178results about How to "Reduce nonlinear distortion" patented technology

Digital filters for processing an input audio signal for extending the audio capabilities and reducing distortion of a loudspeaker. Such extension may include a general sound pressure level (SPL) extension or a more targeted low frequency (LF) extension. This may be accomplished by modifying an original frequency response of the loudspeaker to achieve a particular extension, and applying a digital linear filter based on the modified frequency response to the audio signal. A nonlinear digital filter may also be applied to the audio signal reduce nonlinear distortions of the loudspeaker. The nonlinear digital filter may be based on an inverse of an electro-mechanical model of the loudspeaker. In this manner, a loudspeaker may be driven close to its maximum theoretical power capability without increasing distortion in the loudspeaker output.

A loudspeaker according to the present invention includes: a diaphragm that vibrates back and forth to emit a sound; a magnetic circuit that is provided on a rear side of the diaphragm and has a magnetic gap on a diaphragm side; a voice coil that is directly or indirectly joined to the diaphragm and disposed within the magnetic gap; a magnetic fluid that is loaded within the magnetic gap; and a plurality of first edge pieces that are provided at different positions in an outer circumferential portion of the diaphragm for vibratably supporting the diaphragm, each of the first edge pieces having a non-linear cross-sectional shape.

The invention relates to a power amplifier module comprising a first amplifier 2 having a first front-end 4 and a first backend amplifier stage 5 and a second amplifier 3 having a second front-end 6 and a second backend amplifier stage 7, the first amplifier and the second amplifier being arranged in a Bridge Tied Load (BTL) configuration with feedback over the load,characterized in thatthe first and the second backend amplifier stages having point symmetrical transfer functions with respect to the origin,the input current i1 of the first backend amplifier stage being substantially equal to the input current i2 of the second backend amplifier stage.

The invention relates to a method of reducing Peak-to-Average. Power Ratio in a transmitting device of a filter-hank multi-carrier system, which includes the steps of: performing constellation modulation (210) on data to be transmitted; performing K-point Discrete Fourier Transform (220) on a vector composed of K constellation symbols resulting from the constellation modulation: and performing Offset-Quadrature Amplitude Modulation (230) on a data vector resulting from the Discrete Fourier Transform, wherein the parameter K represents the number of subcarriers allocated for transmission of the data to be transmitted. With the solution proposed by the invention. PAPR of a signal can be significantly reduced without adding a large number of operations to thereby improve the efficiency of a power amplification circuit, to improve effective transmission, power and to alleviate nonlinear distortion of the signal during a power amplification.

A system and method of digitally modeling and significantly reducing the direct coupling between a telephone appliance loudspeaker and microphone employs a handset microphone in a phone appliance, such as, for example, a speakerphone, as a reference channel along with the speakerphone microphone to reduce the direct coupling between the speakerphone loudspeaker and the speakerphone microphone. Using an analog adaptive echo cancellation scheme, the sensitivity and dynamic range of a telephone device microphone to speech from talkers in an enclosure but not from signals emitted by a telephone loudspeaker are enhanced. Nonlinear distortions produced by a telephone appliance loudspeaker are measured by a telephone appliance handset microphone and used in a feedback subsystem, e.g., network, to reduce nonlinear distortions in the telephone appliance loudspeaker and loudspeaker driving circuit.

The present invention provides an advanced adaptive predistortion linearization technique to dramatically reduce nonlinear distortion in power amplifiers over a very wide instantaneous bandwidth (up to 2 GHz) and over a wide range of amplifier types, input frequencies, signal types, amplitudes, temperature, and other environmental and signal conditions. In an embodiment of the invention, the predistortion linearization circuitry comprises (1) a higher-order polynomial model of an amplifier's gain and phase characteristics—higher than a third-order polynomial model; (2) an adaptive calibration technique; and (3) a heuristic calibration technique. The higher-order polynomial model is generated by introducing, for example, a plurality of multi-tone test signals with varying center frequency and spacing into the power amplifier. From the power amplifier's corresponding output, the nonlinearities are modeled by employing a higher-order curve fit to capture the irregularities in the nonlinear transfer function. Different distortion transfer functions can be implemented for different operating conditions. The adaptive calibration technique is based on a feedback analysis technique, which updates the applicable distortion transfer function by analyzing the error signal between the introduced input signal and the output signal in real-time. The heuristic calibration technique implements different distortion transfer functions based on historical operating conditions and optimal configurations of the power amplifier.

