An example method includes controlling an audio source to generate a test tone, controlling a plurality of audio sensors to sense the test tone simultaneously, receiving an output signal from each audio sensor, and determining an acoustic characteristic of each audio sensor based at least in part on the received output signals. The method also includes determining a difference between the acoustic characteristic and a corresponding reference value, identifying at least one audio sensor for which a difference corresponding to the at least one audio sensor is within a predetermined range of the reference value, and generating a compensation factor of the at least one audio sensor based at least in part on the respective output signal of the at least one audio sensor.

An audio synchronization method includes: receiving a first audio signal from a first recording device; receiving a second audio signal from a second recording device; performing a correlation operation upon the first audio signal and the second audio signal to align a first pattern of the first audio signal and the first pattern of the second audio signal; after the first patterns of the first audio signal and the second audio signal are aligned, calculating a difference between a second pattern of the first audio signal and the second pattern of the second audio signal; and obtaining a starting-time difference between the first audio signal and the second audio signal for audio synchronization according to the difference between the second pattern of the first audio signal and the second pattern of the second audio signal.

The present disclosure discloses a method and an apparatus for testing a speaker, an electronic device and a storage medium. A specific implementation includes: obtaining first audio data recorded by a microphone integrated with the speaker in ambient white noise; analyzing the first audio data to derive a first analysis result; and determining whether there is a defect in the microphone according to the first analysis result. Hence, these allow for testing a completed set on an assembled speaker to ensure the consistency of a microphone test and improve the accuracy of the test result.

Focusing sound signals in a shared 3D space uses an array of physical microphones, preferably disposed evenly across a room to provide even sound coverage throughout the room. At least one processor coupled to the physical microphones does not form beams, but instead preferably forms 1000's of virtual microphone bubbles within the room. By determining the processing gains of the sound signals sourced at each of the bubbles, the location(s) of the sound source(s) in the room can be determined. This system provides not only sound improvement by focusing on the sound source(s), but with the advantage that a desired sound source can be focused on more effectively (rather than steered to) while un-focusing undesired sound sources (like reverb and noise) instead of rejecting out of beam signals. This provides a full three dimensional location and a more natural presentation of each sound within the room.

The invention discloses a test circuit, a test method and an audio codec for testing a microphone module that includes a first microphone and a second microphone. The first microphone outputs the first data, and the second microphone outputs the second data. The test circuit includes a comparison circuit, a counter, and a decision circuit. The comparison circuit is configured to compare the first data with the second data and generate a comparison result. The counter is coupled to the comparison circuit and configured to generate a count value based on the comparison result. The decision circuit is coupled to the counter and configured to indicate, based on the count value and a threshold value, whether the microphone module has an error.

Implementations can detect respective audio data that captures an acoustic event at multiple assistant devices in an ecosystem that includes a plurality of assistant devices, process the respective audio data locally at each of the multiple assistant devices to generate respective measures that are associated with the acoustic event using respective event detection models, process the respective measures to determine whether the detected acoustic event is an actual acoustic event, and cause an action associated with the actional acoustic event to be performed in response to determining that the detected acoustic event is the actual acoustic event. In some implementations, the multiple assistant devices that detected the respective audio data are anticipated to detect the respective audio data that captures the actual acoustic event based on a plurality of historical acoustic events being detected at each of the multiple assistant devices.

A method and apparatus are provided for generating a directional output signal from sound received by at least two microphones arranged as microphone array. The method includes transforming the sound received by each of said microphones and represented by analog-to-digital converted time-domain signals provided by each of said microphones into corresponding complex-valued frequency-domain microphone signals each having a frequency component value for each of a plurality of frequency components, and calculating, for each of the plurality of frequency components of the complex-valued frequency-domain microphone signal of at least one of said microphones, a respective tolerance compensated frequency component value by multiplying the frequency component value of the complex-valued frequency-domain microphone signal of said microphone with a frequency-specific real-valued correction factor.

According to certain embodiments, a microphone array having a plurality of microphone elements is calibrated by ensonifying the microphone array at a first direction relative to the microphone array with a first acoustic signal to concurrently generate a first set of audio signals from two or more of the microphone elements and processing the first set of audio signals to calibrate the two or more microphone elements. One or more other sets of audio signals can be generated by ensonifying the microphone array with one or more other acoustic signals at one or more other directions relative to the microphone array, where the two or more microphone elements are calibrated using the first set and the one or more other sets of audio signals. The calibration process can be performed outside of an anechoic chamber using one or more acoustic sources located outside or inside the microphone array.

Microphone capsules for condenser or electret microphones often exhibit individual deviations from a desired ideal behavior, e.g. the frequency response and phase response. Particularly if a plurality of microphone capsules are interconnected to form a microphone array, suitable microphone capsules must be found in a selection process. Some of these deviations can be corrected electronically, e.g. by filtering with a corresponding filter that is individually adapted. An improved microphone capsule, with which an automatic selection and automatic assembly of circuit boards with microphone capsules is facilitated, comprises an electrostatic sound transducer, an amplifier element that outputs an amplified output signal of the electrostatic sound transducer, and at least one electronic memory element. Data obtained by a measurement and relating to the individual frequency response or phase response of the respective microphone capsule can be stored therein. The data can be read out during manufacturing and during operation, which enables automatic sorting of the capsules during production and automatic calibration of the target circuit in operation.

An electronic device according to an embodiment may include: a first sound input device configured to obtain external sound and produce a first signal and a processor operatively connected to the first sound input device. The processor may be configured to: receive the first signal from the first sound input device; produce a first high-frequency signal by passing the first signal through a high-pass filter to; determine a first energy value of the first high-frequency signal; determine a second energy value of the first signal; compare a product of the second energy value of the first signal and the first energy value of the first high-frequency signal with a first threshold value to produce a first result; and determine whether the first sound input device is blocked based on the first result. In addition, various other embodiments may be provided.