Provided are an adaptive equalization apparatus and a method using the same of, for optimizing digital algorithm, responding proactively in dynamic environmental change, and adjusting the monitoring range according to the signal size. The adaptive equalization apparatus includes a first equalizer filter for compensating and outputting for components of a high frequency band of an input signal; a second equalizer filter for, installed in parallel with the first equalizer filter, monitoring the input signal; a size comparison unit for sampling the size of a monitoring signal from the second equalizer filter at each period of an asynchronous clock signal; and a digital control unit for collecting the comparison data in the size comparison unit as changing an equalizer monitoring code to be provided to the second equalizer filter and a reference signal of the size comparison unit, and finding and providing an optimal equalizer control code to the first equalizer filter based on the collected comparison data; wherein the second equalizer filter compensates and outputs the input signal according to an equalizer monitoring code of the digital control unit to find an optimal equalizer control code in the digital control unit.

A method and apparatus for automatic gain control (AGC). A transceiver includes an analog-to-digital converter (ADC) configured to convert a received analog signal to digital signal, a measurement circuitry configured to measure a signal level on the digital signal, an AGC controller configured to generate, based on the measured signal level, an analog gain control signal to change a gain to be applied to a received analog signal in analog domain and a digital gain control signal for digital gain compensation corresponding to the gain change in analog domain, and a digital gain compensation circuitry configured to apply the digital gain compensation based on the digital gain control signal. The digital compensation gain applied to the digital bits follows a ramp profile that is an inverse of a transient response to the gain change in analog domain.

Apparatus, systems, articles of manufacture, and methods for volume adjustment are disclosed herein. An example method includes collecting data corresponding to a volume of an audio signal as the audio signal is output through a device, when an average volume of the audio signal does not satisfy a volume threshold for a specified timespan, determining a difference between the average volume and a desired volume, and applying a gain to the audio signal to adjust the volume of the audio signal to the desired volume, the gain determined based on the difference between the average volume and the desired volume.

Methods and apparatus to perform an automated gain control protocol with an amplifier based on historical data corresponding to contextual data are disclosed. Example apparatus disclosed herein include a first controller to, in response to a trigger to identify an automatic gain control (AGC) parameter for an AGC protocol, determine the AGC parameter for the AGC protocol based on historical data corresponding to contextual data. Disclosed example apparatus also include a processor to perform the AGC protocol based on the selected AGC parameter. Disclosed example apparatus further include a second controller to update the historical data based on the contextual data and a result of the AGC protocol.

Volume leveler controller and controlling method are disclosed. In one embodiment, A volume leveler controller includes an audio content classifier for identifying the content type of an audio signal in real time; and an adjusting unit for adjusting a volume leveler in a continuous manner based on the content type as identified. The adjusting unit may configured to positively correlate the dynamic gain of the volume leveler with informative content types of the audio signal, and negatively correlate the dynamic gain of the volume leveler with interfering content types of the audio signal.

The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency Q+r.sub.0 is generated on the basis of existing components at and +.sub.0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.

The present application provides a method for audio processing, an apparatus, an electronic device and a storage medium, the method includes: performing a segmentation processing on an audio to be processed to obtain N audio segments, N is an integer larger than or equal to 2; obtaining a target sequence according to maximum volume values of each of the audio segments, the target sequence includes: maximum volume values of M audio segments sorted in order of maximum volume values from large to small, M is a positive integer smaller than N; obtaining a volume adjustment parameter of the audio to be processed according to the target sequence, the volume adjustment parameter is used to adjust playing volume of the audio to be processed to target volume. The method avoids the problem that the user frequently adjusts the volume of the audio.

A sound reproducing apparatus and a sound reproducing method are provided. The apparatus includes multiple directional speakers located at different places, an information acquisition unit configured to acquire sensing information of a person to be provided with a reproduced sound, and a processor that determines a position of the person from the acquired sensing information and controls output levels of the multiple directional speakers. The method includes the steps of acquiring sensing information of a person to be provided with a reproduced sound, determining a position of the person from the acquired sensing information, and controlling output levels of the directional speakers. The output levels of the speakers are determined based on the distance from the person.

Exemplary multipath digital microphone described herein can comprise exemplary embodiments of adaptive ADC range multipath digital microphones, which allow low power to be achieved for amplifiers or gain stages, as well as for exemplary adaptive ADCs in exemplary multipath digital microphone arrangements described herein, while still providing a high DR digital microphone systems. Further non-limiting embodiments can comprise an exemplary glitch removal component configured to minimize audible artifacts associated with the change in the gain of the exemplary adaptive ADCs.

A system comprising audio processing circuitry is provided. The audio processing circuitry is operable to receive audio signals. The audio processing circuitry is operable to process the audio signals to detect strength of a chat component of the audio signals and strength of a game component of the audio signals. The audio processing circuitry is operable to automatically control a volume setting based on one or both of: the detected strength of the chat component, and the detected strength of the game component. The combined-game-and-chat audio signals may comprise a left channel signal and a right channel signal. The processing of the combined-game-and-chat audio signals may comprise measuring strength of a vocal-band signal component that is common to the left channel signal and the right channel signal.