Abstract: Example techniques relate to grouping based on acoustic signals in a system including a first network device and a second network device. A first network device may detect a first acoustic signal, perhaps while playing back first audio content in a single-device playback configuration. The first network device sends, to the second network device, data corresponding to the detected first acoustic signal. The second network device may generate one or more instructions based on the sent data corresponding to the detected first acoustic signal and send data representing such instructions to the first network device. Based on the instructions, the first network device transitions from operating in the single-device playback configuration to operating in a grouped playback configuration. In the grouped configuration, the first network device outputs second audio content in synchrony with the third network device.

Abstract: Systems, devices, apparatuses, components, methods, and techniques for identifying and playing media content are provided. An example media-playback device for identifying and playing media content for a user traveling in a vehicle includes an audio identification engine and a media playback engine. Audio content is recorded and identified by comparison to media content databases. The audio content is identified and immediately played on the same device. Additional media content is selected for playback based on user listening preferences.

Abstract: A method of playing media files includes accessing a media library, identifying a plurality of media files in the media library, indexing metadata of the plurality of media files, targeting a first subset of media files in the plurality of media files based on metadata of a selected media file, and playing at least one media file of the first subset of media files.

Abstract: In multichannel audio coding, an improved coding efficiency is achieved by the following measure: the noise filling of zero-quantized scale factor bands is performed using noise filling sources other than artificially generated noise or spectral replica. In particular, the coding efficiency in multichannel audio coding may be rendered more efficient by performing the noise filling based on noise generated using spectral lines from a previous frame of, or a different channel of the current frame of, the multichannel audio signal.

Abstract: A terminal and a method for controlling a stereo sound effect thereof are provided. The terminal includes at least two speakers, and the method includes: presetting a relationship between an angle and a sound effect, the relationship represents a sound effect of each speaker at each corresponding angle value; acquiring, in real time, an angle value of an orientation of a screen of the terminal; and after acquiring the angle value, acquiring the a sound effect of each speaker at the angle value according to the relationship of each speaker, and controlling each speaker to play according to the corresponding sound effect. Mo matter how the terminal rotates, the sound effect played at the same angle value is kept unchanged, so that the terminal can show the consistent stereo sound effect at different angle values.

Abstract: An eye movement feature amount calculating system includes: a moving state input unit that inputs moving state values in a time series indicating a moving state of an eye of an examinee; a saccade period extracting unit that extracts a saccade period in which the eye performs saccadic movement based on time-series variation of the moving state values; and a feature amount calculating unit that divides a period including at least a part of the extracted saccade period into a plurality of periods and calculates a feature amount of eye movement of the subject based on the separate periods.

Abstract: Various embodiments of systems and methods allow a system to embed an item identifier into a content item. A first device can then play an audio trigger that is imperceptible to humans before playing the item identifier. A second device can go into an active listening mode after detecting the audio trigger and record an audio segment contain the embedded item identifier. A system can then decode the item identifier to determine an appropriate context for the second device. The second device can then receive a vocal command or query and respond according to the determined context. In one example, the first device can be a television, and the second device can be a digital assistant (e.g., Amazon Alexa) that detects advertisements played on the television via audio signals embedded in accompanying audio streams. Subsequent user interactions with the digital assistant can then be informed by the context of the recently-heard advertisements.

Abstract: Example techniques relate to a loss of control of a playback queue. An example implementation involves a playback device receiving, from a native controller, an instruction to add audio tracks to a queue, where, in a first mode associated with the native controller, the playback device is configured to play back from the queue. While the audio tracks are in the queue, the playback device receives, from a media player application associated with a particular wireless protocol, an instruction to play back a media item. In response, the playback device is configured to play back in a second mode associated with the particular wireless protocol, where, in the second mode, the queue is not in use. While playing back the media item, the playback device detects a loss of control by the media player application and configures the playback device to play back in the first mode.

Abstract: A computing system includes a virtual desktop server and a client computing device. The virtual desktop server includes a real-time media application to provide real-time communications (RTC) for peer-to-peer networking, and a native RTC engine to execute a portion of the real-time media application when received by the native RTC engine. An API code redirection module redirects intercepted APIs of the real-time media application so that the portion of the real-time media application is redirected away from the native RTC engine to a client RTC engine in the client computing device. The portion of the real-time media application being redirected includes provisioning information directed to a remote media server. A network location service provides provisioning information directed to a nearby media server.

