A speaker has a magnetic angle sensor that detects the displacement of the vibration system of the speaker. A non-linearity compensation filter compensates for output distortion due to a non-linear parameter of the speaker. A transfer function of a linear inverse filter is applied so that output distortion is eliminated from the displacement, detected by the magnetic angle sensor, of the vibration system. A controller drives the speaker by using a test signal, and measures a response from the detected displacement of the vibration system. If error between the measured response and the theoretical value, theoretically determined from the design specifications, of the response is small, the controller uses the theoretical value, theoretically determined from the design specifications, of the non-linear parameter as an active non-linear parameter and sets, in the non-linearity compensation filter, a transfer function that compensates for distortion in an output due to the active non-linear parameter.

The present technique relates to a signal processing device, method, and a program capable of reducing Doppler distortion. The signal processing device includes: a displacement prediction unit that predicts displacement of a diaphragm of a speaker, in a case where the speaker plays back sound based on an audio signal in which a high-frequency signal and a low-frequency signal are mixed, based on the audio signal; and a correction unit that performs time direction correction on the audio signal by performing interpolation processing using at least three samples of the audio signal, based on the displacement obtained from the predicting and a correction time obtained based on an acoustic velocity. The present technique can be applied in audio playback systems.

The present technique relates to a signal processing device, method, and a program capable of reducing Doppler distortion. The signal processing device includes: a displacement prediction unit that predicts displacement of a diaphragm of a speaker, in a case where the speaker plays back sound based on an audio signal in which a high-frequency signal and a low-frequency signal are mixed, based on the audio signal; and a correction unit that performs time direction correction on the audio signal by performing interpolation processing using at least three samples of the audio signal, based on the displacement obtained from the predicting and a correction time obtained based on an acoustic velocity. The present technique can be applied in audio playback systems.

The invention relates to audio transducer technology, including audio tuning systems to be utilised in personal audio devices, such as headphone, earphones, mobile phones and the like. The audio tuning system optimises the frequency response of the personal audio device by using Diffuse Field curve characteristics. The audio transducers of the personal audio device incorporate low resonance designs, including low resonance transducer and diaphragm suspensions to further optimise the sound quality of the device. The invention also relates to an audio transducer diaphragm construction that includes a three-dimensional lattice which may be utilised in any audio transducer application.

The invention relates to audio transducer technology, including audio tuning systems to be utilised in personal audio devices, such as headphone, earphones, mobile phones and the like. The audio tuning system optimises the frequency response of the personal audio device by using Diffuse Field curve characteristics. The audio transducers of the personal audio device incorporate low resonance designs, including low resonance transducer and diaphragm suspensions to further optimise the sound quality of the device. The invention also relates to an audio transducer diaphragm construction that includes a three-dimensional lattice which may be utilised in any audio transducer application.

A method and apparatus for audio signal processing in an audio chain to correct a non-linearity of the electroacoustic transducers in the audio chain by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain using an approximation of the quadratic function. The method accommodates the psychoacoustical characteristics of the human ear by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain approximating by a non-linear fifth degree polynomial function for a pressure change by the human ear up to p_Δ. The method and apparatus reduce non-linearities of the entire audio chain with the human ear, by adding non-linearities in the audio chain so that an audio chain characteristic reduces the non-linearity of the human ear polynomial approximation to the pressure change p_Δ=±1 Pa.

A method and apparatus for audio signal processing in an audio chain to correct a non-linearity of the electroacoustic transducers in the audio chain by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain using an approximation of the quadratic function. The method accommodates the psychoacoustical characteristics of the human ear by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain approximating by a non-linear fifth degree polynomial function for a pressure change by the human ear up to p_Δ. The method and apparatus reduce non-linearities of the entire audio chain with the human ear, by adding non-linearities in the audio chain so that an audio chain characteristic reduces the non-linearity of the human ear polynomial approximation to the pressure change p_Δ=±1 Pa.

Two coils are wrapped in one of numerous different implementations. In one implementation, the two coils are wrapped about a portion of a bobbin that has at least three flanges. The first coil is disposed about a first portion of the bobbin between the first flange and the second flange, and a second coil is disposed about a second portion of the bobbin between the second flange and the third flange.

A bone anchored hearing device includes an electromagnetic vibrator for generating a vibration in order to transmit sound through a bone of a user to an ear of the user; and a compensator for at least in part compensating a distortion in the vibration of the electromagnetic vibrator. Further, a signal processing method for a bone anchored hearing device includes providing, by an input transducer, an electric input signal representing sound of a surrounding of a user of the bone anchored hearing device; processing, by a signal processing unit, the electric input signal and providing a processed electric signal; generating, by an electromagnetic vibrator, based on the processed electric signal, a vibration in order to transmit sound through a bone of the user to an ear of the user; and at least in part compensating, by a compensator, a distortion in the vibration of the electromagnetic vibrator.

A microphone system includes a microphone and a pre-amplification conditioning circuit configured within a housing and comprising a pair of matched JFETs configured in a differential pair with common-source configuration and, when biased, are operable to receive and amplify the differential microphone output signal. The microphone further includes a pair of BJTs configured as a complimentary feedback transistor pair with each of the pair of BJTs coupled in parallel to a corresponding one of the pair of matched JFETs, and a current sink coupled to the matched JFETs and corresponding emitter electrodes of the BJTs and operable to maintain a fixed total direct current through each of the matched JFETs and BJTs, which reduces the JFETs corresponding electrical load, reduces signal noise, and increases a maximum amplified microphone output signal level at the drains of the matched JFETs.