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Capacity and Speech Quality aspects using Adaptive Multi-Rate (AMR)

Olivier Corbun, Magnus Almgren, Krister Svanbrot

Ericsson Radio Systems AB, S - 16480 Stockholm, Sweden
?Ericsson Erisoft AB, S-97128 Lule5, Sweden

olivier.corbun@? / magnus.almgren

AMR (Adaptive Multi-Rate) is an emerging speech codec cellular standard in ETSI [I]. This standard should be ready during
1998 as a Speech GSM evohtion. It is a new concept for achieving a high speech quality maintaining an efficient spectrum
usage. According to the channel quality and the traffic load, the
radio resource algorithm allocates a half-rate or a full-rate channel in order to obtain the best balance between quality and capacity. Within this channel, the codec is quickly adapted to track
changes in the radio link. An AMR system model has been developed to show the impact on speech quality by varying the capacity from only full-rate channels to only half-rate channels. The
aim is also to show the gain provided by an AMR system compared with an existing GSM system using second generation EFR
(Enhanced Full Rate) and HR (Half Rate) coders. The results
show that there is a trade-off between capacity increase and
speech quality degradation. It is also very clear that there is a
potential gain in quality by using AMR compared to existing
speech codecs in GSM systems.

GSM is a worldwide success with more than 100 million wireless
subscribers. This number will probably grow to 600 million by
the beginning of the next millennium. Speech will remain one of
the most important services. The challenge for the operator is to
enhance the speech quality and at the same time to optimize spectrum efficiency particularly j n large urban areas where subscriber
density is very high.
This can be obtained using AMR (Adaptive Multi-Rate) as it has
been shown during the study phase at ETSI [l]. According to
their plan, the first release of AMR specifications should be ready
in february 1999.

power control, has been used. The general system behavior when
an AMR coder is used to switch between and within Half-Rate
(HR) and Full-Rate (FR) traffic channels is assessed. Then, the
speech quality obtained using this AMR system is compared with
an existing system using HR (GSM-HR codec) and FR (GSMEFR codec) channels for an equivalent capacity.


The AMR (Adaptive Multi-Rate) codec allows channel mode
(HR or FR) and codec mode (combination of speech and channel
bit-rates) to vary in order to suit traffic and channel conditions.
The channel mode can be switched in order to increase channel
capacity, replacing for example one full-rate channel with two
half-rate channels, while maintaining a certain lower limit for the
speech quality. These AMR handovers occur much less frequently
than the codec mode changes, probably a few times per minutes.
A representation is drawn to the left in Figure 1. If the speech
quality in HR is acceptable for a given CL,it can be interesting to
trigger an AMR handover from FR to HR in order to gain capacity even if there is a loss in quality. Reciprocally, if the quality
becomes too poor and the capacity is not critical, an AMR handover from HR to FR has to be done.
For each channel mode (HR or FR), the codec mode, i.e. bit partitioning between speech and channel bit-rates, can be varied rapidly to track the channel error rates or the channels CA. This
variation is represented to the right in Figure 1. By decreasing
this coding bit-rate, i.e. switching from codec 3 to 2 or from
codec 2 to 1, the robustness is increased under poor conditions.
The changes must occur quite immediately (several times a second), with no perceptible speech degradation. This process is
equivalent to Link Adaptation [4][5][6].


A half-rate or full-rate channel is allocated according to the need

of capacity and quality and the current situation in each cell.
Then, at a fixed rate (HR OI-FR), the AMR system exploits the
compromise between robustness to errors and clear channel quality by adapting the speech aind channel coding rates according to
the radio conditions.
The aim of this study is to show the relative capacity increase and
the trade-off between quality and capacity which may be
obtained by using AMR. A radio network simulator modeling the
performance of a GSM system, including frequency hopping and

0-7803-4872-9/98/$10.00 0 1998 IEEE






for one channel mode


Figure 1. Channel modes and Codec modes

Due to the flexibility of the system, AMR can be applied in many

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ways. Three potential scenarios are:

- Full-rate only: gives a good robustness to channel errors.
However, there is no capacity gain, except perhaps with a tighter
frequency reuse.
- Half-rate only: gives a maximum capacity. The codec mode
adaptation to channel conditions can provide significant quality
- mixed Full- and Half-rate: allows a trade-off between quality
and capacity enhancements according to radio and traffic conditions.
The algorithm representing the AMR channel and codec modes
selection is drawn in Figure 2.
about every 10 sec.



