Showing papers on "Sampling (signal processing) published in 2015"

Heart rate data collection

Abstract: One innovative aspect is directed to heart rate data collection. In some implementations, a circuit includes a light detector for generating a first electrical signal based on received light. The circuit includes a switching circuit, having a first and a second configuration, configured to receive a first voltage signal based on the first electrical signal and to switch among the first and the second configurations. The circuit includes first and second sampling circuits for sampling a value of the first voltage signal when the switching circuit is in the first configuration and second configurations, respectively. The circuit includes an ambient light cancellation circuit for generating a current signal to counter a first component of the first electrical signal when the first switching circuit is in the first configuration.

Ultra-sensitive graphene strain sensor for sound signal acquisition and recognition

Abstract: A wearable and high-precision sensor for sound signal acquisition and recognition was fabricated from thin films of specially designed graphene woven fabrics (GWFs). Upon being stretched, a high density of random cracks appears in the network, which decreases the current pathways, thereby increasing the resistance. Therefore, the film could act as a strain sensor on the human throat in order to measure one’s speech through muscle movement, regardless of whether or not a sound is produced. The ultra-high sensitivity allows for the realization of rapid and low-frequency speech sampling by extracting the signature characteristics of sound waves. In this study, representative signals of 26 English letters, typical Chinese characters and tones, and even phrases and sentences were tested, revealing obvious and characteristic changes in resistance. Furthermore, resistance changes of the graphene sensor responded perfectly with pre-recorded sounds. By combining artificial intelligence with digital signal processing, we expect that, in the future, this graphene sensor will be able to successfully negotiate complex acoustic systems and large quantities of audio data.

Heart monitors and processes with accelerometer motion artifact cancellation, and other electronic systems

Abstract: A heart monitor includes a single chest accelerometer (210), an analog signal conditioning and sampling section (215) responsive to said accelerometer to produce a digital signal substantially representing acceleration, and a digital processor (220) operable to filter the acceleration signal into a signal affected by body motion and to cancel the body motion signal from the acceleration signal, thereby to produce an acceleration-based cardiac-related signal Other processes and electronic systems are also disclosed

Complementary compressive imaging for the telescopic system

Abstract: Conventional single-pixel cameras recover images only from the data recorded in one arm of the digital micromirror device, with the light reflected to the other direction not to be collected. Actually, the sampling in these two reflection orientations is correlated with each other, in view of which we propose a sampling concept of complementary compressive imaging, for the first time to our knowledge. We use this method in a telescopic system and acquire images of a target at about 2.0 km range with 20 cm resolution, with the variance of the noise decreasing by half. The influence of the sampling rate and the integration time of photomultiplier tubes on the image quality is also investigated experimentally. It is evident that this technique has advantages of large field of view over a long distance, high-resolution, high imaging speed, high-quality imaging capabilities, and needs fewer measurements in total than any single-arm sampling, thus can be used to improve the performance of all compressive imaging schemes and opens up possibilities for new applications in the remote-sensing area.

A Decision-Error-Tolerant 45 nm CMOS 7b 1 GS/s Nonbinary 2b/Cycle SAR ADC

Abstract: A compact decision-error-tolerant 2b/cycle SAR ADC architecture is presented. Two DACs with different designated functions, SIG-DAC and REF-DAC, are implemented to make the structure compact and to eliminate the sampling skew issue. Use of a nonbinary decision scheme with decision redundancies not only increases the ADC speed with a relaxed DAC settling requirement but also makes the performance robust to reference fluctuations and comparator offset variations. The proposed dynamic register and direct DAC control scheme enhance the conversion speed by minimizing logic delay in the SAR decision loop. The proposed comparator-error detection with digital error correction scheme enhances high-speed ADC performance. A prototype 7b ADC fabricated in a 45 nm CMOS process operates at a sampling rate of 1 GS/s under a 1.25 V supply while achieving a peak SNDR of 41.6 dB and maintaining an ENOB higher than 6 up to 1.3 GHz signal frequency. The FoM under a 1.25 V supply is an 80 fJ/conversion-step with a power consumption of 7.2 mW.

