Rfadc
Rfadc
A Dissertation
by
DOCTOR OF PHILOSOPHY
May 2006
A Dissertation
by
DOCTOR OF PHILOSOPHY
Approved by:
May 2006
ABSTRACT
B.E. (Hons.), Birla Institute of Technology and Science (BITS), Pilani, India;
to-digital (A/D) conversion of the radio frequency (RF) signals and running
slight variation of this architecture is the software defined radio architecture in which the
A/D conversion is performed on intermediate frequency (IF) signals after a single down
conversion.
The first part of this research deals with the design and implementation of a
fourth order continuous time bandpass sigma-delta (CT BP ) ADC based on LC filters
for direct RF digitization at 950 MHz with a clock frequency of 3.8 GHz. A new ADC
architecture is proposed which uses only non-return to zero feedback digital to analog
converter pulses to mitigate problems associated with clock jitter. The architecture also
has full control over tuning of the coefficients of the noise transfer function for obtaining
iv
the best signal to noise ratio (SNR) performance. The operation of the architecture is
examined in detail and extra design parameters are introduced to ensure robust operation
of the ADC. Measurement results of the ADC, implemented in IBM 0.25 m SiGe
and 1 MHz, respectively, around 950 MHz while consuming 75 mW of power from
1.25 V supply.
The second part of this research deals with the design of a fourth order CT BP
ADC based on gm-C integrators with an automatic digital tuning scheme for IF
digitization at 125 MHz and a clock frequency of 500 MHz. A linearized CMOS OTA
architecture combines both cross coupling and source degeneration in order to obtain
good IM3 performance. A system level digital tuning scheme is proposed to tune the
ADC performance over process, voltage and temperature variations. The output bit
stream of the ADC is captured using an external DSP, where a software tuning algorithm
tunes the ADC parameters for best SNR performance. The IF ADC was designed in
TSMC 0.35 m CMOS technology and it consumes 152 mW of power from 1.65 V
supply.
v
DEDICATION
ACKNOWLEDGMENTS
experience and I would like to thank a lot of people who have helped me through various
stages of my career.
I would like to thank my advisor Dr Jose Silva-Martinez, who has been a great
limited knowledge in analog and mixed signal design owing to my prior computer
science background during my Bachelors program. The endless doubts from my side
never bothered him and he patiently answered all my questions, which was one of the
main reasons I decided to continue my PhD under his supervision. The long discussions
with him have helped me to understand different aspects of analog design and I am
greatly indebted to him for all his guidance during my graduate studies.
Takis Zourntos, Dr Shankar Bhattacharya and Dr Vinod Srinivasan, for agreeing to serve
The graduate control theory course that I took under Dr Bhattacharya was very
interesting and I would like to thank him for discussions on possible application of non-
linear control theory to my research work. The courses that I took under other professors
vii
in our AMSC group helped me to learn various aspects of IC design. Special mention
and thanks goes to Dr Ugur Cilingiroglu - his courses on basic analog IC design,
semiconductor devices and MOS modeling helped me tremendously because of his clear
explanation of all basic concepts. Our group secretary, Ella Gallagher, has always been a
I would like to thank all my colleagues in the AMSC group for the great
discussions and help throughout my stay. Special mention goes to Chinmaya Mishra
who helped with the design of digital blocks for the RF ADC project and to Xuemei Liu
(from high school to graduate school) who have helped me grow in both technical and
personal aspects. Special mention goes to Mrs Shyla George and Mr C Subramanian
(both high school teachers) and to Dr Santosh Kumar (professor during my Bachelors
Im always grateful for the love and support from my family. One person who
has made the greatest impact in my life is my late grandmother Shakuntala. I would
like to thank my parents and my sister for their constant love, encouragement and their
belief in me, which has been a great source of inspiration throughout my graduate
studies. I would like to thank my wife Preethi, who is always there for me with her
unconditional love and a smiling face all the time. She will always be the shining light in
TABLE OF CONTENTS
Page
DEDICATION ...................................................................................................................v
ACKNOWLEDGMENTS.................................................................................................vi
LIST OF FIGURES...........................................................................................................xi
CHAPTER
I INTRODUCTION......................................................................................1
CHAPTER Page
CHAPTER Page
VI CONCLUSIONS ....................................................................................172
REFERENCES...............................................................................................................178
VITA ..............................................................................................................................186
xi
LIST OF FIGURES
Page
Fig. 2.10. Classification of ADC architectures based on speed and resolution ..........26
Page
Fig. 2.19. Block diagram of the CRFB architecture for the SC modulator.................40
Fig. 2.22. Output spectrum of the SC ADC for 5 MHz span ......................................42
Fig. 2.23. Two tone IMD test for -11 dBr input signal ..............................................43
Fig. 2.24. Plot of SNR vs input amplitude for 270 kHz bandwidth ...........................43
Fig. 3.2. Block diagram of single loop and cascaded ADC architecture.............49
Fig. 3.3. Open loop equivalence of continuous and discrete time ADC .............52
Fig. 3.5. Representation of DAC pulses in time and frequency domain ...................55
Fig. 3.7. Excess loop delay in continuous time bandpass ADC ..........................61
Fig. 3.8. Effect of the bandpass filters finite quality factor on the NTF ..................64
Fig. 3.9. Effect of unequal DAC pulse rise and fall times ........................................65
xiii
Page
Fig. 4.7. Schematic of the feedback DAC blocks with ideal integrator DAC...........84
Fig. 4.8. Schematic of the feedback DAC blocks with lossy integrator DAC ..........85
Fig. 4.9. Output spectrum of the ADC architecture with and without integrator
losses ..........................................................................................................88
Fig. 4.12. Schematic of the second order Q-enhanced LC bandpass filter .................98
Fig. 4.13. Post-layout simulation of the bandpass filter center frequency tuning..104
Fig. 4.15. Post-layout simulation of the bandpass filter two tone test ...................106
Page
Fig. 4.27. Measured single tone output spectrum of the RF ADC ............................125
Fig. 4.29. Noise floor of the measurement setup with grounded inputs....................127
Fig. 4.30. Noise floor of the measurement setup for the two tone input test ............127
Fig. 4.31. Plot of measured SNR of ADC vs input power for the RF ADC .............129
Fig. 4.32. Plot of the measured IM3 vs input power for the RF ADC ......................129
Page
Fig. 5.7. Effect of deviations in center frequency and Q of filter on the NTF ........147
Fig. 5.10. System level implementation of the proposed digital tuning scheme.......152
Fig. 5.18. Output spectrum of the IF ADC from post-layout simulations ................169
xvi
LIST OF TABLES
Page
Table 4.4. Component and current values for the LC bandpass filter.....................103
Table 4.5. Component and current values for the integrator DAC .........................108
Table 4.6. Component and current values for the current DAC..............................109
Table 4.7. Component and current values for the preamplifier gain stage .............116
Table 4.8. Component and current values for the latch stages................................116
Table 4.9. Component and current values for the buffer stages..............................118
Table 5.3. Values of optimized loop coefficients for the IF ADC ..........................144
Page
Table 5.5. Component and current values for the OTA design...............................159
Table 5.6. Component and current values for the current DAC design ..................161
Table 5.7. Component and current values for the preamplifier gain stage .............164
Table 5.8. Component and current values for the latch stages................................166
1
CHAPTER I
INTRODUCTION
1.1. Motivation
The wireless communication industry has experienced tremendous growth since the
early 1990s throughout the world, especially in Europe, Asia and the USA. The wireless
market is poised for more growth in the future, with the demand increasing due to
addition of new services which provide more functionality for the end users. One of the
main reasons for this growth has been the major progress in the field of radio frequency
(RF) integrated circuit design and the design trend is slowly advancing towards a single
chip transceiver with very few external components. The major developments in system-
on-chip (SoC) design, which enables integration of analog and digital designs on the
same chip has led to integration of analog RF, analog baseband and digital signal
processors on the same chip. This has resulted in sleek and low power handset devices
with multi-purpose functionality like videos, voice, pictures and internet access.
The number of wireless standards has also increased drastically along with the
market growth, with each standard catering to specific market segments. The most
common wireless communication applications along with some of their standards are
shown in Fig. 1.1. The personal communication (cell phones) segment includes
standards like GSM (global system for mobile communications), GPRS (general packet
radio service), EDGE (enhanced data rate for GSM evolution), CDMA (code division
_____________
This dissertation follows the style and format of IEEE Journal of Solid-State Circuits.
2
multiple access), AMPS (advanced mobile phone systems), UMTS (universal mobile
division multiple access), DECT (digital European cordless telephone), IS-95 (digital
version of AMPS) etc. The wireless local area network (WLAN) for laptops, desktops
and PDAs include standards like IEEE 802.15, also known as Bluetooth, for the
personal area network (PAN); IEEE 802.11 a/b/g and Wi-Fi for wireless home area
networking, IEEE 802.16 a/e and Wi-Max for wide-area high speed wireless zones, in
the wireless metropolitan Area Network (MAN) segment, and IEEE 802.20 for the wide
navigation systems for locating position of objects (such as commercial systems used in
cars) use the GPS (global positioning system) standard. Another wireless standard that is
gaining popularity recently is IEEE 802.15.3a or UWB (Ultra wide band) for short range
3
high data rate applications in the local area network segment. The number of wireless
standards is increasing everyday with committees being formed in all regions of worlds
The prominent differences between the wireless standards are the geographical
area of usage, location in the frequency spectrum of the uplink and downlink channels
for base stations and user handsets, channel spacing and the number of channels (total
bandwidth of the standard), symbol and chip rates, type of multiple access techniques
these techniques), type of modulation scheme [some examples include BPSK (binary
phase shift keying), QPSK (quadrature phase shift keying), GMSK (gaussian minimum
shift keying) etc..], specifications for minimum signal power delivered by the antenna
(receiver sensitivity) and maximum transmit power at the transmit side, adjacent channel
specifications (blocker specification), etc [1]. There are different governing bodies in
each country which regulate the commercial usage of frequency spectrum, which has
resulted in adoption of different standards in different regions of the world. The wireless
communication service providers all over the world invest lot of infrastructure to support
the wireless standards in base stations for their respective coverage areas. The end user
devices (cell phones, wireless network cards etc) usually support transmission and
reception of signals for only one standard. A major requirement of the next generation of
wireless devices and base stations is a single chip set with minimum external
components that can support multiple standards. This would enable a single wireless
device which makes use of functionality that is supported by multiple service providers.
4
The support for multiple standards would increase the number of features supported by
the mobile devices (internet access, voice, data, pictures etc) with a faster response time.
There are many technical challenges, both at hardware and software level that needs to
Software radio architecture, which is described in detail in the next chapter, has
been proposed as a solution for support of multiple wireless standards in the same chip.
The main goal in a software radio transceiver is to perform the analog to digital (A/D)
and digital to analog (D/A) conversion as close to the antenna as possible, with
reconfigurable software programs running in the backend digital signal processor (DSP).
This research deals with the design of analog to digital converters (ADCs) for use in
The main aim of this research is to explore new design techniques for radio
frequency (RF) and intermediate frequency (IF) ADCs that can be used for digitization
and a continuous time bandpass sigma-delta () ADC architecture was found to be the
best choice for this application. The main drawbacks of ADCs that have been
previously reported in literature were identified and new design techniques are proposed
both at architectural and circuit level to overcome these limitations. The integrated
circuit implementations of the two ADC designs are then presented in detail.
5
The first part of this research presents the design of a direct radio frequency (RF)
ADC at 950 MHz center frequency using 3.8 GHz clock in IBM 0.25 m SiGe BiCMOS
technology. A fourth order continuous time bandpass ADC with LC filter was
implemented using a new architecture with only Non-Return to Zero (NRZ) digital to
analog converter (DAC) to reduce SNR degradation due to clock jitter. The operation of
the proposed ADC architecture is examined in detail and extra parameters are introduced
in the design to improve the operating range of the ADC. Power optimization techniques
are used both at architectural and circuit level to obtain good signal to noise ratio (SNR)
and 1 MHz, respectively around 950 MHz center frequency. The proposed ADC
reported in the literature for similar SNR performance. The measurement results of the
frequencies with low power consumption and enhances the state of art for continuous
time bandpass sigma delta (CT BP ) ADCs operating at GHz sampling frequencies.
The second part of this research deals with the design of an intermediate frequency
(IF) ADC at 125 MHz center frequency and 500 MHz clock in TSMC 0.35 m CMOS
technology. The fourth order continuous time bandpass ADC uses an architecture
with only Non-Return to Zero (NRZ) DAC and gm-C (transconductance-C) filter, which
is ideal for integrated implementation at intermediate frequencies. The gm-C filter uses a
coupling and source degeneration to enhance its linearity performance at 125 MHz. A
new automatic digital tuning algorithm is proposed to tune the ADC at the system level
in order to guarantee the performance over process, voltage and temperature (PVT)
variations. The digital bit stream is captured using an external digital signal processor
(DSP) and the noise transfer function (NTF) is measured using Fast Fourier Transform
(FFT) in the DSP. This information is used to tune the center frequency and quality
factor (Q) of the gm-C bandpass filter, and the DAC currents, which maximizes the SNR
performance for the ADC. The postlayout simulation of the fourth order modulator show
1.3. Organization
presented in Chapter II. The different types of architectures for communication receivers
bandpass (CT BP) ADCs, which are well suited for use in a software radio
are identified. A fourth order LC bandpass ADC which performs direct digitization
of RF signals around 950 MHz with 3.8 GHz clock is presented in Chapter IV. The
proposed ADC architecture is analyzed in detail and the circuit implementation of the
7
ADC building blocks are then presented. A detailed description of the test setup and
discussion of the measurement results are given at the end of the chapter. A fourth order
gm-C bandpass ADC for digitization of IF signals around 125 MHz with 500 MHz
clock is presented in Chapter V. The architecture, circuit implementation and the digital
tuning scheme of the IF ADC are described in detail, followed by simulation and
measurement results. The final chapter draws some conclusion and provides a summary
of this research work, together with possible future research directions for ADCs in
CHAPTER II
RECEIVERS
recover the base band signals that are modulated on a carrier wave at radio frequencies.
The up converted base band signals are transmitted at over air (channel) using power
amplifiers and the signal power at the transmitter output is very high, usually in the Watt
range. The carrier signals are corrupted by noise and adjacent band interferers (that
coexist in the electromagnetic spectrum) as they propagate through the channel and the
signal power at the receiver input (after the antenna) is very low, usually in the nano-
watt range. The two main considerations for signals as they pass through the receiver are
noise and distortion. The lower bound of the signal is determined by the noise floor and
the upper bound is dictated by the distortion components [2]. The lowest signal that can
be detected at the input of the receiver is defined as the sensitivity of the receiver and is
given by
where the NF is the noise figure of the receiver, BW is the bandwidth of the signal and
SNRmin is the minimum signal to noise ratio required at the output of the receiver for a
given bit error rate (BER). The sum of the first three terms gives the total integrated
noise of the system and is usually referred to as the noise floor. Noise figure is a measure
of how much noise the system adds to the signal as it passes through it and is defined as
9
SNRinput
NF = (2.2)
SNRoutput
The signal bandwidth, SNRmin and sensitivity are usually specified for a wireless
standard and the required noise figure of the receiver can be calculated from (2.1).
metrics such as P1dB (1 dB input compression point), IIP3 (third order input intercept
point), IM3 (third order intermodulation distortion), cross modulation etc are used to
nonlinearities, the signals from adjacent channel mix together with the signal in the
required bandwidth, which causes in-band distortion at the output. The wireless
adjacent band carrier can exist without corrupting the in-band signal. At the antenna, the
power of the desired signal maybe in the order of -112 dBm (0.56 Vrms in a 50
impedance) while the power of the interfering signals in adjacent bands maybe in the
order of 30-60 dB higher. Following detection of signals at the antenna, the receiver
must amplify the signals, suppress the interferers, translate the RF signals to base band
and extract the desired information. The two main metrics that are used to evaluate the
sensitivity can process weak desired signals in the presence of noise while a receiver
with high selectivity can process a desired signal in presence of strong interferers at
require receivers to have both high sensitivity and selectivity performance. In general,
10
the filters determine the selectivity and the other analog building blocks determine the
sensitivity of the receiver. Along with these stringent specifications, all portable wireless
devices have an additional requirement of low power dissipation in order to increase the
numerous tradeoffs during the design process, especially between noise, linearity and
power consumption. The first step in the receiver design is to derive the system level
performance requirements from the wireless standard specifications. The next step
would be the choice of the best architecture for the receiver implementation. Then, the
specifications for each building block of the receiver can be determined based on the
are suitable for integrated circuit implementation. The chapter starts with a description of
the superheterodyne architecture, which is the most widely used receiver architecture,
generation of wireless communication devices is a single chip receiver that can support
multiple standards, but superheterodyne or direct conversion architectures are not well
suited for implementing more than one standard. The software radio architecture, which
multiple standards in the same chip, is reviewed in detail. Intermediate frequency (IF)
implementation in current silicon technologies, is also analyzed. The output of all the
receiver architectures is in digital format, which can be readily given as input to the
digital signal processor (DSP) following the receiver. Different ADC architectures for
possible use in software radio implementation are analyzed to identify their advantages
and disadvantages for this application. The chapter ends with a survey of ADCs that
have been reported in literature for radio frequency and intermediate frequency
RF IF Baseband
Antenna (0.8 - 5.4 GHz) (100-200 MHz)
I
Lowpass
Image reject ADC
RF BPF VGA
BPF IF BPF
LO2 Digital
LNA bit stream
o
90
Mixer Q
LO1 Lowpass
ADC
Armstrong in 1918, where the incoming high frequency signal is down converted to a
12
lower frequency using a mixer and processed at lower frequencies. This architecture has
many advantages for processing a high frequency input signal. The channel selection is
implemented by varying the frequency of the local oscillator and range of signals that
can be handled is determined by the tuning range of the oscillator. The down conversion
and filtering of signals is easier to achieve at low frequencies rather than at high
frequencies [3]. The basic architecture proposed by Armstrong has evolved over the
years with minor variations and it is the most popular receiver architecture for wireless
communication receivers.
communication receivers is shown in Fig. 2.1, where the RF signal is converted to base
band in multiple frequency translation steps. The incoming signal at the antenna is pre-
frequency. The quality factor of the BPF is very high and is usually implemented using
off-chip SAW (surface acoustic wave) filters. The signal is then amplified by a low noise
amplifier (LNA), followed by an image reject BPF before the down conversion to an
intermediate frequency by the first local oscillator (LO1). The IF signal is further filtered
by IF BPF and amplified by the variable gain amplifier (VGA) before being frequency
translated to base band into parallel in-phase (I) and quadrature (Q) signals by the
second local oscillator (LO2). The baseband analog signals are then converted into
digital signals using low pass analog to digital converters (ADC) for both I and Q paths.
13
RF IF
Antenna (0.8 - 5.4 GHz) (100-200 MHz)
LO1 LO2
RF IF
(0.8 - 5.4 GHz) (100-200 MHz)
LO1 LO2
The location of the intermediate frequency (fIF) is related to the image problem that
arises during the down conversion process. The desired signal centered at carrier
frequency (fc) is down converted to intermediate frequency fIF using local oscillator
frequency (fc + fIF). However, the signal centered at image frequency (fc + (2*fIF)) is also
down converted to fIF, thereby corrupting the desired signal. A higher location of IF
helps to provide the maximum attenuation of the image frequencies from the image
reject BPF, while a lower IF relaxes the filtering and amplifier requirements for the IF
14
BPF and the VGA. A minor variation of the superheterodyne architecture such as
Weaver image reject [4] architectures has been proposed to overcome the image
standards in the same chip. The typical approach for implementing a multi standard
receiver using superheterodyne architecture is to replicate paths for each standard and
multiplex it in time using analog switches, as shown in Fig. 2.2. The main problem is
that most of the blocks are analog in nature and have to be redesigned for each standard.
