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IEEE JOURNAL OF SOLID-STATE CIRCUITS, VOL. 42, NO. 5, MAY 2007

The Path to the Software-Defined Radio Receiver

Asad A. Abidi, Fellow, IEEE

Abstract—After being the subject of speculation for many years, a software-defined radio receiver concept has emerged that is suit- able for mobile handsets. A key step forward is the realization that in mobile handsets, it is enough to receive one channel with any bandwidth, situated in any band. Thus, the front-end can be tuned electronically. Taking a cue from a digital front-end, the receiver’s flexible analog baseband samples the channel of interest at zero IF, and is followed by clock-programmable downsampling with em- bedded filtering. This gives a tunable selectivity that exceeds that of an RF prefilter, and a conversion rate that is low enough for A/D conversion at only milliwatts. The front-end consists of a wide- band low noise amplifier and a mixer tunable by a wideband LO. A 90-nm CMOS prototype tunes 200 kHz to 20-MHz-wide channels located anywhere from 800 MHz to 6 GHz.

Index Terms—Analog decimator, analog FIR filter, anti-aliasing, bandpass signal processing, boxcar integrator, cognitive radio, decimation, digital AGC, direct conversion receiver, multi-mode receiver, multi-rate signal processing, receiver dynamic range, sampling, Software Defined Radio (SDR), switched capacitor circuit, wideband receiver, wideband sampling, wide tuning local oscillator, zero IF.

I. INTRODUCTION

T HE PRESENT way of adding new bands, modes and ser- vices to mobile wireless handsets does not scale. Each

new mode requires its own radio and baseband chipset, which is stuffed into handsets using ingenious packaging so that the physical volume of every generation either remains constant, or even shrinks. The rate at which new services are introduced will soon outstrip the rate of miniaturization in packaging. What is needed is a flexible, universal radio platform for receive and transmit, which can be programmed to steer to any band, tune to a channel of any bandwidth, and receive any modulation—all within reasonable constraints. It is time to consider a software- defined radio (SDR) seriously. Practical SDR fit for handheld operation has so far remained elusive. Indeed, weighed down by historical baggage, in the RF/analog circuit community the concept had become an object of derision. As first envisaged by Mitola [1], all RF and base- band receive signal processing is digital, enabled by an A/D con- verter (ADC) at the antenna (Fig. 1). It is soon apparent that the ADC in the SDR receiver (SDR RX) must fulfill extraordinary specifications. For example, to digitize the frequency band from 800 MHz to 5.5 GHz, where all of today’s cellular and WLAN channels lie, will require a 12 bit, 11 GS/s ADC. Not only is this ADC impossible today, it will remain so in the foreseeable future because ADC dynamic range and conversion are known to progress at a rate much slower than Moore’s law [2]. Even

Manuscript received November 12, 2006; revised January 29, 2007. The author is with the Electrical Engineering Department, University of Cal- ifornia, Los Angeles, CA 90095-1594 USA (e-mail: abidi@icsl.ucla.edu). Digital Object Identifier 10.1109/JSSC.2007.894307

Digital Object Identifier 10.1109/JSSC.2007.894307 Fig. 1. ultimately flexible device for wireless

Fig. 1.

ultimately flexible device for wireless communication.

Software-defined radio, as envisioned by Mitola [1]. This would be the

if we knew how to build an ADC with these specifications, its power dissipation might be hundreds of watts. To find a way out of this impasse, we must question Mitola’s paradigm. The flexibility and programmability that digital signal pro- cessing (DSP) brings to the radio front-end and baseband, Mitola believes, is essential for it to adapt to any modulation, channel bandwidth, or carrier frequency. What is not evident is that Mitola’s SDR receiver is capable of much more. With a sufficiently large number of digital downconverters operating in parallel, a single SDR RX can receive, concurrently, every channel that the ADC digitizes. To do so, all it needs is a powerful DSP and enough memory, both of which will improve at a rate that tracks Moore’s law. The military might be inter- ested in massively concurrent reception, but the typical mobile civilian is content to receive one channel at a time, sometimes two, while three may be the upper limit—for example, as an incoming cellular call is relayed to a headset by Bluetooth, the user browses the web on the handset or, in the background, downloads a large file via WLAN. Let us set our sights, then, on this modest goal: that it is suffi- cient for a software-defined receiver to receive any one channel, but at an arbitrary carrier frequency up to, say, 6 GHz, with any modulation. This receiver should be less power hungry than Mitola’s, indeed it may even be realizable in current IC tech- nology. One flexible receiver can even receive multiple inde- pendent channels, if they transmit in non-overlapped time slots. Otherwise each concurrent channel will need its own receiver, with possibly some shared circuits across the array. In the sense that multiple receivers will be built on a common platform, the SDR is a step forward compared to traditional circuits that are individually customized. The subject of this paper is how to construct this flexible receiver. We survey the steps that have been taken in recent the years towards this direction—some very hesitantly—culmi- nating in the recent realization of a software-defined RF receiver front-end. But we do not cover the digital front-end which fol- lows the ADC, because others do it better [3]; nor do we touch on the software-defined baseband—both, incidentally, essential

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ABIDI: THE PATH TO THE SOFTWARE-DEFINED RADIO RECEIVER

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ABIDI: THE PATH TO THE SOFTWARE-DEFINED RADIO RECEIVER 955 (a) (b) Fig. 2. (a) HF receiver

(a)

THE PATH TO THE SOFTWARE-DEFINED RADIO RECEIVER 955 (a) (b) Fig. 2. (a) HF receiver [4]

(b)

Fig. 2. (a) HF receiver [4] block diagram, styled after Mitola. With powerful enough DSP, it can receive all HF channels concurrently. (b) Details of the HF receiver, in particular of the digital downconverter which denes the digital front-end. The digital downconverter consists of the complex mixer and number-controlled oscillator (NCO). The cascaded integrator comb (CIC) lter downsamples the wanted channel at zero IF to its symbol rate, with ltering interleaved. The number of stages and taps in the digital front-end are programmable.

for a working receiver. Ultimately the receiver itself is only half of a radio communication device, which is not complete without an efcient software-dened transmitter. At this point the SDR TX is still in the research phase, and there is so little similarity between the two that the TX warrants a detailed description of its own in a future publication.

