2874 results about "Digital filter" patented technology

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.

The invention provides a multi-sensor system that uses an algorithm based on adaptive filtering to monitor a patient's respiratory rate. The system features a first sensor selected from the following group: i) an impedance pneumography sensor featuring at least two electrodes and a processing circuit configured to measure an impedance pneumography signal; ii) an ECG sensor featuring at least two electrodes and an ECG processing circuit configured to measure an ECG signal; and iii) a PPG sensor featuring a light source, photodetector, and PPG processing circuit configured to measure a PPG signal. Each of these sensors measures a time-dependent signal which is sensitive to respiratory rate and, during operation, is processed to determine an initial respiratory rate value. An adaptive digital filter is determined from the initial respiratory rate. The system features a second sensor (e.g. a digital 3-axis accelerometer) that attaches to the patient's torso and measures an ACC signal indicating movement of the chest or abdomen that is also sensitive to respiratory rate. This second signal is processed with the adaptive filter to determine a final value for respiratory rate.

A method for operating a microprocessor controlled, reel type slot machine in which payoff is determined before a final game outcome is displayed to a player. A pre-defined count of random numbers is generated and presented to a digital filter having tap outputs which correspond to paytable payline equations. The minimum number of payline equations is equal to the number of distinct paylines in the paytable plus one. Any changes in game outcome are taken into account by modifying variables in computer memory and as such do not require a change in tables stored in computer memory. Several features for attracting players to the apparatus and increasing the enjoyment of playing a game are included.

A single frequency relay station receives a signal from a master station, and retransmits it. Before retransmitting the signal, the relay station uses a digital filter to subtract from the signal components which arise from multipath interference and coupling interference. The coefficients used by the digital filter are derived from characteristics of the multipath interference and the overall transfer function of the relay station. These are derived by turning off the retransmission, so that the multipath interference can be estimated from the received signal, and commencing the retransmission again, to determine the transfer function. A simplified inverse fast Fourier transform is used to simplify the calculations.

A computer readable memory to direct a computer to improve the perceived audio quality of a speaker included in that computer. The computer readable memory stores a first, second and third set of instructions. The first set of instructions causes the computer to determine the speaker type. The second set of instructions causes the computer to select a set of default filter coefficients for a digital filter based upon the speaker type. Finally, the third set of instructions causes the computer to realize a digital parametric equalizer using a digital filter and the set of default filter coefficients. Thus, the digital filter alters the audio signal that is input to the speaker, thereby improving the perceived quality of the speaker.

The invention provides a multi-sensor system that uses an algorithm based on adaptive filtering to monitor a patient's respiratory rate. The system features a first sensor selected from the following group: i) an impedance pneumography sensor featuring at least two electrodes and a processing circuit configured to measure an impedance pneumography signal; ii) an ECG sensor featuring at least two electrodes and an ECG processing circuit configured to measure an ECG signal; and iii) a PPG sensor featuring a light source, photodetector, and PPG processing circuit configured to measure a PPG signal. Each of these sensors measures a time-dependent signal which is sensitive to respiratory rate and, during operation, is processed to determine an initial respiratory rate value. An adaptive digital filter is determined from the initial respiratory rate. The system features a second sensor (e.g. a digital 3-axis accelerometer) that attaches to the patient's torso and measures an ACC signal indicating movement of the chest or abdomen that is also sensitive to respiratory rate. This second signal is processed with the adaptive filter to determine a final value for respiratory rate.

The present invention is related to methods and apparatus that compensate for quadrature impairments of an analog quadrature modulator and/or demodulator over a relatively wide signal bandwidth. One embodiment pre-distorts baseband signals in a quadrature modulator compensation signal processor (QMCSP) to negate the quadrature impairment of an analog quadrature modulator and corrects a received baseband signal in a quadrature demodulator compensation signal processor (QDCSP) to cancel the quadrature impairment of an analog quadrature demodulator. The QMCSP and the QDCSP contain adaptive digital filter correction structures that pre-compensate and post-compensate, respectively, for the quadrature impairments introduced by the analog quadrature modulator and the analog quadrature demodulator over a relatively wide bandwidth. A phase shifter advantageously shifts the phase of a local oscillator signal to the analog quadrature demodulator to distinguish quadrature impairments introduced by the modulation path from quadrature impairments introduced by the demodulation path.