There is minimal reuse of the components which results in large area and power
RF Baseband
(0.8 - 5.4 GHz)
Antenna
BB LPF
Amplifier
I
Lowpass
ADC
RF BPF
Digital
LO bit stream
LNA
o
90 BB LPF
Amplifier
Q
Lowpass
ADC
The direct conversion receiver architecture directly down converts the signal to
baseband rather than converting it into an IF first, and image rejection is no longer
architecture is illustrated in Fig. 2.3. The RF signal that comes out of the antenna is
filtered by the RF BPF and amplified by the LNA before being down converted directly
to baseband along parallel in-phase (I) and quadrature (Q) signals. The frequency
translation is performed by using two mixers using 0 and 90 phase shifted local
oscillator (LO) signals. The I and Q base band signals are amplified and low pass filtered
before the A/D conversion. This architecture eliminates all intermediate frequency
components and their associated design challenges, especially the image reject problem.
However, two practical implementation problems that have limited the use of direct
conversion architectures are DC offset and flicker noise [5]. DC offsets saturate the base
band circuits and if uncorrected, degrade the bit error rate (BER) performance of the
system. The power spectral density flicker noise of transistors is inversely proportional
to frequency and it severely degrades the performance of base band circuits in direct
conversion receivers. Direct conversion architectures are also not well suited for
implementing multiple standards, as each block has to be replicated for every standard.
The term software radio is used to define a broad class of radio transceivers
software and the fundamental aspects of the radio operation can be reconfigured by
16
upgrading that software [6] . The concept of software radio was introduced by Joseph
Mitola III in 1991 [7] and the block diagram of ideal software radio transceiver
architecture is shown in Fig 2.4. The input RF signal is multiplexed in time between the
transmit and receive paths through the T/R (transmit/receive) switch. The digital signal
information from the DSP is converted to RF output frequencies by the DAC and driven
to the channel by the linear power amplifier in the transmit path. In the receive path, the
RF input signals are directly sampled and digitized by the ADC. The digital bit stream is
given to the backend DSP, where demodulation and other signal processing algorithms
Reconfigurable
programs
Antenna
Linear RF
Power DAC
RF amplifier
T/R Digital
signal Output RF ADC signal
switch
processor
(DSP)
ADC
The ideal software radio receiver architecture shown in Fig. 2.4 has some very
technologies. The input RF signal power varies from a few W to milli Watts, which
17
together with distortion and noise specifications translates to receiver dynamic range of
around 120 dB. The ADC would require greater than 20 bits of resolution at GHz
modified form of the software radio architecture, shown in Fig. 2.5, greatly reduces the
ADC resolution requirements and it is considered for this research. The input RF signal
is processed by the RF BPF and amplified by the LNA. The variable gain amplifier
(VGA) controls the signal amplitude variations to reasonable levels at the input of ADC.
The ADC converts the RF signal to digital output and the analog processing blocks
Transmitter
Reconfigurable
programs
Antenna
Linear RF
Power DAC
RF amplifier
T/R Digital
signal
switch signal
processor
(DSP)
RF BPF
LNA + ADC
VGA
Receiver
The two main components of software radio receiver are the RF front end blocks
and the back end software platform and algorithms running on the DSP. The main
emphasis in a software radio receiver is to perform the A/D and D/A conversion as close
equalization etc for both transmit/receive paths are all determined in software that runs
in the digital signal processor (DSP) [7]. Software radio is the flexible and adaptive radio
architecture that can provide multiple services, support multiple standards and multiple
bands because it can be reconfigured and reprogrammed by the software programs that
run in the DSP. This is a broad definition which includes all digital transceivers and
software supported adaptability for multiple operations and environments [8]. The
ultimate goal in radio receiver design is to directly digitize the RF signal at the output of
the antenna and trends in receiver design have evolved towards this goal by
incorporating digitization closer and closer to the receive antenna at increasingly higher
frequencies and wider bandwidths [9]. The early digitization in the signal path also
eliminates need for additional mixers (for down conversion), filters and analog
processing blocks, which can result in significant savings in power and area.
the problem of gain/phase mismatches between I/Q paths which occurs when the
demodulation is done in analog domain. A new standard can be easily added by writing
a software program for the radio interface and upper layer protocols, provided the first
analog stage can provide a digitized version of the RF signal. This would require the use
of a wideband and high resolution ADC and DAC to cover the frequency spectrum of all
19
standards and their respective channels [10]. The DSP can also be replaced by field
combination of these in order to run the required software programs at the data rate
output of the ADC. The clock frequency of currently available commercial DSPs is in
the 1-2 GHz range and the DSP technology is on pace to support software radio
requirements.
RF IF
Antenna (0.8 - 5.4 GHz) (100-200 MHz)
LO
Software defined radio is the terminology used in literature to describe the idea
sometimes referred to as digital radio receiver. The block diagram of a digital IF receiver
architecture is shown in Fig. 2.6. The RF BPF selects the required channel(s) from the
20
signal appearing at the antenna. After amplification by the LNA, the signal is filtered by
the image reject BPF to attenuate the components at image frequency. This operation is
followed by down conversion to IF by the mixer and LO. The IF BPF further filters the
signals and the VGA adjusts the signal strength to be processed by an IF bandpass ADC,
which directly gives the digital output to DSP. There is no particular standard for the
location of IF signals (though there are standards for location of input RF signals) and
the choice of the intermediate frequency is based on the requirements of the IF circuits.
The IF bandpass sampling solves flicker noise and DC offset problems which are usually
supports only a single channel, with channel selection done by an analog filter.
Mobile user 1
Mobile user 2
Mobile user n
Software radio for base stations Software radio for mobile handsets
phone system. The wireless transmission coverage areas are usually divided into cells,
with each cell covered by base stations. The hand sets transmit the information to the
nearest base station and the signal goes through various base stations before reaching the
target system. In this scenario, transmitters and receivers are required both in individual
hand sets and in the base stations. A software radio receiver to support multiple
shown in Fig 2.7. Low power consumption is absolutely essential for mobile handsets,
but is not a strict requirement for base stations. Large power consumption increases the
temperature. The increase in temperature also raises the thermal noise floor (thermal
noise is proportional to temperature), and degrades the overall signal to noise ratio
(SNR). The thermal noise floor should be at the minimum level for high performance
analog systems. To avoid these disadvantages, low power consumption is necessary even
software radio architecture. The wide range of existing and new standards in the cellular
and mobile marketplace has resulted in the adoption of wide range of hand set and base
station architectures for different systems deployed around the world. The ability to
the final stage of manufacture to tailor it to a particular market, presents great benefits to
equipment manufacturers. The mobile handset can be used worldwide if all wireless
22
standards, which are usually specific in one region, are supported in a single terminal. A
powerful benefit of a software defined radio terminal, from the perspective of the
network operator, is the ability to download new services to the terminal after it has been
purchased and is operational on the network. The software radio can also guarantee
compatibility between second generation and third generation networks [11]. Suppose a
mobile device moves from 3G standard service area (like UMTS) to an area covered
only by 2G standard (like GSM), then the device can still function if it supports both
standards. This is very advantageous economically for service providers, since a large
amount of money has been invested to install and maintain the base stations and the
multi-standard support helps in reuse of existing infrastructure. The ability to adapt key
benefit of software radio. It is possible, for example, to reduce the complexity of the
(quadrature phase shift keying) when channel conditions becomes poor, thereby
improving noise immunity and decoding margin. It is also possible to adapt the channel
coding scheme to prevent particular types of interference, rather than just Gaussian
noise, as the device moves form one area to another. Many other parameters like data
rate, channel and source coding, multiple access schemes and modulation type can be
adapted dynamically to improve quality of wireless service [6]. Cognitive radio is a term
used in literature to define an intelligent radio transceiver that makes use of RF spectrum
dynamically based on its usage [12]. The functionality of cognitive radio encompasses
23
software radio together with some artificial intelligence embedded in it. A software radio
The main focus of this research work is to explore new design techniques at both
architectural and circuit level for silicon implementation of analog to digital converters
for use in software radio communication receivers. The software radio receiver, shown
in Fig 2.8, is the target architecture and the ADCs are designed to operate in this
framework. The RF ADC directly digitizes the RF signal after the VGA, while the IF
ADC converts a frequency translated IF signal to digital output. The main goal of the
design process is to obtain the maximum resolution (number of bits or SNR) for both the
ADCs with minimum power consumption. The different possible ADC architectures are
examined in detail next to identify the best architecture for this application.
Antenna
RF BPF
LNA RF ADC
+ VGA
Antenna
Digital signal
processor
(DSP)
RF BPF
LNA
+ VGA IF ADC
Image reject
BPF
LO1
Research
focus
Analog to digital converters are used to transform analog signals (continuous in time
generalized analog to digital conversion process is shown in Fig. 2.9. The input analog
signal is sampled at a frequency Fs, which converts it into a discrete time signal. The
quantizer block converts the discrete time signal into the nearest digital bit code and the
encoder converts it to N-bit output code. The sampler, encoder and the quantizer blocks
together form the ADC. The process of quantizing a signal is non- reversible and
When the full scale voltage of value VFS is applied to a N bit converter, the quantization
step () is given by (VFS / (2N-1)) and the maximum quantization error is given by
(/2). The quantization error has uniform probability density function, provided the
quantization errors are sufficiently random in nature [13]. The quantization noise power
25
spectral density is uniformly distributed between (Fs/2) and (Fs/2) with a magnitude of
(2 /12). The main performance measure of an ADC, signal to noise ratio (SNR), is
defined as the ratio of the signal power to the integrated noise power over the signal
Another performance metric of an ADC is the dynamic range (DR), which is defined as
the ratio between maximum input signal power with acceptable distortion to the
minimum detectable input signal power level. A signal is said to be detectable if the
signal power is just above the noise power (when SNR = 0 dB). Signal to noise +
distortion ratio (SNDR) is defined as the ratio of the signal power to the sum of noise
and distortion components over the required signal bandwidth. The fundamental limit of
noise floor in circuits is given by the thermal noise. The quantization noise floor of the
ADC is assumed to be above the thermal noise floor and therefore determines the
performance of the ADC. The quantization noise usually dominates the noise floor for
ADCs with narrow or small bandwidth, but for ADCs that handle large signal
bandwidth, the integrated thermal noise floor limits the performance of the ADC.
resolution and power is shown in Fig. 2.10. The sampling frequency, number of bits and
power consumption are three parameters which tradeoff with each other for all ADC
architectures [14]. The flash, folding, sub-ranging and pipeline ADC architectures are in
the very high speed and medium to low resolution category, while the successive
26
5 10 15 20
Resolution (number of bits)
Fig. 2.10. Classification of ADC architectures based on speed and resolution
approximation and sigma-delta ADC architectures are in the high/medium speed and
high resolution category. The incremental and integrating dual ramp ADC architectures
fall in the slow speed and very high resolution category. The power consumption for all
ADCs increases at higher sampling frequencies, and is usually in the Watt range for
GHz sampling frequencies. The RF ADC for software radio would require sampling
frequency in the GHz range while the IF ADC would require sampling frequency in the
range of MHz. There are two broad classifications of ADCs based on the relation of
their sampling frequency to the bandwidth of the input analog signal, namely Nyquist
rate and oversampled ADCs. The sigma-delta ADC is the only architecture in the
oversampled category, while all other architectures in Fig. 2.10 belong to the Nyquist
rate category.
27
trains that repeat every Ts seconds. Mathematically, the sampled discrete time signal is
+
x * (t ) = x(t ) (t nT )
n =
s (2.4)
where n is integer and is the ideal impulse function. In frequency domain, the sampled
+
1
*
X ( f ) = X ( f )* T
n= s
( f n * Fs ) (2.5)
where X(f) is the Fourier transform of the signal x(t). The sampling function modulates
x(t) by carrier signals having frequencies at Fs, 2Fs, 3Fs ..etc and in frequency domain,
this causes the spectrum of x(t) to be repeated at nFs frequencies. The sampling process
of an analog signal in time and frequency domain is shown in Fig. 2.11. When the
analog signal has a band limited frequency Fin, then it must be sampled at a rate that is
greater than 2*Fin in order to avoid aliasing, where signal image components at higher
frequencies fold back into the signal bandwidth. The Nyquist sampling theorem states
that the sampling frequency should be at least 2*Fin in order to be able to recover the
original signal perfectly from the sampled version. All Nyquist rate ADC architectures
Impulse train
Ts
Analog signal x(t) Sampled signal x*(t)
Sampling
Time
-F F Frequency F -F F+F
in 0 in -F 0 F F 2F
in in s in s in s
s
Frequency
The general components of a Nyquist rate ADC consist of anti-alias filter (to
limit the bandwidth of the input signals), sample and hold circuit and multi-bit quantizer.
The quantization noise power of Nyquist rate ADCs extends in frequency from DC to
Fs/2, as shown in Fig 2.12. The signals at the input of RF and IF ADCs are located
around a center frequency and the bandwidth depends upon number of channels or
standards that are required to be converted to digital output. For example, suppose the
RF ADC digitizes signals of 1 MHz around 1 GHz, then the Nyquist rate ADC would
have a sampling frequency of atleast 2 GHz and will digitize the entire band from DC to
1 GHz. Since there is no inherent filtering in a Nyquist rate ADC, a lot of power is
wasted while trying to reduce the quantization noise in the entire Nyquist bandwidth,
29
which contains frequency components outside the required signal bandwidth. This
disadvantage makes Nyquist rate ADC unattractive for software radio applications.
Fin
Power
Quantization noise, rms power =
12
= Quantizer step, 1 LSB
0 FS/2 FS Frequency
The ADC architectures that have a sampling frequency much greater than the
Nyquist sampling criterion belong to the category of oversampled ADCs (Fs >> 2*Fin,
where Fin is the bandwidth of the input signal). The quantization noise power extends in
frequency from DC to Fs/2 and its root mean square value is the same as that of Fig.
2.12. The main difference is that the in-band quantization noise power is reduced as
compared to Nyquist rate since the same power is spread over a much greater frequency
range. The oversampling ratio (OSR) of the ADC is defined as Fs/(2*Fin) and a 2x
feedback loop), which greatly reduces the in-band quantization noise power and results
in a high SNR within the signal bandwidth [15]. The negative feedback together with
Noise shaping
In-band quantization noise
0 FS/2 FS
Frequency
The basic components of a ADC is shown in Fig. 2.14, and it includes a loop
filter in the forward loop followed by a comparator, with a feedback digital to analog
converter (DAC) completing the negative feedback loop. While considering a small
signal model (a linearized model is used for the non-linear comparator block) and
31
assuming a unity transfer function for the DAC and comparator, the output (Dout) is
given by
where Qn is quantization noise, STF and NTF are the signal and noise transfer functions.
Dout H ( z) D 1
STF = = , NTF = out = (2.7)
Vin 1 + H ( z ) Qn 1 + H ( z )
Quantization
noise ( Q )
n Comparator
Loop filter
V Dout
in
H(z)
Feedback
DAC
The digital output (Dout) then passes through decimation and digital filter blocks, where
down sampling is performed to get the final output. The loop filter H(z) is usually
implemented a low pass filter and results in a low pass ADC. When H(z) is a first
order discrete time integrator (low pass filter), the loop filter, STF and NTF is given by
z 1
H LP ( z ) =
1 z 1
(
, STF = z 1 , NTF = 1 z 1 ) (2.8)
32
The NTF has a zero at DC and the quantization noise is high-pass filtered at the output,
while the signal is delayed by one clock cycle at the output. The low pass ADCs
offer the advantage of high resolution in a narrow signal bandwidth, but they are still not
suitable for RF/IF ADCs in Fig. 2.8 as the quantization noise is shaped over a wide
range of frequencies (from DC to the signal bandwidth). The easy solution is to use a
loop filter with a bandpass transfer function which results in the ideal ADC architecture
filter. The following general discrete time low pass to bandpass transformation may be
z 1
z 1 z 1 1 < < 1 (2.9)
1 z 1
This general transformation moves the zero of the NTF from zero to an arbitrary location
below the sampling frequency Fs. For the case when = 0, (2.9) becomes z-1 -z-2. This
transformation is widely used because it preserves the properties of the low pass
modulator and it is convenient for integrated implementation. If the order of the low pass
filter is n, then the transformation results in a bandpass filter of order 2n. Using this
transformation, the loop filter, STF and NTF of a 2nd order bandpass ADC is given by
z 2
H BP ( z ) =
1+ z 2
(
, STF = z 2 , NTF = 1 + z 2 ) (2.10)
33
The zeros of the NTF are moved from DC to (Fs/4), and the STF delays the signal by
two clock cycles. The quantization noise is shaped by a notch transfer function around
(Fs/4), while the input signals have a bandpass transfer function. This noise shaping
results in a high SNR in the signal bandwidth around the center frequency of the
bandpass filter [16]. The oversampling ratio (OSR) for a bandpass ADC is defines as
(Fs/(2*Fin)), where Fin is the bandwidth of the signals around the center frequency.
The analog signals that have to be digitized in software radio receivers are
located at center frequency Fo (which can be at RF or IF) and the signal bandwidth is FB.
The quantization noise is attenuated only in the required signal bandwidth instead of the
whole Nyquist bandwidth (DC to Fs/2), which implies lesser power consumption to
architecture is the best choice for digitization in software radio receivers, as shown in
Fig. 2.15.
Bandpass ADC
Fo
C1 V in
Vout
Gm
C L
1 C2
1
V in
Vout LC filter
2 2 1
Operational
amplifier V in
Gm Vout
Switched capacitor integrator
C
Switch
1 ,2 Non overlapping
Transconductance-C filter
clock phase
capacitor (discrete time) or continuous time mode, as shown in Fig. 2.16. The switched
switches and operational amplifier with local feedback. The center frequency of a
switched capacitor bandpass filter is fixed by capacitor ratios (which are very accurate)
and it does not require any automatic tuning scheme. The DC gain of the amplifier
should be high to guarantee proper closed loop operation of the amplifier. The switching
scheme uses two non-overlapped clock phases and the amplifier should settle to the final
value within one-half of the clock period. The settling time specification requires the
35
amplifier which is working in closed loop to have gain bandwidth (GBW) product in the
range of 10-20 times the clock frequency [17]. For a clock frequency of 2 GHz, the gain
bandwidth requirements would be greater than 20 GHz. The power consumption of such
almost impossible to design a system with these building block specifications. It is also
difficult to combine both high DC gain and GBW in an amplifier because of contrasting
design approaches. Finally, the switches that can be implemented using CMOS or BJT
transistors do not operate properly at GHz sampling frequencies. Due to all these
limitations, switched capacitor bandpass ADCs are usually not preferred above
sampling frequencies of 200 MHz. To illustrate this point further, design overview and
measurement results of a switched capacitor BP ADC which was done in the initial
Switched capacitor bandpass (SC BP) modulators widely used for direct
center frequency is directly related to the clock frequency, which is typically limited by
the speed of the switched capacitor integrator. Various architectural level techniques
such as two-path architecture [18], double sampling resonator [19], double sampling
with pseudo two paths [20] and double delay resonator [21] have been proposed to
36
improve the speed of the amplifier. The amplifier is used in the design of a high speed
SC BP ADC.
