Abstract: A method for operating a hearing device on the basis of a speech signal. An acousto-electric input transducer of the hearing device records a sound containing the speech signal from surroundings of the hearing device and converts the sound into an input audio signal. A signal processing operation generates an output audio signal based on the input audio signal. At least one articulatory and/or prosodic feature of the speech signal is quantitatively acquired through analysis of the input audio signal by way of the signal processing operation, and a quantitative measure of a speech quality of the speech signal is derived on the basis of the property. At least one parameter of the signal processing operation for generating the output audio signal based on the input audio signal is set on the basis of the quantitative measure of the speech quality of the speech signal.

Abstract: A bitstream is obtained by a decoder side, that contains an encoded version of an audio signal and an instantaneous loudness sequence of the audio signal. The instantaneous loudness sequence has not been loudness normalized. A dynamic range control, DRC, gain sequence is produced by applying the instantaneous loudness sequence to a DRC characteristic, with loudness normalization. The DRC gain sequence is applied to the decoded audio signal. Other aspects are also described and claimed.

Abstract: Disclosed embodiments provide glitch prediction based on machine learning algorithms in mixed analog and digital systems, particularly directed to digital microelectromechanical (MEMS) multipath acoustic sensors or microphones, which allow seamless, low latency gain changes without audible artifacts or interruptions in the audio output signal.

Abstract: Embodiments are provided for enabling a user to control multiple media playback systems with a single mobile device. The mobile device may connect with a first or second media playback system based on receiving a first or second household identifier (HHID), respectively. If the mobile device receives both the first and second HHIDs, the mobile device may connect with either the first or second media playback system based on a priority of the two systems. If neither the first nor second HHIDs are received by the mobile device, a user may be prompted to add a third media playback system if the mobile device receives a third HHID. The user interface of the mobile device may receive an input to connect the mobile device to the third media playback system, and a connection may be established between the mobile device and the third media playback system.

Abstract: An audio processing unit (APU) is disclosed. The APU includes a buffer memory configured to store at least one frame of an encoded audio bitstream, where the encoded audio bitstream includes audio data and a metadata container. The metadata container includes a header and one or more metadata payloads after the header. The one or more metadata payloads include dynamic range compression (DRC) metadata, and the DRC metadata is or includes profile metadata indicative of whether the DRC metadata includes dynamic range compression (DRC) control values for use in performing dynamic range compression in accordance with at least one compression profile on audio content indicated by at least one block of the audio data.

Abstract: An example first playback device includes programming to perform functions including: (1) storing an active volume state variable in memory, wherein the active volume state variable corresponds to a current playback volume; (2) storing a volume limit state variable in memory, wherein the volume limit state variable corresponds to a playback volume limit of the first playback device; (3) detecting a command to begin playback of media at a proposed playback volume different from the current playback volume; (4) based on comparing (i) the playback volume limit and (ii) the proposed playback volume, selecting a startup playback volume; (5) playing back media at the startup playback volume; and (6) causing at least a second playback device of the media playback system to play back media at the startup playback volume.

Abstract: Example techniques are described for coding video data by obtaining a block of video data, obtaining an adaptive parameter set, determining a set of adaptive loop filter parameters for a plurality of filters for the block of video data based on the adaptive parameter set, wherein a plurality of adaptive loop parameters of the set of adaptive loop filter parameters are signaled using the same signaling parameter for each of the plurality of filters of the adaptive parameter set, and coding the block of video data using the set of adaptive loop filter parameters. The example techniques can be performed as part of an encoding or decoding process and/or by an encoder or a decoder.

Abstract: A system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of electronic device audible cues. Audible cues are stored as sound files. Ambient noise is detected, and the output of the audible cues is altered based on the ambient noise. Various embodiments include processed sound files that are more robust in noisy environments.

Abstract: Audio processing with audio transparency can include receiving a user content audio signal and receiving a microphone signal. The microphone signal can contain sensed sound of a user environment. Strength of the sensed sound can be increased based on strength of the user content audio signal, to reduce a masking of the sensed sound during playback. The sensed sound and the user content audio signal can be combined in a composite output audio signal used to drive a speaker. Other aspects are also described and claimed.

