PAPERS

Digital-to-Analog Converterwith Low Intersample Transition Distortion and Low Sensitivity to Sample Jitter and Transresistance Amplifier Slew Rate*
MALCOLM OMAR HAWKSFORD, AES Fellow

Department

of Electronic Systems Engineering,

University of Essex, Colchester C04 3SQ, UK

Multibit digital-to-analog convener technology now claims a sample amplitude accuracy of about 20 bit. However, to achieve commensurate performance in digital audio applications, both sample timing and the complete sample waveform must also have corresponding accuracies. The errors due to jitter and slew rate are analyzed, as they are treated as a unified process in the presence of a correlation between the audio signal and sample timing. The concept of jitter-equivalent slew-rate-induced distortion is introduced and an enhanced multibit topology proposed, which offers low sensitivity to both jitter and slew-rate distortion and improves upon waveform reconstruction by exhibiting no waveform discontinuities.

0 INTRODUCTION The fundamentals of sampling theory, uniform amplitude quantization, and dither are well documented [1]. If these processes are implemented correctly, the only errors at the digital-to-analog gateway output are a band limitation of the input signal and a predictable increase in noise level. However, there is evidence [2] that errors resulting from imperfect electronics do have a deleterious effect on sonic performance, even though the perceptual correlations are not completely understood. For example, although multibit digital-to-analog converters (DACs) using error-correction precise level reconstruction, techniques can achieve nonlinearity within the

retiming, this is not always achieved within the constraints of practical circuitry. A technique is presented that lowers the sensitivity of the DAC to sample timing errors and enables the converter to operate with band-limited signals, which eliminates rapid signal transitions and discontinuities. Jitter and slew-rate-induced distortion are analyzed and unifled, as similar errors result from correlation with the intersample values of the audio data. The importance of controlling both pulse shape and timing is also emphasized in sample reconstruction. I JITTER IN DIGITAL-TO-ANALOG CONVERSION

sample transition region resulting from slew-rate and induced jitter can produce impairment [3]- [6]. Jitter on the DAC conversion clock can be nonnoiselike and arises within digital circuits [7] from EMC-related interference and from imperfect phase-locked-loop (PLL) performance responding to correlation between the digital audio data and pulse timing in the digital data stream [8]- [ 10]. Although these problems can be corrected [ 11 ], [ 12] by * Presented at the 93rd Convention of the Audio Engineering Society, San Francisco, CA, 1992 October 1-4; revised 1994 August 20.
J Audio Eng. Soc., Vol. 42, No. 11, 1994 November

In this paper we define two forms of jitter and use the terminology random jitter and correlated jitter, the latter describing a sample time displacement that is correlated with the state of the system. Jitter is strictly a random event. However, the foregoing definitions are now achieving common usage in this subject area. In general a reconstructed sample can undergo both amplitude and time displacement, which together constitute an error vector E, as shown in Fig. 1. However, considering only jitter, two classes of sample format are identified: 1) Samples that are impulsive, of uniform shape, and
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noninteracting, such as switched-capacitor converters 2) 100% duration pulses, allowing nonlinear intersampie interaction within the sample sequence 1.1 Uniform Sampling with Jitter and Noninteracting Pulses Consider a uniform and impulsive data sequence of sampling ratefs Hz. The jitter model for the rth sample of weight A r with jitter ATr is shown in Fig. 2, where location is a difference between a the error andthe time-displaced version,sample the nominal and of impulse weight {Ar/fs} of a sample is defined with respect to a The Fourier transform Er(f) of the error for the rth nminalrectangularpulsefamplitudeArandwidthl/fs" sample is

timing but also affects the sample weight, as the area under a reconstructed pulse changes as it interacts with the two adjacent samples in the sequence. Consider a sample of amplitude At, nominal width 1_, and with leading- and trailing-edge jitter ATr and ATr+I, respectively. The sample construction is shown in Fig. 3. The Fourier transform Pr(f) of the rectangular pulse shown in Fig. 3 can be expressed as

Pr(f)

aj_f{ exp [ -j2wf

( -_ 1

AT r )]

- exp

-j2xrf

1 + ATr+ 1 ( 2fs )]} 1 and {2rtfATr+l} << 1,

which,
Ar e-J2_rf ATr)

for {2rrfATr}

<<

Er(f)

= fss (1 -

(la)

simplifies to - arsin(Trf/fs) fs xrf/fs [cos(Trf_ (ATr AT_+I) Ar L \fs ]

which, for 2xrf ATr << Er(f) = j _ 2_rf ATr .

1, approximates (lb)

Pr(f)

Hence for an N-sample cyclic sequence, the errorE_(LfJ N) is given by EN = j 2"rr _'_ A r ATr e -j 2,,Lr_u However, for AT r = 0 and ATr+ 1 = 0, the target Fourier transform of a rectangular sample Pr(f) is expressed as (2) PT(f) -- Ar sin(_f/f_) fs _rf/f_

-lr_0

Eq. (2) shows that the error spectrum is proportional to the harmonic number L of the sequence repetition frequency fdN Hz, but that the microstructure of the spectrum depends on intermodulation between the pulse weighting sequence {Ar}/Vand the pulse jitter sequence
error

',
,"

{AT,}N. 1.2 Uniform Sampling with Jitter and Samples with Nonlinear Interaction Although sample timing errors give rise to the error spectra described in Section 1.1, it is more common for a DAC to use 100% duration sample reconstruction. This may arise directly from the DAC output or via a sampleand-hold circuit used to eliminate glitches during DAC sampletransitions.Althoughthis strategymaximizes signal energy and improves immunity to system noise, the effect of sample jitter now not only modifies sample

vector _._,.

:E '.7 _'--

Jl ....... 6a -.
......

target ample s location _

relocatedample s j t

Fig. 1. Error vector resulting from simultaneous amplitude and time errors of a sample.

input sequence

Ar

5(0)

(jitter) time displacement

)--

error sequence

Fig. 2. Elementary model of sampling jitter.

