Actuation of a Frequency Modulated MEMS Gyroscope

2014 ◽  
Author(s):  
Michael Xie ◽  
Sangtak Park ◽  
Eihab Abdel-Rahman ◽  
Mustafa Yavuz
Keyword(s):  
Automatic Gain Control ◽  
Signal To Noise Ratio ◽  
Gain Control ◽  
Function Generator ◽  
Pass Filter ◽  
Low Pass Filter ◽  
Mems Gyroscope ◽  
Actuation System ◽  
Rlc Circuit ◽  
Drive Circuit

This paper describes an analog actuation circuit for a novel frequency-modulated MEMS gyroscope. The circuit provides an amplitude-modulated (AM) signal as the input into a RLC resonant drive circuit, which drives the gyroscope. The actuation system is composed an automatic gain control (AGC) loop, a low pass filter, an amplitude modulation component and a resonant drive circuit. The AM signal is composed of a modulating signal that excite a natural frequency of gyroscope drive mode and a carrier signal with a frequency corresponding to the electrical resonant frequency of the RLC circuit. Both feedforward and feedback AGC configurations are used to stabilize the envelope of the signal. However, the breadboard implementations of the feedforward and feedback circuits in their current configurations have similar signal to noise ratio to that of the function generator. To improve the actuation circuit performance, we plan to include the resonant drive circuit within the AGC feedback loop and implement the actuation circuit on PCB.

scholarly journals NOISE STABILITY OF SIGNAL RECEPTION ALGORITHMS WITH MULTIPULSE PULSE-POSITION MODULATION

2018 ◽  
Vol 42 (1) ◽  
pp. 167-174 ◽  
Author(s):  
V. I. Parfenov ◽  
D. Y. Golovanov
Keyword(s):  
Communication Systems ◽  
Pulse Sequence ◽  
Signal To Noise Ratio ◽  
Point Of View ◽  
Estimation Accuracy ◽  
Pulse Position Modulation ◽  
Signal To Noise ◽  
Pass Filter ◽  
Low Pass Filter ◽  
Noise Ratio

An algorithm for estimating time positions and amplitudes of a periodic pulse sequence from a small number of samples was proposed. The number of these samples was determined only by the number of pulses. The performance of this algorithm was considered on the assumption that the spectrum of the original signal is limited with an ideal low-pass filter or the Nyquist filter, and conditions for the conversion from one filter to the other were determined. The efficiency of the proposed algorithm was investigated through analyzing in which way the dispersion of estimates of time positions and amplitudes depends on the signal-to-noise ratio and on the number of pulses in the sequence. It was shown that, from this point of view, the efficiency of the algorithm decreases with increasing number of sequence pulses. Besides, the efficiency of the proposed algorithm decreases with decreasing signal-to-noise ratio.It was found that, unlike the classical maximum likelihood algorithm, the proposed algorithm does not require a search for the maximum of a multivariable function, meanwhile characteristics of the estimates are practically the same for both these methods. Also, it was shown that the estimation accuracy of the proposed algorithm can be increased by an insignificant increase in the number of signal samples.The results obtained may be used in the practical design of laser communication systems, in which the multipulse pulse-position modulation is used for message transmission. 

scholarly journals Audio Watermarking Combined with Compressive Sampling Based on QIM and DST-QR Techniques

2019 ◽  
Vol 19 (1) ◽  
pp. 20
Author(s):  
Irma Safitri ◽  
Gelar Budiman ◽  
Arfidianti Kartika Meiza Putri
Keyword(s):  
Signal To Noise Ratio ◽  
Transformation Process ◽  
Compressive Sampling ◽  
Audio Watermarking ◽  
Pass Filter ◽  
Low Pass Filter ◽  
Digital Copyright ◽  
Sine Transform ◽  
Low Pass ◽  
Speed Change

Abuse is not only done to copy or distribute data but also to the digital copyright labels. There is a way to protect data by inserting or hiding a piece of certain information, namely a watermarking technique. In this paper, we propose audio watermarking with Quantization Index Modulation (QIM) method as an embedding process combined with Compressive Sampling (CS), Discrete Sine Transform (DST) and QR decomposition. Binary image is used as a watermark inserted in host audio. DST is used for transformation process from time domain to frequency domain, while QR is used to decompose onedimension matrix into two-dimension matrix. Meanwhile, CS is used to obtain the compressed watermark file which is done before the embedding process. QIM method is used to embed the watermark file to the audio host file. Simulation results indicated that the proposed audio watermarking technique has good robustness against some attacks such as Low Pass Filter (LPF), resampling and linear speed change. In addition, it provides good performance in terms of imperceptibility with Signal to Noise Ratio (SNR) > 20 dB and capacity C = 689 bps.