Abstract: Performing an operation comprising extracting a first element of encrypted data embedded in a digital recording comprising at least one of audio data and image data captured in a recording environment, generating a first decrypted value, the generating performed by execution of a cryptography algorithm based on a private key and the first element of encrypted data, comparing the first decrypted value and a stored value associated with a first element of the digital recording, and authenticating the digital recording based on the first decrypted value substantially matching the stored value.

Abstract: A system is described for maintaining synchrony of operations among a plurality of devices that have independent clocking arrangements. The system includes a task distribution device that distributes tasks to a synchrony group comprising a plurality of devices that are to perform the tasks distributed by the task distribution device in synchrony. The task distribution device distributes each task to the members of the synchrony group over a network. Each task is associated with a time stamp that indicates a time, relative to a clock maintained by the task distribution device, at which the members of the synchrony group are to execute the task.

Abstract: A method of operating an audio system that includes a non-wearable acoustic system and a wearable acoustic device, each having at least one acoustic driver, includes generating, at the at least one acoustic driver of the non-wearable acoustic system, a first acoustic signal having a range of acoustic frequencies. In response to a request of a change of operational mode of the non-wearable acoustic system, a second acoustic signal having a first sub-range of the acoustic frequencies is generated at the at least one acoustic driver of the non-wearable acoustic system and a third acoustic signal having a second sub-range of the acoustic frequencies is generated at the at least one acoustic driver of the wearable acoustic device. The first sub-range of acoustic frequencies is different from the second sub-range of acoustic frequencies and the range of acoustic frequencies is inclusive of the first and second sub-ranges of acoustic frequencies.

Abstract: Described is a method for initiating a cashless payment transaction from a motor vehicle, wherein in a vehicle-external, stationary device an optical detecting apparatus generates image data of a representation apparatus positioned on the motor vehicle for optically representing a chassis number of the motor vehicle, and a recognition apparatus recognizes the representation apparatus in the image data, and based on the recognized representation apparatus, the chassis number of the motor vehicle is determined, and a retrieval apparatus, depending on the determined chassis number, retrieves user data of a user from a data storage, wherein the user data contain bank account data for a cashless payment, and wherein a control device performs the payment transaction in a cashless way, based on the bank account data.

Abstract: Methods and systems for providing information to a user are described. Multiple mobile devices can individually collect data and feed the data to beacons in a location. The information can include sound data, light data, motion data, health and wellness related data, humidity data, and/or temperature data. The beacons receive this data from multiple users and transmit it to a service provider. The service provider collects the data from the beacons and provides the data to one or more users.

Abstract: There is provided an error concealment unit, method, and computer program, for providing an error concealment audio information for concealing a loss of an audio frame in an encoded audio information. In one embodiment, the error concealment unit provides an error concealment audio information for a lost audio frame on the basis of a properly decoded audio frame preceding the lost audio frame. The error concealment unit derives a damping factor on the basis of characteristics of a decoded representation of the properly decoded audio frame preceding the lost audio frame. The error concealment unit performs a fade out using the damping factor.

Abstract: Method of processing a digital audio input signal containing a sequence of input signal sample values, the method comprising: determining signal difference values between consecutive pairs of input signal sample values in the sequence of input signal sample values; determining a normalization value; dividing each signal difference value in the group of signal difference values by a divider associated with the normalization value; reducing a bit depth of the normalization value; reducing a bit depth of each normalized signal difference value in the group of normalized signal difference values; transmitting the group of reduced normalized signal difference values, and the reduced normalization value; receiving the transmitted group of reduced normalized signal difference values, and the transmitted reduced normalization value; multiplying each reduced normalized signal difference value by a multiplier associated with the reduced normalization value; determining a digital audio output signal containing a sequ

Abstract: A system and method of modifying a binaural signal using headtracking information. The system calculates a delay, a first filter response, and a second filter response, and applies these to the left and right components of the binaural signal according to the headtracking information. The system may also apply headtracking to parametric binaural signals. In this manner, headtracking may be applied to pre-rendered binaural audio.

