Abstract: A method, apparatus and computer program, the method comprising: receiving a plurality of input signals representing a sound space; using the received plurality of input signals to obtain spatial metadata corresponding to the sound space; using the received plurality of input signals to obtain a first spatial audio signal corresponding to the spatial metadata; and associating the first spatial audio signal with the spatial metadata to enable the spatial metadata to be used to process the first spatial audio signal to obtain a second spatial audio signal.

Abstract: An apparatus including circuitry configured for: obtaining media content, wherein the media content includes at least one object data; obtaining priority content information, the priority content information including a priority identification identifying and classifying the at least one object; rendering the at least one object based on the priority content information.

Abstract: An audio processing chip includes a detector circuit, a first-in first-out (FIFO) circuit, and an adjustment circuitry. The detector circuit is configured to detect an audio stream to output an enable signal. The FIFO circuit is configured to start outputting audio data corresponding to a channel in the audio stream to be a first signal. The adjustment circuitry is configured to process the first signal, in order to generate an output signal and transmit the output signal to a speaker.

Abstract: A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

Abstract: A method for generating loudspeaker signals associated with a target screen size is disclosed. The method includes receiving a bit stream containing encoded higher order ambisonics signals, the encoded higher order ambisonics signals describing a sound field associated with a production screen size. The method further includes decoding the encoded higher order ambisonics signals to obtain a first set of decoded higher order ambisonics signals representing dominant components of the sound field and a second set of decoded higher order ambisonics signals representing ambient components of the sound field. The method also includes combining the first set of decoded higher order ambisonics signals and the second set of decoded higher order ambisonics signals to produce a combined set of decoded higher order ambisonics signals.

Abstract: There are two representations for Higher Order Ambisonics denoted HOA: spatial domain and coefficient domain. The invention generates from a coefficient domain representation a mixed spatial/coefficient domain representation, wherein the number of said HOA signals can be variable. An aspect of the invention further relates to methods and apparatus decoding multiplexed and perceptually encoded HOA signals, including transforming a vector of PCM encoded spatial domain signals of the HOA representation to a corresponding vector of coefficient domain signals by multiplying the vector of PCM encoded spatial domain signals with a transform matrix and de-normalizing the vector of PCM encoded and normalized coefficient domain signals, wherein said de-normalizing comprises.

Abstract: A system for and a method of generating an audio image for use in rendering audio. The method comprises accessing an audio stream; accessing positional information, the positional information comprising a first position, a second position and a third position; and generating an audio image. In some embodiments, generating the audio image comprises generating, based on the audio stream, a first virtual wave front to be perceived by a listener as emanating from the first position; generating, based on the audio stream, a second virtual wave front to be perceived by the listener as emanating from the second position; and generating, based on the audio stream, a third virtual wave front to be perceived by the listener as emanating from the third position.

Abstract: Systems and methods are disclosed for audio group identification for conferencing. For example, methods may include joining a conference call using a network interface; accessing an audio signal that has been captured using a microphone; detecting a control signal in the audio signal; and, responsive to detection of the control signal, invoking modification of an audio path of the conference call.

Abstract: A method, system, and computer program to encode and decode a channel coherence parameter applied on a frequency band basis, where the coherence parameters of each frequency band form a coherence vector. The coherence vector is encoded and decoded using a predictive scheme followed by a variable bit rate entropy coding.

Abstract: An audio processor for processing an audio signal includes: an audio signal modifier for modifying the audio signal in response to a user input; a loudness controller for determining a loudness compensation gain based on a reference loudness or a reference gain and a modified loudness or a modified gain, where the modified loudness or the modified gain depends on the user input; and a loudness manipulator for manipulating a loudness of a signal using the loudness compensation gain.

Abstract: An apparatus comprising means for: in response to user input, selecting at least one sound source of a spatial audio scene, comprising multiple sound sources, the spatial audio scene being defined by spatial audio content; selecting at least one related contextual sound source based on the at least one selected sound source; and causing rendering of an audio preview, representing the spatial audio content, that can be selected by a user, wherein the audio preview comprises a mix of sound sources including at least the at least one selected sound source and the at least one related contextual sound source but not all of the multiple sound sources of the spatial audio scene, and wherein selection of the audio preview causes an operation on at least the selected sound source.

