Difference between revisions of "Examples of Communication Systems/Telecommunications Aspects of UMTS"

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==Improvements regarding speech coding ==
 
==Improvements regarding speech coding ==
 
<br>
 
<br>
In the chapter&nbsp; [[Examples_of_Communication_Systems/General_Description_of_GSM|"GSM"]]&nbsp; (''Global System for Mobile Communications'')&nbsp; of this book, several speech codecs have already been described in detail:
+
In the chapter&nbsp; [[Examples_of_Communication_Systems/General_Description_of_GSM|"Global System for Mobile Communications"]]&nbsp; $\rm (GSM)$&nbsp; of this book,&nbsp; several speech codecs have already been described in detail.
  
 
{{BlaueBox|TEXT=
 
{{BlaueBox|TEXT=
 
$\text{Reminder:}$&nbsp;   
 
$\text{Reminder:}$&nbsp;   
 
A speech codec is used to reduce the data rate of a digitized speech or music signal.  
 
A speech codec is used to reduce the data rate of a digitized speech or music signal.  
*In the process, redundancy and irrelevance are removed from the original signal.  
+
#In the process,&nbsp; redundancy and irrelevance are removed from the original signal.  
*The artificial word "codec" indicates that the same functional unit is used for both encoding and decoding.}}
+
#The artificial word&nbsp; "codec"&nbsp; indicates that the same functional unit is used for both,&nbsp; encoding and decoding.}}
  
  
Among others, the&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Adaptive_Multi_Rate_Codec|"Adaptive Multi-Rate Codec"]]&nbsp; (AMR) was introduced, which in the frequency range from&nbsp; $\text{300 Hz}$&nbsp; to&nbsp; $\text{3400 Hz}$&nbsp; dynamically switches between eight different modes (single codecs) of different data rate in the range of&nbsp; $\text{4. 75 kbit/s}$&nbsp; to&nbsp; $\text{12.2 kbit/s}$&nbsp; and is based on&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Algebraic_Code_Excited_Linear_Prediction|"ACELP"]]&nbsp; (''Algebraic Code Excited Linear Prediction'').
+
Among others,&nbsp; the&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Adaptive_Multi_Rate_Codec|"Adaptive Multi-Rate Codec"]]&nbsp; $\rm (AMR)$&nbsp; based on&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Algebraic_Code_Excited_Linear_Prediction|$\rm ACELP$]]&nbsp; $($"Algebraic Code Excited Linear Prediction"$)$&nbsp; was introduced,&nbsp;  
 +
*which in the frequency range from&nbsp; $\text{300 Hz}$&nbsp; to&nbsp; $\text{3400 Hz}$&nbsp;  
  
These AMR codecs are also supported in UMTS Release 99 and UMTS Release 4. Compared to the earlier speech codecs (''Full-Rate, Half-Rate'' and ''Enhanced Full-Rate Vocoder''), they allow.
+
*dynamically switches between eight different modes&nbsp; $($single codecs$)$&nbsp;
*independence from channel conditions and network load,
+
 
*the ability to adapt data rates to conditions,
+
*of different data rate in the range of&nbsp; $\text{4. 75 kbit/s}$&nbsp; to&nbsp; $\text{12.2 kbit/s}$.
*improved flexible error protection in the event of more severe radio interference, and
+
 
*thereby providing better overall voice quality.
+
 
 +
These codecs are also supported in UMTS Release 99 and Release 4.&nbsp; Compared to the earlier speech codecs&nbsp; $($Full Rate,&nbsp; Half Rate,&nbsp;Enhanced Full Rate Vocoder$)$,&nbsp; they allow
 +
#independence from channel conditions and network load,
 +
#the ability to adapt data rates to conditions,
 +
#improved flexible error protection in the event of more severe radio interference, and
 +
#thereby providing better overall voice quality.
 +
 
 +
 
 +
[[File:EN_Bei_T_4_3_S1_v2.png|right|frame|Composition of wideband AMR modes]]
 +
 
 +
In 2001,&nbsp; the&nbsp; "3rd Generation Partnership Project"&nbsp; $\text{(3gpp)}$&nbsp; and the&nbsp; "International Telecommuncation Union"&nbsp; &nbsp; $\text{(ITU)}$&nbsp; specified the new voice codec&nbsp; &raquo;'''Wideband AMR'''&laquo;&nbsp; for UMTS Release 5.&nbsp; This is a further development of AMR and offers
 +
 
 +
*an extended bandwidth from&nbsp; $\text{50 Hz}$&nbsp; to&nbsp; $\text{7 kHz}$&nbsp; <br>$($sampling frequency&nbsp; $\text{16 kHz})$,
 +
 
 +
*a total of nine modes between&nbsp; $\text{6.6 kbit/s}$&nbsp; and&nbsp; $\text{23.85 kbit/s}$&nbsp; <br>$($of which only five modes are used$)$,&nbsp; and
 +
 
 +
*improved voice quality and better (more natural) sound.
  
  
In 2001, the 3gpp Forum&nbsp; (''3rd Generation Partnership Project'')&nbsp; and the ''International Telecommuncation Union'''&nbsp; (ITU)&nbsp; specified the new voice codec&nbsp; '''Wideband-AMR''''&nbsp; for UMTS Release 5. This is a further development of AMR and offers
 
[[File:EN_Bei_T_4_3_S1_v2.png|right|frame|composition of wideband AMR modes]]
 
*an extended bandwidth from&nbsp; $\text{50 Hz}$&nbsp; to&nbsp; $\text{7 kHz}$&nbsp; $($sampling frequency&nbsp; $\text{16 kHz})$,
 
*a total of nine modes between&nbsp; $\text{6.6 kbit/s}$&nbsp; and&nbsp; $\text{23.85 kbit/s}$&nbsp; (of which only five modes are used), and
 
*Improved voice quality and better (more natural) sound.
 
<br clear=all>
 
The table gives an overview of the different modes and their bit-size. You can visualize the quality of these speech coding methods for speech and music with the interactive SWF applet&nbsp; [[Applets:Quality_of_different_voice_codecs_(Applet)|"Quality of different speech codecs"]]&nbsp; (''Note:''&nbsp; "Only suitable for "Windows"! &nbsp; Adobe Flashplayer required!).
 
  
''Note'': &nbsp; The lower cutoff frequency of wideband AMR is specified as&nbsp; $\text{50 Hz}$&nbsp;z, but due to used pre-filters this is usually - and also in the audio demo - raised to&nbsp; $\text{200 Hz}$&nbsp; to reduce the susceptibility to interference and to take into account the characteristics of cell phone speakers and microphones.
 
 
  
 
{{BlaueBox|TEXT=
 
{{BlaueBox|TEXT=
 
$\text{Some features of wideband AMR}$&nbsp;
 
$\text{Some features of wideband AMR}$&nbsp;
  
*Speech data is delivered to the codec as PCM encoded speech with&nbsp; $16000$&nbsp; samples per second. The speech coding is done in blocks of&nbsp; $\text{20 ms}$&nbsp; and the data rate is adjusted every&nbsp; $\text{20 ms}$&nbsp; .
+
#Speech data is delivered to the codec as PCM encoded speech with&nbsp; $16\hspace{0.05cm}000$&nbsp; samples per second.&nbsp;
*The frequency band&nbsp; $\text{(50 Hz}$&nbsp; to&nbsp; $\text{7000 Hz})$&nbsp; is divided into two sub-bands, which are coded differently to allocate more bits to the subjectively important frequencies. The upper band&nbsp; $\text{(6400 Hz}$&nbsp; to&nbsp; $\text{7000 Hz})$&nbsp; is transmitted only in the highest mode $($with&nbsp; $\text{23.85 kbit/s)}$&nbsp;. In all other modes, only frequencies&nbsp; $\text{50 Hz}$&nbsp; to&nbsp; $\text{6400 Hz}$&nbsp; are considered in coding.
+
#The speech coding is done in blocks of&nbsp; $\text{20 ms}$&nbsp; and the data rate is adjusted every&nbsp; $\text{20 ms}$.
*Wideband AMR supports&nbsp; ''Discontinuous Transmission''&nbsp; (DTX). This feature means that transmission is paused during voice pauses, reducing both mobile station power consumption and overall interference at the air interface. This process is also known as&nbsp; ''Source-Controlled Rate'''&nbsp; (SCR).
+
#The band&nbsp; $\text{(50 Hz}$&nbsp; to&nbsp; $\text{7000 Hz})$&nbsp; is divided into two sub-bands,&nbsp; which are encoded differently to allocate more bits to the subjectively important frequencies.  
*The&nbsp; ''Voice Activity Detection''&nbsp; (VAD) determines whether speech is in progress or not and inserts an SID frame&nbsp; (''Silence Descriptor'')&nbsp; even during shorter speech pauses. The subscriber is suggested the feeling of a continuous connection by the decoder inserting synthetically generated background noise&nbsp; (English:&nbsp; ''Comfort Noise'')&nbsp; during speech pauses.}}
+
#The upper band&nbsp; $\text{(6400 Hz}$&nbsp; to&nbsp; $\text{7000 Hz})$&nbsp; is transmitted only in the highest mode $($with&nbsp; $\text{23.85 kbit/s)}$&nbsp;.  
 +
#In all other modes,&nbsp; only frequencies&nbsp; $\text{50 Hz}$&nbsp; to&nbsp; $\text{6400 Hz}$&nbsp; are considered in encoding.
 +
#Wideband AMR supports&nbsp; "discontinuous transmission"'&nbsp; $\rm (DTX)$.&nbsp; This feature means that transmission is paused during voice pauses,&nbsp; reducing both mobile station power consumption and overall interference at the air interface.&nbsp; This process is also known as&nbsp; "Source-Controlled Rate"&nbsp; $\rm (SCR)$.
 +
#The&nbsp; "Voice Activity Detection"&nbsp; $\rm (VAD)$&nbsp; determines whether speech is in progress or not and inserts a&nbsp; "silence descriptor frame"&nbsp; during speech pauses.  
 +
#The subscriber is suggested the feeling of a continuous connection by the decoder inserting synthetically generated&nbsp; "comfort noise"&nbsp; during speech pauses.}}
  
