Abstract: A howling canceller applied to an acoustic system having a speaker and a microphone comprises: a filter insertion unit for inserting a notch filter at a frequency of an audio signal picked up by the microphone; a setting unit for setting the insertion time of the notch filter on the basis of the frequency at which the notch filter is inserted; and a releasing unit for, when the insertion time set by the setting unit has elapsed, releasing the notch filter, the insertion time of which has elapsed. The setting unit sets the insertion time of the notch filter to be shorter as the frequency at which the notch filter is inserted increases.

Abstract: A microphone and a method for calibrating a microphone are disclosed. In one embodiment the method for calibrating a microphone comprises operating a MEMS device based on a first AC bias voltage, measuring a pull-in voltage, calculating a second AC bias voltage or a DC bias voltage, and operating the MEMS device based the second AC bias voltage or the DC bias voltage.

Abstract: A howling canceller is adapted to an acoustic system having a speaker and first and second microphones. The speaker and the first microphone form a first acoustic feedback loop; the speaker and the second microphone form a second acoustic feedback loop. The howling canceller includes a howling suppressing unit for performing suppression processing in such a way that: frequency components at which howling is possibly occurring are detected in each of the sound signals picked up by the first and second microphones; the detected frequency components of the sound signals picked up by the first and second microphones are compared with each other on a per-frequency basis and a frequency component having larger power is detected; and based on the comparison results, the larger power frequency component of at least one of the sound signals picked up by the first and second microphones is suppressed.

Abstract: Methods, systems, and computer program products for simulating propagation of sound in a static scene can be operated for pre-computing a transfer operator for simulating results of sound reflection within a modeled scene, simulating distribution of sound energy from a sound source positioned within the scene, applying the transfer operator to the distribution of sound energy from the sound source to compute simulated sound at a plurality of points within the scene, and simulating collection of sound energy from the plurality of points within the scene to a listener positioned within the scene.

Abstract: An auditory display apparatus is provided that places sounds such that sounds whose fundamental frequencies are close to each other are not adjacent to each other. A sound transmission/reception section receives sound data. A sound analysis section analyzes the sound data, and calculates a fundamental frequency of the sound data. A sound placement section compares the fundamental frequency of the sound data with a fundamental frequency of adjacent sound data, and places the sound data such that a difference in fundamental frequency is maximized. A sound management section manages a placement position of the sound data. A sound mixing section mixes the sound data with the adjacent sound data. A sound output section outputs the sound data obtained by the mixture to a sound output device.

Abstract: Sound is outputted, without giving the sense of incongruity to the user, even if reproduction of animation is stopped by the user. An animation acquiring section 11 acquires animation data D1 representing an animation generated in advance based on a setting operation by the user, and sound data D2 representing a sound to be reproduced in association with the animation. A sound output control section 12 calculates stop time sound information representing a feature of the sound at a point of time at which reproduction of the animation is stopped, using sound attribute information D4, in response to input of a stop command by the user, and determines a predetermined output method of the sound that matches the animation, based on the calculated stop time sound information, to reproduce the sound by the determined output method.

Abstract: Provided are a method of generating and playing an object-based audio content that may effectively store preset information about an object-based audio content, and a computer-readable recording medium for storing data having a file format structure for an object-based audio service. The method of generating the object-based audio content may include: receiving a plurality of audio objects (310) generating at least one preset using the plurality of audio objects (320) and storing a preset parameter with respect to an attribute of the at least preset and the plurality of audio objects (330). The preset parameter may be stored in a form of a box that is defined in a media file format about the object-based audio content. Through this, it is possible to effectively store a preset about a plurality of audio objects.

Abstract: A method of reducing noise in an environment where the noise source is in a fixed location relative to a pair of microphones, such as in a camera with a zoom motor, involves receiving signals x1(t), x2(t) from the respective microphones, and filtering each of the signals x1(t), x2(t) with respective first and second linear filters having filter coefficients obtained by computing eigenfilters corresponding to data samples from the respective microphones for noise only and signal only conditions.

Abstract: A system for controlling an electronic device, including an accessory coupled to the device having a control interface for receiving at least one input and a tone generator, the accessory configured to, in response to an input, generate a corresponding control tone using the tone generator, and send the control tone and a calibration sequence to the device. The device is operable to receive the calibration sequence and control tone. The device includes a zero-crossing detector operable to determine at least one of the start or ending of each of the calibration sequence and the control tone and having a circuit operable to determine the periods of the calibration sequence and the control tone based on the zero-crossing detector output.

Abstract: Embodiments of a method, system, and apparatus for inserting caller identifying information into a video signal includes receiving a digital video stream and receiving caller identifier information from an incoming call associated with a calling device. The method further includes converting the caller identifier information into caller identifying textual information, and inserting the caller identifying textual information into the digital video stream using a visual timecode video insertion format. The method further includes transmitting the digital video stream with the inserted caller identifying textual information.