Cp Cf
Ca
Vi- 1
Vo+ Error
DC gain
Vi+ Vo-
Ca
Cp Cf Slew Limited by
limit GBW and PM
and the overall speed of the filter is determined by the settling performance of the SC
integrating clock phase is shown in Fig. 2.17. The closed loop transfer function during
V0 Av0p
(2.11)
Vi s + (1 + Av0 )p
where
37
Ca Cf
= , = (2.12)
Ca + C f + C p Ca + C f + C p
is the feed forward factor, is the feedback factor, Av0 is the open loop DC gain of the
amplifier, p is the dominant pole of the open-loop amplifier, Cp is the sum of all
parasitic capacitances at the input node of amplifier and Ca ,Cf are the input and
shown in (2.11), the output step response of the SC integrator after the slewing phase is
given by
Ca
Cf
V out (t ) = - 1 e . GBW
( .t
)V + V o ( t0 ) (2.13)
1 in
1 + A
v0
where GBW (=Av0p) is the open-loop gain-bandwidth product of the amplifier and
Vo(t0) is the initial condition at the output node. The amplifiers settling consists of two
phases: slew rate and linear settling dictated by the GBW and phase margin of the
amplifier [22]. The error in the final settling is inversely proportional to amplifiers DC
gain and speed in the linear region is determined by GBW. For a good settling
performance, the amplifier must have a high gain along with high GBW and good phase
margin. The exponential term in (2.10), GBW, should be large such that the output
settles properly within one-half of the clock period (Tclock). For the worst case operating
condition of the SC integrator and a clock frequency of 100 MHz, the amplifier fGBW
should be greater than 980 MHz and 1.3 GHz for a settling accuracy of 0.25 % and
0.033%, respectively. Assuming that the system settles properly within the integrating
38
phase, the settling error is ultimately determined by the factor Av0. For a settling error
below 0.1 %, Av0 must be greater than 60 dB; hence for around 0.2 (-14 dB), Av0
It is difficult to design an amplifier with both high gain and GBW because of
contradicting design requirements. High gain amplifiers use cascode and multi-stage
architectures, low bias currents and long channel devices, whereas high bandwidth
amplifiers use single-stage architectures, high bias currents and short channel devices
[23]. Two basic approaches for designing a high gain amplifier are cascode or vertical
approach and cascade or horizontal approach. The signal swing in cascode amplifiers is
constrained by the power supply voltage, which is a problem for low voltage designs. In
compensation schemes trade bandwidth for stability. The No Capacitor Feed Forward
(NCFF) compensation scheme [24] has been proposed to overcome the limitations of
Miller compensation schemes. A feedforward path, which has the same phase shift as the
direct path, produces a left half plane (LHP) zero. The positive phase shift of LHP zero
is used to cancel the negative phase shift of pole, resulting in a good phase margin. This
scheme shows a great improvement in bandwidth, along with a high gain and good phase
margin. The schematic of the differential amplifier designed in TSMC 0.35 technology
using the NCFF compensation scheme is shown in Fig. 2.18. The post-layout
simulations of the amplifier show DC gain of 80 dB, GBW of 1.4 GHz, phase margin of
62 and a 0.1% settling time of 2 ns while consuming 4.6 mA of current from a 1.25V
power supply.
39
VDD VDD
Vcmfb2 Vcmfb2
Vcmfb1 M4 M4 Vcmfb1 M8 M8
Vo- Vo+
Vbp M3 M3 Vbp
M7 M7
Vbn1 M2 M2 Vbn1
Mb2
Vi+ M1 M1 Vi- Vb2 Ib2
Mb1 VSS
Vb1 Ib1
Vbn2 M6 M6 Vbn2
VSS
Vi+ M5 M5 Vi-
Vbff Ibff
Mbff
VSS
Fig. 2.18. Schematic of the differential amplifier using NCFF compensation scheme
the sampling frequency, for ease of digital demodulation. The coefficients are optimized
based on tradeoff between signal swing in the internal nodes, capacitance spread and the
maximum SNR attainable with stable operation. The value of capacitor in the first stage
is important as its kT/C noise contribution dominates the overall noise performance. The
out of band gain of the noise transfer function (NTF) is an important parameter for
stability; in this design, this parameter is chosen to be 1.5, giving some margin for PVT
variations. Simulations were done in Simulink and Matlab to verify the stability of the
40
Coefficients Value
c1,c2,c3 1, 1, 1
g1,g2 2, 2
Vin
b1 b2
b3
g1 g2
COMPARATOR
1
z 1 1 z
1 c3
c1
1 z 1
c2 1 z 1 1 z 1
1 z 1
a1 a2 a3 a4
Fig. 2.19. Block diagram of the CRFB architecture for the SC modulator
architecture over typical component variations. The block diagram of the architecture is
shown in Fig. 2.19 and the optimized coefficients used in the modulator implementation
are given in Table 2.1. Switches are implemented using only NMOS transistors with a
boosted clock voltage to reduce the switch resistance. The comparator is a twostage
structure, with a low gain first stage followed by a latch. A single-bit DAC is used to
feedback the digital output; the reference voltage used was 0.25 V.
41
technology is shown in Fig. 2.20 and it occupies an area of 1.25 mm2 . The bandpass
modulator was characterized by directly injecting the output digital bit stream into a
42
spectrum analyzer. The clock frequency is at 92 MHz and the IF signal is at 23 MHz.
The NTF of the modulator measured by grounding the inputs is shown in Fig. 2.21. The
noise floor of the modulator is at 127 dB. This level is determined not only by the noise
shaping of the sigma-delta modulator and thermal noise of the transistors, but also the
noise added to the digital levels (from noisy supply voltages) since the output is directly
The output spectrum of the ADC for 5 MHz span with a single tone input at 23 MHz is
shown in Fig. 2.22. The noise level increases mainly due to the noise contribution of the
signal generators used. The two-tone test for signals 200 kHz apart in the band (22.9
MHz and 23.1 MHz) are shown in Fig. 2.23. IM3 is less than 58 dB for two-tone input
signals, each at -11 dBr (normalized with respect to the reference voltage of 0.25 V).
43
The peak SNR of the modulator was 80 dB and 54 dB for bandwidth of 270 KHz and
3.84 MHz, respectively. The plot of SNR of ADC for various input values is shown in
Fig. 2.24. The modulator consumes 47.5 mW of from 1.25 V power supply.
Fig. 2.23. Two tone IMD test for -11 dBr input signal
Fig. 2.24. Plot of SNR vs input amplitude for 270 kHz bandwidth
44
The measurement results of the ADC show that high performance IF digitization
limited by the amplifier requirements and for implementations with more advanced
CMOS technologies cannot exceed 200 MHz for high SNR performance. The linearity
of the switches degrades at high clock frequencies and limits the maximum speed of
the switched capacitor filters. For ex, Operation transconductance amplifiers (OTA) or
gm-cells used in continuous time filters usually require GBW to be the same value as the
filter with lower power consumption at higher signal frequencies, which is one of the
main reasons why continuous time bandpass filters are preferred over switched capacitor
filters. Continuous time bandpass (CT BP) ADCs offer many advantages for low
power digitization in software radio communication receivers and will be used for both
the RF/IF ADC implementations in this research. A brief survey of bandpass ADCs that
shown in Table 2.2 [26]-[35]. The first five entries in the table are ADCs which
perform direct RF digitization (input signals are from 0.8 to 1.3 GHz) and other ADCs
mixer CMOS
From Table 2.2, it is clear that continuous time bandpass (CT BP) ADC is the
dominant choice of architecture for both RF and IF A/D conversion. The RF ADCs with
sampling frequencies in the GHz range (3.2 4.3 GHz) have been implemented in SiGe,
InP, and GaAs technologies. A switched-capacitor CMOS low pass ADC with an
46
integrated mixer is reported in [29] and in spite of low power consumption, it has limited
signal to noise + distortion radio (SNDR) because it has a mixer integrated in the loop
and all harmonics from the mixer directly appear at the digital output. Subsampling is
another technique that has been proposed for RF A/D conversion, but it has the
disadvantage of poor SNDR due to aliasing of high frequency components into the
signal bandwidth [36]. The power consumption of the reported RF ADCs is high (from
0.45 to 6.2 watts) for obtaining good resolution (> 10 bits). The IF ADCs with
sampling frequencies in the GHz range (0.4 4 GHz) have been implemented in SiGe,
The main drawback of the high resolution ADCs in Table 2.2 is that they have
large power consumption, which makes them unsuitable for use in portable wireless
devices. The main aim of this research is to explore new design techniques for RF/IF
continuous time bandpass sigma-delta ADCs to obtain good SNR performance with low
power consumption.
47
CHAPTER III
3.1. Introduction
is shown in Fig. 3.1. The loop filter is a Nth order continuous time bandpass filter and the
comparator together with the DAC completes the feedback path. The center frequency of
the bandpass filter is fo, and the input signals with bandwidth fin is located around the
Clock
Feedback DAC
Dout Dout
STF = NTF =
Vin Qn
Power
Power
Freq Freq
center frequency. The negative feedback loop causes the quantization noise (assumed to
be injected at the comparator input) to have a notch transfer function around fo, while the
signals have a bandpass transfer function. The comparator is the only discrete-time
block, while the filter and DACs are continuous time blocks. The sampling operation
occurs inside the loop and the loop filter provides inherent anti-alias filtering of the input
signals, which is one of the main advantages of a CT BP ADC, along with low power
order and type of continuous time loop filter, clock frequency and resolution of the
comparator. The two most common architectures are single loop and MASH (multi stage
noise shaping, also called cascaded) ADC. The block diagram of a generic single bit
and a cascaded ADC architecture is shown in Fig. 3.2. The single loop architecture,
as the name suggests, has only one feedback loop with a loop filter (H1(s)) and they have
a very simple implementation. However, single loop ADCs with higher order loop
filters (order > 2) exhibit unstable behavior [15]. The signal swing at the output of the
filter increases when a higher order filter is used and can saturate the input node of the
comparator. This condition is called comparator overload and causes the digital output to
stay at a constant logic value. When this condition occurs, the negative feedback loop is
disabled and the ADC operates in open loop, which causes potential stability problems.
A cascaded ADC architecture, shown in Fig. 3.2 (b), overcomes the stability problem
by using a cascade of lower order filters (in this case H2(s) and H3(s)), which achieves
an equivalent higher order ADC while maintaining the signal swing at all internal nodes.
49
Feedback
DAC
Digital noise
Feedback cancellation
DAC filter
DAC
Q
1
HH (s)
(s)
33
Feedback
DAC
Fig. 3.2. Block diagram of single loop and cascaded ADC architecture
The quantization noise of the first stage (Q1) is calculated and is given as input to the
cascaded second stage. The output of the second stage is added to the first stage after
50
being processed by a digital noise cancellation filter. The main disadvantage of multi-
stage architecture is that it is very sensitive to mismatch of filter transfer functions [15].
The digital noise cancellation filter has to be matched perfectly with the analog loop
filters (H2(s) and H3(s)) and a slight mismatch causes a considerable reduction in the
SNR performance. The strict matching requirement becomes more difficult to implement
at high operating frequencies, which restricts the use of cascaded architectures to low
The choice of the order of the loop filter, clock frequency and the comparator
resolution affect the SNR of the ADC. The SNR of a generic Nth order, single loop
where OSR is the oversampling ratio (OSR = fs / (2*fin); fs is the clock frequency, fin is
the bandwidth of the input signal) and B is the number of bits in the comparator. The
order of loop filter determines the overall order of the modulator (N in (3.1)) and SNR
increases for a high order loop filter. The OSR of the ADC can be increased by
increasing the clock frequency, which results in a higher SNR according to (3.1). The
SNR of the ADC increases by 6 dB for a 1 bit increase in the comparator resolution,
which is evident from (3.1). The disadvantages of using a multi-bit comparator is that it
results in higher power consumption together with increased accuracy requirements for
the DAC in the feedback loop. The linearity of the multi-bit DAC in the feedback path
has to be in the same order as the required SNR at the ADC output. A one bit DAC is
51
requited to obtain the desired linearity for a multi-bit DAC [15]. The different types of
LC and active-RC filters. The choice of the type of loop filter depends upon various
chapter, followed by discussion of the main non-idealities and their impact on the ADC
encouraged to read the book for further understanding of the concepts. The main
continuous and discrete time domain techniques for its full description. The analysis
capacitor (SC) BP ADC, since there exists many design tools and techniques to
design SC architectures [37]. The equivalence between a discrete time and continuous
time ADC is illustrated in Fig. 3.3. The main difference between the two
architectures is that signal is sampled right at the input for a SC ADC, whereas the
52
x(n) x(n)
Dout d Dout x(t) c
DAC H(z) DAC H(s)
1
Ts =
Fs
Two structures are equivalent if x(n) = x(n) at the comparator input every clock cycle
d c
Fig. 3.3. Open loop equivalence of continuous and discrete time ADC
signal is sampled inside the loop before the comparator in a CT ADC. The two ADC
architectures are equivalent if the output bit stream is the same when the same inputs are
applied at their inputs, which can be guaranteed when the input to the two comparators
are same at the sampling time instants every clock period (x(n)d = x(n)c).
The open loop structure for both ADCs is obtained by breaking the feedback
loop at the DAC inputs, as shown in Fig. 3.3. The value of x(n)c is obtained by sampling
This can be guaranteed if the impulse response of both the open loop structures is same
L-1 and Z-1 are the inverse Laplace and inverse Z transforms. In time domain, the
equivalence is given by
where h(n), rdac(t) and h(t) are impulse responses of H(z), DAC and H(s), respectively.
transformation as they require the open loop impulse responses to be similar [39]. The
gives the desired SNR with stable performance. This helps in identifying H(z) and DAC
coefficients for the discrete time modulator. Then, H(s) and DAC coefficients in time
domain can be obtained by using (3.3) and the properties of both the structures (like
dynamic range, stability etc) are the same because of the impulse invariant
transformation.
The choice of the DAC pulse shape is an important design consideration in the
the modulator. The different types of DAC pulse shapes that can be used in the
modulator are shown in Fig. 3.4. The non return to zero (NRZ) DAC pulse has a
transition (from high to low or vice versa) only when there is a change in the polarity of
the comparator output. During the clock cycles when there is no change in the bit
polarity, the previous value of the bit stream is maintained in a NRZ DAC pulse stream.
54
CLOCK
1 1 1 0 0 1 0 1
V
Time
V
Return to zero
(RZ) DAC
Time
V
Half delay return to
zero (HRZ) DAC
Time
Fig. 3.4. Different types of DAC pulse shapes
The return to zero (RZ) DAC pulse has the value of the bit (low or high) for half a clock
period and returns to zero value for the other half of the clock period. The half return to
zero (HRZ) exhibits the same behavior as RZ pulse with the difference that it is shifted
The loop transfer function depends on the type of DAC pulse shape, which is
evident from the Rdac(s) term in (3.3). The NTF of a CT BP ADC is determined by
the type of the DAC pulse shape and it plays an important role in determining the ADC
performance. The representation of the different DAC pulse shapes in time domain and
frequency domain is shown in Fig. 3.5. The conceptual difference between the DAC
pulse shapes is the fraction of clock period for which current (charge) is injected back to
55
the loop filter. The type of DAC pulse also affects the non-idealities of the modulator
V
1 e sT
(a) Non Return to Zero (NRZ) DAC H NRZ ( s ) =
s
0 T Time
V sT
(b) Return to Zero (RZ) DAC 1 e 2
H RZ ( s ) =
T Time
s
0 T
2
RZ(t) = 1 for 0<t<(T/2); 0 otherwise
V sT
(c) Half Return to Zero (HRZ) DAC e 2
e sT
H HRZ ( s ) =
T
s
0 T Time
2
HRZ(t) = 1 for (T/2)<t<T; 0 otherwise
power over the required signal bandwidth and a decrease in SNR at the ADC output. The
non-idealities also results in distortion components, which reduces the dynamic range of
the ADC. The non-idealities may also lead to loss of stability in certain cases (clipping
in the internal nodes), which makes it necessary to take the non-idealities into account
The ideal clock signal is periodic in nature with consecutive rising and falling
edges separated in time from each other by exactly Ts seconds. There are many non-
idealities (noise, bandwidth limitations etc.) which cause the clock signal to deviate from
ideal behavior and the clock edges occur at time instants of [n*Ts tj] seconds (n is an
integer). The deviation of edges from ideal behavior every cycle ( tj seconds) is known
as clock jitter and it is usually specified as rms (root mean square) value by considering
the jitter samples over a large number of clock cycles. The clock jitter causes an error in
the sampling time and it is reflected in the ADC output as a noise component, similar to
the quantization noise. When the noise component due to jitter is uncorrelated with
quantization noise, then their noise power add directly and raises the noise floor at the
ADC output. As the clock sampling frequency increases, clock jitter becomes an
increasing fraction of the time period and becomes the dominant limit in the SNR
performance of the ADCs. A simple analytical formula was derived in [40] for a
Nyquist rate ADC, where the maximum number of bits achievable considering only jitter
error is given by
2
Bmax, jitter _ Nyquist = log2 1 bits (3.5)
3f
clock j
where fclock is the clock frequency and j is the rms value of clock jitter.
Clock jitter raises the in-band noise floor of CT BP modulators, similar to the
effect of Nyquist rate ADCs. At high sampling frequencies, the in-band noise
component due to jitter has a flat spectral shaping (white noise) and the advantages of
57
noise-shaping are absent. The edge of DAC current pulses which are fed back to the loop
filter vary in time due to clock jitter and the clock jitter noise undergoes a similar
transfer function as the signal (especially for the DAC connected to the first stage),
which causes the jitter noise to directly appear at the ADC output. The effects of clock
jitter for the different DAC pulse shapes in shown in Fig. 3.6. The clock jitter causes a
small error every cycle during rising or falling clock edges and the power of the error
term depends on the rms value of clock jitter (j). The error term due to NRZ is much
lower than the error term due to RZ/HRZ DACs and is evident from the sum of the
tj
V
NRZ
DAC
Time
Ts
V
RZ
DAC
To
Time
V
HRZ
DAC
Time
Error due to clock jitter every clock edge j RMS value of clock jitter
clock jitter has been derived analytically in [41], where jitter is assumed to be a small
perturbation in the sampling instant with a first order expansion of the non-linear effects
of jitter. The power of the jitter noise in the feedback DAC pulses is calculated for a
sinusoidal input signal and then multiplied by the signal transfer function to obtain the
noise power due to jitter at the ADC output. The SNR for CT BP ADC architecture
with only NRZ DAC is given by (3.6), while the SNR for the ADC architecture with
OSR * V 2 * sinc 2 o Ts
2
in
= 10 log10 (3.6)
SNR NRZ 2
j
4
Ts
OSR * V 2 * sinc 2 o To
in
SNR RZ / HRZ = 10 log 10 2 (3.7)
2 2
Ts j
4
To Ts
where OSR is the oversampling ratio, Vin is the amplitude of input signal, o is the
center frequency of the bandpass filter, Ts is the time period of sampling clock, To is the
time for which RZ DAC pulse is high (usually 50% of Ts) and j is the rms value of
clock jitter. The sinc function appears in the numerator of (3.6) and (3.7) because the
evaluated in a small region around the center frequency [41]. When To = 0.5*Ts, the
SNR for NRZ DAC is 6 dB higher than RZ/HRZ DAC, assuming the same value of j in
59
both cases. NRZ DAC waveforms make a transition only when the bit value changes
while RZ DAC waveforms have a transition every cycle, which results in a higher error
power introduced due to clock jitter for the RZ case. For the architecture which uses a
combination of both RZ and HRZ DAC, the error power term doubles and the SNR of
the architecture with only NRZ DAC improves by 9.3dB (1.5 bits), as compared to the
RZ/HRZ architecture [42]. Simulation results performed in Matlab validate this analysis
and SNR improvement of roughly 2 bits (12 dB) is obtained by using NRZ DAC over
RZ/HRZ DAC architectures. This improvement is significant when the clock frequencies
are in the GHz range and the clock jitter becomes an increasing fraction of the overall
time period.
regenerative latch. The regenerative latch is a positive feedback circuit which amplifies
the voltage difference at the input to a digital output of maximum positive value (+1) or
maximum negative value (-1). The regenerative circuit of the digital latch can be
modeled as a single pole system where the voltage difference at the input increases
exponentially with a time constant that is inversely proportional to the gain bandwidth
(GBW) product of the latch. When the input to the comparator is very small, the
regeneration time of the latch becomes a significant part of the clock period and the
output of the comparator cannot reach one of the decision levels (1 or -1). This causes
the zero crossing time of the output (propagation delay of the latch) to be a variable
60
quantity depending on the input value. This effect is called metastability and it causes
uncertainty in the comparator output. The comparator output drives the DACs and the
comparator metastability has the same effect as clock jitter in that it causes a random
variation of the rising/falling edges and increases the noise floor. The metastability
problem in CT ADC has been analyzed in detail in [37], [42] and it results in
degradation of the SNR at the ADC output. Since the metastability has an effect similar
to the clock jitter, the amount of SNR degradation can be estimated by calculating an
effective rms value for the edge variations in DAC pulses due to metastability only (j
seconds) and substituting that value of j in (3.6) and (3.7). The most common approach
to overcome this problem is to use a cascade of latch stages in order to provide more
positive feedback (gain) for the regeneration process, which results in lower uncertainty
of the digital output. The main drawback of this approach is that an additional half cycle
delay (z-1/2) in the feedback loop results by the addition of two digital latch stages, each
driven by opposite differential clock phases. This problem of additional delay can be
overcome by considering the delay in the loop transfer function of H(z) while using the
The rising edge of the comparator clock starts the process of sampling and
quantization, and the comparator drives the DAC to inject current pulses back to the
continuous time bandpass filter. The ideal response of the DAC current pulses is delayed
in time due to finite switching time of transistors in the feedback path (comparator and
61
Digital
Continuous time
Vin bitstream
Comparator
bandpass filter
Feedback DAC
Clock
Time
DAC blocks). The delay (in time) between the ideal response and the actual response of
the DAC current pulses is called excess loop delay and is illustrated in Fig. 3.7.
e-sd and in discrete domain as z-(d/Ts), where Ts is the time period of the clock. The effect
of delay can be modeled by including the delay block representation in cascade with the
loop transfer function (H(s) or H(z)) of (3.3) , which includes the delay effect while
calculating the impulse invariant transform. The effect of excess loop delay CT BP
ADC has been analyzed in detail in [37] and [43]. The excess loop delay modifies the
loop transfer function and its effect is observed at the ADC output in two ways. The
62
quantization noise floor in a CT BP ADC increases with excess loop delay and
degrades the noise floor. The amount of SNR degradation depends on lot of factors like
the order of ADC, value of delay in each of the DAC feedback paths (each path affects
the SNR differently as they modify the transfer function differently), type pf DAC pulse
shape etc. Another harmful effect of excess loop delay is that it increases the order of the
modulator when the impulse invariant transform of (3.3) includes the delay block. The
increase in the order of the ADC can lead to stability problems when the amount of
delay is very high, which should be carefully taken into account in simulations and the
design process. One possible solution to avoid excess loop delay problems is to adjust
the ADC coefficients of the CT BP ADC in order to restore the loop transfer function
which is modified by the loop delay. Another solution is to decrease the out-of-band gain
energy stored
Q = 2 * per unit time (3.8)
average power dissipated
Q is dimensionless and gives an indication of the amount of losses present in the system.