Abstract: An input at a physical interface of a first playback device is detected. The input may indicate a playback command. Via the first playback device, the second playback device is determined to be currently playing media content. Based on determining that the second playback device is currently playing media content and based on receiving the input at the physical interface, the first playback device is caused to play the media content synchronously with the second playback device.

Abstract: Disclosed is a method of operating an audio signal processing apparatus playing content including an audio signal. The method includes receiving the audio signal, receiving metadata including information related to a loudness of the audio signal, the metadata including loudness distribution information indicating, for each of a plurality of steps separated according to a loudness magnitude, a ratio between an amount of the audio signal corresponding to each of the plurality of steps of the audio signal and a total amount of the audio signal, and adjusting the loudness of the audio signal based on the metadata.

Abstract: A sound output device includes: first circuitry, connected to a first audio device, that outputs sound based on sound output data via the first audio device according to first sound volume data; and second circuitry connected to a second audio device, including an amplifier that amplifies sound to be output from the second audio device according to second sound volume data. The second circuitry controls a value of the second sound volume data to be a fixed value, receives, from the first circuitry, the sound output data and the first sound volume data, and outputs sound based on the sound output data that is received, via the second audio device, according to the first sound volume data that is received.

Abstract: An electronic apparatus and method for enhancement of audio for users with a hearing impairment in a shared listening environment is provided. The electronic apparatus receives first audio content from a media source and detects a first user with a hearing disability as a wearer of the personal listening device. The electronic apparatus modifies one or more features of the first audio content based on an audio enhancement profile associated with the detected first user. Thereafter, the electronic apparatus generates second audio content based on the modification and shares the generated second audio content with the personal listening device.

Abstract: In accordance with embodiments of the present disclosure, a method for processing audio information in an audio device may include reproducing audio information by generating an audio output signal for communication to at least one transducer of the audio device, receiving at least one input signal indicative of ambient sound external to the audio device, detecting from the at least one input signal a near-field sound in the ambient sound, and modifying a characteristic of the audio information reproduced to the at least one transducer in response to detection of the near-field sound.

Abstract: An ANC system is provided, including an AD converter performing an AD conversion on an external noise signal; an ANC signal generator generating an ANC signal for canceling a noise component arriving at ears of a user based on an output signal of the AD converter; a level detector detecting a level of the output signal and causes the ANC signal generator to power down in response to the level; and a zero-cross detector detecting a zero-cross timing of the ANC signal. The level detector starts measuring a time when the level is equal to or less than a first threshold value, and causes the ANC signal generator to perform a power down operation when the zero-cross timing is detected after the measured time exceeds the predetermined value, and causes the ANC signal generator to exit from the power down operation when the level exceeds a second threshold value.

Abstract: A communication apparatus includes a wireless communication unit configured to wirelessly communicate with an external apparatus, a speaker, a memory, and a control unit, wherein in a case where an operation is executed to change the communication apparatus to a predetermined mode in which a sound is not output from the speaker, the control unit performs control such that the communication apparatus is changed to the predetermined mode, wherein in a case where identification information about the external apparatus is stored in the memory, the control unit enables a wireless communication function in response to the execution of the operation to change the communication apparatus to the predetermined mode, and wherein in a case where the identification information about the external apparatus is not stored in the memory, even if the operation to change the communication apparatus to the predetermined mode is executed, the wireless communication function is not enabled.

Abstract: Systems and methods are provided for processing an input dataset or running an inference. The systems and methods may be configured to accept an input dataset, access one or more predefined logic plugins for processing the input dataset, process the input dataset based at least in part on a first predefined logic plugin, and generate the one or more outputs based at least in part of the processing of the input dataset. The one or more outputs may have a different format than a format of the input dataset.

