542results about "Transducer circuit damping" patented technology

Control system for devices such as an audio reproduction system, an actuator device, an electromechanical device and a telephony device. The system includes control circuitry which receives an input signal and a signal indicative of a position of a portion of the controlled apparatus. The control circuit provides an output signal to the controlled apparatus to affect an operation of the controlled apparatus. The output signal provides control of the apparatus to compensate for one or more of: motor factor; spring factor; back electromotive force; and impedance of a coil in a driver of the controlled apparatus. The signal indicative of position is derived by one or more position indicator techniques such as an infrared LED and PIN diode combination, position dependent capacitance of one portion of the controlled apparatus with respect to another portion of the controlled apparatus, and impedance of a coil in the controlled apparatus. The control circuitry is configurable to control transconductance and / or transduction of the system being controlled. A technique is disclosed to detect and measure a cant of a voice coil transducer, the technique including measuring a capacitance between one portion of the voice coil transducer with respect to another portion of the voice coil transducer over a range of movement of the voice coil during operation.

Loudspeakers can be damaged by high drive signals. One reason for this damage is an excess vibration displacement of the coil-diaphragm assembly. This invention describes a novel method for limiting this displacement by a signal processor. In the present invention, a low frequency shelving and notch filter is used to attenuate low frequencies according to a prediction of the loudspeaker displacement. A novel method for calculating coefficient values for a digital implementation of the low frequency shelving and notch filter according to the predicted displacement is described.

Output control system and method includes calculating gain for an input signal, detecting a predetermined condition, and modifying the gain calculation in accordance with one or more kneepoints to provide a varying output signal. A companding ratio is then established in accordance with the kneepoints. System gain is then adjusted, and the companding ratio may be further adjusted by adjusting one of the kneepoints relative to another.

Systems and methods for utilizing inter-microphone level differences to attenuate noise and enhance speech are provided. In exemplary embodiments, energy estimates of acoustic signals received by a primary microphone and a secondary microphone are determined in order to determine an inter-microphone level difference (ILD). This ILD in combination with a noise estimate based only on a primary microphone acoustic signal allow a filter estimate to be derived. In some embodiments, the derived filter estimate may be smoothed. The filter estimate is then applied to the acoustic signal from the primary microphone to generate a speech estimate.

There is provided a noise cancellation system, comprising: an input for a digital signal, the digital signal having a first sample rate; a digital filter, connected to the input to receive the digital signal; a decimator, connected to the input to receive the digital signal and to generate a decimated signal at a second sample rate lower than the first sample rate; and a processor. The processor comprises: an emulation of the digital filter, connected to receive the decimated signal and to generate an emulated filter output; and a control circuit, for generating a control signal on the basis of the emulated filter output. The control signal is applied to the digital filter to control a filter characteristic thereof.

Described is a unit for amplifying and processing audio signals for driving an electro-acoustic transducer (D), comprising: an input for audio signals; a processor for audio signals (107); an output for a signal for driving said electro-acoustic transducer; and an input for at least one operating quantity of the electro-acoustic transducer. The audio-signal processor is programmed for setting a series of parameters defining a transducer to be emulated, the parameters of which define a model of the transducer to be emulated. The input audio signal is processed on the basis of said at least one operating quantity of the electro-acoustic transducer to obtain a behavior of the electro-acoustic transducer that emulates the transducer defined by said series of parameters set.

A sound masking system for shaping the ambient noise level in a physical environment. The sound masking system comprises a networked and distributed system having a number of master units coupled together and to a control unit. One or more of the master units may include satellite sound masking units which function to reproduce the sound masking signal generated by the master sound masking unit. Each of the master units is addressable over the network by the control unit enabling the control unit to program the contour, spectral band, and gain characteristics of the sound masking output signal. The system may also include a remote control unit which provides the capability to tune and adjust each master sound masking unit in situ without requiring physical access through the ceiling installation.