II. PROTO-SDR RECEIVERS AND METHODS

A. HF SDR Receiver, Mitola Style

The UK DERA has developed a prototype SDR receiver [4] for the HF band (330 MHz) used by the military in long-dis- tance communication. This is the only receiver that we know of Mitolas type (Fig. 2(a)). A discussion of its strengths and lim- itations will help to put later developments into perspective. The 30 MHz low-pass lter following the antenna suppresses signals that are out of the HF band. This lters usual role is to limit the total dynamic range incident on the receivers front- end, but here it is also an anti-aliasing lter for the waveform sampling that takes place immediately afterwards. A digitally controlled attenuator aligns the levels of the amplied input spectrum with the available dynamic range of a 12 bit, 75 MHz

A/D converter, which then digitizes the entire HF band. 1 A pow- erful DSP can receive any number of channels concurrently. It is instructive to follow the signal processing ow through the digital front-end, not only for readers from the analog circuit community, but because it has inspired the RF/analog front-end of our SDR receiver. DSP channelizers [5] operate on the complex analytic signal [6] representing information in the desired channel. The channelizer Fig. 2(b) consists of a complex (quadrature) number-controlled digital oscillator tuned to the channel of interest, driving a complex digital mixer to produce a stream of output samples. Problems such as offset or noise that plague zero IF signal processing in analog simply do not exist in the digital domain. The channelizers output samples are at the same high rate as the ADC. They represent the desired (narrow) channel at zero IF, but also unwanted channels that populate the rest of the HF band. Assume that the symbol rate of the wanted narrow channel is some small rational fraction of the ADC sample rate. Equal- ization and demodulation of the wanted channel most naturally

1 This ADC was state-of-the-art for the time, developed for IF conversion with high spur-free dynamic range (SFDR) in wireless basestations. Wireless base- stations always use SDRs (in a more restricted sense than Mitolas) to receive and transmit all channels in the band simultaneously. Radio power dissipation is, however, not of primary concern in basestations.

all channels in the band simultaneously. Radio power dissipation is, however, not of primary concern in
all channels in the band simultaneously. Radio power dissipation is, however, not of primary concern in

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IEEE JOURNAL OF SOLID-STATE CIRCUITS, VOL. 42, NO. 5, MAY 2007

JOURNAL OF SOLID-STATE CIRCUITS, VOL. 42, NO. 5, MAY 2007 Fig. 3. Proto-SDR receiver from Toshiba

Fig. 3. Proto-SDR receiver from Toshiba [8] which uses front-end downconversion to decouple sample rate from carrier frequency. Flexibility of the digital front-end enables receiver to tune individual PCS or DCS channels.

take place at this symbol rate. Therefore, the main task of the

digital front-end is to decimate the sample rate by this large

non-integer factor, and suppress the unwanted channels so that the signal of interest is captured with the shortest wordlength [3], [7] to lower the complexity of later computation. After decimation, portions of the wideband input spectrum

will alias into the now shrunken Nyquist band. The comb lter is

commonly used to protect a narrow frequency band surrounding

DC from aliasing [6]. This low-pass lters characteristic is

dened by a discrete approximation to a function up to the (wide) input Nyquist rate, with nulls at the (lower) output Nyquist rate and its multiples. The lter stopband around the nulls [7] must be wide enough to guard against corruption of the wanted signals spectral tails by aliased spectrum from else-

where. Unwanted spectrum in the lter sidelobes between nulls

will alias into the main lobe, but away from DC.

The input sample stream is downsampled in a sequence of stages, and then interpolated to realize conversion by a frac-

tional sample rate. Some appropriate amount of ltering is usu-

ally merged into each stage.

This HF software-dened receiver prototype performs almost as well as a conventional analog tuned triple-conversion super-

heterodyne. Its main advantage is greater programmability, and the relative ease with which any number of DSP channelizers

can be added to receive greater numbers of channels concur-

rently. Its main limitation stems from the sample rate and dy- namic range of practical ADCs. If the highest receive frequency is limited to half the sample rate, then todays state-of-the-art ADCs limit use of the SDR to no higher than the UHF band (30300 MHz). Yet most of the interesting wireless bands use

– 300 MHz). Yet most of the interesting wireless bands use carrier frequencies well above UHF.

carrier frequencies well above UHF. This limitation prompts us to look farther aeld.

B. Role of Tuning and Downconversion

To break the link between carrier frequency and ADC sample rate, the band of interest can be rst downconverted to within the ADCs Nyquist band. In choosing the LO frequency for down- conversion we sacrice some of Mitolas exibility, but this, it turns out, is a small price to pay for the gains. Although by convention we associate the Nyquist band with what is, more precisely, the (low-pass) rst Nyquist band, equally well it can be a higher Nyquist band that the ADC undersamples. In either case, the spectrum of interest should be limited to less than the Nyquist bandwidth; otherwise aliasing will cause part of the spectrum to fold over. The obligatory RF prelter at the antenna is an obvious way to bandlimit the receiver. Toshiba uses RF prelters for anti- aliasing to realize a mainly digital, exible receiver for PDC and DCS (Fig. 3) [8]. Switch selectable RF lters at 1.5 and 1.9 GHz pass 27-MHz-wide bands which are digitized. The preselected bands capture all PCS or DCS channels. Quadrature (complex) analog mixers with xed LO frequency downconvert the entire band to DC. A complex ADC (two 12 bit, 64 MHz ADCs, one in each of the two quadrature branches) digitizes the entire band, and a number-controlled complex digital oscillator translates the channel of interest to DC. Decimation and channelization follow in DSP. Although the fullband digitization will consume more power than in a single-channel ADC, the resulting exibility justies it.

ABIDI: THE PATH TO THE SOFTWARE-DEFINED RADIO RECEIVER

ABIDI: THE PATH TO THE SOFTWARE-DEFINED RADIO RECEIVER Fig. 4. A use of impulse sampling to

Fig. 4. A use of impulse sampling to effect two unrelated frequency translations at once. This enables the concurrent reception of GPS and GLONASS [10]. Corresponding points on the triangles identify portions of the input frequency spectrum that will coincide after aliasing.

To overcome the limited image rejection of the complex analog downconverter, the 27-MHz-wide band is translated to one side of DC so that the stopband of the RF lter attenuates images that will now lie on the other side of DC. With these measures, the image is actually suppressed by 70 dB. This receiver is one step in the right direction, but as we will see, there is farther to go: we must nd a way to overcome the constraints imposed by xed prelters. First let us take a small detour to explore subsampling and its possible use in concurrent reception.

C. Subsampling or Undersampling

The bandpass sampling theorem says that a conventional ADC can, in principle, digitize a signal with an information band situated compactly within any integer translation in frequency of the rst Nyquist band [6]. This process, called undersampling or sampling translation, exploits the translation of a compact spectrum situated within any one Nyquist band by convolution with an innite train of impulses separated on the frequency axis by to every Nyquist band; [9] gives another possible use of this effect. Owing to nite aperture time, that is, the non-zero transition time from on to off, practical samplers are inefcient in realizing large translations. Now consider two bandlimited spectra, each on carrier fre- quencies located within different multiples of a hypothetical Nyquist band that is wider than the sum of the two bandwidths. After undersampling at the Nyquist rate, both information bands will translate into the rst and higher Nyquist bands; what is more, with the right sample rate they will situate side-by-side with no spectral overlap. A low-pass ADC can now digitize both bands, and after DSP channelization the two signals can be re- ceived concurrently. This idea underlies the front-end of a positioning receiver that takes concurrent xes from GPS (1575 MHz) and GLONASS (1602 MHz) satellites (Fig. 4) [10]. Two prelters after the LNA select these bands but attenuate all out-of-band signals and wideband LNA noise that would otherwise, after sampling, translate and accumulate in the rst Nyquist band. A simple calculation shows that sampling at 24.2 MHz translates the two bands to the range DC to 12 MHz and locates them side-by-side.

to the range DC to 12 MHz and locates them side-by-side. 957 In this example the

957

In this example the sampler substitutes for the customary mixer, but with the property that a single sampling action pro- duces two different incommensurate frequency translations. For

this to work, the RF lter after the LNA must block all other

bands as well as wideband noise. The xed preselect lters are necessary, but once again they limit the receivers exibility.