Abstract: One or more processors of an electronic device detect a communication device electronically in communication with a content presentation companion device operating as a primary display for the electronic device and including a first ultra-wide band component. The one or more processors determine, with a second ultra-wide band tag component carried by the electronic device, a distance between the electronic device and the content presentation companion device using an ultra-wide band ranging process. The one or more processors dynamically enhance an audio performance characteristic of the content presentation companion device as a function of the distance between the electronic device and the content presentation companion device.

Abstract: An ANC system includes an AD converter which performs AD conversion on an external noise signal, an ANC signal generator which generates an ANC signal for canceling a noise component arriving at the ears of a user based on an output signal of the AD converter, and a level detector which detects a level of the output signal and causes the ANC signal generator to power down in response to the level. The level detector measures a time for which the level is equal to or less than a predetermined first threshold value, causes the ANC signal generator or a portion of blocks of the AD converter to power down after the measured time exceeds a predetermined value, and causes the ANC signal generator or a portion of blocks of the AD converter to return from the power down when the level exceeds a predetermined second threshold value.

Abstract: A controllable device, such as a set top box, responds to a transmission received from a one of a plurality of controlling devices of differing capabilities by entering into a one of a plurality of operating modes wherein the one of the plurality of operating modes entered into corresponds to the capabilities of the controlling device from which the transmission originated.

Abstract: A wearable accessory for delivery of personalized audio service includes a first microphone for capturing sound, a first speaker for playing media, a logic circuit for creating a personalized Head Related Transfer Function (HRTF) to enhance listening experience of the media, a communication module for accessing the Internet and to provide access to the media in response to capturing the sound, and a memory storage unit for storing information related to the captured sound. Other embodiments are disclosed.

Abstract: Device for outputting audio signals into passenger compartment of a motor vehicle comprising: an audio output which comprises a plurality of audio output elements which can be or are arranged in an interior of a motor vehicle, a controller which is or can be assigned to the audio output and which is configured for controlling at least one output parameter of the audio signals which can be output via respective audio output elements, the controller is configured to control the at least one output parameter of respective audio output elements of the audio output on the basis of orientation information generated by a detection means which is configured for detecting the orientation and/or movement of a chassis of a motor vehicle relative to a ground, and for generating orientation information which describes a detected orientation and/or movement of the chassis of the motor vehicle relative to the ground.

Abstract: An image capture device may capture voice commands during capture of video. Voice commands may be located within the audio of the video, and the presence of the voice commands may be reduced.

Abstract: A sound volume operation device configured to amplify and output an inputted acoustic signal includes: at least one sound volume adjuster configured to adjustably amplify an inputted acoustic signal; an output level detector configured to detect an output sound volume level of the acoustic signal amplified by the at least one sound volume adjuster; and an illuminator provided at or adjacent to the at least one sound volume adjuster and configured to provide a different illumination state according to the output sound volume level detected by the output level detector.

Abstract: A method of generating audio loudness performed by an audio loudness generation device may include: receiving loudness information on each of a plurality of audio tracks included in one group; predicting an intermediate loudness distribution, which is a loudness distribution for the one group, on the basis of the loudness information on each of the plurality of audio tracks; and generating an integrated loudness for the one group on the basis of the intermediate loudness distribution.

Abstract: Embodiments of active noise control (ANC) headphones and operating methods thereof are disclosed herein. In one example, a headphone includes a speaker, an internal microphone, and a processor. The speaker is configured to play an audio of interest based on an audio source signal. The internal microphone is configured to obtain a mixed audio signal including a noise signal and the audio of interest played by the speaker. The processor is configured to determine a first current system parameter of the headphone at a first time point, and determine if the first current system parameter of the headphone is higher than a predetermined threshold to determine if the headphone is worn by a user.

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: An activity sensing circuit includes an ammeter circuit configured to monitor a supply current drawn specifically by a webcam. A hardware trigger circuit receives a webcam supply current monitoring signal from the ammeter circuit, and detects when the webcam supply current monitoring signal indicates the supply current drawn specifically by the webcam exceeds a first threshold supply current level. In response to detection of the supply current exceeding the first threshold supply current level, the trigger circuit notifies an indicator circuit to provide a user-perceptible indication. The indicator circuit is immune to control by a processor of an information handling system of which it is part.

