Abstract: Described herein is a retrofit digital speaker system comprising two or more retrofitted speaker enclosures, each of the two or more retrofitted speaker enclosures (enclosures) comprising: at least one speaker; and an analog-and-digital interface adapted to receive digitally encoded audio signals, electrical power, and digital command signals, and wherein the digitally encoded audio signals, electrical power, and digital command signals are transmitted over existing two wire analog audio cables, and further wherein the two or more retrofitted speaker enclosures are wired in a daisy chain fashion, via the existing two wire analog audio cables.

Abstract: A method for audio processing includes receiving multiple electrical signals to be transmitted in parallel via multiple respective audio channels. Multiple respective weights are assigned to the multiple electrical signals, wherein at least two of the weights differ from one another. An instantaneous gain is calculated, to be applied to the multiple electrical signals. The instantaneous gain depends on (i) instantaneous amplitudes of the multiple electrical signals, and (ii) the weights assigned to the multiple electrical signals. The instantaneous gain is applied to the multiple electrical signals, and the multiple electrical signals are transmitted via the multiple respective audio channels.

Abstract: A control system may comprise a plurality of audio output devices (e.g., controllable speakers), and a remote control device having at least one button for selecting a preset, where the preset defines different commands for at least two of the audio output devices. The at least two audio output devices may be configured to be controlled according to the different commands (e.g., starting, pausing, or stopping playback, adjusting volume, etc.) in response to an actuation of the button of the remote control device. The control system may also comprise a load control device, such as a dimmer configured to control an intensity of a lighting load to a predetermined intensity in response to the actuation of the button of the remote control device to select the preset. The audio output device may be configured to play a feedback signal indicating an operational characteristic of the dimmer.

Abstract: An audio device includes an amplifier having a volume control circuit, a communication circuit configured to communicate with a host apparatus, a processor, and a memory having stored thereon instructions executable by the processor to cause the audio device to perform: setting a volume value in the volume control circuit; receiving, from the host apparatus through the communication circuit, an instruction value indicating a volume value to be set in the volume control circuit; comparing the received instruction value with a set value that is a volume value set in the volume control circuit; and sending an instruction signal based on a comparison result to the host apparatus through the communication circuit.

Abstract: Example pop noise removal systems and methods include detecting impulsive components in an input signal based on a signal-to-noise ratio spectrum of the input signal, and generating a spectral pop noise removal mask and applying the spectral pop noise removal mask to the input signal if impulsive components in the input signal are detected, the pop noise removal mask configured to suppress the impulsive components in the input signal, when applied.

Abstract: A sound management system for use by a user located within an environment includes a memory device for storing a selection of sounds. The sounds can be music and/or various “colors” of noise (e.g., white, pink, and brown). A controller is used to select one particular stored sound based on a measured biological condition of the user, such as stress or fatigue, or an environmental condition of the environment, such as ambient noise. According to one embodiment, the system is used in conjunction with a sit/stand desk and the sound selection is made in response to changes in the desk height. The selected sound is selected to help abate or mitigate distracting sounds in the environment, such as people talking. The selected sounds are played to the user through headphones worn by the user, or through nearby speakers.

Abstract: Volume leveler controller and controlling method are disclosed. In one embodiment, A volume leveler controller includes an audio content classifier for identifying the content type of an audio signal in real time; and an adjusting unit for adjusting a volume leveler in a continuous manner based on the content type as identified. The adjusting unit may configured to positively correlate the dynamic gain of the volume leveler with informative content types of the audio signal, and negatively correlate the dynamic gain of the volume leveler with interfering content types of the audio signal.

Abstract: An audio processor (1) includes a first filter coefficient calculator (31) that calculates a first filter coefficient so as to correspond to first gains for respective bands set by a user, a second filter coefficient calculator (32) that if values of third gains for respective bands of the first filter coefficient are greater than an absolute value of a second gain set by the user, calculates a second filter coefficient by limiting the values of the third gains for the respective bands to the amplitude value of the second gain, and a filtering unit (35) that filters an audio signal that has been transformed into a frequency-domain signal, using the second filter coefficient.