A CMOS implemented passive mixer circuit for improving linearity performance in wireless communication systems is described, including dual pairs of NMOS FETs and dual pairs of PMOS FETs. Each NMOS FET is connected in parallel with a corresponding PMOS FET. A local oscillator signal is provided to the gate of one FET while a 180-degree phase shifted local oscillator signal is provided to the gate of its complementary FET. Because the complementary FETs are driven by local oscillator signals that are 180 degrees out of phase, the NMOS FET is turned on for at least a portion of the positive cycle of the local oscillator signal and the PMOS FET is turned on for at least a portion of the negative cycle of the 180-degree phase shifted local oscillator signal. Distortion in the mixed output signal is thereby reduced.

The invention discloses an analog sampling switch and an analog-to-digital converter using the same. The analog sampling switch comprises a sampling switch circuit which comprises a sampling switch tube, and is used for controlling conduction state of the sampling switch tube; and a comparator circuit for comparing input signal level and output signal level of the sampling switch tube, and taking a higher level terminal as a source electrode of the sampling switch tube. The analog sampling switch and the analog-to-digital converter can track the source electrode of the switch tube, eliminate nonlinear relation between the switch conduction resistance and input signals, enhance linearity of the whole sampling switch, and meet high-speed and high-precision requirements of a sampling holding circuit.

The invention discloses a wireless optical orthogonal multi-carrier communication method with a low peak to average power ratio. The wireless optical orthogonal multi-carrier communication method comprises the following steps that at a transmitting end, firstly, modulated frequency domain signals are converted into a semi-positive definite planning convex optimization problem by adopting a tone injection algorithm via a relaxation method, and solving is carried out through general convex optimization and randomized methods; secondly, the frequency domain signals are symmetrically mapped to subcarriers in a conjugate mode, and a cyclic prefix is added after reverse fast Fourier transformation; at last, time domain transmission signals are added into direct-current offset to drive a light-emitting diode to emit light. At a receiving end, photovoltaic conversion is achieved through a photodiode; signals are removed from the cyclic prefix and conjugate symmetry parts after amplification, filtering, analog-digital conversion and fast Fourier transformation; the signals are modulated to be recovered in an original planisphere; at last, receiving symbols are obtained through demodulation. According to the wireless optical orthogonal multi-carrier communication method, the peak to average power ratio of a wireless optical communication PFDM system can be effectively reduced, requirements of a power amplifier and an LED for linear degrees are reduced, non-linear distortion is reduced, and receiving performance is improved.

A system for compensating and driving a loudspeaker includes an open loop loudspeaker controller that receives and processes an audio input signal and provides an audio output signal. A dynamic model of the loudspeaker receives the audio output signal, and models the behavior of the loudspeaker and provides predictive loudspeaker behavior data indicative thereof. The open loop loudspeaker controller receives the predictive loudspeaker behavior data and the audio input signal, and provides the audio output signal as a function of the audio input signal and the predictive loudspeaker behavior data.

The digital signal conversion apparatus comprises: an over-sampling circuit that samples the input digital signal at high frequency; a polarity-inversion circuit that inverts the polarity of the sampled digital signal; interpolation circuit (A) and interpolation circuit (B) that perform interpolation of each respective digital signal; noise-shaping circuit (A) and noise-shaping circuit (B) that perform noise shaping on the interpolated signals; PWM conversion circuit (A) and PWM conversion circuit (B) that perform PWM conversion on the noise-shaped signals; and a switching circuit for driving a load based on the PWM signals from PWM conversion circuit (A) and PWM conversion circuit (B).

