The invention provides a concept for combined dynamic range compression and guided clipping prevention for audio devices. An audio decoder for decoding an audio bitstream and a metadata bitstream related to the audio bitstream according to the concept includes an audio processing chain including a plurality of adjustment stages including a dynamic range control stage for adjusting a dynamic range of the audio output signal and a guided clipping prevention stage for preventing clipping of the audio output signal; and a metadata decoder configured to receive the metadata bitstream and to extract dynamic range control gain sequences and guided clipping prevention gain sequences from the metadata bitstream, at least a part of the dynamic range control gain sequences being supplied to the dynamic range control stage, and at least a part of the guided clipping prevention gain sequences being supplied to the guided clipping prevention stage.

The present disclosure relates to processing a plurality of audio signals. A device receives the plurality of audio signals in the frequency domain and determining an overall attenuation multiplier based on the plurality of audio signals and an overall lookup table that relates decibel values to different overall attenuation multipliers. The device determines an attenuation vector comprising a plurality of bin-specific attenuation multipliers, each bin-specific attenuation multiplier respectively corresponding to a different frequency bin of the plurality of frequency bins. The device scales each bin-specific attenuation value in the attenuation vector with the overall attenuation multiplier, and edits each of the audio signals based on the scaled bin-specific attenuation values in the attenuation vector.

A method for reducing gain for audio amplification by an audio system having a tweeter channel, an optional midrange channel, and a woofer channel. Gain of the woofer channel is reduced simultaneously with reducing gain of the tweeter channel, both responsive to detecting the same instance of overloading (overdriving) the woofer channel. Other aspects are also described and claimed.

Methods, systems, and apparatuses are described herein for improved processing audio in a video stream. A system may split audio in a frame of video content into multiple bands based on their audio levels. The system may then dynamically compress and dynamically normalize the audio level in each band. When dynamically compressing the bands, the system may determine, based on stored information, what audio level range is acceptable for an end user and may smooth and maintain the ranges of the audio to be within the acceptable range. The system may include the dynamically normalized and dynamically compressed frames as a second audio track in the video content. A computing device receiving the video content may select the second audio track during playback. If an end user selects the second audio track, the video is delivered with the modified sound of the second audio track.

The present document describes a dynamic range control unit (210) configured to apply dynamic range control, referred to as DRC, to an audio signal (211). The DRC unit (210) is configured to downsample a subband signal (212) derived from the audio signal (211), to provide a downsampled subband signal (321), to determine a DRC gain (329) based on the downsampled subband signal (321), and to apply the DRC gain (329) to the subband signal (212), to provide a compressed subband signal (213) of a compressed audio signal (214).

Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.

The invention provides a loudspeaker nonlinear compensation method. The method includes steps: obtaining system parameter of the loudspeaker, the No. i time-domain excitation voltage signal and the No. i state vector of the loudspeaker; compensating the No. i time-domain excitation voltage signal according to the system parameter and the No. i state vector and obtain i compensation voltage signal; obtaining the No. i+1 state vector according to the calculation of the system parameter and the No. i compensation voltage signal; outputting the No. i compensation voltage signal and record the quantity of the compensation voltage signal; judging whether the quantity of the compensation voltage signal is equal to the preset number value.

A method for speech equalization, comprising the steps of receiving an input audio signal, processing said input audio signal in dependence on frequency and to providing an equalized electric audio signal according to an equalization function, wherein said equalization function comprises at least an actuator part configured to dynamically applying a compensation filter to the received input signal and dynamically applying a transparent filter to the received input signal, and further transmitting an output signal perceivable by a user as sound representative of said electric acoustic input signal or a processed version thereof.

An audio signal processing apparatus that includes a band division section, an analysis section, a gain adjustment amount calculation section, and a gain adjustment section. The band division section is configured to generate a resonance in-band signal by band division on an audio input signal. The analysis section is configured to extract an amount of features from each of the resonance in-band signal and the input signal. The gain adjustment amount calculation section is configured to calculate a gain adjustment amount for a resonance frequency band in the input signal, the gain adjustment amount being calculated based on the amount of features of the resonance in-band signal and the amount of features of the input signal. The gain adjustment section is configured to perform a gain adjustment on the resonance frequency band in the input signal based on the gain adjustment amount.

The invention relates to Higher-Order Ambisonics (HOA) Content Loudness Level Adjustment. A method for adjusting a loudness level of a HOA audio signal on a transmitting side comprises steps of measuring loudness of only the W-channel of the HOA audio signal, and leveling HOA master gain for all channels of the HOA signal according to the measured loudness of the W-channel.