Disclosed is a programmable impedance control circuit, comprising a voltage divider, the voltage divider comprising an MOS array supplied with a first voltage and an external resistance having an external impedance equal to N times said external resistance. The voltage divider outputs a second voltage. A reference voltage generator is provided for generating a third voltage corresponding to N / (N+M) times said first voltage as a reference voltage for said second voltage, and wherein M times internal impedance is used for N times external impedance (N=M or N<> M).
Disclosed is a programmable impedance control circuit, comprising a voltage divider, the voltage divider comprising an MOS array supplied with a first voltage and an external resistance having an external impedance equal to N times said external resistance. The voltage divider outputs a second voltage. A reference voltage generator is provided for generating a third voltage corresponding to N / (N+M) times said first voltage as a reference voltage for said second voltage, and wherein M times internal impedance is used for N times external impedance (N=M or N<> M).
A Method for filter tuning using direct digital sub-sampling is provided. The tuning is accomplished in the digital domain by determining the filter characteristics from the shape of the transfer function. The input signal (1) is passed through the filter (3) and is then sub-sampled by and Analog-to-digital Converter (ADC) (5). The sub-sampled signal (6) is then processed in the digital domain using a digital circuit (7) that is used to determine the center frequency (Fc) and Quality factor (Q) and / or other important filter parameters. The Fc, Q and / or other important filter parameters are then adjusted by generating digital control signals (8) that can be converted to analog signals (10) using Digital-to-analog Converters (DACs) (9).
Aspects of a method and system for dynamic filtering and data conversion adjustments in a receiver are provided. Exemplary aspects of the invention may include a receiver comprising a data converter and one or more filters, and a resolution of the data converter and / or a frequency response of the filters may be varied / configured (e.g., via one or more switching elements) based on one or more characteristics of a received signal. The received signals may be amplified and / or filtered prior to determining the characteristics. The resolution of the data converter and / or a quality factor of one or more of the filters may be reduced when measured interference is below a threshold and increased when measured interference is above a threshold. The resolution of the data converter and / or the frequency response of the filter may be determined based on an error rate associated with the received signals.
The invention discloses an active noise controlsystem based on a variable step LMS algorithm. According to the system, a filter based on the variable step LMS algorithm is applied to identification of a secondary channel; a filter weight value is updated by adjusting a step value of the filter; in order to obtain a relatively fast convergence rate, the step value of self-adaptive filtering in an initial phase is relatively high; and when the self-adaptive filtering approaches a steady state, the step size is reduced, so a relatively low steady state error can be obtained. The variable step LMS algorithm is applied to the identification of the secondary channel of the active noise controlsystem, and the contradiction among the convergence rate, the steady state error and a tracking capability of a time-varying system can be solved well.
An adaptive equalizerfinite impulse response (FIR) filter for high-speed communication channels with modest complexity, where the filter is iteratively updated during a training sequence by a circuit performing the update: {overscore (h)}(t+1)={overscore (h)}(t)+μ[sgn{d(t)}−sgn{z(t)−Kd(t)}]sgn{{overscore (x)}(t)}, where {overscore (h)}(t) is the filter vector representing the filter taps of the FIR filter, {overscore (x)}(t) is the data vector representing present and past samples of the received data x(t), d(t) is the desired data used for training, z(t) is the output of the FIR filter, μ determines the memory or window size of the adaptation, and K is a scale factor taking into account practical limitations of the communication channel, receiver, and equalizer. Furthermore, a procedure and circuit structure is provided for calibrating the scale factor K.
One preferred embodiment of the present invention provides systems and methods for removing noise from an input analog signal in continues time. Briefly described in architecture, one embodiment of the system, among others, can be implemented as follows. An analog filtering system including VLSI circuitry separates an analog input signal into a plurality of sub-band signals. Then, an analog gainsystem including VLSI circuitry calculates a gain for each sub-band signal that suppresses the noise within the sub-band signal. In some preferred embodiments, VLSI circuitry includes floating gate technology. Methods and other systems are also provided.
The invention discloses an instant messagingsystem comprising a signal frequency collection circuit, a comparison compensation circuit and a filtering and voltage stabilization output circuit, wherein the signal frequency collection circuit collects signal frequency in a signal transmission channel in the instant messagingsystem, the comparison compensation circuit compares and adjusts the signal frequency, the signal frequency is proportionally amplified by an operational amplifier AR3 and then input into the filtering and voltage stabilization output circuit, meanwhile a compensator AR4 isdesigned for compensating the level of the operational amplifier AR3, the filtering and voltage stabilization output circuit outputs a signal after performing filtering and voltage stabilization on the signal, that is to say, the signal flows into the signal transmission channel of the instant messagingsystem. According to the system provided by the invention, the problem that the signal is instable and the signal transmission efficiency of the instant messaging system due to the fact that the signal frequency in the signal transmission channel in the instant messaging system has a time-varying characteristic and the frequency conversion is relatively fast is effectively solved.
An adaptive equalizerfinite impulse response (FIR) filter for high-speed communication channels with modest complexity, where the filter is iteratively updated during a training sequence by a circuit performing the update: {overscore (h)}(t+1)={overscore (h)}(t)+μ[sgn{d(t)}−sgn{z(t)−Kd(t)}]sgn{{overscore (x)}(t)}, where {overscore (h)}(t) is the filter vector representing the filter taps of the FIR filter, {overscore (x)}(t) is the data vector representing present and past samples of the received data x(t), d(t) is the desired data used for training, z(t) is the output of the FIR filter, μ determines the memory or window size of the adaptation, and K is a scale factor taking into account practical limitations of the communication channel, receiver, and equalizer. Furthermore, a procedure and circuit structure is provided for calibrating the scale factor K.
Adjustable gain circuits (AGCs) within serial filter stages are initialized to maximum gain. The output of each AGC is then sampled and converted to digital representation for use by control logic in setting the gain for the respective AGC. The gain adjustment decision for each AGC is performed in one shot, sequentially backwards from the last AGC, such that gain may be adapted simply and quickly within a number of cycles equal to the number of AGCs. Performance is enhanced by a fast-adapting cell in which capacitances are switched into the input path and feedback loop of an amplifier to reduce direct current gain within the transfer function through charge sharing dividing down the output voltage.
Systems and methods that reduce phase noise are provided. In one embodiment, a method may include one or more of the following: generating a signal at a particular frequency in which the signal may be associated with a harmonic frequency signal disposed at a harmonic frequency; and selectively attenuating frequency content disposed in a region around the harmonic frequency. The signal may be associated with a second harmonic frequency signal disposed at a second harmonic frequency. Frequency content disposed in a second region around the second harmonic frequency may be selectively attenuated. One or more non-linear operations may be applied to the signal and the applied signal may be transmitted.