Abstract: A device for volume control for at least two audio sources includes a loudness analyzer and a volume regulator. The loudness analyzer is configured to analyze an audio signal of the first one and of the second one of the two audio sources over a time period, to determine a first and a second loudness value, respectively, as a function thereof, and to store same while associating them with the first and the second audio source. The volume regulator is configured to adapt the audio signal of the currently selected first and/or second one of the two audio sources in accordance with the corresponding first and/or second loudness value.

Abstract: Context-aware loudness control of audio content may include choosing from a plurality of loudness level models based on an audio reproduction device, measuring loudness level of the audio content based on the chosen loudness model, processing the real-time loudness measurement of the input audio signal to output real-time loudness level adjustment, processing a momentary loudness measurement of the input audio signal to output a momentary loudness level adjustment, processing a short-term loudness measurement of the input audio signal to output a short-term loudness level adjustment, adjusting the input audio signal based on the real-time, momentary, and short-term loudness level adjustments to output a post-processing input signal, measuring long-term loudness of the post-processing input signal to output a long-term loudness measurement, processing the long-term loudness measurement to output a post-processing level adjustment, and processing the real-time, momentary, short-term, and post-processing level a

Abstract: The present aspects relate to audio management for a web browser. For example, an electronic device such as a mobile device may enable a user to navigate to a webpage within a web browser. In some embodiments, the webpage may automatically initiate presentation of audio content. However, the user may wish to quickly silence the audio content that is playing without locating the media player controls of the audio content within the webpage. Accordingly, the present aspects provide audio control for a web browser. Specifically, an audio control affordance may be displayed at a noticeable and recognizable position or area in the web browser, and based on a determination that a webpage includes audio content and/or determination of playback of the audio content. The user may in turn readily adjust a property of the audio content.

Abstract: A microphone array included in a portable electronic device is used to generate various virtual studio microphones by combining one or more microphone signals to produce one or more acoustic pickup beams. An error is determined in a position of the microphone array relative to an audio source to be recorded. An interface is displayed to instruct a user on repositioning the microphone array relative to the instrument and the instrument is recorded using the repositioned microphone array.

Abstract: Volume limiting systems and methods are operable to limit volume output from media presentation devices. An exemplary embodiment detects a sound using a microphone, wherein the sound corresponds to an audio output of at least one controlled media presentation device, and wherein the microphone is remotely located from the at least one controlled media presentation device; compares a level of the detected sound with a predefined maximum volume limit; generates a volume output limit command in response to the detected sound exceeding the predefined maximum volume limit; and communicates the volume output limit command to the media presentation device. The media presentation device then reduces a volume level of its audio output. In some instances, volume may be limited during user specified periods.

Abstract: Loudness control systems or methods may normalize audio signals to a predetermined loudness level. If the audio signal includes moderate background noise, then the background noise may also be normalized to the target loudness level. Noise signals may be detected using content-versus-noise classification, and a loudness control system or method may be adjusted based on the detection of noise. Noise signals may be detected by signal analysis in the frequency domain or in the time domain. Loudness control systems may also produce undesirable audio effects when content shifts from a high overall loudness level to a lower overall loudness level. Such loudness drops may be detected, and the loudness control system may be adjusted to minimize the undesirable effects during the transition between loudness levels.

Abstract: A method (600) for decoding an encoded audio signal (102) is described. The encoded audio signal (102) comprises a sequence of frames. Furthermore, the encoded audio signal (102) is indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. Different subsets of DRC profiles from the plurality of DRC profiles are comprised within different frames of the sequence of frames, such that two or more frames of the sequence of frames jointly comprise the plurality of DRC profiles.

Abstract: A video game controller can automatically operate as a video game controller with audio capability and a video game controller without audio capability based upon a signal received via a switch in communication with an audio jack of the video game controller when a headset is coupled to the audio jack. Connection of the headset to the controller automatically causes video game audio to be communicated from the video game console to the headset via the controller. The video game audio is communicated at a default volume level to the headset upon coupling of the headset to the video game controller. The volume level of the audio communicated to the headset can be adjustable up or down via a directional pad of the video game controller while an audio control button on the video game controller is pressed.

