Abstract: The invention provides an integrated circuit of a microphone. In one embodiment, the integrated circuit receives a first signal converted from a sound and receives a reference signal with a digital format for echo cancellation. In one embodiment, the integrated circuit comprises a pre-amplifier, an analog-to-digital converter, a digital signal processor, and a post amplifier. The pre-amplifier amplifies the first signal according to a first gain to obtain a third signal. The analog-to-digital converter converts the third signal from analog to digital to obtain a fourth signal. The digital signal processor cancels an echo component from the fourth signal according to the reference signal to obtain a fifth signal, and determines the first gain and a second gain, wherein a product of the first gain and the second gain is kept constant, and the first gain is determined so that an amplitude of the third signal is kept equal to an amplitude of the reference signal.

Abstract: Techniques for performing acoustic echo cancellation are described. An ADC oversamples an analog signal from a microphone and provides a near-end signal having a wider bandwidth than the bandwidth of a communication channel. A subband filter receives and filters the near-end signal, provides an in-band signal having spectral components in a frequency band of interest, and provides an out-of-band signal having spectral components in at least one other frequency band. An adaptive filter receives a reference signal and the in-band signal, derives an echo estimate signal with the reference signal, cancels a portion of the echo in the in-band signal with the echo estimate signal, and provides an intermediate signal. A double-talk detector detects for double talk based on the out-of-band signal and the intermediate signal, e.g., by determining a power ratio based on the powers of the out-of-band and intermediate signals and detecting for double talk based on the power ratio.

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 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 microphone preamplifier circuit is provided in a system on chip. An amplifier comprises a first input end, a second input end, and an output end. A bias voltage is provided by a bias voltage source. A first sensor is coupled to the first input end and the bias voltage source for sensing a first physical parameter and a second physical parameter. A second sensor is coupled to the second input end and the bias voltage source for sensing the first physical parameter, wherein the second sensor is insensitive to the second physical parameter. The output end of the amplifier outputs a difference of the first and second input ends whereby noises and interferences are reduced.

Abstract: A microphone preamplifier circuit is provided. An amplifier comprises a first input end, a second input end, and an output end. A bias voltage is provided by a bias voltage source. A first sensor is coupled to the first input end and the bias voltage source for sensing a first physical parameter and a second physical parameter. A second sensor is coupled to the second input end and the bias voltage source for sensing the first physical parameter, wherein the second sensor is insensitive to the second physical parameter. The output end of the amplifier outputs a difference of the first and second input ends whereby noises and interferences are reduced.

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: A multi-channel audio playback apparatus including a channel interface, a first switching amplifier and a second switching amplifier is provided. The channel interface is used to receive multi-channel digital data and generate first channel digital data and second channel digital data. The first switching amplifier is used to convert the first channel digital data into a first pulse width modulation (PWM) signal according to a first reference signal with a first frequency. The second switching amplifier is used to convert the second channel digital data into a second PWM signal according to a second reference signal with a second frequency. The second frequency is different from the first frequency.

Abstract: A digital microphone is provided, comprising a transducer for converting sound pressure into an analog electrical signal, an analog-to-digital converter for converting the analog electrical signal into a digital signal according to a clock signal, and a clock buffer circuit coupled between the analog-to-digital converter and a clock source for deducting a high frequency component from the clock signal received from the clock source.

Abstract: An improved voice recognition system in which a Voice Keyword Table is generated and downloaded from a set-up device to a voice recognition device. The VKT includes visual form data, spoken form data, phonetic format data, and an entry corresponding to a keyword, and TTS-generated voice prompts and voice models corresponding to the phonetic format data. A voice recognition system on the voice recognition device is updated by the set-up device. Furthermore, voice models in the voice recognition device are modified by the set-up device.

Abstract: The invention provides an integrated circuit attached to a microphone. In one embodiment, the integrated circuit comprises a buffer, a gain stage, an analog-to-digital converter (ADC), and a memory module. The buffer buffers a first signal generated by the microphone, and outputs the first signal as a second signal. The gain stage amplifies the second signal according to an adjustable gain to obtain a third signal. The analog-to-digital converter converts the third signal from analog to digital to obtain a fourth signal as an output of the integrated circuit. The memory module stores the adjustable gain and outputs the adjustable gain to the gain stage for controlling amplification of the gain stage.

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: 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: An electronic device includes a housing, a plurality of microphones, and a plurality of guide tubes. The plurality of microphones are disposed in the housing. The plurality of guide tubes extend from the housing toward the plurality of microphones, whereby the plurality of microphones in the housing receives external sound via the guide tubes.

Abstract: A broadside small array microphone beamforming apparatus comprises first and second omni-directional microphones, a microphone calibration unit, and a directional microphone forming unit. The first and second omni-directional microphones respectively convert voice from a desired near-end talker into first and second signals. The second and first omni-directional microphones and the desired near-end talker are respectively arranged at three points of a triangle. The microphone calibration unit receives the first and second signals and correspondingly outputs first and second calibration signals. The directional microphone forming unit receives the first and second calibration signals to generate a first directional microphone signal with a bidirectional polar pattern. The adaptive channel decoupling unit receives the first calibration signal and the first directional microphone signal to generate a first main channel signal and a first reference channel signal for noise detection.

Abstract: A microphone package includes a carrier, a cap, an integrated circuit chip, and a microphone unit. The cap covers the carrier to form a storage space. The integrated circuit chip is disposed in the storage space. The microphone unit is disposed in the storage space and stacked on the integrated circuit chip.

Abstract: A microphone package includes a carrier, a cap, a first silicon-based microphone, and a first integrated circuit chip. The carrier has a first storage space. The cap has a planar part contacting the carrier and a plurality of flanges extending from the planar part and surrounding the carrier. The first silicon-based microphone is disposed in the first storage space and covered by the cap. The first integrated circuit chip is disposed in the first storage space, electrically connected to the first silicon-based microphone, and covered by the cap.

Abstract: An improved voice recognition system in which a Voice Keyword Table is generated and downloaded from a set-up device to a voice recognition device. The VKT includes visual form data, spoken form data, phonetic format data, and an entry corresponding to a keyword, and TTS-generated voice prompts and voice models corresponding to the phonetic format data. A voice recognition system on the voice recognition device is updated by the set-up device. Furthermore, voice models in the voice recognition device are modified by the set-up device.

Abstract: Techniques for suppressing large echo due to nonlinearity, instable adaptive filter, and so on. An echo cancellation system includes an echo canceller unit, a nonlinear echo detection unit, and a nonlinear echo suppression unit. The echo canceller unit (e.g., an adaptive filter) receives a reference signal and a near-end signal, derives an echo estimate signal based on the reference and near-end signals, cancels a portion of echo in the near-end signal based on the echo estimate signal, and provides an intermediate signal having residual echo. The nonlinear echo detection unit detects for large nonlinear echo in the echo estimate signal. The nonlinear echo suppression unit suppresses large nonlinear echo in the first intermediate signal based on a variable gain, which may be adjusted lower if large nonlinear echo is detected and higher otherwise.