Dolby International Ab

Dolby Laboratories Licensing Corporation

Dobly International Ab

Robin Thesing

Heiko Purnhagen

Kristofer Kjoerling

Karl Jonas Roeden

Harald Mundt

Robert Loring Andersen

Michael Schug

James Michael Silva

Martin Wolters

Klaus Peichl

Lars Villemoes

Per Ekstrand

Leif Sehlstrom

Tobias Friedrich

Karsten Linzmeier

Alexander Stahlmann

Phillip Williams

Stanislaw Gorlow

Reinhold Boehm

Michael Beer

Simon Plain

Leif Sehlström

Jan Mueller

Mark C Leddy

Vinay Mathew

Mark Leddy

James M Silva

There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.

Audio encoder and decoder for interleaved waveform coding

A dynamic range control unit () configured to apply dynamic range control, referred to as DRC, to an audio signal (). The DRC unit () is configured to downsample a subband signal () derived from the audio signal (), to provide a downsampled subband signal (), to determine a DRC gain () based on the downsampled subband signal (), and to apply the DRC gain () to the subband signal (), to provide a compressed subband signal () of a compressed audio signal ().

Method and unit for performing dynamic range control

A technique including receiving and decoding a coded bitstream encoded with audio content including first audio objects corresponding to a first media content type of two consecutive media content types and second audio objects corresponding to a second media content type of the two consecutive media content types, and audio metadata corresponding to the audio content. The audio metadata including first and second audio object gains, for the first and second audio objects, generated in part based on a first fading curve of the first media content type and a second fading curve of the second media content type, respectively. The technique further includes applying the first and second audio object gains to the first and second audio objects, and rendering a sound field represented by the first audio object with the applied first audio object gain and the second audio object with the applied second audio object gain.

Using metadata to aggregate signal processing operations

A method for rendering an audio output based on an audio data stream including M audio signals, side information including a series of reconstruction instances of a reconstruction matrix C and first timing data, the side information allowing reconstruction of N audio objects from the M audio signals, and object metadata defining spatial relationships between the N audio objects. The method includes generating a synchronized rendering matrix based on the object metadata, the first timing data, and information relating to a current playback system configuration, the synchronized rendering matrix having a rendering instance for each reconstruction instance, multiplying each reconstruction instance with a corresponding rendering instance to form a corresponding instance of an integrated rendering matrix, and applying the integrated rendering matrix to the audio signals in order to render an audio output.

Integrated reconstruction and rendering of audio signals

There is provided an audio decoder and a method therein for transforming a digital audio signal from a first frequency domain to a second frequency domain. For each received frame of the digital audio signal, the method identifies an upper limit of the frequency range, and if the upper limit of the frequency range is below the Nyquist frequency of said frame of the digital audio signal by more than a threshold amount, the Nyquist frequency of said frame of the digital audio signal is lowered from its original value to a reduced value by removing spectral bands of said frame of the digital audio signal above the identified upper limit of the frequency range. Thereafter said frame of the digital audio signal is transformed from the first frequency domain to the second frequency domain via an intermediate time domain.

Audio decoder and method for transforming a digital audio signal from a first to a second frequency domain

The present document relates to the technical field of audio coding, decoding and processing. It specifically relates to methods of recovering high frequency content of an audio signal from low frequency content of the same audio signal in an efficient manner. A method for determining a first banded tonality value for a first frequency subband of an audio signal is described. The first banded tonality value is used for approximating a high frequency component of the audio signal based on a low frequency component of the audio signal. The method comprises determining a set of transform coefficients in a corresponding set of frequency bins based on a block of samples of the audio signal; determining a set of bin tonality values for the set of frequency bins using the set of transform coefficients, respectively; and combining a first subset of two or more of the set of bin tonality values for two or more corresponding adjacent frequency bins of the set of frequency bins lying within the first frequency subband, thereby yielding the first banded tonality value for the first frequency subband.

Methods and systems for efficient recovery of high frequency audio content

The present document relates to the technical field of audio coding, decoding and processing. It specifically relates to methods of recovering high frequency content of an audio signal from low frequency content of the same audio signal in an efficient manner. A method for determining a first banded tonality value () for a first frequency subband () of an audio signal is described. The first banded tonality value () is used for approximating a high frequency component of the audio signal based on a low frequency component of the audio signal. The method comprises determining a set of transform coefficients in a corresponding set of frequency bins based on a block of samples of the audio signal; determining a set of bin tonality values () for the set of frequency bins using the set of transform coefficients, respectively; and combining a first subset of two or more of the set of bin tonality values () for two or more corresponding adjacent frequency bins of the set of frequency bins lying within the first frequency subband, thereby yielding the first banded tonality value () for the first frequency subband.

The present document relates to audio decoding and/or audio transcoding. In particular, the present document relates to a scheme for efficiently decoding a number M of audio channels from a bitstream comprising a higher number N of audio channels. In this context a method and system for merging a first and a second source set of spectral band replication (SBR) parameters to a target set of SBR parameters is described. The first and second source set comprise a first and second frequency band partitioning, respectively, which are different from one another. The first source set comprises a first set of energy related values associated with frequency bands of the first frequency band partitioning. The second source set comprises a second set of energy related values associated with frequency bands of the second frequency band partitioning. The target set comprises a target energy related value associated with an elementary frequency band. The method comprises the steps of breaking up the first and the second frequency band partitioning into a joint grid comprising the elementary frequency band; assigning a first value of the first set of energy related values to the elementary frequency band; assigning a second value of the second set of energy related values to the elementary frequency band; and combining the first and second value to yield the target energy related value for the elementary frequency band.

SBR bitstream parameter downmix

A method, an apparatus, a computer readable storage medium configured with instructions for carrying out a method, and logic encoded in one or more computer-readable tangible medium to carry out actions. The method is to decode audio data that includes N.n channels to M.m decoded audio channels, including unpacking metadata and unpacking and decoding frequency domain exponent and mantissa data; determining transform coefficients from the unpacked and decoded frequency domain exponent and mantissa data; inverse transforming the frequency domain data; and in the case M&#x3c;N, downmixing according to downmixing data, the downmixing carried out efficiently.

Audio decoder and decoding method using efficient downmixing

The invention relates to an apparatus () for improving a stereo audio signal of an FM stereo radio receiver (). It is suggested to employ components of a standard HE-ACC v2 decoder () in such apparatus () to efficiently recover the stereo audio signal (L\R') while also exhibiting good signal quality. To that end, parametric stereo parameters () shall be quantized and submitted to such decoder () together with a mono-downmix signal (). The decoder () can then easily recover the parametric stereo parameters () by de-quantizing them via a simple lookup table, and finally creating the stereo output signal (L&#x2032;R&#x2032;) from the de-quantized parametric stereo parameters and the mono-downmix signal (). The apparatus () further includes a parameter estimation stage () which may again be embodied on the basis of components of a standard HE-ACC v2 encoder (). After estimating the parametric stereo parameters () and generating the mono-downmix signal (), the encoder () will not use further encoding other than the already performed PS-coding. An apparatus () according to the invention therefore advantageously includes decoder () and/or encoder () components of a standard HE-ACC v2 codec&#x2014;utilizing its parametric stereo parameter processing capabilities.

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Nuremberg, Germany

Robin Thesing