D. Analog Decimation

In most cases the passband of available RF lters is far wider than the wanted channels bandwidth (Fig. 5(a)). If this lter

also anti-aliases, the wanted channel will be sampled at a much

higher rate than minimum. A/D conversion at a surplus rate wastes precious power, especially when the later channelizer

suppresses much of the digitized spectrum. Power dissipated by

a receiver ADC usually falls off sharply when sample rate is

lowered, because the dynamic range also goes down. This sug-

gests that the samples should rst be decimated to the lowest reasonable rate that captures what is wanted, and then digitized. Much like DSP-based decimation, analog downsampling nar- rows in on the desired band while protecting it from aliasing. Decimation can also be merged with frequency translation by choosing an initial (high) sample rate, some multiple of which

is centered on the band of interest. This translates the band to

DC or to an IF, and downsampling can follow in a discrete-

time (D-T) analog decimation lter. A 2.4 GHz CMOS receiver based on this principle has been published [11]. A cascade of decimation stages also lowers the IF by the decimation factor, a property that has been used in another early receiver [12]; obvi- ously this property is of no use if the signal of interest is already at zero IF. How does an analog decimation lter work? The simplest realization is to load successive samples of the discrete-time

is to load successive samples of the discrete-time waveform into capacitors, and then sum their charges

waveform into capacitors, and then sum their charges (Fig. 5(b)) [13]. This forms an -tap FIR lter with equal tap weights whose transfer function resembles a with nulls at all integer multiples of the decimated-by- (lower) sample rate (Fig. 5(c)). The nulls protect wanted spectrum at and near DC from suffering aliasing distortion, as well as wideband noise from accumulating in the input Nyquist band.

wideband noise from accumulating in the input Nyquist band. E. Sampling With Built-In Anti-Aliasing So far
wideband noise from accumulating in the input Nyquist band. E. Sampling With Built-In Anti-Aliasing So far

E. Sampling With Built-In Anti-Aliasing

So far all the sampling schemes we have described need an anti-aliasing prelter somewhere; it may be the RF prelter, or

it may be another lter downstream. This xed lter restricts

the receiver from tuning to arbitrary carrier frequencies, and

from receiving channels of widely different bandwidths. Con- tinuous-time analog lters are tunable by limited amounts, but the bandwidth of todays wireless channels can span 100:1. For exibility we must nd a way to eliminate a standalone anti-alias lter. Classic impulse sampling, which has become synonymous with sampling, is inherently wideband. This is why it can be used for frequency translation. However, impulse sampling captures, with equal delity, the narrow wanted spectrum and unlimited spectrum that is unwanted. This is why it needs an anti-aliasing prelter. Is there a way, then, to sample wireless channels with built-in

anti-aliasing that is good enough to protect, say, one narrow-

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IEEE JOURNAL OF SOLID-STATE CIRCUITS, VOL. 42, NO. 5, MAY 2007

JOURNAL OF SOLID-STATE CIRCUITS, VOL. 42, NO. 5, MAY 2007 Fig. 5. (a) The available anti-alias

Fig. 5. (a) The available anti-alias lter (here an RF lter) may force a narrowband channel to be sampled at an unrealistically high intial rate. To save power in later hardware, the analog samples should be decimated. (b) A switched capacitor decimator, or downsampler, with built-in comb lter [13]. (c) Illustrating aliasing associated with downsampling, and how the comb decimation lter protects a narrowband around DC from aliasing.

fi lter protects a narrowband around DC from aliasing. Fig. 6. One way to sample a

Fig. 6. One way to sample a narrowband channel with built-in anti-aliasing [14] after it is downconverted to zero IF. The waveshaping built into mixing, followed by integration, realizes a programmable lter. WFG is a weight func- tion generator; MX is a mixer.

band channel? In some early work on software-dened receivers [14], [15], the Poberezhskiys ask this very question. They pro- pose to construct a programmable lter using the mixer as part of a sampler and weighted integrator (Fig. 6). The LO waveform is a discrete approximation to the desired convolution kernel. The output is a stream of anti-aliased, ltered samples. How- ever, the many functions merged into a mixer soon become un- wieldy for a practical circuit realization. Yuan [16] gives a simpler answer to the same question: a sam- pler using what is known to physicists as the boxcar integrator. 2 This is a special case of the Poberezhskiy proposal. It entails, rst, a translation of the input spectrum, assumed to be of un- limited extent, so that the wanted channel is at zero IF. This

2 Physicists detect single events with the boxcar integrator, whereas we look to it to sample bandpass continuous-time waveforms. The boxcar integrating sampler has also been published in the analog circuits literature, but not as a solution to our problem [30].

literature, but not as a solution to our problem [30]. Fig. 7. (a) A simpler method

Fig. 7. (a) A simpler method to sample a narrowband channel at zero IF, based on an integrate-and-dump or windowed integration [16]. Filtering can follow later, often embedded with decimation. (b) The transfer function of this sampler showing protection from aliasing at DC.

is followed by integrating the composite waveform over a time

window of width

(Fig. 7). The result of a windowed integra-

tion comprises an output sample. After readout the integrator is

reset, or dumped, and

later it is ready with the next sample.

This process can continue without interruption by, for instance,

or dumped, and later it is ready with the next sample. This process can continue without
or dumped, and later it is ready with the next sample. This process can continue without

ABIDI: THE PATH TO THE SOFTWARE-DEFINED RADIO RECEIVER

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ABIDI: THE PATH TO THE SOFTWARE-DEFINED RADIO RECEIVER 959 Fig. 8. IF receiver based on windowed

Fig. 8. IF receiver based on windowed integration sampling [18]. To simplify clock generation, the integration window is formed by dividing down some number of cycles of the mixer LO. We point out that this is non-optimal. The IF carrier frequency sets the mixer LO, but protection of the wanted channel at zero IF from aliasing by unwanted channels should set the integration window.

time-interleaving two identical circuits that alternate between integration and readout/reset. This windowed integration sampler takes continuous-time input and produces D-T analog samples at rate . It is a linear, periodically switched circuit. Its transfer function on the continuous input frequency scale is

Therefore, the receiver must rely on a xed RF preselect lter, which immediately limits tunability. The function alone

is not selective enough to displace the preselect lter; we must

continue the search for a powerful but tunable lter to replace it.