  
 
==Application of the CDMA method to UMTS==   
 
==Application of the CDMA method to UMTS==   
 
<br>
 
<br>
UMTS uses the multiple access method&nbsp; ''Direct Sequence Code Division Multiple Access''&nbsp; $\rm (DS-CDMA)$, which has already been discussed in the chapter [[Modulation_Methods/Direct-Sequence_Spread_Spectrum_Modulation#Block_diagram_and_equivalent_low-pass_model|"PN modulation"]]&nbsp; of the book "Modulation Methods".
+
UMTS uses the multiple access method&nbsp; "Direct Sequence Code Division Multiple Access"&nbsp; $\rm (DS-CDMA)$,&nbsp; which has already been discussed in the&nbsp; [[Modulation_Methods/Direct-Sequence_Spread_Spectrum_Modulation#Block_diagram_and_equivalent_low-pass_model|"PN modulation"]]&nbsp;  chapter&nbsp; of the book "Modulation Methods".
  
[[File:P_ID1533__Bei_T_4_3_S2c_v1.png|center|frame|CDMA transmission system for two subscribers '''Korrektur''']]
+
Here follows a brief summary of this method according to the diagram describing such a system in the equivalent low-pass range and highly simplified:
 +
[[File:EN_Bei_T_4_3_S2c.png|right|frame|CDMA transmission system for two subscribers]]
 +
 
 +
*The two data signals&nbsp; $q_1(t)$&nbsp; and&nbsp; $q_2(t)$&nbsp; are supposed to use the same channel without interfering with each other. The bit duration of each is&nbsp; $T_{\rm B}$.
 +
 
 +
*Each of the data signals is multiplied by an associated spreading code &nbsp; $c_1(t)$&nbsp; resp.&nbsp; $c_2(t)$.
  
Here follows a brief summary of this procedure according to the diagram describing such a system in the equivalent low-pass domain and highly simplified:
 
*The two data signals&nbsp; $q_1(t)$&nbsp; and&nbsp; $q_2(t)$&nbsp; are supposed to use the same channel without interfering with each other. The bit duration of each&nbsp; $T_{\rm B}$.
 
*Jedes der Datensignale wird mit einem zugeordneten Spreizcode –&nbsp; $c_1(t)$&nbsp; bzw.&nbsp; $c_2(t)$&nbsp; – multipliziert.
 
 
*The sum signal&nbsp; $s(t) = q_1(t) · c_1(t) + q_2(t) · c_2(t)$&nbsp; is formed and transmitted.
 
*The sum signal&nbsp; $s(t) = q_1(t) · c_1(t) + q_2(t) · c_2(t)$&nbsp; is formed and transmitted.
*At the receiver, the same spreading codes&nbsp; $c_1(t)$&nbsp; and&nbsp; $c_2(t)$&nbsp; are added, respectively, thus separating the signals again.
+
 
*Assuming that the spreading codes are orthogonal and that the AWGN noise is small, the two reconstructed signals at the receiver output are:
+
*At the receiver,&nbsp; the same spreading codes&nbsp; $c_1(t)$&nbsp; resp.&nbsp; $c_2(t)$&nbsp; are added,&nbsp; thus separating the signals again.
 +
 
 +
*Assuming orthogonal spreading codes and a small AWGN noise,&nbsp; the two reconstructed signals at the receiver output are:
 
:$$v_1(t) = q_1(t) \ \text{and} \ v_2(t) = q_2(t).$$
 
:$$v_1(t) = q_1(t) \ \text{and} \ v_2(t) = q_2(t).$$
*For AWGN noise signal&nbsp; $n(t)$&nbsp; and orthogonal spreading sequences, this does not change the error probability due to other participants.
+
*For AWGN noise signal&nbsp; $n(t)$&nbsp; and orthogonal spreading codes,&nbsp; this does not change the error probability due to other participants.
  
  
 
{{GraueBox|TEXT=
 
{{GraueBox|TEXT=
 
$\text{Example1:}$&nbsp;   
 
$\text{Example1:}$&nbsp;   
The graph above shows three data bits&nbsp; $(+1, -1, +1)$&nbsp; of the rectangular source signal&nbsp; $q_1(t)$&nbsp; from subscriber '''1''', each with the symbol duration&nbsp; $T_{\rm B}$.
+
The upper graph shows three data bits &nbsp; $(+1, -1, +1)$ &nbsp; of the rectangular signal&nbsp; $q_1(t)$&nbsp; from subscriber '''1''',&nbsp; each with symbol duration&nbsp; $T_{\rm B}$.
  
[[File:EN Mob T 3 4 S4.png|right|frame|Signals at ''Direct–Sequence'' Bandspread '''Korrektur''' evtl. Beswchreibung anpassen]]  
+
[[File:EN Mob T 3 4 S4.png|right|frame|Signals at&nbsp; "Direct–Sequence Spread Spectrum"]]  
*The symbol duration&nbsp; $T_{\rm C}$&nbsp; of the spreading code&nbsp; $c_1(t)$&nbsp; – which is also called&nbsp; '''chip duration'''&nbsp; is smaller by a factor&nbsp; $4$&nbsp;.  
+
*Here,&nbsp; the symbol duration&nbsp; $T_{\rm C}$&nbsp; of the spreading code&nbsp; $c_1(t)$ &nbsp; &rArr; &nbsp; also called&nbsp; "chip duration"&nbsp; is smaller by a factor&nbsp; $4$.
*The multiplication&nbsp; $s_1(t) = q_1(t) · c_1(t)$&nbsp; results in a chip current of length&nbsp; $12 · T_{\rm C}$.
+
 +
*The multiplication&nbsp; $s_1(t) = q_1(t) · c_1(t)$&nbsp; results in a chip sequence of length&nbsp; $12 · T_{\rm C}$.
  
  
Further one recognizes from this representation that the signal&nbsp; $s_1(t)$&nbsp; is of higher frequency than&nbsp; $q_1(t)$.  
+
One recognizes from this sketch that the signal&nbsp; $s_1(t)$&nbsp; is of higher frequency than&nbsp; $q_1(t)$.  
  
  
*This is why it is also called&nbsp; '''spread spectrum'''&nbsp;.
+
*This is why this modulation method is often also called&nbsp; "spread spectrum".
  
*The CDMA receiver reverses this, which is referred to as&nbsp; '''Band Sprawl'''&nbsp; }}
+
*The CDMA receiver reverses this&nbsp; "spreading".&nbsp; We refer to this&nbsp; "receiver-side spreading"&nbsp; as&nbsp; "despreading". }}
  
  
 
{{BlaueBox|TEXT=
 
{{BlaueBox|TEXT=
 
$\text{Summarizing:}$ &nbsp;   
 
$\text{Summarizing:}$ &nbsp;   
By applying&nbsp; $\rm (DS-CDMA)$&nbsp; to a payload bit sequence.
+
By applying&nbsp; "Direct Sequence Code Division Multiple Access"&nbsp; $\rm (DS-CDMA)$&nbsp; to a data bit sequence&nbsp; $q(t)$&nbsp;
*increases its bandwidth by the&nbsp; ''spreading factor''&nbsp; $J = T_{\rm B}/T_{\rm C}$ &ndash; this is equal to the number of&nbsp; ''chips per bit'';
+
*increases the bandwidth of&nbsp; $s(t) = q(t) \cdot c(t)$&nbsp;  by the&nbsp; &raquo;'''spreading factor'''&laquo; &nbsp; $J = T_{\rm B}/T_{\rm C}$&nbsp; &ndash;&nbsp; this is equal to the number of&nbsp; "chips per bit";
*if the chip rate&nbsp; $R_{\rm C}$&nbsp; is greater than the bit rate&nbsp; $R_{\rm B}$ by a factor&nbsp; $J$&nbsp;;
+
 
*the bandwidth of the entire CDMA signal is greater than the bandwidth of each user by&nbsp; $J$&nbsp;.
+
*the chip rate&nbsp; $R_{\rm C}$&nbsp; is greater than the bit rate&nbsp; $R_{\rm B}$ by a factor&nbsp; $J$;
 +
 
 +
*the bandwidth of the entire CDMA signal is greater than the bandwidth of each user by a factor&nbsp; $J$.
  
  
 
That is:&nbsp; &nbsp; $\text{In UMTS, the entire bandwidth is available to each subscriber for the entire transmission duration}$.  
 
That is:&nbsp; &nbsp; $\text{In UMTS, the entire bandwidth is available to each subscriber for the entire transmission duration}$.  
  
