Abstract: An electronic apparatus is provided having a front side and a rear side oriented in opposite directions along a first axis, and a right-side and a left-side oriented in opposite directions along a second axis that is perpendicular to the first axis. A null control signal is generated based on an imaging signal. A first microphone located near the right-side of an electronic apparatus generates a first signal and a second microphone located near the left-side of the electronic apparatus generates a second signal. The first and second signals are processed, based on the null control signal, to generate a right beamformed audio signal having a first directional pattern having at least one first null, and a left beamformed audio signal having a second directional pattern having at least one second null. A first angular location (?) of the at least one first null and a second angular location (?) of the at least one second null are steered based on the null control signal.

Abstract: An acoustic-feedback detection apparatus includes: a first level detecting section configured to detect a signal level of sound signals obtained from a position in a sound-signal system in which a microphone and speaker are connected; a first extracting section configured to extract, from the sound signals of which the signal level is detected, signals in a band having a bandwidth predetermined for each of at least one predetermined center frequency; a second level detecting section configured to detect a signal level of the signals in each band, the signals being extracted by the first extracting section; and a determining section configured to determine whether or not acoustic feedback is occurring, on the basis of a threshold determined according to the signal level detected by the first level detecting section and a waveform of each signal level detected by the second level detecting section.

Abstract: A method for processing signals for reducing noise components in a vehicular microphone system (500A) compares (517) a plurality of noise component values (511, 513, 515) for at least one frequency band of plurality of frequency bands. At least one frequency band is then downwardly expanded (521) by a predetermined expansion ratio (523) for providing a noise reduced signal (535) when determining a value is below a predetermined threshold (519).

Abstract: A harmonic generating apparatus and the method are provided, which are used to enhance the quality of the bass audio signals. The method includes the steps of: providing a frequency signal having a present level and a preceding level; comparing the present level with the preceding level to generate a compared result; and generating the plurality of harmonics based on the compared result.

Abstract: An electronic device includes a central processing unit, a USB connector, a USB switch, and an audio path selector. The USB switch has a data transmission path and an audio signal transmission path for signal transmission between the central processing unit and the USB connector. The USB switch selects the data transmission path for data transmission according to a first selection signal from the central processing unit, and selects the audio signal transmission path for audio signal transmission according to a second selection signal from the central processing unit. The audio path selector interconnects the central processing unit and the USB switch, and has a first audio path for output of audio signals from the central processing unit to the audio signal transmission path, and a second audio path for output of audio signals from the audio signal transmission path to the central processing unit.

Abstract: In embodiments of the present invention improved capabilities are described for determining a multi-dimensional sound signature for a location within a hypothetical space, comparing the multi-dimensional sound signature to a known multi-dimensional sound signature, and modifying the hypothetical space such that the similarity between the multi-dimensional sound signature for the location within the modified hypothetical space and the known multi-dimensional sound signature is increased.

Abstract: Methods and systems for TDD hum noise cancellation are disclosed and may include correlating a received audio signal to a time division duplexed (TDD) envelope signal, controlling a feedback signal based on the correlating, and cancelling a TDD noise signal via the feedback signal. The amplitude and delay of the feedback signal may be configured for controlling. A least mean square (LMS) filter may be utilized for the correlating and may include a finite impulse response filter. The TDD envelope signal may be generated on the chip or may be received from a source external to the chip. The feedback signal may be added to the received audio signal for the cancelling.

Abstract: A system and method provide at least a single stage optimization process which maximizes the flatness of the net subwoofer and satellite speaker response in and around a cross-over region. A first stage determines an optimal cross-over frequency by minimizing an objective function in a region around the cross-over frequency. Such objective function measures the variation of the magnitude response in the cross-over region. An optional second stage applies all-pass filtering to reduce incoherent addition of signals from different speakers in the cross-over region. The all-pass filters are preferably included in signal processing for the satellite speakers, and provide a frequency dependent phase adjustment to reduce incoherency between the center and left and right speakers and the subwoofer. The all-pass filters are derived using a recursive adaptive algorithm.

Abstract: An approach is provided that receives a sound request from a requesting process. A selective mute setting is identified that corresponds to user enablement actions. If the selective mute setting is disabled the sound request is played at an audio output in response. However, if the selective mute setting being enabled, then steps are performed to determine whether to play the sound request. These steps include determining whether the sound request corresponds to one or more user sound request actions. If it is determined that the sound request corresponds to at least one of the user sound request actions then the sound request is audibly played at the audio output. However, if the sound request does not correspond to at least one of the user sound request actions, then the audio request is suppressed and not played at the audio output.

Abstract: A method of processing a signal is disclosed. The present invention includes receiving (a) downmix signal being generated from plural-channel signal and (b) spatial information indicating attribute of the plural-channel signal in order to upmix the downmix signal and including phase shift flag indicating whether phase of a frame of at least one channel of the plural-channel signal is shifted; obtaining inter-channel phase difference (IPD) coding flag indicating whether IPD value is used to the spatial information from a header of the spatial information; obtaining IPD mode flag indicating whether the IPD value is used to frame of the spatial information from the frame based on the IPD coding flag; obtaining the IPD value of parameter band in the frame, based on the IPD mode flag; upmixing plural-channel signal by applying the IPD value to the downmix signal; and shifting the phase of the frame of the at least one channel of the plural-channel signal based on the phase shift flag.

