Abstract: A system comprises an auditory device processor, an auditory device output mechanism, an auditory input sensor, a database including a reference bank of environmental sounds and corresponding sound profiles, and a memory.

Abstract: A computing device capable of controlling a networked media playback system receives data identifying a first zone player, a second zone player, and a zone scene including a predefined grouping of zone players including at least the first zone player and the second zone player. The computing device displays a representation of available zone players in the networked media playback system including a first selectable indication of the first zone player, a second selectable indication of the second zone player, and a third selectable indication of the zone scene that, when selected, causes the zone scene to be invoked such that the predefined grouping of zone players becomes configured for synchronous playback of media. While displaying the representation, the computing device receives a user input indicating that the zone scene has been selected to be invoked. Based on the user input, the computing device causes the zone scene to be invoked.

Abstract: The present application describes platform containing a dynamic multilayered media structure which is generated by aggregation of media pieces into a plurality of media layers. The platform allows users to interact with media layers of the dynamic multilayered media structure independently of one another. The platform further provides for a dynamic, customized media channel lineup and a user account that allows individual media pieces within the dynamic multilayered media structure to play across separate devices that are linked to the account. The user account allows for inputs including real-time controls over the media, interaction, preferences, adding additional media content, editing media into a condensed form, and curation options with the dynamic multilayered media structure.

Abstract: The present application describes platform containing a dynamic multilayered media structure which is generated by aggregation of media pieces into a plurality of media layers. The platform allows users to interact with media layers of the dynamic multilayered media structure independently of one another. The platform further provides for a dynamic, customized media channel lineup and a user account that allows individual media pieces within the dynamic multilayered media structure to play across separate devices that are linked to the account. The user account allows for inputs including real-time controls over the media, interaction, preferences, adding additional media content, editing media into a condensed form, and curation options with the dynamic multilayered media structure.

Abstract: A multi-channel audio signal coding method includes obtaining a to-be-encoded first audio frame, pairing at least five channel signals according to a first pairing manner to obtain a first channel pair set, obtaining a first sum of correlation values of the first channel pair set, where one channel pair has one correlation value, pairing the at least five channel signals according to a second pairing manner to obtain a second channel pair set, obtaining a second sum of correlation values of the second channel pair set, determining a target pairing manner of the at least five channel signals based on the first sum of correlation values and the second sum of correlation values, and encoding the at least five channel signals based on a channel pair set corresponding to the target pairing manner, where the target pairing manner is the first pairing manner or the second pairing manner.

Abstract: A method for loudness equalization is provided that includes receiving input loudness data at an audio processing system. Converting gain data of the input loudness data to a linear scale at the audio processing system. Determining a reciprocal of a gain-linear loudness value as a function of the converted gain data using the audio processing system. Determining a compression ratio using the audio processing system. Performing temporal smoothing and look ahead processing using the audio processing system. Outputting gain data as a function of the temporal smoothing and look ahead processing using the audio processing system.

Abstract: An audio device being a mobile terminal includes: a playback unit configured to play back music data; a display configured to display an operation element for accepting a user operation related to playback setting with respect to the music data; a touch panel provided on a surface of the display and configured to accept the user operation; and a playback controller configured to decide a kind of the playback setting related to a playback position in the music data, which is based on a timing at which a contact is started or based on a timing at which the contact is finished with the touch panel corresponding to the operation element displayed on the display, based on a direction of a change in the contact position on the touch panel from the timing at which the contact is started to the timing at which the contact is finished.

Abstract: There is provided encoding and decoding methods for representing spatial audio that is a combination of directional sound and diffuse sound. An exemplary encoding method includes inter alia creating a single- or multi-channel downmix audio signal by downmixing input audio signals from a plurality of microphones in an audio capture unit capturing the spatial audio; determining first metadata parameters associated with the downmix audio signal, wherein the first metadata parameters are indicative of one or more of: a relative time delay value, a gain value, and a phase value associated with each input audio signal; and combining the created downmix audio signal and the first metadata parameters into a representation of the spatial audio.

