Title of Invention

A TELECOMMUNICATION NETWORK FOR ACHIEVING TIME SYNCHRONISM BETWEEN A TRANSMITTER AND A RECEIVER IN TIME DIVISION MULTIPLE ACCESS (TDMA) COMMUNICATION SYSTEMS AND METHOD OF APPLYING THE SAME

Abstract A telecommunications network participant,comprising means (302,402) for digitising, as a series of samples, a received signal containing a suceession of symbols arrange in time slots of a repeating time frame, means (305, 405, 407) for measuring time misalignment between the symbols and the samples and means (306, 406, 409) for applying a fractional delay to the position of the oexamples to reduce the misalignment, characterised in that, for each of a plurality of said time slots, the measuring means is arranged to deduce a timing error for the respective slot and the delay applying means is arranged to apply a fractional delay to suppress the timing error of the respective slot.
Full Text



Field of Invention :
The proposed invention describes a mechanism for achieving time synchronism
between a transmitter and a receiver in Time Division Multiple Access (TDMA)
communication systems.
Background of Invaition :
Figure 1 illustrates a mobile telephone 100 that commumcates with a base station (not
shown) using a time division multiplex (TDM) scheme such that bursts containing
series of digital symbols are periodically exchanged between the telephone and the
base station. Figure 1 illustrates the mobile telephone 100 from the standpoint of its
function as a receiver. Figure 1 illustrates certain major processes that are involved in
the processing of received signals and the skilled person will realise that some
processes have- been omitted for the sakes of clarity and brevity. It will be understood
by a person skilled in the art that whilst Figure 1 illustrates a chain of processing
activities, Figure 1 does not necessarily reflect how these processes would be
implemented in a practical device. Usually, the processes shown in Figure 1 would be
implemented as a mixture of hardware and software entities.
A signal burst received at the antenna 109 of the telephone 100 from the base station
is first processed by the Radio-Frequency (RF) unit 101 to produce an analogue
electrical signal which is passed to a mixed-signal processing unit 102 for conversion
to a digital format The mixed signal processing unit 102 performs analogue to digital
conversion (ADC) of the analogue signal at 103, followed by digital low pass filtering
at 104 and then decimation at 105. The decimated signal is then passed to a digital
signal processing block 106. The processing performed by block 106 includes
demodulation at 107 to produce estimates of the transmitted information bits. The
estimated information bits are then put to their intended purpose, typically the
reconstruction of speech.
Object of the Invention :
The purpose of the decimation process 105 is to reduce the number of samples per
second which are provided to the digital signal processing unit 106. This rate
reduction is defined by a decimation factor D. The decimation process 105 outputs
one sample for every D samples from the low-pass filtering process 104. The

sampling phase of the decimation process 105 is the position, within a group of D
samples that are to be decimated, of the sample that is to survive the decimation
process and be provided as an output. Due to implementation complexity issues, the
sampling phase in the mixed-signal processing unit 102 is usually fixed.
Description of Prior Art :
In addition to producing estimates of transmitted information bits, the digital signal
processing block 106 also generates estimates of a timing error indicating any
mismatch between the sampling instants used by the ADC unit 103 and the
boundaries of information symbols in the current burst from the transmitter. The
timing error estimates are then used to correct the timing of the sampling instants that
are used by the ADC unit 103 in the digitisation of the next burst to be received from
the transmitter. Averaging or filtering of the timing error estimates is performed by
process 108 prior to their use in adjusting the sample acquisition timing of the ADC
unit 103. This filtering stage 108 improves the accuracy of the corrections that are
made to the sample acquisition timing of the ADC unit 103. The loop through
filtering stage 108 is known as the timing recovery loop.
The approach described above imposes two constraints on the performance of the
timing-recovery loop. First, the timing resolution of the samples processed by the
digital; signal processing block 106 is set by the frequency at which the mixed-signal
processing unit 102 operates (i.e. the frequency at which ADC unit 103 produces
digital samples) and the decimation factor J} used by process 105. If, in order for the
telephone to demodulate received bursts satisfactorily, a better timing resolution is
required by the digital signal processing section 106, either the decimation factor
needs to be reduced or the operating rate of the mixed-signal unit 102 needs to be
increased. Either of these options will increase the power consumption of the mixed-
signal processing section 102 and will also increase the memory required by the
digital signal processing block 106 to store the digital samples arriving from the
mixed-signal processing stage 102. Second, once the mixed-signal unit 102 starts
generating samples, the period between consecutive samples is effectively constant,
This is due to the fact that usually it is not possible to satisfactorily change the
sampling phase of the decimation process 105 while receiving samples from the
upstream processes 103 and 104, These limitations may lead to receiver performance

