Abstract: A system configured to perform acoustic feedback control to enable a device to perform hearing enhancement while suppressing acoustic feedback. During hearing enhancement, the device may amplify environmental noise based on a unique hearing profile associated with the user, personalizing equalization settings, a dynamic range, and/or other characteristics to optimize playback audio for the user. The acoustic feedback control may include an acoustic feedback cancellation (AFC) component that uses an adaptive filter to estimate and cancel a feedback signal. In addition, the AFC component may perform entrainment prevention by detecting periodic signals and adjusting an adaptation rate of the adaptive filter accordingly. Separately, the device may selectively suppress acoustic feedback by detecting frequency bands representing acoustic feedback (e.g., squeal detection) and applying one or more notch filter(s) to suppress the selected frequency bands (e.g., squeal suppression).

Abstract: Techniques for presence-detection devices to detect movement of a person in an environment by emitting ultrasonic signals using a loudspeaker that is concurrently outputting audible sound. To detect movement by the person, the devices characterize the change in the frequency, or the Doppler shift, of the reflections of the ultrasonic signals off the person caused by the movement of the person. However, when a loudspeaker plays audible sound while emitting the ultrasonic signal, audio signals generated by microphones of the devices include distortions caused by the loudspeaker. These distortions can be interpreted by the presence-detection devices as indicating movement of a person when there is no movement, or as indicating lack of movement when a user is moving. The techniques include processing audio signals to remove distortions to more accurately identify changes in the frequency of the reflections of the ultrasonic signals caused by the movement of the person.

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: A microphone array is provided. The microphone array is disposed on an electrical device, and includes: at least one built-in microphone, being built in the electrical device, having a first frequency spectrum; and an audio processor, coupled to the built-in microphone, for coupling to an external transducer, including: a spectrum estimation unit for estimating a second frequency spectrum of the external transducer.

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: 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: A microphone apparatus is provided, including a body, a main microphone and a reference microphone. The main microphone and a reference microphone are disposed on the body for receiving a sound from a source and a noise from other than the source, wherein the main microphone and the reference microphone are arranged vertically towards the source.

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: The invention provides a full-duplex communication device. In one embodiment, the full-duplex communication device comprises a first adaptive filter, a second adaptive filter, a channel decoupling module, and a frequency processing module. The first adaptive filter having a first tap length filters out echoes of a far-end talker from a first near-end signal carrying voices of a near-end talker according to a far-end signal carrying voices of the far-end talker to obtain a second near-end signal. The second adaptive filter having a second tap length less than the first tap length filters out echoes of the far-end talker from the first near-end signal according to the far-end signal to obtain a third near-end signal. The channel decoupling module processes the second near-end signal to generate a fourth near-end signal and subtracts the second near-end signal from the third near-end signal to obtain a fifth near-end 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: 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: 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: The invention provides a full-duplex communication device. In one embodiment, the full-duplex communication device comprises a first adaptive filter, a second adaptive filter, a channel decoupling module, and a frequency processing module. The first adaptive filter having a first tap length filters out echoes of a far-end talker from a first near-end signal carrying voices of a near-end talker according to a far-end signal carrying voices of the far-end talker to obtain a second near-end signal. The second adaptive filter having a second tap length less than the first tap length filters out echoes of the far-end talker from the first near-end signal according to the far-end signal to obtain a third near-end signal. The channel decoupling module processes the second near-end signal to generate a fourth near-end signal and subtracts the second near-end signal from the third near-end signal to obtain a fifth near-end signal.

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: 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: 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 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: Techniques for canceling echo and suppressing noise using an array microphone and signal processing. In one system, at least two microphones form an array microphone and provide at least two microphone input signals. Each input signal may be processed by an echo canceller unit to provide a corresponding intermediate signal having some echo removed. An echo cancellation control unit receives the intermediate signals and derives a first gain used for echo cancellation. A noise suppression control unit provides at least one control signal used for noise suppression based on background noise detected in the intermediate signals. An echo cancellation and noise suppression unit derives a second gain based on the control signal(s), cancels echo in a designated intermediate signal based on the first gain, and suppresses noise in this intermediate signal based on the second gain. The signal processing may be performed in the frequency domain.

Abstract: A method incorporating the use of a filter that accepts simultaneous masking signals and generates a close replica of temporal masking signals derived from the input simultaneous masking signals. The filter output is then added to the filter input to provide a composite masking signal. This composite masking signal may then be used to establish overall masking threshold levels which can be mapped in the appropriate subband to significantly reduce the amount of coding quantization required without significantly affecting the perceived sound of the reconstructed broadband signal. The filter's transfer function and impulse response define a filter the output of which exhibits two principal characteristics of temporal masking. One such characteristic is decay with the logarithm of time. The other is a rate of decay that is inversely proportional to the duration of the corresponding simultaneous masking.

Abstract: A system for improved digital data compression in an audio encoder. A threshold is established which depends on the bit rate of the input data. A determination is made whether the bit rate is above or below the established threshold. A masking index is calculated for the input data according to a first formula if the input data is being transmitted at a rate at or below the threshold. A second formula is used to calculate the masking index if the input data is being transmitted at a rate above the threshold. The masking index is used to generate a masking threshold, and data deemed insignificant relative to the masking threshold is ignored. In the preferred embodiment of the present invention, a psycho-acoustic modeler, which is included in the encoding section of an encoding/decoding (CODEC) circuit, is used to determine a masking index. The masking index is then used to generate a masking threshold.