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.

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.

In some examples, adaptive speech intelligibility control for speech privacy may include determining, based on background noise at a near-end of a speaker, a noise estimate associated with speech emitted from the speaker, and comparing, by using a specified factor, the noise estimate to a speech level estimate for the speech emitted from the speaker. Adaptive speech intelligibility control for speech privacy may further include determining, based on the comparison, a gain value to be applied to the speaker to produce the speech at a specified level to maintain on-axis intelligibility with respect to the speaker, and applying the gain value to the speaker.

An audio signal processing device comprises: a receiver for receiving an input audio signal; a processor for generating loudness metadata corresponding to the input audio signal; and an outputter for transmitting the loudness metadata generated by the processor. The processor is configured to acquire loudness information analyzed from input content, acquires loudness information about the input audio signal by measuring the loudness of the input audio signal, generates the loudness metadata by converting the loudness information, and transmits, through the outputter, the generated loudness metadata to an output device for outputting the input audio signal.

Example systems, apparatus, and methods receive audio information including a plurality of frames from a source device, wherein each frame of the plurality of frames includes one or more audio samples and a time stamp indicating when to play the one or more audio samples of the respective frame. In an example, the time stamp is updated for each of the plurality of frames using a time differential value determined between clock information received from the source device and clock information associated with the device. The updated time stamp is stored for each of the plurality of frames, and the audio information is output based on the plurality of frames and associated updated time stamps. A number of samples per frame to be output is adjusted based on a comparison between the updated time stamp for the frame and a predicted time value for play back of the frame.

Disclosed is a method of operating an audio signal processing apparatus playing content including an audio signal. The method includes receiving the audio signal, receiving metadata including information related to a loudness of the audio signal, the metadata including loudness distribution information indicating, for each of a plurality of steps separated according to a loudness magnitude, a ratio between an amount of the audio signal corresponding to each of the plurality of steps of the audio signal and a total amount of the audio signal, and adjusting the loudness of the audio signal based on the metadata.

In an audio system with a plurality of input signals, the input signals are prioritized from least important to most important. The level of the aggregate of all of the input signals is compared to a threshold level. If the level is greater than the threshold, the levels of one or more of the input signals are reduced one at a time, in order from the least important input signal to the most important input signal.

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 relates to an apparatus (100) having an input (11, 12, 15, 16) and an output (13, 14, 17, 18) and having an effect device with volume-regulated audio signals from an audio file or audio source, wherein the audio signals have input levels at an input of the effect device and output levels at an output of the effect device. In order to achieve background sound which is as optimal as possible, the invention proposes that each input level is assigned to a preset output level irrespective of the content of the audio file or audio source, wherein the input level is regulated to the output level.

An automatic adjustment method includes: recognizing whether one of a plurality of predetermined sound categories is included in audio data of current game; when one of the predetermined sound categories is included in the audio data, determining volume gain based on the predetermined sound category included in the audio data; and adjusting the volume of the audio data according to the volume gain.