395 results about "Analogue filter" patented technology

A waveform adaptive transmitter that conditions and/or modulates the phase, frequency, bandwidth, amplitude and/or attenuation of ultra-wideband (UWB) pulses. The transmitter confines or band-limits UWB signals within spectral limits for use in communication, positioning, and/or radar applications. One embodiment comprises a low-level UWB source (e.g., an impulse generator or time-gated oscillator (fixed or voltage-controlled)), a waveform adapter (e.g., digital or analog filter, pulse shaper, and/or voltage variable attenuator), a power amplifier, and an antenna to radiate a band-limited and/or modulated UWB or wideband signals. In a special case where the oscillator has zero frequency and outputs a DC bias, a low-level impulse generator impulse-excites a bandpass filter to produce an UWB signal having an adjustable center frequency and desired bandwidth based on a characteristic of the filter. In another embodiment, a low-level impulse signal is approximated by a time-gated continuous-wave oscillator to produce an extremely wide bandwidth pulse with deterministic center frequency and bandwidth characteristics. The UWB signal may be modulated to carry multi-megabit per second digital data, or may be used in object detection or for ranging applications. Activation of the power amplifier may be time-gated in cadence with the UWB source thereby to reduce inter-pulse power consumption. The UWB transmitter is capable of extremely high pulse repetition frequencies (PRFs) and data rates in the hundreds of megabits per second or more, frequency agility on a pulse-to-pulse basis allowing frequency hopping if desired, and extensibility from below HF to millimeter wave frequencies.

In one embodiment, the present invention includes a test signal generator capable of producing an impedance test signal comprising of a sine wave having a known frequency. The test signal generator may include a crystal oscillator, a counter, and a lookup table. The lookup table output is applied to a digital to analog converter and is then low pass filtered using a conventional analog filter to produce a sine wave of a known frequency and voltage amplitude. The test signal flows through the electrode and combines with an electrophysiological signal to form a combined signal. A signal processor is used to isolate the combined signal into the test signal component and the electrophysiological component. The signal processor digitally low pass filters the combined signal and the output of the low pass filter is the electrophysiological signal. The signal processor then digitally bandpass filters the combined signal using a filter with a center frequency which is the same as the test frequency. The output of this filter is then used to calculate the electrode impedance.

A digital to analog converter includes a digital processor having an input port adapted to receive an input signal and an output port coupled to an input port of an analog filter wherein the digital processor includes a digital feedback loop which compares a reference digital voltage with a digital voltage provided by a digital model of the analog filter. Using a completely digital feedback loop which compares an input digital voltage with the digital voltage from the digital model of the analog filter results in a single bit digital to analog converter having improved accuracy for a given clock rate and filter. The next digital state of the converter (i.e. ‘0’ or ‘1’) is selected based upon a comparison of the input (or reference) voltage with the digital voltage provided by the digital feedback loop. The digital converter output is then fed to the analog filter. If the analog filter matches the digital model, then the analog voltage will match the digital voltage, and therefore the reference voltage.

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.

A DSP-based and multi-channel digital filter chip providing crossover filtering and parametric equalization. Crossovers can be low-pass, high-pass, or band-pass filters with programmable cutoff frequencies. Each equalizer band has independently adjustable center frequency band, and levels over a -60 dB to +16 dB range. The chip can work either with a microprocessor controlling the chip or in a stand-alone mode having the filter settings downloaded automatically from an external EEPROM. The chip is ideal for applications that require precise digital filtering or software programmable filters, and is also an alternative to analog filters, eliminating passive components and reducing circuit size.

The invention relates to a cascaded scheme in which an RRC filter, a modified RRC filter or other digital filter is implemented at a relatively low data rate, such as twice the symbol or chip rate, or 2×. Interpolation filters are used to increase the data rate to a higher data rate, such as 8×. Decimation filters are used to reduce the data rate from a higher rate, such as 8×, to a lower rate, such as 2×. The coefficients of the digital filter may be adjusted to compensate for characteristics of other components across the entire filter chain. Most of the implementation complexity of the filter chain is consolidated into the relatively low rate (such as 2×) digital filter while interpolation or decimation filters can be implemented at very low cost. The compensation capability provided by the digital filter makes design of simple decimation or interpolation filters much easier. The compensation capability provided by the digital filter, along with the relatively high over-sampling rate for the output of the interpolation filter (or the input of the decimation filter), also makes the design of the preceding (or following) analog filters much easier.