A self calibrating dipole microphone formed from two omni-directional acoustic sensors. The microphone includes a sound source acoustically coupled to the acoustic sensors and a processor. The sound source is excited with a test signal, exposing the acoustic sensors to acoustic calibration signals. The responses of the acoustic sensors to the calibration signals are compared by the processor, and one or more correction factors determined. Digital filter coefficients are calculated based on the one or more correction factors, and applied to the output signals of the acoustic sensors to compensate for differences in the sensitivities of the acoustic sensors. The filtered signals provide acoustic sensor outputs having matching responses, which are subtractively combined to form the dipole microphone output.

Embodiments disclosed herein enable detection and improvement of the quality of the audio signal using a mobile device by determining the loss in the audio signal and enhancing audio by streaming the remainder portion of audio. Embodiments disclosed herein enable an improvement in the sound quality rendered by rendering devices by emitting an test audio signal from the source device, measuring the test audio signal using microphones, detecting variation in the frequency response, loudness and timing characteristics using impulse responses and correcting for them. Embodiments disclosed herein also compensate for the noise in the acoustic space by determining the reverberation and ambient noise levels and their frequency characteristics and changing the digital filters and volumes of the source signal to compensate for the varying noise levels.

A desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with processor (30). Processor (30) receives aural signals from two sensors (22, 24) each at a different location. The two inputs to processor (30) are converted from analog to digital format and then submitted to a discrete Fourier transform process to generate discrete spectral signal representations. The spectral signals are delayed to provide a number of intermediate signals, each corresponding to a different spatial location relative to the two sensors. Locations of the noise source and the desired source, and the spectral content of the desired signal are determined from the intermediate signal corresponding to the noise source locations. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal with output device (90). Techniques to localize multiple acoustic sources are also disclosed. Further, a technique to enhance noise reduction from multiple sources based on two-sensor reception is described.

An equalizer for equalizing channel multi-path distortion includes digital filters. To improve the convergence speed and tracking ability of the equalizer while lowering noise and power consumption, the digital filters are divided into sections. Various parameters of the sections, such as step-size, shutdown and update rates can be controlled. Control of the various parameters can be realized either in software on an embedded or external processor or by dedicated hardware.

Disclosed herein are methods and systems for encoding digital watermarks into content signals. Also disclosed are systems and methods for detecting and/or verifying digital watermarks in content signals.

According to one embodiment, a system for encoding of digital watermark information includes: a window identifier for identifying a sample window in the signal; an interval calculator for determining a quantization interval of the sample window; and a sampler for normalizing the sample window to provide normalized samples.

According to another embodiment, a system for pre-analyzing a digital signal for encoding at least one digital watermark using a digital filter is disclosed.

According to another embodiment, a method for pre-analyzing a digital signal for encoding digital watermarks comprises: (1) providing a digital signal; (2) providing a digital filter to be applied to the digital signal; and (3) identifying an area of the digital signal that will be affected by the digital filter based on at least one measurable difference between the digital signal and a counterpart of the digital signal selected from the group consisting of the digital signal as transmitted, the digital signal as stored in a medium, and the digital signal as played backed.

According to another embodiment, a method for encoding a watermark in a content signal includes the steps of (1) splitting a watermark bit stream; and (2) encoding at least half of the watermark bit stream in the content signal using inverted instances of the watermark bit stream.

Other methods and systems for encoding/decoding digital watermarks are also disclosed.

An analog-to-digital converter system [50D] processing an input signal, g, which can be either a discrete-time or a continuous-time signal. A first quantizer [154] generates a first digital signal, d0(k), representing the sum of the input signal, g, and a dithering signal, y0. A digital-to-analog converter [156] generates an analog feedback signal, alpha, representing accurately the first digital signal, d0(k). The DAC [156] may be linearized by the use of mismatch-shaping techniques. A filter [158] generates the dithering signal, y0, by selectively amplifying in the signal band the residue signal, r0, defined as the difference of the input signal, g, and the analog feedback signal, alpha. Optional signal paths [166][168] are used to minimize the closed-loop signal transfer function from g to y0, which ideally will be zero. An analog compensation signal, m0, which is described by a well-controlled relationship to the residue signal, r0, is extracted from the filter [158]. Ideally, the closed-loop signal transfer function from g to m0 will be zero, or at least small in the signal band. A second quantizer [160] converts the analog compensation signal, m0, into a second digital signal, dm0(k). The two digital signals, d0(k) and dm0(k), are filtered individually and then added to form the overall output signal, dg(k). The second digital filter [164] has a low signal-band gain, which implies that the sensitivity to signal-band errors caused by the second quantizer [160] will be low. The output signal, dg(k), is a highly-accurate high-resolution representation of the input signal, g. Circuit imperfections, such as mismatch, gain errors, and nonlinearities, will cause only noise-like errors having a very low spectral power density in the signal band.The invention facilitates the implementation of uncalibrated highly-linear high-resolution wide-bandwidth A/D converters [50D], e.g., for use in digital communication systems, such as xDSL modems and other demanding consumer-market products for which low cost is of the essence.