Recall:&nbsp; In GSM, both&nbsp; ''Frequency Division Multiple Access''&nbsp; and&nbsp; ''Time Division Multiple Access''&nbsp; are used as multiple access methods.
+
Recall:&nbsp; In GSM,&nbsp; both&nbsp; "Frequency Division Multiple Access"&nbsp; and&nbsp; "Time Division Multiple Access"&nbsp; are used as multiple access methods.
*Here, each subscriber has only a limited frequency band (FDMA), and.
+
*Here,&nbsp; each subscriber has only a limited frequency band&nbsp; $\rm (FDMA)$,&nbsp; and
*he only has access to the channel within time slots (TDMA)}}.
+
 
 +
*he only has access to the channel within time slots&nbsp; $\rm (TDMA)$.}}
  
 
 
 
 
 
==Spreading codes and scrambling with UMTS== 
 
==Spreading codes and scrambling with UMTS== 
 
<br>
 
<br>
Die Spreizcodes für UMTS sollen
+
The spreading codes for UMTS should
*zueinander orthogonal sein, um eine gegenseitige Beeinflussung der Teilnehmer zu vermeiden,
+
*be orthogonal to each other to avoid mutual interference between subscribers,
*eine flexible Realisierung unterschiedlicher Spreizfaktoren&nbsp; $J$&nbsp; ermöglichen.
 
  
 +
*allow a flexible realization of different spreading factors&nbsp; $J$.
  
Der hier dargelegte Sachverhalt wird auch durch das interaktive Applet&nbsp; [[Applets:OVSF_codes_(Applet)|OVSF–Codes]]&nbsp; verdeutlicht.
+
 
 +
&rArr; &nbsp; The issue presented here is also illustrated by the German-language SWF applet&nbsp; [[Applets:OVSF_codes_(Applet)|"OVSF codes"]].
  
 
{{GraueBox|TEXT=
 
{{GraueBox|TEXT=
$\text{Beispiel 2:}$&nbsp;   
+
$\text{Example 2:}$&nbsp;   
Ein Beispiel hierfür sind die&nbsp; '''Codes mit variablem Spreizfaktor'''&nbsp; (englisch:&nbsp; ''Orthogonal Variable Spreading Faktor'', '''OVSF'''), die Codes der Längen von&nbsp; $J = 4$&nbsp; bis&nbsp; $J = 512$&nbsp; bereitstellen. Diese können, wie in der Grafik zu sehen ist, mit Hilfe eines Codebaums erstellt werden. Dabei entstehen bei jeder Verzweigung aus einem Code&nbsp; $C$&nbsp; zwei neue Codes&nbsp; $(+C \ +\hspace{-0.1cm}C)$&nbsp; und&nbsp; $(+C \ -\hspace{-0.1cm}C)$.
+
An example of this is the &nbsp; &raquo;'''orthogonal variable spreading factor'''&laquo; &nbsp; $\rm  (OVSF)$,&nbsp; which provide codes of lengths from&nbsp; $J = 4$&nbsp; to&nbsp; $J = 512$.  
  
[[File:P_ID1535__Bei_T_4_3_S3c_v1.png|right|frame|Schaubild zur OVSF–Codefamilie]]
+
[[File:P_ID1535__Bei_T_4_3_S3c_v1.png|right|frame|Chart on the OVSF code family]]
  
Anzumerken ist, dass kein Vorgänger und Nachfolger eines Codes benutzt werden darf.
+
These can be created using a code tree, as shown in the diagram.&nbsp; Here,&nbsp; at each branch,&nbsp; two new codes are created from one code&nbsp; $C$:
*Im gezeichneten Beispiel könnten also acht Spreizcodes mit Spreizfaktor&nbsp; $J = 8$&nbsp; verwendet werden.
+
:#&nbsp; $(+C \ +\hspace{-0.1cm}C)$,&nbsp; and
*Auch die vier gelb hinterlegten Codes – einmal mit&nbsp; $J = 2$, einmal mit&nbsp; $J = 4$&nbsp; und zweimal mit&nbsp; $J = 8$&nbsp; – sind möglich.
+
:#&nbsp; $(+C \ -\hspace{-0.1cm}C)$.
*Die unteren vier Codes mit dem Spreizfaktor&nbsp; $J = 8$&nbsp; können aber nicht herangezogen werden, da diese alle mit "$+1 \ \hspace{-0.1cm}1$" beginnen, was bereits durch den OVSF–Code mit Spreizfaktor&nbsp; $J = 2$&nbsp; belegt ist.}}
 
  
 +
Note that no predecessor and successor of a code may be used. 
 +
*So in the drawn example,&nbsp; eight spreading codes with spreading factor&nbsp; $J = 8$&nbsp; could be used.
 +
 +
*Also possible are the four codes with yellow background
 +
:*once with&nbsp; $J = 2$,
 +
:*once with&nbsp; $J = 4$,&nbsp; and
 +
:*twice with&nbsp; $J = 8$.
 +
  
Um mehr Spreizcodes zu erhalten und so mehr Teilnehmer versorgen zu können, wird nach der Bandspreizung mit&nbsp; $c(t)$&nbsp; die Folge mit&nbsp; $w(t)$&nbsp; chipweise nochmals verwürfelt, ohne dass eine weitere Spreizung stattfindet. Der&nbsp; '''Verwürfelungscode''' $w(t)$&nbsp; hat die gleiche Länge und die selbe Rate wie&nbsp; $c(t)$.
+
But the lower four codes with spreading factor&nbsp; $J = 8$&nbsp; cannot be used,&nbsp; because they all start with&nbsp; "$+1 \ -\hspace{-0.1cm}1$",&nbsp; which is already occupied by the OVSF code with spreading factor&nbsp; $J = 2$.}}
  
[[File:EN_Mob_T_3_4_S5.png|left|frame|Zusätzliche Verwürfelung nach Spreizung]]
 
  
Durch die Verwürfelung&nbsp; (englisch:&nbsp; ''Scrambling'')&nbsp; verlieren die Codes ihre vollständige Orthogonalität; man nennt sie ''quasi–othogonal''.  
+
[[File:EN_Mob_T_3_4_S5.png|right|frame|Additional scrambling after spreading]]
 +
To obtain more spreading codes and thus be able to supply more participants,&nbsp; after the band spreading with&nbsp; $c(t)$&nbsp;
 +
*the sequence is chip-wise scrambled again with&nbsp; $w(t)$,&nbsp;
 +
 
 +
*without any further spreading.
 +
 
 +
 
 +
The&nbsp; &raquo;'''scrambling code'''&laquo; $w(t)$&nbsp; has same length and rate as the spreading code&nbsp; $c(t)$.
 +
<br clear=all>
 +
[[File:EN_Mob_T_3_4_S5b_v3.png|right|frame|Typical spreading and scrambling codes for UMTS ]]
 +
 
 +
&rArr; &nbsp; Scrambling causes the codes to lose their complete orthogonality; they are called&nbsp; "quasi-orthogonal".  
  
*Bei diesen Codes ist zwar die&nbsp; [[Modulation_Methods/Spreading_Sequences_for_CDMA#Properties_of_the_correlation_functions|"Kreuzkorrelationsfunktion"]]&nbsp; (KKF) zwischen unterschiedlichen Spreizcodes ungleich Null.
+
*For these codes,&nbsp; although the&nbsp; [[Modulation_Methods/Spreading_Sequences_for_CDMA#Properties_of_the_correlation_functions|"cross-correlation function"]]&nbsp; $\rm (CCF)$&nbsp; between different spreading codes is non-zero.  
*Sie zeichnen sich aber durch eine ausgeprägte&nbsp; [[Modulation_Methods/Spreading_Sequences_for_CDMA#Properties_of_the_correlation_functions|"Autokorrelationsfunktion"]]&nbsp; um den Nullpunkt aus, was die Detektion am Empfänger erleichtert.
 
  
 +
*But they are characterized by a pronounced&nbsp; [[Modulation_Methods/Spreading_Sequences_for_CDMA#Properties_of_the_correlation_functions|"auto-correlation function"]]&nbsp; $\rm (ACF)$&nbsp; around zero,&nbsp; which facilitates detection at the receiver.
  
[[File:EN_Mob_T_3_4_S5b_v3.png|right|frame|Typische Spreiz– und Verwürfelungscodes für UMTS ]]
+
*Using quasi-orthogonal codes makes sense because the set of orthogonal codes is limited and scrambling allows also different users to use the same spreading codes.
  
  
Die Verwendung quasi–orthogonaler Codes macht Sinn, da die Menge an orthogonalen Codes begrenzt ist und durch die Verwürfelung verschiedene Teilnehmer auch gleiche Spreizcodes verwenden können.
+
The table summarizes some data of spreading and scrambling codes.
<br><br><br><br><br><br>
 
Die Tabelle fasst einige Daten der Spreiz– und Verwürfelungscodes zusammen.
 