Abstract: An electronic device comprises a speaker protection module (36) arranged to receive an audio signal (22) and to use one or more parameters relating to a loudspeaker to produce a controlled audio signal (34). An amplifier (40) is arranged to amplify the controlled audio signal to produce an amplified audio signal (24, 26) which is passed to the loudspeaker. The amplitude of the controlled audio signal (34) is such that the amplified audio signal (24, 26) is less than or equal to an amplitude determined to be safe for the loudspeaker in view of the one or more parameters. An ultrasound generator (16) is arranged to generate an ultrasound signal (30) that is mixed (32) with the controlled audio signal (34) before the controlled audio signal (34) is passed to the amplifier (40). The ultrasound generator (16) is capable of producing said ultrasound signal at a plurality of frequencies.

Abstract: A signal processing system and method is disclosed for applying time-based effects to an N-channel audio input signal for reproduction on a set of loudspeakers having a predetermined configuration. A first M-channel audio signal is produced from the N-channel audio input signal. Each channel of the first M-channel audio signal is associated with a subset of the loudspeakers. A second M-channel audio signal is produced from the first M-channel audio signal according to an M×M matrix, each element aij in the M×M matrix including a delay term. A minimum value of the delay term in each element aij is determined according to a distance between at least two loudspeakers in at least one of the i and j subsets of loudspeakers and according to minimum delay value of a time-based delay effect applied to each channel of the second M-channel audio signal. A K-channel audio signal is then produced from the or each second M-channel audio signal.

Abstract: A system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of electronic device audible cues. Audible cues are stored as sound files. Ambient noise is detected, and the output of the audible cues is altered based on the ambient noise. Various embodiments include processed sound files that are more robust in noisy environments.

Abstract: A system configured to perform adaptive feedback reduction, such as howling suppression, to prevent feedback from occurring when two devices are acoustically coupled during a communication session. To reduce feedback between the acoustically coupled devices, a device monitors input audio data from a microphone and determines whether feedback is present in individual frequency subbands. For example, the device determines an expected value for a first subband using neighboring subbands (e.g., averaging subbands surrounding the first subband) and compares the expected value to an actual value for the first subband. If the actual value is noticeably higher than the expected value, the device determines that feedback is present in the first subband and attenuates the first subband. If the feedback is not present, the device passes the first subband without attenuation.

Abstract: The present technology relates to an information processing device, an information processing method, a voice output device, and a voice output method that allow a user to hear a speech from a home agent unit regardless of where the user is. The voice output device outputs, on the basis of an image and noise from a predetermined noise source which are obtained in a unit capable of outputting a voice toward a user, information indicating hearing difficulty of a voice from the unit at a position of the user. The present technology can be applied to a household voice assistant device that performs a speech to a user.

Abstract: When the volume is adjusted in a multi-speaker system, it is desirable that one speaker does not change volume disproportionately with respect to another speaker. A method is presented for adjusting a volume level of one or more speakers. Each speaker can have a non-standardized relationship between logical volume level that is input to the speaker and sound pressure level that is produced by the speaker. A selected volume level, corresponding to a sound pressure level, can be received via a user interface. A stored lookup table can be accessed to convert the sound pressure level to a first product-specific logical volume level for each speaker. The stored lookup table can tabulate the non-standardized relationship between logical volume level and sound pressure level for each speaker. Data corresponding to the first product-specific logical volume level can be transmitted to each speaker.

Abstract: A method of verifying audio notifications generated within a vehicle includes generating an audio notification using an audio generation path that includes at least a controller, an audio notification generator and a speaker, wherein the controller generates an audio notification request utilized by the audio notification generator to generate an audio signal, and wherein the speaker converts the audio signal into the audio notification. The method further includes monitoring sounds within the vehicle using a microphone separate from the audio generation path and analyzing the monitored sounds using an audio analyzer, wherein the audio analyzer verifies whether the audio notification was generated. Feedback is provided to the audio generation path regarding whether generation of the audio notification was verified.

