This document summarizes a master's thesis that designed a low power successive approximation register (SAR) analog-to-digital converter (ADC) for wireless sensor networks. Key aspects of the design included:
- The SAR ADC was designed in a 28nm CMOS process with 12-bit resolution, 500 kS/s sampling rate, and a supply voltage of 0.5V ±10%.
- An asynchronous SAR architecture was used to allow faster conversion times and lower power consumption compared to a synchronous design.
- The comparator and sampling DAC were sized to meet noise constraints while operating at sub-threshold voltages for low power. Time constraints were also considered for bit conversions.
- Models of
Nyquist criterion for distortion less baseband binary channelPriyangaKR1
binary transmission system
From design point of view – frequency response of the channel and transmitted pulse shape are specified; the frequency response of the transmit and receive filters has to be determined so as to reconstruct [bk]
This document discusses digital communication systems and provides an overview of several key topics:
- It introduces line coding techniques and their properties.
- It describes the basic digital communication block diagram and advantages of digital transmission.
- It discusses intersymbol interference, equalization techniques like zero-forcing equalization, and eye patterns.
- It provides information on topics like noise immunity, regenerative repeaters, and pulse shaping to eliminate intersymbol interference.
1. VSB modulation is used for picture transmission in commercial TV in India as it provides a compromise between SSB and DSB. Speech signals use FM modulation for its noise immunity.
2. In a DSB AM system, if the modulation index is doubled, the ratio of sideband power to carrier power increases by a factor of 4.
3. The maximum power efficiency of an AM modulator is 33%.
This document discusses the application of synchronized phasor measurement in real-time wide-area monitoring. It provides an overview of phasor and synchrophasor measurement techniques using Fourier transforms. It also discusses power system stability and transient stability. The document demonstrates the monitoring of a multi-machine system using synchronized phasor measurements by simulating various fault conditions and load changes on a 3-machine, 9-bus system and observing the results with a phasor measurement unit to analyze stability.
This document provides examples and solutions for calculating various parameters related to communications systems. Example 1.1 calculates wavelength for different frequencies. Example 4.9 calculates the signal-to-noise ratio at the output of an FM receiver detector given the input SNR and modulation parameters. The examples cover topics such as noise power, amplifiers, modulation, transmitters and more. They demonstrate calculations for key concepts in communications systems.
This document discusses amplitude modulated communication systems. It describes how a carrier signal is modulated by a baseband modulating signal to allow for information exchange over a channel. There are different types of modulation including continuous wave, pulse, and digital modulation. Amplitude modulation varies the amplitude of the carrier signal based on the instantaneous value of the modulating signal. This allows for multiplexing of multiple messages and use of more practical antenna sizes. Specific amplitude modulation techniques are described like conventional AM, DSB-SC, SSB, and VSB along with their tradeoffs in terms of carrier suppression, bandwidth, cost, and applications.
The document describes the design and simulation of a 2-input NOR gate in 0.25um CMOS technology. The goal is to minimize propagation delay and area. Various transistor widths and lengths are analyzed to determine the optimal sizing. Shared diffusion is used to reduce parasitic capacitance and area. The final design has a propagation delay of 0.65ns and passes DRC and LVS checks.
The document proposes a time-frequency domain approach for pitch estimation of noisy speech that uses an inverse circular average magnitude difference function to weight the autocorrelation function of pre-filtered noisy speech. It estimates the dominant pitch harmonic in the frequency domain using a cosine model of autocorrelation function before optimally fitting a variable period impulse train to the weighted autocorrelation function for pitch estimation. Simulation results using the Keele speech database show the proposed method achieves better pitch estimation accuracy than conventional autocorrelation-based methods, even at low signal-to-noise ratios down to -10 dB.
Nyquist criterion for distortion less baseband binary channelPriyangaKR1
binary transmission system
From design point of view – frequency response of the channel and transmitted pulse shape are specified; the frequency response of the transmit and receive filters has to be determined so as to reconstruct [bk]
This document discusses digital communication systems and provides an overview of several key topics:
- It introduces line coding techniques and their properties.
- It describes the basic digital communication block diagram and advantages of digital transmission.
- It discusses intersymbol interference, equalization techniques like zero-forcing equalization, and eye patterns.
- It provides information on topics like noise immunity, regenerative repeaters, and pulse shaping to eliminate intersymbol interference.
1. VSB modulation is used for picture transmission in commercial TV in India as it provides a compromise between SSB and DSB. Speech signals use FM modulation for its noise immunity.
2. In a DSB AM system, if the modulation index is doubled, the ratio of sideband power to carrier power increases by a factor of 4.
3. The maximum power efficiency of an AM modulator is 33%.
This document discusses the application of synchronized phasor measurement in real-time wide-area monitoring. It provides an overview of phasor and synchrophasor measurement techniques using Fourier transforms. It also discusses power system stability and transient stability. The document demonstrates the monitoring of a multi-machine system using synchronized phasor measurements by simulating various fault conditions and load changes on a 3-machine, 9-bus system and observing the results with a phasor measurement unit to analyze stability.
This document provides examples and solutions for calculating various parameters related to communications systems. Example 1.1 calculates wavelength for different frequencies. Example 4.9 calculates the signal-to-noise ratio at the output of an FM receiver detector given the input SNR and modulation parameters. The examples cover topics such as noise power, amplifiers, modulation, transmitters and more. They demonstrate calculations for key concepts in communications systems.
This document discusses amplitude modulated communication systems. It describes how a carrier signal is modulated by a baseband modulating signal to allow for information exchange over a channel. There are different types of modulation including continuous wave, pulse, and digital modulation. Amplitude modulation varies the amplitude of the carrier signal based on the instantaneous value of the modulating signal. This allows for multiplexing of multiple messages and use of more practical antenna sizes. Specific amplitude modulation techniques are described like conventional AM, DSB-SC, SSB, and VSB along with their tradeoffs in terms of carrier suppression, bandwidth, cost, and applications.
The document describes the design and simulation of a 2-input NOR gate in 0.25um CMOS technology. The goal is to minimize propagation delay and area. Various transistor widths and lengths are analyzed to determine the optimal sizing. Shared diffusion is used to reduce parasitic capacitance and area. The final design has a propagation delay of 0.65ns and passes DRC and LVS checks.
The document proposes a time-frequency domain approach for pitch estimation of noisy speech that uses an inverse circular average magnitude difference function to weight the autocorrelation function of pre-filtered noisy speech. It estimates the dominant pitch harmonic in the frequency domain using a cosine model of autocorrelation function before optimally fitting a variable period impulse train to the weighted autocorrelation function for pitch estimation. Simulation results using the Keele speech database show the proposed method achieves better pitch estimation accuracy than conventional autocorrelation-based methods, even at low signal-to-noise ratios down to -10 dB.
This document discusses active filter circuits using operational amplifiers (op amps). It begins by examining the disadvantages of passive filter circuits, such as inability to amplify and sensitivity of cutoff frequency to load resistance. The document then examines first-order low-pass, high-pass, and bandpass filter circuits using op amps. It discusses designing these filters through component value calculations and scaling. Higher order filters created by cascading multiple first-order sections are also covered. The document concludes by discussing Butterworth filters, which have maximally flat frequency responses.
- Obtained the Fast Fourier Transform of signals.
- Designed and Validated Low Pass, High Pass, and Band Pass filters in compliance with the specifications.
- Produced and compared graphs of the results upon processing.
The document contains details about sampling a bandpass signal with varying center frequency fo from 5 kHz to 50 kHz at a sampling rate of 25 kHz.
It analyzes the ranges of fo for which the sampling rate is adequate by calculating the variation in bandwidth (k) as fo changes. It concludes that the sampling rate of 25 kHz is adequate when fo is between 5-7.5 kHz, 15-20 kHz, 25-32.5 kHz, and 35-50 kHz.
Fundamentals of Passive and Active Sonar Technical Training Short Course SamplerJim Jenkins
This four-day course is designed for SONAR systems engineers, combat systems engineers, undersea warfare professionals, and managers who wish to enhance their understanding of passive and active SONAR or become familiar with the "big picture" if they work outside of either discipline. Each topic is presented by instructors with substantial experience at sea. Presentations are illustrated by worked numerical examples using simulated or experimental data describing actual undersea acoustic situations and geometries. Visualization of transmitted waveforms, target interactions, and detector responses is emphasized.
On The Fundamental Aspects of DemodulationCSCJournals
When the instantaneous amplitude, phase and frequency of a carrier wave are modulated with the information signal for transmission, it is known that the receiver works on the basis of the received signal and a knowledge of the carrier frequency. The question is: If the receiver does not have the a priori information about the carrier frequency, is it possible to carry out the demodulation process? This tutorial lecture answers this question by looking into the very fundamental process by which the modulated wave is generated. It critically looks into the energy separation algorithm for signal analysis and suggests modification for distortionless demodulation of an FM signal, and recovery of sub-carrier signals
(1) The document discusses obtaining first-order low-pass, high-pass, band-reject, and band-pass filters from a first-order all-pass filter.
(2) It shows how to derive the transfer functions for each filter type by adding or subtracting the input and output of one or two cascaded all-pass filter sections.
(3) Key specifications of each filter like cutoff frequencies, gain, and bandwidth are calculated and verified through SPICE simulation, showing good agreement between calculated and simulated responses.
The document discusses diversity techniques for flat fading channels. It describes different types of diversity including space, time, frequency, and polarization diversity. It also discusses selection diversity and maximum ratio combining (MRC). MRC is shown to provide optimal performance by maximizing the output signal-to-noise ratio. For large average SNR, the bit error rate for MRC is proportional to (1/SNR)^L, where L is the diversity order, providing significant diversity gain.