F. Sample Rate, Downsampling, and Filtering

When the wanted channels bandwidth is much less than the sample rate, the lter can protect it from aliasing. It strongly attenuates unwanted signals occupying an equal bandwidth at each null. Unwanted channels between nulls lie in the lter sidelobes, but after sampling these will alias into the rst half of the main lobe away from DC. Aliasing from all sidelobes accumulates in the main lobe [Fig. 7(b)]. As long as this accumulationafter any analog ltering that followsdoes not overload the ADC, it will be eliminated by

a

digital lter in DSP. We dene the stopband loss at the th lter null as

We de fi ne the stopband loss at the th fi lter null as (1) This
We de fi ne the stopband loss at the th fi lter null as (1) This
We de fi ne the stopband loss at the th fi lter null as (1) This
We de fi ne the stopband loss at the th fi lter null as (1) This

(1)

This denes a low-pass lter with a main lobe, and a non- monotonic transfer function consisting of a series of sidelobes that decay at 20 dB/decade, with zeros at The op- eration filters the continuous-time input before sampling, with full anti-aliasing at DC. This anti-aliasing extends to wideband input noise; that is, noise at the output of the preceding wide- band stages is also nulled at aliasing frequencies, and does not accumulate at DC [17]. Windowed integration, we can say, is well suited to sampling a narrow channel at DC immersed in an innite band of unwanted signals. Karvonen et al. [18] report an IF receiver that uses this sampler (Fig. 8). The input IF sets the frequency of mixer switching, and a simple division determines the output sample rate , where is programmable. This circuit merges mixing to zero IF with windowed integrator sampling. It consists of a transconductance amplier with IF input, tri-level switches that commutate the transconductor output current, and an opamp-based current integrator whose output is read and reset after every cycles of commutation. Mixing and inte- gration together can be interpreted as RF (or IF) sampling, particularly when the mixer is built with passive FET switches that can also be thought of as sampling switches. However, we caution against going too far down this line of thinking, because mixing is determined by the carrier frequency, while, as we will show, sampling rate is set by the wanted information bandwidth and strength of unwanted channels. Indeed, it is for the last reason that this receiver is unable to exploit the possibilities latent in integrate-and-dump sampling. In most cases the 20 dB/decade rolloff of the ltersside- lobes is not good enough to suppress strong adjacent channels.

is not good enough to suppress strong adjacent channels. the minimum attenuation relative to DC across
is not good enough to suppress strong adjacent channels. the minimum attenuation relative to DC across
is not good enough to suppress strong adjacent channels. the minimum attenuation relative to DC across
is not good enough to suppress strong adjacent channels. the minimum attenuation relative to DC across
is not good enough to suppress strong adjacent channels. the minimum attenuation relative to DC across
is not good enough to suppress strong adjacent channels. the minimum attenuation relative to DC across
is not good enough to suppress strong adjacent channels. the minimum attenuation relative to DC across
is not good enough to suppress strong adjacent channels. the minimum attenuation relative to DC across

the minimum attenuation relative to DC across the bandwidth

the minimum attenuation relative to DC across the bandwidth surrounding . For the function, this is

surrounding

. For the
. For the

function, this is

(2)

We use this relation to nd the baseline initial sample rate

if no other lter were present. Suppose the desired channel oc-

fi lter were present. Suppose the desired channel oc- cupies the bandwidth . First we must
fi lter were present. Suppose the desired channel oc- cupies the bandwidth . First we must

cupies the bandwidth . First we must construct a specication on stopband loss. We calculate the relative energy of adjacent channels in the same band and in nearby bands picked up in a window of width that moves from DC to arbitrarily large fre- quencies. Then, assuming that this window can alias to DC as a co-channel interferer, we calculate by how much it must be at- tenuated at every frequency if the wanted channel is to reach the target SNR. From (2), the lowest sample rate is found to satisfy the specication on stopband loss at the rst few nulls; usually this sample rate will give more than enough attenuation at the higher order nulls. Depending on the unwanted signal prole, this initial sample rate may be very high, reaching into the GHz. A/D conversion at this rate will consume disproportionate power, or might even be impossible. To alleviate this situation, we might add more ltering, or downsample the output. Both, as we will show, are necessary; both are employed.

we might add more fi ltering, or downsample the output. Both, as we will show, are

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IEEE JOURNAL OF SOLID-STATE CIRCUITS, VOL. 42, NO. 5, MAY 2007

JOURNAL OF SOLID-STATE CIRCUITS, VOL. 42, NO. 5, MAY 2007 Fig. 9. Bluetooth receiver using windowed

Fig. 9.

Bluetooth receiver using windowed integration at carrier frequency, followed by decimation stages [19]. This is not yet a software-dened receiver.

We learn from multi-rate digital signal processing that down- sampling by a certain composite integer is most efciently re- alized with a cascade of stages, each stage downsampling by reasonable factors of that integer and interleaved with anti-alias ltering [7]. In our use, a not widely appreciated outcome is that the embedded ltering cumulatively across a few stages sup- presses adjacent channels by huge amounts. This will turn out to be the most important ltering action in suppressing all un- wanted channels. Just as in DSP implementations, a compact analog comb lter protects a narrow band at DC from aliasing that results from decimation. A passive switched capacitor circuit realizes decimation and downsampling simultaneously. Samples of the incoming analog discrete-time waveform are distributed to an array of either equal or weighted capacitorsthe relative weights will determine the taps of an equivalent FIR lterand after input samples are loaded, all capacitors are shorted to share their charge with an st discharged capacitor. This corresponds to downsampling by , with embedded FIR anti-aliasing. If we can tolerate the signal loss associated with each stage of passive charge redistribution, more than one stage of downsampling and ltering are cascaded very simply as pure charge-transfer circuits with no inter-stage buffers. The st capacitor referred to above could itself be one of equal, or weighted, capacitors in an array, which down- sample by with built-in anti-aliasing. The cascaded ltering progressively deepens the stopband, while all the passbands coincide at DC. The lter passband narrows along the cascade. This, then, is the ltering that will enable the sampling function to nally displace the RF preselect lter.

function to fi nally displace the RF preselect fi lter. G. RF Sampling: The Texas Instruments
function to fi nally displace the RF preselect fi lter. G. RF Sampling: The Texas Instruments
function to fi nally displace the RF preselect fi lter. G. RF Sampling: The Texas Instruments
function to fi nally displace the RF preselect fi lter. G. RF Sampling: The Texas Instruments

G. RF Sampling: The Texas Instruments Receivers

The Bluetooth [19] and GSM receivers [20] from Texas Instruments use integrate-and-dump sampling, followed by analog downsampling and ltering [21]. They are excellent examples of all that has been described so far. A large amount of programmable ltering and decimation gives them more exibility than traditional receivers. Furthermore, the very simple realizations of the RF/analog circuits enable the radio to be integrated on the same chip as the DSP in deep-submicron CMOS. However, both use preselect lters and tuned front-ends for the respective bands, and therefore are not, in our sense, software-dened receivers.

are not, in our sense, software-de fi ned receivers. Fig. 10. Range of variable gain in

Fig. 10. Range of variable gain in the receiver required to capture a GSM input, from the point of view of a delta-sigma ADC with 14 bit resolution. This A/D- centric approach seeks to lighten the burden on analog circuits, and push as much as possible to DSP.