902 J. Audio Eng. Soc., Vol. 42, No. 11, 1994 November

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Thus the error spectrum

ERr(f ) =

PT(f)

Pr(f),

that is,

ERr(f ) =ar[COS(_sf)(ATr-Arr+l)

+ jsinlXrf_(Arr ]

+[fs, Arr+l) 1 . the Fourier transform E_(Lf_/N) is

(3)

Summing the terms over N samples of a cyclic sequence,

ERN

cos

_-

Z r=0 Ar(Arr--

Arr+l)

e-J2_Lr/N

+ jsin

N-IAr
_r=_O

(AT r "ff Arr+l)e

-j2xtLr/N

(4)

Comparing Eqs. (2) and (4) there is now an in-phase component weighted by a cos(wL/N) multiplier that extends the spectrum to dc, where correlation between timing error and signal results in a complicated error spectrum that may not be masked by program material, Alternatively, an impulsive error sequence can be 1ocated at the interface between adjacent samples where the impulse weight is proportional to the pulse-area error resulting from jitter, whereas the impulse timing corresmultaneous ponds to the amplitude and timing modulation, where sijitter. This error impulse sequence has pulse-area error = {Ar+ l - Ar} {fs ATr+I}

tabulated below and computed over N = 4096 samples, where the sampling frequencyfs = 44.1 kHz, fo = fs H_/ z fi = 752f0 Hz

f2 = 1760fo Hz

d = 10 ns

Ar =

{ sin (rfxh fs] 2w

+ sin(2wrf2_ / } _, fs

An error pulse is assumed rectangular with the leading edge located at t = (r + 0.5)/f s and the trailing edge at t = (r + 0.5)/f, + AT_+ I. Thus with reference to the error pulse center the pulse timing error equals AT_+1/2. Hence the Fourier transform of the rth error pulse in the sequence is

ATr=d{sin(2xr_)+

x fa/ sin (2_r rf2'_ }

'

Here fo is the sequence repetition frequency, fl and f2 are the selected signal frequencies, and d is the jitter noise. In this example fo _ 10.8 Hz, fl _- 8.096 kHz,

ERr(f) = [ar+l - Ar][ATr+sf_] exp [-j2_ By forming a summation over an N-sample quence, a discrete transform follows as

( r + O'_5+ O'5 ATr+t) ] 'fs cyclic se-

ERN

r=_0 (Ar+ 1 -- Ar) ATr+

exp

-j

(2r

To illustrate example error characterizations of jitter when mapped onto and correlated with the audio data sequence, sets of error spectra are presented. The first set uses the data and jitter sequences {Ar} N and {ATt} N

and f2 _ 18.949 kHz. In the following simulated results all spectra are referenced to the input sequence {Ar} which is designated 0 N,

erTrea

error r_.

Ar+l

Ar-1 !_ I
aT, _ r-0.5 q
Fig. 3. Construction

i I
m.41. f_ al;+] .4_2 r+0.5 q
samples with jitter.

_t

of 100% rectangular

J. Audio Eng. Soc., Vol. 42, No. 11, 1994 November

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dB. The results shown in Fig. 4 correspond to impulsive samples as described in Section 1.1, those in Fig. 5 to the 100% duration samples described in Section 2.1. The lower frequency components are now predictably of higher level. The second set of results uses similar correlated sequences {At}jr and {ATr}_,but now a weighted random noise sequence is added to the jitter function so that z_Tr= d_sin(2arrfl_+ L \ fs/ sin(2_r rf2_ -J\ fs ] +J, Rand(r)

three-dimensional plot illustrates the effects of varying levels of random jitter noise together with the correlated displacements of the sample locations for the case of 100% samples, where J. is defined as Jn = d
100"25(1-x)

for 1 _<x <_ 16 in unit steps of x and d = 10 ns. Finally, in the third set a modified jitter sequence is simulated to demonstrate the effect of incorporating a slowly varying frequency-modulated jitter component that is superimposed upon the correlated components already described. The frequency modulation is sinusoidal with a frequency equal to the sequence repetition frequency f0 Hz, and

where Rand(r) is a random function with a triangular probability distribution function spanning -1 to + 1, and/n is the noise weighting factor. without 6 and 7 show the corresponding where J_ with and Figs. 100% sample reconstruction, results = 10 ns, meaning that the jitter probability distribution function is triangular and spans - 10 ns to 10 ns. In Fig. 8 a
-60 -80 -100

ATr=d{sin(2_)+ + Jn Rand(r)

sin(2_ \

fs

]}

.................... ,............................................... _ ............................................................................................. _....................................................................

t

............................................. _...........................................................................................................................................................................

- 120 ........................ _ .......................................................................... _ ................................................................................................................... .................. - 140 ................................................................................................ i......................................................................................................................................... -160 ........................ _.................................................................. i............................................. 4 ....................................................

-200 -220

............... ............... _ ........................ i ....................

-260 0

i 200

i 400

i 600

* 800

i 1000

1200

i 1400

i 1600

i 1800

2000

binnumber Fig. 4. Output jitter spectrum; no random jitter, impulsive samples.

-60
i

-60 .................. _ ......................... _................................... :..' .................................... _ ................ _............... ] ............ _ .............

-lOO i i ................................. ............... ....................... .................. i . i ............................... i i .................. . . ] .............. ...............

-_o.............. i ....................................... I.- _ _........................................ .................. i................... . . !........_ I1' ............... ................ -14o_ i..................... ................ i ....................................................... . i......................................... i............... i................. i....... l-li
-1.o _................. _ i ..................... i ,i . _, .................................... ; ...................... ................................ ! ..................... . ......... ..................... _,

. . . . . .i. . . . . . .i. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .t. . Ii. . . . . . .". . . . . . . .