Novel RF Power Amplifier Linearization Proof-Of-Concept Bipolar Ne46134welahdab
The document presents a novel approach for linearizing RF power amplifiers to improve efficiency while maintaining linearity. It describes the nonlinearity of amplifiers and how predistortion is commonly used as a linearization technique. The proposed technique achieves better linearity and efficiency simultaneously with little insertion loss or die area increase. Experimental results on prototype amplifiers show improvements like higher 1dB compression point, 6-10dB lower IM3, and maintained gain and PAE over a wider input power range when the technique is applied.
The document summarizes Lecture 7 which covered:
1) A review of Lecture 6 on PCM waveforms and the remaining portion of Chapter 2 on spectral densities of PCM waveforms and multi-level signaling.
2) An overview of Chapter 3 on baseband demodulation/detection including matched filters, correlators, Bayes' decision criterion, and maximum likelihood detection.
3) Key aspects of line codes including how pulse shaping can control the signal spectrum and ensure symbol transitions, comparisons of line codes based on power spectral density, DC component, and bandwidth.
Using Distortion Shaping Technique to Equalize ADC THD Performance Between ATEsPete Sarson, PH.D
This document discusses using distortion shaping techniques to equalize the THD performance of ADCs between different automatic test equipment (ATEs). It presents a method of generating test signals with an arbitrary waveform generator (AWG) that shifts harmonic distortion components closer to the Nyquist frequency to improve the AWG's harmonic performance and allow more accurate testing of ADC THD specifications. Test results show the proposed method significantly reduces harmonic distortion in the AWG output and allows different testers to achieve identical ADC test results, resolving issues caused by performance differences between ATEs. The technique is especially effective for higher resolution ADCs where noise floors are lower.
Sigma-Delta Analog to Digital ConvertersSatish Patil
In recent years Sigma-Delta ADCs became one of the most popular types of Analog-to-Digital converters. The key features of these are high-speed, high resolution and low operating voltages. These are commonly used in variety of applications like digital audio CDs, CODEC, biomedical sensor applications and wireless transmitters/receivers. The basic principles involved in this technique are oversampling and noise shaping. This report reviews different techniques proposed for high resolution, low power Sigma-Delta ADC. Conventional design of SDM was dominated by discrete time architecture but in modern designs continuous types are also becoming famous because of their low power attributes. Continuous efforts have been taken to reduce the supply voltages of SDM and recently, lowest reported is 250mv.
This document describes a technique for digitally calibrating the current of a digitally controlled oscillator (DCO) to optimize its phase noise performance across process and temperature variations. The phase error (PHE) signal from a digital PLL is digitized and used to estimate the DCO's phase noise. By adjusting the DCO current digitally based on the estimated phase noise, the optimum operating point with minimum phase noise can be identified. Measurement results on a 90nm CMOS chip demonstrate good correlation between the estimated and measured DCO phase noise, validating the digital calibration approach.
The document describes a simulation of an optical communication system using different modulation formats at bit rates of 10.7 Gbps and 42.65 Gbps. The system includes a transmitter with a laser, data source, modulator, and amplifier. A noisy channel is modeled by adding ASE noise. A receiver includes a filter, photodetector, amplifier, and error counting. Eye diagrams, spectra, and BER are analyzed for each modulation format under various channel conditions including OSNR and chromatic dispersion. The impact of increasing bandwidth for PSBT is also examined.
This document discusses correlative-level coding and its applications in baseband pulse transmission systems. Correlative-level coding introduces controlled intersymbol interference to increase signaling rate. It allows partial response signaling and maximum likelihood detection at the receiver. Specific techniques discussed include duobinary signaling and modified duobinary signaling. The document also covers tapped-delay line equalization using adaptive algorithms like least mean square to compensate for channel distortion. Decision feedback equalization and its implementation are summarized as well. Eye patterns are described as a tool to evaluate signal quality in such systems.
This document discusses various types of pulse modulation techniques used in analog and digital communication systems. It begins by defining pulse amplitude modulation (PAM) and describing how the amplitude of pulses varies proportionally to the message signal. It then discusses different types of PAM based on the sampling technique used - ideal, natural, and flat-top sampling. Flat-top sampling uses sample-and-hold circuits and can introduce amplitude distortion known as the aperture effect. The document also covers pulse width modulation (PWM), pulse position modulation (PPM), pulse code modulation (PCM), delta modulation (DM), and their advantages. It explains the sampling theorem and proves it through Fourier analysis. Finally, it discusses bandwidth requirements, transmission, drawbacks
This document discusses various methods of modulating digital and analog data for transmission:
1. It describes digital-to-analog modulation techniques including amplitude shift keying (ASK), frequency shift keying (FSK), phase shift keying (PSK), and quadrature amplitude modulation (QAM).
2. It explains the relationships between bit rate, baud rate, and bandwidth for different modulation schemes. ASK, FSK, and PSK have baud rate equal to bit rate, while higher-order PSK and QAM can have higher bit rates through multiple bits per symbol.
3. Modems and standards like V.32, V.34, and V.90 are discussed in the context of mod
1. The document describes signal processing techniques for Synthetic Aperture Radar (SAR), including SAR signal specifications, different SAR imaging modes, and common SAR processing algorithms like Range Doppler and Chirp Scaling.
2. Key steps in SAR processing are described, such as de-chirping the received signal, applying the stationary phase principle to focus the signal, and using techniques like reference function multiplication to decouple nonlinear dependencies in the signal.
3. Several test conditions are simulated to validate the SAR processing algorithms, including varying parameters like the number and location of targets and the squint angle. Target detection performance is evaluated under each condition.
Concurrent Triple Band Low Noise Amplifier DesignHalil Kayıhan
The document describes the design of a concurrent triple-band low noise amplifier (LNA) that operates at 1.8 GHz, 2.4 GHz, and 5.2 GHz. A cascode structure with a source degeneration inductor is used. The input matching network employs a multi-element LC filter to match the input to 50 ohms across all three bands. Separate output resonance circuits are used for each band. Simulation results show the LNA achieves good input matching and noise figure across bands while providing sufficient gain and linearity.
This document discusses sampling and analog-to-digital conversion. It begins by introducing sampling and the sampling theorem, which states that a continuous-time signal can be uniquely represented by its samples if the sampling frequency is greater than twice the highest frequency present in the signal. It then describes the components and process of analog-to-digital conversion, including periodic sampling, quantization, encoding, and the effects of aliasing if the sampling rate is too low. It also discusses digital-to-analog conversion and reconstruction of the original analog signal from its samples.
The document describes the design of an oscillator using a CMOS operational transconductance amplifier (OTA). It discusses different current mirror circuits used to design the CMOS OTA and selects the cascode current mirror as most suitable. RC phase shift and Wien bridge oscillators are then created using the CMOS OTA. Simulation results show the oscillators generate sine waves close to their calculated frequencies. The CMOS OTA provides advantages like high bandwidth and slew rate over traditional op-amps.
This document summarizes a master's thesis on designing a very low power successive approximation register (SAR) analog-to-digital converter (ADC) for wireless sensor networks. The thesis aimed to design a 12-bit 500 kS/s SAR ADC using a 28nm CMOS process and operating at 0.5V supply voltage. It discusses SAR ADC topology and improvements, state of the art converters, the design of key blocks including the comparator, sampling DAC, and delay block. It also covers timing constraints, noise constraints, sizing of blocks, and calibration techniques. The thesis was conducted in partnership with Synopsys to follow industrial IC design methodologies.
This document discusses active filter circuits using operational amplifiers (op amps). It begins by examining the disadvantages of passive filter circuits, such as inability to amplify and sensitivity of cutoff frequency to load resistance. The document then examines first-order low-pass, high-pass, and bandpass filter circuits using op amps. It discusses designing these filters through component value calculations and scaling. Higher order filters created by cascading multiple first-order sections are also covered. The document concludes by discussing Butterworth filters, which have maximally flat frequency responses.
- Obtained the Fast Fourier Transform of signals.
- Designed and Validated Low Pass, High Pass, and Band Pass filters in compliance with the specifications.
- Produced and compared graphs of the results upon processing.
The document contains details about sampling a bandpass signal with varying center frequency fo from 5 kHz to 50 kHz at a sampling rate of 25 kHz.
It analyzes the ranges of fo for which the sampling rate is adequate by calculating the variation in bandwidth (k) as fo changes. It concludes that the sampling rate of 25 kHz is adequate when fo is between 5-7.5 kHz, 15-20 kHz, 25-32.5 kHz, and 35-50 kHz.
Fundamentals of Passive and Active Sonar Technical Training Short Course SamplerJim Jenkins
This four-day course is designed for SONAR systems engineers, combat systems engineers, undersea warfare professionals, and managers who wish to enhance their understanding of passive and active SONAR or become familiar with the "big picture" if they work outside of either discipline. Each topic is presented by instructors with substantial experience at sea. Presentations are illustrated by worked numerical examples using simulated or experimental data describing actual undersea acoustic situations and geometries. Visualization of transmitted waveforms, target interactions, and detector responses is emphasized.
On The Fundamental Aspects of DemodulationCSCJournals
When the instantaneous amplitude, phase and frequency of a carrier wave are modulated with the information signal for transmission, it is known that the receiver works on the basis of the received signal and a knowledge of the carrier frequency. The question is: If the receiver does not have the a priori information about the carrier frequency, is it possible to carry out the demodulation process? This tutorial lecture answers this question by looking into the very fundamental process by which the modulated wave is generated. It critically looks into the energy separation algorithm for signal analysis and suggests modification for distortionless demodulation of an FM signal, and recovery of sub-carrier signals
(1) The document discusses obtaining first-order low-pass, high-pass, band-reject, and band-pass filters from a first-order all-pass filter.
(2) It shows how to derive the transfer functions for each filter type by adding or subtracting the input and output of one or two cascaded all-pass filter sections.
