A distributed automatic level control function is provided, in which information relating to a common automatic level control parameter is transmitted to each of a plurality of microphone devices, wherein the information transmitted to at least one microphone device is derived from an audio sample of at least one different microphone device. Each microphone device produces the common automatic level control parameter based on the information received by the microphone device and applies the common automatic level control parameter produced by the microphone device to a distributed automatic level controller of the microphone device.

A system, a method for interleaving a digital audio signal including a PWM signal, and a digital audio processing device employing the same in accordance with the present invention are disclosed. The system, the method and the digital audio processing device is capable of minimizing the interference with an RF signal such as an AM signal or an FM signal by interleaving the digital audio signal including the PWM signal.

A digital synthesizer is described that comprises: a ramp generator configured to generate a signal of frequency control words, FCW, that describes a desired frequency modulated continuous wave; a digitally controlled oscillator, DCO configured to receive the FCW signal; a feedback loop; and a phase comparator coupled to the ramp generator and configured to compare a phase of the FCW output from the ramp generator and a signal fed back from the DCO via the feedback loop and output a N-bit oscillator control signal. The digital synthesizer comprises a gain circuit coupled to a multiplier located between the ramp generator and the DCO and configured to apply at least one gain from a plurality of selectable gains to the N-bit oscillator control signal that set a selectable loop gain of the digital synthesizer and thereby set a selectable loop bandwidth; and calculate and apply a gain offset dependent upon the selected gain that is adapted when the selected gain is changed.

Audio information of audio content being listened to by a user is received. An aspect of a listening environment of the user is identified. A volume preset, based on the audio information and the aspect of the listening environment, is determined to be available. A first volume of the audio content being listened to by the user is determined to be different from the volume preset. The first volume of the audio content is adjusted to a second volume.

The present application relates to a method and a circuit for dynamic range control based on peak level and RMS level double-level detection. The method includes: processing an input signal by an equalizer, to obtain an equalized signal; detecting a signal peak level and a signal RMS level of the equalized signal, to obtain the signal peak level and the signal RMS level; adjusting a parameter configuration of a dynamic range controller according to the signal peak level and the signal RMS level; adjusting a dynamic gain of the dynamic range controller according to the parameter configuration, and adjusting the equalized signal by using the adjusted gain, and then outputting the adjusted equalized signal. The method can not only control the dynamic range of the signal but also improve the clipping distortion of the signal.

One aspect of the present disclosure relates to a method for operating an amplifier, the amplifier including a variable resistor coupled between a source of a first input transistor and a source of a second input transistors, and a variable capacitor coupled between the source of the first input transistor and the source of the second input transistor. The method includes adjusting a resistance of the variable resistor to adjust a low-frequency gain of the amplifier, and adjusting a capacitance of the variable capacitor in an opposite direction as the adjustment to the resistance of the variable resistor.

A system and method for digital processing including a gain element to process an input audio signal, a high pass filter to then filter the signal and create a high pass signal, a first filter module to filter the high pass signal and create a first filtered signal and a splitter to split the high pass signal into two high pass signals. The first filter module filters one high pass signals before a first compressor modulates the signal or a high pass signal to create a modulated signal. A second filter module filters the modulated signal to create a second filtered signal that is processed by a first processing module including a band splitter that splits the signal into low and high band signals that are then modulated by compressors. A second processing module processes the modulated low and high band signals to create an output signal.

A speaker and a microphone are provided to a helmet main body. A phase of a sound signal passing through a high pass filter is controlled by a first phase control section, and is amplified by a first amplification section. A phase of a sound signal passing through the low pass filter is controlled by a second phase control section, and is amplified by a second amplification section. These sound signals are synthesized by a synthesis section. A drive section drives the speaker according to a synthesized sound signal. A control section switches phase inversion processing/non-inversion processing of the phase control sections and controls an amplification factor of the amplification sections based on sound power of the sound signal.

The present application relates to a method and a circuit for dynamic range control based on peak level and RMS level double-level detection. The method includes: processing an input signal by an equalizer, to obtain an equalized signal; detecting a signal peak level and a signal RMS level of the equalized signal, to obtain the signal peak level and the signal RMS level; adjusting a parameter configuration of a dynamic range controller according to the signal peak level and the signal RMS level; adjusting a dynamic gain of the dynamic range controller according to the parameter configuration, and adjusting the equalized signal by using the adjusted gain, and then outputting the adjusted equalized signal. The method can not only control the dynamic range of the signal but also improve the clipping distortion of the signal.

A sound signal processing includes obtaining sound signals; amplifying the sound signals based on a plurality of analog gain values to generate a plurality of amplified sound signals, where each amplified sound signal corresponds to a respective analog gain value; digitizing the plurality of amplified sound signals to produce plurality of digital sound signals; determining whether the plurality of digital sound signals are clipped or non-clipping, and selectively outputting the non-clipping sound signals; and applying a dynamic range control program to the non-clipping sound signals to adjust them.