CHAPTER 7

ROLE OF ADAPTIVE MULTIRATE ON WCDMA CAPACITY

ENHANCEMENT

7.1 INTRODUCTION

Originally developed to be used in GSM by the Europe Telecommunications

Standards Institute (ETSI), the AMR speech codec [TS 26.071] was approved within

the 3GPP forum in 1999 to be mandatory for circuit- and packet-switched speech in

UMTS networks. An AMR speech codec adapts the error protection level to the

local radio channel and traffic conditions so that it always selects the optimum

channel and codec mode to deliver the best combination of speech quality and

system capacity. AMR uses Multi-Rate Algebraic Code Excited Linear Prediction

(MR-ACELP) scheme based on two different synthesis filters. It converts a

narrowband speech signal (from 300 to 3,400Hz) to 13-bit uniform Pulse Coded

Modulated (PCM) samples with 8kHz sample rate. This leads to 20 ms AMR frames

consisting of 160 encoded speech samples. This means that the codec can switch

mode, i.e. source bite rate, every 20ms. AMR has 8 coded modes in UMTS systems,

whereas in GSM AMR uses either 6 or 8 modes. The eight source rates vary from

4.75 to 12.2kbps. It also contains a low rate encoding mode, called SIlence

Descriptor (SID), which operates at 1.8kbps to produce background noise and a non-

transmission mode. The AMR codec dynamically adapts its error protection level to

the channel error conditions. For instance, lower speech coding bit rate and more

error protection schemes are used in bad channel conditions.

This principle where AMR strives to change to the best curve associated to a

given AMR mode. It has been shown that the degradation on the audio quality

caused by a lower speech coding rate is compensated by increased robustness with

the channel coding. Note, however, that this channel robustness is more beneficial in

GSM than in UMTS due to the embedded fast power control used in WCDMA

systems [71]. Using a variable-rate transmission scheme also makes it possible to

control the transmission power of the UE, a fact that is particularly useful when the

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UE suddenly attains its maximum transmits power in CDMA, lower bit rates

generally need lower transmit power and vice versa [72].

7.2 AMR FRAME STRUCTURE AND OPERATING MODES

The generic structure of the AMR frame is divided into a header, auxiliary

information and core frame. The header contains the Frame Types and Frame

Quality Indicator fields. The Frame Type can indicate the use of one of the eight

AMR codec modes for that frame, a noise frame, or an empty frame. The Frame

Quality Indicator indicates if the frame is good or bad. The auxiliary information

part includes the Mode Indication, Mode Request and Codec CRC fields. The CRC

field is used for the purpose of error detection calculated over all the Class A bits in

the AMR Core frame. The Core frame part is used to carry the encoded bits divided

into A, B and C classes. In case of a comfort noise frame, comfort noise parameters,

i.e. a SID frame, replace “class A” bits of the core frame while “class B” and “class

C” bits are omitted.

7.2.1 Classification of the encoded bits according to their sensitivity to errors

AMR encoded bits are divided into three indicative classes according to their

importance: A, B and C. The reason for dividing the speech bits into classes is that

they can be subjected to different error protection in the network. Class A contains

the bits that are most sensitive to errors and any kind of errors in these bits typically

result in a corrupted speech frame which should not be decoded without applying

appropriate error concealment. This class is protected by the CRC in auxiliary

information field. Classes B and C contain bits where increasing error rates

gradually reduce the speech quality, but the decoding of an erroneous speech frame

is usually possible without a strongly perceptible quality degradation.

7.2.2 AMR operating modes

The AMR can operate in 8 different modes [73] (source bit rates). It should

be noted that some of these modes are equivalent to the speech codecs currently

used in other mobile communication systems. For instance, the “AMR 12.20kbps”

mode is equal to the ETSI GSM called codec EFR (Enhanced Full Rate Speech [TS

06.60]). Similarly, the “AMR 7.40kbps” mode is equivalent to the IS-641 codec

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used in the USA standard IS-136 (US TDMA). Finally, “AMR 6.7kbps” mode is

equivalent to the codec used in the PDC Japanese standard. Based on the fact that

voice activity in a normal conversation is about 40%, all AMR modes implement a

Voice Activity Detection (VAD) algorithm that detects if each 20 ms-frame contains

speech or not on the transmitting side. VAD works together with the DTX or Source

Controlled Rate (SCR) [TS 26.093] techniques where RF transmission is cut during

speech pauses. When the transmission is cut, “comfort noise” parameters are sent at

a regular rate in AMR frames during discontinuous activity. These frames are

known as Silence Descriptor (SID) frames. The receiver decodes these parameters

and generates locally a “comfort noise”. Without this background noise the

participants in a conversation, might think that their connection is broken during

silence periods. The SCR technique for AMR in UMTS is mandatory and aims at

prolonging the battery life (UE side) and reducing the interference.