Abstract: An audio scene encoder for encoding an audio scene, the audio scene having at least two component signals, has: a core encoder for core encoding the at least two component signals, wherein the core encoder is configured to generate a first encoded representation for a first portion of the at least two component signals, and to generate a second encoded representation for a second portion of the at least two component signals, a spatial analyzer for analyzing the audio scene to derive one or more spatial parameters or one or more spatial parameter sets for the second portion; and an output interface for forming the encoded audio scene signal, the encoded audio scene signal having the first encoded representation, the second encoded representation, and the one or more spatial parameters or one or more spatial parameter sets for the second portion.

Abstract: The present invention relates to a method and an apparatus for processing an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesize an object signal and a channel signal and effectively perform binaural rendering of the synthesized signal.

Abstract: Disclosed is an electronic device including a speaker, a communication circuit, a processor, and a memory. The processor is configured to receive first data including a first audio frame corresponding to a first interval and a second audio frame corresponding to a second interval subsequent to the first interval, using the communication circuit, to store the second audio frame in the memory in response to reception of the first data, to output a first audio signal generated based on the first audio frame, through the speaker, and to store the third audio frame in the memory and output a second audio signal generated based on the second audio frame of the second data through the speaker when second data including the second audio frame and a third audio frame corresponding to a third interval subsequent to the second interval is received using the communication circuit.

Abstract: There are two representations for Higher Order Ambisonics denoted HOA: spatial domain and coefficient domain. The invention generates from a coefficient domain representation a mixed spatial/coefficient domain representation, wherein the number of said HOA signals can be variable. An aspect of the invention further relates to methods and apparatus decoding multiplexed and perceptually encoded HOA signals, including transforming a vector of PCM encoded spatial domain signals of the HOA representation to a corresponding vector of coefficient domain signals by multiplying the vector of PCM encoded spatial domain signals with a transform matrix and de-normalizing the vector of PCM encoded and normalized coefficient domain signals, wherein said de-normalizing comprises.

Abstract: An encoder for encoding secondary media data including metadata and control data for primary media data is shown, wherein the encoder is configured to encode the secondary media data using adding redundancy or bandlimiting and wherein the encoder is configured to output the encoded secondary media data as a stream of digital words. Therefore, the stream of digital words may be formed such that it is capable to resist a typical processing of a digital audio stream. Furthermore, processors for processing a digital audio stream are able to process the stream of digital words, since the stream of digital words may be designed as an audio-like or analog-like digital stream.

Abstract: An audio processing apparatus comprises a receiver (705) which receives audio data including audio components and render configuration data including audio transducer position data for a set of audio transducers (703). A renderer (707) generating audio transducer signals for the set of audio transducers from the audio data. The renderer (7010) is capable of rendering audio components in accordance with a plurality of rendering modes. A render controller (709) selects the rendering modes for the renderer (707) from the plurality of rendering modes based on the audio transducer position data. The renderer (707) can employ different rendering modes for different subsets of the set of audio transducers the render controller (709) can independently select rendering modes for each of the different subsets of the set of audio transducers (703).

Abstract: A system and method for transcoding and mixing of video data in a security video distribution system for a video security system in response to selection of video data. User devices such as mobile user devices can select displayed video data streams sent from the video security system and create new views of the selected video data streams without the prior steps of configuring and pushing new views from a server within the system. At the time of selection, user devices can specify operations to perform on the selected video data, such as zoom and combine. In a preferred embodiment, transcoding and mixing of video data streams within the security video distribution system utilizes shared memory between processes. This has advantages not only in equipment cost savings and efficiency, but also provides for real-time collaboration and sharing of video content in a video security system using mobile user devices.

Abstract: An apparatus comprising at least one processor and at least one memory including a computer program code, the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to: receive at least one audio signal; obtain, associated with the at least one audio signal over at least one frequency band: at least one spatial audio energy ratio parameter; and at least one remainder energy ratio, wherein a sum of the at least one spatial audio energy ratio parameter and the at least one remainder energy ratio over the frequency band equal a determined value; and control a transmission/storage of the at least one spatial audio energy ratio, and the at least one remainder energy ratio.

Abstract: The present invention relates to methods and apparatus for encoding an HOA signal representation (c(t)) of a sound field having an order of N and a number O=(N+1)2 of coefficient sequences to a mezzanine HOA signal representation (wMEZZ(t)). The present invention further relates to methods and apparatus for decoding a reconstructed HOA signal representation from the mezzanine HOA signal representation.