ZOO

-2400

400

;00

800

1000

1200

1400

1600

1800

2000

bin number Fig. 5. Outputjitter spectrum; o random n jitter, 100%samples.

904 J. Audio Eng. Soc., Vol. 42, No. 11, 1994 November

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DIGITAL-TO-ANALOG CONVERTER

-80

-100

-120

-140

-16o ,
-180 0 , '200

,
, 400 600

,
i 600

i
, 1000

i
{ 1200 random

i
, 1400 , 1600 bin , 1800 number samples. 2000 jitter, impulsive

Fig.

6. Output

jitter

spectrum;

including

-60 -70 ............................ r

[ f

-100

j
{ 0 i 200 i 400 i 800 i 800 Fig. 7. Output jitter spectrum;

_
'

,_

_: .............

-120 -130 --140 -150

[ 1000

i 1200

i 1400

i 1600

i 1800

2000

bin number including random jitter, 100% samples,

spectrum an' )litude x_ k \

' 00O
Fig. 8. Jitter spectral family with varying levels of random jitter noise. 905 J. Audio Eng. Soc., Vol. 42, No. 11, 1994 November

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where etfm = 1 +
Jfm

where the integer parameter sin 2rr_) r\ as a

h is scanned over a range

amplitude weighting. of 1 _<h _< 16, which in turn addresses the sine-squared 2 SLEW-RATE-INDUCED DISTORTION

Jr,, being the modulation depth, 0 _<Jfm _ 1. This additional jitter component is introduced

frequency modulation of the periodic jitter sequence, where the modulation depth is Jfm. The results for Jf,_ = 1.8 10 -4 and Jr,, = 1 for J, = 0 ns are shown in Fig. 9, whereas the three-dimensional plots in Figs. 10 and 11 illustrate the modification in spectral form for a range of modulation depths over 1.8 10-4 _ Jfm _ 1 for Jn = 0 ns and J, = 1 ns, respectively. Finally Fig. total shows ofanoise andofdistortion to demonstrate that family spectra varies as a function the 12 level of signal amplitude. Here Jn = 0.1 ns, Jfm = 0, d = 10 ns, and the modified signal function is Ar sin2(h_r_sin(2.trrfl_ + sin(2_rrf2_ _ \ 17 / I. \ f_ ] \ fs / J
-60

The performance requirements of multibit DAC electronics for digital audio systems are stringent. First the reconstruction levels must be accurately specified. Edge jitter must be minimized. While it is a primary function of clock performance, it can result from internal circuitry exhibiting variability on propagation delay and response time as well as electromagnetic interaction between systern modules. Finally the trajectory of the signal between adjacent samples should be determined by a linear network, forming, for example, an exponential curve. However, because of the rapid response times encountered, even when sample-and-hold circuitry is used asa deglitcher, momentary nonlinearity can result in a small

-80 .......... ....... :........................i.............. i f

i......... i ............................... i..........

T r,r -,20......................... i.................... ........................

r'I'r

! i ?: ..... ..
-220 0 200 400 600 800 1000 1200 1400 11500

.
1800 2000

............................ ...............
bin number

Fig. 9. Jitter spectrum with FM jitter component, used to calibrate Fig. 10.

Jn 0ns = sp_\l \\\_ N_,_\\\\\_,\\\/\

2000

Fig. 10. Family of jitter spectra for varying FM depths; no random jitter.
906 J. AudioEng. Soc,, Vol. 42, No. 11, 1994 November

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CONVERTER

change in the pulse area, which is related to intersample values. Because of the small time duration of each nonlinear event, ideal rectangular transitions are assumed with the error modeled using an equivalent sample jitter component ATnr. Fig. 13 shows a nonlinear transition between two 100% rectangular a loss of pulse
l_kAr =

whereby % ATnr = + 2'

(6)

ideal rectangular

pulse

pulses of weight where

Tr '

Ar

and

Ar+l,

con-

Iossofpulsearea I_ Ar

lacedAT.r

area/_kAr,

(Ar+ 1

--

Ar)

i i
I

Im

ConsiderStrained by rectangularpulse wherethe transitionWhich in a a constant slew rate S V/s, resultsdisis placed from its nominal location by ATnr such that {pulse area of Ar+l} -- {pulse area of Ar} = -_,A r that is,

Ar+l __r'w. i _ aTnr-----_ I_t i

r

i _,

fs

gr+l

( _sl

Arn r)

r (j_ I

q_ Arnr)

_AAr_A

distortion. Two Fig. 13.

adjacent

samples

linked

by

dominant

slew-rate

Jn

ns

spectrum amplitude

Jfm

2000

'0 Fig. l 1. Family of jitter spectra for varying FM depths; with random jitter.

spectrum

mplitude

[2O00

Fig. 12. Family of jitter spectra with varying levels of input signal.
d. Audio Eng. Soc., Vol. 42, No. 11, 1994 Novomber 907

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But for a constant slew rate, SAr+_- Ar %

ized by observing distortion products when processing band-limited audio signals. Hence this amplifier stage requires band limitation and slew-rate prevention to yieldlow correlatedequivalentjitter, where it is suggested that wide-bandwidth, open-loop transresistance convertersare the preferredchoicewith possibleprefiltering to reduce the bandwidth of the DAC output (7) current. Also, since most DAC transresistance stages operate with 100% pulses, the effect of jitter (random or slewrate equivalent) is increased. To demonstrate this conjecture, the equations presented in Section 1.2 are used together with Eq. (7) as well as the following data to generate the error spectrum shown in Fig. 14: Number of samples N = 4096 Sampling rate (2 times oversampling) fs = 88.2 kHz Transresistance amplifier slew rate S = 50 V/l_S Sample sequence generator A r = {sin(2_rrfl/fs) + sin(27rrfJf_)} Random jitter J, = 1 ns Correlatedjitter d = 1 10-18 s where f0 = fJN Hz, fl = 992f0 Hz, and f2 = 512fo Hz, that is,fl = 21.36 kHz and f2 = 11.025 kHz. No other correlated jitter source is included. Finally using similar data, a three-dimensional plot is shown in Fig. 15, where f_ is scanned linearly in 32 steps from 689 Hz to 22.05 kHz,f2 = 11.025 kHz, and d = J, = 1 10 -18 s. The surface shows tracks of intermodulation distortion products, where the calibration of the vertical scale can be estimated from Fig. 14 that correspond to trace 31 where f2 = 21.359 kHz. 3 DAC TOPOLOGY WITH LOW JITTER AND SLEW-RATE SENSITIVITY 3.1 System Topology and Function To reduce DAC sensitivity to slew rate and jitter, the rapid signal transition at each sample boundary must be