Abstract: Systems and methods for detecting volume level changes in communications are described herein. In some embodiments, a system comprises a computer system. The computer system comprises at least one processor and a memory coupled to the at least one processor. The memory stores program instructions that are executable by the at least one processor to cause the computer system to perform tasks. The tasks include recording a communication that comprises audio, and analyzing the audio of the communication. The analysis of the audio is operable to detect a change in a volume level of the audio that indicates an occurrence of a potential event of interest. The tasks also include creating and storing an information record corresponding to the communication in a second database. The information record includes an indication of the detected change in the volume level.

Abstract: Various embodiments relate generally to data science and data analysis, computer software and systems, and wired and wireless network communications to interface among repositories of disparate datasets and computing machine-based entities configured to access datasets, and, more specifically, to a computing and data storage platform to implement predict data constraints to validate one or more portions of a dataset, according to at least some examples. For example, a method may include predicting a subset of constraint data to validate a graph-based data arrangement, and analyzing the graph-based data arrangement against a subset of constraint data to determine an action. At least one action may include validating data in a graph-based data arrangement. Also, the method may include integrating graph-based data arrangement into a graph data arrangement responsive to determining data representing a validation.

Abstract: An audiobook quality control system may detect and flag errors in human voice recordings of manuscripts at real-time or near real-time. The system may receive a manuscript and process the manuscript to generate a comparison document. The system may receive audio data from a recording session and may use a speech-to-text transcription tool to generate transcribed audio. Based on comparing the transcribed audio to the comparison document, the system may flag differences as errors. Additionally, the system may flag common recording errors (e.g., consistency, pronunciation, spacing issue, noise, wrong words, etc.). Based on the error types and number of errors, the system may determine to re-record specific sections of the manuscript.

Abstract: System, method, and various embodiments for providing a speaker normalization system are described herein. An embodiment operates by identifying a plurality of audio output devices connected to an audio source. For each audio output device, a set of technical capabilities and a relative position of the audio output device is identified. An audio configuration is calculated based on both the set of technical capabilities and the relative position of each audio output device, and which audio output devices are identified. The calculated audio configuration is provided to each identified audio output device. Audio is provided to each of the identified audio output devices.

Abstract: An embodiment of the present invention prevents false detection of lifting. An embodiment of the present invention includes: an angle determination section (62) configured to determine whether or not an angle of a longitudinal direction of a mobile terminal (1) with respect to a vertical direction is less than or equal to a predetermined value; and a lifting determination section (65) configured to determine whether or not the mobile terminal (1) has been lifted, the lifting determination section carrying out determination in (i) a case where the angle of the longitudinal direction of the electronic device with respect to the vertical direction is greater than the predetermined angle and (ii) a case where (a) the angle of the longitudinal direction of the electronic device with respect to the vertical direction is less than or equal to the predetermined angle and (b) an acceleration of the electronic device in the longitudinal direction satisfies a predetermined condition.

Abstract: A software tool includes a plurality of pre-coded API sensor modules for obtaining information from the sensors coupled to a mobile device and a pre-coded API audio module for playing audio content based on the information obtained from at least one of the sensors.

Abstract: In some implementations, a computing device can generate personalized music selections to associate with any collections of visual media (e.g., photos and videos) stored on the computing device. A user may prefer particular genres of music and listen to some genres more frequently than others. The computing device can create measures of the user's genre preferences and use these measures to select music that is preferred by the user and music that may be significant or relevant to the particular collection of visual media. The computing device may also determine music that was being played when and where the visual media were being created. The computing device may store the visual media and music items in association with each other. The computing device may generate composite media items that combine the visual media and music items. When the visual media are viewed, the selected music item is also played.

Abstract: A system is described for maintaining synchrony of operations among a plurality of devices that have independent clocking arrangements. The system includes a task distribution device that distributes tasks to a synchrony group comprising a plurality of devices that are to perform the tasks distributed by the task distribution device in synchrony. The task distribution device distributes each task to the members of the synchrony group over a network. Each task is associated with a time stamp that indicates a time, relative to a clock maintained by the task distribution device, at which the members of the synchrony group are to execute the task.