When the value of Q is high, the losses in the system are low and vice versa. In a LC
filter, the inductor stores magnetic energy and capacitor stores electrical energy and
energy is exchanged between L and C every cycle in a resonator, with losses due to the
63
resistor. The quality factor for an LC tank can be derived by applying the definition of
R 1
Q LC tan k = ;o = (3.9)
0 L LC
For bandpass filter, Q is also defined as the ratio of center frequency to the -3dB
bandwidth of the bandpass filter. The Q of the bandpass filter affects the noise transfer
second order ADC. The NTF of a second order CT BP ADC with finite Q is given by
s 2 + o s + o2
NTF = Q (3.10)
s 2 + o s + o2 + As
Q
o
NTF @ s = j = (3.11)
o
o + AQ
The value of (3.11) approaches zero for very high values of Q (it approaches large
negative dB values in log magnitude). The effect of the bandpass filters finite quality
factor (Q) on the NTF of a CT BP ADC is shown in Fig. 3.8. The quantization noise
floor for a CT BP ADC reduces for high Q values and results in high SNR. The gain
of the bandpass filter at its center frequency (s=jo) is given by (AQ/o), which
increases for higher values of Q. The large results in increased signal swing in the
internal nodes of the ADC (output node of the bandpass filter). The increased signal
swing levels may cause clipping of the voltages and may result in unstable behavior.
64
Thus, the upper limit for the Q value is determined by the loop stability. The optimum
value of Q is a tradeoff between the maximum attainable SNR and stable operation.
Feedback DAC
| NTF|
dB
Q = 20
Increasing Filter Q
Q = 50
Q = 100
o Freq
Fig. 3.8. Effect of the bandpass filters finite quality factor (Q) on NTF
65
The open loop transfer function of a CT BP ADC depends on the exact shape
of the DAC pulse and non-idealities occur when the rise and fall time of the DAC pulse
shapes are not equal [37]. This effect is shown in Fig. 3.9, where the rise time of the
DAC pulse is high and the fall time is almost zero for two different bit patterns from
Current
+1 -1 +1
Time
Current t rise
+1 +1 -1
Tclk Tclk
Time
Area of DAC pulse when digital output is high
Fig. 3.9. Effect of unequal DAC pulse rise and fall times
66
the comparator output {+1,- 1,+1} and {+1,+1,-1}. The area of the DAC current pulses
when the digital output is high (dark shaded region) and when the digital output is low
(lightly shaded region) for the two bit patterns are different. This effect is also called
inter symbol interference (ISI). The area is important because it represents the time the
feedback DAC current is processed by the continuous time filter. Ideally, the ordering of
the same set of bits should not change the area, but it differs when the rise and fall times
are not the same. This error is injected directly at the input for the first stage DAC,
which is not noise shaped by the feedback loop and results in performance degradation.
Higher order modulators produce non-deterministic bit streams and hence, probabilistic
estimates are required to analyze this effect. The effect of unequal rise/fall times of DAC
pulses on CT ADC was done in [44] and it was shown that asymmetric DAC pulses
results in white noise at the output spectrum, which reduces the quantization noise floor
and the SNR. The rise/fall time of the DAC should satisfy the following condition for a
desired SNR
4 * Tclk * OSR
t rise / fall (3.12)
SNRdesired
where Tclk is the time period of the clock and OSR is the oversampling ratio. There are
two approaches that have been proposed to overcome this problem. The first one is to
use fully differential DAC rather than single-ended DAC outputs, as the sum of two
waveform. The other approach is to use RZ DAC rather than NRZ DAC, since the area
for different bit patterns are same in the case of RZ DAC because they have the property
The center frequency and Q of a continuous time loop filter, unlike a switched
variations. The continuous time bandpass filter could have gm-C, LC or active RC
implementations and all these filter types have a typical 30% variation in the location
of their center frequency and Q. An automatic tuning loop is usually needed to fix the
center frequency and Q to the desired value [45]. The quantization noise floor is
attenuated to the maximum possible extent only when the center frequency and Q are at
the required value and any deviation will results in lower SNR. This issue is addressed in
(SC) architecture (H(z) and DAC coefficients in Fig. 3.3), which is used to derive an
equivalent H(s) using the impulse invariant transformation. The important choices for
the SC architecture are single loop or multi-stage (MASH), number of comparator bits
and the relationship between clock frequency and the frequency of the input signal. The
architecture. The choice of DAC pulse shape plays a very important role in the
ADC architectures for RF digitization will be discussed to identify some of the main
The first topology that is analyzed is a fourth order, single loop, single bit CT BP
ADC architecture that was reported in [28] for digitizing signals around 1 GHz with
a 4 GHz clock frequency and LC bandpass filters were used for the loop filter
implementation. The required H(s) is calculated using (3.3) from a fourth order SC BP
the feedback path. The main issue in using only NRZ DAC in a LC bandpass ADC
is the lack of enough injection nodes to realize the required notch NTF with full control
over tuning of the coefficients. The LC filter is a second order block and has only one
Vin Gs Gs Dout
z 1
s + o2
2
s 2 + o2
k1 k2 k3 k4
RZ DAC
HRZ DAC
injection point where the feedback DAC can be connected in order to realize a transfer
function, in contrast to a gm-C filter implemented using a two integrator loop, which has
two injection points (this difference is explained further in section 4.1 of Chapter IV).
This architecture overcame the problem by using a combination of both RZ and HRZ
DACs, as shown in Fig. 3.10. Two feedback paths have RZ DAC while the other two
have HRZ DAC and this combination results in full control of tuning all the coefficients
of the NTF. The main problem with this architecture is that a combination of RZ/HRZ
DAC is more susceptible to clock jitter and it degrades the SNR of ADC by more than
12 dB (2 bits) as compared to only NRZ DAC. The ADC architecture with RZ/HRZ
to the NRZ DAC implementation. The measured SNR of the ADC reported in [28] was
The second topology that is analyzed is a fourth order, single loop, three bit CT
BP ADC architecture that was reported in [26] for digitizing signals around 1.3 GHz
with a 4.3 GHz clock frequency and LC bandpass filters were used for the loop filter
implementation. The architecture used for the ADC implementation is shown in Fig.
3.11. It uses only NRZ DAC and it overcomes the problem of control of NTF
coefficients by using variable delays of 1.1*Tclk, 1.6*Tclk, Tclk and 0.7*Tclk in each of the
DAC paths (Tclk is the time period of the clock used in the comparator). This approach is
not practical since it becomes difficult to precisely control fractional delays at GHz clock
frequencies. The time period of the clock used in this design was 230 ps and 0.1*Tclk is
70
23 ps. The implementation of this fractional delay which is necessary for accuracy of
this architecture is not possible over PVT variations. The measured SNR of the ADC
reported in [26] was 62 dB in a 1 MHz bandwidth around 1.3 GHz, while dissipating
6200 mW of power and it was implemented in Indium phosphide (InP) HBT technology.
LC filter LC filter
Comparator
Vin Gs Gs Dout
s 2 + o2 s + o2
2
k1 k2 k3 k4
d1 d2 d3 d4
NRZ DAC
th
dn Time delay in the n feedback path T
clk
Time period of comparator clock
intermediate frequency range (50 MHz 200 MHz) because large values of inductors
and capacitors are required, which are accompanied by area penalty. The IF bandpass
filters use gm-C or active-RC implementation which have two injection points and hence
71
can implement the required NTF (using only NRZ DAC) with full control of tuning the
coefficients.
ADC design was presented in this chapter. The design procedure of a CT BP ADC
utilizes the impulse invariant transformation to derive a continuous time modulator from
its discrete time equivalent architecture. The main advantages and non-idealities of the
CT BP ADC were discussed and clock jitter is one of the main non-idealities that
degrades the SNR at high sampling frequencies. The choice of DAC pulse shape
changes the loop transfer function and it affects the performance and non-idealities of
the ADC. The NRZ DAC pulse shape is the optimum choice for a CT BP ADC and it
is used for both the RF and IF ADC designs in this research work.
72
CHAPTER IV
4.1. Introduction
RF input
Antenna
FB
RF BPF
The target application that is chosen for the ADC design is a software radio
architecture operating in the GSM band and is shown in Fig. 4.1. The primary GSM 900
(P-GSM) standard occupies frequencies from 935 MHz to 960 MHz for its mobile
receive (RX) band. The channel spacing (distance between the centers of two adjacent
channels) is 200 kHz and the modulation format used is Gaussian minimum shift keying
(GMSK), which is a frequency modulated signal preceded by a Gaussian low pass filter.
The carrier amplitude in GMSK is not altered according to the digital bit stream and it
uses a constant envelope modulation scheme. The bandwidth specification of the ADC
73
depends on the total number of channels that have to be digitized at its input. The
maximum bandwidth for this case is 25 MHz (the whole band) and the minimum
bandwidth is 200 KHz (single channel). The SNR requirement of the ADC can be
calculated by considering the noise floor (sensitivity at receiver input), signal power and
the required bit error rate (BER) at the receiver output. The required SNR of the ADC
BW
SNRADC = Pc PSDnoise + 10 log NF + 10 log(m) 10 log(OSR ) (4.1)
m
where Pc is the power level of the carrier, BW is the bandwidth of all the channels, m is
the number of channels, (BW/m) is the channel spacing, PSDnoise is the fundamental
noise floor due to atmospheric noise at the antenna (-174 dBm/Hz at room temperature),
NF is the total noise figure from the antenna to the ADC input and OSR is the
oversampling ratio of the ADC. The OSR depends on the choice of the clock frequency,
the noise figure depends on the implementation of the front end blocks (RF bandpass
filter, LNA and VGA) and the other parameters in (4.1) are specified by the GSM
standard.
discussed in Section 3.5. The power consumption is very high (0.45 to 6.2 Watts) and
the CT BP architecture using LC filters cannot use NRZ DAC to reduce clock jitter
effects. The two main objectives of this design is to obtain high resolution with low
power consumption (in the mW range) and to explore alternate architectures for
using a fourth order, single loop and single bit CT BP architecture with LC filters.
The fourth order architecture is chosen as an optimum choice between performance and
stability considerations. Higher order modulator result in higher SNR, but are prone to
unstable behavior [15]. The simple single loop, single bit architecture is chosen to keep
the power consumption to a nominal value, which can increase for more complex
be placed at any frequency between DC and Fs/2, where Fs is the sampling frequency of
the clock. The input signals are located around center frequency Fo with bandwidth Fin,
and the choice of the ratio Fo/Fs is a tradeoff among sampling frequency (related to speed
of the whole system), anti-aliasing filter requirements and the OSR [47]. The optimal
solution to the problem is to place the center frequency (Fo) at one-fourth of the
sampling frequency. This selection of clock frequency offers two main advantages. The
first advantage is that the loop filter implementation becomes simple by using the low
pass to bandpass transformation (z-1 -z-2). The second advantage is that the digital
The OSR (defined as Fs/[2*Fin] for the BP modulator) of the ADC is fixed by the
The choice of architecture and the noise transfer function (NTF) of the ADC
attenuated by the notch NTF as long as it is the dominant noise source at the output,
which is true when a narrow bandwidth is under consideration. For a wide bandwidth
75
system, the integrated thermal noise floor will have a higher level than the quantization
noise which necessitates a different design approach for obtaining the best SNR. The
zeros of the NTF are all placed at a single frequency (center frequency of the filter) for a
narrow bandwidth ADC, which ensures maximum possible reduction of the noise at the
center frequency. The zeros of the NTF can be spread over the entire bandwidth for a
wide bandwidth system, which results in an almost flat transfer function (combination of
several closely spaced shallow notch functions) in the bandwidth of interest. Multi-bit
comparator designs are also commonly used in wideband architectures at the cost of
more power consumption. The digitization of the entire GSM band (25 MHz) is not
considered because of power dissipation constraints. The two bandwidths that are
considered in this design are 200 kHz (1 channel) and 1 MHz (5 channels), and both
Parameter Value
Supply 1.25 V
76
The target specifications of the RF ADC design for software radio application in
the GSM band is shown in Table 4.1. The maximum power level of the carrier (Pc) that
is specified by the GSM standard is -13 dBm. The number of channels is one for 200
kHz bandwidth and five for a 1 MHz bandwidth. The OSR of the ADC is 9500 for a
bandwidth of 200 kHz and 1900 for a bandwidth of 1 MHz. The combined noise figure
(NF) of the all the blocks preceding the ADC depends on the implementation. The
required SNR of the ADC can be calculated by substituting the values for the parameters
in (4.1). Assuming a NF of around 3 dB, the SNR of the ADC should be greater than 64
dB for a 200 kHz bandwidth and greater than 74 dB for a 1 MHz bandwidth. The target
power consumption is less than 100 mW and 0.25 m IBM SiGe BiCMOS technology is
chosen for the ADC implementation. The choice of the technology is based on two
The transition frequency (ft) of the bipolar transistors in this technology is 47 GHz and
they have the potential for high speed operation. The technology also offers the
advantages and flexibility of using both bipolar and CMOS transistors. The models of
the passive (inductors, resistors and capacitors) and active components have very good
correlation with the measured values. This is a big advantage for the ADC
implementation as accurate quality factor (Q) of the LC bandpass filter is essential for
good performance. The combination of these advantages makes this technology a very
The design process of the CT BP ADC begins with identifying the best
bandpass ADC architecture in the switched capacitor domain. The open loop transfer
77
function of the SC modulator is calculated and the open loop transfer function for the
discussed in Section 3.2. The process of calculation of the open loop transfer function
A fourth order SC BP ADC can be realized by using a cascade of two second order
resonator functions, as discussed in Section 2.2.3. The fourth order SC modulator can be
derived from a fourth order low pass SC prototype by the substitution z-1 -z-2, which
transforms the zeros from DC to Fs/4, where Fs is the clock frequency. The unit delay
element (z-1) in the low pass filter is substituted by a two unit delay element (z-2) with an
inversion. The loop filter (H(z)) for the fourth order SC BP modulator is given by
2 z 2 + z 4
H ( z) = (4.2)
(1 + z )
2 2
78
There are two other design consideration which determine the open loop transfer
function of the CT BP ADC. The first design consideration is the type of feedback
DAC implementation NRZ (non return to zero), RZ (return to zero) or HRZ (half
return to zero) DAC, as discussed in Section 3.3. The second design consideration is the
implementation of one clock period delay (z-1) in digital domain after the comparator
(this technique helps to reduce the metastability problem of the comparator, which is
critical especially at GHz clock frequency). The z-1 delay can be accounted in the design
procedure by factoring it out of (4.2). The modified H(z) that is used is given by (4.3),
which together with the clock period delay results in the original transfer function.
2 z 1 + z 3
H mod ified ( z ) = (4.3)
(1 + z )
2 2
NRZ DAC is used for the feedback DAC to mitigate the effects of clock jitter. Then,
H(s) in Fig. 4.2 is derived using the modified H(z), the transfer function of NRZ DAC in
the impulse invariant transform equation given by (3.3). For a fourth order CT BP
ADC with one clock cycle delay in the feedback loop, the open loop transfer function
4
1 s 3 s2 3 2 s 3
3 2
+ + 2 + + + *
2 4 T 16 4 T 8 16 T 3 4 2T (4.4)
H (s) = 2
2 2
s +
2T
79
where T is the time period of the comparator clock and the center frequency of the
bandpass filter, o= (/(2T)). The out of band gain of the noise transfer function (NTF)
is an important parameter which affects stability and as a rule of thumb, should be less
than 2 [49]. In this design, the value is chosen to be 1.4, to give some room for PVT
variations.