In the Bluetooth receiver (Fig. 9), the stage following the LNA merges downconversion, integration and decimation l- tering in the paradigm of RF sampling. The initial sample rate is equal to the wanted channels carrier frequency in the 2.4 GHz band. This positions the rst aliasing channel at 4.8 GHz, well in the RF prelters stopband. Decimation by 8 through a lter similar to [11] and [13] follows, and a ca- pacitor continuously connected to the mixer transconductance realizes a D-T pole. Then follow a rst-order D-T lter that decimates by 4 and a third-order D-T lter that dec- imates by 2 . Finally the output discrete-time waveform is dig- itized. At the nal output rate of 37.5 MHz, some other channel in the receivers 80-MHz-wide passband could alias on to the wanted channelbut by this point the cumulative ltering will have suppressed it to a very low level. We note also that as the rst LO tunes to different channels, the conversion to symbol rate will need different non-integer interpolations, but this is readily managed in todays DSP.

but this is readily managed in today ’ s DSP. III. UCLA SDR R ECEIVE A
but this is readily managed in today ’ s DSP. III. UCLA SDR R ECEIVE A
but this is readily managed in today ’ s DSP. III. UCLA SDR R ECEIVE A
but this is readily managed in today ’ s DSP. III. UCLA SDR R ECEIVE A
but this is readily managed in today ’ s DSP. III. UCLA SDR R ECEIVE A
but this is readily managed in today ’ s DSP. III. UCLA SDR R ECEIVE A
but this is readily managed in today ’ s DSP. III. UCLA SDR R ECEIVE A

III. UCLA SDR RECEIVE ARCHITECTURE

A. Baseband Circuits

We call a receiver SDR if it: (a) tunes to any band (from 800 MHz to 6 GHz is good enough today); (b) tolerates blockers in its tuning range without a prelter at the antenna; (c) uses a low-power ADC to digitize the wanted channel and remaining adjacent channels; and (d) exploits the conversion dynamic range of the ADC to relax ltering and programmable gain in the RF and analog front-end. A receiver based on direct conversion to zero IF with a wide tuning LO simplies later signal processing. We describe, step

ABIDI: THE PATH TO THE SOFTWARE-DEFINED RADIO RECEIVER

961

ABIDI: THE PATH TO THE SOFTWARE-DEFINED RADIO RECEIVER 961 Fig. 11. kHz to tens of MHz.

Fig. 11.

kHz to tens of MHz. The analog front-end following the mixer bears strong resemblance to the digital front-end following the ADC [see Fig. 2(b)].

Software-dened radio receiver, tunable from 800 MHz to 6 GHz and congurable, mainly by choice of sampling clock, to receive channels as wide as

by step, the development of this receivers architecture and circuit blocks [23]. Contrary to the usual receiver design ow which proceeds downstream from the antenna through the RF sections to baseband, in this case we start with the ADC, then proceed upstream. Let us rst examine feature (d) in the context of GSM re- ception (Fig. 10). The 200-kHz-wide wanted channel can be re- ceived at any strength ranging from 102 to 15 dBm, a dy- namic range of 87 dB. What, then, is the minimum range of analog variable gain before the ADC that will capture this dy- namic range, while delegating as much as possible of the re- maining automatic gain control (AGC) function to the DSP? It is relatively easy in todays technology to realize a delta- sigma ADC that dissipates about 10 mW, and yields 14 bit reso- lution (86 dB SNDR) across 100 kHz. A GSM channel must be detected with a minimum SNR of 9 dB for acceptable bit error rate. Suppose the ADC quantization noise limits this SNR. If the ADC full-scale is 1 dBm, it follows that the quantization oor lies at 82 dBm across a 100 kHz bandwidth. The receiver need amplify the minimum GSM signal at the antenna by only 43 dB, that is, to 22 dB above the quantization noise oor, for the cu- mulative SNR to reach 9.2 dB: this is good enough. Now consider the largest allowable GSM signal at the re- ceiver input: this will overload the ADC unless the gain is low- ered. What is the minimum by which the gain must be low- ered? Allowing for 4 dB peak-to-average variation if the signal is EDGE modulated, the gain must be lowered to 12 dB for the waveforms peaks to just reach the ADCs full scale. This means that with only 31 dB of variable gain in the RF/analog front-end, the receiver can make use of surplus dynamic range in the ADC to capture an input signal whose RMS strength changes by 87 dB (Fig. 10) [22]. Of course, this underestimates the real-life situation where strong adjacent channels are also incident on the ADC, but that is the lters job. Therefore, let us turn to this lter. With no RF prelter, the desired channel at zero IF is surrounded by other, possibly very strong, channels in its own band as well as in all other bands. Like the variable gain function just described, the on-chip lter should, within reason, burden analog circuits only lightly, while pushing as much as possible to DSP. Its job is mainly to pre- vent the composite waveform of unwanted channels from over- loading a low power ADC. By lowering the total dynamic range, ltering relaxes the linearity and power consumption of the cir-

relaxes the linearity and power consumption of the cir- cuits that follow. The receiver bene fi

cuits that follow. The receiver benets most if the lter appears immediately after the mixer. To recapitulate, a windowed integration sampler is also

a lter that attenuates every aliasing channel that is a po- tential co-channel interferer for the wanted channel at DC. Between nulls, the sidelobes in its frequency response roll off

at 20 dB/decade and attenuate, but do not suppress, adjacent

channels. Analog circuits following the sampler must operate

in discrete time. This is good, because switched circuits can be

tuned by a clock over orders of magnitude in frequency. The initial sample rate that protects against aliasing will al- most always be too high: it should be lowered by additional ltering before sampling, and by decimation afterwards. This begs the question, though, that to suppress aliases if the rst stage must sample at a high rate, then how can the second stage, which operates at a lower rate, adequately suppress the new set

of co-channel interferers that pass through the rst stages side-

lobes? The answer is that it cannot, unless the stopband of the second decimation lter is deeper. This may need a decimation lter of

fi lter is deeper. This may need a decimation fi lter of higher-order, with a frequency
fi lter is deeper. This may need a decimation fi lter of higher-order, with a frequency

higher-order, with a frequency response where . Then the stopband loss is

with a frequency response where . Then the stopband loss is (3) We have found one

(3)

We have found one lter cascadethere may be othersthat works well for the two extreme cases of GSM and 802.11g

WLAN channels. We believe that by changing clock frequency,

it can be congured as well to work for most other cases when

the channel bandwidth is somewhere in between these two ex- tremes. Although we arrive at this cascade by trial-and-error, in

future it should be possible to synthesize the optimal lter cas- cade by computer. In our SDR receiver (Fig. 11), the lter is a cascade of two poles at the mixer load, windowed integra- tion with an embedded D-T pole, followed by a second-order

, and fi nally a fi rst-order sinc nally a rst-order sinc

lter that decimates by 2 or 3 Next we describe how to congure this lter to receive 802. 11g, which will also help the reader to understand how it works.

The channel of interest occupies 20 MHz in the 2.4 GHz ISM

band. To keep things simple let us assume that other channels

in the same ISM band are comparable in power and will not

overload the ADC. The CDMA band 200 MHz away is the

are comparable in power and will not overload the ADC. The CDMA band 200 MHz away
are comparable in power and will not overload the ADC. The CDMA band 200 MHz away

lter that decimates by 4

are comparable in power and will not overload the ADC. The CDMA band 200 MHz away
.
.