Abstract: A microphone having a multi-channel mixer and one or more connectors usable to receive and/or output audio. Any of the connectors may be used as an input connector, as an output connector, or configured to be switchable between being an input and an output connector. The user of the microphone may be able to conveniently use one or more of the connectors to expand the microphone to become part of a larger setup that uses multiple microphones. For example, an XLR connector of the microphone may be passive, and may be configured such that a user can daisy chain the output from an XLR connector of another microphone into an XLR connector of the microphone. In such an arrangement, an output based on one or both of the microphones may be output through another connector of the microphone, such as a USB connector.

Abstract: An apparatus or method to allow a user to control an audio processing operation of an internal and/or external microphone(s). The method includes providing a configurable user interface which enables mixing of audio signals through interaction with the user interface. The audio signals may be captured with internal or external microphones.

Abstract: Examples of the present disclosure relate to a method, computer-readable medium, and system for vehicle assistive hearing. An example may include transmitting an assistive listening device pairing signal into a vehicle. The method may further include identifying a plurality of digital audio sources in the vehicle, wherein at least one of the audio sources is a vehicle alert system for the vehicle. In an example, the method may categorize the priority of the plurality of digital audio sources. In an example, the method may generate a stream of audio from the plurality of digital audio sources with volume adjusted for each audio source according to the priority of the digital audio source. The method may transmit the stream of audio for a paired assistive listening device.

Abstract: Encoding/decoding an audio signal having one or more audio components, wherein each audio component is associated with a spatial location. A first audio signal presentation (z) of the audio components, a first set of transform parameters (w(f)), and signal level data (?2) are encoded and transmitted to the decoder. The decoder uses the first set of transform parameters (w(f)) to form a reconstructed simulation input signal intended for an acoustic environment simulation, and applies a signal level modification (?) to the reconstructed simulation input signal. The signal level modification is based on the signal level data (?2) and data (p2) related to the acoustic environment simulation. The attenuated reconstructed simulation input signal is then processed in an acoustic environment simulator. With this process, the decoder does not need to determine the signal level of the simulation input signal, thereby reducing processing load.

Abstract: A signal processing apparatus is provided that includes a sound source separation section configured to apply sound source separation processing to a mixed sound signal including a mixture of signals of a plurality of sound sources, and band extension sections configured to apply frequency band extension processing to respective sound source separation signals obtained by separation by the sound source separation section.

Abstract: A frequency-dependent signal limiter using a paired combination of a bandpass filter (or “selection filter”) and an attenuating equalizing filter (or “cutting filter”) limit the amount of distortion generated by an audio output device in an acoustic rendering system. A bank of one or more of these pairs in series can be used to limit selected frequencies identified as primary causes of distortion in the audio output device to specified levels that avoid distortion.

Abstract: A control device is configured to (i) define configuration data corresponding to a zone scene of a media playback system, the configuration data including an identification of at least a first zone player and a second zone player, (ii) cause a remote computing device to (a) receive the configuration data via a wide area network and (b) store the configuration data, (iii) while the first zone player and a third zone player are playing back first audio content in synchrony, display a selectable indication that, when selected, causes the zone scene to be invoked, and (iv) in response to receiving an input indicative of a selection of the selectable indication, cause the remote computing device to invoke the zone scene at the media playback system such that the first zone player and the second zone player play back second audio content in synchrony without the third zone player.

Abstract: An electronic device comprising a circuitry configured to perform (402; 602; 902; 1002) blind source separation (201) based on a received input signal (1) to obtain separated sources; to perform (403; 605; 906; 1005) time-domain filtering (202) on at least one of the separated sources to obtain at least one filtered separated source; and to perform (404; 606; 907; 1006) remixing or upmixing (203) based on the at least one filtered separated source.

Abstract: A multi-sensory spatial audio experience system transforms one or more features of audio data to visualizations and coordinates for presentation of the visualizations with auditory cues of the audio data to create a combination of auditory-visual cues for audio spatialization. The multi-sensory spatial audio experience system presents a graphical depiction of audio source spatial positions with respect to a user depiction via a user interface. The multi-sensory spatial audio experience system determines a spatial feature(s) of input audio data and transforms the audio data spatial feature into visual feature data. In coordination with rendering of the audio data, the multi-sensory spatial audio experience system updates the user interface according to the visual feature data. A user can visually perceive graphical changes in the audio source spatial positions that reinforce the auditory perception of the corresponding audio data.