Vout
V Vout
in Gm
Gm
V
in
C L Gm
C
Gm
Only one injection point for feedback DAC C
The implementation of the transfer function given by (4.4) using LC filters and
NRZ DAC with full tuning of the loop coefficients poses some practical problems, as
Fig. 4.3, which shows two possible second order bandpass filter implementations LC
filter and gm-C filter. The second order LC filter just has one injection node where the
feedback DAC current can be injected in order to implement a transfer function. This is
80
in contrast to a second order gm-C filter realized using two integrator loop, which has
two injection nodes where the feedback DAC current can be injected to implement a
transfer function. The lack of enough injection nodes for transfer function realization is
the main reason why NRZ DAC cannot be used along with LC filters in a CT BP
ADC. The new architecture that is proposed in this research work overcomes this
L C -R L C -R
Comparator
Integrator b1 Current b3
DAC b2 DAC b4
s s
NRZ DAC
Feedback DAC's
The block diagram of the proposed ADC architecture is shown in Fig. 4.4. The
loop filter in the forward path is a fourth order continuous time LC bandpass filter that is
enhancement block (negative resistor, -R) to replenish the losses in the inductor and
81
capacitor. The transconductance blocks, gm1 and gm2, are used to convert the input
voltage into current, which is then injected into the LC tank. The comparator samples the
filter output and gives a single-bit digital bit stream as output. The feedback path is a
linear combination of four paths; two quasi-lossless integrator DAC paths and two
current DAC paths. The DAC blocks convert the digital output to current (analog
domain), which is injected back into the LC tank. The main difference between a current
DAC and the integrator DAC is that the latter is frequency dependent, which is evident
Vin As As
Vout
gm 1
s + o2
2
gm2 s + o2
2 z 1
b1 b3
b2 b4
s s
NRZ DAC
The transfer function of the proposed ADC architecture can be calculated from
Fig. 4.5. Ignoring the losses in the LC tank, the open loop transfer function from the
ADC output (Vout) to the comparator input (calculated by opening the loop) is given by
82
3 2 2 2 2 2
s Ab4 + s ( A g m 2 b2 + Ab3 ) + s( A g m 2 b1 + A o b4 ) + A o b3
H loop ( s ) = (4.5)
2 2 2
(s + o )
where
1 2 1
A= ; o = (4.6)
C LC
The transconductance value of the first filter stage, gm1, does not affect the loop
characteristics since it doesnt appear in (4.5). The main role of gm1 is to transfer the
input voltage into current and its value determines the dynamic range at the input of the
ADC. The transconductance value of the second filter stage, gm2, affects the output
signal swing and the loop transfer function; its value is chosen as an optimum tradeoff
between linearity, noise, signal swing and power consumption considerations. The center
frequency of the bandpass filter (fo) is at of the clock frequency, thereby reducing
digital demodulation to simple multiplication by {-1, 0, 1}. The value of inductor (L)
depends on technology and geometry, and is chosen to optimize its quality factor at the
calculated from (4.6), when L and o are known. One of the main features of this
architecture is that it provides independent control for each of the terms in the numerator
of (4.5) while using only NRZ DAC, which alleviates clock jitter problem. The s3 and
respectively. Once these two coefficients are fixed, the s2 and s terms are controlled by
coefficients b2 and b1, respectively. The value of each of the coefficients in the
numerator of (4.5) affects the poles and zeros of the NTF, and the precise control of each
83
of the coefficients is required in order to guarantee the proper shape of the NTF. The
individual control of each of these coefficients ensures good control of the NTF while
tuning them, which results in good SNR. The open loop transfer function calculated
given by (4.4). The loop coefficients b1, b2, b3 and b4 can be calculated by equating each
coefficients of the numerator in (4.4) and (4.5) respectively. The main feature in the
DAC block which helps in realizing the required transfer function. The operation of the
The integrator DAC in the proposed ADC architecture takes a digital NRZ bit
stream as input and gives an integrated current pulse stream as output. The typical output
bi
s
The integrator DAC shown in Fig. 4.6 has an ideal transfer function (bi/s), but this
representation poses some practical limitations. The ideal integrator has a pole at = 0
and the phase shift at j= 0 is 90. It is difficult to implement ideal integrators due to
finite DC gain of amplifiers and finite quality factor of integrated capacitors. The output
swing of all internal nodes in the ADC should be within the supply voltages (Vdd/Vss)
at all times for proper operation. The lossless integrators are prone to reach saturation
'
g mb1 g mb1
H int,ideal ( s ) =
sC1
Voltage at node Vx
1 1 1 1 1 Comparator
Vx To LC Vdd
output gmb1 gmb1'
resonator
C1
Possibility of output
Integrator DAC clipping to Vdd/Vss
gmb2 To LC
gmb2 resonator
Current DAC Vss
Fig. 4.7. Schematic of the feedback DAC blocks with ideal integrator DAC
The simplified schematic of an ideal integrator DAC along with its transfer
function is shown in Fig. 4.7. The transconductor gmb1 converts the digital output to
current, which is integrated by the capacitor C1. The output voltage at node Vx has to be
converted into current in order to be injected back into the LC tank, and the V-I
conversion is performed by the transconductor gmb1. When a NRZ bit stream is given as
input to an ideal integrator, its effective DC voltage is integrated at node Vx. When a
85
large consecutive number of 1s or 0s are present at the input, the output of the ideal
integrator may clip to either of the supply voltages, as shown in Fig. 4.7. The hard
clipping of the integrator output voltages at node Vx to either of the supply rails disables
the integrator DACs from the negative feedback loop and they either inject or extract a
constant amount of current every clock cycle. When this condition occurs at node Vx, the
loop transfer function of the ADC changes drastically and this results in SNR reduction
g g'
mb1 mb1
C1
1
H int,lossy ( s ) = , a1 = Voltage at node Vy
s + a1 R1C1
Vdd
Comparator Vy
1 1 1 1 1 output gmb1 gmb1' To LC
resonator
C1 R1
gmb2 To LC
Vss
gmb2 resonator
Current DAC Output is between
supply voltages
Fig. 4.8. Schematic of the feedback DAC blocks with lossy integrator DAC
The simplified schematic of a lossy integrator DAC along with its transfer
function is shown in Fig. 4.8. The addition of the resistor R1 to the ideal integrator adds
extra losses and the lossy integrator has a pole at =1/(R1*C1). The use of lossy
integrators, as shown in Fig. 4.8, instead of ideal integrators alleviates the hard clipping
issues as it decreases the integration step and reduces the probability of clipping for the
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same number of consecutive 1s or 0s at the integrator output node Vy. The probability
of clipping at the integrator output is directly related to the integration step (incremental
step of output voltage every clock cycle), which is dependent on the circuit parameters.
For every cycle that the digital input is high(low), a certain amount of current is injected
(extracted) to the capacitor, which causes an increase (decrease) in the output voltage.
Thus, the integration step depends on the value of gmb1, C1 and the losses. A larger step
every cycle will cause the output to clip for a relatively short number of consecutive 1s
or 0s, and vice-versa for a smaller integration step. This concept can be understood
better by considering the time domain response for ideal and lossy integrators to a step
input (consecutive 1s or 0s in time domain). The output at node Vx for a unity step
g mb1 t g mb1
Vx ( t ) = Vx ( 0 ) + dt = Vx ( 0 ) + t (4.7)
C1 0 C1
where Vx(0) is the initial condition of the output at t = 0. The linear ramp eventually hits
the saturation limit (supply voltages) and disables this feedback path. Similarly, the
output at node Vy for a unity step input at any given time, t, is given by
1
t
g mb1 R1C1
Vy ( t ) = Vy ( 0 ) + e (4.8)
C1
where Vy(0) is the initial condition of the output at t = 0. The exponential time constant
settling effect in (4.8) causes the lossy integrator output to have soft clipping, rather than
hard clipping and this extends the operating range of the integrator. The difference in
time domain behavior can be summarized by the following statement if the output
87
given by (4.7) clips to the supply rail for n consecutive 1s or 0s, then the output given
by (4.8) would clip for at least twice the number (2*n) of consecutive 1s or 0s. The
output of the lossy integrator to a step input (consecutive 0s or 1s) is given by (4.8),
settles to 63.2% of its final value in one time constant ( = RC seconds), to 86.6% of the
final value in 2* seconds and to 99.3% of the final value in 5* seconds. The output of
the lossless integrator to a step input is given by (4.7), which is a linear ramp. Let us
consider the case when the ramp signal (output of the lossless integrator) hits the rail in
in the integrator such that the time constant of the RC network at its output is , then the
output reaches only 87% of the final value in 2* seconds (the output doesnt clip for
(resistor R1) improves the ADCs operating range for good SNR performance and makes
it more robust. The operating range of the ADC refers to the integrator DAC tolerating a
wide variety of input bit patterns without undergoing hard clipping at its output. The
effect of the losses in the integrator DAC on the ADC performance was verified using
simulated in Simulink with both ideal and lossy integrator DACs and the output
spectrum of the ADC (calculated using Fast Fourier transform (FFT) of the digital bit
stream) for both cases is shown in Fig. 4.9 where amplitude of the input signal is 120
mV. The ideal integrators are assumed to have infinite DC gain, while the lossy
88
Fig. 4.9. Output spectrum of the ADC architecture with and without integrator losses
integrators have a nominal DC gain of about 40 dB. The behavioral model contains non-
filter, limited signal swing at all internal nodes, clipping of node voltages when it
reaches the supply voltages etc. The solid line in Fig. 4.9 shows the output spectrum of
the ADC with lossy integrator DAC while the dotted line is the output spectrum of the
89
ADC with ideal integrator DAC. For the case when the output of ideal integrator clips
(especially for large signals), the loop transfer function changes drastically and increases
the in-band noise floor, resulting in a severe degradation of SNR at the output of the
ADC.
There are two other approaches to decrease the integration step at node Vx in Fig
4.8. The first approach is to increase the value of the capacitor (C1), while maintaining
the value of gmb1. The value of the loop coefficient is given by (gmb1*gmb1)/C1 and in
order to get the required coefficient, the value of gmb1 has to be increased
due to area constraints, which usually limit the capacitor to a reasonable value. The
second approach is to decrease gmb1 and simultaneously increase gmb1 such that the
overall coefficient remains constant. gmb1 cannot be scaled down arbitrarily as its lower
limit is determined by noise considerations and proper operation at the clock frequency
(in the GHz range). Due to the limitations of these two approaches, the more practical
approach of introducing losses through resistors is used to decrease the integration step
at node Vx.
There are two non-idealities that modify the required open loop transfer function
of the proposed architecture, given by (4.5). The first non-ideality is the finite Q of the
bandpass filters and its main effect is the increase of the quantization noise floor, as
discussed in Section 3.4.4. The other non-ideality which modifies the open transfer
function is the introduction of losses in the integrator DAC. The ideal integrator DAC
transfer functions in Fig. 4.4 ((b1/s) and (b3/s)) are replaced by the lossy integrator DAC
90
transfer functions ( (b1/(s+a1)) and (b3/(s+a2)), a1 and a2 represent the losses in the
integrator ). The loop transfer function is calculated from the ADC output to the
comparator input and the modified open loop transfer function due to the use of lossy
s 3 a 2 Ab3 + s 2 a1 ( a 2 Ab3 + A 2 g m 2 b1 ) + sa 2 ( a1 A 2 g m 2 b1 + A o2 b3 ) + a2 a1 A o2 b3
H error ( s ) =
( s 2 + o2 ) 2 ( s + a1 )( s + a 2 )
(4.10)
where a1 and a2 represent losses in the two integrator DACs, ai = 1/RiCi, refer to Fig.
4.8, o and A are given by (4.6). For operation in a narrow bandwidth around the center
frequency, the error transfer function can be simplified under the assumption that a1, a2
<< o (a1, a2 can be neglected in the denominator, numerator terms which are divided by
s a Ab + a1 ( a 2 Ab3 + A 2 g m 2 b1 )
H error ( s ) 2 3 2 2 2 around = 0 (4.11)
( s + o )
The error transfer function given by (4.11) represents a small deviation from the required
loop transfer function provided the values of a1 and a2 are small. The earlier analysis of
clipping at integrator outputs led to the conclusion that relatively large value of a1 and a2
is required in order to prevent clipping at the integrator output for robust ADC operation.
The value of a1 and a2 used in this design is a tradeoff between these two opposing
considerations and is chosen after extensive behavioral simulations. The value of a1 and
a2 used in the ADC implementation is 6.67*108 rad/sec. At the resonant frequency, the
91
main effects are small deviations of the first two coefficients in the loop transfer
function, as shown in (4.11). These errors can be accounted for by changing the values
of the loop coefficients (b1, b2, b3 and b4) in order to get the required values in (4.5) and
are taken into account when the component values are calculated in the design. The
behavioral simulations of the proposed ADC architecture was used to optimize the
coefficients for best SNR and low power consumption (coefficient scaling) while
maintaining reasonable signal swing at all internal nodes. The details of the behavioral
Simulink and Matlab software simulation tools were used for the behavioral
in Simulink for behavioral simulations is shown in Fig 4.10. The model uses sub-blocks
available from Simulink library to obtain the required loop transfer function. The s-
domain (continuous time) transfer function block is used to realize the bandpass filters.
For the resonator, d1 = o/Q, d2 = o2 = 1/(L*C) and A = (1/C), where o is the center
frequency and Q is the quality factor of the bandpass filter. The integrator DAC with
losses is implemented using the integrator sub-block, and current DAC is implemented
using a simple multiplier block with the required coefficient to convert the digital bit
into current. The saturation block is used at the output of bandpass filters and integrator
blocks to limit the signal swing to the supply voltages (Vdd/Vss). The saturation block is
Input
gm1 Constant
1
A.s s
gm2
s2+d1.s+d2 Pulse Transport
Saturation A.s Generator Delay
xo
s2+d1.s+d2
Transfer Fcn Saturation1
Integrator
Transfer Fcn1
b4
Transfer Fcn3
Saturation3
b3
Comparator
s+a1 4
Transfer Fcn2
Saturation4
b1
s+a1
b2
yout yout
92
93
not required for the current DAC as its output is a digital current pulse stream that
follows the bit stream input. The summation of signals before each bandpass resonator
occurs in current domain, which matches with the circuit implementation. The
summation of the signals for the feedback path has a positive sign in the model, though
the ADC is a negative feedback loop system. The main reason for this implementation is
due to the fact that the overall phase shift of signals in the loop (together with the one
Sampling operation
Input
Digital
Clock Trigger
Transport Relay output
Integrator delay block
block
The comparator is the only block which operates in discrete time mode and it
performs the sampling and quantization operations every clock cycle. The behavioral
model of the comparator in Simulink is shown in Fig. 4.11. The sampling operation is
performed by grounding the input of an integrator block and using the bandpass filter
94
output as the initial condition. The rising edge of the clock signal is used as a trigger to
capture the value of the filter output as the initial condition and since the input is zero,
the output effectively samples the initial condition every clock cycle [50]. The z-1 delay
(one clock period) is implemented using the transport delay block and the single bit
The input signal is a sine wave at 950 MHz and the sampling clock frequency is
3.8 GHz. The initial values of the coefficients are calculated by equating each
coefficients of the numerator in (4.4) and (4.5) respectively (the initial values of the
coefficients b1, b2, b3 and b4 calculated using this approach are 5.53*107, 0.0098, 7.9*107
and 0.0146, respectively). The behavioral simulation in Simulink has various parameters
which determine whether the simulation is in continuous time or discrete time domain.
The continuous time ode5 (Dormant-Prince) solver is used with fixed step option and the
size of the step determines the accuracy of the simulation. A step size of 0.05*Ts
seconds (Ts is the time period of the clock) is used for the model, which ensures very
good accuracy of the behavioral simulations. At the architectural level, the loop
coefficients and filter gain can be scaled down to reduce power consumption. The
scaling of these two values results in scaling down of the signal swing at all the internal
nodes of the ADC. The lower limit for scaling is determined by the noise level for the
required dynamic range at the output of the ADC. For the integrator DACs, the main
consideration is the amount of losses required to ensure that the output doesnt clip a
ADC is done for various values of amplitude for the input sine wave signal. The lowest
95
level is the input amplitude when the output SNR is zero dB (signal power is equal to the
noise power), while the largest level is the input amplitude when the total power of the
distortion components is higher than the noise power (integrated over the signal
bandwidth). The signal swing at all internal nodes of the ADC must be within the supply
voltages over the entire input signal range. This ensures stable operation of the ADC and
Coefficient Value
b1 5.13*107(Amps*rad/Volt*sec)
b2 0.0091 (Amps/Volt)
b3 7.44*107 (Amps*rad/Volt*sec)
b4 0.0139 (Amps/Volt)
The SNR of the output bit stream is calculated by dividing the power of signal
over the quantization noise power in frequency spectrum. The power spectrums are
estimated using the fft function in Matlab, and the bit stream is first preprocessed using a
Hanning window in order to prevent the leakage in frequency spectrum caused by taking
the Fourier transform of a truncated bit stream. The value of the optimized loop
coefficients that is used in the implementation of the ADC is shown in Table 4.2. The
96
values, which decreases the power consumption and also brings down the signal levels
closer to the noise floor. The optimization process should also maintain reasonable
signal swing at all internal nodes of the ADC for all possible values of the input signal.
The coefficients were optimized as tradeoff between power and noise, while obtaining
the best SNR at the output of the ADC. The circuit implementation and simulation
The main building blocks of the CT BP ADC are the LC bandpass filter,
integrator DAC, current DAC and the comparator. The details of the circuit
implementation and post layout simulations results of these building blocks are
presented in this section. The ADC is implemented in IBM 0.25 m SiGe BiCMOS
technology, which offers the flexibility of using both bipolar and MOS transistors. High
speed bipolar junction transistors (BJT) of this technology have a transition frequency
(fT) of 47 GHz and are used for transistor implementation in the signal path.
transconductance stage and then injected into the LC tank. The differential output
V
Vod = I tail RL tanh id (4.12)
1+ 2VT
where is the collector current gain of the BJT, Vid is the input differential voltage, Itail
is the tail current, RL is the load resistor and VT is the thermal voltage (VT = kT/q = 26
electronic charge). The main considerations in the design of the transconductor are
linearity, noise and power consumption. The nominal input voltage range (Vid) for linear
operation is around 20 mVp-p and emitter degeneration resistors (Re) are used to enhance
the input linear range. The effective transconductance of a emitter degenerated BJT
gm
g m,diffpair = (4.13)
1 + g m Re
where gm = (Ic/VT), Ic is the collector current (Ic = Itail/2) and gm*Re is called the emitter
degeneration factor. When gm*Re >> 1, the effective transconductance of the linearized
differential pair is (1/Re). The third harmonic distortion (HD3) of a emitter degenerated
2
1 Vid
HD3 = (4.14)
24 (1 + g m Re )VT
The third order intermodulation distortion (IM3) is three times the value of HD3 [3]. The
HD3 decreases by a factor of (1+ gm*Re)2 as compared to a differential pair without any
linearization, at the cost of higher noise and power consumption. The addition of
resistors Re also increases the noise floor of the differential pair. The total input referred
8qVT2
V 2
in ,noise = (1 + g m Re ) + 8kTrb (4.15)
I tail
where rb is the base resistance of the BJT. The first term of (4.15) refers to the shot noise
contribution of the BJT, which increases by a factor of (1+ gm*Re) as compared to the
case without any degeneration. The emitter degeneration factor used in this design is 5 as
an optimum tradeoff between noise, linearity and power consumption, which results in
an input linear range of 120 mVp-p for the BJT differential pair.
