Abstract: Techniques associated with conforming local and remote media characteristics data to target media presentation profiles are described, including a media characteristic configuration module comprising an audio characteristic manager configured to modify an audio portion of media content data received from a source to conform the audio portion to a target audio characteristic indicated in a target configuration profile, and to modify an audio portion of other media content data received from another source to conform the audio portion to the target audio characteristic indicated in the target configuration profile, a communication facility configured to receive media content data from different sources using different data communication protocols, and a storage device configured to store target configuration profiles and data associated with media content.

Abstract: A real time messaging platform identifies an audio snippet of a time-based media (TBM) event. The messaging platform maintains a real time repository of concurrently broadcasting TBM events as well as a historical repository of previously broadcast TBM events. These repositories contain acoustic fingerprints of their respective TBM events. The messaging platform matches an acoustic fingerprint of the audio snippet with one of the stored acoustic fingerprints to identify the TBM event in the recorded snippet. To identify the TBM event, the messaging platform matches multiple overlapping reference audio segments of the reference audio stream with multiple test audio segments of the audio snippet. This allows the platform to account for time delays between the test and reference audio segments that would otherwise hinder the matching process.

Abstract: Audio processing method and audio processing apparatus, and training method are described. According to embodiments of the application, an accent identifier is used to identify accent frames from a plurality of audio frames, resulting in an accent sequence comprised of probability scores of accent and/or non-accent decisions with respect to the plurality of audio frames. Then a tempo estimator is used to estimate a tempo sequence of the plurality of audio frames based on the accent sequence. The embodiments can be well adaptive to the change of tempo, and can be further used to tracking beats properly.

Abstract: A method for outputting a sound in an electronic device is provided. The method includes generating a loop module corresponding to a loop element; displaying the generated loop module; and outputting a sound included in the displayed loop module.

Abstract: Extracting a common signal from multiple audio signals may include summing a first signal and a second signal to obtain a first+second signal; subtracting the second signal from the first signal to obtain a first?second signal; transforming the first+second signal and the first?second signal to frequency domain representations; calculating absolute value of the frequency domain representations of the first+second signal and the first?second signal; subtracting the absolute value of the frequency domain representation of the first?second signal from the absolute value of the frequency domain representation of the first+second signal to obtain a difference signal; multiplying the difference signal by the frequency domain representation of the first+second signal to obtain a product signal; dividing the product signal by the absolute value of the frequency domain representation of the first+second signal to obtain a frequency domain representation of the common signal; and transforming the frequency domain repre

Abstract: A case for a pair of wireless earbuds includes a lid and a lid sensor configured to generate a detect signal when the lid is moved from a closed position to an open position. The case further includes circuitry configured to, in response to receiving the detect signal, send a signal to the wireless earbuds to turn them on and to initiate a pairing sequence between the wireless earbuds and an electronic device.

Abstract: In general, techniques are described for identifying distinct audio objects from spherical harmonic coefficients (which may also be denotes as higher order ambisonic coefficients). A device comprising one or more processors may perform the techniques so as to identify the distinct audio objects from the spherical harmonic coefficients (SHC) associated with the audio objects based on a directionality determined for one or more of the audio objects.

Abstract: A microphone includes a microelectromechanical system (MEMS) circuit and an integrated circuit. The MEMS circuit is configured to convert a voice signal into an electrical signal, and the integrated circuit is coupled to the MEMS circuit and is configured to receive the electrical signal. The integrated circuit and the MEMS circuit receive a clock signal from an external host. The clock signal is effective to cause the MEMS circuit and integrated circuit to operate in full system operation mode during a first time period and in a voice activity mode of operation during a second time period. The voice activity mode has a first power consumption and the full system operation mode has a second power consumption. The first power consumption is less than the second power consumption. The integrated circuit is configured to generate an interrupt upon the detection of voice activity, and send the interrupt to the host.

Abstract: A portable audio networking system is provided that includes master devices and slave devices in communication over cables adapted to simultaneously transport a digital audio signal, DC power, and a data signal. Master devices are configured to receive a digital audio signal, distribute power, and transceive a data signal, and slave devices are configured to transmit the digital audio signal to master devices, receive the power from master devices, and transceive the data signal with master devices. Slave devices can include wireless audio receivers that receive an RF signal containing an audio signal from wireless audio transmitters. Master devices can include gateway interconnection devices that act as hubs and routers for the system. The reduction in the number of cables compared to traditional systems results in less weight for crew members to carry and easier setup of the system. Remote control of wireless audio transmitters is also enabled.

Abstract: A system includes a video display screen configured to display selected portions of video content on a main portion of the video display screen and on one or more extended portions of the video display screen. The system also includes an audio portion configured to dynamically create one or more sound radiating speaker elements at one or more spatially selected locations with respect to the displayed selected portions of the video content. A method is also provided that provides the video display screen and the audio portion of the system.