CA [de]

Figure 3. Mapping from C/I to Speech Quality

every 0.1 sec

AMR handover


speech & channeI

from a speech quality perspective.

There is already a potential quality improvement using AMR
compared to EFR/HR. As it can be seen in Figure 3, AMR-HR
gives a better speech quality than GSM-HR under good radio
conditions. On the other hand, AMR-FR is more robust than
GSM-EFR under poor radio conditions. This difference ip quality
should be increased by the intelligent switching used in AMR
for selecting the relevant channel mode (HR or FR).


Figure 2. AMR codec algorithm






In order to examine the performance with and without AMR, it is

necessary to have some mapping from channel quality to speech
quality for the different codecs in use. A subjective performance
measure, based on an informal Mean Opinion Score (MOS) of a
few users, has been used. The performance for each codec mode
in the HR and FR channels has been derived from subjective
speech quality assessments. The test was performed using ideal
frequency hopping on a typical urban channel profile, at low
speed. The speech quality has been transformed from the MOS
domain to the equivalent dBQ domain.
Figure 3 shows the performance for the six different codec modes
(three for each channel mode) and the two envelopes used for
AMR. For each traffic channel mode (HR or FR), three codec
modes bit-rates are used. AMR-HR and AMR-FR, which are the
envelopes of three codecs each, are used to get a speech quality.
The performance is only indicating quality for speech, i.e. the
performance under different background noise conditions is not
included. That means that the general results must be interpreted
carefully. Since the transformation from CO to speech quality
does not include dynamic effects, it is of course a simplification

AMR system model


Figure 4.

AMR System Model

In this study, the AMR handover decision, i.e. the choice between
HR and FR, is based on C/I. For each of the channel modes, the


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codec mode adaptation is supposed to be ideal.

A dynamic radio network model is used, giving a Carrier to Interference ratio (C/Isim)value for each connection each half second.
It is not important to have a smaller time interval since AMR handovers can occur just a few times per minute. The sequence of C/
I values for each call is fed into the AMR codecs, as shown in
Figure 4, and transformed into speech quality (grey box) using
the mapping in Figure 3. The simulated Carrier to Interference
values (C/Isim)for each call is subject to a log-normal estimation
error (COer) with a standard deviation of 3 dB. The result is then
filtered. The selection of AMR codec is done using this estimation of the channel quality (C/IeSt).Basically, for a C/Iest value
under a threshold the full-rate AMR codec will be chosen. Otherwise, the half-rate codec i:; used. The decision is delayed by one
time step (half a second) before it is applied.
The position of the threshold which is used in the actual channel
mode selection is calculated according to the desired capacity, i.e.
the proportion of HR channels. Looking at all the COest resulting
from the simulation, the tbreshold is placed to get the wanted
proportion of HR channels as it is illustrated in Figure 5. To limit

with 27 equidistant 3-cell sites. Simple Frequency hopping and

Power control are added. To avoid border effects which would
prevent the use of data from border cells, a wrap-around technique is used. The geographical traffic distribution is uniform
throughout the area and calls are generated according to a Poisson process. The Okumura-Hata model is used to calculate the
path loss between mobile stations and base stations. Two log-normal fading components are added to the path loss, one with long
spatial correlation distance (110 m) modeling terrain variations
and one with short correlation distance (20 m) modeling buildings. The total standard deviation of the shadow fading is 10 dB.
For both components fading values are determined with a lognormal map, ensuring that the radio environment in any single
point remains constant, and that it changes continuously in space.
The fading values between a mobile and different base stations is
uncorrelated which is representative of an urban environment. A
total of 36 frequencies (4 per cell, 3/9 frequency reuse) are simulated in the system (32 time-slots per cell). Cell selection at call
set-up and handovers are based on highest path gain.