Abstract: A High Speed Serial Bus connecting teleconference devices, such as video conference unit, speakerphone, external loudspeakers or microphones is disclosed. The disclosed embodiments enable the distributing and sharing of controls among the linked devices. They enable incremental upgrades or capacity increases of the teleconference system.

Abstract: A method for real-time monitoring of audio signals reception quality includes receiving output signals from a plurality of microphone clusters, each microphone cluster having at least two microphone units to receive audio signals from at least two distinct directions and output corresponding electrical signals; identifying comparative features of output signals for each of the microphone clusters; and selecting at least one microphone cluster based on the identified features. A system for real-time monitoring of audio signals reception quality includes a plurality of microphone clusters, each microphone cluster having at least two microphone units to receive audio signals from at least two distinct directions and output corresponding electrical signals; and a main audio unit to identify comparative features of output signals for each of the microphone clusters and to select at least one microphone cluster based on the identified features.

Abstract: An audio device (e.g. an AV amplifier), with a plurality of audio output terminals according to the HDMI standard, converts audio data of a source device into audio signals suited to audio reproduction abilities (e.g. channel counts) of sink devices. The audio device stores a plurality of audio reproduction ability information describing the predetermined channel count thereof and the channel counts of sink devices, wherein arbitrary audio reproduction ability information precluding audio reproduction ability information of a sink device with the smallest channel count is provided to a source device. The audio device performs down-mixing on audio signals with respect to a sink device whose channel count is smaller than the predetermined channel count, while the audio device performs signal processing on audio signals with respect to a sink device whose channel count is larger than the predetermined channel count.

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 estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit determines a degree of coupling between the user's ear and the transducer and adjusts the adaptive cancellation of the ambient sounds to prevent erroneous and possibly disruptive generation of the anti-noise signal if the degree of coupling lies either below or above a range of normal operating ear contact pressure.

Abstract: A plurality of frequencies are generated at a wireless device and a frequency which a user perceives as the loudest frequency is determined. A first frequency range is selected comprising a first minimum frequency and a first maximum frequency. A ringtone is processed to shift the second frequency range to within the first frequency range, wherein an average frequency of the processed ringtone is the determined loudest frequency.

Abstract: A microphone array device includes a first sound reception unit configured to obtain a first sound signal that is input from a first microphone, a second sound reception unit configured to obtain a second sound signal that is input from a second microphone, a noise state evaluation unit configured to compare the first sound signal and the second sound signal and to obtain an evaluation parameter to evaluate an influence of a non-target sound included in the second sound signal on a target sound included in the first sound signal according to a result of the comparison, a subtraction adjustment unit configured to set a suppression amount for the second sound signal based on the evaluation parameter and to generate a third sound signal; and a subtraction unit configured to generate a signal to be output based on the third sound signal and the first sound signal.

Abstract: Signals are received, over a range of angles, at an input of a device. The signals include a primary signal with a principal direction of arrival and an interfering signal with a respective interfering direction of arrival at the input. Measurements are determined for the received signals over the range of angles. Each measurement relates to a particular angle and indicating the energy of the received signals which are received from the particular angle. For each angle over the range of angles, a value is removed from the measurement for that angle, the value being based on the minimum of: (i) the energy of the measurement for that angle, and (ii) the energy of a corresponding measurement for a corresponding angle mirrored around the principal direction of arrival, whereby the remaining values of the plurality of measurements are indicative of said at least one interfering direction of arrival.

Abstract: A method for automatic volume adjustment performed by a physical computing system includes, with the computing system, determining a virtual proximity between a user and a number of applications running on the computing system, the virtual proximity based in part on a type of one of the number of applications, and with the computing system, automatically adjusting a volume of the number of applications linked with a sound peripheral connecting to the computing system, the volume of each of the number of applications being based on the virtual proximity.

Abstract: Techniques for providing multiple power supplies in electronic devices are disclosed. According to one aspect of the present invention, an appropriate power supply is provided only to accommodate a volume setting. In other words, there are at least two power supplies, one with a low voltage and the other with a high voltage. The high voltage power supply is only applied when there is a need to accommodate a volume setting. Thus the power consumption of the amplifiers is well controlled. As a result, the designs of the device and heat dissipation therein can be simplified and lowered in cost.

Abstract: Systems and methods described herein provide for low latency active noise cancellation which alleviate the problems associated with analog filter circuitry. The present technology utilizes low latency digital signal processing techniques which overcome the high latency conventionally associated with conversion between the analog and digital domains. As a result, low latency active noise cancellation is performed utilizing digital filter circuitry which is not subject to the inaccuracies and drift of analog filter components. In doing so, the present technology provides robust, high quality active noise cancellation.