Abstract: Methods are disclosed for improving sound localization of the human ear. In some embodiments, the method may include creating virtual movement of a plurality of localized sources by applying a periodic function to one or more location parameters of a head related transfer function (HRTF).

Abstract: Disclosed is an electronic device including: a housing (110) in which at least one sound hole (113) and at least one sound hole (114) are formed; an oscillator (120) that is provided inside the housing (110) and outputs a modulated wave for a parametric loudspeaker; a sound demodulation unit (130) that is not provided in the sound hole (113) but provided in the sound hole (114) and demodulates the modulated wave; and a sound hole opening and closing unit (112) that is mounted to the housing (110) and opens and closes the sound hole (113).

Abstract: In an exemplary information processing system including an information processing section, a first loudspeaker, and a second loudspeaker, predetermined music data composed of two channels is separated into first channel music data and second channel music data. A first sound source object associated with one of the first channel music data and the second channel music data, a second sound source object associated with the other one, and a virtual microphone are placed in a virtual 3-dimensional space. The reproduction volume of a sound reproduced from the first loudspeaker, and the reproduction volume of a sound reproduced from the second loudspeaker are determined based on the direction of the virtual microphone and the positional relationship between the virtual microphone and each of the first sound source object and the second sound source object.

Abstract: A voice detection system and method for automatic volume controls and voice sensors is disclosed. More specifically, the invention addresses a situation where the user's own voice undesirably affects the functionality of an automatic volume control for a two-way communication device, such as a cellular telephone. In addition, the invention proposes solutions wherein one (voice) microphone is employed and also, when two (voice and noise) microphones are employed. Further, an algorithm is disclosed that addresses the issue concerning the user's own voice in an AVC pertaining to the two microphone solution. Yet further, a method herein is disclosed that detects the presence of voice in a single non-selective (noise) microphone.

Abstract: Various embodiments relate to a system for adapting a signal gain, which comprises a peak detector configured to receive an audio input signal containing consecutive signal blocks and to dynamically establish a current input signal level of the audio input signal, and a loudness determination unit configured to dynamically determine a perceived loudness of a current signal block of the audio input signal based on a psycho-acoustic model of a human hearing. Furthermore the system comprises a time constant generation unit configured to determine a time constant based on the perceived loudness of the current signal block and the perceived loudness of a preceding signal block, wherein the time constant describing a change of the signal gain. A gain determination unit is configured to determine the signal gain for the current signal block based on the time constant and the current input signal level.

Abstract: In an exemplary information processing system including a plurality of sound output sections, the positional relationship among the plurality of sound output sections is recognized. In addition, a sound corresponding to a sound source object present in a virtual space is generated. The output volume of the sound for the sound source object is determined, for each sound output section, in accordance with the positional relationship among the plurality of sound output sections, and the generated sound is outputted in accordance with the output volume.

Abstract: A flat panel electronic device and an audio playing apparatus thereof are provided. The audio playing apparatus comprises an audio generator, a plurality of speakers, a sensor and a controller. The audio generator is operable to generate a left channel audio and a right channel audio. The plurality of speakers are configured such that at least one pair of speakers is symmetrically disposed at a left side and a right side of the flat panel electronic device no matter how the flat panel electronic device is placed. The sensor is operable to detect a placed state of the flat panel electronic device in the installed state. The controller is operable to receive a detecting signal from the sensor so as to control the at least one pair of speakers to play the left channel audio and the right channel audio correspondingly according to the placed state of the flat panel electronic device.

Abstract: A watch includes a sound input body and a main body. The main body include an interface unit, a first electromagnetic coil unit, a signal processing unit, and a control unit. The sound input body includes a microphone and a second electromagnetic coil unit. The second electromagnetic coil unit generates inducted electromagnetic filed in response to an electric signal generated by the microphone. The first electromagnetic coil unit coupling the second electromagnetic coil unit generates electric signal in response to the electromagnetic field. The signal processing unit filters algorithm of the electric signal. The control unit recognizes the electric signal amplified by the signal amplification unit to execute corresponding function.

Abstract: In a structure of an in-vehicle audio device, the audio system has an audio signal supplying section, an audio signal processing section, a balance adjustment section and an audio signal output section. A right audio signal processing section and a left audio signal processing section in the audio signal processing section perform a process of filtering audio signals supplied from the audio signal supplying section on the basis of a transfer function. The transfer function contains a head related transfer function H(s), a space transmission compensation function G0(s)?1 and a vehicle height compensation function G1(s). The space transmission compensation function G0(s)?1 is an inverse function of a space transfer function G0(s). A filtering process for the audio signal is performed by using the transfer function to guide a driver's visual line to be decreased when a vehicle height is high, and increased when the vehicle height is low.

Abstract: An audio interface adapter device and an audio signal receiving apparatus are provided. The audio interface adapter device comprises an audio interface having a pin 1, a pin 2, a pin 3 and a pin 4. The pin 1 and/or the pin 2 is used as an audio pin of the audio interface. One of the pin 3 and the pin 4 is used as a microphone pin, and the other one of the pin 3 and the pin 4 is used as a ground pin. The audio pin is connected with the pin 3 via a first unidirectional conductive element, and connected with the pin 4 via a second unidirectional conductive element. A conductive direction of the first unidirectional conductive element is from the audio pin to the pin 3, and a conductive direction of the second unidirectional conductive element is from the audio pin to the pin 4.