VDD
Rcm
L V L
f
C C MOS in triode region
Vo- Vo+
Vi+ Q1 Q1 Vi- Q2 V Q2
q
Re Re Mq
I Rp I Rp
E P
V M1 V M2
b1 b2
VSS
Fig. 4.12. The differential pair formed by transistors Q1 performs linear V-I conversion
and the current is injected into the LC tank. The Q-enhancement block to replenish the
losses in the LC tank is realized using a cross-coupled BJT differential pair formed by
transistors Q2, which have emitter degeneration (Rp in parallel with triode transistor) to
enhance its linearity. The linearization minimizes the distortion components that are
introduced in the LC tank from the cross-coupled pair, but reduces the quality factor (Q)
tuning range of the bandpass filter. The gate control voltage (Vq) of a NMOS transistor
(Mq) operating in triode region is used to tune the Q of the bandpass filter. Mq is
connected in parallel with a linear resistor to avoid linearization degradation due to the
1
RMq = (4.16)
Wn
nCox
Ln
(Vgs Vth )
where n is the majority carrier mobility in the n-channel, Cox is the oxide capacitance,
Wn and Ln are the width and length of the NMOS transistor, Vgs is the gate source
voltage and Vth is the threshold voltage. The value of negative resistance of the cross
2
Rneg = (4.17)
g m,eff
where
100
g m2
g m,eff = (4.18)
R
1 + g m 2 R p || Mq
2
The Q-enhancement block increases the noise floor of the differential pair, with shot
noise and base resistance thermal noise contribution from Q2, thermal noise contribution
from triode transistor Mq and resistor Rp. For both the linearized differential pairs, the
tail current is placed between the two degeneration resistors, Re (and Rp). This ensures
that the noise contribution of transistors M1 (and M2) becomes common mode noise and
is rejected at the differential output, but the disadvantage is that additional voltage drops
of 0.5*Ie*Re (and 0.5*Ip*Rp) have to be tolerated in the design. This voltage drop,
along with Vbe of transistor Q1 (Q2) and Vdsat of current mirror transistor M1 (M2),
restrict the voltage swings while maintaining the transistors in the proper region of
operation. The common mode voltage at the filter output is given by Vo,cm = VDD -
It*Rcm, where It = [Ie + Ip + sum of DAC currents connected to the resonator]. The
resistor Rcm sets the common mode voltage of the filter and also for the DAC blocks that
The value of the inductor and its quality factor (Q) depends on various factors
such as geometry (shape, number of turns etc) and technology parameters (thickness and
conductivity of metal layers and insulators, resistivity of substrate etc.) [51]. The IBM
0.25 m SiGe technology offers custom inductors with a pre-determined shape on metal
layers and some variable parameters (number of turns, distance between metal turns,
hollowness etc) that are restricted to a certain range of values. The design of inductors in
this process is simplified by these restrictions, which narrows the design space during
101
the optimization process where the Q of inductor is optimized at the filters center
frequency (950 MHz) while occupying minimum area. The value of the important
parameters for the spiral inductor design is given in Table 4.3. The optimum value of
inductance (L) in Fig. 4.12 for these parameters is 5.1 nH, which results in maximum Q
at 950 MHz. The topmost metal layer is used in order to minimize the capacitance
between the metal and the substrate. A mesh layer is used to shield the spiral inductor
from the substrate, which increases the resistivity of the substrate beneath the spiral and
Parameter Value
Spiral width 15 m
Underpass width 15 m
Shape Octagon
junction varactors. The center frequency of the LC filter is tuned by varying the control
voltage (Vf), which changes the reverse bias voltage (Vr = Vo - Vf) of the varactors. The
depletion layer capacitance of the P-N junction under reverse bias conditions is given by
102
C j0
Cdep = 1 (4.19)
n
1 Vr
V
bp
where Cj0 is the capacitance for zero bias voltage, the value of n is a fitting parameter
with a value between 2 and 3, Vbp is the built-in potential of the P-N junction . The
nominal value of the differential varactor is 2.75 pF (2*C, C according to Fig. 4.12). The
varactors can be tuned by a value of 20% from the nominal value by varying the
control voltage (Vf). The parasitic capacitance at the output of the resonator (due to all
components connected at the resonator output) and is around 110 fF. The overall Q of
the LC filter is usually limited by the Q of the standalone inductor. The Q of the
standalone varactor is around 40, while the Q of the inductors is around 7. The fourth
order LC bandpass filter is implemented as a cascade of two second order LC filters. The
component and current values used in the circuit implementation for the two LC filter
blocks is shown in Table 4.4. The value of Rcm is different for the two filters as the
transconductance of each stage and the value of feedback DAC currents that connect to
them are different. A large shunt capacitor (40 pF) is added in parallel to Rcm in order to
minimize the noise injection into the LC tank. This technology has certain metal density
rules (each of the 5 metal layers need to be present at a certain density in the layout), and
the extra space required to satisfy the density rules are used to implement the shunt
capacitor. Though the area occupied by the shunt capacitor is large, it comes at no extra
penalty and it improves the noise rejection (with the exception of noise from the power
supply). The transconductance of the first stage does not affect the loop transfer function
103
Table 4.4. Component and current values for the LC bandpass filter
Re 110 100
Rp 325 325
Rcm 65 30
IE 2.3 mA 2.5 mA
IP 0.95 mA 0.95 mA
and the required dynamic range at the ADC input determines the maximum range that it
can be scaled down. When the ADC is used in a software radio system, similar to the
one shown in Fig. 4.1, the output of LNA/VGA is usually in current mode and could be
directly injected into the LC tank. This approach results in elimination of the first
transconductor, which would further reduce the power consumption of the ADC.
The dynamic range of the ADC is mainly determined by the dynamic range of
the input transconductance, Q-enhancement block and the DACs that are connected to
104
the first resonator. The lowest signal that can be handled is limited by the total input
noise contribution of these stages, while the maximum signal that can be handled
properly is limited by the distortion components. The extracted simulations for the center
Fig. 4.13. Post-layout simulation of the bandpass filter center frequency tuning
frequency tuning of the second order resonator is shown in Fig. 4.13 and the center
frequency can be tuned from 790 MHz to 1.1.6 GHz. The post-layout simulation result
for the Q tuning of the second order resonator is shown in Fig. 4.14 and the Q of the
resonator can be varied from a value of 7 to 26 using the gate voltage of the triode
transistor.
105
The distortion performance of the filter was calculated by applying two tone input (each
-7 dBm at 949 MHz and 951 MHz) to the second order resonator and the post-layout
simulation result is shown in Fig. 4.15, which indicate IM3 value of 63 dB. The
simulation was performed by injecting the two tones at the input with the output nodes
shorted by voltage sources with proper DC bias and the output current is plotted in Fig.
4.15 (power levels denote power of output current). The combined noise level of the
simulations, integrated in a 200 kHz and 1 MHz bandwidth around 950 MHz are 0.683
Fig. 4.15. Post-layout simulation of the bandpass filter two tone test
The schematic of the integrator DAC is shown in Fig. 4.16. The digital NRZ bit
stream from the comparator outputs is converted into current by the differential pair
formed by transistor Q3, and the current is integrated by capacitor C1. Finite gain of the
differential pair (gm3*[ro3 || rds4]) makes this a lossy integrator, but as explained in
Section 4.3, extra losses are required to introduce soft clipping and improve the
operating range. The resistors R1 are added at the output of the integrator to introduce
extra losses and they also set the output common mode voltage (Vcm, applied at the
107
To LC resonator
VDD
Idac- Idac+
M4 M4 M4
C1
C1
R1 R1
Q4 Q4
Vcm
Q3 Q3
Vcomp+ Vcomp- Rd
I
d1 I I
2 I d2 d2
d1
M3 V V M3 V M5
b3 b3 b5 M5
2
VSS
node between the two resistors R1) of the differential integrator. The output of the
injected back into the LC resonator. This V to I conversion has to be very linear,
especially when the current is injected back into the first LC tank where it directly mixes
with the input signal. Any non-linearity that is introduced by the integrator DAC
connected to the first stage is not suppressed by the feedback loop and it directly appears
at the ADC output. A differential pair (formed by transistors Q4) with emitter
degeneration resistor Rd is used to convert the integrated output to current; a large tail
108
current (3 mA) is used in this stage to maintain its linearity. The overall transfer function
iout I I dac R1 g m4
= dac + I d 1 sign(Vin ) (4.20)
Vin IDAC Vcomp + Vcomp 1 + s (R1 * C1) 1 + g m 4 Rd
2
where sign(Vin) denotes the polarity of the digital output bit stream (+1 for high and -1
for low). The value of coefficients (b1 and b3) of the integrator DACs from Table 4.2
can be used to determine the component values for the circuit implementation. The value
of Id1 is chosen based on the maximum integration step that can be accommodated
Table 4.5. Component and current values for the integrator DAC
R1 1.5 k 1.5 k
Rd 250 178
C1 1 pF 1 pF
Id1 1 mA 1 mA
during each clock cycle for proper operation of the ADC. The tuning of the integrator
DAC coefficient is achieved by changing the value of the current Id1. The emitter
degenerated differential pair is not used for tuning the coefficient because the tuning
range obtained by varying Id2 (varying gm4) is very small. A triode transistor could be
used for the implementation of Rd which will provide a variable resistance for the
coefficient tuning, but the downside is that the non-linearity of the triode transistor will
appear at the ADC output (which is critical especially for the integrator DAC that is
connected to the first resonator). The component and current values used in the circuit
implementation for the two integrator DAC blocks is shown in Table 4.5.
The schematic of the current DAC is shown in Fig. 4.17. The digital NRZ bit
stream from the comparator outputs is converted into current by the differential pair
formed by transistors Q5, and the current is directly injected into the LC resonator. The
current DAC acts like a simple switch by simply injecting/extracting the tail current for
logic 1 or 0 value. The value of coefficients (b2 and b4, which are represented as a
Table 4.6. Component and current values for the current DAC
fraction of the transconductances of LC filter blocks to which they are connected) for the
current DACs from Table 4.2 can be used to determine values of the tail current. The
current DAC coefficient can be tuned by directly changing the value of the current Id3.
The component and current values used in the circuit implementation for the two
To LC resonator
Idac- Idac+
Vcomp+ Q5 Q5 Vcomp-
I
d3
V M6
b6
VSS
4.5.4. Comparator
The single-bit NRZ comparator samples the output of the fourth order filter and
quantizes the sampled value to logic 1 or 0. Both the input and output of the comparator
block are fully differential, which implies a threshold of ground (power supply is 1.25
111
V) for the quantizer. The block diagram for the circuit implementation of the comparator
is shown in Fig. 4.18. The preamplifier gain stage is a simple gain stage and provides
extra amplification for signals before the quantizer. The switching action in the
comparator causes the comparator output to couple back to the input (to the filter
bandpass in this case), which degrades the signal in the form of transient noise. This
phenomenon is called kickback effect and it degrades the signal at the comparator input
(thereby causing errors in the operation of the comparator) for high clock frequencies
[13]. Emitter follower stages are used to isolate the gain stages, and before the DAC to
provide isolation and minimize the comparator kickback effect. The signal swing in the
comparator should be within the power supply ( 1.25 V) and a DC level shifter is
required before the gain stage and the first latch to maintain all transistors in proper
operating region. The latch block samples the filters output and also performs the
quantization operation. Extra latch stages are introduced to decrease metastable behavior
and the extra delay is already accounted in the loop transfer function.
NRZ
Input DC Preamp DC bitstream
Emitter level gain Emitter Latch1 Latch2 Latch3 Emitter
level
follower shift stage follower follower
shift
CLK CLK
The schematic of the preamplifier gain stage is shown in Fig. 4.19 and it consists
of a simple BJT differential pair with load resistors. The DC level shift is achieved by
112
using a C-R high pass network, where the series capacitor 10 pF is used for AC coupling
of signals and the required DC voltage is applied through a 2 K resistor. The pole
1
created by this high pass network ( f = ) is at 8 MHz, and is chosen such that all
2RC
signals propagating through the comparator lie above this frequency. The choice of this
pole location is a tradeoff due to area constraints. The input referred offset of the
preamplifier gain stage should be minimized but it doesnt degrade the performance of
VDD
R2 R2
Vo- Vo+
Vi+ Qef Qef Vi-
C C
Q7 Q7
R R
I I
ef DC level V V ef
dc dc
shifter
I
g
V Mef V Mef V
b,ef b8
M8 b,ef
Emitter VSS
follower
the ADC as it is rejected by the noise shaping feedback loop, similar to the effect of the
The schematic of the latch implementation is shown in Fig. 4.20. When the
differential clock signal is high, the differential pair formed by transistors Q8 is active
and it amplifies the input signal. At the time instant of the falling clock edge, the signal
is sampled and the value at that time instant becomes the initial condition for the next
clock phase. When the clock is low, the positive feedback cross coupled differential pair
formed by transistors Q9 is active and the output reaches either logic 0 or 1. The
VDD
R3 R3
Vo -
Vo +
Q8 Q8
Vin+ Q9 Q9
Vin-
I
d4
V M9
b9
VSS
regeneration time of the latch, defined as the time taken for the output to reach a logic
latch V
Tregeneration = ln o ,latch (4.21)
g m9 R3 1 Vlatch,t =0
where latch is the time constant of the latch (equal to the product of resistance and the
total capacitance at the output node), Vo,latch is the output voltage for either logic 0 or 1
and Vlatch,t=0 is the initial condition at the output. The regeneration time increases for very
small value of inputs and it can be decreased by increasing gm9 (increase in Id4), which
results in higher power consumption. The value of resistor, R3, cannot be increased
arbitrarily as it affects the output signal swing. A cascade of three latches driven by
different clock phases results in z-1/2 delay. The propagation delay in comparator, DACs
and routing parasitics adds to this delay and the resulting worst case total delay (from
simulations) is less than one clock period (z-1 delay). This approach of incorporating
circuit propagation delays (carefully adjusted according to post layout simulations) in the
required overall delay helps to overcome the excess loop delay problem in this design.
The blocks that can be optimized at circuit level are the comparator and transconductor
of the first LC filter, as the current consumption of other blocks are determined by the
loop transfer function. The currents in the comparator are scaled down as much as
possible while maintaining proper operation (no bit error) at 3.8 GHz clock frequency
(time period of 263 ps). The third latch consumes higher current in order to drive the
parasitic capacitance at the base of the emitter follower, which is its subsequent block.
115
The input and output waveforms of the comparator from post-layout simulations are
shown in Fig. 4.21. The differential output swing of the comparator is around 200 mV,
which is sufficient to drive the subsequent DAC stages. The differential offset at the
comparator input is around 0.4 mV and the time delay in the comparator (regeneration
time of the latches) was around 40 ps (both values are from post-layout simulations).
The comparator can resolve input signals greater than 1 mV without any errors at its
output. The output of the comparator has to drive an external load (50 of spectrum
analyzer) for measurement purpose and an output buffer is used to properly drive the bit
stream. The component and current values for the preamplifier and the latch blocks of
the comparator are shown in Table 4.7 and Table 4.8, respectively.
116
Table 4.7. Component and current values for the preamplifier gain stage
Parameter Value
R2 600
R 2 K
C 10 pF
M8 (W/L) 50 m / 0.4 m
Ig 0.9 mA
Ief 1.2 mA
Table 4.8. Component and current values for the latch stages
The output buffer is used to drive the differential bit stream into the spectrum
analyzer load and the clock buffer is used to drive the input clock signal to the
comparator. The schematic of the output buffer is shown in Fig. 4.22 and the clock
VDD
R R
L L
Vo- Vo+
I
buf
V Mbuf
buf
VSS
R Off chip resistor
L
Bondpad impedance
buffer has a similar implementation, except that the load resistors (RL) are implemented
on-chip and the bond pad impedance appears before the input rather than at the output.
DC level shifting blocks, similar to the one shown in Fig. 4.19, are used to set the correct
118
DC bias at the input and output of both the clock and output buffers. The component and
current values for the circuit implementation of the output and clock buffers are shown
in Table 4.9.
Table 4.9. Component and current values for the buffer stages
RL 150 40
Ibuf 5 mA 2.6 mA
The individual blocks described in Sections 4.5.1 to 4.5.5 are connected together
according to the architecture shown in Fig. 4.5, in order to obtain the top level circuit
Cadence by applying a test input signal at 950 MHz. Impedance matching blocks are
used both at the input and output to ensure proper coupling of all signals. The output bit
stream is written to a file using the printvs command available in Analog artist
calculator tool. The output bit stream file is processed in Matlab and the SNR of the
ADC is calculated by using the FFT command. The power spectrum at the ADC output
from post-layout simulations is shown in Fig. 4.23. The number of samples used is
16384 and the output SNR was 56 dB in a 1 MHz bandwidth around 950 MHz. The
119
number of samples is restricted by the limited computing resources, since a top level
extracted simulation can take about 5 days to generate 16384 samples. A higher number
of samples (65536 samples) are required to get a more accurate estimate of signal and
noise power values. The current consumption of the different ADC blocks is shown in
Table 4.10. The largest share of power consumption is from the comparator block and is
about 41%. The stringent linearity requirements increases the power consumption in
120
both the bandpass filter and the integrator DACs. Power optimization is done both at
architecture and circuit level while maintaining the dynamic range and stability of the
ADC. A very straightforward way to increase the signal bandwidth while maintaining
increasing the single bit comparator to a two bit comparator would require four
comparators instead of one, along with more digital blocks for encoding the bit stream.
From Table 4.10, it can be seen that this would increase the overall power consumption
increase in SNR (increasing the comparator by 1 bit increases the SNR by 6 dB). This
indicates that increasing the comparator resolution is not a power efficient technique for
overall power
enhancement
TOTAL 30 mA 100 %
121
LC Filter
Comparator
DACs
Output
Buffer
0.25m SiGe BiCMOS technology (fT = 47 GHz) through the MOSIS educational
program. The QFN64A package available from IPAC is used for this chip, which is a
high speed package that can easily process the maximum frequency of 3.8 GHz without
significant degradation of the signal. The chip micrograph is shown in Fig. 4.24 and the
ADC occupies an area of 1.08 mm2. The four inductors (each 5.1 nH) used in the two
122
LC filters occupy the largest percentage of area (around 40 %). The inductors are placed
atleast 250 m from each other in order to minimize any mutual coupling between them.
The floor plan of the layout was designed to minimize the interconnect lengths between
the various ADC blocks. This is critical, especially for the routing path from the output
of the filter through the comparator and DACs and back to the filter. Any parasitic
resistors and capacitors created due to these metal connections increases the excess loop
delay and have to be minimized as much as possible. Deep trench wells and protective
guard rings are used extensively around every ADC building block to ensure that there is
minimal coupling between them. This is especially critical for the comparator block,
where the switching operating of the clock introduces noise, which can couple through
The block diagram of the measurement setup is shown in Fig. 4.25. A narrow
band balun (EHFFD1618 from Digikey) operating in the frequency range of 880 MHz to
input signals. The single ended clock generator signal is converted to fully differential
frequency range of 3.4 to 4 GHz. The differential output bit stream is converted to a
single ended signal using the a wideband balun (5315 A from Picosecond labs) which
operates in the frequency range of 200 kHz to 17 GHz, which is then injected to the
spectrum analyzer. The CT BP ADC bit stream is equivalent to a PWM (pulse width
modulated) bit stream and its frequency content is spread over a large bandwidth, which
123
Clock
generator
Narrowband
balun
C2 V C2
clk,DC
R2 R2
Vclk+ Vclk-
Signal Spectrum
generator C1 C3 analyzer
Vin+ Vo+
R1 R3
V V
in,DC o,DC
R1 Vin- Vo- R3
C1 Wideband
Narrowband CT BP ADC C3
balun
balun
makes it necessary to use a wideband balun at the output to prevent loss of signal
information during the measurement process. The value of the input and output
impedance of the three baluns is 50 and the impedances of the connecting blocks are
matched for maximum power transfer. The DC voltages for the input, clock and output
signals (Vin,DC, Vclk,DC and Vo,DC) are applied through resistors R1, R2 and R3
the two narrowband baluns. On-chip termination of 50 is provided for the input and
124
clock signal paths (50 resistors at the base of input transconductor and clock buffer
blocks), which ensures proper coupling of signals into the chip. The value of resistor R3
at the ADC output is 150 (3 times the nominal value is used to increase the signal
swing at the output at the cost of increased reflections as an optimal tradeoff). Capacitors
C1, C2 and C3 are used to couple the high frequency AC signals (series coupling) and
provide DC blocking in the input and output ports. An on-chip output buffer that was
described in Section 4.5.5 drives the bit stream into the spectrum analyzer. A printed
circuit board (PCB) was fabricated in order to test the ADC and the photo of the
measurement setup is shown in Fig. 4.26. The signal swing of the NRZ differential
output bit stream at the input of the spectrum analyzer is around 100 mV, after the
Clock
generator
Signal generator
Wideband Balun
Spectrum analyzer
(200 kHz -17 GHz)
losses in the output balun. The value of the output swing scales the absolute value of
both the signal and noise power in the spectrum analyzer (while SNR remains constant if
the noise of the output balun is not dominant), which is the reason for attenuation of the
The single-tone output spectrum of the RF ADC is shown in Fig 4.27. The noise floor of
the ADC (including the signal generator and measurement setup noise) integrated in a
resolution bandwidth of 20 kHz was measured to be -113 dBm at 950 MHz. The output
spectrum of the ADC for two input tones at 949 MHz and 951 MHz, each at -8 dBm, is
shown in Fig. 4.28. A passive power combiner with a gain of -6 dB is used to combine
the two-tone input signals. The IM3 of the ADC for the two tone input was measured to
Fig. 4.29. Noise floor of the measurement setup with grounded inputs
Fig. 4.30. Noise floor of the measurement setup for the two tone input test
128
be -62 dB; unfortunately the noise level of this characterization increases due to the
noise contribution of the two signal generators used. Therefore, the SNR of the ADC is
calculated by dividing the signal power of the two-tone test over the integrated noise
power with a single tone test, while the distortion components are smaller than the noise
level in the required signal bandwidth. The noise floor of the measurement setup with
grounded inputs is shown in Fig. 4.29 and the value is around -132 dBm at 950 MHz.
