A microphone system, comprises a first transducer, for generating a first acoustic signal, and a second transducer, for generating a second acoustic signal. A high-pass filter receives the first signal and generates a first filtered signal, and a low-pass filter receives the second signal and generates a second filtered signal. An adder forms an output signal of the microphone system as a sum of the first filtered signal and the second filtered signal.

A bias structure includes a reference voltage node connected to gate structures of a first NMOS transistor and a second NMOS transistor, a bias voltage node comprising a bias voltage, and a first op amp having a first input connected to the reference voltage, a second input connected to a drain of the first NMOS transistor, and an output connected to gate structures of a first PMOS transistor and a second PMOS transistor. The bias structure further includes a second op amp having a first input connected to the reference voltage, a second input connected to a drain of the second NMOS transistor, and an output connected to a gate structure of a third NMOS transistor and the bias voltage node. The first NMOS transistor matches a transistor of a differential pair of an integrated circuit device.

An example method of operation may include identifying speakers and microphones connected to a network controlled by a controller, assigning a preliminary output gain to the speakers used to apply test signals, measuring ambient noise detected from the microphones, recording chirp responses from all microphones simultaneously based on the test signals, deconvolving all chirp responses to determine a corresponding number of impulse responses, and measuring average sound pressure levels (SPLs) of each of the microphones to obtain a SPL level based on an average of the SPLs.

Example electronic devices that may be implemented to generate an audio output via a resonance are disclosed. A received resonance sample of an audio output is categorized with a neural network to obtain an inference profile. The received resonance sample corresponding with an equalization setting. The equalization setting for the audio output is adjusted based on the inference profile.

An information processing device (100) including a control unit that causes, on a basis of vibration output characteristics of a vibration device (200), the vibration device (200) to output a vibration corresponding to sound data.

In various applications, the system provides a method for processing audio signals, including: receiving, by a processor, a digital audio signal from a recorded audio file; analyzing, by the processor, the digital audio signal to identify pitch distortion caused by changes in momentary sound level; determining, by the processor, an amount of compensation of the audio signal to correct the identified pitch distortion; dynamically adjusting, by the processor, the digital audio signal by the compensation amount to correct the identified pitch distortion; and outputting, by the processor, the digital audio signal to an audio transducer device of a listener to improve a listening experience for the listener of the recorded audio file.

The present disclosure provides a limiter system. The limiter system includes a first equalization filter, a low-pass filter, a first limiter, a high-pass filter, a second limiter, and a mixer. The limiter system provided by the present disclosure further includes a second equalization filter. An audio signal from a signal source first passes through the first equalization filter, the signal equalized for the first time is divided into two signals, one signal is processed by the low-pass filter and the first limiter, the other signal is processed by the high-pass filter and the second limiter, and then the two processed signals enter the mixer to be mixed and outputted. The mixed output signal is subjected to second equalization filtering by the second equalization filter to avoid clipping distortion or to obtain a higher maximum sound level.

A limiter system for an active speaker may include at least one lowpass filter configured to receive an input signal and output a signal lower than a crossover frequency, at least one highpass filter, configured to receive an input signal and output a signal higher than the crossover frequency, a first allpass filter configured to adjust the phase of the signal lower than the crossover frequency, a second allpass filter configured to adjust the phase of the signal higher than the crossover frequency, a first limiter, configured to receive and limit the signal from the first allpass filter, a second limiter, configured to receive and limit the signal from the second allpass filter, and a mixer, configured to mix the signal lower from the first limiter and the signal from the second limiter.

A method for generating a perceptual equalization (EQ) filter applicable to an audio signal to equalize the audio signal, including: generating a full EQ filter for use in performing full equalization on the signal; and modifying the frequency-amplitude spectrum of the full EQ filter in accordance with a dip detection threshold function, thereby generating the perceptual EQ filter, where the dip detection threshold function is indicative of minimum perceivable amplitude of each of at least a number of different dips in the frequency-amplitude spectrum of an acoustic signal. Also, a method for equalizing an audio signal, including: generating a full EQ filter for use in performing full equalization on the signal, modifying the frequency-amplitude spectrum of the full EQ filter in accordance with at least one dip detection threshold value, thereby generating a perceptual EQ filter, and applying the perceptual EQ filter to perceptually equalize the signal.

A microphone system, comprises a first transducer, for generating a first acoustic signal, and a second transducer, for generating a second acoustic signal. A high-pass filter receives the first signal and generates a first filtered signal, and a low-pass filter receives the second signal and generates a second filtered signal. An adder forms an output signal of the microphone system as a sum of the first filtered signal and the second filtered signal.