An echo canceller has a high-pass filter 11, a speaker 3, an A / D converter 6, and an echo canceller 7. The high-pass filter 11 suppresses a low-frequency component in a received signal. The speaker 3 outputs an acoustic sound of a low-frequency suppressed received signal passed through the high-pass filter 11. The A / D converter 6 converts an acoustic echo from a microphone 4 into a transmission signal of a digital form. The echo canceller 7 generates a pseudo echo signal based on the low-frequency suppressed received signal passed through the high-pass filter 11, and eliminates the acoustic echo to be inputted to the microphone 4 from the speaker 3 by subtracting the pseudo echo signal from the digital signal converted by the A / D converter 6.

The invention relates to a method for reducing a peak-to-average power ratio in a filter-bank multi-carrier system. The method comprises the following steps of: 210, performing constellation modulation on data to be transmitted; 220, performing K-point discrete Fourier transformation on K vectors consisting of constellation symbols obtained by the constellation modulation; and 230, performing offset quadrature amplitude modulation on data vectors obtained through discrete Fourier transformation, wherein the parameter K represents the number of sub carriers distributed for transmission of the data to be transmitted. By the scheme, under the condition that the increase of operations is low, the peak-to-average power ratio of a signal can be obviously reduced, so that the efficiency of a power amplification circuit is improved, the effective emission power is increased, and the nonlinear distortion of the signal in a power amplification stage is reduced.

The present invention provides an advanced adaptive predistortion linearization technique to dramatically reduce nonlinear distortion in power amplifiers over a very wide instantaneous bandwidth (up to 2 GHz) and over a wide range of amplifier types, input frequencies, signal types, amplitudes, temperature, and other environmental and signal conditions. In an embodiment of the invention, the predistortion linearization circuitry comprises (1) a higher-order polynomial model of an amplifier's gain and phase characteristics—higher than a third-order polynomial model; (2) an adaptive calibration technique; and (3) a heuristic calibration technique. The higher-order polynomial model is generated by introducing, for example, a plurality of multi-tone test signals with varying center frequency and spacing into the power amplifier. From the power amplifier's corresponding output, the nonlinearities are modeled by employing a higher-order curve fit to capture the irregularities in the nonlinear transfer function. Different distortion transfer functions can be implemented for different operating conditions. The adaptive calibration technique is based on a feedback analysis technique, which updates the applicable distortion transfer function by analyzing the error signal between the introduced input signal and the output signal in real-time. The heuristic calibration technique implements different distortion transfer functions based on historical operating conditions and optimal configurations of the power amplifier.

An adaptive coding scheme for nonlinear channels improves reliability and an efficiency in digital communication networks. The method monitors channel statistics to analyze an extrinsic information transfer chart of the channels. The channel statistics are fed back to the transmitter to adapt forward error correction coding. A parametric analysis method uses a Gaussian mixture model. The statistical information feedback can adapt an ARQ sheme by adding a weighted received signal to the original coded signals to reduce nonlinear distortion. Trellis shaping, can make the transmitting signal preferable for nonlinear channels.

The present invention provides an advanced adaptive predistortion linearization technique to dramatically reduce nonlinear distortion in power amplifiers over a very wide instantaneous bandwidth (up to 2 GHz) and over a wide range of amplifier types, input frequencies, signal types, amplitudes, temperature, and other environmental and signal conditions. In an embodiment of the invention, the predistortion linearization circuitry comprises (1) a higher-order polynomial model of an amplifier's gain and phase characteristics—higher than a third-order polynomial model; (2) an adaptive calibration technique; and (3) a heuristic calibration technique. The higher-order polynomial model is generated by introducing, for example, a plurality of multi-tone test signals with varying center frequency and spacing into the power amplifier. From the power amplifier's corresponding output, the nonlinearities are modeled by employing a higher-order curve fit to capture the irregularities in the nonlinear transfer function. Different distortion transfer functions can be implemented for different operating conditions. The adaptive calibration technique is based on a feedback analysis technique, which updates the applicable distortion transfer function by analyzing the error signal between the introduced input signal and the output signal in real-time. The heuristic calibration technique implements different distortion transfer functions based on historical operating conditions and optimal configurations of the power amplifier.