Abstract: Dynamic loudness equalization of received audio content in a playback system, using metadata that includes instantaneous loudness values for the audio content. A playback level is derived from a user volume setting of the playback system, and is compared with a mixing level that is assigned to the audio content. Parameters are computed, that define an equalization filter that is filtering the audio content before driving a speaker with the filtered audio content, based on the instantaneous loudness values and the comparing of the playback level with the assigned mixing level. Other embodiments are also described and claimed.

Abstract: The present disclosure relates to a signal processing device, a signal processing method, and a program that allows for generating an input signal suitable for a multi-way speaker. A band dividing unit divides an audio signal into signals in a plurality of bands corresponding to respective bands of a plurality of speaker units of the multi-way speaker. A filter processing unit performs wave front synthesis filter processing on each of the audio signals in the respective bands having been divided into. The audio signal in each of the bands after the wave front synthesis filter processing is supplied to the speaker unit of the corresponding band in the multi-way speaker. The present disclosure is applicable, for example, to a signal processing device or other devices.

Abstract: A method suppresses feedback in a hearing instrument. A microphone generates an input signal and a loudspeaker generates an acoustic signal which is partially fed back to the microphone via an acoustic feedback path. An intermediate signal is generated along a main signal path depending on the input signal, and an output signal is formed on the basis of the intermediate signal. A voice activity of a user is monitored, a transfer function of an electro-acoustic closed signal loop formed by the main signal path and the feedback path is estimated, that, depending on the transfer function of the closed signal loop and the voice activity of the user, the intermediate signal is decorrelated to form the output signal. A compensation signal is generated from the output signal and is fed to the input signal for feedback compensation, and the output signal is fed to the loudspeaker for reproduction.

Abstract: To provide a technology capable of changing an output characteristic of music data more efficiently with a simpler operation irrespective of whether or not the music data is being played back, an audio controller (2) identifies a usage environment of a wireless speaker (1) from picked-up image data on an installation room of the wireless speaker (1). Then, an output characteristic suitable for the identified usage environment is selected, and the selected output characteristic is set as an output characteristic of music data to be output from the wireless speaker (1).

Abstract: A noise control method and device are provided that relate to the field of noise control. A noise control method includes: acquiring noise information of an ambient environment; generating a noise control message including the noise information, the noise control message being used to notify other devices to adjust a volume; and sending the noise control message to the other devices. Another noise control method includes: receiving, by a device, a noise control message from an external device; judging, according to the noise control message, whether the device is necessary to perform volume adjustment; and adjusting a volume according to a volume adjustment policy if the device is necessary to perform volume adjustment. The noise control method and device in the embodiments of the present application easily and quickly realize control over ambient noise, thereby improving user experience.

Abstract: Methods and apparatuses for addressing open space noise are disclosed. In one example, a method for masking open space noise includes receiving a sensor data from a sensor arranged to monitor an open space over a time period. The method includes generating a predicted future noise parameter in the open space at a predicted future time from the sensor data. The method further includes adjusting a sound masking noise output from a loudspeaker prior to the predicted future time responsive to the predicted future noise parameter.

Abstract: Some embodiments provide a multi-listener stereo image array which provides multiple separate stereo images of audio content to each of multiple listeners while mitigating interference of audio signals which provide separate stereo images to separate listeners. The array can include sensors which can monitor the environment and can identify relative positions of various listeners and can further control the audio signal patterns generated by the drivers of the array to propagate towards positions associated with particular listeners and be at least partially restricted from propagating towards positions associated with other listeners.

