Abstract: To provide a speaker device that can form a substantially uniform sound field over a range from a long distance to a short distance without significantly increasing a calculation load. A plurality of FIR filters 21 to 2n perform delay control of respective speakers so as to increase a delay time difference between adjacent speakers 51 to 5n in a line array speaker 5 toward one end of the line array speaker 5, and thereby over a wide range from a long distance to a short distance, a sound filed 12 is formed. Also, by adding a common shift delay time Dc to filter coefficients for the FIR filters 21 to 2n, the delay time difference between adjacent speakers 51 to 5n is made less than a sampling period of a sound signal to form a wide and uniform sound field 12.

Abstract: Provided is an encoding/decoding apparatus and method of multi-channel signals. The encoding apparatus and method of multi-channel signals may encode phase information of the multi-channel signals using a quantization scheme and a lossless encoding scheme, and the decoding apparatus and method of multi-channel signals may decode the phase information using an inverse-quantization scheme and a lossless decoding scheme.

Abstract: A multi-channel audio system including first combining circuitry, for combining a first spectral band of a first plurality channels to provide a first bass audio signal stream; second combining circuitry, for combining the first spectral band of a second plurality channels to provide a second bass audio signal stream; and third combining circuitry, for combining a second spectral band, the second spectral band including lower frequencies than the first spectral band, of the first plurality of channels and the second plurality of channels to provide a third bass audio signal stream.

Abstract: An audio interruption and buffering playback system includes a primary audio source for reproducing primary audio content and a secondary audio source for reproducing secondary audio content. A processing device detects an interruption event that includes the secondary audio source reproducing secondary audio content having priority over the primary audio source. The processor mutes the output of the primary audio content in response to the interruption event. A buffer buffers the audio content from the primary audio source during the interruption event. The buffered audio content is reproduced by the processor to the user at an accelerated playback speed following the interruption event.

Abstract: Reflection cancelling boundary microphones are described in which a microphone capsule is mounted within the pressure zone of a vibrating surface and the microphone is tuned in such a way as to cancel pressure waves reflected by the surface while admitting pressure waves generated by the vibration of the surface. One embodiment of the invention includes a microphone capsule configured to be mounted within the pressure zone of a vibrating surface. In addition, the microphone is tuned to cancel pressure waves reflected by the surface while admitting pressure waves generated by the vibration of the surface.

Abstract: Among other things, techniques and systems are disclosed for detecting musical structures, such as downbeats. In one aspect, a method performed by a data processing device includes receiving an input audio signal. The method includes detecting a meter in the received audio signal. Detecting the meter includes generating an envelope of the received audio signal; generating an autocorrelation phase matrix having a two-dimensional array based on the generated envelope to identify a dominant periodicity in the received audio signal; and filtering both dimensions of the generated autocorrelation phase matrix to enhance peaks in the two-dimensional array. The meter represents a time signature of the input audio signal having multiple beats. Additionally, the method includes identifying a downbeat as a first beat in the detected meter.

Abstract: A listening device for processing an input sound to an output sound, includes an input transducer for converting an input sound to an electric input signal, an output transducer for converting a processed electric output signal to an output sound, a forward path being defined between the input transducer and the output transducer and including a signal processing unit for processing an input signal in a number of frequency bands and an SBS unit for performing spectral band substitution from one frequency band to another and providing an SBS-processed output signal, and an LG-estimator unit for estimating loop gain in each frequency band thereby identifying plus-bands having an estimated loop gain according to a plus-criterion and minus-bands having an estimated loop gain according to a minus-criterion.

Abstract: Systems and methods for adjusting a bias voltage and gain of the microphone to account for variations in a thickness of a gap between a movable membrane and a stationary backplate in a MEMS microphone due to the manufacturing process. The microphone is exposed to acoustic pressures of a first magnitude and a sensitivity of the microphone is evaluated according to a predetermined sensitivity protocol. The bias voltage of the microphone is adjusted when the microphone does not meet the sensitivity protocol. The microphone is then exposed to acoustic waves of a second magnitude that is greater than the first magnitude and a stability of the microphone is evaluated according to a predetermined stability protocol. The bias voltage and the gain of the microphone are adjusted when the microphone does not meet the stability protocol.

Abstract: Noise canceling headphones designed so that external noise coming from all directions can be effectively concerned by means of a canceling sound before being heard by a user. These feed-forward noise canceling headphones use noise canceling headphones (10), each of which comprises an air chamber (8) formed from a space that is sealed by a member covering the outer circumference of the housing and provided more outward than a front air chamber (7), and wherein the sound-wave introducing part of a microphone (2) is inside the air chamber (8).

Abstract: Six input channels, to which are allocated 5.1-channel surround signals from a plurality of input sources, are grouped into a surround channel group, and these six input channels are connected to corresponding ones of six surround buses in a one-to-one relationship. Thus, merely grouping the input channels into a surround channel group allows the signals of the individual input channels (5.1-channel surround signals) to be taken out via a plurality of output destinations (5.1-channel speakers) corresponding to the surround buses. Once an instruction is given for changing a value of a parameter, the parameter is controlled in a ganged fashion in all of the input channels of the surround channel group except for an LFE input channel of the surround channel group.