An audio power management system manages operation of audio devices in an audio system. The audio power management system includes a parameter computer, a threshold comparator and a limiter. Audio signals generated with the audio system may be provided to the audio power management system. Based on a measured actual parameter of the audio signal, such as a real-time actual voltage and / or a real-time actual current, the parameter computer can derive estimated operational characteristics of audio devices, such as a loudspeaker included in the audio system. The threshold comparator may use the estimated operational characteristics to develop a threshold and manage operation of one of more devices in the audio system by monitoring the measured actual parameter, and selectively directing the limiter to adjust the audio signal, or another device in the audio system to protect or optimize performance.

A method and an apparatus for compensating for nonlinear distortion are provided to divide audio signals reproduced in a nonlinear speaker system into linear and nonlinear components in a time domain and a frequency domain, and then generate inversely-corrected signals by means of an inverse filtering scheme, so that it is possible to further consider a variety of nonlinear distortion characteristics such as viscous damping and structural damping which have not been reflected in the conventional lumped parameter method, and thus to obtain better sound quality.

An apparatus includes first and second microphone arrangements, arranged to output first and second signals respectively and is operable in a first mode and a second mode. In the first mode, an output signal is generated based on the second signal and a third signal, where the second signal and, optionally, the first signal, can be used to compensate for ambient noise, for example, for noise cancellation when a telephone call is relayed through a speaker. In the second mode, an output signal is generated based on the first and second signals. In this manner, the combination of the first and second microphone arrangements provides a directional sensitivity that can pick up sound from a remote source, for example, in an audio or video recording session. The apparatus may include a sensor to allow automatic switching between one or more of modes, directional sensitivity patterns and types of recording session.

The invention provides a method for determining a set of filter coefficients for an acoustic echo compensator in a beamformer arrangement. The acoustic echo compensator compensates for echoes within the beamformed signal. A plurality of sets of filter coefficients for the acoustic echo compensator is provided. Each set of filter coefficients corresponds to one of a predetermined number of steering directions of the beamformer arrangement. The predetermined number of steering directions is equal to or greater than the number of microphones in the microphone array. For a current steering direction, a current set of filter coefficients for the acoustic echo compensator is determined based on the provided sets of filter coefficients.

A noise control device according to the present invention is capable of reducing noises respectively arriving in a plurality of spaces which are acoustically independent from each other. The noise control device comprises: sound output means, which are respectively provided in the plurality of spaces so as to respectively correspond to the plurality of spaces, each for outputting a sound to a corresponding space; noise detection means, which is provided in at least one of the plurality of spaces, for detecting a noise arriving in the at least one of the plurality of spaces; and signal generation means which is a single means for generating, based on the noise detected by one of the noise detection means, a cancellation signal for canceling the noise, and outputting the generated cancellation signal to each of the plurality of sound output means.

Techniques for estimating adaptive noise canceling (ANC) performance in a personal audio device, such as a wireless telephone, provide robustness of operation by triggering corrective action when ANC performance is low, and / or by saving a state of the ANC system when ANC performance is high. An anti-noise signal is generated from a reference microphone signal and is provided to an output transducer along with program audio. A measure of ANC gain is determined by computing a ratio of a first indication of magnitude of an error microphone signal that provides a measure of the ambient sounds and program audio heard by the listener including the effects of the anti-noise, to a second indication of magnitude of the error microphone signal without the effects of the anti-noise. The ratio can be determined for different frequency bands in order to determine whether particular adaptive filters are trained properly.

A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to measure the output of the transducer in order to control the adaptation of the anti-noise signal and to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit that performs the adaptive noise canceling (ANC) function also detects frequency-dependent characteristics in and / or direction of the ambient sounds and alters adaptation of the noise canceling circuit in response to the detection.

A signal processing device includes an adaptive filter (5), a noise estimation circuit (10), and a double talk detection circuit (81) and operates so that the double talk detection circuit (81) detects a double talk by using the estimated noise obtained by the noise estimation circuit (10). The signal processing device further includes noise estimation means and detects a double talk by using an estimated noise, a microphone signal, and pseudo-echo. An echo removal method and device detect a double talk by using a reliability coefficient expressed as continuous values between 0 and 1. By using continuous values instead of two values 0 and 1, it is possible to reduce the affect of a detection error.