degradation in a number of circumstances. Two such scenarios will .now be
described.
Advantages and Disadvantages over Prior Art :
In the E-GPRS (Enhanced General Packet Radio Service) system, the information is
transmitted in bursts, also referred to as slots, of a fixed size. A burst is defined to
contain 156% symbols ('3GPP TS 05.02, 3rd Generation Partnership Project,
Technical Specification Group GSM/EDGE; Radio Access Network; Multiplexing
and multiple access on the radio path'). Hence, when multi-slot transmission is used
to achieve high throughput, the receiver riming will slip by an extra quarter symbol
for each slot following the first one. The SPSK modulation, which is used in the E-
GPRS system, is very sensitive to timing errors, even when the timing error is sub-
symbol. Hence, if this quarter symbol slip is not corrected at a receiver, transmission
link quality will degrade and link throughput will be reduced. One potential way to
correct this quarter symbol slip is to change the sampling phase of the decimation
process 105 between the different slots. However, this usually is not possible without
having to completely refresh the contents of the preceding low-pass filtering process
104 and such a refresh event will result in the loss of a few symbols of the received
burst thus degrading receiver performance. Hence, the approach described in Figure 1
for the receiver timing recovery loop will not perform very well for the E-GPRS
system.
Another possible scenario where the timing recovery loop of Figure 1 may not
provide adequate performance is when constraints are set on when the timing of a
receiver can be changed. For example, in order to reduce the implementation
complexity, it is possible for a receiver to use the same timing reference for both
uplink and downlink timing. In this case, transmission timings and reception timings
are derived from a single master timing event. In cellular communication systems,
limits can be imposed by a network as to when uplink timing can be changed. For
example, in the E-GPRS system, uplink timing can only be changed if a timing error
greater than half the duration of one symbol is present. If uplink and downlink
timings are locked, this restriction on uplink timing changes can result in a downlink
timing error of up to the duration of half a symbol. Such a residual error can degrade
the performance of the demodulation process 107 and will reduce communications

link throughput. One possible way to address these drawbacks discussed .above is
shown in Figure 2.
The approach used in Figure 2 is similar to the that used in Figure 1 and elements 201
to 205, 208 and 209 of Figure 2 correspond, respectively, to elements 102 to 108 in
Figure 1 and their function will not be described in detail again. However, two new
processes 206 and 208 have been added to the digital signal processing stage 205.
The signal from the mixed-signal processing unit 201 goes first through a low-pass
filtering process 206 and is then decimated using process 207 before being
demodulated at 208. Any potential frequency aliasing that could be introduced by the
decimation process 207 is removed by the low-pass filtering process 206.
The minimum sampling rate at the output of the mixed-signal stage 201 needs to be
set to meet the timing resolution required by the digital signal processing stage 205.
This means that the sampling rate at the output of the mixed-signal processing unit
will usually be significantly higher than the information data rate. The higher
sampling rate at the input to the digital signal processing block 205 makes it possible
for the receiver to adjust the timing of the received signal by selecting different
sampling phases in the decimation process 207, The decision on the sampling phase
to be used in the decimation process can be derived from different sources. It can for
example be based on the estimates of the residual timing error. This overall approach
leads to a timing recovery loop with a potentially very fine time resolution. However,
there are two major obstacles to the implementation of the solution shown in Figure 2.
First, to.obtain the fine time resolution, the mixed-signal processing unit needs to be
operated at clock rates significantly larger than the information date rate. This will
have associated costs in terms of silicon die size and/or power consumption. Second,
because the sampling rate at the input of the digital signal processing will be higher
than the information rate, larger buffers will be required to store the receiver samples.
This will have a deleterious impact on the overall memory requirements of the
receiver.
Sjimary of the Irivantion :
The present invention aims to provide an alternative timing recovery loop for control
of receiver timing.