Abstract: An apparatus comprising a recording medium including a wide sound composition and a method for producing said wide sound composition. The wide sound composition comprises a full composition or a partial composition recorded in combination with a delayed full composition or a partial composition. The method for producing said wide sound composition comprises at least an audio reproducing mechanism, an audio recording device and a recording medium for the storage of a wide sound composition, wherein said audio reproducing mechanism comprises at least a first and second electroacoustic device and a source music and, wherein said source music comprises at least a full composition or a partial composition.

Abstract: The present invention relates to a method and an apparatus for processing an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesize an object signal and a channel signal and effectively perform binaural rendering of the synthesized signal.

Abstract: In one embodiment, a method in a user equipment (UE) includes receiving broadcasted system information from a network node. The broadcasted system information may include explicit configuration information for a communications channel. The method further includes determining a configuration of the communications channel based, at least in part, on the explicit configuration information. The explicit configuration information may be used by the user equipment to override default channel configuration information. The method may further include receiving data over the communications channel.

Abstract: An exemplary audio processing system accesses audio stream data representative of a plurality of virtual sounds that virtually originate within an extended reality world and include real-time captured speech originating from a speaking user experiencing the extended reality world and a prerecorded or synthesized sound originating from a real audio source. The audio processing system also accesses acoustic propagation data representative of characteristics that affect propagation within the extended reality world of the virtual sounds to a listening avatar. As the virtual sounds propagate to the listening avatar, the audio processing system renders, based on the audio stream data and the acoustic propagation data, a composite binaural audio stream that represents simulated reproductions of the virtual sounds. The simulated reproductions of the virtual sounds are customized to account for the characteristics that affect the propagation of the virtual sounds to the listening avatar.

Abstract: An apparatus for downmixing three or more audio input channels to obtain two or more audio output channels is provided. The apparatus includes a receiving interface for receiving the three or more audio input channels and for receiving side information. Moreover, the apparatus includes a downmixer for downmixing the three or more audio input channels depending on the side information to obtain the two or more audio output channels. The number of the audio output channels is smaller than the number of the audio input channels. The side information indicates a characteristic of at least one of the three or more audio input channels, or a characteristic of one or more sound waves recorded within the one or more audio input channels, or a characteristic of one or more sound sources which emitted one or more sound waves recorded within the one or more audio input channels.

Abstract: 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.

Abstract: A method of encoding channel or object based input audio for playback, the method including the steps of: (a) initially rendering the channel or object based input audio into an initial output presentation; (b) determining an estimate of the dominant audio component from the channel or object based input audio and determining a series of dominant audio component weighting factors for mapping the initial output presentation into the dominant audio component; (c) determining an estimate of the dominant audio component direction or position; and (d) encoding the initial output presentation, the dominant audio component weighting factors, the dominant audio component direction or position as the encoded signal for playback.

Abstract: A decoder for generating an audio output signal having one or more audio output channels is provided. The decoder includes a receiving interface for receiving an audio input signal including a plurality of audio object signals, for receiving loudness information on the audio object signals, and for receiving rendering information indicating whether one or more of the audio object signals shall be amplified or attenuated. Moreover, the decoder includes a signal processor for generating the one or more audio output channels of the audio output signal. The signal processor is configured to determine a loudness compensation value depending on the loudness information and depending on the rendering information. Furthermore, the signal processor is configured to generate the one or more audio output channels of the audio output signal from the audio input signal depending on the rendering information and depending on the loudness compensation value. Moreover, an encoder is provided.

Abstract: A method comprising: causing display of a sound-source virtual visual object in a three-dimensional virtual visual space; causing display of a multiplicity of interconnecting virtual visual objects in the three-dimensional virtual visual space, wherein at least some of the multiplicity of interconnecting virtual visual objects interconnect visually a sound-source virtual visual object and a user-controlled virtual visual object, wherein a visual appearance of each interconnecting virtual visual object, is dependent up on one or more characteristics of a sound object associated with the sound-source virtual visual object to which the interconnecting virtual visual object is interconnected, and wherein audio processing of the sound objects to produce rendered sound objects depends on user-interaction with the user-controlled virtual visual object and user-controlled interconnection of interconnecting virtual visual objects between sound-source virtual visual objects and the user-controlled virtual visual object

Abstract: Systems and methods for limiting volume in an audio playback device using a feedback controller are disclosed herein. In one example, a gain stage modulates gain of an audio signal based in part on feedback from a downstream limiter. The gain stage receives a first audio signal as well as a feedback signal from the feedback controller. Based at least in part on the feedback signal from the feedback controller, the gain stage modulates a gain of the first audio signal to provide a second audio signal. The second audio signal is delivered to the limiter, which limits the second audio signal to produce an output signal. The output signal is played back via a transducer. The feedback controller receives a gain reduction value from the limiter and determines a feedback signal to provide to the gain stage upstream of the limiter.