and thus AT,_ - A_+I - Ar 2S

Although this example is idealized, it demonstrates how an equivalent correlated jitter component ATnr can be assigned to the rth sample, and thus the results in Section 1.2 can be used. The analysis ignores spectral changes relating to pulse shape, as these events are of short duration compared with the Nyquist sampling period. The use of an equivalent jitter time defined in association with slew-rate distortion and other related nonlinearities in the current-to-voltage (transresistance) stage of a DAC enables a unification of this class of problem, where Eq. (7) together with Section 1.2 permit specifyingthe performance. Equivalent jitter resulting from slew-rate distortion is potentially more serious than random jitter because of the natural correlation between slew limiting and the data samples. Although with appropriate design tighter limits can be achieved, pulse jitter greater than = 10 ns has been reported to be of audible significance, and there is anecdotal evidence to support a much tighter specification. However, using the 10-ns criteria and assuming by way of example {A_+l -- A_ --- 500 mV}, a transresistance amplifier should exhibit a slew rate greater than 25 V/ixs. Even if slew-rate limiting does not occur, an operational amplifier may be close to its open-loop limits during periods of rapid signal transition, and this may contribute momentary "packets" of distortion. There is little doubt that transresistance stages used in DAC systerns can contribute distortion that is not fully character-

iiii .... il
-i00

iii fillii i i

-801 .............................................................._ ......................................................................................................................

............................................................. _.................................................................................. _ ..........................................

.................i...............................

-tso
0

;
ZOO

]
,too

{
8oo aoo

i
lOOO

,
12OO

,
1400

,
1600

i
1800 ZOO0

bin number
Fig. 908 ]4. Distortion spectrum resulting from slew rate used to calibrate Fig. 15.

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minimized and a degree of intersample isolation introduced to control the distortion described in Section 1.2. Effectively the operating bandwidth of the DAC should be lowered so that a more "analoglike" conversion is achieved at the gateway of data conversion. The wide bandwidth generally encountered is an artifact of topology that is typified by the current-switching DAC in association with a wide-band transresistance converter. The proposed topology consists of two time-interleaved DACs operating with mild oversampling, but where the DACs are configured as multiplying converters (MDACs). The reference inputs of the two DACs are raised cosine waveforms with low inherent jitter. The basic system is shown in Fig. 16 and uses a 4 oversampiing filter to enable initial interpolation of input data. Fig. 17 shows a series of illustrative waveforms to demonstrate operation, The output of the 4 x oversampling filter is multiplexed alternately between two MDACs (MDAC_ and MDAC2) using sampled latches, where conversion occurs on the alternating data sequences D_ and D 2. The DACs therefore run at 2fn_Hz, and output pulses overlap by 1/f,s, f,_ being the Nyquist sampling frequency, Although the data applied to each DAC are held constant for two consecutive samples, examination of the respective raised cosine reference waveforms R_ and R2 shows each reference voltage to be zero on a conversion edge. Consequently assuming that there is no pulse feedthrough in the MDAC, any jitter on the data edge is attenuated. In a practical system, circuitry would arrange for data to be transformed only when the reference is zero. However, because of the near zero slope of the reference voltage waveform in close proximity to its zero value, the timing of data transition is noncritical, Once the data are latched into an MDAC, the reference voltage (controlling the gain of the MDAC) rises from zero, thus causing the output current I_ or 12 to change in direct proportion to, but weighted by, the present data value. When the cosine waveform reaches its peak

value, the reference voltage applied to the other MDAC is now at zero, at which instant its data are updated in a similar manner. The process then proceeds at a uniform rate, with data being updated on the corresponding zero of each raised cosine reference signal. The net result of this process can be summarized as follows. 1) Data conversion only occurs when the reference to an MDAC is zero. Thus the contribution of jitter is minimized. Effectively, the jitter dependence is translated from the digital data to the two raised cosine reference signals. 2) Because the current output of each DAC tracks a raised cosine, the rate of change is reduced compared with rectangular switching. Thus slew-rate-induced distortion and other minor nonlinearities within the transresistance converter are virtually eliminated. 3) Reduction of high-frequency spuriae and the use of 4 x oversampling relax the design of the analog reconstruction filter, and the output signal from the converter is more "analoglike." 4) Because a raised cosine consists only of a dc term and a single spectral line, noise filtering to reduce jitter is simplified. 5) Any imbalance in gain between MDAC_ and MDAC 2 is of little consequence and only causes a mild increase in the spectral replication at 2fns HZ, which because of 4 x oversampling, is located well above the audio band. 6) Reduced bandwidths of signals within the converter mean that circuit layout and parasitic and mutual coupling of circuit elements are less problematic. Cautionary Note. To achieve the performance specifled in the preceding, the cosine weights for all samples should be identical. Consequently the frequency response of the MDAC from reference input to current output must not be code dependent. This is not a fundamental problem, but it does require appropriate attention in the design of the MDAC.

amplitude

f2

trace number (scanning fl) 0 __ ';_ 20 Fig. 15. Scanned distortion spectrum resulting from slew rate. J. Audio Eng. oc., ol.42,No.11,1994 S V November 909

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is given by 3.2 Estimate of Jitter Suppression

AA r
=

Fig. 18(a) shows sample reconstruction for a single DAC, where the data conversion timing is optimum. However, if a timing offset TOand a superimposed jitter component ATr are introduced at the gateway of data conversion, then the waveform shown in Fig. 18(b) result. In this example only even samples in the oversampled data sequence are shown. The result of this timing error is a loss in pulse area, whichcanbe estimated follows.Thepulse-area as error

_ TO+ATr J/=0 (Ar+2 - At) [1 - cos(4,rrfnst)] dt.