Abstract: A controlling method including controlling signal processing of the channel assigned to each of the plurality of first physical controllers, based on a parameter indicated by each of the plurality of first physical controllers, enabling a parameter change operation by the plurality of first physical controllers when a channel group set to be enabled by the third physical controller is designated, and disabling the parameter change operation by the plurality of first physical controllers when a channel group set to be disabled by the third physical controller is designated.

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: Distributed audio mixing may include transmitting a set of parameters from a local location to one or more remote locations at least multiple miles away from the local location for, at each of the one or more remote locations, one or more remote audio sources to be processed according to the parameters to produce respective one or more remote audio mixes; processing one or more local audio sources according to the parameters to produce a local audio mix; receiving the one or more remote audio mixes; and locally summing the one or more remote audio mixes to the local audio mix to obtain a final audio mix.

Abstract: A method of generating a virtual reality (“VR”) experience includes detecting, using external sensors, a real-world spectator that is in a spectator space that is adjacent to a user that is wearing a VR headset and experiencing a VR space. The method also includes detecting a real-world physical state of the spectator, and rendering, in response to detecting, the spectator in the VR space as an avatar that reflects the real-world physical state of the spectator.

Abstract: Methods and systems are provided for independent audio volume control. User input may be received in an audio setup configured for combined-audio, with the user input including one or both of a selection of a desired volume of a first type audio component of the combined-audio signal, and a selection of a desired volume of a second type audio component of the combined-audio signal. Based on the user input, a corresponding mix setting may be determined, with the mix setting including one or more control parameters for adaptively controlling mixing of the first type audio component and the second type audio component to generate the combined-audio signal. The mix setting may be determined based on a data set characterizing an audio device where at least portion of the mixing is performed. The data set may be pre-programmed and/or determined based on testing of the audio device.

Abstract: A display method includes displaying a list of a plurality of groups in a first area which is a first partial area of a display, and a name of a component belonging to at least one of the plurality of groups in a second area which is a second partial area of the display and is different from the first area. In the displaying, switching a display mode between a first display mode and a second display mode is performed in the second area. The first display mode is a mode in which all of the plurality of groups are displayed and a group, out of the plurality of groups, to which the component belongs is displayed in a display appearance different from that of a different group of the plurality of groups. The second display mode is a mode in which, out of the plurality of groups, only the group to which the component belongs is displayed.

Abstract: An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, includes: a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block; a spectral characteristic estimator for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block; a smoothing filter for smoothing the cross-correlation spectrum over time using the spectral characteristic to obtain a smoothed cross-correlation spectrum; and a processor for processing the smoothed cross-correlation spectrum to obtain the inter-channel time difference.

Abstract: A method of managing at least one audio playback equipment group including at least one piece of audio playback equipment is provided with a light-emitting identification interface. The method includes the steps of: using the light-emitting identification interface of the piece of audio playback equipment to display a light pattern characteristic at least of the group to which said piece of equipment belongs and/or characteristic of the role of the piece of equipment within said group in the event of at least one of the pieces of audio playback equipment being selected, and performing an action relating to the associated piece of audio playback equipment.

Abstract: An operation panel for an audio processing apparatus includes a plurality of components including a user interface that selects at least one output channel among a plurality of output channels and presents a first processing parameter, which indicates a send level from an input channel to the selected at least one output channel, and a second processing parameter different from the first processing parameter. The audio processing apparatus further includes a link receiver that receives, for at least two components, among the plurality of components, for setting a first link mode and a second link mode independently the first link mode for linking the presented first processing parameter, and the second link mode for linking the presented second processing parameter.

