Abstract: A device including a magnetic structure, the magnetic structure having a substrate adjacent surface and a second, opposing surface, the magnetic structure having a near field transducer (NFT), wherein the NFT includes gold or an alloy thereof, and is positioned at the second surface an overcoat structure; and a film structure, the film structure positioned between the magnetic structure and the overcoat structure, the film structure having a total thickness of not greater than about 100 ?, and the film structure including: a first interfacial structure having a first and a second opposing surface; a second interfacial structure having a first and a second opposing surface; and an intermediate structure wherein the first surface of the first interfacial structure is positioned adjacent the NFT of the magnetic structure, and the second surface of the second interfacial structure is positioned adjacent the overcoat structure, and the intermediate structure is positioned between the first interfacial structure an

Abstract: A terminal includes: a communication unit configured to communicate with a hands-free device and an audio device provided in a place in which speech is produced; and a control unit configured to perform control such that: i) when a filter generation mode is selected, a test signal is generated, ii) when a microphone signal transmitted from the hands-free device is received, the received microphone signal and the generated test signal are mixed, and the mixed signal is transmitted to the audio device, and iii) when a feedback signal transmitted from the hands-free device is received, a sound feedback removing filter is generated based on the received feedback signal, and the generated sound feedback removing filter is transmitted to the audio device.

Abstract: In one example, a presentation sound box is provided that includes a first box portion, and a second box portion that is movable relative to the first box portion. A sound module is disposed within the box and includes a speaker, a processor, and memory accessible by the processor. The presentation sound box further includes a power source operable to drive the speaker and processor, a trigger device operable to connect and disconnect the power source and sound module, and an amplifying cover configured and arranged to at least partly enclose the speaker.

Abstract: An echo cancellation system performs audio beamforming to separate audio input into multiple directions (e.g., target signals) and generates multiple audio outputs using two acoustic echo cancellation (AEC) circuits. A first AEC removes a playback reference signal (generated from a signal sent a loudspeaker) to isolate speech included in the target signals. A second AEC removes an adaptive reference signal (generated from microphone inputs corresponding to audio received from the loudspeaker) to isolate speech included in the target signals. A beam selector receives the multiple audio outputs and selects the first AEC or the second AEC based on a linearity of the system. When linear (e.g., no distortion or variable delay between microphone input and playback signal), the beam selector selects an output from the first AEC based on signal to noise (SNR) ratios. When nonlinear, the beam selector selects an output from the second AEC.

Abstract: MEMS microphone and vibration sensor dies and packages are described. In an embodiment, a MEMS microphone and vibration sensor die includes a die substrate, a MEMS microphone on the die substrate and a MEMS vibration sensor on the die substrate. The MEMS vibration sensor may include a plurality of beams with different proof masses corresponding to different resonant frequencies, wherein the different proof masses comprise a same material as the die substrate.

Abstract: A limiter for an audio system prevents loud audio signals that exceed a threshold from being output. Output of the audio signals are delayed. When a loud signal exceeds the threshold, the gain applied to the delayed signals is gradually reduced so that by the time the loud signal reaches the output, the gain is at a level that reduces the loud audio signal to be within the threshold. Thereafter the gain is gradually restored to normal over a longer period of time than the audio signals are delayed.

Abstract: Methods for modeling a loudspeaker having a passive radiator include applying a stimulus signal to a speaker within the cabinet, wherein the stimulus is applied over a frequency range. The sound pressure level (SPL) in the cabinet is measured as a function of frequency during application of the stimulus signal. At least one coefficient based on the measured SPL is derived, wherein at least one passive radiator parameter is a function of the at least one coefficient.

Abstract: An audio capture system for a sports camera includes at least one “enhanced” microphone and at least one “reference” microphone. The enhanced microphone includes a drainage enhancement feature to enable water to drain from the microphone more quickly than the reference microphone. A microphone selection controller selects between the microphones based on a microphone selection algorithm to enable high quality in conditions where the sports camera transitions in and out of water during activities such as surfing, water skiing, swimming, or other wet environments.

Abstract: An audio device and a position recognition method of the audio device are disclosed. According to an example embodiment, an audio device includes at least one infrared (IR) transceiver and a controller configured to recognize whether the audio device is positioned to a left or right of another audio device in response to a signal received by the at least one IR transceiver.

Abstract: An echo cancellation system that uses a combined reference signal using a playback reference signal and an adaptive reference signal. The playback reference signal is generated from a playback signal sent to a loudspeaker and the adaptive reference signal is generated using beamforming on microphone inputs corresponding to audio received from the loudspeaker. The system applies a low pass filter to the playback reference signal and applies a high pass filter to the adaptive reference signal to generate the combined reference signal. The system may remove the combined reference signal from target signals associated with the microphone inputs to isolate speech included in the target signals.

