A method of generating audio loudness performed by an audio loudness generation device may include: receiving loudness information on each of a plurality of audio tracks included in one group; predicting an intermediate loudness distribution, which is a loudness distribution for the one group, on the basis of the loudness information on each of the plurality of audio tracks; and generating an integrated loudness for the one group on the basis of the intermediate loudness distribution.

A method (600) for decoding an encoded audio signal (102) is described. The encoded audio signal (102) comprises a sequence of frames. Furthermore, the encoded audio signal (102) is indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. Different subsets of DRC profiles from the plurality of DRC profiles are comprised within different frames of the sequence of frames, such that two or more frames of the sequence of frames jointly comprise the plurality of DRC profiles. The method (600) comprises determining a first rendering mode from the plurality of different rendering modes; determining (609, 610) one or more DRC profiles from a subset of DRC profiles comprised within a current frame of the sequence of frames; determining (611) whether at least one of the one or more DRC profiles is applicable to the first rendering mode; selecting (604) a default DRC profile as a current DRC profile, if none of the one or more DRC profiles is applicable to the first rendering mode; wherein definition data of the default DRC profile is known at a decoder (100) for decoding the encoded audio signal (102); and decoding the current frame using the current DRC profile.

Methods, systems, and computer program products of automatic de-essing are disclosed. An automatic de-esser can be used without manually setting parameters and can perform reliable sibilance detection and reduction regardless of absolute signal level, singer gender and other extraneous factors. An audio processing device divides input audio signals into buffers each containing a number of samples, the buffers overlapping one another. The audio processing device transforms each buffer from the time domain into the frequency domain and implements de-essing as a multi-band compressor that only acts on a designated sibilance band. The audio processing device determines an amount of attenuation in the sibilance band based on comparison of energy level in sibilance band of a buffer to broadband energy level in a previous buffer. The amount of attenuation is also determined based on a zero-crossing rate, as well as a slope and onset of a compression curve.

Disclosed is a method and apparatus for determining one or more operation parameters for a dynamic range compression (DRC) system. The method comprises obtaining, as an input, a parameter indicative of a hearing ability of a user, the parameter relating to a first difference in sound intensity between a maskee at a first frequency and a masker at a second frequency, determining a target value for the parameter, and determining the one or more operation parameters such that a second difference in sound intensity after sound intensity modification by the DRC (between sound intensity of the maskee of the masker) corresponds to the target value for the parameter. The operation parameters are determined such that a dependence of the second difference in sound intensity on the sound intensity of the maskee is minimized for a given range of sound intensities of the maskee.

Volume limiting systems and methods are operable to limit volume output from media presentation devices. An exemplary embodiment detects a sound using a microphone, wherein the sound corresponds to an audio output of at least one controlled media presentation device, and wherein the microphone is remotely located from the at least one controlled media presentation device; compares a level of the detected sound with a predefined maximum volume limit; generates a volume output limit command in response to the detected sound exceeding the predefined maximum volume limit; and communicates the volume output limit command to the media presentation device. The media presentation device then reduces a volume level of its audio output. In some instances, volume may be limited during user specified periods.

The present technology relates to a decoding apparatus, a decoding method and a program which make it possible to obtain sound with higher quality.

A demultiplexing circuit demultiplexes an input code string into a gain code string and a signal code string. A signal decoding circuit decodes the signal code string to output a time series signal. A gain decoding circuit decodes the gain code string. That is, the gain decoding circuit reads out gain values and gain inclination values at predetermined gain sample positions of the time series signal and interpolation mode information. An interpolation processing unit obtains a gain value at each sample position between two gain sample positions through linear interpolation or non-linear interpolation according to the interpolation mode based on the gain values and the gain inclination values. A gain applying circuit adjusts a gain of the time series signal based on the gain values. The present technology can be applied to a decoding apparatus.

Volume limiting systems and methods are operable to limit volume output from media presentation devices. An exemplary embodiment detects a sound using a microphone, wherein the sound corresponds to an audio output of at least one controlled media presentation device, and wherein the microphone is remotely located from the at least one controlled media presentation device; compares a level of the detected sound with a predefined maximum volume limit; generates a volume output limit command in response to the detected sound exceeding the predefined maximum volume limit; and communicates the volume output limit command to the media presentation device. The media presentation device then reduces a volume level of its audio output. In some instances, volume may be limited during user specified periods.

An audio normalization gain value is applied to an audio signal to produce a normalized signal. The normalized signal is processed to compute dynamic range control (DRC) gain values in accordance with a selected one of several pre-defined DRC characteristics. The audio signal is encoded, and the DRC gain values are provided as metadata associated with the encoded audio signal. Several other embodiments are also described and claimed.

This application relates to Class D amplifier circuits. A modulator controls a Class D output stage based on a modulator input signal (Dm) to generate an output signal (Vout) which is representative of an input signal (Din). An error block, which may comprise an ADC, generates an error signal (ε) from the output signal and the input signal. In various embodiments the extent to which the error signal (ε) contributes to the modulator input signal (Dm) is variable based on an indication of the amplitude of the input signal (Din). The error signal may be received at a first input of a signal selector block. The input signal may be received at a second input of the signal selector block. The signal selector block may be operable in first and second modes of operation, wherein in the first mode the modulator input signal is based at least in part on the error signal; and in the second mode the modulator input signal is based on the digital input signal and is independent of the error signal. The error signal can be used to reduce distortion at high signal levels but is not used at low signal levels and so the noise floor at low signal levels does not depend on the component of the error block.

A power amplifier includes a digital-to-analog converter, a loop filter, a driver circuit, a first adjustable reference resistor and a second adjustable reference resistor. A circuit includes an overcurrent protection circuit and a power amplifier, wherein the overcurrent protection circuit is communicatively coupled to the power amplifier. The digital-to-analog converter is configured to receive a digital signal and to output an analog signal, the driver circuit communicatively coupled to the loop filter and at least one of a first output port and a second output port of the power amplifier.