 
<br clear=all>
 
<br clear=all>
[[File:P_ID1537__Bei_T_4_3_S3b_v2.png|right|frame|Generator zur Erzeugung von Goldcodes]]
 
 
{{GraueBox|TEXT=
 
{{GraueBox|TEXT=
$\text{Beispiel 3:}$&nbsp;   
+
[[File:P_ID1537__Bei_T_4_3_S3b_v2.png|right|frame|Generator for creating Gold codes]]
Bei  UMTS werden für die Verwürfelung so genannte&nbsp; '''Goldcodes'''&nbsp; verwendet. Die Grafik aus&nbsp; [3gpp]<ref>3gpp Group: ''UMTS Release 6 Technical Specification'' 25.213 V6.4.0., Sept. 2005.</ref>&nbsp; zeigt das Blockschaltbild zur schaltungstechnischen Erzeugung solcher Codes.  
+
$\text{Example 3:}$&nbsp;   
*Dabei werden zunächst zwei unterschiedliche Pseudonoise–Folgen gleicher Länge&nbsp; $($hier:&nbsp; $N = 18)$&nbsp; mit Hilfe von Schieberegistern parallel erzeugt und mit&nbsp; ''Exklusiv–Oder–Gatter''&nbsp; bitweise addiert.
+
In UMTS, so-called&nbsp; &raquo;'''Gold codes'''&laquo;&nbsp; are used for scrambling. The graphic from&nbsp; [3gpp]<ref>3gpp Group:&nbsp; UMTS Release 6 - Technical Specification 25.213 V6.4.0,&nbsp; Sept. 2005.</ref>&nbsp; shows the block diagram for the circuitry generation of such codes.  
*Im Uplink hat jede Mobilstation einen eigenen Verwürfelungscode und die Trennung der einzelnen Kanäle erfolgt über den jeweils gleichen Code.
+
*Two different pseudonoise sequences of equal length&nbsp; $($here:&nbsp; $N = 18)$&nbsp; are first generated in parallel using shift registers and added bitwise using&nbsp; "exclusive-or"&nbsp; gates''.
*Dagegen hat im Downlink jedes Versorgungsgebiet eines "Node B" einen gemeinsamen Verwürfelungscode.}}
+
 
 +
*In the uplink,&nbsp; each mobile station has its own scrambling code and the separation of each channel is done using the same code.
  
 +
*In contrast,&nbsp; in the downlink, each coverage area of a&nbsp; "Node B"&nbsp; has a common scrambling code.}}
  
==Kanalcodierung  bei UMTS==  
+
 
 +
==Channel coding for UMTS==  
 
<br>
 
<br>
Ebenso wie bei GSM erfahren EFR– und AMR–codierte Sprachdaten im UMTS einen zweistufigen Fehlerschutz, bestehend aus
+
With UMTS,&nbsp; the EFR- and AMR-encoded voice data pass through a two-stage error protection&nbsp; $($similar to GSM$)$,&nbsp; consisting of
*Bildung von CRC–Prüfbits&nbsp; (englisch:&nbsp; ''Cyclic Redundancy Check''), und
+
[[File:EN_Bei_T_4_3_S4a_v1.png|right|frame|Insertion of CRC bits and tailbits in UMTS ]]
*anschließender Faltungscodierung&nbsp; (englisch:&nbsp; ''Convolutional Coding'').
+
#formation of&nbsp; "cyclic redundancy check bits"&nbsp; $\rm (CRC)$,
 +
#subsequent convolutional encoding.
  
  
Diese Verfahren unterscheiden sich jedoch von denjenigen bei GSM durch eine größere Flexibilität, da sie bei UMTS unterschiedliche Datenraten berücksichtigen müssen.
+
However,&nbsp; these methods differ from those used for GSM in that they are more flexible,&nbsp; since for UMTS they have to take different data rates into account.
  
[[File:EN_Bei_T_4_3_S4a_v1.png|right|frame|Einfügen von CRC&ndash; und Tailbits bei UMTS ]]
+
&rArr; &nbsp; For&nbsp; &raquo;'''error detection'''&laquo;,&nbsp;  eight, twelve, sixteen or &nbsp;$24$&nbsp; CRC bits are formed depending on the size of the transport block&nbsp; $\text{(10 ms}$&nbsp; or&nbsp; $\text{20 ms})$,&nbsp; and appended to it.  
Für die&nbsp; '''Fehlererkennung'''&nbsp; mittels CRC werden je nach Größe des Transportblockes&nbsp; $\text{(10 ms}$&nbsp; oder&nbsp; $\text{20 ms})$&nbsp; acht, zwölf, sechzehn oder &nbsp;$24$&nbsp; CRC–Bit gebildet und an diesen angehängt.  
+
 
*Am Ende eines jeden Rahmens werden außerdem acht Tailbits eingefügt, die Synchronisationszwecken dienen.  
+
*Eight tail bits are also inserted at the end of each frame for synchronization purposes.
*Die Grafik zeigt einen beispielhaften Transportblock des '''DCH'''–Kanals mit &nbsp;$164$&nbsp; Nutzdatenbits, an den &nbsp;$16$&nbsp; CRC–Prüfbits und acht Tailbits angehängt werden.
+
 +
*The diagram shows a transport block of the '''DCH''' channel with &nbsp;$164$&nbsp; user data bits,&nbsp; to which &nbsp;$16$&nbsp; CRC bits and eight tail bits are appended.
 
<br clear=all>
 
<br clear=all>
  
Für die&nbsp; '''Fehlerkorrektur'''&nbsp; kommen bei UMTS – je nach Datenrate – zwei verschiedene Verfahren zum Einsatz:
+
&rArr; &nbsp; For&nbsp; &raquo;'''error correction'''&laquo;,&nbsp; UMTS uses two different methods,&nbsp; depending on the data rate:
*Bei niedrigen Datenraten werden wie bei GSM&nbsp; [[Channel_Coding/Basics_of_Convolutional_Coding|"Faltungscodes"]]&nbsp; (englisch:&nbsp; ''Convolutional Codes'')&nbsp; mit den Coderaten&nbsp; $R = 1/2$&nbsp; oder&nbsp; $R = 1/3$&nbsp; verwendet. Diese werden mit acht Speicherelementen eines rückgekoppelten Schieberegisters&nbsp; $(256$&nbsp; Zustände$)$ erzeugt. Der Codiergewinn beträgt mit der Coderate&nbsp; $R = 1/3$&nbsp; und bei niedrigen Fehlerraten ca.&nbsp; $4.5$&nbsp; bis&nbsp; $6$&nbsp; dB.
+
*For low data rates,&nbsp; [[Channel_Coding/Basics_of_Convolutional_Coding|"convolutional codes"]]&nbsp; with code rates&nbsp; $R = 1/2$ &nbsp; or &nbsp; $R = 1/3$&nbsp; are used as with GSM.&nbsp; These are generated with eight memory elements of a feedback shift register&nbsp; $(256$&nbsp; states$)$.&nbsp; The coding gain is approximately&nbsp; $4.5$&nbsp; to&nbsp; $6$&nbsp; dB with code rate&nbsp; $R = 1/3$&nbsp; and at low error rates.
*Bei höheren Datenraten verwendet man&nbsp; [[Channel_Coding/The_Basics_of_Turbo_Codes|"Turbo–Codes"]]&nbsp; der Rate&nbsp; $R = 1/3$. Das Schieberegister besteht hier aus drei Speicherzellen, die insgesamt acht Zustände annehmen können. Der Gewinn der Turbo–Codes ist gegenüber Faltungscodes um&nbsp; $2$&nbsp; bis&nbsp; $3$&nbsp; dB größer und abhängig von der Anzahl der Iterationen. Sie benötigen dafür einen Prozessoren mit hoher Rechenleistung und es kann es zu relativ großen Verzögerungen kommen.
 
  
 +
*For higher data rates,&nbsp; one uses&nbsp; [[Channel_Coding/The_Basics_of_Turbo_Codes|"turbo codes"]]&nbsp; of rate &nbsp; $R = 1/3$.&nbsp; The shift register consists here of three memory cells,&nbsp; which can assume a total of eight states.&nbsp; The gain of turbo codes is larger by&nbsp; $2$&nbsp; to&nbsp; $3$&nbsp; dB than by convolutional codes and depends on the number of iterations.&nbsp; You need a processor with high processing power for this and there may be relatively large delays.
  
Nach der Kanalcodierung werden die Daten wie bei GSM einer&nbsp; [[Examples_of_Communication_Systems/Entire_GSM_Transmission_System#Interleaving_for_speech_signals|"Interleaver"]]&nbsp; zugeführt, um empfangsseitig die durch Fading entstandenen Bündelfehler auflösen zu können. Schließlich werden zur&nbsp; ''Ratenanpassung''&nbsp; der entstandenen Daten an den physikalischen Kanal einzelne Bit nach einem vorgegebenen Algorithmus entfernt&nbsp; (''Puncturing'')&nbsp; oder wiederholt&nbsp; (''Repetition'').
 