Is 50 Hz high enough ECG sampling frequency for accurate HRV analysis

Abstract: With the worldwide growth of mobile wireless technologies, healthcare services can be provided at anytime and anywhere. Usage of wearable wireless physiological monitoring system has been extensively increasing during the last decade. These mobile devices can continuously measure e.g. the heart activity and wirelessly transfer the data to the mobile phone of the patient. One of the significant restrictions for these devices is usage of energy, which leads to requiring low sampling rate. This article is presented in order to investigate the lowest adequate sampling frequency of ECG signal, for achieving accurate enough time domain heart rate variability (HRV) parameters. For this purpose the ECG signals originally measured with high 5 kHz sampling rate were down-sampled to simulate the measurement with lower sampling rate. Down-sampling loses information, decreases temporal accuracy, which was then restored by interpolating the signals to their original sampling rates. The HRV parameters obtained from the ECG signals with lower sampling rates were compared. The results represent that even when the sampling rate of ECG signal is equal to 50 Hz, the HRV parameters are almost accurate with a reasonable error.

High-resolution dynamic speech imaging with joint low-rank and sparsity constraints.

Abstract: Purpose To enable dynamic speech imaging with high spatiotemporal resolution and full-vocal-tract spatial coverage, leveraging recent advances in sparse sampling. Methods An imaging method is developed to enable high-speed dynamic speech imaging exploiting low-rank and sparsity of the dynamic images of articulatory motion during speech. The proposed method includes: (a) a novel data acquisition strategy that collects spiral navigators with high temporal frame rate and (b) an image reconstruction method that derives temporal subspaces from navigators and reconstructs high-resolution images from sparsely sampled data with joint low-rank and sparsity constraints. Results The proposed method has been systematically evaluated and validated through several dynamic speech experiments. A nominal imaging speed of 102 frames per second (fps) was achieved for a single-slice imaging protocol with a spatial resolution of 2.2 × 2.2 × 6.5 mm3. An eight-slice imaging protocol covering the entire vocal tract achieved a nominal imaging speed of 12.8 fps with the identical spatial resolution. The effectiveness of the proposed method and its practical utility was also demonstrated in a phonetic investigation. Conclusion High spatiotemporal resolution with full-vocal-tract spatial coverage can be achieved for dynamic speech imaging experiments with low-rank and sparsity constraints. Magn Reson Med 73:1820–1832, 2015. © 2014 Wiley Periodicals, Inc.

Tuning of Synchronous-Frame PI Current Controllers in Grid-Connected Converters Operating at a Low Sampling Rate by MIMO Root Locus

Abstract: Current controller performance is key in grid-connected power converters for renewable energy applications. In this context, a challenging scenario is arising in multi-megawatt wind turbines, where sampling and switching frequencies tend to be lower and lower as power ratings increase. This strongly affects achievable control time constant. With this perspective, this paper presents a systematic procedure for accurate dynamics assessment and tuning of synchronous-frame proportional–integral current controllers, which is based on linear control for multiple-input–multiple-output (MIMO) systems. The dominant eigenvalues of the system are calculated with explicit consideration of time-delay and cross-coupling terms, two factors which clearly impair the system dynamics when considering a low sampling frequency. The proposed methodology is summarized as follows. First, the plant and controller matrices are modeled in state space. Subsequently, the characteristic polynomial of the closed-loop system is obtained and a computer-aided parametric analysis is performed to calculate the MIMO root locus as a function of the control gain. By its inspection, it is possible to identify the gain, which minimizes the current closed-loop time constant. This tuning is suitable for wind turbine applications, taking into consideration cascaded-control structures and grid-code requirements. The validity and accuracy of the analysis is fully supported by experimental verification.

