A sound controlling system including a user terminal having a sound source, a wireless communication device, a digital to analog converter (DAC) and first processing electronics. The first processing electronics are configured to: provide data of a backing sound to the sound source; control the sound source to generate a sound signal based on the data; receive a first input instruction including a first instruction to transmit the sound signal and a second instruction to play back the backing sound; provide the sound signal to the wireless communication device as the first input instruction being the first instruction, and provide the sound signal to the DAC as being the second instruction; control the wireless communication device to convert the sound signal to a wireless signal and transmit the wireless signal; and convert the sound signal from a digital signal to an analog signal for play back of the backing sound.

In various applications, the system provides a method for processing audio signals, including: receiving a request for audio content; receiving an identifier encoded in a personal audio device comprising a transducer for playing audio; retrieving at least one parameter associated with the identifier; and processing the audio content using at least the request, the identifier and the at least one parameter, wherein the processing is customized for the personal audio device based on the at least one parameter associated with the identifier. In various applications the parameter is, one or more of, associated with a specification of the personal audio device, acoustic metrics of the transducer, relates to control of equalization, relates to permission to enable proprietary sonic processing for enhanced acoustic reception of streaming content, relates to acoustic metrics of the transducer and wherein the identifier is associated with permission to enable proprietary sonic processing for enhanced acoustic reception of streaming content, and/or is stored in a chip on the personal audio device, is retrieved from a server in a network, among other things. In various applications, the personal audio device comprises ear buds and the identifier is stored in a non-volatile memory of the ear buds.

The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output. In these and other embodiments, a feature of responses of the mode filters have been optimized to minimize an aspect of a residual error after a point in time.

The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output. In these and other embodiments, a feature of responses of the mode filters have been optimized to minimize an aspect of a residual error after a point in time.

An audio signal output system including a first audio signal path, a second audio signal path and control electronics that provide a mixed sound from sound sources to the first and second audio signal paths, and control the mixed sound in accordance with a position at which a sound image associated with at least one of the sound sources is localized relative to an orientation of a user's head. The control electronics provide one or more signals corresponding to the mixed sound to the first and second audio signal paths based on one or more transfer functions associated with one or more values corresponding to at least one of a distance from the user to the at least one sound source, an angle of the user with respect to the at least one sound source, and a size of a space in which the user is located.

In various applications, the system provides a method for processing audio signals, including: receiving, by a processor, a digital audio signal from a recorded audio file; analyzing, by the processor, the digital audio signal to identify pitch distortion caused by changes in momentary sound level; determining, by the processor, an amount of compensation of the audio signal to correct the identified pitch distortion; dynamically adjusting, by the processor, the digital audio signal by the compensation amount to correct the identified pitch distortion; and outputting, by the processor, the digital audio signal to an audio transducer device of a listener to improve a listening experience for the listener of the recorded audio file.

A system including: an electronic memory device and a processor. The processor is configured to: control a communication device to receive first input information indicating a first instruction, the first instruction corresponding to control of sound associated with at least one first sound source; control a transmitter of the communication device to transmit first information to an audio output device, the first information corresponding to the first input information indicating the first instruction; control the communication device to receive second input information indicating a second instruction corresponding to control of sound associated with the at least one second sound source; and control the transmitter to transmit second information to the audio output device, the second information including or corresponding to an audio signal associated with the at least one second sound source and processed according to the second input information indicating the second instruction.

In general, the present invention relates to a method and system for synthesizing artificial reverberation using modal analysis of a room or resonating object. In one embodiment of the inventive system, a collection of resonant filters is employed, each driven by the source signal, and their outputs summed. With filter resonance frequencies and dampings tuned to the modal frequencies and decay times of the acoustic space or resonating object being simulated, and filter gains set according to the source and listener positions within the space or object, any number of acoustic spaces and resonant objects may be simulated.

A system including: an electronic memory device and a processor. The processor is configured to: control a communication device to receive first input information indicating a first instruction, the first instruction corresponding to control of sound associated with at least one first sound source; control a transmitter of the communication device to transmit first information to an audio output device, the first information corresponding to the first input information indicating the first instruction; control the communication device to receive second input information indicating a second instruction corresponding to control of sound associated with the at least one second sound source; and control the transmitter to transmit second information to the audio output device, the second information including or corresponding to an audio signal associated with the at least one second sound source and processed according to the second input information indicating the second instruction.

An audio signal output system including a first audio signal path, a second audio signal path and control electronics that provide a mixed sound from sound sources to the first and second audio signal paths, and control the mixed sound in accordance with a position at which a sound image associated with at least one of the sound sources is localized relative to an orientation of a user's head. The control electronics provide one or more signals corresponding to the mixed sound to the first and second audio signal paths based on one or more transfer functions associated with one or more values corresponding to at least one of a distance from the user to the at least one sound source, an angle of the user with respect to the at least one sound source, and a size of a space in which the user is located.