Abstract: A schematic block diagram of an audio encoder for encoding a multichannel audio signal is shown. The audio encoder includes a linear prediction domain encoder, a frequency domain encoder, and a controller for switching between the linear prediction domain encoder and the frequency domain encoder. The controller is configured such that a portion of the multichannel signal is represented either by an encoded frame of the linear prediction domain encoder or by an encoded frame of the frequency domain encoder. The linear prediction domain encoder includes a downmixer for downmixing the multichannel signal to obtain a downmixed signal. The linear prediction domain encoder further includes a linear prediction domain core encoder for encoding the downmix signal and furthermore, the linear prediction domain encoder includes a first joint multichannel encoder for generating first multichannel information from the multichannel signal.

Abstract: A stereo parameter conditioner performs a conditioning operation on a first value of a stereo parameter and a second value of the stereo parameter to generate a conditioned value of the stereo parameter. The first value is associated with a first frequency range, and the second value is associated with a second frequency range. The conditioned value is associated with a particular frequency range that is a subset of the first frequency range or a subset of the second frequency range.

Abstract: A method is provided for exchanging data between a user device and an audio output device over a wireless link. A user device detects that a trigger event for initializing a low latency mode has occurred at the user device, the low latency mode configured to cause the audio output device to reduce a size of a render buffer to be used for rendering audio data. In response to the detecting, the user device transmits an indication to the audio output device to initialize the low latency mode, the indication causing the audio output device to reduce the size of the render buffer to be used for rendering the audio data. The user device then transmits the audio data to the audio output device over the wireless link.

Abstract: A method implemented by a processing device for processing an ambisonic content including a plurality of ambisonic components of a plurality of orders defining a succession of ambisonic channels in each of which an ambisonic component is represented. The method includes: frequency filtering of the ambisonic components in a plurality of frequency bands, compiling an ambisonic decoding matrix, processing the ambisonic decoding matrix in order to extract, by matrix dimension reduction, a plurality of ambisonic decoding sub-matrices each associated with an ambisonic order and a frequency band selected for this ambisonic order, respective applications of the decoding sub-matrices to the ambisonic components in each selected frequency band, and a reconstruction, band by band, of the results of said respective applications, in order to deliver a plurality of decoded signals, each associated with a sound source.

Abstract: A decoding device includes: a separating unit separating first encoded data, a spectrum including a low-band spectrum of audio signals having been encoded, and second encoded data, a high-band spectrum of a higher band having been encoded, based on the first encoded data; a first decoding unit decoding the first encoded data and generating a first decoded spectrum; a first amplitude normalizer dividing amplitude of the first decoded spectrum into sub-bands, normalizing the spectrum of each sub-band by the largest amplitude of the first decoded spectrum within each sub-band, and generating a normalized spectrum; an addition unit adding noise spectrum to the normalized spectrum and generating a noise-added normalized spectrum; a second decoding unit decoding the second encoded data using the noise-added normalized spectrum, and generating a second noise-added spectrum; and a converter performing time-frequency conversion regarding a spectrum coupled based on the first decoded spectrum and second noise-added spe

Abstract: An example audio decoding device includes processing circuitry and a memory device coupled to the processing circuitry. The processing circuitry is configured to receive, in a bitstream, encoded representations of audio objects of a three-dimensional (3D) soundfield, to receive metadata associated with the bitstream, to obtain, from the received metadata, one or more transmission factors associated with one or more of the audio objects, and to apply the transmission factors to the one or more audio objects to obtain parallax-adjusted audio objects of the 3D soundfield. The memory device is configured to store at least a portion of the received bitstream, the received metadata, or the parallax-adjusted audio objects of the 3D soundfield.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: Disclosed are a multi-channel audio signal processing method and a multi-channel audio signal processing apparatus. The multi-channel audio signal processing method may generate N channel output signals from N/2 channel downmix signals based on an N-N/2-N structure.

Abstract: The present invention is directed to apparatus and methods for decoding Higher Order Ambisonics (HOA) audio signals. HOA audio signals may be decompressed based on perceptual decoding to determine at least an HOA representation corresponding to the HOA audio signals. A rotated transform may be determined based on a rotation of a spherical sample grid. A rotated HOA representation may be determined based on the rotated transform and the HOA representation. The rotated HOA representation may be rendered to output to a loudspeaker setup.