Abstract: The present invention relates generally to the field of digital content, and more particularly to an improved system and methods for producing, mixing, and recording content. In particular, the system may include a plurality of input channels to which one or more audio sources may be linked via a user interface. Each audio source may be assigned to a physical fader or a virtual fader for adjusting a level of an audio signal received from the audio source. Advantageously, the system may connect via a wired and/or wireless connection with a number of audio devices and may further facilitate assigning each audio source to an input channel such that content creators, such as podcasters, are able to produce, mix, and record audio contents more efficiently and effectively.

Abstract: The present invention provides a signal processing device that allows a user, who adjusts a signal level to match a reference level, to adjust the signal level while checking a degree of difference between the signal level and the reference level. A signal processing device according to the present invention comprises: an input part 20 to which an input signal is input; a storage 40 that stores therein one or more setting values of a parameter to be used for amplifying the input signal and a reference value of the setting value; an amplification part 30 that generates an output signal by amplifying a level of the input signal based on the setting value; an output part 70 that outputs the output signal; an adjustment part 50 that adjusts the setting value; and a display 60 that displays a display result determined based on the setting value adjusted by the adjustment part and the reference value.

Abstract: A method for configuring a modular safety switching apparatus having electronic modules comprises: a) selecting a configuration data set of the apparatus from a database and transmitting the configuration data set to a computing apparatus, wherein the configuration data set is processed by a computer program to generate a target configuration of the apparatus and the electronic modules; b) displaying the target configuration; c) plugging one of the electronic modules into one of the module slots of the module row; d) setting a rotary switch position of one or more rotary switches of the electronic module; e) determining an actual configuration of the electronic module configured according to c) and d); f) comparing the actual configuration with the target configuration; g) displaying whether a deviation exists between the actual configuration and the target configuration; and h) repeating c) to g) for each additional electronic module.

Abstract: An output buffer set is coupled to a first pair of pins and configured to output art audio output signal to the first pair of pins. A direct current (DC) shifting buffer set is coupled between the audio codec circuit and the first pair of pins, and is directly coupled to the output buffer set and the first pair of pins. An audio port is coupled to the first pair of pins. The DC shifting buffer set is configured to receive a first audio input signal from the audio port through the first pair of pins. The audio codec circuit is configured to record the audio output signal back and mix the recorded audio output signal with a second audio input signal, or mix the first audio input signal with a third audio input signal.

Abstract: A real-time service playback method includes, during running a real-time service, in response to receiving an input instruction, obtaining a playback speed and a time coordinate of a first anchor point, the time coordinate of the first anchor point being a start point of playback, and in response to the input instruction, stopping playing a real-time content of the real-time service, based on the time coordinate of the first anchor point, calculating a time coordinate of a second anchor point in future, and reading and storing a real-time content between a time coordinate value of the first anchor point and a time coordinate value of the second anchor point to play according to the playback speed until the playback reaches the time coordinate value of the second anchor point to resume the real-time service. The time coordinate of the second anchor point is an end point of the playback.

Abstract: In at least one embodiment, an active noise cancellation system. The system includes a first controller, a data bus, and a second controller. The first controller receives first information from a plurality of noise sensing devices and second information from an audio system. The first controller employs a time division multiplexing scheme to generate a multiplexed stream of data including the first information and the second information. The data bus transmits the multiplexed stream of data on a single data channel. The second controller receives the multiplexed stream of data on the single data channel and separates the first information from the second information on the multiplexed stream of data to perform ANC functionality.

Abstract: Mixer devices for efficiently processing multitrack audio including: at least one audio input port for receiving local audio from an audio input device as at least one local audio input; a memory; a user interface, the user interface including at least one adjuster; and at least one control unit electrically coupled to said receiver, said transmitter; said at least one audio input port, said memory, and said user interface; the at least one control unit programmed to control local gain of the local audio and local mixing. In another aspect, the mixer devices also remotely control the gain of bodypack receivers capable of receiving local audio input in analog or digital form. A novel display is also incorporated to facilitate such control surrounding the adjustor capable of performing the control.