Digital Signal Processing and Spectral Analysis for Scientists: Concepts and Applications

Abstract: Introduction.- Discrete-time Signals and Systems.- Transforms of Discrete-time Signals.- Sampling of Continuous-time Signals.- Spectral Analysis of Deterministic Discrete-Time Signals.- Digital Filter Properties and Filtering Implementation.- FIR Filters Design.- IIR Filters Design.- Statistical Approach to Signal Analysis.- Non-Parametric Spectral Methods.- Parametric Spectral Methods.- Singular Spectrum Analysis (SSA).- Non-Stationary Spectral Analysis.- Discrete Wavelet Transform (DWT).- De-noising and Compression byWavelets.- Exercises with Matlab.

A 14 b 35 MS/s SAR ADC Achieving 75 dB SNDR and 99 dB SFDR With Loop-Embedded Input Buffer in 40 nm CMOS

Abstract: This paper presents a 14 bit 35 MS/s successive approximation register (SAR) ADC that achieves a nearly constant 74.5 dB peak SNDR up to Nyquist and an SFDR of 90/99 dB for inputs near Nyquist and at low-frequencies, respectively. The ADC employs a loop-embedded input buffer that shields the large sampling capacitor from the input and thereby eases the ADC drive requirements significantly. Since the buffer's nonlinearity is cancelled by the SAR operation, a pair of basic source followers can be used, adding only 12.5 mW (23% of the total power) to the power budget. The ADC includes a bandgap reference and a self-calibrated current steering DAC to close the SAR loop, which eliminates the need for a low-impedance off-chip reference. The design occupies 0.236 mm $^{2}$ in 40 nm CMOS and consumes a total power of 54.5 mW from its 1.2/2.5 V supplies, leading to an SNDR-based Schreier FOM of 159.5 dB at Nyquist.

Compressive sensing of signals generated in plastic scintillators in a novel J-PET instrument

Abstract: The J-PET scanner, which allows for single bed imaging of the whole human body, is currently under development at the Jagiellonian University. The discussed detector offers improvement of the Time of Flight (TOF) resolution due to the use of fast plastic scintillators and dedicated electronics allowing for sampling in the voltage domain of signals with durations of few nanoseconds. In this paper we show that recovery of the whole signal, based on only a few samples, is possible. In order to do that, we incorporate the training signals into the Tikhonov regularization framework and we perform the Principal Component Analysis decomposition, which is well known for its compaction properties. The method yields a simple closed form analytical solution that does not require iterative processing. Moreover, from the Bayes theory the properties of regularized solution, especially its covariance matrix, may be easily derived. This is the key to introduce and prove the formula for calculations of the signal recovery error. In this paper we show that an average recovery error is approximately inversely proportional to the number of acquired samples.

Wideband Rapid Interferer Detector Exploiting Compressed Sampling With a Quadrature Analog-to-Information Converter

Abstract: We introduce a rapid interferer detector that uses compressed sampling (CS) with a quadrature analog-to-information converter (QAIC). By exploiting bandpass CS, a blind sub-Nyquist sampling approach, the QAIC offers an energy efficient and rapid interferer detection over a wide instantaneous bandwidth. The QAIC front end is implemented in 65 nm CMOS in 0.43 mm 2 and consumes 81 mW from a 1.1 V supply. It senses a frequency span of 1 GHz ranging from 2.7 to 3.7 GHz (PCAST Band) with a resolution bandwidth of 20 MHz in 4.4 µs, 50 times faster than traditional sweeping spectrum scanners. Rapid interferer detector with the bandpass QAIC is two orders of magnitude more energy efficient than traditional Nyquist-rate architectures and one order of magnitude more energy efficient than existing low-pass CS methods. Thanks to CS, the aggregate sampling rate of the QAIC interferer detector is compressed by 6.3 $\times$ compared to traditional Nyquist-rate architectures for the same instantaneous bandwidth.