After integration,

Z_Ar----(Ar+2_

At)r _T0 + ATr_ 1,

sin[4,rrfns(To+ ATr)]_ J 4_rfn_ (8a)

D1

11

NyquistPCM

2 fas(even)

_
O/P

dalAf

filter

I
4f Fig.

'"7
LATCH 16. Basic

?
_

reference

+_/

MDAC2_
MDAC topology.

7'

two-interleaved

Input data

T
| T

(Nyquist pcm' fns)

4 fimesinterpolateddata

T T T T T T T T I

I T T T _ T T T T T

Rlraisedcsinel_ference

j_/'_/_/'_/_/-_/'_j_/'_j_

R2raisedcsinerefevence

/-_j_/-_/_/_/_/_/'_/'_/-_

onM_AtC_u_Snetdc_a_lweighled/_/,l__/_j_/_/_

MDAC_ laised cosine weighled utput_'urrent:dataO2

/_/_

/_/_

/_/_/_]_/_/

cosine interpolation composite signal

Fig. 17. Illustrative waveforms showing raised cosine interpolation in time-interleaved DAC topology.
910 d. Audio Eng. Soc., Vol. 42, No. 11, 1994 November

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which, for 4.rrf.,(T 0 + ATr) <<

1, simplifies to

The Fourier transform Fc(f) then follows as

AAr

= (Ar+ 2 -

A r)

(_fns) 2

(TO +ATr)

(8b)
re' r

Fo(f) = 0.5
- l/4fos

[l + cos(4xrfnst) ] e -j2'_ft dt

Defining an equivalent time offset and jitter the rectangular pulse shown in Fig. 18,
(Ar+ 2 -Ar ) rer _ _u_ r

error

for that is, 1

yielding
8
rer = 5 ('Irfns)2

Fc(f) = 8-_
(r
+ Arr)3

[ _,2f.s

\2fJ

(9)

+
that is,
rer

I_ _,2f.s+

TO + ATr

8 3 [xrf"s (TO + ATr)]2

(10)

where sine(x) = sin(x)/x. The time and the corresponding Fourier transform of the time-limited raised cosine pulse are shown in Fig. 19. The Fourier transform shows that there is a significant response to 3f.s, but for 3f. s and above there is attenuation. However, because a 4 oversampling filter is prescribed, the first spectral replication in the final reconstructed signal is centered on 4fns Hz and extends -fas/2 Hz. Consequently the inherent attenuation offered by the raised cosine waveform at 4f_s (noting that at exactly 4f_s Hz the Fourier transform is zero) significantly suppresses the first spectral replication and thus relaxes the design of the analog recovery filter. Indeed, the spectrum in Fig. 19 suggests that the analog filter can be designed to have a band-reject response centered on 4f_s Hz or, alternatively, a twin resonant circuit with rejection bands centered on 3.5fa s and 4.5f.s, respectively, followed by a mild low-pass filter response to reduce out-of-band noise and spuriae. Finally, because of the form of the raised cosine transform Fc(f) described by Eq. (12) and to enable a flat audio passband, mild linear-phase equalization should be included in the oversampling filter to match the inverse response over the frequency band of 0-0.5f_s Hz. It may also be expedient to include a minor correction for the analog reconstruction filter transfer function, although in practice this will be small. 3.4 MDAC Nonlinearity in Reference Signal Path

Eq. (9) estimates the equivalent timing error using the raised cosine sample format compared with the case using rectangular samples, whereas Eq. (10) reconfigures this result to show the corresponding reduction in dependence of the timing errors. For example, let (TO + ATr) = 200 ns, fas = 44.1 kHz

whereby the effective reduction in timing error _ 2 x 10 -3. This demonstrates that a DAC topology with a response or settling time of only 200 ns is adequate, The analysis demonstrates a remarkable reduction in sensitivity to jitter within the digital data stream. This improvement is partially dependent on low jitter in the raised cosine waveform and a reference voltage that accurately attains a zero value at the minima of the raised cosine function. However, the form of the raised cosine waveform with only two spectral lines, dc and 2fas Hz, means that band-pass filtering can achieve a low inherent jitter performance that is considerably more effective than smoothing a high-frequency rectangular clock, as the equivalent noise bandwidth can be made lower because of the lower number of contributing harmonics, Also the band-limited raised cosine reference waveforms can be applied directly to the MDACs without additional noise-inducing counters and logic circuits. These factors, together with an almost total independence of digital data jitter, are the principal attributes of this new DAC topology, 3.3 Spectral Response of Raised Cosine Modulated Samples and Requirements on Analog

Reconstruction Filters
The requirements for analog signal recovery subsequent to digital-to-analog conversion can be determined by analyzing the combination of 4 x oversampling and the overlapping raised cosine weighting that is associated with each sample. The impulse response he(t) of the time-limited raised cosine generator can be expressed as hc(t) = 0.5 {1 + cos(4_rfr,st }rectl/2f_ (t) .
J. Audio Eng. Soc., Vol. 42, No. 11, 1994 November

(11)