A loudspeaker system with a transducer and an enclosure with at least one resonant chamber, including a resonant-chamber resonance frequency, at which a displacement characteristic of a vibratile diaphragm of the transducer has a minimum. The loudspeaker system further includes a multi-mode signal processor with a set of signal processes wherein in one example, a variable gain, frequency selective dynamic filter reduces the gain in a high displacement frequency range upon an output in the frequency range exceeding an overload amplitude threshold. The gain reduction is compensated for by complimentary frequency generating signal processes, including at least one of a harmonics generator and a transpositional gain controller, with the signal processor adapted to match the resonant chamber enclosure by substantially maintaining or increasing acoustic output at the resonant chamber resonance and generating harmonics associated with fundamental frequencies in the gain reduced frequency range to maintain a perception of tonal quality and physical bass impact, increasing low frequency capability while minimizing audible overload distortion.

The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.

An earpiece for real-time audio processing of ambient sound includes an ear bud that provides passive noise attenuation to the earpiece such that exterior ambient sound is substantially reduced within an ear of a wearer, an exterior microphone that receives ambient sound and converts the received ambient sound into analog electrical signals, and an analog-to-digital converter that converts the analog electrical signals into digital signals representative of the ambient sounds. The earpiece further includes a digital signal processor that performs a transformation operation on the digital signals according to instructions received from a mobile device, the transformation operation transforms the digital signals into modified digital signals, a digital-to-analog converter that converts the modified digital signals into modified analog electrical signals, and a speaker that outputs the modified analog electrical signals as audio waves.

A Personal Noise Reduction for enclosed cabins is disclosed. According to one embodiment, an input audio source corresponding to sound received from multiple microphones situated equidistantly in both directions in a two dimensional plane, is converted to a digital signal via an analog to digital (A / D) convertor. The AID converted audio is analyzed for content to identify ambient noise. The frequency, amplitude and phase of the identified ambient noise is subsequently determined. A Noise correction sound wave is generated with negative phase of that corresponding to the identified ambient noise. The noise correction sound wave is added to the identified noise to create a noise corrected sound.

There is provided a filter circuit for producing a noise reduction signal for reducing a noise signal collected by a microphone, including: a digital section including an analog / digital conversion section configured to convert the noise signal into a digital noise signal, a digital filter section configured to produce a digital noise reduction signal based on the digital noise signal, and a digital / analog conversion section configured to convert the digital noise reduction signal into an analog noise reduction signal; an analog path connected in parallel to said digital section and configured to output the noise signal as it is or after processed by an analog filter; and a synthesis section configured to synthesize the analog noise reduction signal outputted from said digital / analog conversion section of said digital section and the analog signal outputted from said analog path to produce a noise reduction signal to be used for noise reduction.

The invention provides one sound producing system including a speaker, a driving circuit and an audio controller. The speaker is used for producing an audible sound. The driving circuit is connected to the speaker. The audio controller is configured to receive an input audio signal and pre-amplify the input audio signal with an amplify gain to obtain a pre-amplified audio signal; operate a multiband dynamic range control on the pre-amplified audio signal to obtain an output audio signal; convert the output audio signal to a driving voltage; provide the driving voltage to the speaker through the driving circuit; detect over the driving circuit to obtain at least one excursion parameter to determine an estimated excursion of the speaker in response to the driving voltage; and adjust the amplify gain according to the estimated excursion.

An apparatus for improving a perceived quality of sound reproduction of an audio output signal is provided. The apparatus has an active noise cancellation unit for generating a noise cancellation signal based on an environmental audio signal, wherein the environmental audio signal has noise signal portions, the noise signal portions resulting from recording environmental noise. Moreover, the apparatus has a residual noise characteristics estimator for determining a residual noise characteristic depending on the environmental noise and the noise cancellation signal. Furthermore, the apparatus has a perceptual noise compensation unit for generating a noise-compensated signal based on an audio target signal and based on the residual noise characteristic. Moreover, the apparatus has a combiner for combining the noise cancellation signal and the noise-compensated signal to obtain the audio output signal.