7.3 DYNAMIC AMR MODE ADAPTATION

The AMR mode adaptation in UMTS networks means using different AMR

coding for the data stream. Mode adaptation can independently be applied in the

uplink and the downlink. At any point in time, a different AMR mode can be used in

each direction and this can be dynamically changed during a voice conversation.

7.3.1 Location of the AMR speech codec in UMTS networks

The AMR speech codec is located in the Transcoder (TC) function defined to

be in the UMTS core network and as such, logically controlled by Non-Access

Stratum protocols. From the transfer point of view, this means that all AMR coded

data is going to be transmitted not only via Iub and air interface but also via Iu-

interfaces. Note, however, that the AMR mode control that generates the AMR

mode command cannot be located in the TC, since this control entity needs

information from the air interface to make a decision about valid AMR modes – the

AMR mode command is used to change the current AMR mode to the new one. The

only element in the network which can provide this type of information is the

UTRAN. Note that in GSM networks the control of the codec mode is provided by

the BTS. This solution is not applicable in UTRA due to the soft-handover

procedure defined for dedicated traffic channels. Therefore, the AMR mode control

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function is part of the RNC, and more precisely a part of layer 3 functionality.

Within the radio interface, the rate on the speech connection is either decreased or

increased depending on the new valid AMR mode by changing the valid Transport

Format (TF) in the corresponding MAC-d entity.

7.3.2 AMR mode adaptation in the downlink

The RNC generates the AMR mode adaptation command based on existing

radio conditions in the downlink as reported by the UE from radio quality

measurements and from traffic volume measurements. The command is sent to the

encoder inside the TC via the Iu interface.

7.4 RESOURCE ALLOCATION FOR AN AMR SPEECH CONNECTION

An AMR speech connection can be initiated either by the UE or by the

network. When the UE requests resources from the network, a first negotiation is

made based on NAS procedures in order to configure the call connection. The CN

will determine the QoS, needed which will be then indicated to the UTRAN inside

the RANAP RAB ASSIGNMENT REQUEST message. Based on this request, RNC

can define the requested.

RAB and associated Radio Bearer(s) (RB). Depending on whether the

requested AMR base speech connection supports the concept of Unequal Error

Protection (UEP) or Equal Error Protection (EEP), the RNC assigns either one or

three RBs (including one or three DCHs), respectively, for the user plane. In the

control plane, RRC may allocate one or none signalling radio bearer according to the

alternative method used to change the AMR mode.

The speech codec in UMTS will employ the AMR technique. These services

are initially provided through the circuit switched core network in WCDMA, but

they can later be provided also through the packet switched core network. The

multirate speech coder is a single integrated speech codec with eight source rates

such as 12.2 (GSM-EFR), 10.2, 7.95, 7.40 (IS-641), 6.70 (PDC-EFR), 5.90, 5.15

and 4.75kbps. The AMR bit rates can be controlled by the radio access network. To

facilitate interoperability with existing cellular networks, some of the modes are

same as in existing cellular networks. The 12.2kbps AMR speech codec is equal to

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the GSM EFR codec, 7.4kbps is equal to the US-TDMA speech codec, and 6.7kbps

is equal to the Japanese PDC codec. The AMR speech codec is capable of switching

its bit rate every 20 ms speech frame upon command.

The bit rate of the AMR speech connection can be controlled by the radio

access network depending on the air interface loading and the quality of speech

connections [74]. During busy hours, it is possible to use lower AMR bit rates to

offer higher capacity while providing slightly lower speech quality. Also, if the

mobile is running out of the cell coverage area and using its maximum transmission

power, a lower AMR bit rate can be used to extend the cell coverage area. With the

AMR speech codec it is possible to achieve a trade-off between the network

capacities, coverage and speech quality according to the operator’s requirements.