Nonlinearity within an MDAC can be modeled by assuming a perfect DAC combined with a dynamic nonlinear network in the reference input, as shown in Fig. 20. Because the modified MDAC has now been desensitized to distortion components generated at the data transition, the residual errors are solely dependent on the accuracy of pulse amplitude reconstruction and the nonlinearity within the circuitry associated with the reference (gain-defining) input. If the MDAC is assumed ideal, then nonideality can be modeled by a nonlinear network in the reference channel where pulse amplitude errors can be accounted for by allowing the data input to modulate this network. We thus identify two possibly interrelated error mechanisms, which can cause distortion in the reconstructed output. However, this model
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is also useful conditions that cal audio band. input channel,

as it allows us to define succinctly the prevent distortion from entering the critiIf the only nonlinearity is in the reference and the data input in no way alters this

and consider adjacent samples of V0 such that the amplitude separation of A,+ _ and Ar is maximized when using 4 x oversampling (that is, 4f. s Hz). Thus

monic distortion to the result cosine an addition Thus the raised waveform. of hatnonlinearity, thenMDAC we is onlyexpect to observe at the output of the would a modest level of spectral replication of the input about output harmonics, which can readilyand removed by the these analog reconstruction filter be thus is of little consequence. It is only when the data sequence modifies the nonlinearity that level reconstruction errors occur, which will result in output distortion. However, this is fundamental to all multilevel DACs, and this system is no exception to this class of distortion. 3.5 Estimate of Maximum Rate of Change of Signal at Output of Transresistance Converter Consider two adjacent samples Ar and Ar+ 1 separated by the oversampled time interval 1/4f, s. The DAC attempts to edit these samples by a half-cosine wave interpolation, as shown in Fig. 17. The amplitude of this half-cosine segment is therefore {0.5(Ar+ | - At) } such that over the sample interval 1/4f,_ the reconstructed signal is V0 = Ar
+ 0.5(Ar+ 1 Ar) I1 -

A, = -A m sin () rr

Ar+ 1 -- A m sin

_(_r) . of the

Hence from Eq. (13) the minimum slew transresistance converter is

rate

Smi n

Smi n =

4"rrf._A msin _- .

(14)

By way of an example, let A m = 2N/2 V andf, s = 44.1 kHz, whereby Stain = 1.12 V/IJ-S. This basic analysis shows that for a standard 2 V,_s output signal the maximum rate of change of the output signal for the transresistance converter is constrained. Consequentlya performancecommensurate withlow inband distortion is simpler to achieve and should be compared with the example given in Section 2. 4 EXPERIMENTAL RAISED COSINE DAC

cos(4_rfnst)] .

The maximum

slope of the output signal is therefore

dt dV max= 2_rfns(Ar+l -- Ar)

(13)

Assume a maximum amplitude-coded sine wave of frequencyfJ2 Hz that has the analog form
V o -- A m sin(_rf.st)

21,An experimental raised cosine DAC parts. The in Fig. which uses commercially available is shown design employes a Micro Power Systems MP7616 16-bit CMOS four-quadrant multiplying DAC having the basic architecture shown in Fig. 22. (Although this is a 16-bit device, it is only of marginal performance for highquality applications offering a current settling time of 2

Ar+2 --

2At+ 2-

Atrectangular

II
data reconstruction

t
(a)

2Ar--_

/ _,

raised cosine data reconstruction

2A_+2- -f'_
error area I

error area

2At_

LXA_\

A__

AT

(b) Fig. 18. (a) Rectangular and raised cosine sample reconstruction (samples weighted to have same area). (b) Effect of timing offset and jitter on reconstructed raised cosine samples.
912 J. AudioEng. Soc., Vol. 42, No. 11, 1994 November

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!xs to 0.01% of FSR. However it was the only device available at the time of experimentation.)To lower tolerance on resistor matching, the MP7616 features 15 equivalued current sources with inputs decoded from the four MSBs. The remaining 12 bit are then converted using a binary weighted tree. A key feature of this DAC in the present application is the bipolar reference input, which is driven by one of the two raised cosine waveforms. SPDIF serial digital data are decoded by a Yamaha receiver (YM3623), and the sampling rate is increased eight times using a Burr Brown DF1700. The oversampled data are next converted to a parallel format and, via two alternately clocked 16-bit latches, input to the two 16-bit MDACs. Complementary parallel data allow the use of in-phase raised-cosine waveforms as defined in Fig. 21, where complementary dc offsets result in a zero-mean output current when the two MDAC outputs are summed, thus simplifying the design of the transredistance converter as no dc correction is required, Raised cosine waveform phase alignment is maintained by including the cosine waveform generator within a PLL, as shown in Fig. 23. The voltage-controlled oscillator (VCO) of the PLL drives two analog gates to produce a symmetrical square wave, which is subsequently band limited by a second-order bandpass filter.' Two in-phase signals are formed on secondary windings coupled to the tuned circuit, which are dc off-

set to form the two raised cosine waveforms. The PLL and the bandpass filter are also instrumental in reducina jitter. This system validated the operation of the raised cosine DAC, and measurement confirmed cosinusoidal interpolation between adjacent samples, thus relaxing the slew-rate requirements for the current-to-voltage converter. In this sense the DAC can be seen to filter the MDACoutputcurrentwaveform,but withoutcompromising noise performance, which occurs when a filter is placed between the DAC and the transresistance converter. Also such filters do not reduce the jitter present in the source data. However, in this experimental model the resolution of the DAC and its settling time were limiting factors. 5 CONCLUSION This paper has described a technique of time interleavina two MDAC converters using complementary raised cosine generators applied to the reference input. The effect of this process is to replace the normalrectangular pulses in a 4 x oversampled converter with raised cosine weighted samples that are time limited to span two consecutive samples in the oversampled data stream. The effect of this process is a reduced sensitivity to data timing jitter and transition distortion as well as a reduction in the slew-rate requirement of the transresistance amplifier. It was also shown that the requirements of the analog recovery filter at the DAC are relaxed. Theelimination edge-transitionistortion of d andthe

i' ,

III'IIII ii i.: 4 ii i!_i

i i ...... ..i ; : _i

ii.i.