(3) Key specifications of each filter like cutoff frequencies, gain, and bandwidth are calculated and verified through SPICE simulation, showing good agreement between calculated and simulated responses.
The document discusses diversity techniques for flat fading channels. It describes different types of diversity including space, time, frequency, and polarization diversity. It also discusses selection diversity and maximum ratio combining (MRC). MRC is shown to provide optimal performance by maximizing the output signal-to-noise ratio. For large average SNR, the bit error rate for MRC is proportional to (1/SNR)^L, where L is the diversity order, providing significant diversity gain.
Novel RF Power Amplifier Linearization Proof-Of-Concept Bipolar Ne46134welahdab
The document presents a novel approach for linearizing RF power amplifiers to improve efficiency while maintaining linearity. It describes the nonlinearity of amplifiers and how predistortion is commonly used as a linearization technique. The proposed technique achieves better linearity and efficiency simultaneously with little insertion loss or die area increase. Experimental results on prototype amplifiers show improvements like higher 1dB compression point, 6-10dB lower IM3, and maintained gain and PAE over a wider input power range when the technique is applied.
The document summarizes Lecture 7 which covered:
1) A review of Lecture 6 on PCM waveforms and the remaining portion of Chapter 2 on spectral densities of PCM waveforms and multi-level signaling.
2) An overview of Chapter 3 on baseband demodulation/detection including matched filters, correlators, Bayes' decision criterion, and maximum likelihood detection.
3) Key aspects of line codes including how pulse shaping can control the signal spectrum and ensure symbol transitions, comparisons of line codes based on power spectral density, DC component, and bandwidth.
Using Distortion Shaping Technique to Equalize ADC THD Performance Between ATEsPete Sarson, PH.D
This document discusses using distortion shaping techniques to equalize the THD performance of ADCs between different automatic test equipment (ATEs). It presents a method of generating test signals with an arbitrary waveform generator (AWG) that shifts harmonic distortion components closer to the Nyquist frequency to improve the AWG's harmonic performance and allow more accurate testing of ADC THD specifications. Test results show the proposed method significantly reduces harmonic distortion in the AWG output and allows different testers to achieve identical ADC test results, resolving issues caused by performance differences between ATEs. The technique is especially effective for higher resolution ADCs where noise floors are lower.
Sigma-Delta Analog to Digital ConvertersSatish Patil
In recent years Sigma-Delta ADCs became one of the most popular types of Analog-to-Digital converters. The key features of these are high-speed, high resolution and low operating voltages. These are commonly used in variety of applications like digital audio CDs, CODEC, biomedical sensor applications and wireless transmitters/receivers. The basic principles involved in this technique are oversampling and noise shaping. This report reviews different techniques proposed for high resolution, low power Sigma-Delta ADC. Conventional design of SDM was dominated by discrete time architecture but in modern designs continuous types are also becoming famous because of their low power attributes. Continuous efforts have been taken to reduce the supply voltages of SDM and recently, lowest reported is 250mv.
This document describes a technique for digitally calibrating the current of a digitally controlled oscillator (DCO) to optimize its phase noise performance across process and temperature variations. The phase error (PHE) signal from a digital PLL is digitized and used to estimate the DCO's phase noise. By adjusting the DCO current digitally based on the estimated phase noise, the optimum operating point with minimum phase noise can be identified. Measurement results on a 90nm CMOS chip demonstrate good correlation between the estimated and measured DCO phase noise, validating the digital calibration approach.
The document describes a simulation of an optical communication system using different modulation formats at bit rates of 10.7 Gbps and 42.65 Gbps. The system includes a transmitter with a laser, data source, modulator, and amplifier. A noisy channel is modeled by adding ASE noise. A receiver includes a filter, photodetector, amplifier, and error counting. Eye diagrams, spectra, and BER are analyzed for each modulation format under various channel conditions including OSNR and chromatic dispersion. The impact of increasing bandwidth for PSBT is also examined.
This document discusses correlative-level coding and its applications in baseband pulse transmission systems. Correlative-level coding introduces controlled intersymbol interference to increase signaling rate. It allows partial response signaling and maximum likelihood detection at the receiver. Specific techniques discussed include duobinary signaling and modified duobinary signaling. The document also covers tapped-delay line equalization using adaptive algorithms like least mean square to compensate for channel distortion. Decision feedback equalization and its implementation are summarized as well. Eye patterns are described as a tool to evaluate signal quality in such systems.
This document discusses various types of pulse modulation techniques used in analog and digital communication systems. It begins by defining pulse amplitude modulation (PAM) and describing how the amplitude of pulses varies proportionally to the message signal. It then discusses different types of PAM based on the sampling technique used - ideal, natural, and flat-top sampling. Flat-top sampling uses sample-and-hold circuits and can introduce amplitude distortion known as the aperture effect. The document also covers pulse width modulation (PWM), pulse position modulation (PPM), pulse code modulation (PCM), delta modulation (DM), and their advantages. It explains the sampling theorem and proves it through Fourier analysis. Finally, it discusses bandwidth requirements, transmission, drawbacks
This document discusses various methods of modulating digital and analog data for transmission:
1. It describes digital-to-analog modulation techniques including amplitude shift keying (ASK), frequency shift keying (FSK), phase shift keying (PSK), and quadrature amplitude modulation (QAM).
2. It explains the relationships between bit rate, baud rate, and bandwidth for different modulation schemes. ASK, FSK, and PSK have baud rate equal to bit rate, while higher-order PSK and QAM can have higher bit rates through multiple bits per symbol.
3. Modems and standards like V.32, V.34, and V.90 are discussed in the context of mod
1. The document describes signal processing techniques for Synthetic Aperture Radar (SAR), including SAR signal specifications, different SAR imaging modes, and common SAR processing algorithms like Range Doppler and Chirp Scaling.
2. Key steps in SAR processing are described, such as de-chirping the received signal, applying the stationary phase principle to focus the signal, and using techniques like reference function multiplication to decouple nonlinear dependencies in the signal.
3. Several test conditions are simulated to validate the SAR processing algorithms, including varying parameters like the number and location of targets and the squint angle. Target detection performance is evaluated under each condition.
Concurrent Triple Band Low Noise Amplifier DesignHalil Kayıhan
The document describes the design of a concurrent triple-band low noise amplifier (LNA) that operates at 1.8 GHz, 2.4 GHz, and 5.2 GHz. A cascode structure with a source degeneration inductor is used. The input matching network employs a multi-element LC filter to match the input to 50 ohms across all three bands. Separate output resonance circuits are used for each band. Simulation results show the LNA achieves good input matching and noise figure across bands while providing sufficient gain and linearity.
This document discusses sampling and analog-to-digital conversion. It begins by introducing sampling and the sampling theorem, which states that a continuous-time signal can be uniquely represented by its samples if the sampling frequency is greater than twice the highest frequency present in the signal. It then describes the components and process of analog-to-digital conversion, including periodic sampling, quantization, encoding, and the effects of aliasing if the sampling rate is too low. It also discusses digital-to-analog conversion and reconstruction of the original analog signal from its samples.
The document describes the design of an oscillator using a CMOS operational transconductance amplifier (OTA). It discusses different current mirror circuits used to design the CMOS OTA and selects the cascode current mirror as most suitable. RC phase shift and Wien bridge oscillators are then created using the CMOS OTA. Simulation results show the oscillators generate sine waves close to their calculated frequencies. The CMOS OTA provides advantages like high bandwidth and slew rate over traditional op-amps.
This document summarizes a master's thesis on designing a very low power successive approximation register (SAR) analog-to-digital converter (ADC) for wireless sensor networks. The thesis aimed to design a 12-bit 500 kS/s SAR ADC using a 28nm CMOS process and operating at 0.5V supply voltage. It discusses SAR ADC topology and improvements, state of the art converters, the design of key blocks including the comparator, sampling DAC, and delay block. It also covers timing constraints, noise constraints, sizing of blocks, and calibration techniques. The thesis was conducted in partnership with Synopsys to follow industrial IC design methodologies.
This document discusses the layout of analog CMOS integrated circuits. It focuses on the layout of transistors and basic cells. Key topics covered include:
- Layout of a single transistor, use of multiple fingers, and interdigitated transistors for matching.
- Common centroid layouts and dummy devices to reduce mismatch.
- Ensuring matched interconnect resistance, capacitance, and parasitics.
- Stacked layout of analog cells with stick diagrams to represent multiple transistors.
- Two examples of laying out basic cells - a two-stage op-amp and folded cascode. Design considerations like transistor sizing and grouping are discussed.
The document discusses layout design rules for integrated circuits. It provides guidelines for feature sizes and spacings to ensure fabricated circuits meet intended designs. This includes minimum line widths, separations between layers, and allowances for misalignment. The document also notes two key checks that must be completed to validate a mask design: a design rule check to verify rules are followed, and circuit extraction to confirm masks produce the correct interconnected circuit.
The document contains notes on Analog and Digital VLSI Design from a course taught at BITS Pilani in Fall 2013. It includes a disclaimer noting that the information is provided for educational purposes only without any guarantees. It also states that the content was prepared by Akshansh Chaudhary based on a course by Prof. Vijaya Gunturu and includes copyright information and a link for more resources.
This document discusses the design and characteristics of CMOS voltage comparators. It begins by defining the basic requirement of a comparator to compare an analog input voltage to a reference voltage and output a binary signal. It then covers comparator static characteristics like gain, offset voltage, resolution and noise. Dynamic characteristics of propagation delay and slew rate are also discussed. Different comparator circuit topologies like open-loop, regenerative and high-speed designs are presented. The document provides small-signal models of common comparator circuits and examines the effects of hysteresis. It concludes by presenting the typical architecture of high-speed comparators using preamplifier and latch stages to minimize propagation delay.