This plot includes the noise of the input and output baluns, ADC, output buffer and the
PCB. The noise floor of the measurement setup for the two tone test is shown in Fig.
4.30. This plot includes the noise of the two signal generators in addition to the noise
sources in Fig. 4.29. The noise floor increases to -121 dBm at 950 MHz due to the noise
The measured SNR is 63 dB and 59 dB for bandwidths of 200 kHz and 1 MHz
respectively, around 950 MHz with a clock frequency of 3.8 GHz. The approach of
directly measuring the SNR from the spectrum analyzer captures all noise in the analog
domain that are not contributed by the ADC at 950 MHz, which increases the noise
floor. Another disadvantage with this approach is that it does not capture the jitter effects
properly. The noise from the jitter is added at the input of the comparator and the effect
is similar to the quantization noise because it is rejected at the output of the ADC. Since
the jitter noise is rejected by the noise shaping effect of the feedback loop, the effect of
jitter is not measured accurately by this approach. A better approach would have been to
capture the bit stream using a logic analyzer or data acquisition board, but this was not
performed due to lack of measurement equipment at GHz clock frequency. The plot of
129
Fig. 4.31. Plot of measured SNR of ADC vs input power for the RF ADC
Fig. 4.32. Plot of the measured IM3 vs input power for the RF ADC
130
output SNR over 200 kHz bandwidth vs input power is shown in Fig. 4.31 and the plot
of measured IM3 vs the input power is shown in Fig. 4.32. The performance of the ADC
architecture reduces clock jitter problems by using only NRZ DACs and achieves good
Ref [26], this design has similar SNR performance, while the power consumption is
further reduced by about two orders of magnitude. The significant reduction in power
consumption is due to factors such as supply scaling and power optimization. The power
supply is scaled down by a factor of four, which scales the power consumption but it
HBT BiCMOS
also results in design challenges due to limited room for signal swing. The power
optimization techniques used in this design further improves the power consumption
performance as compared to the designs that have been reported in literature. The ADC
design in Ref [26] has a 3-bit comparator, while the ADC in this research work has a 1-
bit comparator. Despite the difference in the comparator resolution, the current design
achieves almost the same resolution as compared to Ref [26] in a 1-MHz bandwidth,
The measurement results of the ADC shows that high resolution narrowband A/D
metrics of the ADC improves the state of the art for data conversion at RF frequencies,
but significant challenges still have to be overcome before the ADC can be used in a
software radio transceiver. The design challenges and possible future research directions
CHAPTER V
5.1. Introduction
shown in Fig. 5.1. There is no specific standard for the location of the intermediate
frequency in all wireless standards and the choice depends on the circuit implementation.
A high IF (125 MHz) is chosen in this research work to push the state of art for the IF
ADC design in CMOS technology. The bandwidth for the ADC depends on the number
of channels and the type of wireless standard. The IF ADC is designed for the GSM
IF input
FB
Antenna
LO1
standard (channel bandwidth of 200 kHz) and five channels are considered, which
receiver, which has resulted in adoption of different IF frequency locations for the
receiver implementation for the same wireless standard. The IF can usually vary from 5
MHz to 200 MHz, frequencies lower than 5 MHz are not preferred due to flicker noise
problems whereas frequencies higher than 200 MHz tend to increase the power
consumption for the IF circuits. The bandpass ADCs that have been reported in
switched capacitor circuits, which are not suitable when higher values of IF are used in
the receiver, as discussed in Section 2.2.3. The continuous time implementation is more
suited for high IF receivers with low power consumption. The three types of possible
continuous time filter implementations are LC, active RC and gm-C filters. Integrated
implementation of LC filters is not possible for this frequency range as large values of
inductor and capacitor is required, which comes with huge area penalty. The active RC
filter implementation requires high DC gain amplifiers with GBW in the GHz range,
which comes with the penalty of large power consumption. The gm-C filter is the best
The literature review of bandpass ADCs in Table 2.2 showed that high
indium phosphide (InP) and silicon germanium (SiGe) with large power consumption.
The trend in wireless receiver design is to integrate the analog and digital signal
134
designed in CMOS technology have been reported in literature and their performance
SNDR 42 dB 54 dB
Power 39 mW 30 mW
implement at high IF with the constraint of low power consumption. The linearity of the
OTA in the first stage is very important since its distortion components appear directly at
the output without any noise shaping. The distortion components of the subsequent OTA
stages are filtered by the loop and they have less stringent linearity requirements. The
approach that has been adopted in continuous time low pass ADC designs is to use
an active-RC filter only in the first stage for linearity enhancement and then use gm-C
filter in subsequent stages for low power consumption. This approach is not practical for
high IF CT BP ADC as the active-RC stage requires a high gain amplifier with gain-
bandwidth product in the GHz range, which results in huge power consumption. The
center frequency and the quality factor of the continuous time loop filter in the ADC
changes due to process, temperature and supply voltage (PVT) variations, which reduces
the SNR at the output of the ADC. An automatic tuning loop is required to fix the center
The main design specifications for the IF ADC are shown in Table 5.2. The
center frequency of the ADC is 125 MHz and it uses a Fs/4 architecture (Fs is the
sampling frequency), which results in a clock frequency of 500 MHz. The first objective
of this design is to develop a very linear transconductor for the gm-C bandpass filter
implementation with a center frequency of 125 MHz. The second objective of this
design is to propose a system level digital tuning loop to automatically tune the filters
center frequency, Q and other loop parameters in order to obtain the best SNR
performance for the ADC. The digital tuning approach offers several advantages over
136
other conventional approaches in the analog domain and can be easily integrated in
Parameter Value
The gm-C bandpass filter can be derived from a LC bandpass filter by replacing
the inductor with a gyrator, which emulates the action of inductive impedance for any
blocks (gm) and capacitor is shown in Fig. 5.2. The value of the effective inductor is
given by
C
Leff = (5.1)
g m2
137
Vx Vx
gm
L eff
gm
c
The implementation of a second order gm-C bandpass filter using the gyrator
block is shown in Fig. 5.3. Assuming ideal transconductance blocks (infinite output
resistance and DC gain), the transfer function of the second order bandpass filter based
g m1s
C
= 1 2
Vout (5.2)
Vin g
s 2 + m2
C1C2
The center frequency of the second order filter (o) is given by
g m2
o = (5.3)
C1C2
where C1 and C2 represent the effective capacitance at the output and intermediate nodes
(shown in Fig 5.3). The quality factor of the transfer function given by (5.2) is infinite,
which reduces when the finite output resistance of the transconductor blocks is taken
138
g m2
V in Vout
g m1
c
1
g m2
c
2
into account. Assuming that each transconductor block in Fig. 5.3 has the same finite
Vout g m1 ( sC2 + g o )
= 2 (5.4)
Vin s C1C2 + s(2C1 + C2 ) g o + (2 g o2 + g m2 2 )
2 go2 + g m2 2 (5.5)
o =
C1C2
The quality factor of the filter, which was infinite in (5.2), is given by
The finite output impedance of the OTA modifies both the center frequency and Q of the
filter. A higher order filter can be obtained by the cascade of the second order resonator
blocks.
139
This design was a joint research work by the author and his colleague. The author
was responsible for the design of the ADC architecture, digital tuning scheme and the
circuit implementation of the comparator. The details of the architecture for the IF ADC
signal to noise ratio (SNR) that can be obtained with limited signal swing at all internal
nodes (related to stability). The main architectural level choices for the ADC that affects
Chapter III. A fourth order, single-loop and single-bit architecture is chosen for the CT
swing and SNR. A higher order loop filter increases the SNR, but it could suffer from
potential stability problems as large signal swing may saturate the internal nodes. A
single-loop and single-bit architecture is chosen over the multi-stage and multi-bit
The first step in the design is to identify the transfer function of a switched
capacitor BP ADC which has the desirable loop transfer function with the required
SNR and good stability performance. The next step is to perform the impulse invariant
time) transfer function, which is described in Section 3.2. The transformation ensures
that the properties of both the structures are the same in terms of performance. A fourth
order switched capacitor bandpass filter can be obtained from a low pass prototype by
performing the substitution z-1 -z-2, which transform a nth order low pass filter to
bandpass filter of order 2n. The transfer function of the bandpass loop filter obtained
from the low pass filter is given by (4.2). A full clock cycle delay (z-1) is used after the
comparator, which gives more time for the comparator output to settle, thereby
alleviating any problems with metastability. This z-1 delay is taken into account while
the transfer function is calculated and the modified z-domain loop filter transfer function
that is used in the design is given by (4.3). The choice of the DAC pulse shape affects
the ADC performance, as discussed in Section 3.3. A Non Return to Zero (NRZ) DAC
pulse shape is used for the IF ADC which reduces problems associated with clock jitter.
The s-domain transfer function of the NRZ DAC (given in Fig. 3.4) and the z-domain
loop filter transfer function given by (4.3) are used in the impulse invariant transform of
(3.3) to find the open loop transfer function of the ADC. The open loop s-domain
transfer function of the ADC calculated using the previous step is given by (4.4).
shown in Fig. 5.4. The loop filter in the forward path is a fourth order gm-C bandpass
141
filter (cascade of two second order gm-C resonators, the basic second order resonator is
shown in Fig. 5.3). The comparator z-1 with delay samples the filter output and
quantizes the sampled value to a single-bit output. The digital bit stream (voltage mode)
g m2
Vin
g m2
g m1
c
g m1 Vout
c 1
z
g m2
c g m2
Comparator
c
k k k k
1 2 3 4
NRZ DAC
is converted into current by the four current DACs (k1 k4) and injected back into the
filter to complete the negative feedback loop. Assuming that the transconductance
blocks (gm1, gm2) are ideal, the open loop transfer function from the ADC output (Vout) to
s 3 Ak 3 + s 2 ( A2 g m1 k1 + Ao k 4 ) + s ( A 2o g m1 k 2 + Ao2 k 3 ) + A o3 k 4
H loop ( s ) = (5.7)
( s 2 + o2 ) 2
142
1 g
A= ; o = m 2 (5.8)
C C
The transconductance of the input OTA (V-I converter at the filter input) does not affect
the loop transfer function, but affects the dynamic range at the input of the ADC. This
architecture provides individual control for tuning each of the terms in the numerator of
(5.7). The s3 and constant terms in numerator of (5.7) are controlled by coefficients k3
and k4, respectively. Once these two coefficients are fixed, the s2 and s terms are
controlled by coefficients k1 and k2, respectively. The noise transfer function (NTF) of
Vout 1
NTF ( s ) = = (5.9)
Qn 1 + H loop ( s )
where Qn is the quantization noise (assumed to be dominant and injected at the
comparator input) and Hloop(s) is given by (5.7). The shape of the NTF can be controlled
precisely by tuning each of the numerator terms in the open loop transfer function, which
results in the minimum possible integrated noise power over the entire bandwidth and
good SNR. The transconductance value used in the filter (gm1, gm2) is chosen as an
optimum tradeoff between linearity, noise, signal swing and power consumption
considerations. The loop coefficients k1, k2, k3 and k4 can be calculated by equating each
Simulink and Matlab software simulation tools were used for the behavioral
simulation of the IF ADC. The Simulink behavioral model of the fourth order gm-C
bandpass IF ADC architecture is shown in Fig 5.5. The model uses sub-blocks available
Input
gm2
C.s+g1 gm2
Saturation3
C.s+g1
Transfer Fcn1
Saturation4
Transfer Fcn5 0
gm1
gm2 Constant
gm1
C.s+g1 gm2
Saturation1 1
C.s+g1 s
Transfer Fcn
Saturation2 Pulse Transport
Transfer Fcn4 Generator Comparator
xo Delay
Integrator
k3
k1
yout yout
k4
k2
143
144
from Simulink library to obtain the required loop transfer function. The s-domain
(continuous time) transfer function block is used to realize the integrator blocks (1/s).
The integrator blocks have finite DC gain, which is represented by the term g1 in the
denominator of the integrator transfer function. The current DAC blocks (k1, k2, k3 and
k4) are implemented using a simple multiplier block with the required coefficient to
convert the digital bit into current. The saturation block is used at the output of the
integrator blocks in order to limit the signal swing within the supply voltages (Vdd/Vss).
The saturation block is not required for the current DAC as its output is a digital current
pulse stream that follows the bit stream input. The implementation of the comparator
block is similar to the approach shown in Fig. 4.11 and described in section 4.4. The z-1
delay (one clock period) is implemented using the transport delay block and the single
bit quantizer is implemented using the relay block. The loop coefficients and the
transconductance of each stage are scaled down for low power consumption while
Coefficient Value
k1 0.3183
k2 0.3183
k3 0.8408
k4 0.75
145
maintaining the required signal swing and dynamic range. The final values of the
optimized loop coefficients that are used in the IF ADC are shown in Table 5.3. The
power spectrum of the single bit output of the IF ADC is shown in Fig. 5.6. The input
signal is a sine wave at 125 MHz and the sampling clock frequency is 500 MHz. The
measured SNR at the output of the ADC was 85 dB in a bandwidth of 1 MHz. The SNR
was calculated for 65536 samples and the simulation used finite DC gain of 40 dB for
each of the gm-C integrator blocks in Fig. 5.5. A step size of 0.05*Ts seconds (Ts is the
time period of the clock) is used for the model, which ensures very good accuracy of the
behavioral simulations. The simulation parameters that are used for the Simulink model
The SNR of the CT BP ADC is calculated by dividing the signal power over
fb
fo +
2
STF ( f )
fb
fo
2
SNR ADC = f
(5.10)
fo + b
2
NTF ( f )
fb
fo
2
where STF and NTF are the signal and noise transfer functions, fo is the center frequency
and fb is the signal bandwidth. The power of STF is usually a constant value over the
signal bandwidth and the SNR depends on the integrated noise power in the bandwidth.
The ideal NTF of the ADC can be derived from (5.9) and is given by
( s 2 + o2 )2
NTF ( s ) = 2 (5.11)
( s + o2 ) 2 + H num ( s )
where Hnum(s) is the numerator of (5.7) and o is given by (5.8). The magnitude of the
ideal NTF at s=jo is 0 (negative infinity in log scale), which leads to the best SNR
performance. The effect of finite Q of the filter on the ADC performance was discussed
in Section 3.4.4 and it increases the noise floor, which results in SNR degradation. The
147
Q n Quantization noise
Q n * NTF dB Q n * NTF dB Q n * NTF dB
fb
fb
fb
(a) Ideal NTF (b) Deviation in center frequency (c) Deviation in quality factor
Fig. 5.7. Effect of deviations in center frequency and Q of filter on the NTF
effect of deviation in center frequency and Q of the filter on the integrated noise power
is shown in Fig. 5.7, where the shaded area represents the noise power in the signal
bandwidth. Fig. 5.7 (a) shows the ideal NTF (very high Q and accurate center frequency
(fo)), the noise power at fo is very low and integrated noise power is at a minimum value.
Fig. 5.7 (b) shows the NTF for high Q and a deviation in fo, both the noise power at fo
and the integrated noise power in the signal bandwidth increases, which degrades the
SNR. Fig. 5.7 (c) shows the NTF for correct fo and a lower Q value, both the noise
power at fo and the integrated noise power in the signal bandwidth increases for this case
too, resulting in lower SNR. The shape of NTF given by (5.11) also depends on the loop
coefficients (k1, k2, k3 and k4); due to the presence of Hnum(s) in the denominator, and
these coefficients can change the integrated noise power. Thus, the center frequency and
148
Reference
Master filter
Comparator
DAC
Q of filter, together with the DAC coefficients have to be maintained at the desired value
The center frequency and Q of a gm-C filter, given by (5.5) and (5.6), depends on
the value of capacitor and OTA parameters (transconductance and output resistance).
The center frequency and Q of the gm-C filter typically changes by 25% over PVT
149
variations and a tuning scheme is usually required to compensate for these variations. A
comprehensive overview of typical automatic tuning schemes for obtaining the desired
center frequency and Q of continuous time filters are presented in [54] and [55]. The
most common approach to tune a CT BP ADC is shown in Fig. 5.8. The continuous
time bandpass filter in the forward path is disabled from the feedback loop. The center
frequency and Q of the standalone filter is set using typical tuning schemes (such as
master-slave filter tuning scheme, refer to [54], [55] for more details) and the tuned filter
is then enabled in the feedback loop. The main problem with this approach is that this
does not guarantee the best SNR performance for the ADC because it does not
compensate for any non-idealities in the feedback path (comparator and DAC blocks).
The center frequency and Q of the NTF (which determines the SNR) may change from
the required value, though the filter in the forward path has been tuned, due to other non-
idealities like excess loop delay in comparator, incorrect DAC coefficients etc. A direct
background digital tuning scheme was proposed in [56] and [57] to automatically tune
the center frequency and Q of a CT BP ADC. The scheme was based on the spectral
estimation of the digital output in background during the operation of the ADC. The
main problem with such an approach is that it becomes difficult to correctly estimate the
noise floor (power of NTF) and its frequency location when a strong signal component is
The block diagram of the proposed digital system level tuning scheme for the CT
BP ADC is shown in Fig. 5.9. The output digital bit stream from the ADC is
captured off chip using a data acquisition board. The captured bit stream is processed by
150
On-chip
components
Data
DAC External
acquisition
DSP
board
Control signals
for tuning
Digital
Continuous time
Vin bitstream
bandpass filter
Comparator
Feedback
coefficients
k1 k2 k3 k4
(switched current DAC)
an external digital signal processor (DSP), where the center frequency and Q of the CT
between the estimated and required values of center frequency and Q of the ADC is
calculated in order to generate the digital tuning control signal. The digital control word
is converted into an analog signal using an on-chip DAC, which is used to tune the filter
and the DAC currents for obtaining the best SNR performance of the ADC. This
automatic digital tuning scheme uses a system level approach because of the fact that the
filter, comparator and DAC non-idealities are taken into account when the output bit
stream is used for estimation of the power spectrum. The data acquisition and DSP
151
functions are performed off-chip for ease of implementation for this proof of concept
The system level implementation of the proposed digital tuning scheme is shown
in Fig. 5.10. The input test tone at the required center frequency fo is applied at the input
of the comparator. The measured output spectrum directly gives the NTF of the ADC.
The strength of the test tone can be very small and should be sufficient just to activate
the loop. The Discrete Fourier transform (DFT) of a bit stream (x[n]) of length N is
given by
N -1 j 2nk
X (k) = x[n]e N
, k = 0...N 1 (5.12)
n =0
The DFT can be implemented using the Fast Fourier transform (FFT) in a computer
when N is a power of two. The periodogram, P(k), of the input bit stream of length N is
1 2
P (0) = 2
X (0) (5.13)
N
N
1
[ 2 2
]
N
P( k ) = 2 X ( k ) + X ( N k ) , k = 1... 1
2
2
k 1 N
P = 2 X
2 N 2
where X(k) is given by (5.12). The power spectrum given by the periodogram is defined
at (N/2)+1 uniformly spaced frequency points between 0 and the Nyquist rate (Fs/2),
which implies that the width of the frequency bins are (Fs/N). The estimated power
152
f0
before calibration after calibration
Feedback
DAC
DAC
Tune gm and OTA gain Digital control
Fig. 5.10. System level implementation of the proposed digital tuning scheme
spectrum is used to determine the deviation from the required values and digital control
signals are generated to tune the ADC. The coarse tuning of filters center frequency is
transconductance is changed for the fine tuning of the center frequency. The OTA gain is
changed to vary the Q of the filter and the feedback DACs are tuned by modifying their
current values. The digital control signals are updated such that the tuning scheme
converges to a final value, which is ensured by using the adaptive LMS (Least mean
square) algorithm. The LMS algorithm tries to minimize the mean square error (MSE),
which is given by
153
1
MSE = (d (t ) y(t ))2 (5.14)
2
where d(t) is the desired response and y(t) is the actual response of the system in time
domain. The rate of change of the tuning signal (Wi) during the ith iteration determined
dWi
= [d (t ) y (t )]gi (t ) (5.15)
dt
where is the step size and gi(t) is the gradient of the tuning signal in time domain. The
algorithm for the digital tuning scheme is given by the following steps :
input.
ii) Compute the power spectrum of the captured output bit stream using (5.13),
which gives the power in (N/2)+1 frequency bins. Increase N for a finer
iii) Find the jth frequency bin which has the lowest power of all the frequency
bins. This gives the estimate for the location of center frequency of the
estimated center frequency. This helps in averaging out any errors during the
estimation process.
iv) The digital control tuning signal is calculated using (5.15) based on the
difference between estimated and required center frequency. The step size
is chosen such that the automatic tuning scheme converges within a required
154
time frame. Iterate (ii) (iv) till the center frequency is tuned to the desired
value
v) Calculate the power spectrum using (5.13) and find the lowest power value
(Plow) of all frequency bins. Find the two frequency bins to the either side of
the center frequency which has the power value (Plow + 3 dB), and their
vi) The digital control tuning signal is calculated using (5.15) based on the
chosen such that the automatic tuning scheme converges within a required
time frame. Iterate (v) (vi) till the Q is tuned to the desired value
vii) The DAC current coefficients are fine tuned to reduce the noise power at the
center frequency using the power estimation and correction steps, similar to
the above two cases. This fine tuning should not change the center frequency
and Q of the NTF that was already tune din steps (i) (vi).
