Timing Recovery in Digital Receivers Using Multirate Digital Filters
Overview
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Table of Contents
- 3.5 Timing Recovery in Digital Receivers Using Multirate Digital Filters
- 3.5.1 Timing Recovery in a Classical Analog Receiver
- 3.5.2 Timing Recovery in the Digital Domain Only
- 3.5.3 Early-Late Gate Synchronizer
- 3.5.4 Timing Offset Control Using the Early-Late Gate Principle
- Relevant NI products
- Buy the Book
3.5 Timing Recovery in Digital Receivers Using Multirate Digital Filters
Multirate signal processing is very useful for creating flexible timing recovery techniques
for radios supporting multiple waveforms. Multirate digital signal processing timing recovery
techniques may also reduce complexity compared to conventional timing recovery


techniques that rely on altering the sample rate of the ADC. A feedback loop is used in
conventional techniques to adjust ADC timing, which spans across both analog and digital
domains, greatly complicating the overall design. Multirate digital filters can be used
in digital signal processing based receivers to avoid crossing between domains and to allow
for simplified timing recovery, providing a more flexible approach to support multiple
waveforms.
3.5.1 Timing Recovery in a Classical Analog Receiver
In the classic I&Q receiver, shown in Figure 3.59, a PLL is used to control the phase
of a VCO. By altering the phase, the output of the matched filter is sampled at the optimal
instance. Note that this structure is somewhat inflexible since software control of the analog
matched filter is difficult to accomplish.
A more modern approach is shown in Figure 3.60. The signal is sampled prior to the
matched filtering operation. However, the timing recovery requires feedback between the
analog and digital domains, which complicates the overall design.
3.5.2 Timing Recovery in the Digital Domain Only
A multirate digital signal processing approach can be used to avoid the need to create
a feedback loop with the ADC. The idea is simple: the digitized signal is interpolated
using a polyphase filter and the sample point with the maximum energy/power is chosen,
as illustrated in Figure 3.61. Two possible approaches for this implementation exist. In
the first, the sampling process is performed asynchronously at two samples/symbol and
the subsequent polyphase filter interpolates the digitized signal to a higher sampling rate.
The correct sample is chosen from the interpolated signal as indicated in Figure 3.61 and
the data stream is then downsampled to the symbol rate by selecting the optimal point in
the symbol. Since only a limited number of interpolated samples are actually needed to
determine the actual sample, a full interpolation filter can be avoided.
A second approach is to sample the input at a rate higher than the Nyquist rate and then
decimate it, using a polyphase filter, to the desired output data rate. Timing information is
recovered by controlling the starting index of the input samples presented to the decimating
polyphase filter’s commutator. An advantage of using a polyphase filter approach is that,
along with the downsampling operation, it performs a spectral translation by multiples of
the output data rate since the signal is being resampled.
For a given signal modulation and roll-off factor α of a raised-cosine pulse, the interpolation
factor and the number of stages of interpolation required depends on the number
of samples per symbol. The interpolation factor should be large enough to keep the implementation
loss due to missampling within a specified limit. As the roll-off factor decreases
for a given modulation type, timing becomes a more critical issue and the number of stages
of interpolation must increase to prevent significant sampling timing error. For example,
for 256-QAM modulation with
an interpolation factor of I = 271 is required,
compared to I = 500 for the same modulation with
, to keep the loss below 0.12
dB [43]. The direct method of implementation of the latter case is to operate at one thousand
samples per symbol by using five hundred polyphase stages to increase the sample



Figure 3.61: Illustration of Interpolated Nyquist Pulse.
rate by a factor of five hundred. However, the unnecessary computation involved can be
avoided by using only the specific taps in the polyphase filter that output samples close to
the optimum sample point. Such an implementation can be done using the early-late gate
synchronizing technique.
3.5.3 Early-Late Gate Synchronizer
For illustrative purposes, consider the simpler problem of detecting the optimal sampling
time of a rectangular pulse that is match filtered. The early-late gate synchronizing technique
exploits the symmetry of the signal or the symmetry of the matched-filtered signal.
The matched-filtered signal output R(t) typically has a symmetric shape (neglecting distortion
and noise). The optimal timing is obtained when R(t) is sampled at t = T. If we
take two measurements R(T0 + d) and R(T0 − d) and if T0 = T, the two measurements
are equal and the optimal sample would be located halfway between the two samples, as
illustrated in Figure 3.62. Using this property, an error signal to a timing recovery loop can
be created:
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The variable Δ is called the early/late decision statistic and is an approximation of
the scaled derivative of the matched-filter output. If Δ > Δ thresh early, the sampling
clock should be decreased by some amount. If Δ < Δthresh late, then the sampling clock
should be increased. This simple principle, which was illustrated in continuous time, can
be extended to discrete time.
3.5.4 Timing Offset Control Using the Early-Late Gate Principle
The fundamental goal of symbol synchronization is to sample the pulse at its peak value.
This peak can be determined by estimating the derivative of the sampled signal. The first
derivative is calculated as demonstrated—taking the difference of the early and late measurements
that encompasses the estimated location of the peak. In Figure 3.63, the earlylate
gate derivative measurement
for the kth polyphase output at time n can
be computed from the outputs of k − 1 and k +1 filter stages [43, 44].

Figure 3.62: The Matched Filtering Process Assuming a Positive Rectangular-Shaped
Pulse.
The decision statistic here is the product of the kth polyphase stage and its derivative.
This product is necessary to invert the derivative when the output is negative. Thus, the
decision statistic will be independent of the desired symbol’s value. Driving this product
to zero by means of a feedback loop leads to maximum likelihood timing recovery. Notice
that the product goes to zero if either the derivative is zero (at the peak) or the kth stage
output (signal) is zero.
The process of forming the derivative can be implemented by means of a polyphase
derivative matched filter with weights defined as the difference in weights of the k − 1
and k + 1 stages shown in Figure 3.63. That is, the derivative can be implemented as a
polyphase filter and thus only one filter actually needs be computed to supply information
to the feedback loop. Such a filter in conjunction with a polyphase-matched filter can implement
the system of Figure 3.63. A generic DSP-based receiver that uses the polyphase
filter approach for timing recovery is shown in Figure 3.64. Note that there is no feedback
between the digital and analog domains. Furthermore, this approach is well-suited
for multiple channel systems, such as FDMA systems, since each channel can have its own
interpolation filter. In this case, the polyphase filters can also aid in the receiver channelization
as well as in timing recovery.
This approach to synchronization has the maximum flexibility. The analog section
becomes more straightforward since the analog processing can be generic, and matched
filters for specific pulse-shaping and timing considerations are handled within the more
flexible digital domain. Note that the resampling process translates the signal by multiples
of the data rate. This reduces the carrier offset and subsequently reduces the complexity of
the digital signal processing based carrier recovery technique since the residual frequency
offset is lower after resampling.


Relevant NI products
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