Abstract: Systems and methods for detection, classification, and diagnosis of vocal anomalies in vocal streams are disclosed. Discussed are a method for generating a biometric voiceprint for analyzing user vocal streams to detect and classify vocal anomalies and a method for notifying the appropriate party, where the notification is based on the diagnosis reported by the system. Manual classification in the event of automatic classification failure is discussed, where the manual classification data can be used as training data to improve the functionality of the classification model.

Abstract: Automatic control of media presentation parameters is provided by using one or more of real-time audio playback measurement data from microphones and audience facial and body expression interpretation from video and infrared cameras, in conjunction with artificial intelligence for interpretation and evaluation of facial and body expression and predetermined perceptual audio models. Media presentation parameters can include, for example, speaker volume, audio equalization, feedback elimination, play/pause, and other audio content-related aspects of presentation. In some embodiments, additional environmental parameters can be modified to enhance audience experience, such as, for example, temperature, lighting, and the like, in response to audience facial and body expression.

Abstract: A microphone assembly comprising: a housing including a base, a cover, and a sound port; a MEMS transducer element disposed in the housing, the transducer element configured to convert sound into a microphone signal voltage at a transducer output; and a processing circuit. The processing circuit comprising a transconductance amplifier comprising an input node connected to the transducer output for receipt of the microphone signal voltage, the transconductance amplifier being configured to generate an amplified current signal representative of the microphone signal voltage in accordance with a predetermined transconductance of the transconductance amplifier; and an analog-to-digital converter comprising an input node connected to receive the amplified current signal, said analog-to-digital converter being configured to sample and quantize the amplified current signal to generate a corresponding digital microphone signal.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for dynamic volume adjustment via audio classification. Examples methods include analyzing, with a neural network trained model, a parameter of an audio signal associated with a first volume level to determine a classification group associated with the audio signal, determining an input volume of the audio signal, the selection based on the classification group associated with the audio signal, applying a gain value to the audio signal, the gain value based on the classification group and the input volume, the gain value to modify the first volume level to a second volume level, and applying a compression value to the audio signal, the compression value to modify the second volume level to a third volume level that satisfies a target volume threshold.

Abstract: Example techniques may involve headphone interaction. An implementation may include while headphones are disconnected from a control device, the control device receiving an indication of particular audio content being played back by a first zone of the media playback system. While the particular audio content is being played back by the one or more playback devices of the media playback system, detecting that headphones have been connected to the first control device. Headphones may be connectable to the control device via either (a) an analog headphone jack or (b) a point-to-point personal area network connection. In response to detecting that the headphones are connected to the control device: the control device (i) causing the one or more playback devices of the first zone to stop playback of the particular audio content; (ii) retrieving the particular audio content and (iii) resuming playback of the particular audio content via the connected headphones.

Abstract: Control techniques are disclosed for devices that are members of ad hoc rendering networks. A first network device may receive a data record representing state of the network. The data record may contain data identifying device(s) that are members of the network, any grouping(s) of the device(s) defined for the network, and a play state of the device(s). When the first network device receives the data record, it may store the data record locally at the first player device. The first player device may determine whether the data record possesses a command altering a play state of a device. If so and if the command is addressed to the first player device, the first player device may execute the command at the first player device. The data record may be relayed among all devices in the ad hoc rendering network, and each device may perform this same process. In this manner, state updates propagate to all devices that are members of the rendering network, and all such devices store state data of all other devices.

