Abstract: The invention relates to simulation of sound fields in enclosures, for instance for application in listening tests, where test subjects assess the sound quality or other sound perception characteristics of the sound field. According to a specific embodiment, the system comprises a binaural synthesis portion which synthesizes sound for instance from a sound-reproduction equipment based on measured impulse responses of an actual room stored in a data base (31) and a binaural recording portion comprising a data base 32 for storing binaural recordings of other sound signals made in the room. Data from these databases are mixed (41) and reproduced by means of a headphone (39) provided with a head tracker (42) for tracking the movements of the listener's head. The invention furthermore comprises the use of cross-fading functions (36, 37) to enable the dynamic listening conditions, where the movements of the listener's head are taken into account during the simulation process.

Abstract: A remote controlled robot system that includes a robot and a remote control station. The robot includes a binaural microphone system that is coupled to a speaker system of the remote control station. The binaural microphone system may include a pair of microphones located at opposite sides of a robot head. The location of the microphones roughly coincides with the location of ears on a human body. Such microphone location creates a mobile robot that more effectively simulates the tele-presence of an operator of the system. The robot may include two different microphone systems and the ability to switch between systems. For example, the robot may also include a zoom camera system and a directional microphone. The directional microphone may be utilized to capture sound from a direction that corresponds to an object zoomed upon by the camera system.

Abstract: Provided is an image sensing apparatus in which directivity can be changed in accordance with the position of the shooter. The image sensing apparatus includes a video shooting unit, at least four microphones placed around the video shooting unit, and an audio signal processing unit adapted to synthesize audio signals from two microphones among the at least four microphones, thereby outputting a synthesized audio signal from a specific direction of the video shooting unit. The apparatus further includes a selecting unit adapted to select the position of a shooter. The audio signal processing unit includes a directivity changeover unit adapted to change over directivity by changing a combination of the two microphones in accordance with a selection output signal the said selecting unit.

Abstract: A karaoke device is included within an enclosure and having a voice pickup element integrated into the enclosure, the voice pickup element for converting sound waves into an electrical signal. An audio input signal passes into the enclosure and connects with an electronic circuit for amplifying the electrical signal, for controlling the amplitude of the electrical signal and for mixing the electrical signal and the audio input signal into a mixed audio signal which passes out of the enclosure. A video input signal passes into the enclosure and directly connects to a video output signal that passes out of the enclosure.

Abstract: The present invention provides a sound generating method of generating sound signals related to a video signal, which comprises a step of generating independently each of the sound signals matched to a horizontal direction and a vertical direction of a video, and a step of allowing the horizontal and the vertical sound signals that have been generated to be reproduced independently with horizontal sound output means and vertical sound output means, respectively.

Abstract: A wireless receiving apparatus that includes a connector including a plurality of terminals. A predetermined terminal is assigned as an identification terminal for identifying an electronic appliance connected to the connector, and a circuit for generating an identification signal of a level corresponding to a voltage level of the predetermined terminal when the electronic appliance is connected to the connector, and for identifying the electronic appliance connected to the connector based on the level of the identification signal.

Abstract: A control method to automatically adjust the volume of a sound transmitter based on the measurement and the computation of the acoustic statistics of the room where the sound is emitted. The system comprises at least one sensor, a calculator determining statistics of the signal collected by the at least one sensor and a controller using these statistics provided by the calculator to adjust the volume of the transmitter in the room.

Abstract: The invention provides a method for manufacturing array microphones. First, signal delays of a plurality of microphones are measured. The microphones are then categorized into a plurality of categories according to the signal delays. A plurality of array microphones are then assembled with a number of component microphones selected from the same categories.

Abstract: The present invention provides a means for two or more remotely-located individuals to communicate information about the spatial coordinates of a location of mutual interest in a more rapid, robust, and intuitive manner than is possible with any current voice communication system.

Abstract: An apparatus for processing an audio signal and method thereof applied to an audio playback system are disclosed. The apparatus comprises a decoder, an error-correcting circuit and an audio correcting module. The method for processing audio signals in accordance with the present invention decodes the audio signal to generate a decoded signal by the decoder. Then, the error-correcting circuit performs an error-correcting algorithm in the decoded signal to generate an error indication signal and an output audio signal. And the audio correcting module corrects the output audio signal to generate a corrected audio signal when the error indication signal indicates that the output audio signal has error.

