Disclosed is a method for increasing a perceived loudness of an audio data signal comprising the steps of obtaining a first digital audio data signal; determining at least one temporal amplitude peak in the first digital audio data signal; generating a second digital audio data signal by reducing the at least one temporal amplitude peak in the first digital audio data signal based on a predicted perceptual difference model representing a predicted perceptual difference between the first digital audio data signal and a peak reduced version of the first digital audio data signal; and generating a third digital audio data signal by amplifying the second digital audio data signal so that a peak of the second digital audio data signal has a predetermined signal value, wherein a perceived loudness of the third digital audio data signal is larger than a perceived loudness of the first digital audio data signal.

A method for loudness equalization is provided that includes receiving input loudness data at an audio processing system. Converting gain data of the input loudness data to a linear scale at the audio processing system. Determining a reciprocal of a gain-linear loudness value as a function of the converted gain data using the audio processing system. Determining a compression ratio using the audio processing system. Performing temporal smoothing and look ahead processing using the audio processing system. Outputting gain data as a function of the temporal smoothing and look ahead processing using the audio processing system.

Some embodiments of the invention are directed to enabling a user to easily identify the frequency range(s) at which sound masking occurs, and addressing the masking, if desired. In this respect, the extent to which a first stem is masked by one or more second stems in a frequency range may depend not only on the absolute value of the energy of the second stem(s) in the frequency range, but also on the relative energy of the first stem with respect to the second stem(s) in the frequency range. Accordingly, some embodiments are directed to modeling sound masking as a function of the energy of the stem being masked and of the relative energy of the masked stem with respect to the masking stem(s) in the frequency range, such as by modeling sound masking as loudness loss, a value indicative of the reduction in loudness of a stem of interest caused by the presence of one or more other stems in a frequency range.

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Methods for performing dynamic range compression (DRC) on audio in a manner intended to produce output audio for playback by systems or devices with limited power handling capabilities and preferably also to reduce or prevent undesirable artifacts (e.g., pumping and/or breathing) in the output audio. Some embodiments perform the DRC so as to maximize average loudness (while preventing loss of quieter elements) during playback, and also to reduce or prevent distortion. Other aspects are systems or devices configured to perform embodiments of the method. In some embodiments, reduced DRC is applied when average loudness of the input audio approaches (or matches or exceeds) a target (e.g., a knee point for DRC, or a signal level near to a maximum playback level of the intended playback system), since such input audio is assumed to have already been compressed, and otherwise applying full DRC to the input audio.

The present disclosure provides a limiter system. The limiter system includes a first equalization filter, a low-pass filter, a first limiter, a high-pass filter, a second limiter, and a mixer. The limiter system provided by the present disclosure further includes a second equalization filter. An audio signal from a signal source first passes through the first equalization filter, the signal equalized for the first time is divided into two signals, one signal is processed by the low-pass filter and the first limiter, the other signal is processed by the high-pass filter and the second limiter, and then the two processed signals enter the mixer to be mixed and outputted. The mixed output signal is subjected to second equalization filtering by the second equalization filter to avoid clipping distortion or to obtain a higher maximum sound level.

A limiter system for an active speaker may include at least one lowpass filter configured to receive an input signal and output a signal lower than a crossover frequency, at least one highpass filter, configured to receive an input signal and output a signal higher than the crossover frequency, a first allpass filter configured to adjust the phase of the signal lower than the crossover frequency, a second allpass filter configured to adjust the phase of the signal higher than the crossover frequency, a first limiter, configured to receive and limit the signal from the first allpass filter, a second limiter, configured to receive and limit the signal from the second allpass filter, and a mixer, configured to mix the signal lower from the first limiter and the signal from the second limiter.

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.

An apparatus for controlling a dynamic compressor of a hearing aid includes a combination signal analyzer for determining the binaural similarity of a left and right audio signal and an amplification adjuster for providing an amplification value for a band of the left or right audio signal in dependence on the binaural similarity and a level of the left or right audio signal in the band.

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.