The invention relates to a stereo sound effect enhancement system, which comprises a signal separation module, a dialogue enhancement module, a decorrelation processing module, a signal synthesis module, an environment sound effect module, a virtual bass enhancement module and a dynamic range control module. The system is used for preprocessing a received stereo signal and outputting the stereo signal to a playing device to obtain a desired sound effect.

The invention relates to a dynamic compensation method for reducing LED (Light Emitting Diode) nonlinear distortion of a visible light Flip-OFDM (Orthogonal Frequency Division Multiplexing) communication system. The method comprises the following steps: segmenting a modulated subcarrier data vector into a plurality of non-overlapped subsequences; multiplying by a corresponding weighting coefficient, extending to be a hermitian symmetric sequence and then performing IFFT (Inverse Fast Fourier Transform); selecting a weighting coefficient for keeping the negative signal amplitude of a combined signal to be minimum; decomposing the polarity of the combined signal into a Flip-OFDM positive signal and a flip negative signal and limiting the amplitude; carrying an end cut signal of the positive signal at a zero signal position corresponding to the flip negative signal as a compensation signal; restoring the positive signal by a receiving end through the compensation signal. By using the method, PAPR (Peak to Average Power Ratio) in the Flip-OFDM system can be effectively suppressed, and the LED nonlinear distortion is reduced, so that the usability and the reliability of the system are greatly improved.

A compensated peak-to-average ratio non-linear distortion new algorithm based on a compression perception technology belongs to the communication technology field. The algorithm comprises the following steps of (1) based on a CS theory, using an empty carrier wave to carry out peak-to-average ratio amplitude limiting compensation; and (2) carrying out OMP error tolerance value estimation based on an LS algorithm. The algorithm has advantages that (1) a receiver can accurately recover an amplitude of a signal before amplitude limiting; (2) generation of an amplitude limiting noise is greatly reduced, and simultaneously non-linear distortion of a linear amplifier is reduced so that system performance is effectively increased; and (3) algorithm complexity is low and an obvious improvement effect is possessed for an aspect of amplitude limiting noise reduction. The improved amplitude limiting algorithm based on compression perception is better than a common amplitude limiting algorithm, certain practicality is possessed and a certain reference value is possessed for a high-speed real-time OFDM underwater sound communication system.

A system for compensating and driving a loudspeaker includes an open loop loudspeaker controller that receives and processes an audio input signal and provides an audio output signal. A dynamic model of the loudspeaker receives the audio output signal, and models the behavior of the loudspeaker and provides predictive loudspeaker behavior data indicative thereof. The open loop loudspeaker controller receives the predictive loudspeaker behavior data and the audio input signal, and provides the audio output signal as a function of the audio input signal and the predictive loudspeaker behavior data.

The invention discloses a wine bucket shaped loudspeaker box, comprising a loudspeaker box body and a loudspeaker box bracket; the loudspeaker box body comprises a wine bucket shaped box body and a loudspeaker arranged at the front end surface of the box body, a phase-reversing tube is arranged in the box body, a phase-reversing hole is arranged at the rear end surface of the box body, one end of the phase-reversing tube is arranged at the position of the phase-reversing hole and is vertically and fixedly connected with the rear end surface of the box body, the loudspeaker is electrically connected with an external sound source by a loudspeaker wiring terminal, wave mode sound absorption wool is arranged on the inner wall of the box body, the wine bucket shaped loudspeaker box eliminates the harmonic distortion and resonance of tone quality, caused by flat box body surfaces and edge corners of a common resonance, increases low-frequency diving power and reduces nonlinear distortion of audio frequency, the design of the wine bucket shaped box body changes the disadvantage that the common loudspeaker box has poor shock resistance, promotes the mechanical property of the loudspeaker box structure and can support high audio-frequency power; in addition, the wine bucket shaped appearance design of the loudspeaker box is artistic and elegant, and the function and the decoration are perfectly combined.

