Abstract: An acoustic apparatus includes a back plate, a diaphragm, and at least one pillar. The diaphragm and the back plate are disposed in spaced relation to each other. At least one pillar is configured to at least temporarily connect the back plate and the diaphragm across the distance. The diaphragm stiffness is increased as compared to a diaphragm stiffness in absence of the pillar. The at least one pillar provides a clamped boundary condition when the diaphragm is electrically biased and the clamped boundary is provided at locations where the diaphragm is supported by the at least one pillar.

Abstract: An ultrasonic actuation apparatus includes a piezoelectric transducer producing a first ultrasonic signal; a second transducer; and a platen, the platen being directly and/or acoustically coupled to the piezoelectric transducer and the second transducer. The second transducer may be a MEMS microphone. The second transducer is configured to receive the first ultrasonic signal at a first time, and a second ultrasonic signal at second time. The second ultrasonic signal has been modified from the first ultrasonic signal in correspondence with an object being in contact with the platen.

Abstract: The disclosure relates to a microphone assembly comprising a multibit analog-to-digital converter configured to receive, sample, and quantize a microphone signal to generate N-bit digital microphone samples representative of the microphone signal at a first sampling frequency. The microphone assembly also comprises a first digital-to-digital converter configured to receive, quantize, and noise-shape the N-bit digital microphone samples to generate a corresponding M-bit Pulse Density Modulated (PDM) signal, wherein N and M are positive integers, and N>M. The microphone assembly may comprise a SoundWire compliant data interface configured to repeatedly receive samples of the M-bit PDM signal and write bits of the M-bit PDM signal to a SoundWire data frame.

Abstract: The present disclosure generally relates to acoustic assemblies. One acoustic assembly includes a base and a first die disposed on the base. The first die comprises a microelectromechanical system (MEMS) microphone that includes a first diaphragm and a first back plate. The MEMS microphone has a barometric release. The acoustic assembly also includes a second die disposed on the base. The second die comprises a pressure sensor. The acoustic assembly further includes a cover coupled to the base and enclosing the first dies and the second die. A back volume is formed between the base, the first die, the second die, and the cover. The pressure sensor is configured to sense a pressure of the back volume.

Abstract: The present disclosure relates generally to microphones and related components. One example micro electro mechanical system (MEMS) motor includes a first diaphragm; a second diaphragm that is disposed in generally parallel relation to the first diaphragm, the first diaphragm and second diaphragm forming an air gap there between; and a back plate disposed in the air gap between and disposed in generally parallel relation to the first diaphragm and the second diaphragm.

Abstract: A microphone includes a first micro electro mechanical system (MEMS) motor, the first MEMS motor including a first diaphragm and a first back plate; and a second MEMS motor including a second diaphragm and a second back plate. The first diaphragm is electrically biased relative to the first back plate according to a first voltage, the second diaphragm is biased relative to the second back plate according to a second voltage, and a magnitude of the first voltage is different from a magnitude of the second voltage.

Abstract: Systems and methods for voice authentication are provided. An example method starts with dynamically generating authentication information and a prompt associated therefor. The authentication information is generated to emphasize differences between an authorized user and others. The authentication information may be generated based on at least one of a user profile, an acoustic environment in which a mobile device is located, and a history of interactions with the mobile device. The authentication information may be an answer to a question which the authorized user would uniquely be able to provide, or it may be a distinctive password. The prompt, for the authentication information, is provided to a user attempting to use the mobile device. The method proceeds with capturing an acoustic sound of a speech of the user and detecting the authentication information in the speech. Based on the detection, a confidence score is determined.

Abstract: Systems and methods for assisting automatic speech recognition (ASR) are provided. An example method includes generating, by a mobile device, a plurality of instantiations of a speech component in a captured audio signal, each instantiation of the plurality of instantiations being in support of a particular hypothesis regarding the speech component. At least two instantiations of the plurality of instantiations are then sent to a remote ASR engine. The remote ASR engine is configured to recognize at least one word based on the at least two of the plurality of instantiations and a user context, according to various embodiments. This recognition can include selecting one of the instantiations of the speech component from the plurality of instantiations. The plurality of instantiations may be generated by noise suppression of the captured audio signal with different degrees of aggressiveness.

Abstract: Systems and methods for acoustic echo cancellation are provided. An example method includes receiving a reference signal. The reference signal represents at least one sound captured inside an enclosure of a loudspeaker. The loudspeaker is operable to play back a far end signal. The method also includes receiving an acoustic signal. The acoustic signal represents at least one sound captured outside the enclosure of the loudspeaker. The acoustic signal includes at least a near end signal and the far end signal. The method enables attenuation, using the reference signal, the far end signal in the acoustic signal. The reference signal can be captured by a low sensitivity microphone placed inside the enclosure of the loudspeaker. The attenuation of the far end signal may include subtractive cancellation.

Abstract: At a microphone, voice activity is detected in a data stream while simultaneously buffering audio data from the data stream to create buffered data. A signal is sent to a host indicating the positive detection of voice activity in the data stream. When an external clock signal is received from the host, the internal operation of the microphone is synchronized with the external clock signal. Buffered data stream is selectively sent through a first path, the first path including a buffer having a buffer delay time representing the time the first data stream takes to move through the buffer. The data stream is continuously sent through a second path as a real-time data stream, the second path not including the buffer, the real-time data stream beginning with the extended buffer data at a given instant in time. The buffered data stream and the real-time data stream are multiplexed onto a single data line and transmitting the multiplexed data stream to the host.