  
[[File:EN_Bei_T_4_3_S4b.png|right|frame|Fehlerkorrekturmechanismen bei UMTS]]
+
After channel coding,&nbsp; the data is fed to an&nbsp; [[Examples_of_Communication_Systems/Entire_GSM_Transmission_System#Interleaving_for_speech_signals|"interleaver"]]&nbsp; as in GSM,&nbsp; in order to be able to resolve bundle errors caused by fading on the receiving side.&nbsp; Finally, for&nbsp; "rate matching"&nbsp; of the resulting data to the physical channel,&nbsp; individual bits are removed&nbsp; $($"puncturing"$)$&nbsp; or repeated&nbsp; $($"repetition"$)$&nbsp; according to a predetermined algorithm.
 +
 
 
{{GraueBox|TEXT=
 
{{GraueBox|TEXT=
$\text{Beispiel 4:}$&nbsp;   
+
[[File:EN_Bei_T_4_3_S4b.png|right|frame|Error correction mechanisms in UMTS]]
Die Grafik zeigt zunächst die Zunahme der Bits durch einen Faltungs– oder Turbocode der Rate&nbsp; $R =1/3$, wobei aus dem&nbsp; $188$–Bit–Zeitrahmen (nach der CRC–Prüfsumme und den Tailbits) ein&nbsp; $564$–Bit–Rahmen entsteht.
+
$\text{Example 4:}$&nbsp;   
*Danach folgt eine erste externe Verschachtelung und dann eine zweite interne Verschachtelung.  
+
The graph first shows the increase in bits due to a convolutional or turbo code of rate&nbsp; $R =1/3$, where&nbsp; the&nbsp; $188$&nbsp; bit time frame&nbsp; $($after the CRC checksum and tail bits$)$&nbsp; becomes a&nbsp; $564$&nbsp; bit frame.
*Nach dieser wird der Zeitrahmen in vier Unterrahmen mit jeweils&nbsp; $141$&nbsp; Bit aufgeteilt und diese werden anschließend durch eine Ratenanpassung an den physikalischen Kanal angepasst.}}
+
 
 +
*Followed by a first&nbsp; $($external$)$&nbsp; nesting and then a second&nbsp; $($internal$)$&nbsp; nesting.
 +
 +
*After this,&nbsp; the time frame is divided into four subframes of&nbsp; $141$&nbsp; bits each,&nbsp; and these are then matched to the physical channel by rate matching.}}
  
  
Line 174: Line 217:
 
==Frequency responses and pulse shaping for UMTS== 
 
==Frequency responses and pulse shaping for UMTS== 
 
<br>
 
<br>
[[File:UMTS_Bild_1.png|right|frame|Blockschaltbild des optimalen Nyquistentzerrers bei idealem Kanal  '''Korrektur'''|class=fit]]
+
[[File:EN_Bei_T_4_3_UMTS1_v2.png|right|frame|Block diagram of the optimal Nyquist equalizer at ideal channel|class=fit]]
In diesem Abschnitt gehen wir von folgendem Blockschaltbild eines Binärsystems bei idealem Kanal aus &nbsp; &rArr; &nbsp; $H_{\rm K}(f) = 1$.
+
In this section,&nbsp; we assume the following block diagram of a binary system with ideal channel &nbsp; &rArr; &nbsp; $H_{\rm K}(f) = 1$.
  
Insbesondere gelte:
+
In particular,&nbsp; let hold:
  
*Das&nbsp; ''Sendeimpulsfilter''&nbsp; wandelt die binären&nbsp; $\{0, \ 1\}$ Daten in physikalische Signale. Das Filter wird beschrieben durch den Frequenzgang&nbsp; $H_{\rm S}(f)$, der formgleich mit dem Spektrum eines einzelnen Sendeimpulses ist.
+
*The&nbsp; "transmitter pulse filter"&nbsp; converts the binary&nbsp; $\{0, \ 1\}$ data into physical signals.&nbsp; The filter is described by the frequency response&nbsp; $H_{\rm S}(f)$,&nbsp; which is identical in shape to the spectrum of a single transmitted pulse.
  
  
*Bei UMTS ist das Empfangsfilter&nbsp; $H_{\rm E}f) = H_{\rm S}(f)$&nbsp; an den Sender angepasst&nbsp; (''Matched–Filter'')&nbsp; und der Gesamtfrequenzgang&nbsp; $H(f) = H_{\rm S}(f) · H_{\rm E}(f)$&nbsp; erfüllt das&nbsp; [[Digitalsignalübertragung/Eigenschaften_von_Nyquistsystemen#Erstes_Nyquistkriterium_im_Frequenzbereich|erste Nyquistkriterium]]:
+
*In UMTS,&nbsp; the receiver filter&nbsp; $H_{\rm E}f) = H_{\rm S}(f)$&nbsp; is matched to the transmitter&nbsp; $($"matched filter"$)$&nbsp; and the overall frequency response&nbsp; $H(f) = H_{\rm S}(f) \cdot H_{\rm E}(f)$&nbsp; satisfies the&nbsp; [[Digital_Signal_Transmission/Properties_of_Nyquist_Systems#First_Nyquist_criterion_in_the_frequency_domain|"first Nyquist criterion"]]:
 
:$$ H(f) = H_{\rm CRO}(f)  =  \left\{ \begin{array}{c}    1 \\  0 \\  \cos^2 \left( \frac {\pi \cdot (|f| - f_1)}{2 \cdot (f_2 - f_1)} \right)\end{array} \right.\quad
 
:$$ H(f) = H_{\rm CRO}(f)  =  \left\{ \begin{array}{c}    1 \\  0 \\  \cos^2 \left( \frac {\pi \cdot (|f| - f_1)}{2 \cdot (f_2 - f_1)} \right)\end{array} \right.\quad
\begin{array}{*{1}c} {\rm{f\ddot{u}r}} \\ {\rm{f\ddot{u}r}}\\  {\rm sonst }\hspace{0.05cm}.  \end{array}
+
\begin{array}{*{1}c} {\rm{for}} \\ {\rm{for}}\\  {\rm else }\hspace{0.05cm}.  \end{array}
\begin{array}{*{20}c} |f| \le f_1,  \\ |f| \ge f_2,\\  \\\end{array}$$
+
\begin{array}{*{20}c} |f| \le f_1,  \\ |f| \ge f_2.\\  \\\end{array}$$
 +
 
 +
This means: &nbsp; Consecutive pulses in time do not interfere with each other &nbsp; ⇒ &nbsp; no&nbsp; [[Digital_Signal_Transmission/Causes_and_Effects_of_Intersymbol_Interference|"intersymbol interference"]]&nbsp; $\rm  (ISI)$&nbsp; occur.&nbsp; The associated time function is:
 +
 
 +
:$$h(t) = h_{\rm CRO}(t) ={\rm sinc}(t/ T_{\rm C}) \cdot \frac{\cos(r \cdot \pi t/T_{\rm C})}{1- (2r \cdot t/T_{\rm C})^2}\hspace{0.4cm} \text{with  } \hspace{0.4cm} r =  \frac{f_2 - f_1}{f_2 + f_1}. $$
 +
 +
*"CRO" here stands for&nbsp; [[Linear_and_Time_Invariant_Systems/Some_Low-Pass_Functions_in_Systems_Theory#Raised-cosine_low-pass_filter|"raised cosine low-pass"]].
 
   
 
   
Das bedeutet: &nbsp; Zeitlich aufeinander folgende Impulse stören sich nicht gegenseitig  &nbsp; &nbsp; es treten keine&nbsp; [[Digitalsignalübertragung/Ursachen_und_Auswirkungen_von_Impulsinterferenzen|Impulsinterferenzen]]&nbsp; (englisch:&nbsp; ''Intersymbol Interference'', ISI) auf. Die zugehörige Zeitfunktion lautet:
+
*The sum&nbsp; $f_1 + f_2$&nbsp; is equal to the inverse of the chip duration&nbsp; $T_{\rm C} = 260 \ \rm ns$,&nbsp; so it is equal to&nbsp; $3.84 \ \rm MHz$.
 +
   
 +
*The&nbsp; "rolloff factor"&nbsp; has been determined to&nbsp; $r = 0.22$&nbsp; for UMTS.&nbsp; The two&nbsp; "corner frequencies"&nbsp; are thus
  
:$$h(t) = h_{\rm CRO}(t) ={\rm si}(\pi \cdot t/ T_{\rm C}) \cdot \frac{\cos(r \cdot \pi t/T_{\rm C})}{1- (2r \cdot  t/T_{\rm C})^2}. $$
+
:$$f_1 = {1}/(2 T_{\rm C}) \cdot (1-r) \approx 1.5\,{\rm MHz},$$
 +
:$$f_2 ={1}/(2 T_{\rm C}) \cdot (1+r) \approx 2.35\,{\rm MHz}.$$
 
   
 
   
*„CRO” steht hierbei für&nbsp; [[Linear_and_Time_Invariant_Systems/Einige_systemtheoretische_Tiefpassfunktionen#Cosinus-Rolloff-Tiefpass|Cosinus–Rolloff]]&nbsp; (englisch:&nbsp; ''Raised Cosine'').
+
*The required bandwidth is&nbsp; $B = 2 \cdot f_2 = 4.7 \ \rm MHz$.&nbsp; Thus,&nbsp; there is sufficient bandwidth available for each UMTS channel with&nbsp; $5 \ \rm MHz$.
*Die Summe&nbsp; $f_1 + f_2$&nbsp; ist gleich dem Kehrwert der Chipdauer&nbsp; $T_{\rm C} = 260 \ \rm ns$, also gleich&nbsp; $3.84 \ \rm MHz$.  
 