According to one aspect, the invention provides a telecommunications network
participant, comprising means for digitising, as a series of samples, a received signal
containing a succession of symbols, means for measuring time misalignment between
the symbols and the samples and means for applying a fractional delay to the
positions of the samples to reduce the misalignment.
The invention also consists in a method of digitising a received telecommunications
signal, the method comprising digitising, as a series of samples, a received sipal
containing a succession of symbols, measuring time misalignment between the
symbols and the samples and applying a fractional delay to the positions of the
samples to reduce the misalignment.
The invention may permit a fine timing resolution to be achieved without the
necessity of a high sampling rate. In turn, this may lead to a reduction in power
consumption and memory requirements.
In certain embodiments, the timing of the digitisation of the samples is adjusted to
suppress the misalignment.
In certain embodiments, the samples are shifted in time, either forwards or backwards,
by one or more integer symbol positions to suppress the misalignment. For example,
by applying an integer sample shift of +1 sample positions and a fractional delay of %
of a symbol position, a net misalignment of +lA sample positions can be corrected.
In certain embodiments, the received signal has a format such that the symbols are
arranged in bursts, a respective timing error is deduced for each of one or more bursts
and a fractional delay is applied to the positions of the samples to suppress the timing
error or errors.
In certain embodiments, the received signal has a format such that the symbols are
arranged in bursts, a timing error is deduced for each of several bursts and the errors

are combined to produce a resultant error and a fractional delay is applied to the
positions of the samples to suppress the resultant error.
In certain embodiments, the received signal has a format in which the bursts are
grouped into a repeated time frame, the time frame containing a number of time slots,
each time slot containing a burst and, for each of a plurality of said time slots, a
timing error is deduced for each of one or more bursts in the respective slot and a
fractional delay is applied to the positions of the samples to suppress the timing error
or errors of the respective slot.
In certain embodiments, the received signal has a format in which the bursts are
grouped into a repeated time frame, the time frame containing a number of time slots,
each time slot containing a burst and, for each of a plurality of said time slots, a
timing error is deduced for each of several bursts and timing errors of the respective
slot are combined into a resultant error for the respective slot and a fractional delay is
applied to the positions of the samples to suppress the resultant error of the respective
slot.'.
In certain embodiments, timing errors from different time slots are combined to
produce a sampling timing error and the timing of the digitisation of the samples is
adjusted to suppress the sampling timing error.
In certain embodiments, a timing error for a burst is deduced by calculating the
position of a known training sequence in ihe burst and measuring the time offset
between said position and an ideal position in the burst of said training sequence.
In certain embodiments, a desired fractional delay is applied to samples by
appropriately reconfiguring a fractional delay filter providing the delay. Several pre-
stored configurations of the filter may be provided such that the filter can be
reconfigured by selecting for the filter the configuration that most closely matches the
desired fractional delay.

In certain embodiments, the digitisation of the signal takes place at substantially the
same rate as the information modulation rate of the received signal.
The invention is applicable to various communication systems and is particularly well
suited to the E-GPRS system.
The invention can be realised in hardware, in software on a processor, or a
combination thereof.
Description of accompanying drawings :
The invention can be utilised in, for example, a base station or a mobile telephone.
By way of example only, certain embodiments of the invention will now be described
with reference to the accompanying Figures, in which:
Figure 1 shows a conventional mobile telephone, focussing on aspects of related to its
timing recovery loop;
Figure 2 shows a potential modification that could be made to the timing recovery
loop of Figure 1 to improve the time resolution of the loop;
Figure 3 presents a timing loop according to the present invention;
Figure 4 illustrates another embodiment of a timing loop according to the present
invention: and
Figure 5 shows the two alternative formats for information bursts transmitted by a
GSM/GPRS/E-GPRS base-station.
Figure 3 illustrates an adaptation of the timing recovery loop described earlier with
reference to Figure 1. Elements 301 to 304,305 and 307 correspond, respectively, to
elements 102 to 107 of Figure 1 and their functions will not be described again in
detail. Whereas in Figure 2 the sampling rate at the output of the mixed-signal
processing stage is set to provide the timing resolution required by the digital signal
processing block 305, the sampling rate at the output of the mixed-signal unit 301 is

