The present invention provides methods and systems for narrow bandwidth digital processing of an input audio signal. Particularly, the present invention includes a high pass filter configured to filter the input audio signal. A first compressor then modulates the filtered signal in order to create a partially processed signal. In some embodiments, a clipping module further limits the gain of the partially processed signal. A splitter is configured to split the partially processed signal into a first signal and a second signal. A low pass filter is configured to filter the first signal. A pass through module is configured to adjust the gain of the second signal. A mixer then combines the filtered first signal and the gain-adjusted second signal in order to output a combined signal. In some embodiments, a tone control module further processes the combined signal, and a second compressor further modulates the processed signal.

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.

A method and a device are provided for modifying audio signals in accordance with hearing capabilities of an individual who is listening to audio signals played by a music player. The method comprises the steps of: providing a music player operative to play audio signals, wherein the music player comprises a processor configured to modify audio signals that are about to be played, by taking into account the hearing capabilities of the individual; providing information that relates to the hearing capabilities of the individual; forwarding the information that relates to the hearing capabilities of the individual, from an electronic device to the music player; and using the music player processor to modify audio signals when the individual is listening to audio signals being played by the music player, wherein the audio signals are modified before they are played by taking into account the individual's hearing capabilities.

The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.