Abstract: An adjustment of an audio signal based on an orientation variation amount is carried out according to a surrounding audio condition. An audio signal processing device includes an audio signal analysis unit, an orientation variation analysis unit, and an audio signal adjustment unit. The audio signal analysis unit acquires an audio signal and sets a target value for an audio adjustment on the basis of the audio signal. The orientation variation analysis unit acquires orientation information and generates an orientation variation amount on the basis of the orientation information. The audio signal adjustment unit adjusts the audio signal toward the target value according to the orientation variation amount.

Abstract: Various arrangements for performing dynamic volume control are provided. Audio characteristics of audio content being output to a user may be identified. Adjustments made to an audio volume setting by the user while the audio content is being output to the user can be monitored. A machine learning model can be trained based on the adjustments made to the audio volume setting by the user that are mapped with the audio characteristics of the audio content. After the machine learning model is trained, the audio volume setting can be adjusted based at least in part on the trained machine learning model analyzing audio content.

Abstract: Audio system height channel up-mixing that is configured to develop two or more height channels from audio sources that do not include height-related encoding. The up-mixing involves determining correlations and normalized channel energies between input audio signals. At least two height channels (e.g., left and right height audio signals) are developed from the correlations and normalized energies.

Abstract: To provide an audio reproduction device including: a signal processing circuit that generates multi-channel audio data; an attenuation amount control unit that is connected to the signal processing circuit and controls an attenuation amount of audio data of each channel; and a speaker device that is driven by a multi-channel audio signal from the attenuation amount control unit and has multi-channel speakers including a woofer, in which the attenuation amount control unit estimates, from an audio data value of each channel, a total value of currents flowing through the speaker device, and controls the attenuation amount in such a way that the estimated total value of currents does not exceed a predetermined value that has been set in advance, and the attenuation amount is controlled in accordance with a priority that has been set in advance in such a way that relationships between the channels are maintained.

Abstract: An electronic device and method for audio recommendation and generation are disclosed. The electronic device receives textual information that indicates a plurality of scenes for video content, and determines a first plurality of features for the plurality of scenes. The electronic device determines a set of positions in the textual information based on the determined first plurality of features. A set of audio files are to be inserted at the set of positions related to a set of scenes of the plurality of scenes. The electronic device determines, by an artificial intelligent (AI) engine, the set of audio files for the set of scenes, based on a second plurality of features and the first plurality of features related to the set of scenes. The electronic device controls a display device to display first information corresponding to the set of positions and second information corresponding to the set of audio files.

Abstract: An example involves detecting, during a time period by a receiving device, an audio signal comprising a superposition of a plurality of audio signals transmitted by a plurality of respective playback devices. The example further comprises determining a time-frequency representation of the audio signal and obtaining data associating playback devices with respective reference audio signals, each reference audio signal having a predefined time-frequency representation. The example further comprises, based on the time-frequency representation of the audio signal and the data, determining that the audio signal comprises a superposition of a first audio signal transmitted by a first playback device and a second audio signal transmitted by a second playback device.

Abstract: Described herein is an audio decoder for decoding a bitstream of encoded audio data, wherein the bitstream of encoded audio data represents a sequence of audio sample values and comprises a plurality of frames, wherein each frame comprises associated encoded audio sample values, the audio decoder comprising: a determiner configured to determine whether a frame of the bitstream of encoded audio data is an immediate playout frame comprising encoded audio sample values associated with a current frame and additional information; and an initializer configured to initialize the decoder if the determiner determines that the frame is an immediate playout frame, wherein initializing the decoder comprises decoding the encoded audio sample values comprised by the additional information before decoding the encoded audio sample values associated with the current frame.

Abstract: According to an example embodiment, a method for processing a multi-channel input audio signal representing a sound field into a multi-channel output audio signal representing said sound field in accordance with a predefined loudspeaker layout is provided, the method comprising the following for at least one frequency band: obtaining spatial audio parameters that are descriptive of spatial characteristics of said sound field; estimating a signal energy of the sound field represented by the multi-channel input audio signal; estimating, based on said signal energy and the obtained spatial audio parameters, respective output signal energies for channels of the multi-channel output audio signal according to said predefined loudspeaker layout; determining a maximum output energy as the largest of the output signal energies across channels of said multi-channel output audio signal; and deriving, on basis of said maximum output energy, a gain value for adjusting sound reproduction gain in at least one of said channe

Abstract: An apparatus for audio signal processing audio objects within at least one audio scene, the apparatus comprising at least one processor configured to: define for at least one time period at least one contextual grouping comprising at least two of a plurality of audio objects and at least one further audio object of the plurality of audio objects outside of the at least one contextual grouping, the plurality of audio objects within at least one audio scene; and define with respect to the at least one contextual grouping at least one first parameter and/or parameter rule type which is configured to be applied with respect to a common element associated with the at least two of the plurality of audio objects and wherein the at least one first parameter and/or parameter rule type is configured to be applied with respect to individual element associated with the at least one further audio object outside of the at least one contextual grouping, the at least one first parameter and/or parameter rule type being appli

Abstract: 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.