Abstract: An audio processing device includes: a sound level measuring circuit arranged to operably generate multiple sound level values, wherein the multiple sound level values respectively correspond to the sound levels generated by an audio playback device at multiple time points or the sound levels received by a microphone at multiple time points; an audio dose calculating circuit coupled with the sound level measuring circuit and arranged to operably generate an audio dose value corresponding to a measuring period based on the multiple sound level values and contents of a weighting table; a control circuit coupled with the audio dose calculating circuit and arranged to operably compare the audio dose value with a dose threshold to determine whether to generate a control signal or not; and an indication signal generating circuit coupled with the control circuit and arranged to operably generate a corresponding indication signal according to the control signal.

Abstract: Methods, systems, and computer program products for network-based processing and distribution of multimedia content of a live performance are disclosed. In some implementations, recording devices can be configured to record a multimedia event (e.g., a musical performance). The recording devices can provide the recordings to a server while the event is ongoing. The server automatically synchronizes, mixes and masters the recordings. The server performs the automatic mixing and mastering using reference audio data previously captured during a rehearsal. The server streams the mastered recording to multiple end users through the Internet or other public or private network. The streaming can be live streaming.

Abstract: The present technology relates to a signal processing apparatus and method, and a program that can easily determine a localization position of a sound image. A signal processing apparatus includes: an acquisition unit configured to acquire information associated with a localization position of a sound image of an audio object in a listening space specified in a state where the listening space viewed from a listening position is displayed; and a generation unit configured to generate a bit stream on the basis of the information associated with the localization position. The present technology can be applied to the signal processing apparatus.

Abstract: A system comprises an auditory device processor, an auditory device output mechanism, an auditory input sensor, a database including a reference bank of environmental sounds and corresponding sound profiles, and a memory.

Abstract: The present invention provides for an analog audio patchbay comprising a first plurality of connectors capable of receiving a plurality analog electronic signals, a second plurality of connectors capable of transmitting a plurality of analog electronic signals, and a circuit board extending from the first plurality to the second plurality.

Abstract: Sound device receives a first sound signal from a first sound processor, generates a second sound signal based on the first sound signal, and transmits the second sound signal to a second sound processor. The second sound processor performs signal processing to the second sound signal to generate a third sound signal. The sound device receives the third sound signal from the second sound processor, checks a state of the second sound processor based on a signal received from the second sound processor, transmits a fourth sound signal based on the third sound signal to the first sound processor when determining that the state of the second sound processor is normal, and generates a fifth sound signal based on the first sound signal or the second sound signal to transmit it to the first sound processor when determining that the state of the second sound processor is abnormal.

Abstract: A system for generating channel-compensated features of a speech signal includes a channel noise simulator that degrades the speech signal, a feed forward convolutional neural network (CNN) that generates channel-compensated features of the degraded speech signal, and a loss function that computes a difference between the channel-compensated features and handcrafted features for the same raw speech signal. Each loss result may be used to update connection weights of the CNN until a predetermined threshold loss is satisfied, and the CNN may be used as a front-end for a deep neural network (DNN) for speaker recognition/verification. The DNN may include convolutional layers, a bottleneck features layer, multiple fully-connected layers and an output layer. The bottleneck features may be used to update connection weights of the convolutional layers, and dropout may be applied to the convolutional layers.

Abstract: An audio signal processing method performs signal processing on a first audio signal to be outputted to a first device that a performer uses, the first audio signal on which the signal processing has been performed being a second audio signal, receiving a setting that causes the first audio signal to send to a monitor bus which is for to output the second audio signal, and performing signal processing on the second audio signal, which is received via the monitor bus and is to be outputted to a second device different from the first device, such that a sound quality of a sound to be outputted by the second device is closer to sound quality of a sound to be outputted by the first device than in a case where the signal processing is not performed on the second audio signal.

Abstract: A system for generating audio content in dependence upon an input audio track comprising audio corresponding to one or more sound sources, the system comprising an audio input unit operable to input the input audio track to one or more models, each representing one or more of the sound sources, and an audio generation unit operable to generate, using the one or more models, one or more audio tracks each comprising a representation of the audio contribution of the corresponding sound sources of the input audio track, wherein the generated audio tracks comprise one or more variations relative to the corresponding portion of the input audio track.