Abstract: A control mechanism may control the height and/or position of a read/write head configured to interact with a rotating information storage surface. A computation unit may compute a detected topography from gap measurements using a dynamic filter including a model of read/write head dynamics. A sensor may detect gap measurements of a side read/write track while the read/write head is interacting with a current read/write track. A memory may store the detected topography. The control mechanism may adjust the height of the read/write head based on the detected and/or stored topography. The control mechanism may be a reactionless control mechanism configured to apply a counterforce to offset movements of the read/write head and/or a slider.

Abstract: A failure detection system includes an omnidirectional microphone array device having a plurality of microphone elements and a directivity control device that calculates a delay time of a voice propagated from a sound source to each microphone element and forms a directivity of the voice using the delay time and the voice collected by the omnidirectional microphone array device, and detects a failure of the microphone element. A smoothing unit calculates an average power of one microphone element. An average calculator calculates a total average power of a plurality of usable microphone elements included in the omnidirectional microphone array device. A comparison unit compares whether or not a difference between the average power and the total average power exceeds a range of ±6 dB, and determines whether the microphone element is in failure based on the comparison result.

Abstract: The present disclosure describes a system using an audio mixer for managing endpoint parameters of an input audio device and an output audio device. The system includes a data terminal, multiple endpoints, and the audio mixer, where the data terminal provides multiple data files including configuration parameters of endpoints. The endpoints include a first endpoint and a second endpoint. The audio mixer is configured to receive the data files from the data terminal, process the data files to obtain a first set of configuration parameters corresponding to the first endpoint and a second set of configuration parameters corresponding to the second endpoint, convert the first set and the second set of configuration parameters into a predefined common data format, and organize the predefined common data structure into a generic data structure.

Abstract: A sound modification system is disclosed that includes one or more audio sensors coupled with a processing device and arranged to detect sounds within an environment, and one or more audio output devices coupled with the processing device. The processing device operates to generate an audio signal based on sounds detected from within one or more selected directions for sound modification within the environment, and to output the generated audio signal using the one or more audio output devices. The output generated audio signal combines with the detected sounds to produce a desired sound modification.

Abstract: A method for noise cancellation. The method includes providing a sound signal output device having a listening device output, a microphone input, and signal processing circuitry. The signal processing circuitry is adapted for causing the sound signal output device to perform in a noise-cancellation mode, including making available to the sound signal output device, at the microphone input, a noise-containing signal which is representative of ambient acoustical noise, deriving a noise-cancelling signal from the noise-containing signal, and making available to the personal listening device, at the listening device output, the noise-cancelling signal.

Abstract: Stereophonic sound is reproduced by acquiring image depth information indicating a distance between at least one object in an image signal and a reference location, acquiring sound depth information indicating a distance between at least one sound object in a sound signal and a reference location based on the image depth information, and providing sound perspective to the at least one sound object based on the sound depth information.

Abstract: The application relates to a hearing system comprising a hearing device (HD), and a programming device (PD), and a feedback path analyzer (FPA) for providing an estimate of a feedback path from an output transducer to an input transducer of the hearing device. The hearing device (HD) and the programming device (PD) each comprises a programming interface (PI, PD-PI) allowing the exchange of data between the devices. The feedback path analyzer (FPA) is located in the programming device (PD), and the hearing device (HD) comprises respective input (IBUF) and output (OBUF) buffers for storing corresponding input and output time segments of the electric input signal or a signal derived therefrom and of the processed output signal or a signal derived therefrom, respectively.

Abstract: A gain control system is responsive to a user gain setting and to an effective signal level to calculate and adaptively vary an output gain to avoid output clipping. The effective signal level is calculated by smoothing the energy of an input audio signal in accordance with selected attenuation factors. In times during which fast transient sounds are occurring, attenuation factors corresponding to relatively low levels of attenuation are selected. In times during which fast transient sounds are not detected, attenuation factors corresponding to relatively higher levels of attenuation are selected. The effective signal level is held at a constant level during periods of silence in order to avoid increasing the output gain during these periods. The output gain is calculated based on a comparison of the effective signal level to a compression threshold.

Abstract: A system and a method of indicating the reception of voice in a hearing assistance system that is constructed and arranged to assist a user to better hear the voice of another person. The hearing assistance system includes a detector that is capable of determining whether or not speech has been received by the hearing assistance system. In response to detecting the reception of the voice of another person by the hearing assistance system, the reception of the voice of another person by the hearing assistance system is visually indicated.

Abstract: A device is disclosed, the device comprising a least one microphone, at least one output audio transducer driver, at least one processor, a storage comprising a set of instructions executed locally, wherein the set of instructions causes the processor to: continuously measure an ambient sound pressure level when the device is an on state, calculate a sound pressure level of the output audio transducer driver when the output audio transducer is active, combine the two sound pressure levels into a total sound pressure level.