Abstract: Systems, devices, and methods are described herein for adjusting a relationship between dialog and non-dialog signals in an audio program. In an example, information about a long-term dialog balance for an audio program can be received. The long-term loudness dialog balance can indicate a dialog-to-non-dialog loudness relationship of the audio program. A dialog loudness preference can be received, such as from a user, from a database, or from another source. A desired long-term gain or attenuation can be determined according to a difference between the received long-term dialog balance for the audio program and the received dialog balance preference. The long-term gain or attenuation can be applied to at least one of the dialog signal and the non-dialog signal of the audio program to render an audio program that is enhanced according to the loudness preference.

Abstract: Systems and methods for adjusting audio levels in a plurality of audio signals are disclosed. One or more input signals containing one or more audio signals are depacketized. The audio signals are analyzed to determine if audio levels of the signals correspond to a desired audio level for at least some of the signals. The audio levels of signals that do not correspond are adjusted. Packets that do not require adjustment and the resulting adjusted packets are combined to form one or more output signals corresponding to the one or more input signals.

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: Provided is a signal processing device including a display control unit for causing a display to display an image corresponding to a specified place, a sound-collection-signal input unit for inputting a sound collection signal of a sound collection unit that collects a user sound produced with microphones surrounding the user, an acoustic-signal processing unit for performing a first acoustic-signal process for reproducing a sound field where the user sound is sensed as if the sound were echoing in the place on the signal input by the sound-collection-signal input unit, based on a first transfer function measured in the place to indicate how a sound emitted on a closed surface inside the place echoes in the place and then is transferred to the closed-surface side, and a sound-emission control unit for causing a sound based on the processed signal to be emitted from speakers surrounding the user.

Abstract: An audio encoding device is described herein. The audio encoding device includes a compressor that is based on a nonlinear filter. In particular, the nonlinear filter may be selected from the class of edge-preserving smoothing filters, which avoids common artifacts of conventional compressors. Edge-preserving smoothing filters have been used in image processing algorithms for their de-noising properties while preserving edges in the image. These properties are useful for audio compression because macro-dynamic loudness changes can be tracked precisely while micro-dynamic loudness changes can be ignored for the compression. Due to these advantages, more aggressive compression can be achieved with less distortion.

Abstract: The present invention relates to a head mounted electronic device, including a headphone and a display portion. The headphone includes a C-shaped elastic belt including two coupling ends; two movable members, each being rotatably coupled to a corresponding coupling end along a first axis; and two speakers, each being rotatably coupled to a corresponding movable component along a second axis. The display portion includes a main body, configured to produce and project images outwardly; and two coupling components, coupled to two opposite ends of the main body and each being rotatably coupled to a corresponding speaker along a third axis. Each of the two speakers is configured to rotate along the first axis and the second axis defined by a corresponding movable member, respectively, when the speaker rotates along its respective third axis in response to the movement of the display portion.

Abstract: A computer-implemented method includes identifying a first set of utterances from a plurality of utterances. The plurality of utterances is associated with a conversation and transmitted via a plurality of audio signals. The computer-implemented method further includes mining the first set of utterances for a first context. The computer-implemented method further includes determining that the first context associated with the first set of utterances is not relevant to a second context associated with the conversation. The computer-implemented method further includes dynamically muting, for at least a first period of time, a first audio signal in the plurality of audio signals corresponding to the first set of utterances. A corresponding computer system and computer program product are also disclosed.

Abstract: Systems and methods for leveling loudness variation in an audio signal are described. Embodiments use both a perceptual leveling algorithm and a standards-based loudness measure together to minimize audio process artifacts and ensure that the measured loudness of the processed audio is close to a required measure, according to a particular standard measurement of loudness. These systems and methods can be used either offline or in real-time.

Abstract: A user device receives a first media item that is associated with a second media item. The device determines that a playback state for the device indicates that the device is paired with an alternative playback device. The device sends the playback state in a request for the second media item, receives the second media item in a first format for playback on the user device and in a second format for playback on the alternative playback device. The device determines whether the playback state of the device is the same. The device displays a first graphical representation of the second media item in the first format on the user device if the user device is no longer paired with the alternative playback device and displays a second graphical representation of the second media item in the second format on the user device if the user device is still paired with the alternative playback device.