set merely to the rate required by the demodulation process 307 in the absence of any
timing error. This will be typically of the same order as the information data rate. By
keeping the sampling rate at the output of the mixed-signal unit 301 to a low value,
power consumption and memory requirements are reduced. However, this approach
means that a residual timing error can be present in the signal input to the digital
signal processing block 305. This error is removed by fractional delay process 306
which acts on the signal en route to the demodulation process 307.
Detailed Description of preferred erfcodinEnts :
In this embodiment, the fractional delay process 306 implements a fractional delay
filter. Such filters are designed to have a flat amplitude response and a linear phase
response across the bandwidth of the input signal. Hence, information passing
through process 306 is not modified but simply delayed. By selecting the slope of the
linear phase response a delay of any given value can be applied to the sampling
points. The operation of process 306 can be regarded as equivalent to that of a perfect
interpolator (as long as the sampling rate satisfies the Nyquist criterion). Of specific
interest to the proposed invention is the fact that the delay introduced by the fractional
delay process can be lower than the period of the samples at the output of the mixed-
signal processing unit. Hence, the fractional delay process 306 can implement a
fraction-delay filter to correct any residual timing error in the signal supplied by the
mixed-signal unit 30 1.
The value of the delay introduced by the fractional delay process 306 is derived from
estimates of the residual timing error calculated by the digital signal processing block
305. A timing error estimate is produced for each burst that is processed by block
305. These timing error estimates are passed to a low-pass filtering, or averaging,
process 308 such that the accuracy of those estimates can be improved,
Improvements to the accuracy of the corrections being made by the fractional delay
process 306 can be achieved by selecting the timing error estimates used by the low-
pass filtering process 308. If noisy timing error estimates are excised and not used to
calculate the value of the correction to be applied by the fractional delay process 306,
the residual timing error at the input to the demodulation process 307 can be reduced.

excluded from the filtering process 308. For example, the digital signal processing
block 305 could be configured to send the filtering process 308 a timing error only if
the burst to which that error relates has been demodulated with less than a certain
proportion of errors.
The timing error estimates are also used to correct the timing of the digital sample
acquisition timing by the ADC unit 302. As is done with the timing corrections made
in the digital signal processing section, the timing error estimates are first low-pass
filtered in process 309 before being used to adjust the digital sample acquisition
timing of the ADC unit 302.
Various approaches can be used by the digital signal processing block 305 to derive
estimates of the residual timing error in the filtered signal emerging from process 306.
For example, in some digital communications systems the transmitter embeds a
sequence of known symbols in the block of information constituting a burst. This is
the case in the E-GPRS system, where a sequence of 26 symbols collectively referred
to as a training sequence and which is known to the receiver is inserted in the middle
of each information burst. This training sequence can be used by the digital signal
processing block 305 in a known manner to estimate how far the timing of the
received burst is from the ideal value and thereby produce a timing error estimate for
each burst.
The fractional delay filter can be implemented using either an Infinite Impulse
Response (TJR) or- Finite Impulse Response (FIR) structure. Filtering techniques
based in the frequency domain could also be used for the implementation.
The timing resolution of the delay introduced by the fractional delay process 306
allows complexity to be traded-off against performance. For a given timing
correction, it is possible to calculate adaptively the required configuration for the
fractional delay filter that needs to be implemented by process 306. Hence, the
configuration of the fractional delay filter could be calculated for each new burst
using a timing correction established on the basis of timing errors measured for earlier
bursts. Such an approach should lead to good performance in terms reducing the

residual timing error at the input to the demodulation process 307. However, the
implementation of this solution could require the adaptive derivation of a new
fractional delay filter for each burst received at the digital signal processing block 305
and hence could prove too complex. The implementation complexity of the fractional
delay process 306 can be reduced if the resolution of the timing corrections that are to
be applied is reduced. Reducing the resolution of the timing corrections that are to be
applied by the fractional delay process 306 limits the number of possible corrections,
If the number of possible correction values is low enough, it is possible to pre-
calculate and store a configuration of the fractional delay filter for each of the possible
correction values. In such a case, it is not necessary to calculate a new configuration
for each incoming burst, rather the fractional delay process 306 only needs to engage
the stored fractional delay filter configuration with the timing correction value that is
closest to the desired one, One consequence of using such an approach is that even
after the fractional delay correction, the timing of the received signal will not be
perfect. Hence, this could slightly degrade the performance of the demodulation
process. However, this performance degradation can be kept to a minimum by
carefully selecting the timing correction resolution of the stored filter configurations.
Figure 4 illustrates how the timing recovery loop of Figure 3 can be adapted to cope
with ■multi-slot reception in an E-GPRS receiver. Figure 4 illustrates a processing
scheme for the reception of two slots but it is possible to extend it to any desired
number of slots. Elements 401 to 404 and 4 14 correspond, respectively, to elements
301 to 304 and 309 of Figure 3 and their fimctions will not be described again in
detail. Two versions 405 and 408 of digital signal processing block 305 are included,
each for handling a respective one of the received slots. Each of blocks 405 and 408
includes a respective fractional delay process 406 and 409 for adjusting the sample
timing in the manner of process 306 and a respective demodulation process 407 and
410 for producing estimates of the transmitted information bits from the respective
slots: The fractional delay processes 406 and 409 implement delays that can be
adapted on the basis of timing errors deduced by their respective digital signal
processing blocks 405 and 408. The timing errors are filtered by respective averaging
filters 411 and 412 prior to their use in adapting their respective fractional delay
processes 406 and 409. It is important to note that the two signal processing blocks

