Dolby Laboratories Licensing Corporation

Dolby International Ab

Juan Felix Torres

David S McGrath

Daniel Arteaga

Ming Wang

Giulio Cengarle

Stefanie Brown

Stefan Bruhn

Michael William Mason

Antonio Mateos Sole

Michael John Smithers

Timothy Alan Port

Michael Eckert

Adam J Mills

Sebastian P B Holzapfel

Rishabh Tyagi

Andrew Robert Owen

Heiko Purnhagen

Nicholas Luke Appleton

Mark David De Burgh

William Thomas Rowley

Brian George Arnott

Jyri Tapani Pakarinen

Antonio Mateos Solé

Brian B Lee

Ho Young Sung

Mark David deBurgh

Sebastian PB Holzapfel

David S Mcgrath

Encoding/decoding an immersive voice and audio services (IVAS) bitstream comprises: encoding/decoding a coding mode indicator in a common header (CH) section of an IVAS bitstream, encoding/decoding a mode header or tool header in the tool header (TH) section of the bitstream, the TH section following the CH section, encoding/decoding a metadata payload in a metadata payload (MDP) section of the bitstream, the MDP section following the CH section, encoding/decoding an enhanced voice services (EVS) payload in an EVS payload (EP) section of the bitstream, the EP section following the CH section, and on the encoder side, storing or streaming the encoded bitstream, and on the decoder side, controlling an audio decoder based on the coding mode, the tool header, the EVS payload, and the metadata payload or storing a representation of same.

Encoding and decoding IVAS bitstreams

Methods and systems for improving signal processing by smoothing the covariance matrix of a multi-channel signal by setting a forgetting factor based on the bins of a band. A method and system for resetting the smoothing based on transient detection is also disclosed. A method and system for resampling for the smoothing during a banding transition is also disclosed.

Systems and methods for covariance smoothing

Embodiments are disclosed for bitrate distribution in immersive voice and audio services. In an embodiment, a method of encoding an IVAS bitstream comprises: receiving an input audio signal; downmixing the input audio signal into one or more downmix channels and spatial metadata; reading a set of one or more bitrates for the downmix channels and a set of quantization levels for the spatial metadata from a bitrate distribution control table; determining a combination of the one or more bitrates for the downmix channels; determining a metadata quantization level from the set of metadata quantization levels using a bitrate distribution process; quantizing and coding the spatial metadata using the metadata quantization level; generating, using the combination of one or more bitrates, a downmix bitstream for the one or more downmix channels; combining the downmix bitstream, the quantized and coded spatial metadata and the set of quantization levels into the IVAS bitstream.

Bitrate distribution in immersive voice and audio services

The disclosure herein generally relates to capturing, acoustic pre-processing, encoding, decoding, and rendering of directional audio of an audio scene. In particular, it relates to a device adapted to modify a directional property of a captured directional audio in response to spatial data of a microphone system capturing the directional audio. The disclosure further relates to a rendering device configured to modify a directional property of a received directional audio in response to received spatial data.

Audio processing in immersive audio services

The present document describes a method () for generating a bitstream (), wherein the bitstream () comprises a sequence of superframes () for a sequence of frames of an immersive audio signal (). The method () comprises, repeatedly for the sequence of superframes (), inserting () coded audio data () for one or more frames of one or more downmix channel signals () derived from the immersive audio signal (), into data fields () of a superframe (); and inserting () metadata () for reconstructing one or more frames of the immersive audio signal () from the coded audio data (), into a metadata field () of the superframe ().

Methods and devices for generating or decoding a bitstream comprising immersive audio signals

The disclosed embodiments enable converting audio signals captured in various formats by various capture devices into a limited number of formats that can be processed by an audio codec (e.g., an Immersive Voice and Audio Services (IVAS) codec). In an embodiment, a simplification unit of the audio device receives an audio signal captured by one or more audio capture devices coupled to the audio device. The simplification unit determines whether the audio signal is in a format that is supported/not supported by an encoding unit of the audio device. Based on the determining, the simplification unit, converts the audio signal into a format that is supported by the encoding unit. In an embodiment, if the simplification unit determines that the audio signal is in a spatial format, the simplification unit can convert the audio signal into a spatial 'mezzanine' format supported by the encoding.

Transforming audio signals captured in different formats into a reduced number of formats for simplifying encoding and decoding operations

The present document relates to methods and apparatus for rendering input audio for playback in a playback environment. The input audio includes at least one audio object and associated metadata, and the associated metadata indicates at least a location of the audio object. A method for rendering input audio including divergence metadata for playback in a playback environment comprises creating two additional audio objects associated with the audio object such that respective locations of the two additional audio objects are evenly spaced from the location of the audio object, on opposite sides of the location of the audio object when seen from an intended listener's position in the playback environment, determining respective weight factors for application to the audio object and the two additional audio objects, and rendering the audio object and the two additional audio objects to one or more speaker feeds in accordance with the determined weight factors. The present document further relates to methods and apparatus for rendering audio input including extent metadata and/or diffuseness metadata for playback in a playback environment.

Rendering of immersive audio content

An audio object including audio content and object metadata is received. The object metadata indicates an object spatial position of the audio object to be rendered by audio speakers in a playback environment. Based on the object spatial position and source spatial positions of the audio speakers, initial gain values for the audio speakers are determined. The initial gain values can be used to select a set of audio speakers from among the audio speakers. Based on the object spatial position and a set of source spatial positions at which the set of audio speakers are respectively located in the playback environment, a set of non-negative optimized gain values for the set of audio speakers is determined. The audio object at the object spatial position is rendered with the set of optimized gain values for the set of audio speakers.

Adaptive panner of audio objects

Systems, methods, and computer program products of audio processing based on Adaptive Intermediate Spatial Format (AISF) are described. The AISF is an extension to ISF that allows spatial resolution around an ISF ring to be adjusted dynamically with respect to content of incoming audio objects. An AISF encoder device adaptively warps each ISF ring during ISF encoding to adjust angular distance between objects, resulting in increase in uniformity of energy distribution around the ISF ring. At an AISF decoder device, matrices that decode sound positions to the output speaker take into account the warping that was performed at the AISF encoder device to reproduce the true positions of sound sources.

Audio processing in adaptive intermediate spatial format

An apparatus and method of excursion protection of a loudspeaker. The method includes attenuating selected bands in a transform domain, controlled by a feedback signal resulting from an excursion transfer function that has been modified according to the real-time operational characteristics of the loudspeaker. In this manner, the system reduces the amount of wideband attenuation needed to address the predicted excursion, resulting in a better listening experience.

Loudspeaker excursion protection

Systems and methods for manual characterization of perceived transducer distortion are described. The system includes a signal generator, a controller, a multi-band compressor, and an audio transducer. The signal generator is configured to generate a test signal for a frequency band. The test signal includes at least two simultaneous frequency-modulated tones, which may be combined together in an amplitude-modulated envelope. The at least two simultaneous frequency-modulated tones have different frequencies within the frequency band. The controller is configured to receive user input indicating a gain value. The multi-band compressor is coupled to the signal generator. The multi-band compressor is configured to adjust an amplitude of a component of the test signal based on the gain value. The audio transducer is coupled to the multi-band compressor. The audio transducer is configured to generate an audio signal based on the test signal.

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Darlinghurst, Australia

Juan Felix Torres