A pulse width modulator for producing a PWM signal having reduced nonlinear distortion through interpolation processing with fewer computation steps is provided. The computational processing, W={1+α(X2−X1)}{0.25 (X1+X2)}+0.5, is performed in accordance with successive sample values (X1, X2) in a PCM data train to determine a weighting factor (W). The computational processing, Xq=0.5·X0(W2−W)+X1(1−W2)+0.5·X2(W2+W), is performed using the sample values (X1, X2), a sample value (X0) previous to the sample value (X1), and the weighting factor (W) to thereby determine an interpolated sample value (Xq) having an amplitude close to that of an original analog signal (X(t)) generating the PCM data train. A point in time (tq) at which a reference signal (R(t)) takes on the interpolated sample value (Xq) is then determined to produce a PWM signal (Spwm) which is logically inverted at the point in time (tq).

In each single-pole double-throw membrane switch, there is a microwave T type head composed of microstrip lines including one signal line as input end of high frequency signal, and two connected offset signal lines in symmetrical T shape structure. A same single-pole single-throw membrane switch is connected to each offset signal line. In microwave T type head, total of effective length b of each offset signal line and length a from offset signal line to middle point of coplane wave-guide signal line is equal to 1 / 4 wavelength of input high frequency signal. Control circuit controls on / off of two sets of single-pole single-throw to realize function of single-pole double-throw. Comparing with MEMS membrane switch, the invention solves high threshold voltage and interference of control voltage on transmission signal. Advantages are: low threshold voltage, isolation and insertion loss, and high reliability, and repeatability.

The invention belongs to the technical field of communication, and relates to an OFDM-based Radar and Communication (RadCom) system. The method is based on the traditional OFDM system, and is characterized in that at a transmitting end, bit streams to be sent are modulated into data symbols, local waveform retention designing is carried out according to a data bandwidth ratio through the data symbols and a random phase sequence to obtain RadCom frequency domain signals, the signals are then mapped into the time domain through IFFT, Cyclic Prefix (CP) addition is carried out on the mapped signals, and signals obtained through the CP addition are sent to a channel through a radio frequency front end; at a receiving end, CP removal is carried out on the received signals, signals obtained through the CP removal are mapped into the frequency domain through FFT, the mapped signals are equalized to compensate for channel distortions, symbol extraction is carried out on the equalized symbols according to the data bandwidth ratio, and bit information is finally obtained through symbol demodulation; and meanwhile, the frequency domain signals at the transmitting end and the receiving end areused for radar processing. The system provided by the invention has the advantages that the partly retention loop algorithm is introduced, so that the bandwidth can be flexibly allocated under the premise that the advantages of a communication system are maintained, the PAPR can be effectively reduced, and the spectrum utilization efficiency can be improved.

The typical micro-transducer dedicated for mobile phones and headphones produces at high output in small back enclosure significant distortion in the acoustical output. These typical micro-transducers are designed for very high sensitivity and their mechanical suspension compliances are designed for worst case scenario, free air environment or a very large back enclosure (were the air stiffness from the back enclosure is much lower than the mechanical transducer suspension stiffness). This leads to poor low frequency response, high distortion, non-linear SPL characteristic and unnecessary high system resonance fre-quency when the transducer is placed in a small back enclosure. The Soft Suspension Long throw (SSL) micro-transducer has been designed with an improved non-circular shaped magnetic system, a low density / high Young's modulus thermal conducting diaphragm, a large-area non-circularly shaped diaphragm area, a non-circular shaped voice coil and a soft long-throw suspension which has been designed so that the SSL micro-transducer only functions properly when mounted in a small back enclosure. The combination of these inventions leads to a transducer architecture offering significant improvements in overall distortion, high RMS power handling, extended bass reproduction, also in compact volumes, and an overall flat frequency response. The combined result of several inventions is a significant overall improvement in perceived sound quality.