405 and 408 do not necessarily correspond to two separate physical entities as they
could be software processes running in time-shared fashion on a single processor or
other hardware block which is time-shared between the different time slots hardware
element. Although the slots that are to be demodulated by blocks 405 and 407 are not
received contemporaneously, the processing performed by blocks 405 and 407 may
overlap in time to some (greater or lesser) degree.
Each of digital signal processing blocks 405 and 408 provides an estimate of the
residual timing error in its corresponding time slot. The separate timing errors from
blocks 405 and 408 are then brought together in a combining process 413 (typically
by averaging) to generate a single residual timing error, This combined timing error
is then subjected to averaging over a number of bursts by filtering process 414 and the
filtered result' 'is used to adapt the digital sample acquisition timing that is used by
ADC process 402. Thus, the adaptation of ADC sample acquisition timing is done
according to the timing error values from all of the time slots involved in the multi-
slot transmission. Hence, the timing of the mixed-signal processing unit 401 is
updated such that the average timing error across the different time slots involved in
the multi-slot transmission is driven to zero. However, the correct timing for each of
these individual time slots individually is normally different from this average value.
This is why the timing corrections made by the fractional delay unit 406 and 409 are
derived from the timing error estimate from the time slot to which the correction will
be applied. Hence, the correction values of the fractional delay processes 406 and 409
can differ from one another. This means that even though the decimation process 404
in the mixed signal processing unit 401 will generate samples with a fixed period, the
distance in time between symbols corresponding to different time slots can be
adjusted with a resolution which is only limited by the resolution of the corrections
made by the fractional delay processes 406 and 409.
In a preferred implementation, three pre-calculated fractional delay filter
configurations are stored. Those different configurations correspond to timing
corrections of lA} Vz and 3A of the modulation symbol period. This means that each
fractional delay process 406, 409 can use those configurations to make corrections
with a resolution equal to ± lh of a modulation symbol. This time resolution

provides a good trade-off between implementation complexity and demodulation
performance. Timing corrections of integer values of the symbol duration can easily
be made by simply changing the symbol position within the burst that denotes the
start of the part of the burst that has to be demodulated.
As explained earlier, in the E-GPRS system, the transmitter formats information in
slots of symbols. Eight such slots are then grouped together to form a TDMA frame
with duration equal to 4.165 ms. Each slot is normally specified to correspond to
156'/4 modulation symbols. However, two different options have been defined in the
standard ('3GPP TS 45.010, Technical Specification 3rd Generation Partnership
Project; Technical Specification Group GSM/EDGE Radio Access Network; Radio
subsystem synchronisation') as to how base-stations can group together the different
slots in a singie TDMA frame. Those different formats are illustrated in Figure 5 for
two types of base stations. It can be seen that in both cases, the number of modulation
symbols in a TDMA frame is always equal to 1250. However, the number of symbols
across two consecutive slots is not identical for the two formats. In one of the two
formats, each slot is always 156% symbols long. In the other format, the length of
consecutive slots alternates between 156 and 157 symbols. This means that the
timing corrections to be made by the receiver will need to be different depending on
which of these formats is used in a received transmission. The problem is
exacerbated in that a receiver will not be informed .of the timing format of the
consecutive incoming bursts and hence the receiver will need to adapt to the selected
format in a blind fashion. The scheme proposed in Figure 4 achieves this goal by
keeping the calculation of the timing corrections being made by the fractional delay
processes 406 and 409 separate for the different slots involved in a multi-slot
transmission. Consequently, the corrections made by the fractional delay processes
406 and 409 can be different for the different slots involved in a multi-slot
transmission and will automatically match the transmission format.