Abstract: An external clock signal having a first frequency is received. A division ratio is automatically determined based at least in part upon a second frequency of an internal clock. The second frequency is greater than the first frequency. A decimation factor is automatically determined based at least in part upon the first frequency of the external clock signal, the second frequency of the internal clock signal, and a predetermined desired sampling frequency. The division ratio is applied to the internal clock signal to reduce the first frequency to a reduced third frequency. The decimation factor is applied to the reduced third frequency to provide the predetermined desired sampling frequency. Data is clocked to a buffer using the predetermined desired sampling frequency.

Abstract: A microphone includes a microelectromechanical system (MEMS) circuit and an integrated circuit. The MEMS circuit is configured to convert a voice signal into an electrical signal, and the integrated circuit is coupled to the MEMS circuit and is configured to receive the electrical signal. The integrated circuit and the MEMS circuit receive a clock signal from an external host. The clock signal is effective to cause the MEMS circuit and integrated circuit to operate in full system operation mode during a first time period and in a voice activity mode of operation during a second time period. The voice activity mode has a first power consumption and the full system operation mode has a second power consumption. The first power consumption is less than the second power consumption. The integrated circuit is configured to generate an interrupt upon the detection of voice activity, and send the interrupt to the host.

Abstract: A far end signal from a far end user and broadcasting the far end signal at a loudspeaker is received. A signal for use in an echo transfer function is determined, and the signal selected from between: the far end signal or an echo reference signal received from a near field microphone disposed in close proximity to the speaker. The near field microphone sensing the far end signal that is broadcast from the speaker while sensing the near-end speech and ambient noise at insignificant energy levels compared to the speaker signal. An echo transfer function is determined based at least in part upon the selected signal, and the echo transfer function represents characteristics of an acoustic path between the loudspeaker and a far field microphone that is disposed at a greater distance from the speaker than the near field microphone. An estimated echo is determined based at least in part upon the echo transfer function.

Abstract: The present technology substantially reduces undesirable effects of multi-level noise suppression processing by applying an adaptive signal equalization. A noise suppression system may apply different levels of noise suppression based on the (user-perceived) signal-to-noise-ratio (SNR) or based on an estimated echo return loss (ERL). The resulting high-frequency data attenuation may be counteracted by adapting the signal equalization. The present technology may be applied in both transmit and receive paths of communication devices. Intelligibility may particularly be improved under varying noise conditions, e.g., when a mobile device user is moving in and out of noisy environments.

Abstract: A surface mount package for a micro-electro-mechanical system (MEMS) microphone die is disclosed. The surface mount package features a substrate with metal pads for surface mounting the package to a device's printed circuit board and for making electrical connections between the microphone package and the device's circuit board. The surface mount microphone package has a cover, and the MEMS microphone die is substrate-mounted and acoustically coupled to an acoustic port provided in the surface mount package. The substrate and the cover are joined together to form the MEMS microphone, and the substrate and cover cooperate to form an acoustic chamber for the substrate-mounted MEMS microphone die.

Abstract: A driver, includes a driver block, a controller block, and a comparison block. The driver block includes an adjustable current source configured to produce a digital output stream. The controller block is coupled to the driver block. The comparison block is coupled to the driver block and the controller block. The comparison block is configured to compare the digital output stream to a reference value at a time delayed with respect to a master clock and based upon the comparison cause the controller block to adjust a strength of the driver block.

Abstract: Provided are systems and methods for contextual switching of microphones in audio devices. An example method includes detecting a change in conditions for capturing an acoustic signal by at least two microphones. A configuration is associated with the at least two microphones. The example method provides determining that the change in conditions has been stable for a pre-determined period of time. In response to the determination, the method changes the configuration associated with the at least two microphones. The conditions may include an absence or presence of far-end speech, reverberation, low/high signal-to-noise ratio, low/high signal-to-echo ratio, type of background noise, and so on. The changing of the configuration includes assigning a primary microphone and a secondary microphone based on a change in conditions. Based on the changed configuration, tuning parameters may be adjusted for noise suppression and acoustic echo cancellation.

Abstract: Described are noise suppression techniques applicable to various systems including automatic speech processing systems in digital audio pre-processing. The noise suppression techniques utilize a machine-learning framework trained on cues pertaining to reference clean and noisy speech signals, and a corresponding synthetic noisy speech signal combining the clean and noisy speech signals. The machine-learning technique is further used to process audio signals in real time by extracting and analyzing cues pertaining to noisy speech to dynamically generate an appropriate gain mask, which may eliminate the noise components from the input audio signal. The audio signal pre-processed in such a manner may be applied to an automatic speech processing engine for corresponding interpretation or processing.

Abstract: A die is manufactured using complementary metal-oxide semiconductor (CMOS) techniques to create transistors, electrical pathways, and microelectromechanical system (MEMS) structures. The MEMS structures include springs, plates, mechanical stops, and structural supports, which can be combined to form complex MEMS structures including microphones, pressure sensors, accelerometers, resonators, gyroscopes, and the like.

Abstract: A microelectromechanical (MEMS) microphone includes a MEMS motor and a gain adjustment apparatus. The MEMS motor includes at least a diaphragm and a charge plate and is configured to receive sound energy and transform the sound energy into an electrical signal. The gain adjustment apparatus has an input and an output and is coupled to the MEMS motor. The gain adjustment apparatus is configured to receive the electrical signal from the MEMS motor at the input and adjust the gain of the electrical signal as measured from the output of the gain adjustment apparatus. The amount of gain is selected so as to obtain a favorable sensitivity for the microphone.