Abstract: A method of processing a signal is disclosed. The present invention includes receiving a downmix signal generated from plural channel signal and spatial information indicating attribute of the plural channel signal to upmix the downmix signal; obtaining inter-channel phase difference (IPD) coding flag indicating whether IPD value is used to the spatial information from header of the spatial information; obtaining IPD mode flag based on the IPD coding flag from the frame of the spatial information, the IPD mode flag indicating whether the IPD value is used to a frame of the spatial information; obtaining the IPD value of parameter band of parameter time slot in the frame, based on the IPD mode flag; smoothing the IPD value by modifying the IPD value by using IPD value of previous parameter time slot; and generating plural channel signal by applying the smoothed IPD value to the downmix signal.

Abstract: A passive sound pressure level (SPL) limiter is provided that can be used with audio sources of varying drive levels and headsets, earbuds, etc. of varying sensitivity. The SPL limiter includes a control circuit that includes a rectifier and, in most configurations, a step-up transformer. The SPL limiter also includes a limiting circuit that utilizes a transistor to shunt current from the audio source in proportion to a control signal output by the control circuit. The control circuit may further include a low pass filter, for example an RC filter, and one or more fast limit diode paths. The limiting circuit may further include a feedback network to increase the linear behavior of the limiting circuit.

Abstract: A mixing control device has a plurality of input channels for processing and mixing audio signals according to parameters specified in the respective input channels. In the mixing control device, a plurality of channel strips are grouped into a first portion and a second portion, each channel strip having a fader operable for controlling a parameter. An allocating part allocates the plurality of the input channels to the plurality of the channel strips so that the fader of the channel strip is operable to control the parameter of the input channel allocated to the channel strip. An instructing part provides an instruction to switch the parameter to be controlled by the fader.

Abstract: A passive stethoscope chest piece has a chest piece housing containing a passive piezoelectric (piezo) element mounted within a metal housing. The piezo element converts body signals to an electrical representation. A pair of electrical connections attached to the metal plate and the piezoelectric layer of the piezo element pass the electrical representation outside the housing for processing by external circuitry. This abstract is not to be considered limiting, since other embodiments may deviate from the features described in this abstract.

Abstract: The present invention relates to an apparatus and method for localizing a sound source in real time. The apparatus for localizing a sound source in real time includes a sound signal acquisition unit for acquiring sound signals through two or more channels. A sample delay storage unit stores a plurality of pieces of data, sampled from the sound signals acquired through respective channels, for a predetermined period of time. A correlation calculation unit calculates correlations between the channels from the plurality of pieces of sampled data stored in the sample delay storage unit. A sound source direction calculation unit calculates an azimuth angle of the sound source using both the correlations between the channels and location relationships of the sound signal acquisition unit. Accordingly, the present invention can localize a sound source in real time.

Abstract: A method for reducing keyboard noise in conferencing equipment which includes a microphone and a keyboard, the method including: providing, at a noise reduction apparatus, an audio signal originating from the microphone; detecting, at the noise reduction apparatus, an operation of the keyboard; providing, at the noise reduction apparatus, a pre-stored estimate of a keyboard audio noise resulting from the operation of the keyboard upon detection of the operation of the keyboard; and calculating, at the noise reduction apparatus, a noise reduced output signal based on the estimate of the keyboard audio noise and the audio signal.

Abstract: An audio system 10 has a plurality of audio sources 12a-12c for outputting audio signals different from each other. A coefficient device 15d adjusts the gain of the audio signal output from one of the audio sources. A nonvolatile memory 15i stores, as internal gains, gains to be set to the coefficient device in correspondence with the audio sources. When a switch changeover section 17 selects one of the audio sources, the CPU 15h reads from the nonvolatile memory the internal gain corresponding to the audio source selected, and sets the internal gain to the coefficient device.

Abstract: A binaural decoder for an MPEG surround stream, which decodes an MPEG surround stream into a stereo 3D signal, and a decoding method thereof. The method includes dividing a compressed audio stream and head related transfer function (HRTF) data into subbands, selecting predetermined subbands of the HRTF data divided into subbands and filtering the HRTF data to obtain the selected subbands, decoding the audio stream divided into subbands into a stream of multi-channel audio data with respect to subbands according to spatial additional information, and binaural-synthesizing the HRTF data of the selected subbands with the multi-channel audio data of corresponding subbands.

Abstract: A plural-channel audio signal (e.g., a stereo audio) is processed to modify a gain (e.g., a volume or loudness) of a speech component signal (e.g., dialogue spoken by actors in a movie) relative to an ambient component signal (e.g., reflected or reverberated sound) or other component signals. In one aspect, the speech component signal is identified and modified. In one aspect, the speech component signal is identified by assuming that the speech source (e.g., the actor currently speaking) is in the center of a stereo sound image of the plural-channel audio signal and by considering the spectral content of the speech component signal.

Abstract: A sound source localization system includes a plurality of microphones for receiving a signal as an input from a sound source; a time-difference extraction unit for decomposing the signal inputted through the plurality of microphones into time, frequency and amplitude using a sparse coding and then extracting a sparse interaural time difference (SITD) inputted through the plurality of microphones for each frequency; and a sound source localization unit for localizing the sound source using the SITDs. A sound source localization method includes receiving a signal as an input from a sound source; decomposing the signal into time, frequency and amplitude using a sparse coding; extracting an SITD for each frequency; and localizing the sound source using the SITDs.