Abstract: Various methods and systems for estimating a room impulse response between an audio source and an array of microphones are described. In one example, a method includes receiving audio signals at a microphone of an array of microphones. The audio signals correspond to each of the microphones in the array of microphones. The method also includes determining a room impulse response in a subspace that is compatible with a geometry of the array of microphones based on the received audio signals.

Abstract: A media-playback device includes: a media-output device that plays media content items; a physiological measurement device programmed to measure at least one physiological measurement of a user of the media-output device; and a physiological control engine configured to: identify a current physiological measurement for the user; and cause the media-output device to modify playback of the media content items based upon the current physiological measurement.

Abstract: A sound source locator efficiently employs a distributed physical or logical microphone array to determine a location of a source of a sound. In some instances, the sound source locator is deployed in an augmented reality environment. The sound source locator detects sound at a plurality of microphones, generates a signal corresponding to the sound, and causes attributes of signal as generated at the plurality of microphones to be stored in association with the corresponding microphone. The sound source locator uses these stored attributes to identify multiple groups of the plurality of microphones from which delays between the times the signal is generated can be used to compute the location of the source of the sound.

Abstract: A microphone array processing system and method carried out in the system. In one embodiment, the system includes: (1) a beamformer configured to perform adaptive beamforming on gain-compensated signals received from a plurality of microphones, the adaptive beamforming including dynamic range compression and diagonal loading of a sample correlation matrix based on order statistics and (2) a postfilter configured to receive an output of the beamformer and reduce noise components remaining from the beamforming.

Abstract: Embodiments of the present invention disclose a method, computer program product, and system for real-time determination of attentiveness of an audience in a room to a speaker. A computer determines amounts of wireless activity of a computing device within the room of the audience over time during the presentation. Based in part on the amounts of wireless activity of the computing device, the computer determines and initiates display of measures of attentiveness of the audience over the time during the presentation, to display changes in the measures of attentiveness of the audience to the speaker during the presentation. In another embodiment, the computer determines amounts of attentiveness of the audience within the room over time during the presentation utilizing sensors located throughout the room.

Abstract: Certain implementations of the disclosed technology may include systems and method for handling application notifications. According to an example implementation, a method is provided. The method can include receiving a plurality of audio feeds from a respective plurality of applications, and determining a priority status for one or more of the respective applications. Based on the determined priority status, the method may further include determining a first prioritized audio feed and a second prioritized audio feed from the plurality of applications. The method includes detecting, in a signal associated with the first prioritized audio feed, a signal gap having a gap start. The method includes modifying, based at least in part on the signal of the first prioritized audio feed, one or more parameters associated with at least the second prioritized audio feed, and outputting, to an output device, at least a portion of the first prioritized audio feed.

Abstract: An apparatus and a method for outputting an audio signal in a portable terminal are provided. The portable terminal for outputting an audio signal includes an ear jack, a modem, and an amplifier. The ear jack connects the portable terminal with one of an earphone and an external apparatus. The modem detects whether one of the earphone and the external apparatus is connected to the ear jack. When connection of the external apparatus is detected, the amplifier high power-amplifies the audio signal and outputs the same to the external apparatus according to a control of the modem.

Abstract: An adaptive gain control system and related operating method for digital audio samples is provided. The method is suitable for use with a digital media encoding system that transmits encoded media streams to a remotely-located presentation device such as a media player. The method begins by initializing the processing of a media stream. Then, the method adjusts the gain of a first set of digital audio samples in the media stream using a fast gain adaptation scheme, resulting in a first group of gain-adjusted digital audio samples having normalized volume during presentation. The method continues by adjusting the gain of a second set of digital audio samples in the media stream using a steady state gain adaptation scheme that is different than the fast gain adaptation scheme, resulting in a second group of gain-adjusted digital audio samples having normalized volume during presentation.

Abstract: A control system for selectively operating a sound system provided in a walk-around character costume. e.g., to play audio files in an order defined by a dialog tree that can be navigated by a performer wearing the costume. The control system includes a trigger mechanism provided in the costume operable by a performer wearing the costume to generate and transmit at least one trigger signal to navigate through a dialog tree. A controller receives the trigger signal and transmits a control signal causing the sound system to playback pre-recorded dialog. The system includes a trigger processing module run by the controller for processing the trigger signal and, in response to such processing, for navigating pre-recorded audio for a show to select the set of pre-recorded dialog. The trigger mechanism includes a first and a second pressure sensor in a thumb and index finger of a glove of the costume.

Abstract: In one embodiment, a multimedia audio inserter removes the left and right audio channel data from a multimedia signal, for example an HDMI® signal and replaces it with an alternate audio signal, which may be the center channel audio signal provided by a digital surround sound receiver. In exemplary applications, the resulting, modified multimedia signal may be provided to a digital television that is part of a surround sound system, enabling the television to play the center channel signal through its speakers. Optionally, the multimedia audio inserter may provide a copy of the multimedia signal at a multimedia feedthrough output. In exemplary applications, the multimedia feedthrough output may be used to play left and right channel audio, present in the HDMI® signal, through dedicated left and right speakers.