Table 7.1 Capacity Calculation parameters with AMR

Parameters Values

Chip rate (W) 3.84Mcps

Voice Bit rate (R) 12.2,7.95,4.75kbps

Video Bit rate (R) 64 kbps

Voice Activity factor(υj ) 0.58

Total interference (i) 0.55

Orthogonality factor(α) 0.9

Voice Bit energy to noise density ratio (Eb/No) 5dB,2.7dB

Video Bit energy to noise density ratio (Eb/No) 6.5dB,4.1dB

7.5 RESULTS AND DISCUSSION

The objective of this simulation is to analyze the utility based CAC for

different services in WCDMA network with AMR for different decision decoders.

The simulation model is based on downlink load factor 0.7 with 70% voice users

and 30% video users. The Table 7.1 shows the Bit Error Rate to the Bit energy to

noise density ratio (Eb/No) of QPSK modulation of WCDMA network for

convolution coding with soft and hard decision scheme. To maintain QoS for voice

service the BER is 10-3

and the corresponding Eb/No values are 5dB and 2.7dB for

convolutional code hard and soft decision scheme respectively. Similarly for video

service the BER is 10-5

and the corresponding Eb/No values are 6.5dB and 4.1dB for

convolution code hard and soft decision scheme respectively.

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Figure 7.1a Number of users with AMR– CC- Hard Decision

Figure 7.1b Number of users with AMR– CC- Hard Decision

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Figure 7.1c Number of users with AMR– CC- Hard Decision

Figure 7.1a, b, c shows the capacity of WCDMA network for voice service

with AMR of 12.2kbps, 7.95kbps, and 4.75kbps respectively for convolutional code

hard decision (CCHD) decoder scheme. The number of voice users for load factor

0.7 of different AMR data rates and offered load for network capacity is given in

Table 7.2.

Table 7.2 Number of voice users and offered with AMR– CCHD

Data Rate Number of Users Offered load

12.2kbps 185 7.77 Erlangs

7.95kbps 284 11.88 Erlangs

4.75kbps 474 19.91 Erlangs

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Figure 7.2a Number of users with AMR– CC- Soft Decision

Figure 7.2b Number of users with AMR– CC- Soft Decision

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Figure 7.2c Number of users with AMR– CC- Soft Decision

Figure 7.2a, b, c shows the capacity of WCDMA network for voice service

with AMR of 12.2kbps, 7.95kbps, and 4.75kbps respectively for convolutional code

soft decision (CCSD) decoder scheme. The number of voice users for load factor 0.7

of different AMR data rates and offered load for network capacity is given in

Table7.3.

Table 7.3 Number of voice users and offered load with AMR– CCSD

Data Rate Number of Users Offered load

12.2kbps 314 13.17 Erlangs

7.95kbps 482 20.21 Erlangs

4.75kbps 806 33.76 Erlangs

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Figure 7.3 Blocked video users for CCHD with AMR –7.95kbps

Figure 7.4 Blocked video users for CCHD with AMR –4.75kbps

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Figure 7.3 and 7.4 explains the percentage of blocked users for the service

utility combination of 30% video users and 70% voice users. The load factor for

video user is 0.21 and for voice user is 0.49. At particular instant if the number of

user is 100, the percentage of blocked video user is 86% without the utility function

and 77% with utility function. The utility based CAC scheme reduces 9% of blocked

video users by means providing resources from the unutilized voice service.

Figure.7.5 and 7.6 explains the percentage of blocked users for the service

utility combination of 30% video users and 70% voice users. The load factor for

video user is 0.21 and for voice user is 0.49. At particular instant if the number of

user is 100, the percentage of blocked video user is 75% without the utility function

and 60% with utility function. The utility based CAC scheme reduces 15% of

blocked video users by means providing resources from the unutilized voice service.

Figure 7.5 Blocked video users for CCSD with AMR –7.95kbps

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Figure 7.6 Blocked video users for CCSD with AMR –4.75kbps

7.6 SUMMARY

The performance of WCDMA network for different services such as

voice 4.75kbps, 7.95kbps and 12.2kbps and Video 64kbps are calculated for the

downlink load factor value of 0.7. The quality of service for voice and video users

are maintain by selecting appropriate value of BER for convolutional code with soft

and hard decision scheme. The number of users admitted is evaluated for

convolutional code with soft and hard decision scheme for the same load factor. The

network capacity enhanced with convolutional code with soft decision scheme as

well as the utility based CAC scheme.