962

closest region of concern, because it contains channels at much higher powers used for cellular communication. If we assume as

a worst case that every CDMA channel transmits at peak power,

then the lter must attenuate by 50 dB across the entire band merely to prevent ADC overload [Fig. 12(a)]. If the initial sampling frequency also falls in this band, then the attenuation must rise by another 30 dB in the 20 MHz-wide interval around this frequency so that aliasing CDMA channels located there do not overwhelm the wanted 802.11g channel. Similar considerations lead to notches of various depths in the lter specication at multiples of the sampling frequency. It is hard to realize a continuous-time lter whose stopband is deep enough to encompass these notches. An FIR lter, with periodic nulls in its transfer function, is better suited to the job. The windowed integration sampler is exactly such an FIR lter. Suppose we choose an initial sample rate of 480 MHz. This means that notches must be added to the lter specication at 480 MHz and its multiples, each notch 20 MHz wide; and the rst notch, because it lies in the CDMA band, must be 80 dB

deep [Fig. 12(a)]. Now let us develop the lter. To start with, we realize that at 480 MHz, the samplers own lter response is not se- lective enough to fulll the lter spec anywhere. However two real poles preceding the sampler at frequencies just beyond the wanted channels band edge, say at 15 and 20 MHz, add an- other 50 dB to the cascade loss at 480 MHz. This is now enough to meet the spec at the rst null [Fig. 12(b)]. The poles can be realized with switch-selectable capacitors shunting the mixers resistor loads. It is important that a linear passive RC circuit is used, because it must lter amplied unwanted signals without distortion. Once suppressed, they are relatively benign. With these added poles, the lter specication is met every- where except around 240 MHz, which is, coincidentally, half the sample rate. There is a simple way using only one extra capac- itor to add loss at this frequency. It makes use of the D-T pole arising from a capacitor at the samplers transconductor output, in parallel with the periodically reset integration capac- itor which forms a discrete-time resistor [Fig. 12(c)] [19]. At sample rate the pole frequency is

[Fig. 12(c)] [19]. At sample rate the pole frequency is (4) As this is a discrete-time
[Fig. 12(c)] [19]. At sample rate the pole frequency is (4) As this is a discrete-time
[Fig. 12(c)] [19]. At sample rate the pole frequency is (4) As this is a discrete-time
[Fig. 12(c)] [19]. At sample rate the pole frequency is (4) As this is a discrete-time
[Fig. 12(c)] [19]. At sample rate the pole frequency is (4) As this is a discrete-time

(4)

As this is a discrete-time pole, its transfer function on the axis of continuous input frequency is the superposition of images of

a single-pole rolloff around the sample frequency and its multi-

ples. This leads to a periodically repeating frequency response, with minima at half the sampling frequency, exactly where help is needed, and its odd multiples. A D-T pole located at 25 MHz

gives 20 dB of additional loss at 240 MHz. Now the cascade lter response meets the specications everywhere. But A/D conversion at 480 MHz wastes power when the spec- trum of interest is only 10 MHz wide. The only remedy is downsampling. Let us say that our target sample rate is 40 MHz, a comfort- able clock rate for a Nyquist ADC with modest resolution that might dissipate 10 mW. We factor downsampling into two steps:

dissipate 10 mW. We factor downsampling into two steps: IEEE JOURNAL OF SOLID-STATE CIRCUITS, VOL. 42,

IEEE JOURNAL OF SOLID-STATE CIRCUITS, VOL. 42, NO. 5, MAY 2007

, followed by 3 . The stopband specication evolves at each

4

step, in that the number of notches goes up by the decimation

factor. For every decimation the specication determines the

order of that lter. In the case of the rst decimation by 4 we nd by trial-and-error that a lter is needed [Fig. 12(d)];

but

[Fig. 12(e)]. The latter lter is simpler because it benets from

cumulative attenuation in the preceding cascade. After two stages of decimation, notches appear in the lter specication every 40 MHz apart. The cascade of decimation lters produces the very sharp transition required by the rst notch [Fig. 12(e)], and it suppresses all unwanted channels so well that a 40 MHz Nyquist ADC can capture the lter output with a dynamic range of only 8 bits. The cascade lter looks complex, but it is easy to implement

as a charge transfer circuit (Fig. 13). A transconductor after the mixer integrates the input waveform on the sampling capacitor that is selected sequentially from one of 8 parallel channels. This capacitor, in turn, comprises four equal unit capacitors in parallel. After sampling, charge from any number from one to four of these unit capacitors is transferred to the next stage to realize the tap weights of an FIR lter. How to synthesize the lter? This is the equivalent to

the

fore, must be the convolution of the lters unit rectangle impulse response with itself; this is an isosceles triangle span- ning two sample periods. After decimation by 4 each output sample is formed by summing seven successive input samples weighted by 1-2-3-4-3-2-1; and for uninterrupted throughput, the next summation starts when the rst sum is only half formed. With 4 unit capacitors and 8 channels, time interleaving is seamless. As 3, 2, and 1 unit capacitors in three channels share their charges to convolve input samples with the falling edge of the triangle, the other 1, 2, and 3 unit capacitors in the same channels convolve the same input samples with the rising edge

of the next triangle. Thus, every unit capacitor is fully employed. Summation entails connecting all weighted capacitors across

discharged capacitor in the next stage. Four or three channels of equal unit capacitors in the second stage realize ltering

or 2

and

a

cascade of two lters, whose impulse response, there-

for the next decimation by 3 , a lter is good enough

for the next decimation by 3 , a fi lter is good enough decimation by either
for the next decimation by 3 , a fi lter is good enough decimation by either
for the next decimation by 3 , a fi lter is good enough decimation by either
for the next decimation by 3 , a fi lter is good enough decimation by either
for the next decimation by 3 , a fi lter is good enough decimation by either
for the next decimation by 3 , a fi lter is good enough decimation by either
for the next decimation by 3 , a fi lter is good enough decimation by either
for the next decimation by 3 , a fi lter is good enough decimation by either

decimation by either 3

by 3 , a fi lter is good enough decimation by either 3 . For the

. For the capacitances actually used in the circuit, the cascade

the capacitances actually used in the circuit, the cascade DC gain presented to the wanted channel

DC gain presented to the wanted channel is [31]

the cascade DC gain presented to the wanted channel is [31] (5) Therefore, the passive portions

(5)

Therefore, the passive portions of this baseband circuit atten- uate the integrating samplers own gain (1) by only 16/17, that is, by slightly less than 1. Furthermore, we can show that if the overall gain given by (5) is, say, 3 or more, the input-referred voltage noise density at zero IF is 4 , completely over- whelming any noise involved in the charge transfers. is the noise factor of the transconductor circuit. At a given sample rate, this gain can be varied with or with the sampling capacitance . The transconductor is built up of a set of four unit cells selectable with a 2 bit word, and

a unit sampling capacitor consists of a xed capacitor in parallel with a 3 bit switch-selectable binary-weighted array. Thus, the

capacitor consists of a fi xed capacitor in parallel with a 3 bit switch-selectable binary-weighted array.
capacitor consists of a fi xed capacitor in parallel with a 3 bit switch-selectable binary-weighted array.
capacitor consists of a fi xed capacitor in parallel with a 3 bit switch-selectable binary-weighted array.
capacitor consists of a fi xed capacitor in parallel with a 3 bit switch-selectable binary-weighted array.
capacitor consists of a fi xed capacitor in parallel with a 3 bit switch-selectable binary-weighted array.
capacitor consists of a fi xed capacitor in parallel with a 3 bit switch-selectable binary-weighted array.

ABIDI: THE PATH TO THE SOFTWARE-DEFINED RADIO RECEIVER

963

ABIDI: THE PATH TO THE SOFTWARE-DEFINED RADIO RECEIVER 963 Fig. 12. preselection. (a) – (e). Evolution

Fig. 12.

preselection.