Abstract: A microphone array system (16) for sound acquisition from multiple sound sources in a reception space surrounding a microphone array (18) that is interfaced with a beamformer module (28) is disclosed. The microphone array (18) includes microphone transducers (22) that are arranged relative to each other in N-fold rotationally symmetry, and the beamformer includes beamformer weights that are associated with one of a plurality of spatial reception sectors corresponding to the N-fold rotational symmetry of the microphone array (18). Microphone indexes of the microphone transducers (18) are arithmetically displaceable angularly about the vertical axis during a process cycle, so that a same set of beamformer weights is used selectively for calculating a beamformer output signal associated with any one of the spatial reception sectors. A sound source location module (30) is also disclosed that includes a modified steered power response sound source location method.

Abstract: A signal processing device includes at least two sound signal input elements, an orthogonal transformer, a phase difference determiner, and a virtual spectrum generator. Two sound signals in a time domain are received from the two respective sound signal input elements. The orthogonal transformer orthogonally transforms the two sound signals in the time domain into two respective spectral signals in a frequency domain. The phase difference determiner determines a phase difference between the two spectral signals in the frequency domain. The virtual spectrum generator generates a virtual spectral signal in the frequency domain in accordance with the phase difference and one of the two spectral signals.

Abstract: A sound signal processing device that is capable of suitably extracting main sound from mixed sound in which unnecessary sound (for example, leakage sound and reverberant sound) is mixed with the main sound. More specifically, a mixed sound signal in the time domain including first sound and second sound, and a target sound signal in the time domain including sound corresponding to at least the second sound, which have temporal relation in their entirety or in part, are each divided into a plurality of frequency bands. A level ratio between the two signals is calculated at each frequency. Based on the level ratio, a signal of the first sound that is included in the mixed sound signal is extracted.

Abstract: The present invention discloses a microphone array structure able to reduce noise and improve speech quality and a method thereof. The method of the present invention comprises steps: using at least two microphone to receive at least two microphone signals each containing a noise signal and a speech signal; using FFT modules to transform the microphone signals into frequency-domain signals; calculating an included angle between a speech signal and a noise signal of the microphone signal, and selecting a phase difference estimation algorithm, a noise reduction algorithm or both to reduce noise according to the included angle; if the phase difference estimation algorithm is used, calculating phase difference of the microphone signals to obtain a time-space domain mask signal; and multiplying the mask signal and the average of the microphone signals to obtain the speech signals of the microphone signals. Thereby is eliminated noise and improve speech quality.

Abstract: Arrangements in a device for use with a service involving play out of audio from a headset are provided. The arrangements include an application for providing the play out of audio to a user of the device. The application has at least two modes of operation including first and second active modes. In the first active mode, the headset is in full physical contact with the user. In the second active mode, the headset may not be in full physical contact with the user. The application is configured to switch between the first and second modes of operation based on whether the headset is detected to be in full physical contact with the user. In the first active mode, the user is entitled to full use and/or operation of the service. In the second active mode, the user is entitled to modified use and/or operation of the service. Related methods are also provided.

Abstract: Systems, methods, and other embodiments associated with selective notch filtering for howling suppression are described. According to one embodiment, an apparatus includes a howling detector that detects howling by performing a time domain analysis of speech signals to identify a speech signal that may be exhibiting howling and performing frequency domain analysis to confirm that the speech signal is exhibiting howling. The apparatus also includes a suppression selector configured to select a suppression technique, a signal processor configured to process the speech signal according to a selected suppression technique. The apparatus outputs, without suppression-related processing, speech signals that are not identified based on the time domain analysis.

Abstract: A POP noise suppressing circuit includes a blocking capacitor, an outputting element connected to the blocking capacitor, a current generating unit, a charging unit connected to the current generating unit, a discharging unit connected to the current generating unit, a switching unit connected to the charging unit and the discharging unit and a power amplifying unit connected between the switching unit and the blocking capacitor. The current generating unit provides two slowly increasing currents respectively to the charging unit and the discharging unit. The switching unit is for switching between the charging unit and the discharging unit. The charging unit forms a charging current. The discharging unit forms a discharging current. The voltage of the blocking capacitor increases smoothly in a process of powering on and decreases smoothly in a process of powering off. A POP noise suppressing method is also provided.

Abstract: Provided is an encoding device (1) including: a pitch contour analysis unit (101) which detects information, a dynamic time-warping unit (102) which generates, based on the information, pitch change ratios (Tw_ratio in FIG. 18) within a range (86) including a range (86a) of the pitch change ratios corresponding to absolute pitch differences of 42 cents or larger; a first lossless coding unit (103) which codes the generated pitch parameters (102x); a time-warping unit (104) which shifts a pitch of a signal according to the information; and a second encoding unit which codes a signal (104x) obtained by the shifting.

Abstract: Audible signals (m2) which are transmitted via an uncontrolled transmission path (11) to an individual's ear wearing the control signal processing path (1) of a hearing device are compensated by audible signals (m3) which are produced by an additional controlled signal processing path (20). Signal processing parts are commonly exploited (25b, 25d, 7/27) by both controlled signal processing paths (1, 20).

Abstract: A transmission device includes a transmission power setting unit that respectively sets a transmission power for transmitting a main sound signal indicating a main sound and transmission powers for transmitting one or two or more subsidiary sound signals which pertain to the main sound signal and which realize a predetermined sound effect, and a first communication unit that respectively transmits the main sound signal and the one or two or more subsidiary sound signals on time division channels using the same frequency band, based on the transmission powers set by the transmission power setting unit, wherein the transmission power setting unit sets the transmission power for the main sound signal to a first reference value, and sets the transmission powers for the subsidiary sound signals to a second reference value smaller than the first reference value.