The convergence time for the tuning scheme depends on the value of N (number of
output samples), step size () and the speed of the DSP. The main disadvantage is that
this is an offline tuning scheme and the ADC cannot be used while the tuning scheme is
activated. The offline nature of the tuning scheme necessitates the tuning scheme
The circuit implementation and simulation results of the main building blocks of
the ADC are discussed in this section. As mentioned earlier, the author was responsible
only for the circuit implementation of the comparator. The other building blocks are
described briefly and more details of the circuit implementation can be found in [59].
The fourth order bandpass filter in the forward path is implemented using a gm-C
filter, as shown in Fig. 5.4. The main requirement for implementing the bandpass filter is
a very linear OTA stage with high DC gain which operates properly at 125 MHz (center
frequency of the bandpass filter). The target specifications of the OTA in order to obtain
the required SNR of the ADC can be obtained from system level simulations. A brief
Parameter Value
DC gain > 30 dB
linearization techniques for extending their linear range of operation can be found in [60]
and [61]. A complementary differential pair which utilizes both NMOS and PMOS
pair is source degeneration, which extends the linear range at the cost of increased noise
technique to improve the linearity of OTA structures [63]. The proposed OTA structure
combines the techniques used in [62] and [63] together with source degeneration in order
The schematic of the linearized fully differential OTA used in the IF ADC design
is shown in Fig. 5.11. NMOS transistors M1 and PMOS transistors M2 together with
resistors R1, R2 and the triode transistor M7 form the basic complementary differential
pair with source degeneration. PMOS transistors M2 are cross coupled in the differential
pair in order to obtain non-linearity cancellation [63]. The effective small signal
g m1 g
g m,eff = m2 (5.16)
1 + Nn 1 + N p
where gm1 and gm2 are the small signal transconductances of transistors M1 and M2. Nn
R
N n = g m1 R1 + M 7 (5.17)
2
N p = g m 2 R2 (5.18)
where RM7 is the resistance of transistor M7 operating in triode region and is given by
(5.19).
VDD
Vbp
M6
M6
M8
M9 M9
R2 R2
M2 M2
Vin+ Vin-
Vo- Vo+
M1 Vcn M1
R1 R1
M7
Vbn M3 M3
VSS
1
RM7 = (5.19)
nCox (Vgs Vth )M 7
W
L M 7
where n is the mobility of the NMOS transistor, Cox is the gate oxide per unit area, W
and L are the width and length of M7, Vgs is the gate source voltage and Vth is the
threshold voltage of M7. The gate control voltage Vcn varies the resistor of M7, which in
turns changes the effective transconductance (gm,eff) of the OTA by about 5% without
feedback, which leads to negative resistance at the output. PMOS transistor M8 operates
in triode region and its resistance is similar to (5.18), where the NMOS parameters are
provides linear operation for the negative resistance. The output resistance of the OTA is
given by (5.20), the negative resistance term is given by (5.21) and the positive
1
Rout = (5.20)
1 1
R
pos Rneg
1 1 (5.21)
Rneg = 1+
g m9 W
2 pCox (Vgs Vth )M 8
L M 8
159
R pos = [ g m2 ro 2 ( R 2 // ro 4 )] // g m6 ro 6 ro 5 // g m1 ro1 ( R1 + W
1
) // (5.22)
2 nCox (
V gs Vth )M 7
L M 7
where gmi is the transconductance of the ith transistor, roi is the output resistance of the ith
transistor and p is the mobility of the PMOS transistor. The DC gain of the OTA is
given by (5.23).
Table 5.5. Component and current values for the OTA design
M7 (W/L) 86 m /0.4 m 0
R1 425 -
R2 140 -
160
where gm,eff is given by (5.16) and Rout is given by (5.20). The gate control voltage of
triode transistor M8 is used to vary the DC gain of the OTA, which changes the Q of the
bandpass filter. The component and current values for the linearized OTA design are
shown in Table 5.5. The simulated transconductance of the OTA is around 2 mA/V, DC
gain is greater than 30 dB, IM3 is less than -84 dB and the power consumption is around
10 mW for a supply of 1.65 V. The OTA shown in Fig. 5.11 is used for gm2 in Fig. 5.4,
while the OTA without the negative resistance (without the cross coupling transistors
M9) is used for gm1 in Fig 5.4 to construct the fourth order gm-C bandpass filter. The
controlled by switches while the fine tuning is obtained by tuning the effective
transconductance (gate control of Vcn). The Q of the filter is obtained by varying the
gate control voltage of transistor M8 (Vcp). The tuning range of the center frequency of
the bandpass filter from simulations was 82 MHz to 142 MHz, while the Q can be varied
from the nominal value of 50 by 15%. The details of the OTA design, along with
The schematic of the current DAC is shown in Fig. 5.12. It consists of a simple
NMOS differential pair (transistors M1) with PMOS current load (transistors M2) which
converts the digital bit stream into current pulses to be injected back into the filter. The
161
VDD
M5 M5 M2 M2
R R Vo- Vo+
Vo- Vo+ Io- Io+
M4 M4 Vin+ M1 M1 Vin-
I d1 I d1
Vbn M6 M3
VSS
Common mode feedback
common mode feedback circuit is used to set the required common mode voltage of the
differential pair. The output common mode voltage is sensed by resistors R, compared
Table 5.6. Component and current values for the current DAC design
with the required reference voltage (gate of other transistor M4, zero in this case) and
the control voltage (gate of diode transistor M5) is fed back to the gate of transistor M2.
The value of R is around 20 K and is chosen such that it doesnt affect the output
resistance of the OTA (which is around 10 K), since the DAC output is directly
connected to the output of the OTA. The component and current values for the current
DAC design is shown in Table 5.6. The DACi (i = 1 .. 4) block refers to the DAC of the
5.4.3. Comparator
The block diagram of the comparator implementation is shown in Fig. 5.13. The
preamplifier gain stage provides moderate gain for the signals before they are processed
by the latch. It also acts like a buffer between the filter and subsequent latches, which
prevent the kickback effect [13]. The kickback effect occurs when the transient
switching noise of the latch couples back to the input (and to the filter) through parasitic
capacitors. The output voltage of the latches undergoes a DC level shift before they are
injected into the DAC blocks. The schematic of the preamplifier gain stage is shown in
Input Preamp
DC level NRZ bit stream
gain Latch1 Latch2 Latch3 shift
stage
CLK CLK
VDD
Vbp M4
M2 M2
Vo- Vo+
Vi+ Vi-
R R
M1 M1
Vbn M3
VSS
M1 and PMOS transistors M2, along with current mirror transistors M3 and M4. The
resistors R (with one input tied to output and the other grounded) are used to set the
common mode voltage at the high impedance output node. The offset current due to
mismatch in the differential pair flows into the resistor and sets the common mode
where gmi is the transconductance of the ith transistor and roi is the output resistance of
the ith transistor. The value of R is 2.5 K and is in the same range as the output
164
resistance of the differential pair. The component and current values used in the
Table 5.7. Component and current values for the preamplifier gain stage
The schematic of the CMOS latch stage is shown in Fig. 5.15. It is implemented
using current mode logic for high speed operation at the clock frequency of 500 MHz.
The gate of the differential pair formed by transistors M7 controls the operation of the
sampling and quantization phase. When the clock is high, the differential pair formed by
transistors M5 is enabled as all the bias current is steered through it (other differential
pair is off) and the input signals are amplified. When the clock goes low, the cross
and the signals that were sampled at the falling edge of the clock are regenerated by the
positive feedback to either a logic zero or one (quantization phase). The available
regeneration time for the latch is 1 ns (one-half of the clock period) and the output settles
to either of the logic values within that time period. The regenerative time constant of
165
VDD
R1 R1
Vo -
Vo +
Vin+ M5 M5 Vin-
M6 M6
clk+ M7 M7 clk-
I
d1
V M8
b8
VSS
the latch determines time delay in the latch (the output settles within five time
2Ctot ,out
regeneration = (5.25)
g m6
Where Ctot,out is the sum of all parasitic capacitances at the output node and gm6 is the
transconductance of transistor M6. gm6 is proportional to the square root of the bias
current and the latch settles faster at the cost of higher power consumption. A cascade of
three latches driven by different clock phases as shown in Fig. 5.13 results in z-1/2 delay.
The propagation delay in comparator, DACs and routing parasitics adds to this delay
166
and the resulting worst case total delay (from simulations) is less than one clock period
(z-1 delay). This approach of incorporating circuit propagation delays (carefully adjusted
according to post layout simulations) in the required overall delay helps to overcome the
excess loop delay problem in this design. The component and current values used in the
comparator design is shown in Table 5.8. The power consumption of the latch was
optimized for good performance (no error in single-bit detection), which involved
Table 5.8. Component and current values for the latch stages
optimizing the bias current of the latch for a specific regeneration time. A higher drive
current is used in the third latch stage as it has to drive the subsequent DAC stages. A
lower value of load resistor (R1) is used for the third stage to make sure the signal swing
is within 400 mV for the required operation at 500 MHz clock frequency. The post-
layout transient simulation of the comparator is shown in Fig. 5.16. The differential
swing at the comparator output is around 350 mV and the nominal regeneration time
for the latch is around 180 ps for an input signal of 100 V (preamplifier adds gain to
this signal before the latch). The DC level shifting block is implemented using C-R
The top level circuit implementation of the IF ADC is obtained by connecting the
circuit implementation of all the building blocks according to the architecture shown in
Fig. 5.4. The microphotograph of the IF ADC chip (core ADC + digital blocks) is shown
in Fig. 5.17 and it occupies an area of 4.16 mm2. The chip was designed in TSMC 0.35
m CMOS technology through the MOSIS educational program. The OTA, DAC and
the comparator blocks form the core of the ADC design. The capacitor array is used
DACs
Comparator OTAs
Clock driver
Cap array
Demux
for coarse tuning of the center frequency of the bandpass filter and the digital blocks
include the clock driver and the demultiplexer (Demux). The clock driver is used to
drive an external clock signal to the comparator and the demultiplexer performs a serial
to parallel conversion (higher to lower clock rate) of the output bit stream for external
input signal at 125 MHz. The output bit stream file is processed in Matlab and the SNR
of the ADC is calculated by using the FFT command. The power spectrum of the output
bit stream of the ADC from post-layout simulations is shown in Fig. 5.18. The number
of samples used is 16384 and the output SNR was 50 dB in a 1 MHz bandwidth around
125 MHz. The number of samples is restricted by the limited computing resources, since
a top level extracted simulation can take about 5 days to generate 16384 samples. A
higher number of samples (65536 samples) are required to get a more accurate estimate
of signal and noise power values. The ADC designed in TSMC 0.35 m CMOS
technology consumes 152 mW of power from 1.65 V power supply. The chip is
currently under characterization and measurement results were not available at the time
5.6. Conclusions
The design of a fourth order CT BP ADC for IF digitization around 125 MHz
was presented in this chapter. The continuous time loop filter in the forward path was
implemented using gm-C filter and a NRZ DAC pulse shape was used to minimize the
effects of clock jitter. The design and simulation of the ADC architecture was presented.
A new system level automatic digital tuning scheme was proposed to overcome
problems associated with analog tuning techniques. The tuning scheme captures the
output bit stream using an external data acquisition board and it is processed in a DSP in
order to tune the center frequency and Q of the NTF of the ADC for good SNR
171
gives good linearity performance at 125 MHz was discussed, along with other building
CHAPTER VI
CONCLUSIONS
6.1. Summary
The strong growth in the wireless communications industry has led to the
development of a large number of wireless standards for various market segments (cell
phones, wireless networking, global positioning etc.). The main requirement of the next
generation wireless devices is the support of multiple standards on the same chipset with
negligible increase in power consumption and this issue is addressed in this work.
A brief introduction to the different wireless market segments and the different
standards was presented in Chapter I. The role of analog to digital converter blocks in
communication receivers was discussed in Chapter II. The different types of traditional
be ineffective for supporting multiple standards. The software radio architecture has
been proposed as an alternative to the traditional receiver architectures and it can support
variation of this architecture is the software defined radio architecture (also called digital
IF architecture) where the A/D conversion is performed on IF signals after a single down
conversion. The main types of A/D converter architectures were examined to determine
the optimal choice for a software radio application. A switched capacitor bandpass
frequency of 23 MHz. The clock frequency used was 92 MHz and the power
bandpass ADC implementation has the potential to achieve high resolution, but the
clock frequency cannot exceed 200 MHz even if more advanced silicon technologies are
used. A continuous time bandpass ADC architecture is found to a good choice for
digitization of IF and RF signals with higher clock frequencies, which makes it suitable
for both software radio and digital IF architectures. A literature survey of ADCs for RF
and IF digitization was presented in the last section of Chapter II. Though the reported
RF/IF ADC implementations achieve high resolution, the main bottleneck is high power
consumption (in the Watt range) and they are not suitable for use in portable wireless
devices.
ADC was presented in Chapter III. The design procedure uses an impulse invariant
equivalent discrete time (switched capacitor) architecture. The effect of various non-
idealities (clock jitter, excess loop delay, comparator metastability, finite quality factor
of the bandpass loop filter, unequal rise/fall time of DAC pulses) on the performance of
minimize their effects. The final section of Chapter III reviews some of the reported
architectures for direct RF digitization and identifies some of their major problems that
digitization at 950 MHz center frequency was presented in Chapter IV. A new
feedback path is proposed which enables the use of non return to zero (NRZ) DAC
pulses with full control of tuning all the coefficients in the noise transfer function (NTF).
The NRZ DAC pulse in the feedback path minimizes the effects of clock jitter and the
tuning of NTF coefficients ensures good SNR performance. The effect of using ideal
integrator DAC in the proposed architecture is examined in detail and the analysis
showed that the SNR performance degrades when long string of consecutive 1s or 0s
were present in the output bit stream. A modified version of the architecture using lossy
optimization techniques were used both at the architectural and at the circuit level to
minimize the overall power consumption of the ADC. The proposed ADC architecture
was implemented in IBM 0.25 m SiGe BiCMOS technology and measurement results
show SNR of 63 dB and 59 dB for bandwidths of 200 kHz and 1 MHz respectively,
around 950 MHz with a clock frequency of 3.8 GHz. The IM3 of the ADC was
measured to be -62 dB for -8 dBm two input tones at 949 MHz and 951 MHz. The ADC
automatic digital tuning scheme for IF digitization at 125 MHz center frequency was
presented in Chapter V. The architecture uses a cascade of two gm-C resonators with
four current DACs in the feedback path, which gives full control over tuning of all
175
coefficients in the NTF. A linearized CMOS OTA architecture combines both cross
coupling and source degeneration in order to obtain good IM3 performance at 125 MHz.
A system level automatic digital tuning scheme is proposed to tune the ADC
performance over PVT variations. A test signal is applied at the input of the comparator
rather than the ADC input, which helps to estimate the NTF directly by analyzing the
output bit stream. The bit stream is captured using an off-chip data acquisition board and
processed by an external DSP. The tuning algorithm runs on the DSP and it generates
the required control signals to tune the center frequency and Q of the filter. Additionally,
the tuning scheme also changes the feedback DAC coefficients to ensure the best
possible SNR at the ADC output. One of the main advantages of this tuning scheme is
that the estimation and correction process takes into account the non-idealities of all the
blocks (filter, comparator and DAC) in the loop, which results in the best performance
for the ADC. The ADC was designed in TSMC 0.35 m CMOS technology and it
consumes 152 mW of power from 1.65 V power supply. The simulation results show
SNR of 50 dB for a 1 MHz bandwidth around 125 MHz for a clock frequency of 500
The measurement results of the RF ADC designed in this research work shows
that high resolution narrowband (200 kHz 1 MHz) direct RF digitization is possible
with low power consumption. However, the ADC design falls short of meeting the
requirement of the ideal software radio architecture is a wideband ADC design with
variable bandwidth and low power consumption, which is not feasible in currently
available IC technologies. The next step in this research would be to explore techniques
for a wideband ADC design with the constraint of low power consumption. Another
interesting topic of research is to explore methods for obtaining both variable loop
bandwidth and center frequency in order to support multiple standards. The current trend
analog, RF and digital base band circuits in the same chip (system on chip approach).
The ADC design in this research work was done in SiGe BiCMOS technology and only
bipolar junction transistors (BJT) were used in the signal path. The RF ADC design can
good research topic. The main problem in CMOS RF ADC design is the requirement of
a very linear CMOS transconductor block with good performance at 1 GHz. In BJTs,
of a MOS device is proportional to the square root of drain current. This difference in
transconductance values makes the CMOS transconductor design with low power
consumption constraint a very challenging task at GHz frequency range. The output of
combination of both) before it can be given to the ADC. The implementation of these
interesting research topic. There are many significant technical hurdles at the hardware
177
radio solution.
The digital tuning scheme for the IF ADC design currently uses several tuning
interesting topic for future research. Based on the sensitivity, the tuning algorithm can be
adapted to simplify the tuning approach which decreases the overall time required for
tuning. The time required for tuning is an important parameter as this is an offline tuning
scheme (which requires disabling of the ADC during the tuning time) and any
improvement in the algorithm to reduce the time for tuning offers a significant advantage
for the proposed tuning scheme. Another possibility that could be explored in the IF
ADC design is the potential use of an active-RC implementation only in the first stage,
which has much better linearity as compared to a gm-C filter. This approach is usually
used in CT low pass ADCs where the inherent feedback around the operational
amplifier in the active RC stage provides a linear block for the first stage. The bandpass
filter works at higher frequency and would require demanding specifications for the
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VITA
Bharath Kumar Thandri received the B.E. (Hons.) degree in Computer Science
from Birla Institute of Technology and Science (BITS), Pilani, India in 1998 and his M.S.
degree in electrical engineering from Texas A&M University, College Station, Texas,
Bangalore from June 1998 to August 1999 in the DSP design group on the development
Cypress semiconductors, Austin, TX from Nov 2001 to Dec 2002 in the mixed-signal
CAD flow development and on the design of optical communication transceiver ICs. He
did two internships the first in Texas Instruments (India), Bangalore during Fall 1997
in the DSP group and the second in Texas Instruments, Dallas, Texas during Fall 2000 in
the Computer peripherals and control products group. He received his Ph.D. degree from
the AMSC group of the Electrical Engineering Department, Texas A&M University in
Jan 2006 as a Staff design engineer where he is involved in the design of RF transmitters
for wireless base stations. His main research interests are in the design of high
performance and high frequency analog circuits for communication and signal