1. The document discusses the process for designing and fabricating analog integrated circuits using CMOS technology. It covers topics like MOS transistor operation, CMOS fabrication process steps, and SPICE device modeling parameters.
2. The CMOS fabrication process involves growing gate oxides, depositing polysilicon gates, implanting sources and drains, and depositing multiple metal interconnect layers.
3. SPICE device models are needed to simulate the behavior of transistors and circuits during the design process prior to fabrication. Device parameters are provided for a 0.8 micron CMOS process.
The document provides an overview of analog layout design. It discusses that analog circuits require careful attention to geometry during layout due to process variations. The analog design flow includes electrical design, physical design involving layout, and fabrication/testing. Key considerations for analog layout include minimizing parasitic resistances and capacitances, reducing noise, and ensuring matching between identical components using techniques like common-centroid layout. Resistors and capacitors must be carefully laid out to minimize non-ideal effects and provide accurate values.
This document presents the design of a high performance folded cascade OTA and sample and hold circuit. The OTA is designed to achieve 10-bit resolution while operating at a 28 MHz sampling frequency. Simulation results show the OTA achieves a high open loop gain of 72 dB and bandwidth of 112 MHz, with a phase margin of 73 degrees. A low resistance transmission gate switch is designed to reduce charge injection and clock feedthrough effects during sampling. The circuit is implemented in a 130 nm CMOS technology.
This document discusses the analysis and optimization of a PLL loop for USB3 and DisplayPort applications. It provides the block diagram and describes simulations performed to analyze stability, phase step response, reference clock noise, VCO noise, fractional-N effects, and charge pump mismatch. Equations for loop filter components like C1, R1, and C2 are derived based on parameters like reference clock frequency, VCO gain, and charge pump current. Optimization results targeting bandwidth, peaking, drift, and area are presented for different configurations including integer and fractional-N modes with 240MHz and 24MHz reference clocks.
Design of 17-Bit Audio Band Delta-Sigma Analog to Digital ConverterKarthik Rathinavel
• Systematically designed a delta sigma ADC with CIFF modular architecture in MATLAB Simulink with an ENOB of 19-bits.
• Designed a decimation filter to remove noise in the digital output of the delta sigma modulator.
• Observed the effect of non-idealities on the modulator such as finite gain, finite bandwidth, slew rate, analog noise and capacitor mismatch.
The document describes the design of a 12-bit digital to analog converter (DAC). It includes a binary weighted resistor ladder circuit to convert the digital input to an analog voltage, and an operational amplifier circuit to drive the output load. Simulation results show the DAC can operate at up to 25MHz with good linearity and accuracy. Layout design considerations are discussed to optimize circuit performance and minimize parasitics.
Modelling and Simulation of a SAR ADC with Internally Generated Conversion Si...iosrjce
This paper presents the modeling and simulation of a 833.33 kS/s, 51.279µW successive
approximation register(SAR) Analog to Digital Converter(ADC) using 0.18μm CMOS technology that uses
internally generated signal for approximation for low power applications. The ADC is powered by single supply
voltage of 1V. In our scheme, comparator output time and bit settling time of the Digital to Analog
Converter(DAC) are utilized to generate a signal level such that the next step of the conversion can take place.
This model is significant for Globally Asynchronous Locally Synchronous(GALS) system integration.
A Low Phase Noise CMOS Quadrature Voltage Control Oscillator Using Clock Gate...IJERA Editor
This project presents the low phase noise cmos quadrature voltage control oscillator using clock gating technique. Here the colpitts vco is used to split the capacitance in the Qvco circuit producing quadrature output. The startup condition in the oscillator is improved by using 𝐺𝑚enhancement [12].This QVCO performs the operation anti phase injection locking fordevice reuse [8]. The new clock gating technique is used to reduce the power with thepower supply 1.5v. The QVCO uses a 0.5m𝐴with phase error of 0.4𝑜and exhibits a phase noise of -118dBc/HZ at 1MHZ offset at the centre frequency of 500MHZ.
A Low Phase Noise CMOS Quadrature Voltage Control Oscillator Using Clock Gate...IJERA Editor
This project presents the low phase noise cmos quadrature voltage control oscillator using clock gating technique. Here the colpitts vco is used to split the capacitance in the Qvco circuit producing quadrature output. The startup condition in the oscillator is improved by using 퐺푚enhancement [12].This QVCO performs the operation anti phase injection locking fordevice reuse [8]. The new clock gating technique is used to reduce the power with thepower supply 1.5v. The QVCO uses a 0.5m퐴with phase error of 0.4표and exhibits a phase noise of -118dBc/HZ at 1MHZ offset at the centre frequency of 500MHZ.
Index terms: current switching, clock gating, phase noise, Qvco
Design and Implementation of Area Optimized, Low Complexity CMOS 32nm Technol...IJERA Editor
A numerically controlled oscillator (NCO) is a digital signal generator which is a very important block in many Digital Communication Systems such as Software Defined Radios, Digital Radio set and Modems, Down/Up converters for Cellular and PCS base stations etc. NCO creates a synchronous, discrete-time, discrete-valued representation of a sinusoidal waveform. This paper implements the development and design of CMOS look up Table based numerically controlled oscillator which improves the performance, reduces the power & area requirement. The design is implemented with CMOS 32 nm Technology with Microwind 3.8 software tool. In addition, it can be used for analog circuit also enables the integration of complete system on chip. This paper also describes the design of a NCO which is of contemporary nature with reasonable speed, resolution and linearity with lower power, low area. For all about Pre Layout simulation has been realized using 32nm CMOS process Technology.
This summarizes a document describing a 6-bit 200MHz flash analog-to-digital converter (ADC) with redundancy. It uses small latched comparators and redundancy to reduce offset errors. Simulations show it achieves 5.3 effective number of bits, consumes 1.56mW of power, and has a figure of merit of 0.2 pJ/conversion step, making it suitable for applications requiring low power consumption. Redundancy allows selecting the closest comparator to the ideal threshold for each level, reducing offset effects.
This document summarizes a wide-range frequency synthesizer using a phase-rotating technique for digital TV tuners. Key points:
- The synthesizer was fabricated in 0.18-μm CMOS and can provide frequencies from 55-960 MHz to satisfy ATSC specifications using an LC-VCO and band selection.
- It achieves fractional functionality using a phase-rotating concept instead of a conventional ΔΣ interpolation to reduce hardware complexity and power consumption.
- Measured results showed a phase noise of -123 dBc/Hz at 1 MHz offset and lock time below 6 μs from 640-930 MHz.
This document describes a 10-bit 165MSPS video ADC. It uses a time-interleaved SAR architecture with multiple low-power SAR ADCs operating in parallel to achieve the high sampling rate. Each SAR ADC samples every nth cycle, resulting in an effective sampling rate that is n times higher than an individual ADC. The design achieves low power consumption and area through its digital nature and use of scaled CMOS technology. It provides offset correction through a calibration technique to improve accuracy for video applications.
Designed a Switched Capacitor Low Pass Filter with a sampling frequency of 60 Hz.
Simulated the filter to have a ripple within 0.2 dB under 3.6 MHz and a stopband attenuation of atleast -51 dB after 7.2 MHz.
Applied dynamic range optimization, Dynamic Range Scaling and Chip Area scaling to get maximum output swing while occupying minimum area on chip.
Tested the filter with non-idealities of the amplifier, such as finite gain, bandwidth, offset voltage, charge injection, etc.
A low-power 2-bit/step operation technique is proposed which uses dynamic threshold configuring comparator instead of multiple digital-to-analog converters (DACs). Power and area overhead is minimized by successively activated comparators. The comparator threshold is configured by simple Vcm biased current source, which keep the ADC free from power supply variations over 10%. To implement power efficient and high performance analog-to-digital converters the designers are urged to design an optimized dual tail comparator. In this paper, It is shown that in the proposed dual tail comparator both the power and delay time is significantly reduced.
Dsp 2018 foehu - lec 10 - multi-rate digital signal processingAmr E. Mohamed
This document discusses multi-rate digital signal processing and concepts related to sampling continuous-time signals. It begins by introducing discrete-time processing of continuous signals using an ideal continuous-to-discrete converter. It then covers the Nyquist sampling theorem and relationships between continuous and discrete Fourier transforms. It discusses ideal and practical reconstruction using zero-order hold and anti-imaging filters. Finally, it introduces the concepts of downsampling and upsampling in multi-rate digital signal processing systems.
The application wavelet transform algorithm in testing adc effective number o...ijcsit
In evaluating Analog to Digital Convertors, many parameters are checked for performance and error rate.
One of these parameters is the device Effective Number of Bits. In classical testing of Effective Number of
Bits, testing is based on signal to noise components ratio (SNR), whose coefficients are driven via
frequency domain (Fourier Transform) of ADC’s output signal. Such a technique is extremely sensitive to
noise and require large number of data samples. That is, longer and more complex testing process as the
device under test increases in resolutions. Meanwhile, a new time – frequency domain approach (known as
Wavelet transform) is proposed to measure and analyze Analog-to-Digital Converters parameter of
Effective Number of Bits with less complexity and fewer data samples.
This document discusses the design and operation of an all-digital phase locked loop (ADPLL). It covers topics such as the digitally controlled oscillator (DCO) core design, noise modeling in the ADPLL, tuning the ADPLL for GSM, impairments like capacitor mismatch and compensation techniques.
The document describes the design of a maximally flat microwave filter with an attenuation of 12dB at 1.3333 times the cutoff frequency of 3.8GHz. It involves calculating the filter order and element values using lowpass prototype filter design equations. The lumped element filter is then transformed into an equivalent distributed filter using transmission line sections. Simulation shows the distributed filter deviates from the ideal lumped response at high frequencies due to neglected capacitance. Adjusting the line lengths proportionally brings the response closer to the lumped filter specification.