A power amplifier system including a composite digital predistorter (DPD) ensuring optimized linearity for the power amplifier is described. In this system, a digital-to-analog converter (DAC), an analog filter, a first mixer, and the power amplifier are serially coupled to the composite DPD. A second mixer, a receive gain block, and an analog-to-digital converter (ADC) are serially coupled to the output of the power amplifier. A DPD training component is coupled between the inputs of the composite DPD and the ADC. The composite DPD includes a memory-based DPD, e.g., a memory polynomial (MP) DPD, a memoryless-linearizing DPD, e.g., a look-up table (LUT) DPD, and two multiplexers.

Methods and apparatuses for obtaining position and energy information without pileup. Signal integration, which is triggered by a present event, stops when a subsequent event is detected. A weighted value for estimating the total energy in a scintillation is calculated, which includes the energy of the current event and a residual energy from previous events. Remnant correction is used to calculate a pile-up free energy from two consecutive weighted values. An analog filter may be applied to reduce noise. Dynamic digital weighting of integrated values, and/or digital integration may be used during data processing. Pileup can be avoided in conjunction with several types of applications, including multi-zone detector applications and coincidence detection applications. High-resolution timing techniques are also disclosed that facilitate one's ability to avoid pileup.

There are provided a method and an apparatus for performing a self-interference cancellation by determining a filter coefficient of an analog filter operated in an analog domain; and canceling self-interference generated in a received signal received by the node by a transmitted signal transmitted from the node, based on the filter coefficient.

A multistandard RF transceiver is disclosed that may optionally include selectable mixers; selectable amplifiers; a configurable analog filter; and a configurable analog to digital converter. The multistandard RF transceiver may also include a data interface for sending data to a host controller and a control interface for receiving configuration commands from the host controller. The configuration commands identify a wireless standard that is to be implemented by the RF receiver. An RF processor processes an RF signal wherein the processed RF signal is output to the host controller on the data interface.

A technique for suppressing quantization noise generated due to digitizing an analog circuit in a PLL circuit is provided. The PLL circuit comprises: a digital phase frequency detector which detects (compares) phases and frequencies of a reference signal and a frequency-divided signal and converts the same to a digital value; a digital loop filter which eliminates high-frequency noise components from an output of the digital phase frequency comparator; a digital-analog converter which converts a digital value of an output of the digital loop filter to an analog value; an analog filter which eliminates a high-frequency noise component from an output of the digital-analog converter; a voltage controlled oscillator whose frequency is controlled based on an output of the analog filter; and a frequency divider which divides the frequency of the voltage controlled oscillator and outputs the frequency-divided signal.

A method of eliminating interference components in a microphone signal by generating a compensation signal and subtracting the compensation signal from the microphone signal. The compensation occurs completely in the frequency domain and the output signal is processed in the frequency domain as well. Measures for reducing the expenditure during the signal processing are specified. For example, advantageous modifications provide that a filter setting, obtained during a preceding speech pause, be used for eliminating interference in a voice signal and/or that the simulation filter be divided into several partial filters for long pulse responses. In particular, the invention is suitable for eliminating interference signal components, e.g., caused by a radio or the like, from a voice input signal in a motor vehicle, the source signal of which is available as reference signal.

A broadband multibit digital radio-frequency (RF) signal is synthesized digitally. to convert the digital signal to a high-power analog signal for RF transmission. Each bit (or cluster of bits) of the digital signal is first separately amplified using a fast switching amplifier with a controlled dc power supply voltage. The DC voltages are weighted to match the significance of the bits, and controlled by a set of calibrated DC reference sources to maintain high precision. The amplified digital signals from the various bits are then combined and passed through an appropriate analog filter to generate the RF signal to be transmitted. Such a signal can exhibit broad bandwidth, high dynamic range, excellent linearity, and low noise. Preferred embodiments of this system can incorporate superconducting electronic elements. For ultimate precision, a set of primary or secondary DC voltage standards can be used to regulate the switching amplifier supply voltages.

A clock recovery circuit in a digital display unit for recovering a time reference signal associated with analog display data. The clock recovery circuit includes a phase-locked loop (PLL) implemented in digital domain and an analog filter to eliminate any undesirable frequencies from the output signal of the PLL. The PLL includes independent control loops to track long term frequency drifts of the time reference signal and the transient phase differences respectively. By providing such independent control loops, the generated clock can be better synchronized with the time reference signal.