Integrated All Electrical Pulse Wave Velocity and Respiration Sensors Using Bio-Impedance

Abstract: In this paper, we propose wire-free integrated sensors that monitor pulse wave velocity (PWV) and respiration, both non-electrical vital signs, by using an all-electrical method. The key techniques that we employ to obtain all-electrical and wire-free measurement are bio-impedance (BI) and analog-modulated body-channel communication (BCC), respectively. For PWV, time difference between ECG signal from the heart and BI signal from the wrist is measured. To remove wires and avoid sampling rate mismatch between ECG and BI sensors, ECG signal is sent to the BI sensor via analog BCC without any sampling. For respiration measurement, BI sensor is located at the abdomen to detect volume change during inhalation and exhalation. A prototype chip fabricated in 0.11 μm CMOS process consists of ECG, BI sensor and BCC transceiver. Measurement results show that heart rate and PWV are both within their normal physiological range. The chip consumes 1.28 mW at 1.2 V supply while occupying 5 mm×2.5 mm of area.

Acceleration of high angular and spatial resolution diffusion imaging using compressed sensing with multichannel spiral data.

Abstract: Purpose To accelerate the acquisition of simultaneously high spatial and angular resolution diffusion imaging. Methods Accelerated imaging is achieved by recovering the diffusion signal at all voxels simultaneously from under-sampled k-q space data using a compressed sensing algorithm. The diffusion signal at each voxel is modeled as a sparse complex Gaussian mixture model. The joint recovery scheme enables incoherent under-sampling of the 5-D k-q space, obtained by randomly skipping interleaves of a multishot variable density spiral trajectory. This sampling and reconstruction strategy is observed to provide considerably improved reconstructions than classical k-q under-sampling and reconstruction schemes. The complex model enables to account for the noise statistics without compromising the computational efficiency and theoretical convergence guarantees. The reconstruction framework also incorporates compensation of motion induced phase errors that result from the multishot acquisition. Results Reconstructions of the diffusion signal from under-sampled data using the proposed method yields accurate results with errors less that 5% for different accelerations and b-values. The proposed method is also shown to perform better than standard k-q acceleration schemes. Conclusions The proposed scheme can significantly accelerate the acquisition of high spatial and angular resolution diffusion imaging by accurately reconstructing crossing fiber architectures from under-sampled data. Magn Reson Med 73:126–138, 2015. © 2014 Wiley Periodicals, Inc.

A non-uniformly under-sampled blade tip-timing signal reconstruction method for blade vibration monitoring.

Abstract: High-speed blades are often prone to fatigue due to severe blade vibrations. In particular, synchronous vibrations can cause irreversible damages to the blade. Blade tip-timing methods (BTT) have become a promising way to monitor blade vibrations. However, synchronous vibrations are unsuitably monitored by uniform BTT sampling. Therefore, non-equally mounted probes have been used, which will result in the non-uniformity of the sampling signal. Since under-sampling is an intrinsic drawback of BTT methods, how to analyze non-uniformly under-sampled BTT signals is a big challenge. In this paper, a novel reconstruction method for non-uniformly under-sampled BTT data is presented. The method is based on the periodically non-uniform sampling theorem. Firstly, a mathematical model of a non-uniform BTT sampling process is built. It can be treated as the sum of certain uniform sample streams. For each stream, an interpolating function is required to prevent aliasing in the reconstructed signal. Secondly, simultaneous equations of all interpolating functions in each sub-band are built and corresponding solutions are ultimately derived to remove unwanted replicas of the original signal caused by the sampling, which may overlay the original signal. In the end, numerical simulations and experiments are carried out to validate the feasibility of the proposed method. The results demonstrate the accuracy of the reconstructed signal depends on the sampling frequency, the blade vibration frequency, the blade vibration bandwidth, the probe static offset and the number of samples. In practice, both types of blade vibration signals can be particularly reconstructed by non-uniform BTT data acquired from only two probes.