Only the speech quality for uplink is shown in the following
chapters. However, the results obtained on the downlink are very
similar. Comparable results can be found in [3] where the mean
speech quality versus the number of user per cell is plotted for
gross rate link adaptation.




Figure 5. C:hannelmode decision

the number of AMR handover, a hysteresis around the threshold
is used. There is no AMR handover in the safety margin, i.e. two
times the hysteresis. The C A value has to pass through this margin to provoke an AMR handover.

Capacity/Quality aspects using AMR

The aim is to display the trade-off between the capacity (Erlang)

and the Speech Quality (dBQ) resulting from an AMR usage. In
order to obtain some figures representing this relation, the speech
quality has been evaluated for different capacities in the system.
The relative capacity, i.e. capacity compared with the one
obtained by using only FR channels, is shown in the following

B. Traflc description
The general radio network simulator models the performance of a
basic GSM system with a rlelatively large number of traffic channels. The amount of offered traffic is adjusted depending on three
parameters: the number of servers in the system (32 FR servers in
our system), the desired proportion of half-rate channels
(depends on the wanted capacity) and the blocking rate (2%
used). The traffic load is set to get all the different capacities in
the system from all the channels in FR mode to all the channels in
HR mode. A certain capacity corresponds to a certain proportion
of HR/FR channels. So, by moving the channel mode switching
threshold, different proportion HR/FR channels, i.e. capacities,
can be obtained. The different capacities are comparable by keeping the blocking rate constant in all the simulations. More traffic
considerations can be found in [2].

C. Parameters set-up
A Radio Network simulation tool is used to simulate a system






Relallve capacity

Figure 6. Up-Link mean speech quality vs.

relative capacity using AMR


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In Figure 6, the average uplink speech quality (over all the speech
quality reports from the different calls resulting from each simulation) is plotted versus the relative capacity. As a reference the
average quality of the AMR coder only operating on the FR
channel achieves a quality slightly under 27 dBQ. As the relative
capacity increases, the average speech quality is monotonously
decreased. At a 50% capacity increase, the average quality is
decreased by approximately 3 dBQ compared to only using AMR
in the FR channel. The average speech quality of the AMR coder
operating on the HR channel is under 22 dBQ.
In Figure 7, speech quality pdf at different percentiles of the
speech quality distribution in the system are plotted as a function
of relative capacity. The percentiles can (for simplicity) be interpreted as the proportion of users experiencing the corresponding
quality or worse.

' . , . ,. .......~


. . . ... . .

. . . . . , ., .. , . . ...,

. .,. . . ..


. . . .:.







more. At a capacity increase of 50%, the AMR speech quality is a

bit more than 24 dBQ and the GSM speech quality is 21 dBQ.
By introducing HR channels and increasing the capacity, the
speech quality using AMR is below the quality resulting from the
use of only EFR (26 dBQ). The gain can be seen in this case only
as a capacity gain but not as a speech quality increase. AMR
seems to provide a good improvement if HR channels are introduced in the system.
C. Handoff rate impact


. ...

. . ..." ................... .... . .... , ..... . ....... . . ... . , . . .. , .., ........ , , , ... .. ,.. ...,.. ..... . .

Relallve capadty

Figure 8. Average Speech Quality for an existing system

and a n AMR concept

..:.......... .......:...._.............;....,...........




Figure 7. Percentile Speech quality vs. relative capacity

At a capacity increase of 25%, the 10% and 30% levels are
affected with 2 dBQ decrease and the 3% level is less affected
with a decrease of 1 dBQ. This indicates that the users with high
speech quality experience higher degradation at moderate capacity increases. At a capacity increase of 50%, the 3%, 1096, and
30% levels are affected (3 to 4 dBQ). Beyond 80% capacity
increase, the quality at the 3% level is significantly affected, indicating that users with already poor quality experience a further
degradation, to a higher extent than users with rather good speech

The frequency of channel mode changes have been characterized

at different percentiles of the speech quality distribution. A hysteresis of 5 dB on the estimated C/I is applied to further limit the
number of AMR handovers. This hysteresis is large enough to get
a reasonable number of handovers (few per minute). As can be


B. AMR vs. existing EFWHR solution

In Figure 8, a comparison between an AMR codec, and a "regular" GSM system using the two codecs GSM-EFR and GSM-HR
is made. This two codecs are included in the AMR codecs (see
Figure 3).