An apparatus for repeating signals includes a receive antenna for receiving input signals, processing circuitry for processing the input signals to form repeated signals, and a transmit antenna for transmitting the repeated signals. The processing circuitry includes an adaptive digital filter configured to generate cancellation signals that are added to the input signals to cancel unwanted feedback signals from the input signals. A frequency shifting circuit adds a frequency shift to the input signals, after the addition of the cancellation signals, to form repeated signals that are frequency shifted from the input signals. A digital signal processor is coupled to the adaptive digital filter for digitally adapting the filter. The digital signal processor utilizes the frequency shift of the transmission signals to adapt the adaptive digital filter.

A reference voltage register stores a reference value Vref indicating the output voltage Vout of a power supply circuit. A digital filter amplifies the difference between the output voltage Vout from the power supply circuit and the reference value Vref stored in the reference voltage register. A pulse width computation unit computes the duty such that the output voltage Vout matches the reference value Vref, and generates the on-time of a pulse signal corresponding to the duty. A reference value operation unit updates the reference voltage register depending on the output current from the power supply circuit or the temperature around the power supply circuit.

The invention provides a system for measuring respiratory rate (RR) from a patient. The system includes an impedance pneumography (IP) sensor, connected to at least two electrodes, and a processing system that receives and processes signals from the electrodes to measure an IP signal. A motion sensor (e.g. an accelerometer) measures at least one motion signal (e.g. an ACC waveform) describing movement of a portion of the patient's body to which it is attached. The processing system receives the IP and motion signals, and processes them to determine, respectfully, frequency-domain IP and motion spectra. Both spectra are then collectively processed to remove motion components from the IP spectrum and determine RR. For example, during the processing, an algorithm determines motion frequency components from the frequency-domain motion spectrum, and then using a digital filter removes these, or parameters calculated therefrom, from the IP spectrum.

The invention provides a system for measuring respiratory rate (RR) from a patient. The system includes an impedance pneumography (IP) sensor, connected to at least two electrodes, and a processing system that receives and processes signals from the electrodes to measure an IP signal. A motion sensor (e.g. an accelerometer) measures at least one motion signal (e.g. an ACC waveform) describing movement of a portion of the patient's body to which it is attached. The processing system receives the IP and motion signals, and processes them to determine, respectfully, frequency-domain IP and motion spectra. Both spectra are then collectively processed to remove motion components from the IP spectrum and determine RR. For example, during the processing, an algorithm determines motion frequency components from the frequency-domain motion spectrum, and then using a digital filter removes these, or parameters calculated therefrom, from the IP spectrum.

The invention discloses a harmonic amplitude measurement method and system of a power signal. The method comprises the following steps: performing digital filtering on an imaginary vector sequence and a real vector sequence to generate an imaginary vector filter sequence and a real vector filter sequence; equally dividing the real vector filter sequence and the imaginary vector filter sequence into two sections of sequences to generate a real vector filter front-section sequence, a real vector filter rear-section sequence, an imaginary vector filter front-section sequence and an imaginary vector filter rear-section sequence respectively; integrating the real vector filter front-section sequence, the imaginary vector filter front-section sequence, the real vector filter rear-section sequence and the imaginary vector filter rear-section sequence in sequence to solve phase and phase difference; converting the phase difference and reference frequency into fundamental wave frequency of the power signal. The product of the fundamental wave frequency and a harmonic factor is used as the reference frequency to perform harmonic wave amplitude measurement. By implementing the method and the system, mixed frequency interference elements in the imaginary vector sequences and the real vector sequences can be inhibited to generate the harmonic wave amplitude with high precision.