}ii 1.11117 i iii : i i'ii ZlZll

clocks to the raised cosine generators are seen as pivotal presented to the transresistance stage. A signal (the ref-

i i .... ii iiii ii ,_oo

in the design, as is the reducedbandwidthof the signal tral lines (dc and jitter dependence to filter to suppress transformation of 2JnsHz) is simpler fromonlytwo specerencegeneratoroutput)that consistsof the DAC case noiseand spuriae,the sourceofjitter, thanin the data of a broad-band square wave. Even if a square wave timingsignalis combfilteredto containonlythefundabandwidthassociatedwith each harmonicwhich is greater thanthatof theraised cosine. To support the proposal for a DAC with low jitter sensitivity, ananalysiswaspresented thatdescribes the mechanismits harmonics,there is stilldistortion mentalandby which jitter canintroduce a finite noise into the audio band. It was shown that jitter that mapped to pulse-area distortion as well as producing timing errors resulted in greater low-frequency distortion. For jitter that only mistimeda sampleevent, the distortion spec-

_0

0_ii_ ! ?

::i:: -2

: :_ :::: o

: '::

: ::i:: 5 :: 2

Fig. 19. Raised cosine waveform in both time and frequency domains,

'1
generator Raised cosine Data input _ network Nonlinear l
rererence input

Id esi
"_

current

output

_,l

MDAC)

Fig. 20. Basic model of MDAC nonlinearity. d.Audio Eng.Soc., ol.42,No.11,1994 V November 913

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J. Audio Eng.Soc., ol. V 42,No.11,1994 November

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CONVERTER

trum was proportional to the frequency, whereas the inclusion of area modulation extended this response to dc. Analysis and computer simulation revealed that the distortion was more problematic where a correlation exists between timing error and program material. Indeed, in many practical digital systems the use of PLLs with inadequate timing recovery can yield correlation, even though this process can be highly nonlinear. For example, where a PLL responds to a change in the data sequence, even though this is in coded form, correlation can exist and a highly complicated jitter spectrum result, which even though of very low level, does not necessarily fall under the auditory mask causing perturbations of the detection thresholds within a number of critical bands [13]. It can be argued that an inherent design limitation of the SPDIF serial code is the nonscrambling of serial data by coding to break the correlation between audio data and bit-pattern-induced jitter. If scrambling were used, any resulting jitter that is related to the serial bit pattern would be decorrelated, and therefore would produce only a noiselike residue of benign character. This is a conjecture developed in a supporting paper [8]. were included as well as the effect of adding low-frequency modulation. Fig. 8 demonstrated that the correExample results for both and random jitter lated and can be considered correlated did not intermodu- a late and random jitter components essentially additive for constant-amplitudeinput sequence. However,the inclusion of low-frequency (sinusoidal) modulation of the jitter sequencemimickinga low-frequency error in a PLL, for example, produced significant levels of intermodulation. with and10 and 11 a random jitterthis interacFigs. tion both without demonstrated sequence. However, because all the jitter-induced distortion spectra are dependent on the amplitude of the signal seto the quence,signal level, whereby proportion. To demonstrate if the signal is with respect distortion the spectral in directshould all be read reduced, the changes levels this inherent characteristic, Fig. 12 showed the distortion spectrum as a function of signal level, where the

form of the distortion remains the same, but in direct proportion to the signals. This is true whether or not there is correlation between signal and jitter. To illustrate this feature, Fig. 12 included both random and correlated components. A powerful extension of the jitter analysis was to consider edge transition distortion resulting from slewrate limiting in the transresistance converter, or indeed inherent within the DAC, and to translate this to an equivalent edge jitter when using 100% rectangular samples. The transformation revealed that edge jitter, slew rate, and related transition distortion fall into a common regime and that a similar analysis procedure can be used. However, with DAC transition distortion the correlation with the signal will almost certainly be higher, implying a greater subjective significance. Although slew rate is a dominant distortion, it should be recalled that the operational amplifier, at the edge transition, is operating near open loop, so although the transition may appear well

vc

r--_,'_Iil

-I 01 , _o..r-I
analog

band-pass filter =352kHzcf

_ , ] ',_..

raised cosineo/p 5(1 + cos(2_f0) sv +

raised

gat_ ,z ,

i.21

cosineo/p

'Vc I

' $(] ' cosa_ft))

fp Phase-lockloopipl[

8timesNyqist (=352kHz)

Fig. 23. Raised cosine generation using bandpass filter and phase-locked loop.

