Abstract: The present disclosure provides a radio and method for automatically adjusting, in response to ambient noise, the volume setting of a radio. The radio comprises a microphone for receiving ambient audio and generating a microphone signal; a codec for receiving the microphone signal and generating a processor signal representative of the ambient audio; and processor circuitry operable to receive the processor signal, determine an ambient audio level in response to the processor signal, and determine an adjusted radio volume to generate a radio volume control signal, wherein the adjusted radio volume is determined by calculating a difference between a baseline volume level and ambient audio level and adding a current volume level setting, wherein the codec is also operable to receive the radio volume control signal and generate an output signal to adjust the radio volume setting to maintain a net difference between the adjusted radio volume and ambient audio.

Abstract: Methods and systems are provided for matrixed audio settings. In an audio system that comprises at least one audio output element operable to output audio signals, a first user input that comprises a selection of audio mode may be received, and based on the selected audio mode, one or more audio settings applicable during generating, processing, and/or outputting of the audio signals may be determined. At least one user control element may be configured to enable receiving of a second user input that comprises a selection between the determined one or more corresponding audio settings. The audio system (or at least the audio output element thereof) may be a headset.

Abstract: Systems and methods for processing multichannel audio signals include receiving a multichannel time-domain audio input, transforming the input signal to plurality of multi-channel frequency domain, k-spaced under-sampled subband signals, buffering and delaying each channel, saving a subset of spectral frames for prediction filter estimation at each of the spectral frames, estimating a variance of the frequency domain signal at each of the spectral frames, adaptively estimating the prediction filter in an online manner using a recursive least squares (RLS) algorithm, linearly filtering each channel using the estimated prediction filter, nonlinearly filtering the linearly filtered output signal to reduce residual reverberation and the estimated variances, producing a nonlinearly filtered output signal, and synthesizing the nonlinearly filtered output signal to reconstruct a dereverberated time-domain multi-channel audio signal.

Abstract: Disclosed herein is a lighting system including a luminaire having a lighting device and a sound reduction device. The lighting device includes an illumination output surface, which is at least partially reflective with respect to an audio wave from outside the luminaire. The lighting device also includes an illumination light source configured to generate illumination light for emission through the illumination output surface for illumination of an area. The sound reduction device includes a pick up microphone and an audio output source. The pick up microphone is configured to detect incoming audio waves in a vicinity of the luminaire. The lighting system further includes a circuitry including a sound reduction controller coupled to the pick up microphone and the audio output source of the sound reduction device. The sound reduction controller is configured to operate the audio output source to control sound at least in vicinity of the illuminated area associated with the incoming audio waves.

Abstract: A method for reducing vibration noise by an electronic device is described. The method includes obtaining an audio signal. The audio signal includes vibration noise. The method also includes obtaining vibration data from one or more motion sensor signals. The method further includes processing the vibration data to produce microphone-response matched vibration data based on a transfer function. The method additionally includes reducing the vibration noise based on the microphone-response matched vibration data.

Abstract: The present embodiment relates to a double-faced display device generating sound by directly vibrating a display panel and includes: a first display panel and a second display panel disposed at the front and rear surface thereof, respectively; a single sound generating actuator connected to the first display panel and the second display panel to simultaneously vibrate the first display panel and the second display panel and thus generate sound; and a support part configured to support the sound generating actuator, the first display panel, and the second display panel. The present embodiment does not require a separate speaker and thus can achieve a thin or slim display device, and can improve the localization or the quality of sound output in opposite directions in the double-faced display device.

Abstract: A modular speaker system includes a plurality of speaker modules configured prior to use to function in reproducing various sound channels depending upon specific desires of a user. Each of the plurality of speaker modules includes a substantially cylindrical housing including an open first end and a closed second end, as well as a cylindrical sidewall extending between the first end and the second end. Each speaker module also includes a first driver secured within the first end of the housing, the first driver being mounted within the first end of the housing for directing sound outwardly, and a second driver positioned within the housing at a central position between the first end and the second end of the housing. A plurality of circular rings are positioned around the housing allowing for control of the speaker module in various ways.

Abstract: An electronic system and an audio processing method are provided. The electronic system includes a physiological signal sensor, a processing circuit, an audio processing circuit and an audio output device. The physiological signal sensor is utilized for measuring a physiological signal of a user. The processing circuit is utilized for determining an audio setting according to the physiological signal. The audio processing circuit is utilized for receiving an audio signal and adjusting the audio signal according to the audio setting so as to generate an audio output signal. The audio output circuit is utilized for playing the audio output signal.

Abstract: A tactile sound device includes a transducer to convert an electrical signal into motion. One or membranes are coupled to the transducer and are adapted to transfer vibrations from the transducer to a user's body. A first sensor monitors the vibrations of the transducer. One or more circuits generate the electrical signal based on a signal received from the first sensor that monitors the vibrations of the transducer.

Abstract: An eye cover includes ear plugs integrated into the head strap to transmit audio signals. The audio portion of the device is battery-operated and preprogrammed with a variety of sounds traditionally considered emotionally soothing.