Neural mechanisms of adaptive gain control in a joint control loop: muscle force and motoneuronal activity

1997 ◽  
Vol 200 (9) ◽  
pp. 1383-1402 ◽  
Author(s):  
R Kittmann
Keyword(s):  
Muscle Force ◽  
Gain Control ◽  
Open Loop ◽  
Chordotonal Organ ◽  
Joint Control ◽  
Modulation Amplitude ◽  
Control Loop ◽  
Pass Filter ◽  
Low Pass Filter ◽  
Change In Mean

An adaptive gain control system of a proprioceptive feedback system, the femur­tibia control loop, is investigated. It enables the joint control loop to work with a high gain but it prevents instability oscillations. In the inactive stick insect, the realisation of specific changes in gain is described for tibial torque, for extensor tibiae muscle force and for motoneuronal activity. In open-loop experiments, sinusoidal stimuli are applied to the femoral chordotonal organ (fCO). Changes in gain that depend on fCO stimulus parameters (such as amplitude, frequency and repetition rate), are investigated. Furthermore, spontaneous and touch-induced changes in gain that resemble the behavioural state of the animal are described. Changes in gain in motoneurones are always realised as changes in the amplitude of modulation of their discharge frequency. Nevertheless, depending on the stimulus situation, two different mechanisms underlie gain changes in motoneurones. (i) Changes in gain can be based on changes in the strength of the sensorimotor pathways that transmit stimulus-modulated information from the fCO to the motoneurones. (ii) Changes in gain can be based on changes in the mean activity of a motoneurone by means of its spike threshold: when, during the modulation, the discharge of a motoneurone is inhibited for part of the stimulus cycle, then a change in mean activity subsequently causes a change in modulation amplitude and gain. A new neuronal mechanism is described that helps to compensate the low-pass filter characteristics of the muscles by an increased activation, especially by a sharper distribution of spikes in the stimulus cycle at high fCO stimulus frequencies.

The Human Controller as an Adaptive, Low Pass Filter

1967 ◽  
Vol 9 (2) ◽  
pp. 141-147 ◽  
Author(s):  
Gordon H. Robinson
Keyword(s):  
Signal To Noise Ratio ◽  
Noise Signal ◽  
Manual Control ◽  
Future Research ◽  
Research Needs ◽  
Descriptive Model ◽  
Pass Filter ◽  
Low Pass Filter ◽  
Independent Variables ◽  
Low Pass

Data is presented on the ability of a human controller to track a signal contaminated with noise. Signal frequencies and signal-to-noise ratio are the independent variables. An optimal, adaptive filter is presented for comparison. A descriptive model is derived based on known human characteristics in manual control. Future research needs are discussed.

Weak Signal Detection System Based on Lock-in Amplifier Technology

2013 ◽  
Vol 389 ◽  
pp. 489-493
Author(s):  
Yong Lv ◽  
Chun Hui Niu ◽  
Yue Qiang Li ◽  
Qing Shan Chen ◽  
Xiao Ying Li ◽  
...  
Keyword(s):  
Signal Detection ◽  
Detection System ◽  
Signal To Noise Ratio ◽  
Weak Signal ◽  
Weak Signal Detection ◽  
Pass Filter ◽  
Low Pass Filter ◽  
Low Pass ◽  
Phase Sensitive ◽  
Lock In

In order to detect the weak signal deeply buried in the noise, a weak signal detection system based on lock-in amplifier is proposed. The system includes the preamplifier circuit, active low pass filter circuit, AC amplifying circuit and phase sensitive demodulation circuit. Test results show that it can greatly increase the signal-to-noise ratio (SNR) up to 12.7db.

A 2nd-Order Continuous Time Delta-Sigma Modulator for Photoplethysmography Analog Front-End

2020 ◽  
Author(s):  
Eka Fitrah Pribadi ◽  
Rajeev Kumar Pandey ◽  
Paul C.-P. Chao
Keyword(s):  
Transfer Function ◽  
Continuous Time ◽  
Supply Voltage ◽  
Signal To Noise Ratio ◽  
Pass Filter ◽  
Low Pass Filter ◽  
Delta Sigma Modulator ◽  
Delta Sigma ◽  
Low Pass ◽  
Order Continuous

Abstract A brief presents a 2nd order continuous-time delta-sigma modulator (CT-DSM) using a low pass filter to reduce the slew rate requirement of the output swing of the first integrator. By adding the low pass filter, the desired transfer function of the CT-DSM is altered. Thus a feed-forward based compensation circuit is introduced to transform the altered transfer function to the original condition. The CT-DSM is designed with a bandwidth of 100 Hz to satisfy the requirement of photoplethysmogram (PPG) detection. The CT-DSM is simulated using CMOS 180 nm technology with the layout area 460 μm × 460 μm. The circuit uses a 1.8 V supply voltage and consumes 35.61 μW. The signal-to-noise ratio of the CT-DSM is 101.2 dB, while the SFDR is 99.1 dB.