Abstract: In one aspect, when it is detected that a viewer of an audio video program is interrupted by something, the program may be increased in volume, paused, etc.

Abstract: Image data relating to real-world objects or persons is collected from a scene while collecting audio data relating to the real-world objects or persons from the same scene. The audio data is used to derive sound objects corresponding to the real-world objects or persons. The image data is used to derive video objects corresponding to the real-world objects or persons. Based on the sound objects and the video objects, candidate salient objects are generated. A salient object is selected from among the candidate salient objects. Perceptual enhancement operations are performed on the selected salient object.

Abstract: An earplug unit includes a signal processing unit. The earplug unit is inserted into an external auditory canal and then used. In the earplug unit, a speaker is arranged at an eardrum side, and a microphone is arranged at a side opposite to the eardrum side. The microphone collects a sound in which a user's head shape or auricle characteristic is reflected. An output signal of the microphone is amplification-processed by a volume adjusting unit and then supplied to the speaker. An amplified sound is output from the speaker. A frequency characteristic is smoothed in a band larger than at least an audible frequency band in a system from the microphone to the speaker.

Abstract: The present application refers to a head-mounted electronic device including a headphone and a display portion. The headphone includes a C-shaped elastic belt and has two connection ends, and two sound generators connected to the connection ends respectively. The display portion includes a main body configured for generating and projecting images, and two connection members which are connected to opposite ends of the main body respectively and are rotatably connected to the sound generators respectively. Each connection member arranged between the corresponding sound generator and the main body defines a notch. When the display portion is rotated relative to the headphone, possible distortion of the display portion is limited to the connection member where the notch is defined, protecting the main body and the sound generator from deformation, and thus facilitating the display portion to rotate relative to the headphone.

Abstract: The audio interrupter device (2) interfaces between any audio source (1) and headphones (3) to alert the user to external environmental sounds that exceed a threshold by temporarily halting audio transmission. The threshold is adjustable either through an automatic adjustment option that bases the threshold on the ambient sound level or through manual sensitivity adjustment. The device includes an audio plug (11) and an audio jack (12) that are connected to analog switches (10) that have the ability to connect and disconnect their corresponding conductors. An audio signal from a microphone (4) is amplified and input to a microcontroller (7), which compares the audio signal sample value to the sound interrupting threshold. If the audio signal sample exceeds the threshold, the analog switches disconnect the corresponding conductors of the audio plug and audio jack.

Abstract: A system and apparatus for noticing and creating relational settings, actions, profiles, and tasks by tying resources to a location based on user behavior.

Abstract: A method includes the steps of calculating a power spectrum from an auditory stimulus, filtering the power spectrum to obtain an effective power spectrum, calculating an intensity pattern from the effective power spectrum, calculating a median intensity pattern from the intensity pattern, determining an initial set of pruned detector locations, examining the initial set of pruned detector locations to determine an enhanced set of pruned detector locations, and calculating an excitation pattern from the effective power spectrum using the enhanced set of pruned detector locations. By determining the enhanced set of pruned detector locations from the initial set of pruned detector locations and computing the excitation pattern therefrom, the computational complexity of the above method can be significantly reduced when compared to conventional approaches while maintaining the accuracy thereof.

Abstract: An audio output device which outputs an audio signal that offsets a direct sound in the first speaker to offset the direct sound generated in the second speaker is provided. The audio output device according to various exemplary embodiments includes a first speaker configured to output a first audio signal, the first speaker being disposed at a first side of the audio output device, a second speaker configured to output a second audio signal, the second speaker being disposed at a second side of the audio output device, and an audio signal processor configured to control the first speaker to output the third audio signal to remove the element provided in the same direction with the first side among the second audio signal.

Abstract: The present invention relates to a mobile terminal and controlling method thereof. After setting information related to an executing operation of content has been saved, an item for executing the content is displayed. When the item is selected, the content is run in a setting state corresponding to the setting information. Particularly, operating state information on a specific operation timing point of content is registered as a shortcut and saved. An item of the shortcut is displayed on a screen. When the item is selected, the content is directly executed from a timing point corresponding to the operating state information. Accordingly, when the content is executed, the content can be executed from a user-desired part by skipping a part unnecessary for a user.