Methods and systems are provided for acoustic echo cancellation in electronic devices. The echo cancellation may comprise applying, as a first step, echo cancellation filtering to an acoustic input obtained via an acoustic input element (e.g., microphone), and applying, as a second step, echo suppression to the acoustic input, wherein the echo suppression comprises suppressing residual echo in the acoustic input. The echo cancellation filtering may comprise identifying and / or filtering out echo components, both linear and nonlinear, in the acoustic input, with the echo components corresponding to an echo signal caused by an acoustic output outputted via the acoustic output element (e.g., speaker). A sensor signal, generated by a vibration sensor that detects vibrations in the electronic device including vibrations caused by the outputting of the acoustic output, may be used as reference signal in the echo cancellation filtering and / or the echo suppression.

A system and method for reducing or entirely canceling background or ambient noise from a voice transmission from a communications device. A communications device, such as a mobile telephone, is configured with an ambient noise compensation signal generator that is connected between a microphone and a mixer. The original output of the microphone and a compensation signal generated by the ambient noise compensation signal generator are mixed together prior to being passed to a transmitter. In one embodiment a buffer is provided between the microphone and the mixer to help synchronize the timing of the signals to be mixed. In another embodiment a second microphone is employed to detect ambient noise.

A sound producing system includes a loudspeaker to generate an audible signal. A transducer generates a pressure feedback signal from the audible signal. A controller produces a controlled signal for application to the loudspeaker. The controller derives the controlled signal from the pressure feedback signal and an audio input signal. The controlled signal improves the Sound Pressure Level (SPL) frequency response and reduces distortion within a selected bandwidth of the loudspeaker.

A method of estimating diaphragm excursion of an electrodynamic loudspeaker may be performed using audio signals. An audio output signal may be applied to a voice coil of the electrodynamic loudspeaker through an output amplifier to produce sound. A detected voice coil current and a determined voice coil voltage may be applied to a linear adaptive digital loudspeaker model that has a plurality of adaptive loudspeaker parameters. The parameter values of the adaptive loudspeaker parameters may be computed based on the linear adaptive digital loudspeaker model and applied to a non-linear state-space model of the electrodynamic loudspeaker. For the non-linear state-space model, a predetermined non-linear function may be applied to at least one of the plurality of received parameter values to compute at least one non-linearity compensated parameter value of the adaptive loudspeaker parameters, to determine an instantaneous excursion of the diaphragm.

A speaker system according to the present invention includes: a cabinet; at least one speaker unit fixed to the cabinet; and a gas adsorbent which is situated inside the cabinet and which is made from a porous material. The speaker unit is configured with moisture-proof component parts. In the speaker system, a tubular structure which has a tubular hollow for allowing ventilation between an inside and an outside of the cabinet is provided. A resonant frequency which is determined by an acoustic impedance of the tubular structure and an acoustic impedance of the cabinet is lower than a minimum resonant frequency of an acoustic impedance of the speaker system.

A method for echo cancellation in multichannel audio signals includes receiving a plurality of time-domain signals, including multichannel audio signals and at least one reference signal, transforming the time-domain signals to K under-sampled complex-valued subband signals using an analysis filter bank, and performing, for each of the K under-sampled complex-value subband signals, linear echo cancellation of the reference signal from each channel using an acoustic echo canceller. A probability of acoustic echo dominance is produced using a single-double talk estimator, and a semi-blind multichannel source separation is performed based on the probability and independent component analysis (“ICA”) to decompose the audio signals into a near-end source signal and a residual echoes using subband semi-blind source separation. The residual echo components are removed from the near-end source signal using a spectral filter bank, and the subband audio signals are reconstructed to a multichannel time-domain audio signal using a subband synthesis filter.