(a)(e). Evolution of the sampler, baseband lter, and decimation stages to capture 20-MHz-wide 802.11g channel at zero IF in the absence of RF band

DC gain through the sampler and lter cascade can be changed with a digital word in 6 dB steps by a total of 30 dB.

Sampling of the continuous-time input waveform is not sensi- tive to the duty cycle of the clock waveform; only the repetition

964

964 Fig. 13. Circuit of entire analog baseband section, including sampler, two stages of decimation, and

Fig. 13. Circuit of entire analog baseband section, including sampler, two stages of decimation, and lters. The use of only a single active element is notable. Inset at upper right shows time-interleaved operation of FIR lter associated with rst decimation.

rate matters. All that is required is that switches should be on for long enough to complete the charge transfer from a capac- itor on one side to the other. When the current sampling switch is off, the input current waveform continues to integrate on and then redistributes the integrated charge with all at once. In this way the input waveform is sampled without interruption. This circuit now embodies all functions of the analog base- band: sampling, decimation and associated lters, and variable gain. The input transconductor is the only active circuit; the rest of the circuit consists of passive switched capacitors, which are very linear and scale well with shrinking CMOS technology. This baseband lter in the SDR receiver assumes the role of the RF passive prelter (Fig. 14). Whereas an RF prelters characteristics are xed, 3 this lter circuits cutoff frequency is clock programmable over decades. We must not overlook, though, the fact that programmability relates to chip area: (5) tells us that to maintain constant over a range of clock frequencies, must scale by the same range. This im- plies a corresponding spread in component values. The mea- sured on-chip selectivity is impressive [23] (Fig. 14), outstrip- ping most other types of on-chip lter.

14), outstrip- ping most other types of on-chip fi lter. B. RF Circuits Much innovation at
14), outstrip- ping most other types of on-chip fi lter. B. RF Circuits Much innovation at
14), outstrip- ping most other types of on-chip fi lter. B. RF Circuits Much innovation at
14), outstrip- ping most other types of on-chip fi lter. B. RF Circuits Much innovation at

B. RF Circuits

Much innovation at the transistor level lies behind the RF circuits in the SDR receiver, especially in the wideband LNA and mixer. A detailed discussion of circuits here will cause us to stray from our main theme, which deals with the evolution of a new receiver architecture. Ref. [23] and antecedent publications describe, in detail, the RF circuits at the transistor level. Here we will discuss circuit imperfections because they limit how well the receiver performs in wideband operation, as it is used here. The price of ltering after the LNA and mixer is that large unwanted signals can drive these circuits into nonlinearity, dis- torting the wanted channel. The two most important distortions are cross-modulation due to third-order nonlinearity in either the LNA or mixer, and AM detection by the second-order non- linearity at the mixers output. Cross-modulation is best under- stood in terms of gain compression. It is well-known that a large

3 This is the case today. Tunable MEMs-based lters are in the research phase.

IEEE JOURNAL OF SOLID-STATE CIRCUITS, VOL. 42, NO. 5, MAY 2007

JOURNAL OF SOLID-STATE CIRCUITS, VOL. 42, NO. 5, MAY 2007 Fig. 14. Filter con fi gured

Fig. 14. Filter congured for two different receive conditions: 802.11g, where unwanted CDMA channels band pose greatest threat; and GSM, where the lter must suppress 0 dBm blockers anywhere outside the band. Measured lter re- sponse meets specications to very large offsets.

unwanted signal at any frequency applied to an amplier with compressive nonlinearity causes the small-signal gain to drop by an amount dependent on the square of its amplitude [24].

It follows that modulation of this amplitude also modulates the

small-signal gain. In this way the unwanted signal imposes itself by subjecting a small wanted signal to modulated gain. AM de- tection in direct conversion receivers is better understood. The SDR receiver is vulnerable all the time to distortion due to AM detection and cross-modulation arising from an unwanted signal anywhere. This is why in a wideband zero IF receiver these two forms of distortion are more insidious than others. As the signal at the output of the non-selective wideband LNA has been amplied, it will cause greater distortion in the mixer that follows. Of course the mixer amplies still further but its output is ltered, and even a mild amount of ltering (as long as the lter attenuation is greater than the mixer gain) is enough to sharply lower distortion in the following circuits. In general, we may say that distortion is a problem in the receiver only up to the point that there is no ltering. In our receiver, this means

that it is a problem up to the mixer output. Thus, mixer linearity is paramount. The mixer must be based on a sound circuit topology. The best topology we have found is the double-balanced, passive FET mixer driven by a large square-wave-like LO voltage to commutate signal current [25]. This suppresses both icker noise in the mixer core as well as second-order intermodulation distortion. There is a simple way

to understand that the two are related. Offset between cross-cou-

pled mixer FETs can be modeled as a non-zero voltage source in series with the gate of one FET. It is offset, usually, which causes imbalance and mismatch that leads to second-order inter- modulation effects. By thinking of the offset as slowly varying and random, we may also model icker, or , noise. Then it follows that methods to mitigate the offset from inducing inter- modulation, such as driving the gate with square wave-like LO, also suppress noise at the mixer output [26]. There is also the problem of harmonic mixing, unique to

a wideband receiver. If the mixer is driven by an LO, say, at 900 MHz, the square-wave commutation also contains smaller

amounts of third and fth harmonic at 2700 MHz and 4500 MHz

900 MHz, the square-wave commutation also contains smaller amounts of third and fi fth harmonic at
900 MHz, the square-wave commutation also contains smaller amounts of third and fi fth harmonic at
900 MHz, the square-wave commutation also contains smaller amounts of third and fi fth harmonic at
900 MHz, the square-wave commutation also contains smaller amounts of third and fi fth harmonic at

ABIDI: THE PATH TO THE SOFTWARE-DEFINED RADIO RECEIVER

ABIDI: THE PATH TO THE SOFTWARE-DEFINED RADIO RECEIVER Fig. 15. Frequency synthesis plan using three stagger-tuned

Fig. 15. Frequency synthesis plan using three stagger-tuned oscillators and dividers to cover the SDR receivers full tuning range. Each oscillator tunes only by 15%.

that will downconvert unwanted signals at these frequencies to

zero IF, superimposing them upon, and possibly overwhelming the wanted signal at 900 MHz that has been mixed to zero IF. Bagheri et al. [23] discuss this in depth, and describe a way to lessen unwanted harmonic mixing. Last, the complete receiver needs a frequency synthesizer with wide tuning range. A single oscillator that tunes from

800 MHz to 6 GHz must be of the relaxation or ring type,

but it will consume exorbitant power [27] to satisfy the very

low phase noise for, say, GSM. An oscillator takes much less power, but its tuning range is by comparison very limited. Switched capacitor banks help to expand tuning range, but only up to the point that spectral purity enters into the tradeoff. We have found that three oscillators centered at staggered frequencies, each with 15% tuning range, and followed by

a bank of dividers and a multiplexer, can cover the band from

800 MHz to 6 GHz (Fig. 15). Only one oscillator is active at a

time. It is not wise here to synthesize frequencies by addition or subtraction with single-sideband (SSB) mixers, because residual spurs at unwanted sidebands are sure to lie in this receivers wide passband. With realistic mismatches, spurs can be as high as 40 dBc [28]. Preceded by a low-noise wideband amplier [29], the ele- ments described so far comprise the rst single-chip CMOS software-dened receiver (Fig. 11) [23]. To receive GSM at