In an active hybrid transformer circuit connected to both transmit and receive sides, a current driver is connected to a transmission path circuit through a common connection point of a load resistor and a replica resistor while a replica driver is connected to the replica resistor of which another connection point is directly connected to the receive side. The replica driver may be formed by a DAC of a current output type. A current ratio of the replica driver and the current driver is selected at a predetermined value so as to prevent a transmit signal from leaking into the receive side.

The invention discloses an X-waveband power amplifier based on GaN. The X-waveband power amplifier comprises MIN capacitors, a filter capacitor, a thin-film resistor, a stable resistor, a spiral inductor, a microstrip transmission line, a T-shaped connector and a GaN-based HEMT. A signal input end (RFIN) is connected with an input end (1) of the GaN-based HEMT (H) through the first MIN capacitor (C1), the microstrip transmission line (W) and the T-shaped connector (T). A drain electrode control end (Vds) is connected with an output end (2) of the GaN-based HEMT through the microstrip transmission line (W) and the T-shaped connector (T). The microstrip transmission line (W) is a 100-micrometer high-impedance microstrip transmission line. A 100-microfarad electrolytic capacitor is selected as the filter capacitor. The GaN-based HEMT (H) is an NRF01-02a HEMT tube core. The thin-film resistor is made of NiCr material. The X-waveband power amplifier is matched with the input end in a conjugate mode through a load traction method so as to solve the problem that negative resistance occurs at the port of the transistor; the X-waveband power amplifier is stable in work, has the bandwidth of 3.6 GHz to 8.0 GHz, the maximum gain of 11.04 dB, the maximum output power of 33 dBm and the maximum PAE of 29.2%, and is small in voltage standing wave ratio.

Proposed are a method and device for generating a discrete pilot signal of a digital audio broadcast signal. The method includes: A, in accordance with a system frame structure, filling a pilot symbol obtained by subjecting a bit stream pair to QPSK constellation mapping to a position of an effective sub-carrier where a discrete pilot element is located according to a predetermined mode; B, subjecting the processing result of step A to IFFT to obtain a time-domain OFDM symbol; and C, multiplexing the time-domain OFDM symbol and a beacon, and obtaining a physical frame signal containing a discrete pilot signal through logical frame framing, subframe allocation and physical layer signal frame generation. The device comprises: a filling module, a transformation module and a framing module. Adopting the method and device, the peak-to-average power ratio of a transmitting signal may not increase, reducing the nonlinear distortion and improving the signal quality.

In an active hybrid transformer circuit connected to both transmit and receive sides, a current driver is connected to a transmission path circuit through a common connection point of a load resistor and a replica resistor while a replica driver is connected to the replica resistor of which another connection point is directly connected to the receive side. The replica driver may be formed by a DAC of a current output type. A current ratio of the replica driver and the current driver is selected at a predetermined value so as to prevent a transmit signal from leaking into the receive side.

An approach for improved compensation for nonlinear distortion in multicarrier satellite systems is provided. Source reflecting encoded and modulated sequences of source data symbols are received. Each source signal is predistorted, and a transmit filter is applied to each predistorted source signal. Each filtered signal is translated to a carrier frequency, and the translated signals are combined into a composite signal for transmission via a multicarrier transponder. The final predistorted version of each source signal is generated via an iterative process of a number of stages, wherein, for a given stage and for each source signal, the process comprises: receiving a prior predistorted version of each source signal from a preceding stage; processing each prior predistorted source signal based on all of the received prior predistorted source signals, wherein the processing is performed based on a characterization of one or more characteristics of the multicarrier satellite transponder.