*Der&nbsp; ''Rolloff–Faktor''&nbsp; (wir bleiben bei der in&nbsp; $\rm LNTwww$&nbsp; gewählten Bezeichnung&nbsp; $r$, im UMTS–Standard wird hierfür&nbsp; $\alpha$&nbsp; verwendet)
 
  
:$$r =  \frac{f_2 - f_1}{f_2 + f_1} $$
 
 
:wurde bei UMTS zu&nbsp; $r = 0.22$&nbsp; festgelegt. Die beiden Eckfrequenzen sind somit
 
  
:$$f_1 = {1}/(2 T_{\rm C}) \cdot (1-r) \approx 1.5\,{\rm MHz}, \hspace{0.2cm}
+
{{BlaueBox|TEXT=
f_2 ={1}/(2 T_{\rm C}) \cdot (1+r) \approx 2.35\,{\rm MHz}.$$
+
$\text{Conclusion:}$&nbsp; The graph shows.  
 +
[[File:P_ID1547__Bei_T_4_3_S5b_v1.png|right|frame|Raised cosine spectrum and impulse response]]
 +
*on the left,&nbsp; the&nbsp; $($normalized$)$&nbsp; Nyquist spectrum&nbsp; $H(f)$,  
 
   
 
   
*Die erforderliche Bandbreite beträgt&nbsp; $B = 2 · f_2 = 4.7 \ \rm MHz$. Für jeden UMTS–Kanal steht somit mit&nbsp; $5 \ \rm MHz$&nbsp; ausreichend Bandbreite zur Verfügung.
+
*on the right,&nbsp; the corresponding Nyquist pulse&nbsp; $h(t)$,&nbsp; whose zero crossings are equidistant with distance&nbsp; $T_{\rm C}$.  
  
  
[[File:P_ID1547__Bei_T_4_3_S5b_v1.png|right|frame|Cosinus–Rolloff–Spektrum und Impulsantwort]]
+
$\text{It should be noted:}$
{{BlaueBox|TEXT=
+
# The transmission filter&nbsp; $H_{\rm S}(f)$&nbsp; and the matched filter&nbsp; $H_{\rm E}(f)$&nbsp; are each&nbsp; [[Digital_Signal_Transmission/Optimization_of_Baseband_Transmission_Systems#Root_Nyquist_systems|"root raised cosine"]].  
$\text{Fazit:}$&nbsp;  Die Grafik zeigt
+
#Only the product&nbsp; $H(f) = H_{\rm S}(f) \cdot H_{\rm E}(f)$&nbsp; leads to the raised cosine.&nbsp; This also means:
*links das (normierte) Nyquistspektrum&nbsp; $H(f)$, und
+
# The impulse responses&nbsp; $h_{\rm S}(t)$&nbsp; and&nbsp; $h_{\rm E}(t)$&nbsp; by themselves do not satisfy the first Nyquist condition.  
*rechts den zugehörigen Nyquistimpuls&nbsp; $h(t)$, dessen Nulldurchgänge im Abstand&nbsp; $T_{\rm C}$&nbsp; äquidistant sind.
+
#Only the combination of the two&nbsp; $($in the time domain the convolution$)$&nbsp; leads to the desired equidistant zeros.}}
<br clear=all>
 
$\text{Es ist zu beachten:}$
 
* Das Sendefilter&nbsp; $H_{\rm S}(f)$&nbsp; und das Matched–Filter&nbsp; $H_{\rm E}(f)$&nbsp; sind jeweils&nbsp; [[Digitalsignalübertragung/Optimierung_der_Basisbandübertragungssysteme#Wurzel.E2.80.93Nyquist.E2.80.93Systeme|Wurzel–Cosinus–Rolloff–förmig]]&nbsp; (englisch:&nbsp; ''Root Raised Cosine''). Erst das Produkt&nbsp; $H(f) = H_{\rm S}(f) · H_{\rm E}(f)$&nbsp; führt zum Cosinus–Rolloff.
 
*Das bedeutet auch: &nbsp; Die Impulsantworten&nbsp; $h_{\rm S}(t)$&nbsp; und&nbsp; $h_{\rm E}(t)$&nbsp; erfüllen für sich allein die erste Nyquistbedingung nicht. Erst die Kombination aus beiden (im Zeitbereich die Faltung) führt zu den gewünschten äquidistanten Nulldurchgängen.}}
 
  
==Modulationsverfahren bei UMTS==  
+
==Modulation methods for UMTS==  
 
<br>
 
<br>
Die bei UMTS eingesetzten&nbsp; '''Modulationsverfahren'''&nbsp; können wie folgt zusammengefasst werden:
+
The modulation techniques used in UMTS can be summarized as follows:
*In der Abwärtsrichtung&nbsp; (''Downlink'')&nbsp; wird zur Modulation&nbsp; ''Quaternary Phase Shift Keying''&nbsp; (QPSK) verwendet &ndash; sowohl bei ''FDD''&nbsp; als auch bei ''TDD''. Dabei werden Nutzdaten (DPDCH–Kanal) und Kontrolldaten (DPCCH–Kanal) zeitlich gemultiplext.
+
#In the downlink:&nbsp;  "Quaternary Phase Shift Keying"&nbsp; is used for modulation&nbsp; both in&nbsp; "frequency division duplex"&nbsp; $\rm (FDD)$&nbsp; and in&nbsp; "time division duplex""&nbsp; $\rm (TDD)$.
*Bei ''TDD''&nbsp; wird das Signal in Aufwärtsrichtung&nbsp; (''Uplink'')&nbsp; ebenfalls mittels QPSK moduliert, nicht aber bei ''FDD''. Hier wird vielmehr eine&nbsp; '''zweifache binäre PSK'''&nbsp; (englisch:&nbsp; ''Dual Channel–BPSK'') verwendet.
+
#Here,&nbsp; user data&nbsp; $($DPDCH channel$)$&nbsp; and control data&nbsp; $($DPCCH channel$)$&nbsp; are multiplexed in time.
 +
#With TDD,&nbsp; the signal is modulated in the uplink  also by means of QPSK,&nbsp; but not with&nbsp; FDD.&nbsp;  
 +
#Here,&nbsp;  a&nbsp; "dual channel binary phase shift keying"&nbsp;  is used &nbsp; &rArr;  &nbsp;  different channels are transmitted in&nbsp; "in-phase"&nbsp; and&nbsp; "quadrature components".
 +
#Thus,&nbsp; two chips are transmitted per modulation step.&nbsp; The gross chip rate is therefore twice the modulation rate of&nbsp; $3.84$ Mchip per second.
  
  
Bei&nbsp; ''Dual–Channel BPSK''&nbsp; wird zwar ebenfalls der QPSK–Signalraum genutzt, aber in ''Inphase''– und ''Quadratur–Komponente''&nbsp; werden unterschiedliche Kanäle übertragen. Pro Modulationsschritt werden also zwei Chips übertragen. Die Brutto–Chiprate ist daher doppelt so groß wie die Modulationsrate von&nbsp; $3.84$ Mchip pro Sekunde.
+
{{GraueBox|TEXT=
 +
$\text{Example 5:}$&nbsp;
 +
The graph shows in the equivalent low-pass domain this&nbsp; "I/Q multiplexing method",&nbsp; as it is also called:
 
[[File:EN_Mob_T_3_4_S6.png|right|frame|Modulation and pulse shaping for UMTS|class=fit]]
 
[[File:EN_Mob_T_3_4_S6.png|right|frame|Modulation and pulse shaping for UMTS|class=fit]]
{{GraueBox|TEXT=
+
 
$\text{Beispiel 5:}$&nbsp;
+
#The spread useful data of the DPDCH channel is modulated onto the inphase component.
Die Grafik zeigt dieses I/Q–Multiplexing–Verfahren, wie es auch bezeichnet wird, im äquivalenten Tiefpassbereich:
+
#The spread control data of the DPCCH channel is modulated onto the quadrature component.
*Die gespreizten Nutzdaten des DPDCH–Kanals werden auf die Inphase–Komponente und die gespreizten Kontrolldaten des DPCCH–Kanals auf die Quadratur–Komponente moduliert und übertragen.
+
#After modulation,&nbsp; the quadrature component is weighted by the root of the power ratio&nbsp; $G$&nbsp; between the two channels to minimize the influence of power differences between&nbsp; $I$&nbsp; and&nbsp; $Q$.
*Nach der Modulation wird die Quadratur–Komponente mit der Wurzel des Leistungsverhältnisses&nbsp; $G$&nbsp; zwischen den beiden Kanälen gewichtet, um den Einfluss des Leistungsunterschieds zwischen&nbsp; $I$&nbsp; und&nbsp; $Q$&nbsp; zu minimieren.
+
#Finally,&nbsp; the complex sum signal&nbsp; $(I +{\rm j} \cdot Q)$&nbsp; is multiplied by a scrambling code that is also complex.}}
*Abschließend wird das komplexe Summensignal&nbsp; $(I +{\rm j} · Q)$&nbsp; mit einem ebenfalls komplexen Verwürfelungscode multipliziert.}}
 
  
  
 
{{BlaueBox|TEXT=
 
{{BlaueBox|TEXT=
$\text{Fazit:}$&nbsp; Ein Vorteil der zweifachen BPSK–Modulation ist die Möglichkeit der Verwendung&nbsp; '''stromsparender Verstärker'''.
+
$\text{Conclusion:}$&nbsp; An advantage of dual channel BPSK modulation is the&nbsp; ''' possibility of usinglow-power amplifiers'''.
*Zeitmultiplex von Nutz– und Kontrolldaten wie im&nbsp; ''Uplink''&nbsp; ist aber im&nbsp; ''Downlink''&nbsp; nicht möglich.  
+
*But time division multiplexing of user and control data as in the uplink&nbsp; '''is not possible in the downlink'''.  
*Ein Grund hierfür ist der Einsatz von&nbsp; ''Discontinuous Transmission''&nbsp; (DTX) und die damit verbundenen zeitlichen Einschränkungen.}}
+
 
 +
*One reason for this is the use of&nbsp; "Discontinuous Transmission"&nbsp; $\rm (DTX)$&nbsp; and the associated time constraints.}}
  
 
   
 
   
==Single–User–Empfänger==
+
==Single-user receiver==
 
<br>
 
<br>
Aufgabe eines CDMA–Empfängers ist es, aus der Summe der gespreizten Datenströme die gesendeten Daten der einzelnen Teilnehmer zu separieren und zu rekonstruieren. Dabei unterscheidet man zwischen&nbsp; ''Single–User''–Empfängern und&nbsp; ''Multi–User''–Empfängern.
+
The task of a CDMA receiver is to separate and reconstruct the transmitted data of the individual subscribers from the sum of the spread data streams.&nbsp; A distinction is made between&nbsp; "single-user receivers"&nbsp; and&nbsp; "multi-user receivers".
  