Abstract: A method for performing voice dictation with an earpiece worn by a user includes receiving as input to the earpiece voice sound information from the user at one or more microphones of the earpiece, receiving as input to the earpiece user control information from one or more sensors within the earpiece independent from the one or more microphones of the earpiece, inserting a machine-generated transcription of the voice sound information from the user into a user input area associated with an application executing on a computing device and manipulating the application executing on the computing device based on the user control information.

Abstract: An example implementation involves a playback device receiving digital data representing audio content, the digital data encoded in a first format. The playback device causes one or more speaker drivers to playback the audio content. The playback device decodes a portion of the received digital data to convert the portion of the received digital data from the first format to a second format and transmits, via a network interface to a computing device of an identification system, the decoded portion of the received digital data. The playback device receives, from via the network interface from the identification system, metadata corresponding to the audio content, and in response, causes a control device to display a graphical representation of the received metadata, wherein causing the control device to display the graphical representation comprises sending, via the network interface to the control device, the received metadata to the control device.

Abstract: An earbud includes a housing that includes a driver assembly positioned within the housing forming a front volume in front of the driver and a back volume behind the driver. An acoustic insert is positioned behind the driver assembly and attached to an interior surface of the housing such that it forms a bass channel that is routed from the back volume to a vent in the housing.

Abstract: A noise attenuating device for reducing acoustic resonance in a cavity of an aircraft with an opening in a skin of the aircraft defining a first surface area and an opening peripheral edge includes a connection mechanism for securing the noise attenuating device in proximity to the opening in the skin of the aircraft and an exterior surface. The exterior surface defines a second surface area that is less than the first surface area of the opening and a device peripheral edge, which has a first section having an edge profile complimentary to at least a portion of the opening peripheral edge and a second section having an indented edge profile in relation to the first section such that when the noise attenuating device is secured within the opening, at least one gap is formed between the device peripheral edge and the opening peripheral edge.

Abstract: The present disclosure provides systems and methods for applying a volume gain to change the playback volume of a device by calculating a smoothed noise estimate. The smoothed noise estimate may calculated using an exponential smoothing method. The exponential smoothing method may include applying a weight factor to both a background noise estimate and a historical smoothed noise estimate. The weight factor may place more emphasis on either the background noise estimate or the historical smoothed noise estimate depending on the value of the weight factor. The value of the weight factor may change based on the type of environment the device is in, the speed of the background noise transitions, and/or historical data. The calculated smoothed noise estimate may be used to calculate a volume gain. The volume gain may be applied over a period of time such that the playback volume of the device changes smoothly.

Abstract: An apparatus for generating a sound field description using an input signal having a mono-signal or a multi-channel signal comprises: an input signal analyzer for analyzing the input signal to derive direction data and diffuseness data; a low-order components generator for generating a low-order sound field description from the input signal up to a predetermined order and mode; a mid-order components generator for generating a mid-order sound field description above the predetermined order or at the predetermined order and above the predetermined mode and below or at a first truncation order using a synthesis of at least one direct portion and of at least one diffuse portion using the direction data and the diffuseness data; and a high-order components generator for generating a high-order sound field description having a component above the first truncation order using a synthesis of at least one direct portion.

Abstract: An automated artificial intelligence (AI) and template-based audio record mixing system and process are disclosed. The automated AI and template-based audio record mixing system and process use provided templates and AI to mix records. The artist maintains full control of how they want to record sound when utilizing the automated AI and template-based audio record mixing system and process.

Abstract: The present disclosure relates to the field audio coding, an in particular to an audio decoder having at least two decoding modes, and associated decoding methods and decoding software for such audio decoder. In one of the decoding modes, at least one dynamic audio object is mapped to a set of static audio objects, the set of static audio objects corresponding to a predefined speaker configuration. The present disclosure further relates to a corresponding audio encoder, and associated encoding methods and encoding software for such audio encoder.

Abstract: An encoder includes a signal detection circuit and a signal encoding circuit. The signal encoding circuit is configured to encode the Nth-frame downmixed signal when the signal detection circuit detects that an Nth-frame downmixed signal includes a speech signal, or when the signal detection circuit detects that the Nth-frame downmixed signal does not include a speech signal, encode the Nth-frame downmixed signal when the signal detection circuit determines that the Nth-frame downmixed signal satisfies a preset audio frame encoding condition, or skip encoding the Nth-frame downmixed signal when the signal detection circuit determines that the Nth-frame downmixed signal does not satisfy a preset audio frame encoding condition.

Abstract: Provided are a device control method and apparatus. The method is applied to an audio device, and includes: receiving an acoustic signal set, determining a propagation characteristic of an acoustic signal in the acoustic signal set, determining, according to the propagation characteristic, a device parameter associated with audio play quality to be used by the audio device, and controlling the audio device to play audio with the device parameter.