Abstract: An electronic apparatus is provided that has a rear-side and a front-side, a first microphone that generates a first signal, and a second microphone that generates a second signal. An automated balance controller generates a balancing signal based on an imaging signal. A processor processes the first and second signals to generate at least one beamformed audio signal, where an audio level difference between a front-side gain and a rear-side gain of the beamformed audio signal is controlled during processing based on the balancing signal.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for stereophonic acoustic echo cancellation. The method includes collecting, at a same time, a first audio sample of an audio source from a first omnidirectional microphone and a second audio sample of the audio source from a second omnidirectional microphone. The method includes delaying the second audio sample by a first amount of time to yield a delayed second audio sample and combining the delayed second audio sample with the first audio sample to produce a first channel, then delaying the first audio sample by a second amount of time to yield a delayed first audio sample and combining the delayed first audio sample with the second audio sample to produce a second channel. Then the method includes outputting the first channel and the second channel as a stereo audio signal of the audio source.

Abstract: An audio signal processing apparatus includes first, second and third omni-directional microphones each of which receives sound and generates an omni-directional audio signal and which are spaced apart by a predetermined distance, a first adder section that adds audio signals generated by the first, second and third omni-directional microphones and generates an audio signal having an omni-directivity in the whole circumferential direction, a first subtractor section that subtracts audio signals generated by the first and third omni-directional microphones and generates an audio signal having a directivity in the right-left direction, a second adder section that adds audio signals generated by the first and third omni-directional microphones, a second subtractor section that subtracts an audio signal generated by the second omni-directional microphone from the audio signal added by the second adder section and generates an audio signal having a directivity in the front-back direction, and an output section tha

Abstract: A method determines a bias reduced noise and interference estimation in a binaural microphone configuration with a right and a left microphone signal at a time-frame with a target speaker active. The method includes a determination of the auto power spectral density estimate of the common noise formed of noise and interference components of the right and left microphone signals and a modification of the auto power spectral density estimate of the common noise by using an estimate of the magnitude squared coherence of the noise and interference components contained in the right and left microphone signals determined at a time frame without a target speaker active. An acoustic signal processing system and a hearing aid implement the method for determining the bias reduced noise and interference estimation. The noise reduction performance of speech enhancement algorithms is improved by the invention. Further, distortions of the target speech signal and residual noise and interference components are reduced.

Abstract: Described are systems and methods performed by computer to reduce crosstalk produced by loudspeakers when rendering binaural sound that is emitted from the loudspeakers into a room. The room may have sound-reflecting surfaces that reflect some of the sound produced by the loudspeakers. To reduce crosstalk, a room model stored by the computer, is accessed. The room model models at least sound reflected by one or more of the physical surfaces. The room model is used to calculate a model of an audio channel from the loudspeakers to a listener. The model of the audio channel models sound transmission from the loudspeakers to the listener. The computer uses the model of the audio channel to cancel crosstalk from the loudspeakers when rendering the binaural sound.

Abstract: A video-audio recording and reproducing apparatus (101) has a built-in stereo microphone (21a, 21b) and an external microphone connection terminal (32). The external microphone connection terminal (32) is connected to a binaural microphone (3) to be attached to the ears of a photographer (300). When the binaural microphone (3) is used to collect ambient sounds, an audio signal to be recorded on a recording medium is switched from an audio signal from the built-in stereo microphone (21a, 21b) to a binaural audio signal from the binaural microphone (3). The photographer (300) puts the binaural microphone (3 (31a, 31b)) on his or her ears and collects ambient sounds around the photographer (300) including a sound emanating from an object. The object is photographed with a camera unit (11). The recording medium records the binaural audio signal, a photographed video signal, and a binaural flag signal.

Abstract: Surround sound recording is a tedious task requiring the use of many microphones. The invention aims at enabling the use of two-channel microphones (or stereo microphones) for multi-channel surround recording. A conventional stereo microphone, or a two-channel microphone specifically optimized for use with the proposed algorithm, is used to generate two signals. A post-processor is applied to the microphone generated signals to convert them to multi-channel surround. This aim is achieved through a method to generate multiple output audio channels (y1, . . . , yM) from two microphone generated audio channels (x1, x2), in which the number of output channels is equal or higher than two, this method comprising the steps of: determine directions of sound components related to the microphone characteristics determine compensation gains of sound components related to the microphone characteristics generating the output audio channels, y1, . . .