Abstract: An apparatus and method for controlling an audio power circuit. The audio power circuit includes an audio amplifier having a power input and a speaker connected to the audio amplifier. The audio power circuit also includes a control circuit configured to be connected to a battery and the power input and to control a supply of power to the power input. The audio power circuit further includes a thermal protection circuit connected between the audio amplifier and the speaker, the thermal protection circuit configured to generate a thermal protection signal and provide the same to the control circuit, wherein the control circuit is further configured to control the supply of power based on the thermal protection signal.

Abstract: A method and system dynamically adjusts the audio of an audio and video signal to improve its overall sound quality and dialog intelligibility. Some embodiments use gain, equalization, audio signal compression and spatial enhancement (reverb) on individual channels of a multichannel audio signal.

Abstract: Certain example embodiments relate to an acoustic wall assembly that uses active and/or passive sound reverberation to achieve noise-disruptive functionality, and/or a method of making and/or using the same. With the active approach, sound waves in a given frequency range are detected by a sound masking circuit. Responsive to detection of such sound waves, an air pump (e.g., speaker) is used to pump air in the wall assembly to actively mask the detected sound waves via reverberation and/or the like. The wall assembly may include one, two, or more walls, and the walls may be partial or full walls. With the passive approach, sound waves in a given frequency range are disrupted via features (e.g., holes, slits, etc.) formed in and/or on a wall itself. These techniques may be used together or separately, in different example embodiments.

Abstract: The present embodiment relates to a double-faced display device generating sound by directly vibrating a display panel and includes: a first display panel and a second display panel disposed at the front and rear surface thereof, respectively; a single sound generating actuator connected to the first display panel and the second display panel to simultaneously vibrate the first display panel and the second display panel and thus generate sound; and a support part configured to support the sound generating actuator, the first display panel, and the second display panel. The present embodiment does not require a separate speaker and thus can achieve a thin or slim display device, and can improve the localization or the quality of sound output in opposite directions in the double-faced display device.

Abstract: It is disclosed a speaker module, including a module housing, a speaker assembly and a heat-dissipating member. The module housing has an inner cavity, the inner cavity includes a front cavity and a rear cavity spaced apart from the front cavity, and the front cavity is communicated with an external space. The speaker assembly is mounted in the inner cavity, the speaker assembly includes a vibration diaphragm spacing the front cavity and the rear cavity apart. The heat-dissipating member has a heat-conducting section and a heat-dissipating section, the heat-conducting section is in contact with the speaker assembly, the heat-dissipating section stretches into the front cavity. The vibration of the vibration diaphragm allows air in the front cavity to flow to increase the heat-dissipating rate of the heat-dissipating section. The heat-dissipating section may have a heat-dissipating structure for increasing a heat-dissipating area, and the heat-dissipating structure may be a heat-dissipating fin.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: Systems and methods are described for analyzing and resolving feedback caused by having multiple audio links in a conference room. An audio system may detect the presence of duplicated audio caused by multiple audio links in the conference room. Marker signals may be injected into the conference room or over the network in response to detecting the duplicated audio. Echoes of the marker signals may be received, and the system may determine which case corresponds to the detected duplicated audio based on the received echo of the marker signals. Based on the determined case, operation of at least one of the speaker and the microphone may be modified. After the modification, audio playback in the conference room may be monitored to verify that far end audio playback is taking place and that the duplicated audio has been resolved.

Abstract: Ambient sound is converted into digital signals. A processor performs active noise cancellation and/or a transformation operation that is distinct from the active noise cancellation on the digital signals. The active noise cancellation and the transformation operation transform the digital signals into modified digital signals. The modified digital signals are converted into modified analog signals. The modified analog signals are outputted as audio waves. An interior microphone is configured to output an output signal to the processor in response to receiving the modified analog signals. In response to receiving the output signal from the interior microphone, the processor is configured to determine whether the modified digital signals produce desired audio waves.

Abstract: A method of producing a spatialized stereo audio file from an original stereo audio file comprises creating a data base of impulse responses realized in at least one physical space divided into left, right, front, back, up and down sides relative to a sound acquisition position, with at least one pair of acquisition microphones placed at the sound acquisition position, with at least two pairs of source loudspeakers placed at sound source positions; the sound acquisition position is situated at the left-right median plane of the physical space, the sound source positions are distributed symmetrically by pairs relative to the sound acquisition position, the data base of impulse responses comprising at least one left/right impulse response pair, the left and right impulse responses being obtained by a deconvolution of the direct acquired signal from all the source loudspeakers distributed at the respective left and right side of the physical space.

Abstract: Various embodiments describe techniques for switching microphones in a multiple microphone system. The techniques incorporate sampling audio signals from multiple microphones, determining a microphone that has the greatest incoming amplitude during the analysis window, and switching the microphone to that greatest amplitude microphone. The transition point for switching microphones may be determined when either the amplitude of the incoming signal is within an error bound of zero or at a “zero-crossing” in the input amplitude stream.