Systems and methods for processing an audio signal are provided for replay on an audio device. An audio signal is spectrally decomposed into a plurality of subband signals using band pass filters. Each of the subband signals are provided to a respective modulator and subsequently, from the modulator output, provided to a respective first processing path that includes a first dynamic range compressor, DRC. Each subband signal is feedforward compressed by the respective first DRC to obtain a feedforward-compressed subband signal, wherein the first DRC is slowed relative to an instantaneous DRC. Subsequently, each feedforward-compressed subband signal is provided to a second processing path that includes a second DRC, wherein the feedforward-compressed subband signal is compressed by the respective second DRC and outputted to the respective modulator. Modulation of the subband signals is then performed in dependence on the output of the second processing path. Finally, the feedforward-compressed subband signals are recombined.

An apparatus and method of excursion protection of a loudspeaker. The method includes attenuating selected bands in a transform domain, controlled by a feedback signal resulting from an excursion transfer function that has been modified according to the real-time operational characteristics of the loudspeaker. In this manner, the system reduces the amount of wideband attenuation needed to address the predicted excursion, resulting in a better listening experience.

An electronic audio device has a microchip has embedded therein, circuitry for enhancing the sound outputted therefrom and which is associated, such as a radio, television, headphones, earbuds and the like.

The invention relates to methods and apparatus for adjusting a level of an audio signal. An audio signal is divided into a plurality of frequency bands. Modification parameters are obtained for at least one of the plurality of frequency band. Gain factors are derived for at least one of the plurality of frequency bands, the gain factors determined based on the amplitude scale factors. The gain factors are smoothed. A level of noise from noise compensation factors is determined. The gain factors are applied to at least one of the frequency bands to generate gain adjusted frequency bands. The level of noise is adjusted based on the gain adjusted frequency bands. At least one of the frequency bands is filtered with a filter generated with the filter coefficients. The plurality of frequency bands is synthesized to generate an output audio signal.

Reduced-size guided-surface acoustic wave (SAW) resonators are disclosed. Guided-SAW resonators can achieve high acoustic coupling and acoustic quality Q, but may have a larger surface area compared with a traditional temperature compensated (TC)-SAW resonator. In an exemplary aspect, a guided-SAW device is fabricated with a metal-insulator-metal (MIM) capacitor to produce a guided-SAW which has the same high Q with a surface area which is the same or less than traditional TC-SAW resonators.

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.

A method for adapting a gain of at least one channel audio input signal in order to generate an N channel audio output signal, wherein at least one channel audio input signal includes a speech input channel, in which speech signal components, if present in the at least one channel audio input signal, are present, and comprising other audio input channels. A perceived loudness of the at least one channel audio input signal is dynamically determined and it is determined whether speech signal components are present in the speech input channel. If this is the case the gain of the speech input channel is adapted differently compared to the gain of the other audio input channels.

Systems and methods for limiting volume in an audio playback device using a feedback controller are disclosed herein. In one example, a gain stage modulates gain of an audio signal based in part on feedback from a downstream limiter. The gain stage receives a first audio signal as well as a feedback signal from the feedback controller. Based at least in part on the feedback signal from the feedback controller, the gain stage modulates a gain of the first audio signal to provide a second audio signal. The second audio signal is delivered to the limiter, which limits the second audio signal to produce an output signal. The output signal is played back via a transducer. The feedback controller receives a gain reduction value from the limiter and determines a feedback signal to provide to the gain stage upstream of the limiter.

The invention discloses an algorithm-based audio optimization method, an intelligent terminal and a storage device. The method includes steps of converting an original audio file on time domain into an audio file on frequency domain through Fourier transform; before extracting and matching frequency range and amplitude information of an audio signal with those of different types of existing audio test standard sound sources to determine type of the audio signal; matching corresponding frequency mapping function by function library for transform to obtain a processed audio file, obtaining an optimized audio file by inverse Fourier transform. The present invention searches a similar type of sound source by comparing and matching original audio with standard sound sources, determines the frequency mapping function for transform, maps the frequency, compresses or expands relevant frequency, and automatically performs tuning to achieve effect of improving sound quality.

The present disclosure relates to processing a plurality of audio signals. The method includes receiving 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 method further includes determining 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 method further includes scaling each bin-specific attenuation value in the attenuation vector with the overall attenuation multiplier, and editing each of the audio signals based on the scaled bin-specific attenuation values in the attenuation vector.