Abstract: A system and method is described herein in which an audio source wirelessly transmits audio content to a first audio sink over one wireless link and to a second audio sink over another wireless link. The two audio sinks also exchange forward error correction (FEC) streams over a wireless link between the two audio sinks, wherein the FEC streams are generated by FEC encoding the audio content received from the audio source. The audio sinks advantageously use the exchanged FEC information to synchronize the playback of the audio content as well as to improve the robustness of the wireless links with the audio source in a manner that does not consume additional bandwidth on those links.

Abstract: An output circuit of a bidirectional side microphone element includes an inverting amplifier inverting a phase and outputting an inverted signal, adds a non-inverted output signal of the side microphone element to an output signal of a middle microphone element having unidirectivity to produce a signal for one channel of the left and right channels; and adds an inverted output signal of the side microphone element being the output signal from the inverting amplifier to the output signal of the middle microphone element to produce another signal for the other channel. An input resistor and a feedback resistor to the inverting amplifier are dividable. The division ratio of the input resistor to the feedback resistor is varied to change the levels of the non-inverted output signal and the inverted output signal of the side microphone element, and to change the angle between the left and right directional axes.

Abstract: A system enhances spatialization in which spatial information about sound sources at an originating location is represented in an audio signal. The system applies a phase difference analysis to the signals received from an array of spaced apart input devices or microphones to derive spatial or directional information about the relative directions of one or more satellite input devices or microphones. The signals from the satellite input devices or microphones are mixed by a function of their respective directions to generate a multichannel output signal. When processed by a remote or local system, the output signal provides a representation of the relative directions of the sound sources at the originating location at a receiving location.

Abstract: A system and method for generating, transmitting and distributing audio signals. The system and method may include an array having a plurality of microphones and a plurality of speakers as well as a first processor disposed proximate the array. The system and method can also have a second processor disposed remotely from the array and communicatively coupled with the array, at least one remotely located device having at least one microphone and at least one speaker and one or more remotely located speakers. Also, there may be an audio signal that is generated by one of a microphone on the array and a microphone on the at least one remotely located device, the location of the generation of the audio signal determined by one of the first processor and the second processor, generated audio signal transmitted to at least one of the speakers on the array, the at least one speaker on the remotely located device and the one or more remotely located speakers.

Abstract: A method for generating and playing audio signals and a system for processing audio signals are disclosed. The method for generating audio signals includes: generating distance information about an audio signal corresponding to a view point position, according to obtained auxiliary video and direction information about the audio signal, where the auxiliary video is a disparity map or a depth map; encoding the direction information and distance information about the audio signal, and sending the encoded information. The apparatus for generating audio signals includes an audio signal distance information obtaining module and an audio signal encoding module. With the present invention, the position information, including direction information and distance information, about the audio signal may be obtained accurately in combination with a three-dimensional video signal and a three-dimensional audio signal, without increasing the size of a microphone array, and the audio signal is sent and played.

Abstract: Provided is a sound pickup apparatus that increase the flexibility in installation of microphones, while enabling sound from a sound source to be picked up in stereo without using the information on the current positions of the microphones.

Abstract: A measuring apparatus that measures sound arrival delay time from a speaker to a microphone on the basis of a result obtained by collecting signals output from the speaker by means of the microphone includes: measuring means for measuring the sound arrival delay time that makes a control such that a first sine wave signal having a first frequency and a second sine wave signal having a second frequency different from the first frequency are output from the speaker, is input with the first sine wave signal and the second sine wave signal collected by the microphone and then mixes the first sine wave signal and the second sine wave signal so as to generate a third sine wave signal having a frequency corresponding to a difference between the first frequency and the second frequency, and measures the sound arrival delay time on the basis of the third sine wave signal.

Abstract: There is provided an audio reproduction apparatus. The audio reproduction apparatus comprises a first acquisition unit, a determination unit, a calculation unit, a second acquisition unit, and a generation unit. The configuration of the audio reproduction apparatus enables changing a head-related transfer function for signal processing in audio signal reproduction in accordance with the state of the audio recording apparatus at the time of audio signal acquisition.

Abstract: A stereo capacitor microphone unit includes: two unidirectional microphone units integrally formed with respective fixed electrodes of the unidirectional microphone units facing each other; and an insulating spacer that is interposed between the fixed electrodes and provided with a gap formed at a portion of an outer periphery towards radial direction. The gap communicates fixed electrode rear spaces of the respective unidirectional microphone units with an external space to serve as a common rear acoustic terminal for the unidirectional microphone units.