Im Downlink von UMTS wird stets ein&nbsp; ''Single–User''–Empfänger verwendet, da in der Mobilstation eine gemeinsame Detektion aller Teilnehmer wegen der Vielzahl aktiver Teilnehmer sowie der Länge der Verwürfelungscodes und des asynchronen Betriebs zu aufwändig wäre.
+
In the UMTS downlink,&nbsp; it is always used a&nbsp; &raquo;'''single-user receiver'''&laquo;,&nbsp; since in the mobile station a joint detection of all subscribers would be too costly
 +
*due to the large number of active subscribers
  
Ein solcher Empfänger besteht aus einer Bank unabhängiger Korrelatoren. Jeder einzelne der insgesamt&nbsp; $J$&nbsp; Korrelatoren gehört zu einer spezifischen Spreizfolge. Die Korrelation wird meist in einer so genannten&nbsp; ''Korrelatordatenbank''&nbsp; softwaremäßig gebildet.  
+
*as well as the length of the scrambling codes and the asynchronous operation.
  
Dabei erhält man am Korrelatorausgang die Summe aus
 
*der&nbsp; ''Autokorrelationsfunktion''&nbsp; des Spreizcodes und
 
*der&nbsp; ''Kreuzkorrelationsfunktion''&nbsp; aller anderen Teilnehmer mit dem teilnehmereigenen Spreizcode.
 
  
 +
Such a receiver consists of a bank of independent correlators.
 +
*Each one of the total&nbsp; $J$&nbsp; correlators belongs to a specific spreading sequence.
  
[[File:P_ID1549__Bei_T_4_3_S6a_v1.png|right|frame|Single–User–Empfänger mit Matched–Filter '''Korrektur''']]
+
*The correlation is usually formed in a so-called&nbsp; "correlator database"&nbsp; by software.
Die Grafik zeigt die einfachste Realisierung eines solchen Empfängers mit Matched–Filter.
+
 
*Das Empfangssignal&nbsp; $r(t)$&nbsp; wird zunächst mit dem Spreizcode&nbsp; $c(t)$&nbsp; des betrachteten Teilnehmers multipliziert, was als&nbsp; ''Bandstauchung''&nbsp; oder&nbsp; ''Entspreizung''&nbsp; bezeichnet wird (gelbe Hinterlegung).
+
 
*Danach folgt die Faltung mit der Impulsantwort des Matched–Filters&nbsp; (''Root Raised Cosine''), um das SNR zu maximieren, und die Abtastung im Bittakt&nbsp; ( $T_{\rm B}$ ).
+
Thereby one receives at the correlator output the sum of
*Abschließend erfolgt die Schwellenwertentscheidung, die das Sinkensignal&nbsp; $v(t)$ und&nbsp; damit die Datenbit des betrachteten Teilnehmers liefert.
+
[[File:EN_Bei_T_4_3_S6a_v2.png|right|frame|Single-user receiver with matched filter]]
 +
*the&nbsp; "auto-correlation  function"&nbsp; of the spreading code and
 +
 
 +
*the&nbsp; "cross-correlation function"&nbsp; of all other users with their own spreading code.
 +
 
 +
 
 +
The graphic shows the simplest realization of such a receiver with matched filter.
 +
 
 +
#The received signal&nbsp; $r(t)$&nbsp; is first multiplied by the spreading code&nbsp; $c(t)$&nbsp; of the considered subscriber,&nbsp; which is called&nbsp; "despreading"&nbsp; $($yellow background$)$.
 +
#Followed by convolution with the matched filter impulse response&nbsp; $($"Root Raised Cosine"$)$&nbsp; to maximize SNR,&nbsp; and sampling in bit clock&nbsp; $(T_{\rm B})$.
 +
#Finally, the threshold decision is made, which provides the sink signal&nbsp; $v(t)$ and&nbsp; thus the data bits of the considered subscriber.
  
  
 
{{BlaueBox|TEXT=
 
{{BlaueBox|TEXT=
$\text{Bitte beachten Sie:}$&nbsp;
+
$\text{Please note:}$&nbsp;
  
Beim AWGN–Kanal haben die Bandspreizung beim Sender und die daran angepasste Bandstauchung beim Empfänger wegen&nbsp; $c(t)^2 = 1$&nbsp; keinen Einfluss auf die Bitfehlerwahrscheinlichkeit. Wie in&nbsp; [[Aufgaben:Aufgabe_4.5:_PN-Modulation| Aufgabe 4.5]]&nbsp; gezeigt wird, gilt auch mit Bandspreizung/Bandstauchung bei optimalem Empfänger unabhängig vom Spreizgrad&nbsp; $J$:
+
*For the AWGN channel,&nbsp; spreading at the transmitter and the matched despreading at the receiver have no effect on the bit error probability because of&nbsp; $c(t)^2 = 1$.&nbsp; As shown in&nbsp; [[Aufgaben:Exercise_4.5:_Pseudo_Noise_Modulation|$\text{Exercise 4.5}$]],&nbsp; even with spreading/despreading at the optimal receiver, regardless of spreading factor&nbsp; $J$:
  
:$$p_{\rm B} = {\rm Q} \left( \hspace{-0.05cm} \sqrt { {2 \cdot E_{\rm B} }/{N_{\rm 0} } } \hspace{0.05cm} \right )\hspace{0.05cm}. $$
+
:$$p_{\rm B} = {\rm Q} \left( \hspace{-0.05cm} \sqrt { {2 \cdot E_{\rm B} }/{N_{\rm 0} } } \hspace{0.05cm} \right )\hspace{0.05cm}. $$
 
   
 
   
Dieses Ergebnis lässt sich wie folgt begründen<br>Die statistischen Eigenschaften von weißem Rauschen&nbsp; $n(t)$&nbsp; werden durch die Multiplikation mit dem&nbsp; $±1$–Signal&nbsp; $c(t)$&nbsp; nicht verändert.}}
+
*This result can be justified as follows:&nbsp; The statistical properties of white noise&nbsp; $n(t)$&nbsp; are not changed by multiplication with the&nbsp; $±1$&nbsp; signal&nbsp; $c(t)$.}}
  
==Rake–Empfänger==
+
==Rake receiver==
 
<br>
 
<br>
Ein weiterer Empfänger für die Single–User–Detektion ist der&nbsp; '''Rake–Empfänger''', der bei einem Mehrwegekanal zu deutlichen Verbesserungen führt. Die Grafik zeigt seinen Aufbau für einen Zweiwegekanal mit&nbsp;
+
Another receiver for single-user detection is the&nbsp; &raquo;'''rake receiver'''&laquo;,&nbsp; which leads to significant improvements for a multipath channel.
[[File:P_ID1560__Bei_T_4_3_S6b_v1.png|right|frame|Struktur des Rake–Empfängers (Darstellung im äquivalenten Tiefpassbereich) '''Korrektur''']]
+
[[File:EN_Bei_T_4_3_S6b_v2.png|right|frame|Structure of the rake receiver&nbsp; $($shown in the equivalent low-pass domain$)$]]
*einem direkten Pfad mit Koeffizient&nbsp; $h_0$&nbsp; und Verzögerungszeit&nbsp; $τ_0$,
+
 
*einem Echo mit Koeffizient&nbsp; $h_1$&nbsp; und Verzögerungszeit&nbsp; $τ_1$.
+
The diagram shows its setup for a two-way channel with&nbsp;
 +
*a direct path with coefficient&nbsp; $h_0$&nbsp; and delay time&nbsp; $τ_0$,
 +
 
 +
*an echo with coefficient&nbsp; $h_1$&nbsp; and delay time&nbsp; $τ_1$.
  
  
Zur Vereinfachung werden hier die Koeffizienten&nbsp; $h_0$&nbsp; und&nbsp; $h_1$&nbsp; als reell angenommen. Aufgrund der Darstellung im äquivalenten Tiefpassbereich könnten diese auch komplex sein.
+
For simplicity,&nbsp; the coefficients&nbsp; $h_0$&nbsp; and&nbsp; $h_1$&nbsp; are assumed to be real.&nbsp; Due to the representation in the equivalent low-pass domain,&nbsp; these could also be complex.
  
Aufgabe des Rake–Empfängers ist es, die Signalenergien aller Pfade&nbsp; (in diesem Beispiel nur zwei)&nbsp; auf einen einzigen Zeitpunkt zu konzentrieren. Er arbeitet demnach wie eine&nbsp; ''Harke''&nbsp; für den Garten, was auch die deutsche Übersetzung für „Rake” ist.
+
&rArr; &nbsp; The task of the rake receiver is to concentrate the signal energies of all paths&nbsp; $($in this example only two$)$&nbsp; to a single instant.&nbsp; It works accordingly like a&nbsp; "rake"&nbsp; for the garden.
 