High Speed All Optical Nyquist Signal Generation and Full-band Coherent Detection

Abstract: Spectrum efficient data transmission is of key interest for high capacity optical communication systems considering the limited available bandwidth. Transmission of the high speed signal with higher-order modulation formats within the Nyquist bandwidth using coherent detection brings attractive performance advantages. However, high speed Nyquist signal generation with high order modulation formats is challenging. Electrical Nyquist pulse generation is restricted by the limited sampling rate and processor capacities of digital-to-analog convertor devices, while the optical Nyquist signals can provide a much higher symbol rate using time domain multiplexing method. However, most optical Nyquist signals are based on direct detection with simple modulation formats. Here we report the first experimental demonstration of high speed all optical Nyquist signal generation based on Sinc-shaped pulse generation and time-division multiplexing with high level modulation format and full-band coherent detection. Our experiments demonstrate a highly flexible and compatible all optical high speed Nyquist signal generation and detection scheme for future fiber communication systems.

An efficient phase analysis-based wavenumber linearization scheme for swept source optical coherence tomography systems

Abstract: This letter reports an efficient phase analysis-based direct time domain resampling scheme for swept source-based optical coherence tomography (SS-OCT) systems. The unwrapped phase values, representing non-linear frequency sweeping of the laser, are extracted from the calibration signal generated by a Mach?Zehnder interferometer. Equidistant wavenumber spaces are calculated by normalizing obtained phase values followed by scaling to the maximum number of sampling points. The non-uniform fractional time index values corresponding to the uniformly distributed phase values are computed directly from the linearizer coordinates in order to eliminate the use of the polynomial fitting approach that is used in existing phase-based time domain resampling methods. This proposed linearization scheme shows a significant improvement in performance in terms of accuracy and speed in comparison with the major existing schemes. The robustness of the algorithm, as well as its impact on the resolution and sensitivity, are illustrated using an in-house-developed SS-OCT system and by performing imaging of a human finger nail and an eye model as test samples.

Adaptive tool servo diamond turning for enhancing machining efficiency and surface quality of freeform optics.

Abstract: Fast tool servo/ slow tool servo (FTS/STS) diamond turning is a very promising technique for the generation of freeform optics. However, the currently adopted constant scheme for azimuth sampling and side-feeding motion possesses no adaptation to surface shape variation, leading to the non-uniform surface quality and low machining efficiency. To overcome this defect, this paper reports on a novel adaptive tool servo (ATS) diamond turning technique which is essentially based on the novel two-degree-of-freedom (2-DOF) FTS/STS. In the ATS, the sampling interval and the side-feeding motion are actively controlled at any cutting point to adapt the machining process to shape variation of the desired surface, making both the sampling induced interpolation error and the side-feeding induced residual tool mark be within the desired tolerances. Characteristic of the required cutting motion suggests that besides the conventional z-axis servo motion, another servo motion along the x-axis synthesizing by the c-axis is mandatory for implementing the ATS. Comparative studies of surface generation of typical micro-structured surfaces in FTS/STS and ATS are thoroughly conducted both theoretically and experimentally. The result demonstrates that the ATS outperforms the FTS/STS with improved surface quality while simultaneously enhanced machining efficiency.

Power signal frequency detection method and system based on amplitude modulation

Abstract: The invention discloses a power signal frequency detection method and a power signal frequency detection system based on phase modulation. The method comprises the steps: sampling a power signal according to preset signal time span and preset sampling frequency, so as to obtain an input signal sequence; measuring the frequency of the input signal sequence, so as to obtain the initial frequency of the power signal, and subtracting the input signal sequence and the plus or minus 1 Pi phase-shift sequence of the input signal sequence by using the initial frequency as reference frequency, so as to obtain two phase modulation sequences of which phases change along with the input signal frequency; respectively mixing, filtering and integrating the two phase modulation sequences, so as to obtain the phases of the two phase modulation phases for frequency measurement. By implementing the power signal frequency detection method and the power signal frequency detection system, frequency measurement results with higher accuracy can be obtained.