Abstract: An emotive advisory system for use by one or more occupants of an automotive vehicle includes a directional speaker array, and a computer. The computer is configured to determine an audio direction, and output data representing an avatar for visual display. The computer is further configured to output data representing a spoken statement for the avatar for audio play from the speaker array such that the audio from the speaker array is directed in the determined audio direction. A visual appearance of the avatar and the spoken statement for the avatar convey a simulated emotional state.

Abstract: A stereo microphone includes: a supporter that is made of an elastic material and fixes two unidirectional microphone units at certain angles; and a moving part that applies external force to a part of the supporter by being operated. An angle formed by main axes of the respective unidirectional microphone units is changed by deformation of the supporter by the external force.

Abstract: The present invention relates to an apparatus and method for redeeming otherwise closed and concealed information contained in audio signals. This includes both the primary reference signal, and a plurality of redundant duplicate signals, substantially identical in all respects to the primary reference signal except in relation to magnitude and phase, for the purpose of unfolding, or opening the audio signal content into layers that result in an omni-directional acoustic signal, representing the sound as it would behave in nature. The audio reproduction system uses an in-phase circuit and a separate phase layering technique circuit to drive independent multiple mixed channels, to produce an open, substantially complete sound from a discrete audio signal, for the purpose of enabling a substantially complete audio signal to be formed, or to transform existing incomplete audio signals into a substantially complete audio signal.

Abstract: This invention relates to a system (200) for determining directionality of a sound. The system (200) comprises a first audio device (202) placed on one side of a user's head (100) and having a first microphone unit (110, 112) for converting said sound to a first electric signal, a second audio device (204) placed on the other side of the user's head (100) and having a second microphone unit (114, 116) for converting said sound to a second electric signal, and comprises a transceiver unit (220, 238) for interconnecting the first and second audio device and communicating the second electric signal to the first audio device (202). The first audio device (202) further comprises a first comparator (222) for comparing the first and second electric signals and generating a first directionality signal from the comparison.

Abstract: In a method and an acoustic system that generate a stereo signal for each of multiple. separate sources, a blind source separation of at least two microphone signals is conducted to acquire BSS filters. Each of the microphone signals is filtered with its own filter transfer function that is the quotient of a power density spectral portion of the respective sound source and the overall power density spectrum of the respective microphone signal, such that two stereo signals are obtained for each microphone signal. An approximation of the signals to be separated, for example for each of two hearing devices, is thereby possible.

Abstract: Certain embodiments of the invention may include systems, methods, and apparatus for recording three dimensional audio. According to an example embodiment of the invention, the method may include orienting a three-dimensional (3-D) microphone with respect to a predetermined spatial direction, selectively receiving sounds from one or more directions corresponding to directional receiving elements, recording the selectively received sounds in a 3-D recorder having a plurality of recording channels, recording time code in at least one channel of the 3-D recorder; and mapping the recorded channels to a plurality of output channels.

Abstract: A microphone system including a main unit for controlling the entire system and microphones having cascade connections from the main unit assuming the main-unit side upstream and the opposite side downstream. The microphone includes communication control means for controlling data transmitted between the main unit and microphones, sound input means for converting collected sound into a digital signal, echo cancellation means for eliminating an echo component in the sound signal, and sound-information generation means for updating the sound information by adding the sound signal of the self-microphone to the sound information of the downstream microphones and upstream transmitting the up data including the updated sound information. The microphone transmits the down data transmitted from the main unit to the down-most microphone in sequence in accordance with the cascade connection and transmits the up data from the down-most microphone to the main unit in reverse sequence.

Abstract: The invention concerns a method for reproducing spatial impression of existing spaces in multichannel or binaural listening. It consists of following steps/phases: a) Recording of sound or impulse response of a room using multiple microphones, b) Time- and frequency-dependent processing of impulse responses or recorded sound, c) Processing of sound to multichannel loudspeaker setup in order to reproduce spatial properties of sound as they were in recording room, and (alternative to c), d) Processing of impulse response to multichannel loudspeaker setup, and convolution between rendered responses and an arbitrary monophonic sound signal to introduce the spatial properties of the measurement room to the multichannel reproduction of the arbitrary sound signal, and is applied in sound studio technology, audio broadcasting, and in audio reproduction.