Abstract: A video content controller includes a memory and a microprocessor. The memory is configured to store non-transitory computer-readable instructions and video data representing a temporally-varying scene having a plurality of scene-regions. The microprocessor adapted to execute the instructions to (i) receive a current stream-segment of a video stream corresponding to a first scene-region of the plurality of scene-regions during a current playback-time interval, (ii) download, to a memory, a first video segment of a tagged scene-region within the first scene-region, (iii) receive a first display-region signal indicating at least one of a selection and a feature of the first scene-region, (iv) combine the current stream-segment and the first video segment as a video signal, and (vi) transmit, in response to the first display-region signal, the video signal to a display device.

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: Provided are an apparatus and method for decoding audio, The method includes receiving a bitstream consisting of audio packets; decoding an audio packet included in the received bitstream, extracting a type of the decoded packet; obtaining system data from a packet of which system metadata corresponds to the extracted packet type; and transmitting the obtained system data to a system engine, wherein the system data includes at least one of information regarding a type of the system engine and a length of the system data.

Abstract: A method for controlling a scene sound effect and the electronic device includes after an device is turned on, starting a service having monitoring function; the device monitors the audio track of the device by means of the service having monitoring function, and determines whether the audio track includes an audio output; a mapping exists between the audio tracks and the applications in the device; if the device determines that the audio track includes an audio output, the device then determines, on the basis of the mapping, the application mapped to the audio track; the device obtains the scene sound effect corresponding to the application, and set the scene sound effect as the current sound effect of the device. The setting of the scene sound effects is autonomous, thereby simplifying operations and enhancing the utilization efficiency of the device while ensuring higher accuracy of scene sound effect.

Abstract: An example playback device includes a housing having a front side, a back side, a first end, and a second end. The playback device also includes an IR receiver positioned on the front side of the housing, a first IR emitter positioned on the back side of the housing and oriented such that a first IR signal emitted from the first IR emitter is directed toward the second end of the housing, and a second IR emitter positioned on the back side of the housing and oriented such that a second IR signal emitted from the second IR emitter a) is directed toward the first end of the housing and b) crosses the first IR signal emitted from the first IR emitter. The first and second IR emitters are communicatively coupled to the IR receiver within the housing and configured to retransmit an IR control signal received by the IR receiver.

Abstract: An example computing device is configured to perform functions including receiving calibration data corresponding respectively to a plurality of playback devices, where each playback device in the plurality of playback devices is located in a respective playback environment other than a first playback environment. The functions also include receiving playback device characteristic data respectively indicating at least one playback device characteristic for each playback device of the plurality of playback devices. The functions also include, based on at least the calibration data and the playback device characteristic data, determining updated playback device configuration information and transmitting data indicating the updated playback device configuration information to a given playback device located in the first environment.

Abstract: Example techniques involve controlling playback volumes. An example implementation includes displaying a group volume control of a synchrony group, the group volume control comprising a group volume indicator on a group volume slider and receiving, via the displayed group volume control, input data representing a selection of the group volume control. The implementation also includes displaying, concurrently with the group volume control, a first volume control comprising a first volume indicator on a first volume slider and a second volume control comprising a second volume indicator on a second volume slider. The implementation also includes receiving, via the displayed group volume control, input data representing an input to modify a first volume setting of the synchrony group to a second volume setting and transmitting instructions to cause the synchrony group to modify volume settings according to the second volume setting of the synchrony group.

Abstract: A method for controlling a scene sound effect and the electronic device includes after an device is turned on, starting a service having monitoring function; the device monitors the audio track of the device by means of the service having monitoring function, and determines whether the audio track includes an audio output; a mapping exists between the audio tracks and the applications in the device; if the device determines that the audio track includes an audio output, the device then determines, on the basis of the mapping, the application mapped to the audio track; the device obtains the scene sound effect corresponding to the application, and set the scene sound effect as the current sound effect of the device. The setting of the scene sound effects is autonomous, thereby simplifying operations and enhancing the utilization efficiency of the device while ensuring higher accuracy of scene sound effect.

Abstract: Apparatuses, arrangements, and methods therein for generation of comfort noise are disclosed. In short, the solution relates to exploiting the spatial coherence of multiple input audio channels in order to generate high quality multi channel comfort noise.