14.9 Sub-sampling all-digital fractional-N frequency synthesizer with −111dBc/Hz in-band phase noise and an FOM of −242dB

Abstract: The noise performance of an all-digital phase-locked loop (ADPLL) is limited by the resolution of the time-to-digital converter (TDC). Most TDC research in the past focused on the arrival time difference between the edges of the divider feedback and the reference signal [1-2]. This results in coarser TDC resolution and worse ADPLL noise performance. This paper presents a fractional-/VADPLL that employs a new time-to-digital conversion technique based on sub-sampling phase detection. It is accomplished by directly sampling the analog voltage signal at the PLL's high frequency node and converting it into a digital code. This achieves a higher time resolution with less power.

Compressive Multiplexing of Correlated Signals

Abstract: We present a general architecture for the acquisition of ensembles of correlated signals. The signals are multiplexed onto a single line by mixing each one against a different code and then adding them together, and the resulting signal is sampled at a high rate. We show that if the M signals, each band limited to W/2 Hz, can be approximated by a superposition of R <; M underlying signals, then the ensemble can be recovered by sampling at a rate within a logarithmic factor of RW, as compared with the cumulative Nyquist rate of MW. This sampling theorem shows that the correlation structure of the signal ensemble can be exploited in the acquisition process even though it is unknown a priori. The reconstruction of the ensemble is recast as a low-rank matrix recovery problem from linear measurements. The architectures we are considering impose a certain type of structure on the linear operators. Although our results depend on the mixing forms being random, this imposed structure results in a very different type of random projection than those analyzed in the low-rank recovery literature to date.

Methods and systems for down-converting a signal using a complementary transistor structure

Abstract: Methods, systems, and apparatuses for down-converting a modulate carrier signal to a demodulated baseband signal by sampling the energy of the carrier signal are described herein. Briefly stated, such methods systems, and apparatuses operate by receiving a modulated carrier signal and using pulses with apertures to control a switch so as to (a) transfer energy from the modulated carrier signal and accumulate the transferred energy in a capacitor when the switch is closed during the apertures of the pulses and (b) discharge some of the previously accumulated energy from the capacitor into load circuitry at least when the switch is open. The demodulated baseband signal is generated from (i) accumulating energy transferred to the capacitor each time the switch is closed during the apertures of the pulses, and (ii) discharging some of the previously accumulated energy into the load circuitry each time the switch is opened.

Angular spectrum-based wave-propagation method with compact space bandwidth for large propagation distances.

Abstract: Rigorous propagation methods enable diffraction calculations at high NA. However, for the case of large propagation distances and full NA calculations of a signal, common solutions require zero padding or upsampling. This Letter overcomes these problems by introducing a sampling scheme based on compact space bandwidth product representation, which adjusts the sampling frequency of input and propagated field according to the evolution of the generalized space bandwidth product. This sampling concept allows proposing a novel AS method enabling high efficiency, high accuracy, and high-NA diffraction computations at larger propagation distances without need of zero padding or upsampling. The method has several advantages: (1) high accuracy for larger propagation distances; (2) reduced sampling with minimal computation effort; (3) zooming capability; and (4) both focusing and defocusing propagations possible.

Analysis, Blind Identification, and Correction of Frequency Response Mismatch in Two-Channel Time-Interleaved ADCs

Abstract: In this paper, novel blind identification and compensation architectures for the frequency response mismatch of a two-channel time-interleaved analog-to-digital-converter (TI-ADC) are proposed. First, detailed modeling of the frequency response mismatch is carried out, establishing a direct connection and similarity to the well-known in-phase/quadrature mismatch problem. Stemming from this modeling, the proposed blind mismatch identification and compensation architectures are then developed building on complex statistical signal processing. Compared to the existing methods in the literature, the proposed solutions can identify and correct the frequency response mismatch in a fully blind manner for the full digital bandwidth (BW) of the ADC system, and are also applicable in sub-sampling TI-ADC devices and RF sampling. The efficiency of the proposed solutions is verified and demonstrated using comprehensive measurements of actual RF-sampling TI-ADC hardware with gigahertz-scale instantaneous BW.