This document discusses analog to digital conversion fundamentals. It defines analog and digital signals and introduces analog to digital converters (ADCs) which convert continuous analog signals to discrete digital values. It discusses important ADC characteristics like resolution, quantization error, differential and integral nonlinearity. Sample and hold circuits are explained along with errors like aperture uncertainty. Specifications of digital to analog converters (DACs) like output voltage, resolution and nonlinearity errors are also covered.
1. Very low power consumption SAR ADC for wireless sensor networks
(Master Thesis Extended Abstract)
Tiago Pádua
Instituto Superior Técnico - Taguspark, Av. Prof. Doutor Cavaco Silva — 2744-016 Porto Salvo, Portugal
e-mail: tiago.padua@tecnico.ulisboa.pt
Abstract1
— This study aims to design analog-to-digital
converter based on successive approximations (SAR ADC), with
very low power consumption, in an advanced CMOS technology
(28 nm), following the industrial methodologies IC. This project
was conducted in partnership with Synopsys, operating at a
supply voltage of 0.5 V ± 10% with a resolution of 12 bits and
500 kS/s sampling frequency.
Keywords— Analog-to-Digital Converter (ADC),
Successive-Approximation Register (SAR), low power
consumption, Integrated Circuit (IC).
I. INTRODUCTION
Nowadays, there is a demand for analog-to-digital converters
(ADCs) with low power consumption, namely for wireless
sensor networks. Since is difficult and expensive to develop
new types of battery, electronic circuits need to consume less.
Over the past decade, the research projects show that the SAR
topology is suitable for this situation. Thus, the main goal of
this work is to develop an Asynchronous SAR ADC with 12
bit resolution. It aims to operate at 500 kS/s sampling
frequency, with a supply voltage of 0.5 V ± 10% and a
differential full-scale input range of 1 V. The design ought to
cover 16 PVT corners, with temperature range from -40 to 125
ºC, and a SNR above 65 dB.
The present paper is organized as follows: Section II: The
typical SAR topology and the asynchronous one (that is used)
are depict. It also lists the noise and time specifications.
Section III: Features the comparator, according to its topology,
time and noise specifications. Section IV: Describes the
Sampling DAC employed, accordingly to its topology and the
sizing of its capacitors and switches. It also mentions the
employed clock-boost. Section V: Summarizes the state
machine with a delay cell that mimics the DAC switchings.
Section VI: Addresses the required calibrations for the
converter. Section VII: Lists the conclusions and future work.
II. ADC ARQUITECTURE
The SAR ADC operates through a binary search algorithm,
estimating a digital word that represents, as closer as possible,
its input signal voltage. The word is created by testing each bit
individually, so that in each step the selected bit origins a
reference voltage (produced by the internal DAC) which will
be compared with the stored input signal voltage. This process
I present my sincere gratitude to my mentor Dr Pedro Figueiredo.
is repeated N times (ADC resolution) until the last bit
conversion.
The traditional converter operates in a synchronous way,
where each bit conversion takes the same amount of time,
regarding the slowest bit conversion. Whether an
asynchronous one has a variable time for each bit conversion,
since it only uses the necessary time so that the comparator can
make a decision. This way, the asynchronous converter is
generally faster converting a signal. It should also have a lower
power consumption, since it would allow the blocks to be
disabled sooner. Hence we selected the asynchronous way, but
its sizing is made the same way as a synchronous.
The synchronous operation requires a high-frequency clock
signal which based on the converter’s resolution (𝑁) limits the
sampling frequency (𝑓𝑆) as
𝑓𝑆[𝐻𝑧] =
𝑓𝐶𝐿𝐾
𝑁 + 1
, (1)
where 𝑓𝐶𝐿𝐾 is the required clock signal frequency and 𝑁 is the
number of clocks necessary for every bit conversion, plus one
clock for sampling. The maximum clock frequency (𝑓𝐶𝐿𝐾𝑚𝑎𝑥)
must ensure that all bits are converted, so its estimation must
consider the time of the slowest bit conversion for every bit
conversion. Therefore the 𝑓𝐶𝐿𝐾𝑚𝑎𝑥 is determined based on the
worst (slowest) comparison time per bit (𝑡 𝐶𝑂𝑀𝑃𝑤𝑜𝑟𝑠𝑡), the
DAC settling time (𝑡 𝐷𝐴𝐶) and the sampling time (𝑡 𝑆), yielding
𝑓𝐶𝐿𝐾𝑚𝑎𝑥[𝐻𝑧] =
1
𝑁(𝑡 𝐶𝑂𝑀𝑃𝑤𝑜𝑟𝑠𝑡 + 𝑡 𝐷𝐴𝐶) + 𝑡 𝑆
. (2)
Since, at extremely low supply voltage, the transistors of the
comparator and sampling DAC operate at sub-threshold
region, the switches ON-resistance is high, originating a large
𝑡 𝐶𝑂𝑀𝑃𝑤𝑜𝑟𝑠𝑡, 𝑡 𝐷𝐴𝐶 and 𝑡 𝑆.
The SAR ADC described in this work is composed by five
main blocks, as depicted in Fig. 1. At first, the differential
input signal is sampled by the Sampling DAC (controlled by a
synchronous external clock), and afterwards the stored signal
will be successively processed in an asynchronous way. The
comparator decides, based on the differential output voltage of
the DAC, returning 𝑞 (and its inverse value) to the SAR block
(which stores the bit decision in a register) and to the delay
block, which generates an internal (asynchronous) clock. This
internal clock ensures, for every bit conversion cycle, a
constant time for settling the voltages of the DAC capacitors
and a variable time for each comparator bit decision. After all
bits are converted, the final word is synchronously delivered
2. to the output, the sampling phase is restarted and the cycle
repeats.
Sampling
DAC
SAR
VREF±
[d13,...d0]
VDACP
VDACN
delay
q
q
bout
CLKsynch
CLKasynch
Resistive
ladder
vIN
vIP
vDD
Fig. 1. Implemented SAR DAC block diagram.
This ADC was sized regarding as top guidelines, the time
and the noise constraints. We attributed most of the clock
period for conversion and the remaining for sampling. In turn,
the noise restriction was used to limit the noise that each block
was allowed to produce. Thus, limiting the value of the noise
capacitor of the comparator and the unit capacitor of the DAC.
A. Time constraints
The time operations use 𝑓𝑆 = 500 𝑘Hz as guideline, which
makes a clock period of 2 μs. Then, missing 10% of the clock
period to input signal sampling (𝑡 𝑠𝑎𝑚𝑝), the remaining 90% are
used for conversion (𝑡 𝑐𝑜𝑛𝑣), resulting in
𝑡 𝑠𝑎𝑚𝑝 = 0.1𝑇𝐶𝐿𝐾 = 200 ns
𝑡 𝑐𝑜𝑛𝑣 = 0.9𝑇𝐶𝐿𝐾 = 1800 ns
. (3)
In turn, stipulating that each word conversion takes 12
decisions (given the resolution) plus 2 more due to
redundancy, makes 128 ns for every bit conversion. Since that
time comprises the settling time of the DAC (𝑡 𝐷𝐴𝐶) and the
time for the comparator to decide (𝑡 𝑐𝑜𝑚𝑝), we attributed
{
𝑡 𝐷𝐴𝐶 = 0.55𝑡 𝑏𝑖𝑡 = 70 ns
𝑡 𝑐𝑜𝑚𝑝 = 0.45𝑡 𝑏𝑖𝑡 = 58 ns
. (4)
B. Noise constraints
Besides the quantization noise present in ideal ADCs, real
ADCs also have noise due to internal components and external
factors. Yet, the dominant contribution corresponds to thermal
noise, which is given by 𝑘𝑇 𝐶⁄ . The noise produced comes
from the three typical blocks, meaning from sampling
(𝑣 𝑛,𝑠𝑎𝑚𝑝
2̅̅̅̅̅̅̅̅̅), quantization (𝑣 𝑛,𝑄
2̅̅̅̅̅) and the comparator (𝑣 𝑛,𝑐𝑜𝑚𝑝
2̅̅̅̅̅̅̅̅̅).
We can distribute the noise using the SNR expression, as
𝑆𝑁𝑅 = 10 log10 (
𝑃𝑠𝑖𝑔𝑛𝑎𝑙
𝑣 𝑛,𝑄
2̅̅̅̅̅ + 𝑣 𝑛,𝑠𝑎𝑚𝑝
2̅̅̅̅̅̅̅̅̅ + 𝑣 𝑛,𝑐𝑜𝑚𝑝
2̅̅̅̅̅̅̅̅̅
). (5)
In turn, we can relax the comparator and sampling noise by
selecting a 12 bit resolution, obtaining 𝑣 𝑛,𝑄
2̅̅̅̅̅ = (70.5 μV) 2
.
Then, using the specified SNR and assuming that the
comparator noise is the same as the sampling noise, results in
𝑣 𝑛,𝑠𝑎𝑚𝑝
2̅̅̅̅̅̅̅̅̅ + 𝑣 𝑛,𝑐𝑜𝑚𝑝
2̅̅̅̅̅̅̅̅̅ = (186 μV)2
.