As can be seen, there is a quite large increase in quality by using

AMR. For a relative capacity of one, the difference is half a dBQ.
This is due to the small difference in quality comparing AMR-FR
and GSM-FR (see Figure 3) for the C/I values we get from the
simulation. Then, the difference quickly reaches 3 dBQ and



Relative capaclty

Figure 9. Speech quality vs. capacity

for different hysteresis


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seen in Figure 9 the quality on the 3% and 10% levels is affected

by the handover constrairits in the range 25% to 75% capacity
increase. The gain by using extremely rapid handover may
however be compromised by the decrease in quality due to signalling and handover interruption. The gain on the average
speech quality has not been proved. This has a larger impact on
the users experiencing a ]poor quality like on the 3% level. A
restriction of the handover rate (by introducing a hysteresis, see
A) to a realistic value of 3 HO/call/min in average, at a capacity
increase of 25%, the speech quality performance decreases
slightly, but should in any case be satisfactory.

impact on the quality than the handover degradation itself at high

handover rates.

The authors wish to thank Erik Ekudden for his support and the
presentation of these results he made at ETSI and Anders
Uvliden for his active participation in this work.

ETSI TC SMG, AMR Study Phase Report verl.O,
Budapest, October, 1997, Meeting no. 23, TD -197.

E Khan, D.Zeghlache Performance Analysis of Link

Adaptation in Wireless Personal Communication Systems, IEEE ICC97, Montreal 1997
J. Irvine, P. Cosimini, J. Dunlop Implementation considerations for gross rate link adaptation, IEEE VTC,
Atlanta 1996, pp 1766-1770
J. Dunlop, J. Irvine, P. Cosimini Estimation of the Performance of Link Adaptation in Mobile Radio, IEEE
Vehicular Technology Conference, VTC95, Chicago




1 75

J. Dunlop, P. Cosimini, G. Graham, E. Le Strat Estimation of the performance of an adaptive air interface
in mobile radio, Proc. RACE Mobile Telecommunications Workshop, Amsterdam, 17-19 May 1994, pp. 4751

2 25

E. Yuen, P. Ho, V. Cuperman Variable Rate Speech

and Channel Coding for Mobile Communication,
IEEE Vehicular Technology Conference, VTC94,
Stockholm 1994

Relalive c a ~ c l N

Figure 10. Intracell handover rate (#/call/min)

The number of AMR handovers performed by the AMR codec is
shown in Figure 10. The capacity is increasing by moving the
threshold from the right to the left (Figure 5). The top of the
curve with no handover limitation corresponds to a high density
around the threshold, i.e. a lot of CA values. By having an hysteresis around the threshold of 5 dB, we considerably limit the number: of switches between HR and FR. The maximum value is 3
HO/call/min which is realistic.

T.B. Minde, S. Bruhn. E. Ekudden, H. Hermansson

Requirements on speech coders imposed by speech
service solutions in cellular systems, IEEE Workshop,
Pocono Manor, Pennsylvania, USA, 7-10 September

The AMR system capacity increase has been evaluated in a simulation model. As expected, a1 trade-off between capacity increase
and quality degradation is possible. For moderate capacity
increases the reduction of quality is noticed mostly by users in
high quality situations. The average quality in the system is
slightly lowered. The quality for users in low quality areas is significantly reduced for capacity increase exceeding approximately
In general, in a situation where both channel modes (half-rate and
full-rate) are used, there is potentially a large gain in quality with
AMR, compared to what can be achieved by allocating existing
GSM-EFR and GSM-HR cotlecs.
The AMR handover rates have to be limited to a reasonable number. The simulations show irhat the impact on quality is quite
small by limiting this rate. It is likely that this reduction has less

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