Methods and apparatus for reducing timing-skew errors in time-interleaved analog-to-digital converters

Abstract: A time-interleaved (TI) analog-to-digital converter (ADC) architecture employs a low resolution coarse ADC channel that samples an input analog signal at a Nyquist rate and facilitates background calibration of timing-skew error without interrupting normal operation to sample/convert the input signal. The coarse ADC channel provides a timing reference for multiple higher resolution TI ADC channels that respectively sample the input signal at a lower sampling rate. The coarse ADC digital output is compared to respective TI ADC digital outputs to variably adjust in time corresponding sampling clocks of the TI ADC channels so as to substantially align them with the sampling clock of the coarse ADC channel, thus reducing timing-skew error. In one example, the coarse ADC output provides the most significant bits (MSBs) of the respective TI ADC digital outputs to further improve conversion speed and reduce power consumption in these channels.

Nonlinear Analysis of Discretization Effects in a Digital Current Mode Controlled Boost Converter

Abstract: Digital current mode control finds wide spread application in point of load power converters in DC nano-grid because of its technical benefits. However, finite current-loop sampling effects introduce undesirable sub-harmonic oscillations. This paper presents an analytical framework to investigate such nonlinear phenomena in a digitally current mode controlled boost converter. Discrete-time models for multi-sampled current loops and uniform sample with compensating ramp are derived under continuous conduction mode. We show that the discrete-time maps for such systems are discontinuous in nature. While the error voltage using a proportional-integral controller stays within the zero-error-bin (ZEB), the reference current becomes constant and 1-D maps of the inner current-loop can be used for stability analysis. Uniform sampling may lead to chaos, period doubling or stable period-1 behavior depending on slope of the compensating ramp. Multi-sampled current loop imposes several borders in the discrete parameter space and may eventually lead to high periodic behavior. In a counter-based compensating ramp, staircase effects may lead to sub-harmonic oscillation. Such instability eventually brings the error voltage outside the ZEB and 2-D map models have to be used for further investigating the nonlinear phenomena. A boost converter prototype was made. Digital current mode control is realized using an FPGA device. Test results demonstrate close agreement with the analysis.

Sinusoidal parameter measurement method and system of power signal

Abstract: The invention discloses a sinusoidal parameter measurement method and system of a power signal. The method comprises the following steps: performing preliminary measurement on fundamental wave frequency of a sampling data sequence to acquire preliminary fundamental wave frequency and multiplying a cosine function and a sine function of the preliminary fundamental wave frequency serving as the reference frequency by the sampling data sequence respectively to generate a real vector sequence and an imaginary vector sequence; performing digital filtering on the real vector sequence and the imaginary vector sequence to generate a real vector filter sequence and an imaginary vector filter sequence, and further integrating to generate a real vector integral value and an imaginary vector integral value; converting the real vector integral value and the imaginary vector integral value into corresponding sinusoidal parameters according to a preset sinusoidal parameter conversion rule. By implementing the method and the system, mixed frequency interference elements in the real vector sequence and the imaginary vector sequence can be inhibited to generate high-precision real vector and imaginary vector sequence integral values so as to finally obtain the sinusoidal parameters with higher precision.

System and method for determining audio context in augmented-reality applications

Abstract: An augmented-reality audio system generates information regarding the acoustic environment by sampling audio signals. Using a Gaussian mixture model or other technique, the system identifies the location of one or more audio sources, with each source contributing an audio component to the sampled audio signals. The system determines a reverberation time for the acoustic environment using the audio components. In determining the reverberation time, the system may discard audio components from sources that are determined to be in motion, such as components with an angular velocity above a threshold or components having a Doppler shift above a threshold. The system may also discard audio components from sources having an inter-channel coherence above a threshold. In at least one embodiment, the system renders sounds using the reverberation time at virtual locations that are separated from the locations of the audio sources.