摘要:
A system includes an analog-to-digital converter (ADC) for converting an analog input signal to a digital signal output and a nonlinearity corrector for correcting nonlinear error in the digital signal output to produce a corrected digital signal output. A source of the nonlinear error is associated with the ADC, wherein an analog calibration signal is introduced to the source of the nonlinear error during conversion of the analog calibration signal to a digital calibration output having the nonlinear error. After conversion of the analog calibration signal to the digital calibration output, a calibration circuit calculates expected values of correlation sums in response to the digital calibration output and determines correction coefficients using the expected values of the correlation sums. The calibration circuit provides correction data based upon the correction coefficients to the nonlinearity corrector.
摘要:
Provided are, among other things, systems, apparatuses, methods and techniques for converting a continuous-time, continuously variable signal into a sampled and quantized signal. One such apparatus includes an input line for accepting an input signal that is continuous in time and continuously variable, multiple processing branches coupled to the input line, and an adder coupled to outputs of the processing branches. Each of the processing branches includes a continuous-time quantization-noise-shaping circuit, a sampling/quantization circuit coupled to an output of the continuous-time quantization-noise-shaping circuit, a digital bandpass filter coupled to an output of the sampling/quantization circuit, and a line coupling an output of the digital-to-analog converter circuit back into the continuous-time quantization-noise-shaping circuit. A center frequency of the digital bandpass filter in each the processing branch corresponds to a minimum in a quantization noise transfer function for the continuous-time quantization-noise-shaping circuit in the same processing branch.
摘要:
Provided is an apparatus for converting a continuous-time, continuously variable signal into a sampled and quantized signal, which includes an input line for accepting an input signal, multiple processing branches coupled to the input line, and an adder coupled to outputs of the plurality of processing branches. Each of the processing branches includes a sampling/quantization circuit and a digital bandpass interpolation filter having an input coupled to an output of the sampling/quantization circuit. The digital bandpass interpolation filters in different ones of the processing branches have frequency responses that are centered at different frequencies. The digital bandpass interpolation filter in at least one of the processing branches includes: (i) a quadrature downconverter, (ii) a first lowpass filter and a second lowpass filter, (iii) a first interpolator and a second interpolator, each having an input for inputting a variable interpolant value, and (iv) a quadrature upconverter.
摘要:
The present invention is directed to a method and a hearing device for extending a usable frequency range of an analog input signal (i) being processed by a hearing device, the method comprising the steps of converting the analog input signal (i) to a first output signal (o1) and to an intermediate signal (om), the first output signal (o1) having a final sampling rate and the intermediate signal (om) having an intermediate sampling rate that is greater than the final sampling rate, applying a band-pass filter unit (31) to the intermediate signal (om) in order to obtain a filtered intermediate signal (omf), a lower cut-off frequency of the band-pass filter unit (31) being above half the final sampling rate, an upper cut-off frequency of the band-pass filter unit (31) being below half the intermediate sampling rate, and shifting a spectrum of the filtered intermediate signal (omf) to a frequency range being below the final sampling rate to obtain an intermediate output signal (om2).
摘要:
Provided are, among other things, systems, methods and techniques for converting a continuous-time, continuously variable signal into a sampled and quantized signal. According to one implementation, an apparatus includes multiple processing branches, each including: a continuous-time quantization-noise-shaping circuit, a sampling/quantization circuit, and a digital bandpass filter. A combining circuit then combines signals at the processing branch outputs into a final output signal. The continuous-time quantization-noise-shaping circuits include adjustable circuit components for changing their quantization-noise frequency-response minimum, and the digital bandpass filters include adjustable parameters for changing their frequency passbands.
摘要:
Provided are, among other things, systems, methods and techniques for converting a continuous-time, continuously variable signal into a sampled and quantized signal. According to one representative embodiment, an apparatus includes multiple continuous-time quantization-noise-shaping circuits, each in a separate processing branch and having an adder that includes multiple inputs and an output; an input signal is coupled to one of the inputs of the adder; the output of the adder is coupled to one of the inputs of the adder through a first filter; and the output of a sampling/quantization circuit in the same processing branch is coupled to one of the inputs of the adder through a second filter, with the second filter having a different transfer function than the first filter.
摘要:
Analog signals can be fully encoded as an asynchronous time sequence generated by a time encoding machine. With knowledge of the parameters of the time encoding machine, the asynchronous time sequence can be decoded using a non-linear time decoding machine. Such a system can be extended into an encoder/decoder in which a signal is processed in M separate channels. An input signal is applied to the encoder where the signal is provided to an M channel encoder circuit including a filter bank having a total bandwidth partitioned among M adjacent or overlapping filters. Each of the M filters are coupled to a corresponding one of M time encoding machines. The encoder output is represented by M sets of time encoded trigger values. The input signal can be recovered from the M sets of time encoded trigger values by applying the trigger signals to a corresponding M channel decoder which includes M TDMs and filters. The TDMs recover the continuous signal from each channel. The filter outputs xm are then amplitude scaled sm and are combined to recover the input signal. By partitioning the signal bandwidth into M channels, the average pulse rate from each TDM is substantially reduced.
摘要:
A direct intermediate frequency sampling analog-to-digital and digital-to-analog converter comprised of (1) a bank of precision analog wavelet generator circuits, (2) synchronization circuits and calibration circuits, and (3) digital measurement circuits. In combination these circuits provide high dynamic range and high sample rate direct intermediate frequency sampling conversion between the discrete-time and continuous-time domains.
摘要:
A method and apparatus are disclosed for converting a signal between the analog and digital domains using frequency interleaving. The disclosed frequency interleaving techniques can be similarly applied to convert analog signals to the digital domain or vice-versa. An analog-to-digital converter decomposes the input broadband signal into N frequency bands that are separately sampled (quantized) before a Fourier transform is applied to convert the signal into the digital domain. Each of the frequency bands can be sampled in the corresponding narrow passband using narrow-band converters, such as passband Sigma-Delta converters, or can be returned to baseband prior to sampling. The various analog samples are then converted to the digital domain using an inverse Fourier transform, or another combining technique. Both sampling and analog-to-digital conversion can be performed at a speed that is N times slower than the input frequency. The disclosed frequency interleaving technique decomposes the input signal into frequency bands that are digitized separately at a slower rate. A disclosed calibration scheme corrects for phase and gain mismatches.
摘要:
A photonically sampled analog-to-digital converter using parallel channels of sampling and quantizing. The parallel combination achieves cancellation of the spurs that result from the nonlinear transfer function of the samplers. The samplers feed a dual-detector optoelectronic receiver that has differential inputs for suppression of laser intensity noise. The outputs of the multiple photonic samplers are averaged to reduce the effects of shot or thermal noise from the optoelectronic receiver of a sampler. The errors produced by the quantization process can be reduced by using a delta-sigma modulator-based analog-to-digital convertor as the quantizer which provides noise-spectrum shaping and filtering.