Abstract: An audio apparatus is provided. The audio apparatus includes at most one electroacoustic transducer; and an audio controller, coupled to the electroacoustic transducer, for actively controlling the electroacoustic transducer to function as a loudspeaker or a microphone, wherein the loudspeaker converts output electrical signals to output sounds, and the microphone converts input sounds to input electrical signals.

Abstract: An audio device is provided, employing a defect detection method to detect defectiveness within a microphone array. The microphone array comprising a first microphone and a second microphone, respectively, generates a first audio signal and a second audio signal from ambient audio signals. An error detector is provided to detect functions of the first and second microphones based on the first and second audio signals to generate a status signal. A digital signal processor (DSP) processes the first and second audio signals based on the status signal. If the status signal indicates that only the first microphone or the second microphone is defective, the DSP switches to a single microphone mode in which only the remaining normal microphone is enabled. If the status signal indicates that both the first and second microphones are defective, the DSP generates an error indication signal and stops processing the first and second audio signals.

Abstract: The invention provides a phase calibration module, calibrating phase mismatch between microphone signals output by a plurality of microphones of an array microphone. In one embodiment, the phase calibration module comprises a subband filter, a delay calculation module, and a delay compensation filter. The subband filter extracts a high frequency component and a low frequency component from each of the microphone signals to obtain a plurality of high-frequency component signals and a plurality of low-frequency component signals. The delay calculation module calculates delays between the low-frequency component signals. The delay compensation filter then compensates the low-frequency component signals for phase mismatches therebetween according to the calculated delays to obtain a plurality of calibrated low-frequency component signals.

Abstract: A silicon-based microphone structure, flip-chip mounted a substrate, has a backside and a via hole formed on the backside. A conductive layer covers the backside and the inner walls of the via hole. A solder ring is attached to the silicon-based microphone structure and electrically connected to the conductive layer.

Abstract: The invention provides an apparatus capable of automatic volume adjustment. In one embodiment, the apparatus comprises a speaker, an array microphone located in the vicinity of the speaker, a beamforming module, a signal-to-noise ratio calculation module, and a volume adjustment module. The speaker first broadcasts a first audio signal. The array microphone converts a sound into a plurality of second audio signals. The beamforming module derives a speaker sound signal and an ambient noise signal from the second audio signals, wherein the speaker sound signal mainly comprises speaker sound components generated by the speaker and the ambient noise signal mainly comprises noises other than the speaker sound components. The signal-to-noise ratio calculation module calculates a ratio of a first power of the speaker sound signal to a second power of the ambient noise signal to obtain a signal-to-noise ratio.

Abstract: The invention provides a dynamic range control module installed in a speech processing apparatus. In one embodiment, the dynamic range control module comprises a buffer, a voice activity detector, a peak calculation module, and an amplitude adjusting module. The buffer buffers a speech signal to obtain a delayed speech signal. The voice activity detector determines a syllable from the delayed speech signal. The peak calculation module calculates peak amplitude of the syllable. The amplitude adjusting module determines an attenuation factor corresponding to the syllable according to the peak amplitude in the syllable, and adjusts amplitude of the whole syllable with the same gain according to the attenuation factor to obtain an adjusted speech signal.

Abstract: The invention provides an electronic apparatus. In one embodiment, the electronic apparatus comprises a video camera, a video processing module, an array microphone, and an audio processing module. The video camera adjusts a focal length according to a zoom-in/out control signal and captures an image of a plurality of target objects according to the focal length to generate a raw video signal. The video processing module generates the zoom-in/out control signal to adjust the focal length of the video camera. The array microphone comprises a plurality of microphones converting sounds of the target objects into a plurality of raw audio signals. The audio processing module adjusts a beamwidth for beamforming according to the zoom-in/out control signal, and performs a beamforming process according to the beamwidth on the raw audio signals to generate a first audio signal comprising mainly voice components of the target objects.

Abstract: The invention provides a method for directing operation of a microphone system. In one embodiment, the microphone system comprises a plurality of component modules. First, a diagnostic test is performed to determine a diagnostic result indicating whether the component modules have failed the diagnostic test. Whether a plurality of required component modules corresponding to a current application mode for operating the microphone system have failed the diagnostic test is then determined according to the diagnostic result, wherein the application mode requires cooperation of the required component modules selected from the component modules of the microphone system.