900 MHz, it uses a 9 MHz (decimated from an initial

72 MHz) sigma-delta ADC resolving 14 bit in 100 kHz; and to receive 802.11g at 2.4 GHz, it uses an 8 bit, 40 MHz (deci- mated from an initial 480 MHz) Nyquist ADC. In todays state-of-the-art, each ADC consumes about 10 mW. The re- ceivers cascade IIP3 of 6 dBm and IIP2 of 65 dBm, limited by the LNA and mixer circuits, are good enough to enable it to demodulate a minimum GSM signal in the presence of an un- wanted 20 dBm CDMA (amplitude-modulated) channel.

of an un- wanted 20 dBm CDMA (amplitude-modulated) channel. IV. D ISCUSSION AND O PEN P
of an un- wanted 20 dBm CDMA (amplitude-modulated) channel. IV. D ISCUSSION AND O PEN P
of an un- wanted 20 dBm CDMA (amplitude-modulated) channel. IV. D ISCUSSION AND O PEN P
of an un- wanted 20 dBm CDMA (amplitude-modulated) channel. IV. D ISCUSSION AND O PEN P
of an un- wanted 20 dBm CDMA (amplitude-modulated) channel. IV. D ISCUSSION AND O PEN P
of an un- wanted 20 dBm CDMA (amplitude-modulated) channel. IV. D ISCUSSION AND O PEN P

IV. DISCUSSION AND OPEN PROBLEMS

Over the 10 years since software-dened radio was rst pro-

posed, developments following a somewhat erratic course have nally led to a practical prototype. Success has depended on ar- chitectural innovation, but also on a new generation of front-end circuits, such as the wideband low-noise amplier and mixer. The architectural innovations stem from a deconstruction of Mitolas original proposal, which reveals that SDR based on

a giantADC and powerful enough DSP can receive every

channel concurrently. This is far more than what mobile tele- phones need, where it is good enough to be able to receive two or three channels concurrently from different bands. In this con- text, we can re-dene SDR as being able to tune to one channel

965

of any bandwidth, in any band from 800 MHz to 6 GHz. This would make it a universal receiver for almost all mobile com- munication today. Tuning the channel of interest to zero IF greatly simplies the receive RF/analog signal processing. The low-pass windowed integration sampler, also known as the integrate-and-dump or boxcar integrator, takes on new life as a device to sample a nar- rowband channel at zero IF, while protecting it from aliasing an unlimited band of other signals. When these other signals are larger than the wanted one, as is usually the case, a high initial sampling frequency gives sufcient protection. Almost always this is too high to enable A/D conversion at low power. Yet the wanted narrowband channel alone can be digitized at a low rate. Decimating the analog samples closes this gap in rates. However, decimation itself needs an anti-aliasing lter, which is usually embedded in the decimator circuit. By distributing the usually large decimation factor over two or more stages, the cascaded anti-alias lters greatly suppress unwanted channels that otherwise would accumulate near the wanted channel by aliasing. This lowers dynamic range, and power, in the A/D con- verter that follows. In contrast to the elements used in conventional receivers, the circuits described so far are all very exible. For example, the LO generator is widely tunable; sampling rate depends on an arbitrarily chosen clock frequency; decimation factor, as well as the number of stages, can be selected by the user. The remaining RF and analog signal processing should stay true to this spirit. An RF low-noise amplier can be called exible if it presents uniform gain and noise gure in every communication band. We have developed such a 90-nm CMOS wideband amplier up to 6 GHz. The mixer input stage offers uniform transconductance to inputs up to 6 GHz and beyond. Analog AGC must also be kept to a minimum, and like ltering, should be pushed into the DSP as much as possible. In fact, the functions of sampler, multi-stage decimator, and AGC function can all be combined into a single passive charge-transfer circuit. Although the on-chip lter is programmable and its se- lectivity surpasses a xed RF prelter, the front-end LNA and mixer are exposed to all signals, wanted or not. The two circuits must be very linear to protect the wanted channel from corruption caused through cross-modulation or AM detection due to large, unwanted signals in any band. So far we have shown that the new receiver architecture is fea- sible for SDR. However, many open problems remain, and are worthy of future research. Wireless standards such as CDMA operate in full duplex, that is, the transmitter and receiver op- erate simultaneously through a common antenna. Although the transmitter uses its own frequency band, the transmit signal level is so high that it will saturate the receiver through the common antenna terminal. CDMA handsets today use an RF duplexer lter at the receiver input to attenuate the transmit band. But our SDR philosophy forbids the use of xed lters. Some other solution must be found. The exible receiver platform described so far receives any one channel at a time. It can also receive two or more arbi- trarily spaced channels concurrently, if the received frames fall in non-overlapping time slots, by simply hopping the frequen- cies of the LO and sampling clock. However, we have not yet

966

found a hardware-efcient way to concurrently receive multiple continuously transmitted channels. The software-dened radio is a work in progress. It promises to keep researchers in circuits and systems busy for a while.

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Electr. Eng. Dept., Univ. of California, Los Angeles, 2006. Asad A. Abidi (S ’ 75 –

Asad A. Abidi (S75M80SM95F96) received the B.Sc. (with honors) degree from the Imperial Col- lege, London, U.K., in 1976, and the M.S. and Ph.D. degrees in electrical engineering from the University of California, Berkeley, CA, in 1978 and 1981, re- spectively. From 1981 to 1984, he was with Bell Laboratories, Murray Hill, NJ, as a Member of Technical Staff at the Advanced LSI Development Laboratory. Since 1985, he has been with the Electrical Engineering Department, University of California, Los Angeles (UCLA), where he is a Professor. He was a Visiting Faculty Researcher at Hewlett Packard Laboratories in 1989. His research interests are in CMOS RF

design, high speed analog integrated circuit design, data conversion, and other techniques of analog signal processing. Dr. Abidi was the Program Secretary for the IEEE International Solid-State Circuits Conference (ISSCC) from 1984 to 1990, and the General Chairman of the Symposium on VLSI Circuits in 1992. He was the Secretary of the IEEE Solid-State Circuits Council from 1990 to 1991. From 1992 to 1995, he was the Editor of the IEEE JOURNAL OF SOLID-STATE CIRCUITS. He received an IEEE Millennium Medal, the 1988 TRW Award for Innovative Teaching, and the 1997 IEEE Donald G. Fink Award, and is co-recipient of the Best Paper Award at the

1995 European Solid-State Circuits Conference, the Jack Kilby Best Student

Paper Award at the 1996 ISSCC, the Jack Raper Award for Outstanding Tech- nology Directions Paper at the 1997 ISSCC, the Design Contest Award at the

1998 Design Automation Conference, an Honorable Mention at the 2000 De-

sign Automation Conference, and the 2001 ISLPED Low Power Design Contest Award. He is a Fellow of the IEEE, a member of the National Academy of En- gineering, and he was named one of the top ten contributors to the ISSCC.