Upload
doanhanh
View
226
Download
1
Embed Size (px)
Citation preview
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
95
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
96
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
97
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
98
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.
99
Figure 7.1a Number of users with AMR– CC- Hard Decision
Figure 7.1b Number of users with AMR– CC- Hard Decision
0 50 100 150 200 2500
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
Number of Users
Dow
nlin
k L
oad F
acto
r
Voice- 12.2kbps
0 50 100 150 200 250 300 350 4000
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
Number of Users
Dow
nlin
k L
oad F
acto
r
Voice- 7.95kbps
100
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
0 100 200 300 400 500 600 7000
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
Number of Users
Dow
nlin
k L
oad F
acto
r
Voice- 4.75kbps
101
Figure 7.2a Number of users with AMR– CC- Soft Decision
Figure 7.2b Number of users with AMR– CC- Soft Decision
0 50 100 150 200 250 300 350 400 4500
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
Number of Users
Dow
nlin
k L
oad F
acto
r
Voice- 12.2kbps
0 100 200 300 400 500 600 7000
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
Number of Users
Dow
nlin
k L
oad F
acto
r
Voice- 7.95kbps
102
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
0 200 400 600 800 1000 12000
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
Number of Users
Dow
nlin
k L
oad F
acto
r
Voice- 4.75kbps
103
Figure 7.3 Blocked video users for CCHD with AMR –7.95kbps
Figure 7.4 Blocked video users for CCHD with AMR –4.75kbps
104
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
105
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.