Abstract: A hearing device includes: first and second primary microphones for provision of first and second primary microphone input signals; a secondary microphone for provision of a secondary microphone input signal; a mixing module for provision of a mixer output based on a primary mixer input and/or a secondary mixer input, primary mixer input being based on the first and second primary microphone input signals, the secondary mixer input being based on the secondary microphone input signal; a processing unit configured to provide an electrical output signal; and a receiver for converting the electrical output signal to an audio output signal; wherein the mixing controller is configured to increase an amount of the secondary mixer input in the mixer output for a time period after a presence of a sound event is determined, and reduce the amount of the secondary mixer input in the mixer output after the time period.

Abstract: Systems and methods for programmable pulse density modulation (PDM) components enable backwards compatibility while maintaining reasonable tolerances. A system includes a programmable PDM device, a PDM master device and a bus communicably coupling the programmable PDM device to the PDM receiver. The PDM device may include an audio sensor, audio input circuitry, a delta-sigma converter and a PDM transmitter and receiver. The PDM transmitter and receiver may send out PDM data from the PDM device and receive programming data from the PDM Master device. The PDM device may further include register space controlled by the PDM master device, a buffer storing audio data for wakeup word systems that store audio data when the PDM receiver is powered down, a bus holder to hold the previous value on the bus if no device is driving it, and/or a clock multiplier to multiply the incoming clock by a factor.

Abstract: A microphone system, comprises a first transducer, for generating a first acoustic signal, and a second transducer, for generating a second acoustic signal. A high-pass filter receives the first signal and generates a first filtered signal, and a low-pass filter receives the second signal and generates a second filtered signal. An adder forms an output signal of the microphone system as a sum of the first filtered signal and the second filtered signal.

Abstract: A digital mixer is disclosed having a plurality of displays and which applies a mixing process on an audio signal. The digital mixer includes an input interface, one or more processors, an output interface, at least a first display and a second display, and an operator. The operator includes a mode switching button for switching between a first mode and a second mode. The one or more processors are configured to cause parameter setting screens of audio signals of different channels to be displayed respectively on the first display and the second display when the mode switching button is set to the first mode, and cause different parameter setting screens of an audio signal of a single channel to be displayed respectively on the first display and the second display when the mode switching button is set to the second mode.

Abstract: A signal processor and a signal processing method thereof. The signal processor obtain a first audio signal of a first channel, obtains a second audio signal of a second channel. The signal processor performs a first signal processing on the input first audio signal. The signal processor, when it does not obtain the second audio signal, performs a second signal processing on the input first audio signal having undergone the first signal processing and output a further processed first audio signal.

Abstract: An audio signal processing apparatus includes a first bus, a second bus, and a controller that assigns a first input channel to the first bus, assigns a second input channel to the second bus, receives an amount of feed from the first input channel to the first bus, and receives an amount of feed from the second input channel to the second bus.

Abstract: The present disclosure relates to multimedia data playing methods and terminal devices. One example method includes obtaining, by a terminal device, a first sound effect plug-in, where the first sound effect plug-in corresponds to a first sound effect, and the terminal device includes a first application and a second application that are used to play multimedia data, receiving, by the terminal device, a first instruction used to indicate that a user of the terminal device selects the first sound effect option, wherein where the first sound effect option is an option corresponding to the first sound effect, when first multimedia data is to be played by the first application or the second application, processing, by the terminal device, the first multimedia data by using the first sound effect plug-in according to the first instruction to play the first multimedia data with the first sound effect.

Abstract: A method for playing audio source, executed by one or more processors, comprising: playing an audio source set including a first audio source and a second audio source; displaying a first object for an interaction with a user; obtaining an input for the first object; and adjusting at least a part of the audio source set based on the input; wherein in order that a selective change to the audio source set is given by the interaction with the user, when a first input related to the first object is obtained, the audio source set is adjusted so that an output level of the first audio source selected by the first input is highest, wherein in order that a random change to the audio source set is given by the interaction with the user, when a second input related to the first object is obtained, a random audio effect is applied to at least one audio source included in the audio source set.