Abstract: An electronic computing device includes a processing unit and system memory. The system memory includes instructions which, when executed by the processing unit, cause the electronic computing device to: receive a communication containing audio data; automatically identify one or more sounds in the audio data of the communication that can be correlated with a geographical location; use the identification of the one or more sounds to identify the geographical location for the audio data; and verify an identity of an individual using the geographic location from the audio data.

Abstract: A system, method, and computer-readable medium are disclosed for performing a dynamic application optimization operation, comprising: instrumenting a plurality of system parameters of a client information handling system for monitoring; instructing a user to execute a particular application on the client information handling system; obtaining a plurality of samples of the plurality of system parameters; performing a machine learning operation using the plurality of samples of the plurality of system parameters, the machine learning operation training a machine learning model to generate a profile for the particular application and an operating mode of the particular application; applying the profile to the client information handling system to provide a new information handling system configuration, the new information handling system configuration optimizing the information handling system for the particular application.

Abstract: An example playback device includes a first interface for receiving a first audio signal from a first audio source; a second interface for receiving a second audio signal from a second audio source; and a processor configured to: cause the playback device to playback the second audio signal; determine that the first audio signal is present at the first interface; in response to determining that the first audio signal is present at the first interface, (i) cease playback of the second audio signal being played by the playback device and (ii) cause the playback device to playback the first audio signal; receive an instruction to stop the playback device from playing the first audio signal while the first audio signal is still present at the first interface; and arm the playback device such that a subsequent presence of the first audio signal at the first interface causes the playback device to play the first audio signal.

Abstract: Provided are a method for performing measurement and a device supporting the method. According to an embodiment of the present invention, a method comprises: receiving a reference signal (RS); receiving a measurement configuration including a set of filter coefficients from a network, wherein each filter coefficient in the set of filter coefficients is configured per periodicity of the RS; performing a measurement based on the RS; selecting a filter coefficient in the set of filter coefficients based on the periodicity of the RS; and reporting a result of the measurement filtered based on the selected filter coefficient.

Abstract: Systems and methods for audio output control are disclosed. Audio may be output via a speaker of a communal device associated with a first portion of an environment. A user may provide a user utterance indicating an intent to add another device in a second portion of the environment to the audio-output session, and/or an intent to move the audio-output session from the first device to the second device, and/or an intent to remove a device from an audio-output session. Based on this determined intent, audio-session queues may be associated and dissociated from devices and device states may be altered to effectuate the intent of the user utterance.

Abstract: An example playback device in a first zone of a media playback system receives a first indication that the first zone has been added to a first zone scene including a first preconfigured grouping of zones including the first zone and a second zone. The playback device receives a second indication that the first zone has been added to a second zone scene including a second preconfigured grouping of zones including the first zone and a third zone. After a given one of the first and second zone scenes has been selected for invocation, the playback device receives an instruction to operate in accordance with the given zone scene, and based on the instruction, begins operating in accordance with the given zone scene such that the playback device is configured to play back audio in synchrony with one or more other playback devices in the media playback system.

Abstract: A method is provided including determining a first listening point of a user in an audio space, wherein the audio space comprises at least the first listening point and a second listening point; rendering audio associated with at least one first audio object of the first listening point based on a position and/or orientation of the user relative to the first listening point; in response to receiving an indication of a switch from the first listening point to a second listening point, controlling the rendering of the audio based at least on signaling associated with at least the first audio object, wherein the signaling comprises one or more conditions indicating whether playback of the first audio object is to continue during and/or after the switch to the second listening point.

Abstract: An acoustic processing device performs acoustic processing in a space having a first listening position and a second listening position apart from the first listening position. The acoustic processing device includes a first input unit, a first output unit, and a signal processing circuit. The first input unit receives a first sound source signal to be reproduced at the first listening position. The first sound source signal includes a first left-channel signal and a first right-channel signal. The first output unit outputs the first sound source signal to a first speaker group including a first speaker disposed near a place of a head of a listener at the first listening position. The signal processing circuit draws out a first common signal common to both the first left-channel signal and the first right-channel signal and outputs the first common signal to the first speaker.

