reconstructed audio signal sampled at a target sampling frequency independent of the audio frame rate. By its frame-rate adaptability, the system can be configured to operate frame-synchronously in parallel with a video processing system that accepts plural video frame rates.

Audio decoder utilizing sample rate conversion for audio and video frame synchronization

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. The one or more portions of specific loudness metadata relating to the specific playback environment determine loudness adjustment gains from the one or more portions of specific loudness metadata for the specific playback environment, apply the loudness adjustment gains as a part of overall gains applied to the audio sample data to generate output audio data.

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.

Audio stream mixing with dialog level normalization

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 () 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 (). In case a first gain value () is not sufficient for protection, the respective first gain value () 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 (), the method may add gain values sufficient for protection against signal clipping.

Signal clipping protection using pre-existing audio gain metadata

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.

Authentication of data streams

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Nuremberg, Germany

Alexander Groeschel