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: An audio processing apparatus is provided, comprising: a main microphone for receiving sounds from a source and noises from non-source sources and generating a main input; a reference microphone for receiving the sounds and the noises and generating a reference input; a short-time Fourier transformation (STFT) unit for applying short time Fourier transformation to convert the main input of a time domain signals into a main signal of a frequency domain and convert the reference input of the time domain signals into a reference signal of the frequency domain; a sensitivity calibrating unit for performing sensitivity calibration on the main signal and the reference signal and generating a main calibrated signal and a reference calibrated signal; and a voice active detector (VAD) for generating a voice active signal according to the main calibrated signal, the reference calibrated signal and a direction of arrival (DOA) signal.

Abstract: The invention provides a microphone circuit. In one embodiment, the microphone circuit comprises a transducer, a biasing resistor, a pre-amplifier, and a switch circuit. The transducer is coupled between a ground and a first node for converting a sound into a voltage signal output to the first node. The biasing resistor is coupled between the ground and the first node. The pre-amplifier is biased with a biasing voltage and coupled between the first node and a second node, and amplifies the voltage signal to obtain an output signal at the second node. The switch circuit is coupled between the first node and the ground, couples the first node to the ground when the microphone circuit is reset, and decouples the first node from the ground after a voltage status of the microphone circuit is stable, thus clamping a voltage of the first node to the ground to prevent generation of a popping noise when the microphone circuit is reset.

Abstract: The invention provides a method for analog-to-digital conversion in a microphone circuit. First, a first gain is determined. A first analog signal is then amplified according to the first gain to obtain a second analog signal. The second analog signal is then converted from analog to digital to obtain a first digital signal. A second gain is then determined according to the first gain so that a product of the first gain and the second gain is kept constant. The first digital signal is then amplified according to the second gain to obtain a second digital signal.

Abstract: A method for showing an array microphone effect includes the steps of obtaining an original acoustic signal from array microphones, and visualizing the original acoustic signal to obtain a figure. The original acoustic signal includes a crystal voice, out-beam noises, background noise, and/or an echo. The figure includes a plurality of graphic components representing the crystal voice, out-beam noise, background noise, and/or echo, respectively.

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: An array microphone system includes a first omni-directional microphone, a second omni-directional microphone, a gain control, and a beam former. The first omni-directional microphone faces a first direction. The second omni-directional microphone faces a second direction opposing the first direction. When receiving sound, the first omni-directional microphone and the second omni-directional microphone respectively generate a first signal and a second signal. The gain control amplifies the second signal to transform into a third signal, wherein strength of the third signal is equal to that of the first signal when the sound comes from the first direction. The beam former separates an in-beam sound signal and an out-beam sound signal from the first signal and the third signal.

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 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: The invention provides a sound processing device. In one embodiment, the sound processing device comprises a first microphone, a first analog-to-digital converter, a second microphone, and a second analog-to-digital converter. The first microphone detects a first sound pressure to generate a first analog audio signal. The first analog-to-digital converter converts the first analog audio signal to a first digital audio signal. The second microphone detects a second sound pressure to generate a second analog audio signal. The second analog-to-digital converter converts the second analog audio signal to a second digital audio signal, encodes a third digital audio signal according to the second digital audio signal, receives the first digital audio signal and a clock signal, outputs data bits of the third digital audio signal when the clock signal oscillates to a logic low level, and outputs data bits of the first digital audio signal when the clock signal oscillates to a logic high level.

Abstract: The invention provides an analog-to-digital converter. In one embodiment, the analog-to-digital converter receives a first audio signal from a microphone sensor, and receives a first channel selection signal and a clock signal, and comprises a toggle detection module, a first data processing module, a second data processing module, and a multiplexer. The toggle detection module detects whether the first channel selection signal is toggling between a logic low level and a logic high level to generate a control signal. The first data processing module processes the first channel selection signal to generate a second channel selection signal. The second data processing module converts the first audio signal from analog to digital to generate a second audio signal.

Abstract: The invention provides a sound processing device. In one embodiment, the sound processing device comprises a first microphone, a first analog-to-digital converter, a second microphone, and a second analog-to-digital converter. The first microphone detects a first sound pressure to generate a first analog audio signal. The first analog-to-digital converter converts the first analog audio signal from analog to digital to obtain a first digital audio signal. The second microphone detects a second sound pressure to generate a second analog audio signal.