Abstract: Embodiments of this application provide a volume adjustment method and device, and a system. The method includes: A first terminal device sends a first playback request to a server, where the first playback request carries a model of the first terminal device and an identifier of the first audio. The server determines a first volume value of the first audio based on the identifier of the first audio and the model of the first terminal device. The server determines, based on the first volume value and a preset volume threshold of the first terminal device, a first volume gain. The first terminal device determines, based on the first volume gain and a currently set second volume value, a first target volume value. The first terminal device adjusts volume of playing the first audio to the first target volume value.

Abstract: A method for normalizing platform-adaptive audio includes encoding input video content and generating video stream data as original data to store the video stream data in storage; generating loudness metadata for audio data of the video content and storing the loudness metadata in the storage; receiving a request for the video content from a client; searching the storage for video stream data of the video content corresponding to the request, the loudness metadata, and a device profile corresponding to device information included in the request; and transmitting, to the client, a response including the video stream data, the loudness metadata, and the device profile that are found in the storage.

Abstract: Disclosed are systems and methods for generating a long-term signal level estimate during automatic gain control. In an example method, a computing device receives an input signal comprising feed data, the input signal derived from an upstream input signal. The computing device selects a first subset of the feed data and divides the first subset of the feed data into a plurality of analysis intervals. The computing device calculates a statistical average of each analysis interval and generates a first Gaussian distribution based on the statistical average of each analysis interval of the plurality of analysis intervals. Based on one or more characteristics of the first Gaussian distribution, the computing device determines the long-term signal level estimate, generates a first stage gain based on the long-term signal level estimate, and applies the first stage gain to the input signal.

Abstract: Controlling the output of audio based on the mode of an electronic device having a sound mode, silent mode, and mute mode, along different audio output routes. In the sound mode, all audio can be output, while only registered or authorized audio can be output in the silent mode. In the mute mode, no audio can be output. The sound and silent modes can be enabled using an interface of the device, while the mute mode can be enabled using an accessory coupled to the device. To disengage the mute mode, a user can provide a corresponding instruction using the accessory or providing an instruction on the device related to volume control. Other aspects are also described and claimed.

Abstract: Methods for performing dynamic range compression (DRC) on audio in a manner intended to produce output audio for playback by systems or devices with limited power handling capabilities and preferably also to reduce or prevent undesirable artifacts (e.g., pumping and/or breathing) in the output audio. Some embodiments perform the DRC so as to maximize average loudness (while preventing loss of quieter elements) during playback, and also to reduce or prevent distortion. Other aspects are systems or devices configured to perform embodiments of the method. In some embodiments, reduced DRC is applied when average loudness of the input audio approaches (or matches or exceeds) a target (e.g., a knee point for DRC, or a signal level near to a maximum playback level of the intended playback system), since such input audio is assumed to have already been compressed, and otherwise applying full DRC to the input audio.

Abstract: A method for processing an audio signal includes setting target loudness; receiving metadata of an audio signal from a server; and adjusting the loudness of the audio signal by using the received metadata, so that the loudness of the audio signal corresponds to the set target loudness.

Abstract: Audio systems, methods, and computer readable media for managing audio headroom are provided. At least two input signals are received and combined. The combined signals is compared to a threshold value and may be adjusted if the combined signal exceeds the threshold value. A gain is applied to the first input signal based upon the combined signal being above the threshold value. The gain may be an attenuating gain. Accordingly, a gain is applied to at least one input signal to reduce the input signal when a combination of the input signals exceeds a threshold value.

Abstract: A display device including a display panel including a substrate and a pixel array layer disposed on a first surface of the substrate, a sound generation device disposed on a second surface of the substrate opposite to the first surface of the substrate, and a circuit board disposed on the second surface of the substrate, in which the circuit board includes a first hole overlapping the sound generation device.

Abstract: A method performed by an audio source device that is communicatively coupled with an audio output device, the method includes obtaining an audio signal; receiving a volume adjustment; applying the volume adjustment to the audio signal; applying dynamic range compression on the volume adjusted audio signal; and applying an inverse volume adjustment to the compressed audio signal, where the inverse volume adjusted audio signal is to be transmitted to the audio output device.

Abstract: A system for automatically adjusting a volume of a loudspeaker includes a loudspeaker and a processor. An infrared sensor, which generates a sensing signal when sensing IR radiation from a heat-radiating source, is disposed on the loudspeaker. The sensing signal provides a size information commensurate with a size of the heat-radiating source. The processor receives a plurality of the sensing signals at different time points to obtain a plurality of corresponding size information, realizes a change of distance between the heat-radiating source and the loudspeaker according to a change of the size information, and generates a volume-adjusting signal according to the change of distance between the heat-radiating source and the loudspeaker. The volume-adjusting signal is transmitted to the loudspeaker to adjust the volume of the loudspeaker accordingly.

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.