Abstract: An audio system, including at least one approximating circuit, the at least one approximating circuit configured to apply an approximated gain to the input signal, the least one approximating circuit outputting an approximated signal; a limiter configured to reduce a magnitude of the input signal by a gain determined by an amount the magnitude of the approximated filtered signal exceeds a predetermined threshold, the limiter outputting a limited input signal; a filter being respectively configured to receive the limited input signal and to respectively apply a filter gain, the filters outputting a plurality of filtered signals, wherein the approximated gain is substantially similar to the filter gain of one of the plurality of equalization filters; and a speaker array comprising a plurality of speakers, each speaker configured to receive a respective filtered signal of the plurality of filtered signals and to transduce the filtered signal to an audio signal.

Abstract: Provided are an electronic device and an operation method thereof. The operation method of an electronic device for processing an audio signal may include obtaining viewing environment information related to sound intelligibility, processing an input audio signal by separating the input audio signal into a first channel including a primary signal and a second channel including an ambient signal based on the viewing environment information, processing the input audio signal based on a frequency band and based on the viewing environment information, and generating an output signal based on processing the input audio signal.

Abstract: Provided are a method and a device for identifying a sound level of audio content, the method including receiving an audio signal and obtaining a first hash address based on a sound level of the audio signal, providing a hash address database in which the sound level of the audio signal is made to correspond to a hash address among a plurality of hash addresses, based on a first sound level corresponding to the first hash address and the sound level of the audio signal corresponding to the first sound level, cumulatively counting a first count value of the first hash address, and identifying the sound level of the audio content received during a first time period based on count information corresponding to a total amount of the first count value accumulated during the first time period.

Abstract: An audio reproduction method comprises providing equal loudness response data defining a dependency between sound pressure level and frequency, resulting in sounds of equal loudness, including for sounds of a notional negative loudness; generating a required spectral weighting profile from the response data to represent a difference in frequency response between a first audio transducer and a second, different, audio transducer; applying a spectral weighting to an audio signal in dependence upon the derived frequency response; and reproducing the spectrally weighted audio signal using the second audio transducer at an audio reproduction level.

Abstract: Provided are a method and an apparatus for processing audio data in a sound field. The method includes: acquiring the audio data in the sound field; processing the audio data through a preset restoration algorithm to extract audio data information about the sound field carried by the audio data; acquiring motion information about a target, and generating, through a preset processing algorithm, target-based sound field audio data based on the audio data information and the motion information about the target.

Abstract: A biasing circuit providing power to a microphone is disclosed. The biasing circuit includes a first impedance element, a second impedance element, a detection circuit and a control circuit. The first impedance element has a first impedance and is coupled between a first power node and a first terminal of the microphone. The second impedance element has a second impedance and is coupled between a second terminal of the microphone and a second power node. The detection circuit is coupled between the first and second terminals and generates a detection signal according to an analog signal generated by the microphone. The control circuit adjusts the first and second impedances according to the detection signal.

Abstract: An image processing apparatus includes an image reading portion or an image forming portion, a detection processing portion, and an operation restricting portion. The image reading portion reads an image from a document sheet. The image forming portion forms an image on a sheet. The detection processing portion detects whether or not a speaker that is provided outside the image processing apparatus is in an output state. The operation restricting portion restricts an operation of the image processing apparatus when the detection processing portion detects that the speaker is in the output state.

Abstract: A sound output apparatus is provided. The sound output apparatus according to an embodiment includes an inputter, an outputter, a storage configured to store mapping information between a volume level and a dynamic range of the sound output apparatus and a processor configured to obtain a dynamic range corresponding to a volume level currently set to the sound output apparatus based on the mapping information, process a sound signal input through the inputter based on the dynamic range and output the processed sound signal through the outputter.

Abstract: Disclosed herein are system, method, and tangible computer readable medium for creating a desired audio effect for a user. The method includes operations including: causing a plurality of speakers to play test signals, each test signal being specific to one of the speakers; receiving from a remote device recorded frequency responses of the speakers resulting from the playing of the test signals; creating one or more filters to match an audio profile selected by a user; applying the filters to the recorded frequency responses to obtain filtered transformations of the speakers; and transmitting the filtered transformations to the speakers; wherein the filtered transformations are applied at the speakers to thereby achieve the user audio profile.