Systems and methods for generating a haptic output from an audio signal having a continuous stream of sampled digital audio data are provided. A haptic processing system receives the digital audio data, analyses the digital audio data for processing and extracts haptic signals for generating a haptic effect through an actuator. The method includes passing the digital audio signal on through dynamic processor(s), adjusting the dynamic range of the digital audio signal, extracting the signal envelope of the audio data, synthesising low-frequency signals from the extracted signal envelope, and enhancing the low-frequency content using a resonator. The haptic output is generated by mixing the digital audio signal with outputs from the different modules of the haptic processing system. An analytics module monitors, controls and adjusts the processing of the digital audio signal at the noise gate module, the compressor module and the envelope module to enhance the haptic output.

Current microphone designs are still using concepts developed in the early 20th century. A transducer followed by a simple buffer with high impedance, low power, and low gain. Even when tubes were replaced by solid state devices the same practice continued, with the low gain triode being replaced by an even smaller power, lower current FET buffer. The invention offers a microphone system intended for professional audio applications that dispenses with these, low power concepts and incorporates elements of a microphone, microphone pre amplifier, and high power mixing console/pre amp line drivers into a single housing. By combining transducer, high fidelity pre amp circuit, and high power line driving stage with a high voltage, high current external power supply, the capabilities of microphones can be greatly increased in many ways. Our approach can be implemented in a variety of circuit topologies including discrete transistors, tubes and integrated circuits, our concept allows microphones to connect directly to analog or digital recording devices without the need for external mixers or pre amplifiers. The design not only allows for unprecedented simplicity in the work flow of the modern recording studio, which combines the use of stand-alone analog records, standalone digital recorder, and computer workstation based recording options, but also eliminates redundant circuits that are present in microphones, microphone pre amps, and mixing consoles, while offering the customer increased value.

Current microphone designs are still using concepts developed in the early 20th century. A transducer followed by a simple buffer with high impedance, low power, and low gain. Even when tubes were replaced by solid state devices the same practice continued, with the low gain triode being replaced by an even smaller power, lower current FET buffer. The invention offers a microphone system intended for professional audio applications that dispenses with these, low power concepts and incorporates elements of a microphone, microphone pre amplifier, and high power mixing console/pre amp line drivers into a single housing. By combining transducer, high fidelity pre amp circuit, and high power line driving stage with a high voltage, high current external power supply, the capabilities of microphones can be greatly increased in many ways. Our approach can be implemented in a variety of circuit topologies including discrete transistors, tubes and integrated circuits, our concept allows microphones to connect directly to analog or digital recording devices without the need for external mixers or pre amplifiers. The design not only allows for unprecedented simplicity in the work flow of the modern recording studio, which combines the use of stand-alone analog records, standalone digital recorder, and computer workstation based recording options, but also eliminates redundant circuits that are present in microphones, microphone pre amps, and mixing consoles, while offering the customer increased value.

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.

An active sound management system comprises a transducer that senses an actual sound or vibration from a sound generating source and generates a transducer signal in response to sensing the actual sound or vibration; a harmonic extractor device that extracts a plurality of harmonics from the transducer signal; and a harmonic modifier device that adjusts a feature of the extracted harmonic to be within a predetermined threshold with respect to a target harmonic corresponding to a desired sound.

An active sound management system comprises a transducer that senses an actual sound or vibration from a sound generating source and generates a transducer signal in response to sensing the actual sound or vibration; a harmonic extractor device that extracts a plurality of harmonics from the transducer signal; and a harmonic modifier device that adjusts a feature of the extracted harmonic to be within a predetermined threshold with respect to a target harmonic corresponding to a desired sound.

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.

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.

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.

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.