scholarly journals A CMOS Transmitter Analog Baseband for 5G Mobile Communication

Electronics ◽  
2019 ◽  
Vol 8 (11) ◽  
pp. 1319
Author(s):  
Yen ◽  
Chen ◽  
Wei ◽  
Chung
Keyword(s):  
Mobile Communication ◽  
Gain Control ◽  
Cmos Technology ◽  
Control Technique ◽  
Constant Current ◽  
Constant Current Density ◽  
Pass Filter ◽  
Low Pass Filter ◽  
Analog Baseband ◽  
5G Mobile Communication

CMOS analog baseband circuits including a low-pass filter (LPF) and a programmable gain amplifier (PGA) are designed and implemented for the fifth-generation (5G) mobile communication. The super source follower topology is adopted to achieve a wideband LPF with good linearity, while the constant current density gain control technique is used to implement gain cells of the PGA. The circuits are integrated as an analog baseband for a 5G transmitter (TX) and fabricated using TSMC 90-nm CMOS technology. The analog baseband exhibits the bandwidth from 1.03 to 1.05 GHz when the voltage gain is varied from −18.9 dB to 3.8 dB in 1-dB steps. The gain step errors are within −0.7 dB to 0.9 dB. In the highest gain mode, the analog baseband achieves the IP1dB of −10 dBv and the IIP3 of −0.2 dBv. Over the band of interest, the NF of the analog baseband is 24.4–40.0 dB.

scholarly journals A Novel Sample Based Quadrature Phase Shift Keying Demodulator

2014 ◽  
Vol 2014 ◽  
pp. 1-7
Author(s):  
Asraf Mohamed Moubark ◽  
Sawal Hamid Md Ali
Keyword(s):  
Phase Noise ◽  
Signal To Noise Ratio ◽  
Local Oscillator ◽  
Coherent Detection ◽  
Signal Frequency ◽  
Pass Filter ◽  
Low Pass Filter ◽  
Incoming Signal ◽  
Shift Keying ◽  
Low Pass

This paper presents a new practical QPSK receiver that uses digitized samples of incoming QPSK analog signal to determine the phase of the QPSK symbol. The proposed technique is more robust to phase noise and consumes up to 89.6% less power for signal detection in demodulation operation. On the contrary, the conventional QPSK demodulation process where it uses coherent detection technique requires the exact incoming signal frequency; thus, any variation in the frequency of the local oscillator or incoming signal will cause phase noise. A software simulation of the proposed design was successfully carried out using MATLAB Simulink software platform. In the conventional system, at least 10 dB signal to noise ratio (SNR) is required to achieve the bit error rate (BER) of 10−6, whereas, in the proposed technique, the same BER value can be achieved with only 5 dB SNR. Since some of the power consuming elements such as voltage control oscillator (VCO), mixer, and low pass filter (LPF) are no longer needed, the proposed QPSK demodulator will consume almost 68.8% to 99.6% less operational power compared to conventional QPSK demodulator.

scholarly journals Climatic information archived in ice cores: impact of intermittency and diffusion on the recorded isotopic signal in Antarctica

2019 ◽  
Author(s):  
Mathieu Casado ◽  
Thomas Münch ◽  
Thomas Laepple
Keyword(s):  
Time Scales ◽  
Signal To Noise Ratio ◽  
Ice Core ◽  
Ice Cores ◽  
Reanalysis Data ◽  
Climatic Conditions ◽  
Pass Filter ◽  
Low Pass Filter ◽  
Frequency Changes ◽  
Isotopic Signal

Abstract. The isotopic signal (δ18O and δD) imprinted in ice cores from Antarctica is not solely generated by the temperature sensitivity of the isotopic composition of precipitation but also contains the signature of the intermittency of precipitation patterns as well as of post-deposition processes occurring at the surface and in the firn. This leads to a proxy signal recorded by the ice cores that may not be representative of the local climatic variations. Due to precipitation intermittency, the ice cores only record brief snapshots of the climatic conditions, resulting in aliasing of the climatic signal, and thus a large amount of noise which reduces the minimum temporal resolution at which a meaningful signal can be retrieved. The analyses are further complicated by isotopic diffusion which acts as a low pass filter that dampens any high frequency changes. Here, we use reanalysis data (ERA-Interim) combined with satellite products of accumulation to evaluate the spatial distribution of the transfer function that describes the formation of the isotopic signal across Antarctica. The minimum time scales at which the signal-to-noise ratio exceeds unity range from less than a year at the coast to a thousand years further inland. Based on solely physical processes, we were thus able to define a lower bound for the time scales at which climate variability can be reconstructed from ice core water isotopic compositions.

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