Ergo, 𝑣 𝑛,𝑠𝑎𝑚𝑝
2̅̅̅̅̅̅̅̅̅ = 𝑣 𝑛,𝑐𝑜𝑚𝑝
2̅̅̅̅̅̅̅̅̅ ≃ (131.5 μV)2
. Consequently,
knowing the sampling noise it is possible to determine the
overall sampling capacitance using
III. COMPARATOR
A dynamic comparators only consumes energy during the
decision phase, being more power efficient than a high gain
operational amplifier. Therefore, a dynamic comparator is
used, based on [21], where part of its circuit is illustrated in
Fig. 2. It consists of a differential pair – 𝑀1𝑃 and 𝑀1𝑁 – (that
charge 𝐶 𝐹 making an integrator) and a pair of back to back
inverters – 𝑀4𝑃, 𝑀6𝑃, 𝑀7𝑃 and 𝑀4𝑁, 𝑀6𝑁, 𝑀7𝑁 – (or cross
coupled inverters). Moreover, the integrator was sized to place
the transistors of the differential pair in the sub-threshold
region (namely in the weak inversion region); combined with
a sufficient current supply that obey with the temporal goals.
M2N
M3NM3P
M1P
VDD
M2P
M1N
latch
latch
vINvIP
CFP CFN
Differential
pair
vFP vFN
VDD
M7P M7N
M5P M5N
M4P M4N
M6NM6P
latchlatch
Back
to back
inverters
vOP vON
vS
Fig. 2. Part of the dynamic comparator schematic.
The comparator circuitry is designed firstly ensuring the
noise restriction and secondly the time, which comprises the
phases of integration and regeneration, denoted in Fig. 3. To
do so, a theoretical model is required to instruct a first
approach and then simulate the circuit. Hence, the study from
[18], regarding the comparator modeling and transient
behavior at the regeneration phase, and [22], [23], regarding
the transconductance model for the MOSFET in the weak
inversion region, are combined.
latch
L
H
vFP,N
L
H
vOP
L
H
vON
...
...
...
...
...
t
...
Integration time
Regeneration time
ResetReset
Propagation
time
Fig. 3. Integration and regeneration phases.
A. Theoretical first approach
The integrator switches (𝑀1𝑃/𝑀1𝑁 and 𝑀2𝑃/𝑀2𝑁) were
designed to be in the sub-threshold region (preferably in the
𝐶𝑠𝑎𝑚𝑝 =
8
3
𝑘𝑇
𝑣 𝑛,𝑠𝑎𝑚𝑝
2̅̅̅̅̅̅̅̅̅
≃ 688 fF. (6)
3. weak inversion), where in a slow rate operation the current
consumption is best leveraged. Then, the (noise) capacitors
(𝐶 𝐹) are sized in order to comply with the comparator noise
restriction. Hence, assuming, that the comparator output noise
is mainly given by the noise produced from the comparator
integration nodes, as stated in [18], yields
𝜎2
(𝑣𝑜(𝑡)) ≈ 𝜎𝑖𝑛𝑡
2
(𝑡) =
4𝑘𝑇𝛾𝑔 𝑚
𝐶 𝐹
2 𝑡. (7)
In turn, as illustrated in Fig. 4, the comparator input referred
noise –𝜎 𝑛𝑖
2
(𝑡) – is quantified by
𝜎 𝑛𝑖
2
(𝑡) =
𝜎2
(𝑣𝑜(𝑡))
𝐺2(𝑡)
≈
4𝑘𝑇𝛾
𝑔 𝑚 𝑡
, (8)
where 𝐺(𝑡) is the integration voltage gain that neglecting 𝑔 𝐷𝑆1
is given by
𝐺(𝑡) =
𝑣 𝐹 𝑑𝑖𝑓𝑓(𝑡)
𝑣 𝑑
=
𝑔 𝑚1
𝐶 𝐹
𝑡, (9)
and in the end of the integration is
𝐺(𝑡𝑖𝑛𝑡) = 2𝑔 𝑚1
Δ𝑉𝐹
𝐼𝑆𝑆
=
𝑔 𝑚1
𝐼 𝐷
∆𝑉𝐹. (10)
G
σ2
σ2
ni
Comparator
vI
Fig. 4. Comparator output and input referred noise model.
Furthermore, one can relate the transistors size through its
current, which is given, in the weak inversion region, as [22]
𝐼 𝐷 =
𝑊
𝐿
𝐼 𝑀 𝑒𝑥𝑝 (
𝑉 𝐺𝑆−𝑉 𝑀
𝑛𝜙 𝑡
) [1 − 𝑒𝑥𝑝 (
−𝑉 𝐷𝑆
𝜙 𝑡
)], (11)
where 𝐼 𝑀 and 𝑉 𝑀 are the current and voltage for the
representative point that ends the weak inversion. Moreover,
knowing the transconductance in this region as
𝑔 𝑚 =
𝐼 𝐷
𝑛𝜙 𝑡
⟺ 𝐼 𝐷 = 𝑔 𝑚 𝑛𝜙 𝑡, (12)
where 𝑛 is a transistor intrinsic parameter and 𝜙 𝑡 is the thermal
voltage,
𝜙 𝑡[𝑚𝑉] = 25.9 × (T/300), (13)
if we substitute the equation (10) and (12) in (8), we get the
comparator input referred noise at the integration time as
𝜎 𝑛𝑖
2
(𝑡𝑖𝑛𝑡) ≈
4𝑘𝑇𝛾
𝐶 𝐹
𝐼 𝐷
𝑔 𝑚∆𝑉𝐹
=
4𝑘𝑇𝛾
𝐶 𝐹
𝑛𝜙 𝑡
∆𝑉𝐹
. (14)
This way, we can determine a guideline value to size 𝐶 𝐹,
using the noise restriction as the input referred noise, meaning
𝜎 𝑅
2
= 𝜎 𝑛𝑖
2
= (131.5 μV)2
, and 𝑇 = 398°K (125℃), 𝑛 = 1.6,
∆𝑉𝐹 = 300 mV and 𝛾 = 1, through (14) we obtain the
theoretical value as
𝐶 𝐹 =
4𝑘𝑇𝛾
𝜎 𝑛𝑖
2 (𝑡𝑖𝑛𝑡)
𝑛𝜙 𝑡
∆𝑉𝐹
≈ 230fF. (15)
B. Time restrictions
Once stablished the necessary current supply, the size of the
differential pair is adjusted by simulation to comply with the
integration time restriction, taking in consideration the relation
of 𝑊 𝐿⁄ from equation (11). Likewise, the transistors of the
back to back inverters are adjusted with the remaining time for
the regeneration phase.
However during the complete ADC sizing, was required to
increase the comparator input common mode voltage, what
consequently delayed the comparator response, causing the
slowest corner to fail the time constraint. So, instead of
resizing the comparator, we aggregated an extra capacitor to
the differential pair (as depicted in Fig. 5) that supplies more
current and, thus, accelerating the integration phase.
vS
CL
φ1 φ2'
φ1
Cap turbo
φ2
M1PvIP
CFP
M2P
latch
vFP
VDD
VDD
Fig. 5. Comparator turbo capacitor that speeds up the
integration stage.
This solution is proven for the worst case as illustrated by
Fig. 6, when the turbo is disabled, and by Fig. 7, when it is
enabled.
Fig. 6. Transient simulation of comparator operation in
corner 14 (slowest), having 73uV at input, with turbo OFF.
Fig. 7. Transient simulation of comparator operation in
corner 14 (slowest), having 73uV at input, with turbo ON.
C. Noise restriction
Besides complying the operating times of the comparator
with the time restriction, the noise restriction must be
confirmed. So, a transient noise simulation is performed for all
corners, where it is possible to obtain the standard deviation
value and, consequently, the estimated comparator output
noise. This is done through a stochastic process by triggering
the comparator 500 times with a maximum noise frequency of
15 GHz, at a fixed input signal of 1 mV, whose standard
deviation value (i.e. the output referred noise) is given by [18]
𝜎2
(𝑋(𝑡)) = 𝐸(𝑋(𝑡)2) − 𝐸(𝑋(𝑡))
2
, (16)
4. where 𝑋(𝑡) is a Gaussian distributed stochastic process and
𝐸(𝑋(𝑡)) is its expected value, which is equivalent to the
relation between the differential output voltage root mean
square (𝑅𝑀𝑆) and its average (𝐴𝑉𝐺) values as
𝜎(𝑋(𝑡)) = √𝑅𝑀𝑆2 − 𝐴𝑉𝐺2. (17)
In turn, we can express the comparator input referred noise
as determined by [18] [24]
𝜎 𝑛𝑖(𝑡) = 𝜎(𝑋(𝑡)) 𝐺 𝑒𝑓𝑓(𝑡)⁄ , (18)
where 𝐺 𝑒𝑓𝑓(𝑡) is the effective gain of the regeneration nodes
at a given sampling instant (as illustrated in Fig. 8). Then,
applying these concepts for the same selected corner as before,
yields Table I and Table II, respectively without and with
turbo, which confirm that the produced noise is below the
noise constraint (𝝈 𝑹), within a sufficient margin.
Table I - Comparison of the produced input referred noise
between main corners, with turbo OFF.
Input referred noise from main corners with turbo OFF
File_tr 0 5 7 14
V 0,5 0,45 max Min
T 50 max max Min
Corner Typ Noisy Faster Slowest
𝑹𝑴𝑺 [𝐦𝐕] 50.4 50,96 50.84 51,04
𝑨𝑽𝑮 [𝐦𝐕] −50.25−50.57 −50.52 −50,97
𝝈 [𝐦𝐕] 4 6.3 5.7 2.8
𝑮𝒆𝒇𝒇 50.1 50.2 50.8 50.2
𝝈 𝒏𝒊 [𝛍𝐕] 79.3 𝟏𝟐𝟔 111 56.1
𝝈 𝑹 [𝛍𝐕] 𝟏𝟑𝟏, 𝟓
Margin: 𝝈 𝑹 − 𝝈 𝒏𝒊 [𝛍𝐕] 52,2 𝟓, 𝟓 20,5 75,4
Table II - Comparison of the produced input referred noise
between main corners, with turbo ON.
