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 system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of electronic device audible cues. Audible cues are stored as sound files. Ambient noise is detected, and the output of the audible cues is altered based on the ambient noise. Various embodiments include processed sound files that are more robust in noisy environments.

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.

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.

The present invention discloses a signal processing device, a signal processing method and a speaker. The signal processing device comprises a multi-band dynamic range controller, wherein the multi-band dynamic range controller receives an audio signal and includes a first band splitting unit and a resonant band adjustment unit; the first band splitting unit is configured to split the audio signal into multiple bands and obtain at least one resonant band therefrom, which has a resonant frequency band signal in a resonant frequency range of the audio signal; and the resonant band adjustment unit is configured to adjust the resonant frequency band signal based at least on a resonant band dynamic range control gain and output an adjusted resonant frequency band signal for combination with other band signals into a compression output signal.

A vacuum tube subwoofer extraction circuit system includes: a front-end circuit; and a vacuum tube subwoofer extraction circuit connected to the front-end circuit, the vacuum tube subwoofer extraction circuit including: a small signal amplification vacuum tube for receiving an input signal from the front-end circuit and outputting an audio signal; and a passive filtering circuit connected to the small signal amplification vacuum tube and adapted to perform a filtering process on the audio signal so as to output a filtered signal, wherein the front-end circuit includes a switch power circuit for providing low voltage to the vacuum tube subwoofer extraction circuit, wherein the small signal amplification vacuum tube does not have gain effect on the input signal.

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 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.

A method or apparatus for delivering audio programming such as music to listeners may include identifying, capturing and applying a listener's audiometric profile to transform audio content so that the listener hears the content similarly to how the content was originally heard by a creative producer of the content. An audio testing tool may be implemented as software application to identify and capture the listener's audiometric profile. A signal processor may operate an algorithm used for processing source audio content, obtaining an identity and an audiometric reference profile of the creative producer from metadata associated with the content. The signal processor may then provide audio output based on a difference between the listener's and creative producer's audiometric profiles.