WE CLAIM:
1. A telecommunications network participant, comprising means (302; 402) for
digitizing, as a series of samples, a received signal containing a succession of symbols
arranged in time slots of a repeating time frame, means (305; 405,407) for measuring
time misalignment between the symbols and the samples and means (306; 406,409)
for applying a fractional delay to the positions of the samples to reduce the
misalignment, characterized in that, for each of a plurality of said time slots, the
measuring means is arranged to deduce a timing error for the respective slot and the
delay applying means is arranged to apply a fractional delay to suppress the timing
error of the respective slot.
2. A telecommunications network participant as claimed in claim 1, wherein the
digitizing means is arranged to adjust the timing of the digitization of the samples to
suppress the misalignment.
3. A telecommunications network participant as claimed in claim 1 or 2, further
comprising means for shifting the samples in time by one or more integer symbol
positions to suppress the misalignment.
4. A telecommunications network participant as claimed in claim 1, 2 or 3,
wherein the received signal has a format such that the symbols are arranged in bursts,
the measuring means is arranged to deduce for each of one or more bursts a respective
timing error and the delay applying means is arranged to apply a fractional delay to
suppress the timing error or errors.
5. A telecommunications network participant as claimed in claim 1, 2 or 3,
wherein the received signal has a format such that the symbols are arranged in bursts,
the measuring means is arranged to deduce a timing error for each of several bursts

and to combine the errors to produce a resultant error and the delay applying means is
arranged to apply a fractional delay to suppress the resultant error.
6. A telecommunications network participant as claimed in claim 4 or 5, wherein
the received signal has a format in which the bursts are grouped into a repeated time
frame, the time frame containing a number of time slots, each time slot containing a
burst and, for each of a plurality of said time slots, the measuring means is arranged to
deduce a timing error for each of one or more burst in the respective slot and the
delay applying means is arranged to apply a fractional delay to suppress the timing
error or errors of the respective slots.
7. A telecommunications network participant as claimed in claim 4 or 5, wherein
the received signal has a format in which the bursts are grouped into a repeated time
frame, the time frame containing a number of time slots, each time slot containing a
burst and, for each of a plurality of said time slots, the measuring means is arranged to
deduce a timing error for each of several bursts in the respective slot and to combine
the errors to produce a resultant error for the respective slot and the delay applying
means is arranged to apply a fractional delay to suppress the resultant error of the
respective slot.
8. A telecommunications network participant as claimed in claim 6 or 7, wherein
the participant further comprises means (413) for combining timing errors from
different time slots to produce a sampling timing error and the digitizing means is
arranged to adjust the timing of the digitization of the samples to suppress the
sampling timing error.
9. A telecommunications network participant as claimed in any one of claims 6 to
7, wherein the measuring means deduces a timing error for a burst by calculating the
position of a known training sequence in the burst and measuring the time offset

between the said position and an ideal position in the burst of the said training
sequence.
10. A telecommunications' network participant as claimed in any one of claims 1 to
9, wherein the delay applying means is arranged to apply a desired fractional delay to
samples by appropriately reconfiguring a fractional delay filter providing the delay.
11. A telecommunications network participant as claimed in claim 10, wherein the
delay applying means includes several pre-stored configurations of the filter and the
delay applying means is arranged to reconfigure the filter by selecting for the filter the
configurations that most closely matches the desired fractional delay.
12. A method of digitizing a received telecommunication signal, the method
comprising digitizing, as a series of samples, a received signal containing a succession
of symbols arranged in time slots of a repeating time frame, measuring time
misalignment between the symbols and the samples and applying a fractional delay to
the positions of the samples to reduce the misalignment, characterized in that, for each
of a plurality of said time slots, the measuring step comprises deducing a timing error
for the respective slot and the delay applying step comprises applying a fractional
delay to suppress the deduced timing error of the respective slot.
13. A method as claimed in claim 12, wherein the digitizing step is arranged to
adjust the timing of the digitization of the samples to suppress the misalignment.
14. A method as claimed in claim 12 or 13, further comprising a translating step of
shifting the samples in time by one or more integer symbol positions to suppress the
misalignment.