Input referred noise from main corners with turbo ON
File_tr 0 5 7 14
V 0,5 0,45 max Min
T 50 max max Min
Corner Typ Fast Faster Slowest
𝑹𝑴𝑺 [𝐦𝐕] 51,37 50,8 51,6 50,1
𝑨𝑽𝑮 [𝐦𝐕] −51.22 −50,41 −51,18 −50,07
𝝈 [𝐦𝐕] 3,87 6,27 6,52 2,05
𝑮𝒆𝒇𝒇 50,45 50,07 50,88 50,2
𝝈 𝒏𝒊 [𝛍𝐕] 76,8 𝟏𝟐𝟓 𝟏𝟐𝟖 41
𝝈 𝑹 [𝛍𝐕] 𝟏𝟑𝟏, 𝟓
Margin: 𝝈 𝑹 − 𝝈 𝒏𝒊 [𝛍𝐕] 54,7 𝟔, 𝟓 𝟑, 𝟓 90,5
Fig. 8. Transient noise simulation of the comparator in corner
5, having turbo ON, with and without noise for a fixed input
voltage of 1 mV, acquiring 500 samples with a noise
frequency of 15 GHz and static gain of 50.
D. Kickback noise
The kickback effect or noise [25] is any disturbance coming
from an internal stage that causes a re-action to the previous
stage. The kickback might not be a critical issue for general
signals; however, when we are dealing with a small signal, in
the comparator input, it makes the difference between a right
and bad decision. The Fig. 9 shows half of the differential pair,
where we can state that this type of disturbance propagates to
the input terminal, mainly from the current flowing in the
capacitances (𝐶 𝐺𝑆 and 𝐶 𝐺𝐷) inherent to the transistor. It can
also be originated by charge variations at the gate of 𝑀1𝑃,
when it changes the operating region (cut-off, triode and
saturation). Note that these currents exist because 𝑣 𝑆 and 𝑣 𝐹𝑃
vary.
CFP
M2P
latch
vFP
vS
CGD
CGS
icGS
icGD
vIP M1P
Cb
Ca
vS_copy
vFP_copy
Cturbo
VDD
VDD
M4N
M4P
VDD
M3N
M3P
VDD
M6N
M6P
VDD
M5N
M5P
latchin
Fig. 9 - Half of the comparator differential pair with parasitic
and counter-kickback mechanism.
Having the 𝑀1𝑃 transconductance constant, during the
integration stage, the solution employed to minimize this effect
goes through the reproduction of the reverse behavior caused
by the parasitic capacitances. Hence, we add two capacitors to
the input node, both with a similar value of the MOS parasitic
capacitance and on the opposite terminal plate it is applied a
signal that tries to replicate an inverted phase of the respective
node (namely 𝑣 𝑆_𝑐𝑜𝑝𝑦 and 𝑣 𝐹𝑃_𝑐𝑜𝑝𝑦). The Fig. 10 and Fig. 11
illustrates the result of the kickback effect, during the LSB
decision (𝑠𝑡_0), without and with the counter-kickback (𝐶𝐾)
solution, while the comparator turbo mode is OFF and ON
respectively. We can observe that the DAC differential output
voltage - 𝑣 𝐷𝐴𝐶 (or the comparator input voltage), during the
comparator integration phase (depicted by the rise of 𝑣𝑓𝑝),
have a more steady behavior with this solution than without it.
5. Although, the simulation without turbo has a more clear
impact of the effect reduction than the one with turbo, the
important is to ensure that 𝑣 𝐷𝐴𝐶 has less deviation until the
regeneration phase begins. In both cases, the node common
voltage,
𝑣 𝐶𝑀 =
𝑣 𝐶𝑀 𝐷𝐴𝐶𝑃 + 𝑣 𝐶𝑀 𝐷𝐴𝐶𝑁
2
, (19)
maintains its behavior, but without turbo 𝑣 𝐷𝐴𝐶 reduces its
oscillation (in comparison with the case without the
counter-kickback solution) approximately 4 times. In turn,
with turbo we have 𝑣 𝐷𝐴𝐶 almost flat during most of the
integration stage.
Fig. 10. Comparator input without and with counter-kickback
(CK) effect, in typical corner, having turbo OFF.
Fig. 11. Comparator input without and with counter kickback
(CK) effect, in typical corner, having turbo ON.
IV. SAMPLING DAC
The sampling DAC was made with two capacitive arrays,
having one array illustrated in Fig. 12. Instead of a typical
array, this one uses pseudo-binary scaled due to use of
redundancy. The redundancy allows a recover of eventual
wrong bit decision, which is only possible when the
comparator takes more than one decision over a range of
voltages, where the search algorithm has already passed.
CP2
VCMcomp
CB
vDACP LSB
VRefN VRef/2 VRefP
1.5 1.11
CP1
vIP
vDACP
VRef/2VRefN VRefP
MSB
C12C11C10C9C8C7C6C5C4C3C2C1C0
ISBLSB
VRef/2VRefN VRefP
VCMcomp
vDACN
Fig. 12. One branch of the Sampling DAC block.
As stated before, this ADC is conceived to have a 12-bit
resolution, for that it demands 12 decisions. However, 2 more
decision are added to make redundancy, and rectify wrong
MSB decisions, when determining the inferior decision [26].
For that to happen, we use 12 capacitors plus 1 (for digital
calibration), employing an array split in 3 groups – MSB, ISB
and LSB, as shown in Fig. 12, our redundancy is created by
multiplying the capacitors of the last two groups with scale
factors.
Attached to each capacitor, there is a block of switches that
need to ensure a minimum error for sampling as for
conversion. Thus, the switches need to have a small ON
resistance, such as it allows to commute the capacitor bottom
plate to a selected reference voltage within the required settling
times. It should also be small to minimize the voltage drop at
its terminal (preventing different reference voltages). There is
a mutual dependency between the capacitors and their
respective switches to ensure this settling. So, we first sized
the capacitors, accordingly to its noise constraint, and then, we
size the switches to meet the times with minimal error.
In order to facilitate these commutations and reduce the
power consumption, every capacitor in the DAC array are
placed to a mid-scale value (𝑉𝑅𝐸𝐹 2⁄ ), when the sampling ends.
For that to happen each capacitor is split in half and each one
is put at different reference voltages. Thus, after sampling, the
positive reference voltage (𝑉𝑅𝐸𝐹𝑃) is applied directly to one
half of 𝐶 𝑢𝑛𝑖𝑡 (𝐶 𝑎) and the negative reference (𝑉𝑅𝐸𝐹𝑁) to the
other half (𝐶 𝑏), i.e.
{
𝐶 𝑎 + 𝐶 𝑏 = 𝐶 𝑢𝑛𝑖𝑡 ⟺ 𝐶 𝑎 = 𝐶 𝑏 = 𝐶 𝑢𝑛𝑖𝑡 2⁄
(𝑉𝑅𝐸𝐹𝑃 − 𝑉𝑅𝐸𝐹𝑁) 𝐶 𝑎 (𝐶 𝑎 + 𝐶 𝑏) = 𝑉𝑅𝐸𝐹 2⁄⁄
. (20)
A. Sampling DAC Capacitors
The first step to size these capacitors is to determine the unit
capacitor (𝐶 𝑢𝑛𝑖𝑡), which is defined by the ratio of the total
sampling capacitance (𝐶𝑆𝑎𝑚𝑝), from (6), and their binary
weighted sum. Thus, choosing the contribution of ISB and
MSB group capacitors for sampling (stated in Table I), the
binary weighted sum is given by 256.65, which results in
Secondly, using a bridge capacitor in the middle of the
capacitor array, they were sized accordingly to their group.
This bridge capacitor solves size discrepancies, allowing the
LSB capacitors to be larger, being less affect by parasitic
capacitors, as well as the MSB to be smaller. In a third place,
our redundancy is created by multiplying the capacitors of the
last two groups with scale factors. The capacitors of the MSB
group are binary weighted scaled, since they have a
multiplicative factor of 1. However to create redundancy, the
capacitors of the ISB group have a multiplicative factor of 1.11
and LSB have a factor of 1.5. While the ISB have
multiplicative factor in their capacitance value, the LSB
simply increased and their scale factor is produced by
adjusting the size of the bridge capacitor.
𝐶 𝑢𝑛𝑖𝑡 =
𝐶𝑆𝑎𝑚𝑝
256.65̅̅̅̅̅̅̅̅̅
≃ 2.7 fF. (21)
6. Table III - Index and binary weight of the Sampling DAC
capacitors
Group C bit index C bit weight
MSB 𝐶9, 𝐶10, 𝐶11, 𝐶12 16𝐶 + 32𝐶 + 64𝐶 + 128𝐶
ISB 𝐶5, 𝐶6, 𝐶7, 𝐶8 1.11𝐶 + 2.22𝐶 + 4.44𝐶 + 8.88𝐶
LSB 𝐶0, 𝐶1, 𝐶2, 𝐶3, 𝐶4 0.25𝐶 + 0.5𝐶 + 𝐶 + 2𝐶 + 4𝐶
Samp 𝐶5 to 𝐶12 256.65C
B. Sampling DAC Switches
After sizing the switches for the MSB capacitor, respecting
the time constraints, the following switches were scaled
binary. However due to leakage current their topology had to
be customize as depicted in Fig. 13. In addition, every
capacitor have their top plate connected to a shared node
(𝑉𝐷𝐴𝐶𝑃) that is linked to other two switches, the comparator
biasing switch (𝐶𝑀) and the one that short-circuits the
comparator input (𝑆𝐶).