VDD

I
reference input Equally weighted current sources

I I I I I I I I I I I I I I t
To switches _._ O lut2 '0 aFB

I*
1 2 MSB 3 4 5 6 7 8 9

To 12-bit DAC

lO

11

12

13

14

15

16 LSB

GND

Fig. 22. MP7616 four-quadrant

J. Audio Eng. Soc., Vol. 42, No. 11, 1994 November

multiplying DAC.
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behaved, there may nevertheless be some distortion induction. In this paper the results presented are relatively extreme and do not take account of any softening of the DAC transitions, either internally or by using a transresistance stage with a capacitive feedback element or current input filter. However, the simulations do indicate complicated distortion spectra that are increased in magnitude due to the fundamental correlation between signal and jitter equivalence, as illustrated in Fig. 15, where it can also be observed that the distortion terms are not well matched to psychoacoustic masking thresholds, thus possibly gaining in subjective significance. It should be noted that when correlation between signal and jitter was considered and the distortion calculated by Eq. (5), the differential of the signal sequence was multiplied by the jitter AT r, which was made directly proportional to the signal. However, for slew-rate-induced distortion the jitter equivalence described by Eq. (7) is proportional also to the differential of the input sequence. Therefore the resulting distortion is proportional to the square of the differential of the input, a subtle but possibly significant difference that implies a greater intermodulation distortion dependence on highfrequency signal components, The time-interleaved dual DAC topology is potentially less sensitive to many of these problems. Provided the MDACs can achieve accurate level reconstruction and their output/reference input frequency response is not code dependent, then even if jitter exists and data transition distortion would normally occur with rectangular samples, these errors are of little consequence, as was demonstrated by the equivalent timing displacement estimated in Section 3.2. Thus the relatively low bandwidth excitation of the transresistance converter together with the relaxation of the analog filter topology in association with a standard 4 or 8 oversampling filter in the digital domain should result in near theoretic performance. Section 4 presented an experimental system to confirm operation, although performance restrictions of the available four-quadrant MDAC should be noted. It is possible that an MDAC may exhibit code-dependent distortion, although the relaxation of slew-rate-dependent distortion is potentially of greater benefit. Also, although analog filters can be used to band-limit the output waveform of a DAC prior to the transresistance stage, hence reduce slew-rate-induced distortion, this does not address jitter or distortion present within the DAC settling period after conversion. Such circuits usually imply a high-frequency noise penalty due to a shunt impedance at the transresistance input. There are DACs having 20-bit amplitude resolution that were designed for high-quality digital audio and military systems. If these can be modified to include access to the reference input and thus allow operation as a precision MDAC, there now exists a means to virtually eliminate the vestiges of a number of inherent imperfections, which, although of low level, can still pervade DAC systems and represent an ultimate performance bound.
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6 ACKNOWLEDGMENT The author would like to thank Andrew McCarthy and Phillipe Dolman for their work on constructing a prototype DAC as part of their B.Eng. program in the Department of Electronic Systems Engineering at the University of Essex. 7 REFERENCES [ 1] M. O. J. Hawksford, "An Introduction to Digital Audio," in Proc. lOth Int. AES Conf. (London, 1991 Sept.), pp. T3-T42. [2] J. R. Stuart, "Estimating the Significance of Errors in Audio Systems," presented at the 91st Convention of the Audio Engineering Society, J. Audio Eng. Soc. (Abstracts), vol. 39, p. 1011 (1991 Dec.), preprint 3208. [3] S. Harris, "The Effects of Sampling Clock Jitter on Nyquist.Sampling Analog-to-Digital Converters, and on Oversampling Delta-Sigma ADCs", J. Audio Eng. Soc., vol. 38, pp. 537-542 (1990 July/Aug.). [4] P. van Willenswaard, "Industry Update," Stereophile, vol. 13, pp. 78-83 (1990 Nov.). [5] J. A. Atkinson, "Jitter, Bits and Sound Quality," Stereophile, vol. 13, pp. 179-181 (1990 Dec.). [6] R. Harley, "Industry Update," Stereophile, vol. 14, pp. 38-45 (1991 Sept.); vol. 16, p. 65 (1993 Feb.); vol 16, pp. 47-91 (1993 Sept.). [7] E. Meitner and R. Gendron, "Time Distortions within Digital Audio Equipment Due to Integrated Circuit Logic Induced Modulation Products," presented at the 91st Convention of the Audio Engineering Society, J. Audio Eng. Soc. (Abstracts), vol. 39, p. 992. [8] C. Dunn and M. O. J. Hawksford, "Is the AES/ EBU/SPDIF Digital Audio Interface Flawed?," presented at the 93rd Convention of the Audio Engineering Society, J. Audio Eng. Soc. (Abstracts), vol. 40, p. 1040 (1992 Dec.)., preprint 3360. [9] J. Dunn, "Jitter: Specification and Assessment in Digital Audio Equipment," presented at the 93rd Convention of the Audio Engineering Society, J. Audio Eng. Soc. (Abstracts), vol. 40, p. 1040 (1992 Dec.), preprint 3361. [10] J. Dunn, "Considerations for Interfacing Digital Audio Equipment to the Standard AES-3, AES-5, AES11 ," in Proc. lOth Int. AES Conf. (London, 1991 Sept.), pp. 115-126. [11] R. D. Fourre, "Jitter, Jitter, Jitter .... "Application Note AP-03, Ultra Analog Inc., Fremont, CA (1992 Sept.). [12] M. O. J. Hawksford, Letter in response to R. Adams, "Comments on 'Chaos, Oversampling, and Noise-Shaping in Digital-to-AnalogConversion,'" J. Audio Eng. Soc. (Letters to the Editor), vol. 38, pp. 767-768 (1990 Oct.). [ 13] J.R. Stuart, "Predicting the Audibility, Delectability, and Loudness of Errors in Audio Systems," presented at the 91st Convention of the Audio Engineering Society, J. Audio Eng. Soc. (Abstracts), vol. 39, pp. 1010-1011 (1991Dec.),preprint3209.
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THE AUTHOR

Malcolm Omar Hawksford is director of the Centre for Audio Research and Engineering and a professor in the Department of Electronic Systems Engineering at the University of Essex, where his interests encompass audio engineering, electronic system design, and signal processing. Professor Hawksford studied electrical engineering at the University of Aston in Birmingham where he gained a First Class Honours B.Sc. and Ph.D. The Ph.D. program was supported by a BBC Research Scholarship and investigated the application of delta modulation for color television and the development of a time-compression/time-multiplex system for combining luminance and chrominance signals. During his time at Essex University, he has undertaken research on amplitier studies, digital signal processing, and loudspeaker systems. Since 1982 research into digital crossover systems and loudspeaker equalization has been pursued

which has led to an advanced digital and active loudspeaker system being produced by the University Cornpany, Wivenhoe Enterprises, under the name Essex Audio. Research has also encompassed oversampling and noise shaping as a means of analog-to-digital and digital-to-analog conversion that includes digital linearization of PWM encoders. Professor Hawksford has published in the Journal of the Audio Engineering Society on topics that include error correction in amplifiers, oversampling techniques, and MLS techniques. His supplementary activities include writing for Hi-Fi News and Record Review and designing high-end analog and digital audio equipment. He is a chartered engineer and is a Fellow of the AES, the Institution of Electrical Engineers, and the Institute of Acoustics. He is also technical adviser to HFN and Record Review and a technical consultant to LFD audio, UK.

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