Abstract: There is provided a signal processing device including: an audio signal acquisition portion that acquires audio signals; an external signal acquisition portion that acquires external signals; an output signal generation portion that generates output signals from the audio signals and the external signals; a mode setting portion that sets an external mode as an operation mode; and a fade control portion that controls the output signal generation portion in accordance with the operation mode. When the external mode is set, the fade control portion causes the output signal generation portion to generate the output signal for one of the right ear and the left ear of the user from at least the external signal, and also to generate the output signal for the other ear from at least the audio signal.

Abstract: A communication system, comprising: a transmission unit comprising at least two microphones, with a separate audio signals channel for each microphone, a first ear unit and, a second ear unit worn at the each side of the user's head, each ear unit comprising a receiver unit; the transmission unit comprising means for transmitting at least a first channel and a second channel of the audio signals to the first and second ear unit, at least one of the receiver units being capable of receiving the at least first and second audio signal channel, at least one of the ear units comprising audio signal processing means for generating processed audio signals received via the at least first and second audio signal channel, with the first ear unit and the second ear unit comprising means for stimulating the user's hearing at the right ear and the left ear, accordingly.

Abstract: The two-way wireless speaker system of this invention increases sound fidelity by enabling speakers to acknowledge receipt of audio data packets. This provides increased functionality because the audio hub can receive data not only from wired inputs, but also wireless transmission from computer, cell phone, and other sources. Audio hub can use information from speaker to customize/adjust audio signal for each speaker independently, giving better audio quality and synchronization among speakers.

Abstract: A method and apparatus for capturing image and sound during interactivity with a computer program is provided. The apparatus includes an image capture unit that is configured to capture one or more image frames. Also provided is a sound capture unit. The sound capture unit is configured to identify one or more sound sources. The sound capture unit generates data capable of being analyzed to determine a zone of focus, at which to process sound to the substantial exclusion of sounds outside of the zone of focus. In this manner, sound that is captured and processed for the zone of focus is used for interactivity with the computer program.

Abstract: An acoustic apparatus without increasing noise etc. even when plural directional microphones collect sounds from a place of the same distances is provided. Sound signals output from the microphone arrays are subjected to phase shift by phase shift circuits 211A to 211H, and the sound signals are combined by an adder 212, The phase shift circuits 211A to 211H performs phase shifts according to installation positions of the microphone arrays. The phase shift circuit 211A makes the shift 0 degree, the phase shift circuit 211B makes the shift 45 degrees, the phase shift circuit 211C makes the shift 90 degrees, and sequentially to the phase shift circuit 211H, the shifts are made according to rotational angles.

Abstract: Disclosed is an object based audio contents generating/playing apparatus. The object based audio contents generating/playing apparatus may include an object audio signal obtaining unit to obtain a plurality of object audio signals by recording a plurality of sound source signals, a recording space information obtaining unit to obtain recording space information with respect to a recording space of the plurality of sound source signals, a sound source location information obtaining unit to obtain sound location information of the plurality of sound source signals, and an encoding unit to generate object based audio contents by encoding at least one of the plurality of object audio signals, the recording space information, and the sound source location information, thereby enabling the object based audio contents to be played using at least one of a WFS scheme and a multi-channel surround scheme regardless of a reproducing environment of the audience.

Abstract: Head Related Transfer Functions (HRTFs) of an individual are measured in rapid fashion in an arrangement where a sound source is positioned in the individual's ear channel, while microphones are arranged in the microphone array enveloping the individual's head. The pressure waves generated by the sounds emanating from the sound source reach the microphones and are converted into corresponding electrical signals which are further processed in a processing system to extract HRTFs, which may then be used to synthesize a spatial audio scene. The acoustic field generated by the sounds from the sound source can be evaluated at any desired point inside or outside the microphone array.

Abstract: A binaural sound reproduction apparatus includes a correction filter operable to filter an input sound signal that is recorded using a binaural recording microphone and to supply the filtered signal to a headphone, an adaptive filter to which the input sound signal is supplied, and a difference detector determining a difference between a sound signal that is obtained by collecting a sound reproduced by the headphone using a sound-collecting microphone that is the same as the binaural recording microphone, or that has a similar characteristic to that of the binaural recording microphone, and a sound signal output from the adaptive filter, and for transmitting the difference to the adaptive filter. The adaptive filter determines the inverse of a synthesis characteristic from the headphone to the sound-collecting microphone based on the input sound signal and the difference, and sets the determined characteristic as a characteristic of the correction filter.