Ca
Cb
MP1
MN1
MN3
φSAMP
MN2
φSAMP
MP2
vCb
MPSW
MNSW
VREFN
MNSS
φSS
φSAMP φSAMP
MP1
MN1
MN3
φSAMP
MN2
φSAMP
MP2
vCa
MPSW
MNSW
VREFN
φCON V
φCON V
MPSS
φSAMP φSAMP
φSS
vIN
φCON V
φCON V
MDUMN
MDUMP
Fig. 13. Sampling DAC switches block for one bit-capacitor
(split in 𝐶 𝑎 and 𝐶 𝑏) from side 𝑉𝐷𝐴𝐶𝑃 of the array.
Here, instead of using just one NMOS as a sampling switch,
there are five in a T-scheme. Both 𝑀 𝑁1 and 𝑀 𝑁2 correspond to
the sampling switch, whether 𝑀 𝑁3 creates a path for current
leakage that is caused by the input signal during the conversion
phase. Moreover, the PMOS added in parallel are used to help
the sampling connectivity for larger input signals, since their
inner resistance behaves inversely to the NMOS.
Now, we illustrate in Fig. 14 the Sampling DAC switches
during sampling, focusing just one of the capacitors in that
situation. Now, the operating transistors behave as resistors,
namely the sampling (𝑀 𝑁1, 𝑀 𝑁2) and biasing switches of the
DAC, and the ones that are cut-OFF behave as current sources,
namely the rest of the switches (𝑀 𝑁3, 𝑀 𝑃𝑆𝑊, 𝑀 𝑁𝑆𝑊, 𝑀 𝐷𝑈𝑀𝑃,
𝑀 𝐷𝑈𝑀𝑁, 𝑀 𝑃𝑆𝑆 and 𝑀 𝑁𝑆𝑆), due to their continuous leakage
current.
Likewise, we represent in Fig. 15 the situation during the
conversion phase, where, now, only the reference switches
(𝑀 𝑃𝑆𝑊, 𝑀 𝑁𝑆𝑊) behave as resistors and the rest as current
sources. The 𝑀 𝑁3 will drain the leakage interference from the
input signal and the 𝑀 𝐷𝑈𝑀𝑃/𝑀 𝐷𝑈𝑀𝑁 will try to counter-balance
the leakage from the reference switch that is not selected.
φSAMP - Sampling phase
VDD
Cbit
VCM
vIN
vDACP
vDACN
RSC
RCM
Cp
RMN1 RMN2
vIP
Ileak NSW/DU MP
Ileak N3
Remaining
CDACPIleak PSW/DUMP
Fig. 14. Sampling DAC switches during sampling phase.
φCON V - Conversion phase
Cbit
VCM
vIN
Cp
vDACP
RN3
RPSW
VDD
VDD
CbitvIP
Cp
vDACN
RN3 RNSW
VCM
VREFP
VREFN
Ileak N1
Ileak DUMP
Ileak N2
Ileak NSW/DU MN
Ileak PSW/DU MP
Ileak N2Ileak N1
Ileak DUMN
Remaining
CDACP
Remaining
CDACN
Ileak SC
Ileak CM
Ileak CM
Fig. 15. Sampling DAC switches during conversion phase.
In addition, the switches used during sampling required a
better biasing voltage to ensure smaller resistance and quicker
settling times. For that matter we employed the clock-boost
illustrated in Fig. 16, which basically combines the sampling
input voltage with a percentage of the circuit voltage supply.
MN7
MP4
MN5
MP5MP3
vBST
MP2
clk vBST
MN4
vIN
MCL
MN3
clk
MP6
MN6
clk
MP1
MN1
MN2
vBST
nx
top
bott
nx
clk
Fig. 16. Clock-boost circuit schematic.
V. STATE MACHINE
Although the ADC has an asynchronous behavior, it is only
resultant of the different time of each bit conversion. In fact
after each bit decision the comparator is disabled for a fixed
time (that allows the voltage of the capacitors to stabilize). The
state machine (illustrated in Fig. 17) assures this fixed time
7. thanks to a custom delay cell, which mimics the behavior of
the DAC switching references (depicted in Fig. 18).
pulseCLKext
CLKstate
qd
q
en
q
cmprdy
cmprdy
next
VDD
latch
st_samplepulse
pulse
cmprdy
next
VDD
st_sample
st_sample
st_vos
DAC mimic
delay cell
Shrink
pulse
st_reset
Fig. 17. Delay block logic circuit representation.
DAC mimic delay cell
MN1
Ca
MP2
MN3
Vrefp
Cb
MN4
MP4
Vrefn
p1g
p2g n2g
n1g
VDD
p1g dac_rdy
set rst
rst set
MP3
Vrefn
MN2
cap
cbn
MP1
Vrefp
Fig. 18. Delay cell that mimics the DAC switchings.
This state machine is centralized in a delay block that,
depending on the comparator outcome, creates the control
signals of the comparator (𝑙𝑎𝑡𝑐ℎ̅̅̅̅̅̅̅) and the ring counter
(𝐶𝐿𝐾𝑠𝑡𝑎𝑡𝑒). It produces a total of 17 consecutive states of the
per period, as shown in Fig. 19.
Previous conversion Next conversionSampling
CLKext
...
...
...
...
...st_13
st_0 ...
st_vos ...
st_reset
st_sample
ph_sc
...CLKstate
...latch
Fig. 19. ADC sequence example of sampling and conversion
states for a slow conversion case.
VI. CALIBRATIONS
About the non-linearity issues we can state that their origin
is mainly from the sampling DAC block, due to capacitors
mismatches. However, it can be mitigated by a digital
calibration. On the other hand, the comparator might insert an
offset voltage, which can cause wrong bit decisions. Although
those possible errors could be recovered by redundancy, the
digital calibration that we intended to use requires a
comparator with low offset voltage.
The static non-idealities of linear circuits are decomposed in
offset, gain errors and non-linear errors, so ADCs are
inherently non-linear, but they can be modeled as a linear
system, by highlighting the sources of those errors in the
circuit. Thus, the Fig. 20 pinpoints the gain errors
(𝐺 𝐸𝑠ℎ, 𝐺 𝐸𝑑𝑎𝑐) in the ADC, which estimated through a digital
calibration (of the sampling DAC) can fix the output digital
word (i.e. the ADC transfer function, regardless of the DAC).
vI SAR
ddac
qSampling
DAC
Vsdac
Ideal
ADC
SAR
ddac
q
Vsdac
bout
GEsdac
bout Calib
GEsdac
~
Real
ADC
Real
ADC Digital
calibration
DAC
vI S/H
GEdac
-+GEsh
Vdac
Vsh
Fig. 20. Sources of non-linearity and gain errors solved with
digital calibration.
Likewise, the Fig. 21 pinpoints the offset error in the ADC,
which can be referred just as one source at the ADC inputs,
however, since the sampling DAC is treated with a digital
calibration, the real impact of this error falls upon the
comparator decision, hence the ADC only requires an offset
voltage calibration for the comparator.
vOS (SH+DAC+COMP)
vOS COMP
SARvI
ddac
qSampling
DAC
VDAC
vOS SH
vOS DAC bout
SARvI
ddac
qSampling
DAC
VDAC
bout
Real
ADC
Ideal
ADC
Real
ADC
Fig. 21. Offset voltage source.
VII. CONCLUSIONS AND FUTURE WORK
The challenge of this project was to create a functional SAR
ADC with transistors supplied at 0.5 V, though their nominal
voltage were 0.9 V. Briefly, it turns difficult to obtain low
ON-resistance switches and to contain the significance of
leakage current. Thus, it results in slow switchings and
increased difficulty to contain the charge in the nodes.
A. Achievements
The comparator was aimed to operate at a maximum sample
rate of 500 kS/s, based on 16 corners plus the typical one.
However, corner 2 had to be discard, since it implies a scenario
that requires around 4 times larger sizes of the transistors than
the second slowest corner (14). Even so, this case was only
8. ensured by using a turbo mode that adds an extra capacitor to
integration node of the comparator. This provides a higher
voltage than the supply, accelerating the integration phase and,
thus, the overall conversion time. Although, this method
brings the advantage of internally producing a voltage higher
than the supply (without requiring other external voltage), at
the expense of more occupied area. It requires a greater
complexity in its control circuit as well as in the adjustment of
the counter-kickback effect.
The sampling DAC required a custom topology for its block
of switches, in order to mitigate and balance the leakage
current, when those switches are OFF. In addition,
clock-boosts are employed to drive the gates of the transistors
used for sampling.
The last efforts in this work were fixed in an attempt to solve
the offset calibration with an analog calibration. Unfortunately
the explored topology was successful due to leakage current.
Revising the state of art works, they suggest a re-sizing of the
comparator with transistors with bigger dimension. Since the
transistors become less susceptible to offsets at the cost of
increasing their consumption. However, as seen in this project,
enlarging the transistors sizes is not enough to solve this
matter. Hence it leads to believe that, to maintain the explored
topology, we should use transistors with less leakage current,
such FinFets. Or else apply another topology more complex.
Revising the primary goal of this project, it was not
complied. Since top simulations have not been acquired, which
allow the ADC characterization, the designed converter cannot
be proved to be operational. Nor a proper solution for the
comparator offset calibration was found. Nonetheless, we can
point out the difficulties faced, between corners, to size each
block of the ADC, with some suggestion for topologies to deal
with the significant leakage current.
B. Future Work
Improve the comparator turbo mechanism.
Simulate and re-size the blocks with non-ideal reference
voltages;
Optimize the delay chains, instead of using simple chains
of inverters with higher length values and re-adjust the
DAC delay cell;
Optimize the circuit that generates the driving phases of
the switches, namely the short-circuit pulse before the
offset calibration state;
Improve the comparator counter kickback voltage
mechanism by creating a more complex logic that mimics
better the integration voltage node, and also adjusting in
two operating modes for using turbo ON and OFF;
Explore a better (and functional) calibration topology for
the comparator offset voltage.
Design and simulate the respective layouts of every
block.
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