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.

Example embodiments provide a process that includes one or more of receiving an audio signal at a feedback compressor circuit, multiplying the received audio signal with a power feedback signal to create a product audio signal, wherein the feedback signal comprises a low-pass filtered signal, applying a power amplifier to the product audio signal, and providing the amplified product audio signal as an output signal to a speaker.

A system for encoding and applying Dynamic Range Control/Compression (DRC) gain values to a piece of sound program content is described. In particular, a set of DRC gain values representing a DRC gain curve for the piece of content may be divided into frames corresponding to frames of the piece of content. A set of fields may be included with an audio signal representing the piece of content. The additional fields may represent the DRC gain values using linear or spline interpolation. The additional fields may include 1) an initial gain value for each DRC frame, 2) a set of slope values at particular points in the DRC curve, 3) a set of time delta values for each consecutive pair of slope values, and/or 4) one or more gain delta values representing changes of DRC gain values in the DRC gain curve between points of the slope values.

In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

A method for dynamically controlling enhancement of an audio stream is provided, where the audio stream defines a sequence of audio segments over time. Each audio segment defines a waveform having a plurality of waveform attributes. For each audio segment of the sequence of audio segments, the method includes: (i) determining a set of waveform-attribute values of the audio segment's waveform attributes, (ii) computing a first distance between the determined set of waveform-attribute values and a first predefined set of waveform-attribute values representative of speech, and computing a second distance between the determined set of waveform-attribute values and a second predefined set of waveform-attribute values representative of music, (iii) using the computed first and second distances as a basis to classify the audio segment as primarily speech or rather primarily music, and (iv) controlling, based on the classifying, whether or not to enhance the audio segment for output.

Methods, apparatus, systems and articles of manufacture are disclosed for dynamic volume adjustment via audio classification. Example apparatus include at least one memory; instructions; and at least one processor to execute the instructions to: analyze, with a neural network, a parameter of an audio signal associated with a first volume level to determine a classification group associated with the audio signal; determine an input volume of the audio signal; determine a classification gain value based on the classification group; determine an intermediate gain value as an intermediate between the input volume and the classification gain value by applying a first weight to the input volume and a second weight to the classification gain value; apply the intermediate gain value to the audio signal, the intermediate gain value to modify the first volume level to a second volume level; and apply a compression value to the audio signal, the compression value to modify the second volume level to a third volume level that satisfies a target volume threshold.

A headset having game, chat and microphone audio signals is provided with a programmable signal processor for individually modifying the audio signals and a memory configured to store a plurality of user-selectable signal-processing parameter settings that determine the manner in which the audio signals will be altered by the signal processor. The parameter settings collectively form a preset, and one or more user-operable controls can select and activate a preset from the plurality of presets stored in memory. The parameters stored in the selected preset can be loaded into the signal processor such that the sound characteristics of the audio paths are modified in accordance with the parameter settings in the selected preset.

In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Example embodiments provide a process that includes one or more of receiving an audio signal from a feedback path of a feedback compressor circuit, determining whether an auxiliary attenuation value applied to the feedback compressor circuit has changed since a last audio signal was received, responsive to determining the auxiliary value has changed, determining a current operational state value of the LPF needs to be modified based on the changed auxiliary attenuation value, modifying the operational state value of the LPF, and applying the audio signal to the modified LPF.

Provided is a method for audio peak reduction using an all-pass filter, including: determining a delay parameter m and a gain parameter g based on a formula (1):

[00001]$\begin{array}{cc}\underset{m,g}{\arg \min }\underset{n}{\max }.Math.y_{m,g}\left(n\right).Math.,& \left(1\right)\end{array}$

absolute peak map

[00002]$Y\left(m,g\right)=\underset{n}{\max }.Math.y_{m,g}\left(n\right).Math.,$

y.sub.m,g(n) represents a processed signal with a time-domain response function and is calculated based on a formula (2): y.sub.m,g(n)=(h.sub.s*x)(n)(2), where h.sub.s represents an impulse response function, x (n) represents an input signal, and h.sub.s is calculated based on formula (3):

[00003]$\begin{array}{cc}{H}_S\left(z\right)=\frac{g+z^{-m}}{1+{\mathrm{gz}}^{-m}}.& \left(3\right)\end{array}$

This method is widely used in the reproduction, storage and broadcasting of sound, and the computational complexity is small, which i