Abstract: The invention provides an audio interface device and method utilizing a single audio jack connector for plugging into a three-wire analog microphone or a three-wire digital microphone, thereby reducing dimensions and production costs thereof and an electronic system using the same. The audio interface comprises an audio jack connector, having first to third contacts electronically connected with the three-wire microphone plugged thereinto; and an audio processing device, detecting a type of the three-wire microphone plugged into the audio jack connector, outputting a clock signal to the three-wire microphone and receiving a digital audio signal from the three-wire microphone when the type is digital; or receiving analog audio signals from the three-wire microphone when the type is analog.

Abstract: The invention provides a microphone. The microphone receives a first sound signal and at least one second electrical signal and outputs a third electrical signal. In one embodiment, the microphone comprises a transducer and a signal processor. The transducer converts the first sound signal to a first electrical signal. The signal processor has a first input terminal receiving the first electrical signal and at least one second input terminal receiving the at least one second electrical signal, and derives the third electrical signal from the first electrical signal and the second electrical signal. In one embodiment, the at least one second electrical signal is derived from a t least one second sound signal by at least one second microphone located in the vicinity of the microphone. In another embodiment, the at least one second electrical signal comprises a wind noise signal derived from wind pressure by a pressure sensor located in the vicinity of the microphone.

Abstract: The invention provides an audio interface device and method to acquire audio signal from either an analog or digital microphone through a common connector. The audio interface comprises a connector, electrically connected with a microphone plugged thereinto, an analog readout circuit, acquiring an analog audio signal from an analog microphone, a digital readout circuit, acquiring a digital audio signal from a digital microphone, and a decision circuit, selectively connecting the analog or digital readout circuit to the connector to acquire the analog or digital audio signal respectively according to whether the microphone plugged into the connector is analog or digital.

Abstract: An electronic device includes a case and a microphone array. The case includes a front surface, a rear surface, and side edges connecting the front surface and the rear surface. The microphone array includes a first microphone module and a second microphone module disposed in the case and arranged in a row not parallel to the side edges of the case.

Abstract: An electronic device includes a housing, a circuit board, and a microphone array. A plurality of acoustic openings is defined in the housing. The circuit board is placed in the housing. The microphone array is placed on the circuit board for receiving external sound via the acoustic openings in the housing.

Abstract: A telephone system comprising an analog telephone and a full-duplex speakerphone adapter is disclosed. The analog telephone comprises an ordinary analog telephone subscriber circuit for transmitting and receiving analog signals and a handset for users. The full-duplex speakerphone adapter is coupled between the analog telephone and a central office, uses a subscriber loop interface circuit through the first telephone line to couple to the ordinary analog telephone subscriber circuit of the analog telephone and uses a telephone hybrid interface circuit through the second telephone line to couple to a wall jack to communicate with the central office.

Abstract: A telephony system equipped with an echo cancellation module is disclosed. A hybrid interface circuit outputs an outbound signal to a wall jack while receiving an inbound signal from the wall jack. The inbound signal may contain a line echo of the outbound signal caused by the impedance mismatches in the hybrid interface circuit. A line echo canceller containing an adaptive filter is used to cancel the line echo in the inbound signal based on the learned line echo path characteristics captured in the training period. During a brief period after the telephony system is activated, the line echo canceller enters into a calibration mode wherein its adaptive filter is trained to learn the line echo path characteristics. During the calibration mode, the line echo canceller generates a calibration signal as the outbound signal and receives the line echo from the hybrid interface circuit to perform learning of the line echo path characteristics.

Abstract: An embodiment of an acoustic echo cancellation system is disclosed. The system comprises an echo cancellation unit, a second filter and a subtraction unit. The echo cancellation unit comprises a first attenuator, a first filter and a first subtractor. The first attenuator has a first down-scaling factor for attenuating a first signal. The first filter generates a first echo signal estimate based on the attenuated first signal. The first subtractor generates a third signal by subtracting the first echo signal estimate from a second signal. The second filter generates a second echo signal estimate based on the first signal. The subtraction unit subtracts the second echo signal estimate from the third signal.

Abstract: A voice communication device with an integrated framework structure for echo cancellation and noise reduction is disclosed. A microphone receives a local input signal while a speaker is outputting a local output signal. The local input signal and output signal are all decomposed into a plurality of subband signals by filter banks for conducting individual processing of echo cancellation and noise reduction per subband. The subband echo canceller is followed by a DFT unit to split the cancellation result into a plurality of narrow frequency bins whereby the noise reduction is performed. The noise reduction results are recombined by an IDFT unit for residual echo removal in a subband non-linear processor. The final output is obtained from a synthesis filter bank that synthesizes the subband signals after echo cancellation and noise reduction into a full-band signal.

Abstract: A device includes a main body, a microphone module, and an acoustic guard. The microphone module is connected to the main body and comprises a first acoustic opening and a second acoustic opening, wherein the microphone module receives sound waves via both the first and second acoustic openings. The acoustic guard covers the second acoustic opening and comprises a first sound inlet via which the sound waves enter the acoustic guard and reach the second acoustic opening.