A lithology while drilling and reservoir characteristics recognizing method comprises the following steps: (1), establishing data analysis software and hardware equipment; (2), mounting vibration signal measuring equipment; (3), transmitting vibration acoustic wave signals by a vibration transducer to an analog filter; (4), filtering useless frequency bands in the signals and transmitting useful frequency bands to a high fidelity amplifier by the analog filter; (5), amplifying, strengthening, and transmitting the signals to a digital filter by the high fidelity amplifier; (6), transmitting power spectral density of the signals to a wavelet transformation unit by a power spectrum analysis unit, (7), performing wavelet decomposition, reconstruction and noise elimination on the signals by the wavelet transformation unit; and (8), unfolding restored true signals in spectrographs and energy spectrum charts by a Fourier transformation unit, and determining multiple kinds of lithology and reservoir fluid characteristics by recognizing each group of spectrum lines. The method detects and filters the vibration acoustic wave signals, extracts stratum information, and obtains the lithology and the reservoir fluid characteristics through noise elimination and spectral analysis.

A DC offset cancellation circuit employs multiple feedback factors for canceling DC offset by increasing convergence time. The DC offset cancellation circuit features an active low-pass analogue filter including a variable resistor, an amplifier, a capacitor pair and a comparator, the amplifier being coupled to an output of the variable resistor. Negative feedback is provided from the output of the variable resistor to the input signal source, the capacitor pair is coupled to the amplifier, one input of the comparator is coupled to an output of the amplifier, and a second input of the comparator is coupled to a reference voltage source such that a comparison signal is output by the comparator to the variable resistor after comparing the negative feedback signal with a reference voltage.

There is provided an over-sampling D/A converter which has a mute function for fixing an average DC potential of an analog output signal to a predetermined potential, and comprises a SIGMA DELTA modulator for receiving a multibit digital signal to which a DC offset value is added and then outputting a one-bit non-return-to-zero signal, a signal generator for receiving the non-return-to-zero signal and the clock signal, then generating a return-to-zero signal which is a logical multiplication of the non-return-to-zero signal and the clock signal and a complementary signal of the return-to-zero signal which is a logical addition of the non-return-to-zero signal and an inverted signal of the clock signal, and then adding the return-to-zero signal to the complementary signal of the return-to-zero signal to thus output a polar-return-to-zero signal, and an analog filter for receiving the polar-return-to-zero signal and then outputting an analog signal. Accordingly, because an average DC potential of an output signal at the time of the mute operation-ON is set substantially equal to the average DC potential of the output signal at the time of the mute operation-OFF, variation in the average DC potential in the output signal due to ON/OFF of the mute operation can be prevented. As a result, generation of audible click noises can be prevented.

A system and method is used to tune filters, for example, analog filters in a sigma-delta modulator ADC. A known dither signal is used, for example a digital dither signal. Through adding of the dither to the modulator loop, the digital output of the sigma delta modulator ADC contains a filtered version of the digital dither. This signal can be used to reveal characteristics of the modulator-loop, including characteristics of a continuous-time filter in the modulator. Therefore, using the known digital dither signal and the output signal of the modulator, the continuous-time loop filter can be tuned. The tuning can be done in multiple ways, for example, by using standard LMS adaptive filter techniques to estimate the actual response of the continuous-time loopfilter and adjust the continuous-time loopfilter to the desired response.

The invention concerns a time-continuous delta-sigma analog-digital converter for the conversion of an analog input signal into a digital output signal, comprising an analog filter for the filtering of the analog input signal, a clocked operated quantiser, which contains at least one comparator (34) and which quantises the filtered analog signal outputted through the analog filter for the generation of the digital output signal, and a feedback arrangement with at least one digital-analog converter, which supplies to the analog filter at least one feedback signal on the basis of the digital output signal. According to the invention a calibration device (32) linked with the quantiser is stipulated, which is designed to determine at a predetermined point in time an offset error of the comparator (34) and subsequently to compensate for this (Itrim).

The invention relates to an electronic circuit for solving problems of phase linearity involved with transmitting a signal through a digital-to-analog converter module. By connecting the following in cascade: an interpolator; a sigma-delta modulator; a digital-to-analog converter block made up of a 1-bit digital-to-analog converter and a multiband finite impulse response filter; and an analog filter, it is possible to filter out quantization noise while providing a linear response at the output from the circuit.

A digital tuning system (250) for changing a cutoff frequency of an analog filter (132) includes digital synthesizers (292 and 294) for producing a two-tone calibration signal (196) applied to an input of the filter after a quality factor of the filter is increased. The filter includes at least one R/C circuit with two resistors (304 and 306) for changing the quality factor and arrays (308 and 310) of capacitors for changing the cutoff frequency. The amplitude of the magnitude responses (409 and 411) of the filter to each tone (405 and 407) is measured by a two discrete Fourier transform single-frequency bin power detection circuits (253 and 254) while the filter is sequenced through a plurality of capacitance settings. An optimal capacitance for the R/C circuit is selected by comparing, to a pre-selected value, a difference between the responses of the filter to each tone, for each capacitance setting.