Abstract: Improved tools for authoring and rendering audio reproduction data are provided. Some such authoring tools allow audio reproduction data to be generalized for a wide variety of reproduction environments. Audio reproduction data may be authored by creating metadata for audio objects. The metadata may be created with reference to speaker zones. During the rendering process, the audio reproduction data may be reproduced according to the reproduction speaker layout of a particular reproduction environment.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding a stereo audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and a discrete representation of the stereo audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.

Abstract: An apparatus for decoding a surround audio signal includes a Bitstream De-multiplexer for unpacking a bitstream into spatial parameters and core parameters, a set of core decoders for decoding the core parameters into a set of core signals, a matrix derivation unit for deriving the rendering matrix from the spatial parameters and playback speaker layout information, and a renderer for rendering of the decoded core signals to playback signals using the rendering matrix.

Abstract: An encoder and an encoding method for a multi-channel signal, and a decoder and a decoding method for a multi-channel signal are disclosed. A multi-channel signal may be efficiently processed by consecutive downmixing or upmixing.

Abstract: A method for parametric encoding of a multi-channel digital audio signal. The method includes encoding a mono signal from channel-reduction processing applied to the multi-channel signal and encoding spatialisation information of the multi-channel signal. The channel-reduction processing includes the following steps, implemented for each spectral unit of the multi-channel signal: extracting at least one indicator characterizing the channels of the multi-channel digital audio signal; selecting, from a set of channel-reduction processing modes, a channel-reduction processing mode in accordance with the value of the at least one indicator characterizing the channels of the multi-channel audio signal. Also provides are a corresponding encoding device and a processing method which includes the channel-reduction processing.

Abstract: The present technology relates to an audio processing apparatus capable of downmixing 7.1-ch audio data to 2-ch audio data. A coefficient for downmixing 7.1-ch audio data to 2-ch audio data is set from a coefficient for downmixing 7.1-ch audio data to 5.1-ch audio data specified by a Moving Picture Experts Group 4 (MPEG4) audio standard and a coefficient for downmixing 5.1-ch audio data to 2-ch audio data specified by the standard, and stored in a 2-ch downmixing coefficient unit 22. A 2-ch downmixing unit 21 downmixes 7.1-ch audio data to 2-ch audio data using a coefficient stored in the 2-ch downmixing coefficient unit 22. The present technology can be applied to an audio processing apparatus.

Abstract: An encoding method for a multi-channel audio signal, an encoding apparatus for performing the encoding method, and a decoding method for a multi-channel audio signal and a decoding apparatus for performing the decoding method are disclosed. A method and apparatus of bypassing an MPEG Surround (MPS) standard operation and using an arbitrary tree when a number of audio signals of N channels exceeds a channel number defined in an MPS standard, is disclosed.

Abstract: An apparatus for processing an encoded audio signal having a plurality of downmix signals associated with a plurality of input audio objects and object parameters E. The apparatus includes a grouper configured to group the downmix signals into groups of downmix signals associated with a set of input audio objects. The apparatus includes a processor configured to perform at least one processing step individually on the object parameters Ek of each set of input audio objects in order to provide group results. Further, there is a combiner configured to combine the group results or processed group results in order to provide a decoded audio signal. The grouper is configured to group the downmix signals so that each input audio object belongs to just one set of input audio objects. The invention also refers to a corresponding method.

Abstract: A method and system are implemented in a stereo sound signal encoding system for time domain down mixing right and left channels of an input stereo sound signal into primary and secondary channels. Correlation of the primary and secondary channels of previous frames is determined, and an out-of-phase condition of the left and right channels is detected based on the correlation of the primary and secondary channels of the previous frames. The left and right channels are time domain down mixed, as a function of the detection, to produce the primary and secondary channels using a factor ?, wherein the factor ? determines respective contributions of the left and right channels upon production of the primary and secondary channels.

Abstract: A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

Abstract: A method is described which decodes a downmix matrix for mapping a plurality of input channels of audio content to a plurality of output channels, the input and output channels being associated with respective speakers at predetermined positions relative to a listener position, wherein the downmix matrix is encoded by exploiting the symmetry of speaker pairs of the plurality of input channels and the symmetry of speaker pairs of the plurality of output channels. Encoded information representing the encoded downmix matrix is received and decoded for obtaining the decoded downmix matrix.