15. A method as claimed in claim 12, 13 or 14, wherein the received signal has a
format such that symbols are arranged in bursts, the measuring step comprises
deducing for each of one or more burst a respective timing error and the delay
applying step comprises applying a fractional delay to suppress the timing error or
errors.
t
16. A method as claimed in claim 12,13 or 14, wherein the received signal has a
format such that symbols are arranged in bursts, the measuring step comprises
deducing a timing error for each of several bursts and combining the errors to produce
a resultant error and the delay applying step comprises applying a fractional delay to
suppress the resultant error.
17. A method as claimed in claim 15 or 16, wherein the received signal has a
format in which the burst are grooved into a repeated time frame, the time frame
containing a number of time slots, each time slot containing a burst and, for each of a
plurality of said time slots, the measuring step comprises deducing a timing error for
each of one or more bursts in the respective slot and the delay applying step comprises
applying a fractional delay to suppress a timing error or errors of the respective slot.
18. A method as claimed in claim 15 or 16, wherein the received signal has a
format in which the bursts are grooved into a repeated time frame, the time frame
containing a number of time slots, each time slot containing a burst and, for each of a
plurality of said time slots, the measuring step comprises deducing a timing error for
each of several bursts in the respective slot and combing the errors to produce a
resultant error for the respective slot and the delay applying step comprises applying a
fractional delay to suppress the resultant error of the respective slot.
19. A method as claimed in claim 17 or 18, wherein the method further comprises
combing timing errors from different time slots to produce a sampling timing error

and the digitizing step comprises adjusting the timing of the digitization of the
samples to suppress the sampling timing error.
20. A method as claimed in any one of claims 15 to 19, wherein the measuring step
deduces a timing error for a burst by calculating the position of a known training
sequence in the burst and measuring the time upset between said position and an ideal
position in the burst of said training sequence.
21. A method as claimed in any one of claims 12 to 20, wherein the delay applying
step comprises applying a desired fractional delay to samples by appropriately
*
reconfiguring a fractional delay filer providing the delay.
22. A method as claimed in claim 21, wherein the delay applying step comprises
reconfiguring the filter by selecting for the filter one of several pre-stored
configurations of the filter that most closely matches the desired fractional delay.



Abstract


A telecommunications network for achieving time synchronism between a
transmitter and a receiver in Time Division Multiple Access (TDMA)
communication systems and method of applying the same"
A telecommunications network participant,comprising means (302,402) for
digitising, as a series of samples, a received signal containing a suceession of
symbols arrange in time slots of a repeating time frame, means (305, 405, 407)
for measuring time misalignment between the symbols and the samples and
means (306, 406, 409) for applying a fractional delay to the position of the
oexamples to reduce the misalignment, characterised in that, for each of a plurality
of said time slots, the measuring means is arranged to deduce a timing error for
the respective slot and the delay applying means is arranged to apply a fractional
delay to suppress the timing error of the respective slot.

Documents:

00287-kolnp-2007-correspondence-1.1.pdf

00287-kolnp-2007-form-1-1.1.pdf

00287-kolnp-2007-form-26.pdf

00287-kolnp-2007-form-3-1.1.pdf

00287-kolnp-2007-form-5-1.1.pdf

00287-kolnp-2007-pct request.pdf

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Patent Number 263205
Indian Patent Application Number 287/KOLNP/2007
PG Journal Number 42/2014
Publication Date 17-Oct-2014
Grant Date 14-Oct-2014
Date of Filing 24-Jan-2007
Name of Patentee MSTAR SOFTWARE R&D (SHENZHEN) LTD.
Applicant Address 4F,BLOCK C, SHENZHEN TECH-INNOVATION INTERNATIONAL BLDING.,10TH ROAD OF KEJI, SOUTH AREA,HIGH-TECH INDUSTRIAL PARK,SHENZHEN P.R.CHINA-518057
Inventors:
# Inventor's Name Inventor's Address
1 FATEMI-GHOMI, NAVID 6,TENBY ROAD, FRIMELY,SURREY, GU16 8UT
2 VALDON, CYRIL 139, JACKMANS PLACE, LETCHWORTH, HERTFORDSHIRE SG6 1RG
PCT International Classification Number H04B7/26
PCT International Application Number PCT/GB2005/003111
PCT International Filing date 2005-08-05
PCT Conventions:
# PCT Application Number Date of Convention Priority Country
1 0418133.5 2004-08-13 U.K.