Abstract: A hands-free voice communication apparatus with integrated speakerphone and earpiece is described. The speakerphone includes a loudspeaker and zero, one or multiple microphones. The earpiece includes a small loudspeaker, one or more microphones, a wireless module, and a digital signal processing (DSP) module. The hands-free device support hands-free voice communication via the speakerphone or the earpiece. The earpiece is functional when removed from the speakerphone case and may be used for voice communication with privacy. When the earpiece is placed back in the speakerphone case, the speakerphone is functional and may be used for voice communication with convenience. The DSP module performs digital signal processing for both the earpiece and the speakerphone, e.g., using different sets of parameters. The speakerphone may be a standalone device or may be part of a communication device. The wireless module communicates with the communication device, which provide end-to-end communication with a far-end user.

Abstract: The invention provides a system for calibrating phase and gain mismatches of an array microphone. The array microphone is installed in a voice interface device and comprises a plurality of microphones. The system comprises a loudspeaker and a computing equipment. The loudspeaker plays a segment of sound to be received by the array microphone. The computing equipment controlls the voice interface device which converts the segment of sound to a plurality of audio signals with the microphones of the array microphone, records the audio signals outputted by the voice interface device at bypass mode without any signal processing, calculates delays between the audio signals, and instructs the voice interface device to adjust phase mismatches between the audio signals according to the delays.

Abstract: A small-footprint hands-free speakerphone apparatus includes a loudspeaker, at least one main microphone, at least one reference microphone, and openings for receiving sound. The loudspeaker is disposed within a speaker chamber. Each microphone is mounted in a microphone boot formed by an acoustic opaque resilient material. Each main microphone may be omni-directional or directional. Each reference microphone is omni-directional. An opening is formed in front of each microphone for receiving sound for the microphone. An opening is formed in back of each directional main microphone for receiving sound and forming directivity for the microphone. An opening is formed in the microphone boot for each reference microphone to receive sound from the loudspeaker. An opening may also be formed in the loudspeaker chamber for passing sound to the reference microphone. The apparatus may include one or more interface units to provide interface to external devices.

Abstract: The invention provides a beam forming method for a small array microphone apparatus to generate cone beam pattern by processing a combined bi-directional beam pattern of two virtual bi-directional microphones formed through at least three omni-directional microphones arranged in an L-shape. The invention also provides a small array microphone apparatus using the beam forming method according to the invention to suppress noise by processing a combined bi-directional beam pattern of two virtual bi-directional microphones formed through at least three omni-directional microphones arranged in an L-shape, thereby outputting a clear audio signal with cone beam pattern.

Abstract: A broadside small array microphone beamforming unit comprises a first omni-directional microphone to generate a signal X1(t), a second omni-directional microphone to generate a signal X2(t), a first delay unit delaying the signal X1(t) to generate a signal X1(t?T), a second delay unit delaying the signal X2(t) to generate a signal X2(t?T), a first substrator subtracting the signal X1(t?T) from the signal X2(t) to generate a signal R(t)=X2(t)?X1(t?T), a second substrator subtracting the signal X2(t?T) from the signal X1(t) to generate a signal L(t)=X1(t)?X2(t?T), a third delay unit delaying the signal R(t) to generate a signal R?(t)=R(t?D), a gain function unit convoluting the signal L(t) with a gain function G(t) to generate a signal L?(t)=L(t)*G(t?i), and a substrator subtracting the signal L?(t) from the signal R?(t) to generate a signal B?(t)=R?(t)?L?(t).

Abstract: A microphone module includes a microphone and a boot. The microphone has a front surface and a rear surface. The boot includes an internal storage space with the microphone disposed therein, an acoustic passage connected to the internal storage space, and an opening also connected to the internal storage space. The acoustic passage runs substantially parallel to the front surface of the microphone, and the opening exposes the rear surface of the microphone.

Abstract: An electronic device includes a case, a sound absorbing tunnel, and a uni-directional microphone. The case defines a front acoustic opening and a rear acoustic opening. The sound absorbing tunnel is disposed in the case. The uni-directional microphone is disposed in the case, including a front to receive external sound via the front acoustic opening, and a rear to receive the external sound via the rear acoustic opening and the sound absorbing tunnel.

Abstract: A microphone array comprises a circuit board, a first microphone, and a second microphone. The circuit board comprises a first layer, a third layer, and a second layer sandwiched between the first and third layers. The first layer comprises a first shielding part with a fixed electric potential. The third layer comprises a second shielding part with the fixed electric potential. The second layer comprises an electrically conductive part running between the first and second shielding parts. The first microphone is attached to the first layer of the circuit board. The second microphone is attached to the first layer of the circuit board and electrically connected to the first microphone through the electrically conductive part of the second layer of the circuit board.

Abstract: The invention provides an Internet communication device. The Internet communication device plays a remote audio signal received via a network and transmits an audio signal back to the remote party to complete the communication. The Internet communication device comprises a line-in speech detection module and a line-in channel control module. The line-in speech detection module detects whether the remote audio signal is speech or not to generate a remote speech detection result. The line-in channel control module then attenuates the remote audio signal if the remote speech detection result indicates that the remote audio signal is not speech, thus, all noise including non-stationary noise is removed from the remote audio signal.

Abstract: The invention provides a method for mixing signals with an analog-to-digital converter. The analog-to-digital converter receives a plurality of analog signals. First, the plurality of analog signals are respectively converted to a plurality of first datastreams with a plurality of first delta-sigma modulators. The plurality of first datastreams are then mixed to generate at least one second datastream. The at least one second datastream is then converted to at least one third datastream with at least one second delta-sigma modulator. The at least one second delta-sigma modulator and the plurality of first delta-sigma modulators are triggered by the same clock signal.