Abstract: A computer-implemented method for data transmission at reduced data volumes includes sampling a data stream in consecutive time intervals and generating consecutive data packages, calculating, via principal component analysis, projections of its data points onto a set of principal components, aligning, via dynamic time warping, the projected data points to respective projected data points of an earlier generated data package and calculating distances between the aligned data points, determining and comparing the maximum of the calculated distances to a threshold value, and transmitting the newly generated data package to a data sink and setting the newly generated data package as the earlier generated data package for the next aligning step if the maximum distance exceeds the threshold value or otherwise inhibiting the transmission and retaining the earlier generated data package for the next aligning step.

Abstract: Systems and methods provide for analyzing at least one feature of each segment of a plurality of segments of the audio stream to generate for each segment a prediction value indicating whether there is music in the segment, generating a probability value that there is music in the audio stream based on aggregating the prediction values of the plurality of segments, and causing the audio stream to be identified based on determining that the probability value that there is music in the audio stream meets a predetermined threshold.

Abstract: A user can utter a voice command in an environment where multiple audio playback devices are located to have audio output on a single device, or a predefined group of devices in a synchronized manner. In instances when the voice command uttered by the user does not specify a target for audio output, an implicit target selection algorithm can evaluate one or more criteria to determine an appropriate target for output of the audio corresponding to the voice command. An example criterion is met if a predetermined time period has lapsed since a last utterance was detected by a device in the environment. However, other criteria can be evaluated for determining a target output device(s).

Abstract: An audio system processes a multi-channel input audio signal into a stereo signal for left and right speakers, while preserving the spatial sense of the sound field of the input audio signal. The multi-channel input audio signal includes a first left-right channel pair including a left input channel and a right input channel, and a second left-right channel pair including a left peripheral input channel and a right peripheral input channel. Subband spatial processing may be applied to the first and second left-right channel pairs. A first crosstalk processing is applied to the first left-right channel pair to generate first crosstalk processed channels. A second crosstalk processing is applied to the second left-right channel pair to generate second crosstalk processed channels. A left output channel and a right output channel are generated from the first and second crosstalk processed channels. The crosstalk processing may include crosstalk cancellation or crosstalk simulation.

Abstract: An audio signal processing apparatus comprises: a plurality of channels each configured to perform signal processing on an input audio signal; an auxiliary signal processor (e.g., an effecter circuitry) that inputs an audio signal from one or more of the plurality of channels and performs auxiliary signal processing (e.g., effecter impartment processing) on the input audio signal; a plurality of channel strips each associated with one of the plurality of channels and including a plurality of manual controls for adjusting signal processing parameters of the one channel associated with the channel strip; and a controller that, in response to an assignment instruction given by a user, assigns at least one parameter related to the auxiliary signal processor (e.g., the effecter circuitry) to at least one of the plurality of manual controls included in at least one of the plurality of channel strips.

Abstract: A digital media search and presentation service will receive audio streams from multiple audio content sources and capture text segments, each of which corresponds to words in a snippet of the audio stream. When a client device receives a search request, the service will identify text segments that correspond to the search request. The service will select one or more of the text segments as candidate search results. The service will cause the client device to output user-selectable fields, each of which includes information corresponding to the selected text segments in a user-selectable interface by which a user may select the audio content sources for each of the text segments. When a user selects one of the fields, a media player of the client device will access and play audio content that the audio content source is transmitting at the time.

Abstract: An audio system presented herein includes a transducer array, a sensor array, and a controller. The transducer array presents audio content to a user. The controller controls the transducer array to adjust a level of tactile content imparted to the user via actuation of at least one transducer in the transducer array while presenting the audio content to the user. The audio system can be part of a headset.

Abstract: A method for controlling a scene sound effect and an electronic device are provided. The method comprises: after an electronic device is turned on, a monitoring service is started; the electronic equipment monitors creation and release of an audiotrack of the electronic equipment by means of the monitoring service to determine whether the audiotrack of the electronic equipment has audio output, the audiotrack of the electronic equipment has a mapping relationship with an application in the electronic equipment; in the case that the electronic equipment determines that the audiotrack of the electronic equipment has audio output, the electronic equipment determines the application having the mapping relationship with the audiotrack of the electronic equipment according to the mapping relationship; the electronic equipment acquires a scene sound effect corresponding to the application and sets a sound effect of the electronic equipment to be the scene sound effect.