Abstract: In one aspect, an audio processing apparatus is disclosed. The apparatus includes an audio decoder, a filterbank, and a processor. The audio decoder decodes an encoded audio signal to obtain a time-domain audio signal, the encoded audio signal including a plurality of spectral components. The filterbank splits the time-domain audio signal to obtain a plurality of complex-valued subband samples in a first frequency region. The processor generates a plurality of subband samples in a second frequency region based at least in part on the complex-valued subband samples in the first frequency region, adaptively groups at least some of the plurality of subband samples in the second frequency region with an adaptive time resolution or an adaptive frequency resolution, and determines a spectral profile of at least some of the subband samples in the second frequency region based on the groups.

Abstract: Embodiments of systems and methods are described for providing backwards compatibility for legacy devices that are unable to natively render non-channel based audio objects. These systems and methods can also be beneficially used to produce a reduced set of audio objects for compatible object-based decoders with low computing resources.

Abstract: An overall wireless broadcast system utilizing headphones that react in terms of light colors and light impulses when attuned to specific broadcast channels is provided. Such a system provides a unique performance capability in relation to the whims and selections of headphone wearers and listeners at an event with multiple attendees, ostensibly creating a spontaneous, continually changing visual result for spectators and participants alike. The headphones utilized for such a purpose are particularly adapted to allow for channel selections based upon different broadcast music possibilities (such as from a DJ or music device located in sufficiently close proximity for a headphone wearer to receive the broadcast signal to the degree that the broadcast music can be properly heard within the headphones themselves) such that the headphones themselves become modified in their own presentation based upon such a channel selection.

Abstract: A personal audio device, such as a wireless telephone, includes an adaptive noise canceling (ANC) circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone is also provided proximate the speaker to measure the ambient sounds and transducer output near the transducer, thus providing an indication of the effectiveness of the noise canceling. A processing circuit uses the reference and/or error microphone, optionally along with a microphone provided for capturing near-end speech, to determine whether the ANC circuit is incorrectly adapting or may incorrectly adapt to the instant acoustic environment and/or whether the anti-noise signal may be incorrect and/or disruptive and then take action in the processing circuit to prevent or remedy such conditions.

Abstract: A method for identifying an accessory coupled to an electronic device. The method includes applying at least one detection pulse to the audio accessory, each detection pulse being spectrally shaped to be generally inaudible to a human user, receiving at least one response signal corresponding to each detection pulse that is indicative of the impedance of the accessory, and based on the impedance, identifying the accessory.

Abstract: To solve the problems with the prior art that a multi-microphone array cannot inhibit broad-band noises well and cannot be used in the increasingly widespread broad-band communication, embodiments of the present invention disclose a method, a device and a system for eliminating noises with multi-microphone array.

Abstract: Haptic effects are dynamically generated for content presentation on a device through analysis of the content. During content playback, audio data for the content may be analyzed to determine low frequency audio data. The low frequency audio data is mapped from a low frequency range to a haptic control frequency range of one or more haptic actuators included in the device. This mapping may be used to generate a control signal to drive the one or more haptic actuators. The haptic effects and the content may be synchronized to one another during the presentation of the content on the device. The haptic actuator control signal may be amplified proportionally to the amplitude of the low frequency audio data.

Abstract: The present invention discloses a flat speaker output device and a method for starting the same. Wherein, a plurality of flat speakers utilizes an initial delay unit and a plurality of intermediary delay units connected in series. The initial delay unit connects with the power controller and a first one of the flat speakers. The intermediary delay units respectively connect with the residual each flat speakers. The power controller controls a power source to the initial delay unit to delay the start of the first one of flat speakers, and outputs the power source to the intermediary delay units to sequentially delay the starts time of the residual each flat speaker. The present invention can sequentially start flat speakers without using a high-output power supply device and thus decrease the required capacity of the external power supply device.

Abstract: A method of processing an input audio signal, the method comprising forming a plurality of output audio signals from the input audio signal, wherein each output audio signal is formed by performing respective processing on the input audio signal, wherein for a first output audio signal there is a target audio equalization operation comprising a target filter twice, wherein for the first output audio signal, the respective processing comprises a first audio equalization operation, the first audio equalization operation being the target audio equalization operation modified to compensate for phase shifts that correspond to zeros of the transfer function of the target audio equalization operation, wherein for each output audio signal other than the first output audio signal, the respective processing comprises a compensation filter that compensates for phase shifts that correspond to poles of the transfer function of the target audio equalization operation.

Abstract: A watch includes a main body and a sound input body. The sound input body includes a microphone, a signal transmitting coil unit, a power receiving coil unit, and a power receiving unit. The main body includes an interface unit, a signal receiving coil unit coupling the signal transmitting coil unit, a signal amplification unit, a power transmitting unit, a power transmitting coil unit, and a control unit. The microphone converts a voice signal input to an electric signal, which is transmitted to the control unit by the signal transmitting coils coupling with the signal receiving coils. The control unit executes corresponding function in response to the electric signal. The electric signal obtained by the control unit is transmitted to the power receiving unit by the power transmitting coils coupling with the power receiving coils to power the microphone.