Abstract: The invention concerns the processing of audio data. The invention is characterized in that it consists in: (a) encoding signals representing a sound propagated in three-dimensional space and derived from a source located at a first distance (P) from a reference point, to obtain a representation of the sound through components expressed in a spherical harmonic base, of origin corresponding to said reference point, (b) and applying to said components compensation of a near-field effect through filtering based on a second distance (R) defining, for sound reproduction, a distance between a reproduction point (HPi), and a point (P) of auditory perception where a listener is usually located.

Abstract: The use of sound signals from sounds collected by four non-directional sound collectors facilitates calculation for generating surround sound signals, for example, when multi-channel surround sound signals are generated, and prevents the size of the entire apparatus from increasing. Surround sound signals are generated from sound signals of sounds collected by a microphone unit composed of a first sound collector, a second sound collector disposed posterior to the first sound collector, a third sound collector, and a fourth sound collector.

Abstract: This invention is a method for binaural localization using a cascade of resonators and anti-resonators to implement an HRTF (head-related transfer function). The spectrum of the cascade reproduces the magnitude spectrum of a desired HRTF. The proposed method provides a considerably more computationally efficient implementation of HRTF filters with no detectable deterioration of output quality while saving memory when storing a large quantity of HRTFs due to the parameterization of its resonators and anti-resonators. Finally, the method offers additional flexibility since the resonators and anti-resonators can be manipulated individually during the design process, making it possible to interpolate smoothly between HRTFs, reduce spectral coloring or achieve higher accuracy at perceptually relevant frequency regions. These HRTF are useful in stereo enhancement and multi-channel virtual surround simulation.

Abstract: A microphone comprises a three-pin socket, a first USB socket and a USB adapter. The first USB socket is fixed within the three-pin socket. The three-pin socket comprises an inserted block and three pins. The three pins are fixed on the inserted block. The first USB socket is embedded in the inserted block. The USB adapter comprises a USB plug which matches the first USB socket and a second USB socket which has a different dimension from the first USB socket. The USB plug and the second USB socket are connected electrically. The microphone can be normally used at all occasions. The dimension of the finished product is reduced so as to save a very large space. The microphone can be electrically connected with a USB data line with a USB plug having a different dimension from the first USB socket.

Abstract: To provide a microphone unit for stereophonic recording capable of adjusting an intersecting angle of sound collecting axes in accordance with conditions without impairing directivity and sound quality of the microphone. Respective rotation bases (408, 508) are rotatable about a center axis with respect to mounting bases (404, 504), respectively, which are mounted to a main body (20) of a portable sound recorder (10) through respective brackets (402, 502). Respective top ends of respective knurls (414, 514) of leading ends of the respective rotation bases (408, 508) has a shape being cut to form an inclined surface from a radial direction, and respective microphones (418, 518) are retained so that diaphragms (420, 520) are in consonance with the cut surface.

Abstract: The invention makes use of impulse responses of the performance venue to process a recording or other signal so as to emulate that recording having being recorded in the performance venue. In particular, by measuring or calculating the impulse responses of a performance venue such as an auditorium between an instrument location within the venue and one or more soundfield sampling locations, it then becomes possible to process a “dry” signal, being a signal which has little or no reverberation or other artifacts introduced by the location in which it is captured (such as, for example, a close microphone studio recording) with the impulse response or responses so as to then make the signal seem as if it was produced at the instrument location in the performance venue, and captured at the soundfield sampling location.

Abstract: A mechanism is provided that monitors secondary microphone signals, in a multi-microphone mobile device, to warn the user if one or more secondary microphones are covered while the mobile device is in use. In one example, smoothly averaged power estimates of the secondary microphones may be computed and compared against the noise floor estimate of a primary microphone. Microphone covering detection may be made by comparing the secondary microphone smooth power estimates to the noise floor estimate for the primary microphone. In another example, the noise floor estimates for the primary and secondary microphone signals may be compared to the difference in the sensitivity of the first and second microphones to determine if the secondary microphone is covered. Once detection is made, a warning signal may be generated and issued to the user.

Abstract: To determine the direction of incidence for a signal from an acoustic signal source using a directional microphone system, which has at least two microphones, two or more directional microphone signals are produced, each having a direction-dependent sensitivity distribution with a minimum in one direction. The directional microphone signals are assessed with regard to a quantity to determine the directional microphone signal which is most influenced by the associated direction dependent sensitivity distribution. The direction of incidence is determined as being the direction in which the minimum of the sensitivity distribution of this directional microphone signal is located.