Abstract: Volume limiting systems and methods are operable to limit volume output from media presentation devices. An exemplary embodiment detects a sound using a microphone, wherein the sound corresponds to an audio output of at least one controlled media presentation device, and wherein the microphone is remotely located from the at least one controlled media presentation device; compares a level of the detected sound with a predefined maximum volume limit; generates a volume output limit command in response to the detected sound exceeding the predefined maximum volume limit; and communicates the volume output limit command to the media presentation device. The media presentation device then reduces a volume level of its audio output. In some instances, volume may be limited during user specified periods.

Abstract: The present disclosure relates to an earphone apparatus and method. The earphone apparatus includes an earpiece, including a speaker, configured for arrangement relative to a user's ear for listening to audio from the speaker, a bio-sensor, a motion sensor, and a controller configured to determine an activity state of the user based on aggregated sensor data from the bio-sensor and the motion sensor over time, and to control audio output to the user based on the determined activity state.

Abstract: The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Abstract: The application relates to a loudspeaker system including an input unit providing an electric input audio signal; an equalization unit for modifying said electric input audio signal in dependence on frequency and to provide an equalized electric audio signal according to a predefined equalization function, a loudspeaker unit for converting said equalized electric audio signal to an acoustic output sound signal, and a user interface for modifying a volume level of said output sound signal in a multitude (N) of steps V0, V1, . . . , VN. The application further relates to a method communication device comprising the loudspeaker system and to its use. The present application provides an improved loudspeaker system in which the equalization unit is configured to apply a specific equalization function EQ0, EQ1, . . . , EQN to the electric input audio signal in each of said multitude of steps V0, V1, . . . , VN of the volume level.

Abstract: A sound field control system SY according to the present invention is provided with: a signal supply unit 210 which supplies a test signal sequence to a speaker; and an analysis unit 150 which picks up a sound including a test sound sequence emitted from a speaker based on a test signal sequence using a microphone, and which determines the sound pickup time of an initial test sound among test sounds included in the test sound sequence, wherein the analysis unit 150 detects the initial sound from the sound pickup signal picked up using the microphone, determines whether a sound is detected after the elapse of a sound emission interval, from the sound pickup time of the detected initial sound, between the initial test sound defined by the test signal sequence and an n-th (where n is an integer such that n?2) test sound emitted from the same speaker as the initial test sound, and, upon determining that the sound has been detected, determines the sound pickup time of the initial sound as the sound pickup time of

Abstract: The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Abstract: The portable music studio is a modular apparatus that records and mixes music in real time. The portable music studio comprises a dock and a controller. The dock is a switching device that receives a first set of one or more audio signals from a first set of one or more audio sources and routes the each of the received audio signals as a first multichannel audio signal to the controller. The controller is a logic device that independently records each channel of the multichannel signal and directly receives and records a second set of one or more audio sources. The controller mixes this plurality of recorded audio sources. The portable music studio further generates musical audio input directly from the controller. The controller is separable from the dock to allow for recording, mixing and making audio files at a remote location.

Abstract: Signals in an automobile audio system having at least two near-field speakers located close to an intended position of a listener's head are adjusted such that in a first mode, audio signals are distributed to the near-field speakers according to a first filter that causes the listener to perceive a wide soundstage, and in a second mode, the audio signals are distributed to the near-field speakers according to a second filter that causes the listener to perceive a narrow soundstage. A user input of a variable value is received and, in response, distribution of the audio signals is transitioned from the first mode to the second mode, the extent of the transition being variable based on the value of the user input.

Abstract: Systems and methods for leveling loudness variation in an audio signal are described. Embodiments use both a perceptual leveling algorithm and a standards-based loudness measure together to minimize audio process artifacts and ensure that the measured loudness of the processed audio is close to a required measure, according to a particular standard measurement of loudness. These systems and methods can be used either offline or in real-time.