Patent classifications
H04S2420/03
Inter-channel encoding and decoding of multiple high-band audio signals
A device includes an encoder configured to generate a first high-band portion of a first signal based on a left signal and a right signal. The encoder is also configured to generate a set of adjustment gain parameters based on a high-band non-reference signal and a synthesized signal. The high-band non-reference signal corresponds to one of a left high-band portion of the left signal or a right high-band portion of the right signal.
Method and device for processing audio signal, using metadata
Disclosed is a device for processing an audio signal, which renders an audio signal. The device for processing an audio signal includes a processor. The processor receives metadata including an audio signal and first element reference distance information and renders a first element signal on the basis of the first element reference distance information, wherein the first element reference distance information indicates the reference distance of an element signal. The audio signal is capable of including a second element signal which may be simultaneously rendered with the first element signal, and the metadata is capable of including second element distance information indicating the distance of the second element signal. The number of bits required for representing the first element reference distance information is smaller than the number of bits required for representing the second element distance information.
ELECTRONIC DEVICE, METHOD AND COMPUTER PROGRAM
An electronic device comprising circuitry configured to analyze the results of a stereo or multi-channel source separation to determine one or more time-varying parameters, and to create spatially dynamic audio objects based on the one or more time-varying parameters.
ENABLING STEREO CONTENT FOR VOICE CALLS
Disclosed are systems and methods to modify the Bluetooth mono HFP protocol to support bi-directional stereo operation for high bandwidth audio including 12-KHz wide-band, 16-KHz super wide-band (SWB), and 24-KHz full band (FB) audio. The techniques leverage the larger packet size and longer duty cycle of the 2-EV5 transport packet and expand the block size of the audio frames generated by the AAC-ELD codec to increase the maximum data throughput from the 64 kbps of the mono HFP protocol to 192 kbps using a stereo HFP protocol. The increased throughput not only supports stereo operations, but allows the transport of redundant or FEC packets for increased robustness against packet loss. In one aspect, the AAC-ELD codec may be configured for dynamic bit rate switching to flexibly perform trade-offs between audio quality and robustness against packet loss. The stereo HFP may configure the maximum throughput based on the desired audio quality.
Audio apparatus and method of operation therefor
An audio apparatus, e.g. for rendering audio for a virtual/augmented reality application, comprises a receiver (201) for receiving audio data for an audio scene including a first audio component representing a real-world audio source present in an audio environment of a user. A determinator (203) determines a first property of a real-world audio component from the real-world audio source and a target processor (205) determines a target property for a combined audio component being a combination of the real-world audio component received by the user and rendered audio of the first audio component received by the user. An adjuster determines a render property by modifying a property of the first audio component indicated by the audio data for the first audio component in response to the target property and the first property. A renderer (209) renders the first audio component in response to the render property.
QUANTIZATION OF SPATIAL AUDIO DIRECTION PARAMETERS
A method for spatial audio signal encoding comprising: obtaining a plurality of audio direction parameters, wherein each parameter comprises an elevation value and an azimuth value and wherein each parameter has an ordered position; deriving for each of the plurality of audio direction parameters a corresponding derived audio direction parameter (SP) comprising an elevation and an azimuth value, corresponding derived audio direction parameters (SP) being arranged in a manner determined by a spatial utilization defined by the elevation values and the azimuth values of the plurality of audio direction parameters; rotating each derived audio direction parameter (SP) by the azimuth value (φ.sub.0) of an audio direction parameter in the first position of the plurality of audio direction parameters and quantizing the rotation to determine for each a corresponding quantized rotated derived audio direction parameter; changing the ordered position of an audio direction parameter to a further position coinciding with a position of a rotated derived audio direction parameter when the azimuth value of the audio direction parameter is closest to the azimuth value of the further rotated derived audio direction parameter compared to the azimuth values of other rotated derived audio direction parameters, followed by determining for each of the plurality audio direction parameters a difference between each audio direction parameter and their corresponding quantized rotated derived audio direction parameter; and quantizing a difference for each of the plurality of audio direction parameters, wherein a difference quantization resolution for each of the plurality of audio direction parameters is defined based on a spatial extent of the audio direction parameters.
APPARATUS AND METHOD FOR PROCESSING MULTI-CHANNEL AUDIO SIGNAL
An apparatus for processing audio includes at least one processor configured to obtain a down-mixed audio signal from a bitstream, to obtain down-mixing-related information from the bitstream, to de-mix the down-mixing-related information by using down-mixing-related information, and to reconstruct an audio signal including at least one frame based on the de-mixed audio signal. The down-mixing-related information is information generated in units of frames by using an audio scene type.
Apparatus and Method for Reproducing a Spatially Extended Sound Source or Apparatus and Method for Generating a Description for a Spatially Extended Sound Source Using Anchoring Information
An apparatus for reproducing a spatially extended sound source having a defined position or orientation and geometry in a space has an interface for receiving a listener position. The apparatus having a projector for calculating a projection of a two- or three-dimensional hull associated with the sound source onto a projection plane using the listener position, information on the geometry of the sound source, and on the position of the sound source; a sound position calculator for calculating positions of at least two sound sources for the spatially extended sound source using the projection plane; and a renderer for rendering the at least two sound sources at the positions to obtain a reproduction of the sound source having two or more output signals, configured to use different sound signals for the different positions.
METHODS, APPARATUS AND SYSTEMS FOR REPRESENTATION, ENCODING, AND DECODING OF DISCRETE DIRECTIVITY DATA
The present disclosure relates to a method of processing audio content including directivity information for at least one sound source, the directivity information comprising a first set of first directivity unit vectors representing directivity directions and associated first directivity gains. The disclosure further relates to corresponding methods of encoding and decoding audio content including directivity information for at least one sound source.
AUDIO APPARATUS AND METHOD OF OPERATION THEREFOR
An audio apparatus, e.g. for rendering audio for a virtual/ augmented reality application, comprises a receiver (201) for receiving audio data for an audio scene including a first audio component representing a real-world audio source present in an audio environment of a user. A determinator (203) determines a first property of a real-world audio component from the real-world audio source and a target processor (205) determines a target property for a combined audio component being a combination of the real-world audio component received by the user and rendered audio of the first audio component received by the user. An adjuster (207) determines a render property by modifying a property of the first audio component indicated by the audio data for the first audio component in response to the target property and the first property. A renderer (209) renders the first audio component in response to the render property.