<br clear=all>
 
<br clear=all>
Legt man einen Diracimpuls zur Zeit&nbsp; $t = 0$&nbsp; an den Kanaleingang an, so gibt es am Ausgang des Rake–Empfängers drei Diracimpulse:  
+
If one applies a Dirac delta impulse at time&nbsp; $t = 0$&nbsp; to the channel input,&nbsp; there will be three Dirac delta impulses at the output of the rake receiver:  
 
:$$ s(t) = \delta(t) \hspace{0.3cm}\Rightarrow\hspace{0.3cm}
 
:$$ s(t) = \delta(t) \hspace{0.3cm}\Rightarrow\hspace{0.3cm}
 
y(t) = h_0 \cdot h_1 \cdot \delta(t - 2\tau_0) +  (h_0^2 + h_1^2) \cdot \delta(t - \tau_0 -  \tau_1)+
 
y(t) = h_0 \cdot h_1 \cdot \delta(t - 2\tau_0) +  (h_0^2 + h_1^2) \cdot \delta(t - \tau_0 -  \tau_1)+
 
h_0 \cdot h_1 \cdot \delta(t - 2\tau_1) .$$
 
h_0 \cdot h_1 \cdot \delta(t - 2\tau_1) .$$
 
    
 
    
*Die Signalenenergie konzentriert sich auf den Zeitpunkt&nbsp; $τ_0 + τ_1$. Von den insgesamt vier Wegen tragen zwei dazu bei (mittlerer Term).  
+
*The signal energy is concentrated at the time&nbsp; $τ_0 + τ_1$.&nbsp; Of the total four paths,&nbsp; two contribute&nbsp; $($middle term$)$.
*Die Diracfunktionen bei&nbsp; $2τ_0$&nbsp; und&nbsp; $2τ_1$&nbsp; bewirken zwar Impulsinterferenzen. Ihre Gewichte sind aber deutlich kleiner als die des Hauptpfades.
+
 +
*The Dirac delta functions at&nbsp; $2τ_0$&nbsp; and&nbsp; $2τ_1$&nbsp; do cause momentum interference.&nbsp; However, their&nbsp; weights are much smaller than those of the main path.
  
  
 
{{GraueBox|TEXT=
 
{{GraueBox|TEXT=
$\text{Beispiel 6:}$&nbsp;   
+
$\text{Example 6:}$&nbsp;   
Mit den Kanalparametern&nbsp; $h_0 = 0.8$&nbsp; und&nbsp; $h_1 = 0.6$&nbsp; beinhaltet der Hauptpfad $($mit Gewicht&nbsp; $h_0)$&nbsp; nur&nbsp; $0.82/(0.82 + 0.62) = 64\%$&nbsp; der gesamten Signalenergie. Mit Rake–Empfänger und den gleichen Gewichten lautet die obige Gleichung:
+
With channel parameters&nbsp; $h_0 = 0.8$&nbsp; and&nbsp; $h_1 = 0.6$&nbsp; the main path&nbsp; $($with weight&nbsp; $h_0)$&nbsp; contains&nbsp; $0.82/(0.82 + 0.62) = 64\%$&nbsp; of the total signal energy.  
 +
*With rake receiver and the same weights,&nbsp; the above equation is:
 
   
 
   
 
:$$ y(t) = 0.48  \cdot \delta(t - 2\tau_0) +  1.0 \cdot \delta(t - \tau_0 -  \tau_1)+
 
:$$ y(t) = 0.48  \cdot \delta(t - 2\tau_0) +  1.0 \cdot \delta(t - \tau_0 -  \tau_1)+
 
0.48 \cdot \delta(t - 2\tau_1) .$$
 
0.48 \cdot \delta(t - 2\tau_1) .$$
  
Der Anteil des Hauptpfades an der Gesamtenergie beträgt nun&nbsp; ${1^2}/{(1^2 + 0.48^2 + 0.48^2)} ≈ 68\%.$}}
+
*The share of the main path in the total energy  amounts in this simple example to&nbsp; ${1^2}/{(1^2 + 0.48^2 + 0.48^2)} ≈ 68\%.$}}
  
  
Rake–Empfänger werden zur Implementierung in mobilen Geräten bevorzugt, sind aber bei vielen aktiven Teilnehmern nur begrenzt leistungsfähig. Bei einem Mehrwegekanal mit vielen&nbsp; $(M)$&nbsp; Pfaden hat auch der Rake&nbsp; $M$&nbsp; Finger. Der Hauptfinger&nbsp; (''Main Finger'')&nbsp; – auch ''Searcher''&nbsp; genannt – ist dafür verantwortlich, die individuellen Pfade der Mehrfachausbreitung zu identifizieren und einzuordnen. Er sucht die stärksten Pfade und weist diese zusammen mit ihren Steuerinformationen anderen Fingern zu. Dabei wird die Zeit– und Frequenzsynchronisation aller Finger kontinuierlich mit den Kontrolldaten des empfangenen Signals verglichen.
+
Rake receivers are preferred for implementation in mobile devices,&nbsp; but have a limited performance when there are many active participants.  
 +
#In a multipath channel with many&nbsp; $(M)$&nbsp; paths,&nbsp; the Rake has also&nbsp; $M$&nbsp; fingers.  
 +
#The main finger&nbsp; &ndash; also called&nbsp; "searcher"&nbsp; &ndash;  is responsible for identifying and ranking the individual paths of multiple propagation.  
 +
#It searches for the strongest paths and assigns them to other fingers along with their control information.  
 +
#In the process,&nbsp; the time and frequency synchronization of all fingers is continuously compared with the control data of the received signal.
  
==Multi–User–Empfänger ==
+
==Multi-user receiver ==
 
<br>
 
<br>
Bei einem Single–User–Empfänger wird nur das Datensignal eines Teilnehmers entschieden, während alle anderen Teilnehmersignale als zusätzliches Rauschen betrachtet werden. Die Fehlerrate eines solchen Detektors wird jedoch dann sehr groß sein, wenn große&nbsp; ''Intrazellinterferenzen''&nbsp; (viele aktive Teilnehmer in der betrachteten Funkzelle) oder&nbsp; ''Interzellinterferenzen''&nbsp; (stark störende Teilnehmer in Nachbarzellen) vorliegen.
+
In a single-user receiver,&nbsp; only the data signal of one subscriber is decided,&nbsp; while all other subscriber signals are considered as additional noise.&nbsp; However,&nbsp; the bit error rate of such a detector will be very large
 +
*if there is large&nbsp; "intracell interference"&nbsp; $($many active subscribers in the considered radio cell$)$
  
Dagegen treffen&nbsp; '''Multi–User–Empfänger'''&nbsp; (Mehrbenutzerempfänger)&nbsp; eine gemeinsame Entscheidung für alle aktiven Teilnehmer. Deren Eigenschaften können wie folgt zusammengefasst werden:
+
*or large&nbsp; "intercell interference"&nbsp; $($highly interfering subscribers in neighboring cells$)$.
*Ein Multi–User–Empfänger betrachtet die Interferenzen anderer Teilnehmer nicht als Rauschen, sondern nutzt auch die in den Interferenzsignalen enthaltenen Informationen zur Detektion.
 
*Der Empfänger ist aufwändig zu realisieren und die Algorithmen sind äußerst rechenintensiv. Er beinhaltet eine extrem große Korrelatordatenbank gefolgt von einem gemeinsamen Detektor.
 
*Dem Multi–User–Empfänger müssen die Spreizcodes aller aktiven Teilnehmer bekannt sein. Diese Voraussetzung schließt einen Einsatz im UMTS–Downlink (also bei der Mobilstation) aus. Dagegen sind den Basisstationen alle teilnehmerspezifischen Spreizcodes a priori bekannt, so dass im Uplink die Mehrbenutzerdetektion auch tatsächlich zur Anwendung kommt.
 
*Manche Detektionsalgorithmen verlangen zusätzlich die Kenntnis anderer Signalparameter wie Energien und Verzögerungszeiten. Der gemeinsame Detektor – das Herzstück des Empfängers – ist dafür verantwortlich, den jeweiligen passenden Detektionsalgorithmus anzuwenden. Beispiele für die Mehrbenutzerdetektion sind&nbsp; ''Decorrelating Detection''&nbsp; und&nbsp; ''Interference Cancellation''.
 
  
  
==Near–Far–Effekt==   
+
In contrast,&nbsp; &raquo;'''multi-user receivers'''&laquo;&nbsp; make a joint decision for all active subscribers.&raquo; Their characteristics can be summarized as follows:
 +
#Such a multi-user receiver does not consider the interference from other participants as noise,&nbsp; but also uses the information contained in the interference signals for detection.
 +
#The receiver is expensive to implement and the algorithms are extremely computationally intensive.&nbsp; It contains an extremely large correlator database followed by a common detector.
 +
#The multi-user receiver must know the spreading codes of all active users.&nbsp; This requirement precludes use in the UMTS downlink&nbsp; $($i.e.,&nbsp; at the mobile station$)$.&nbsp; In contrast,&nbsp; all subscriber-specific spreading codes are known a-priori to the base stations,&nbsp; so that multi-user detection is only used in the uplink.
 +
#Some detection algorithms additionally require knowledge of other signal parameters such as energies and delay times.&nbsp; The common detector&nbsp; &ndash; the heart of the receiver &ndash;&nbsp; is responsible for applying the appropriate detection algorithm in each case.
 +
#Examples of multi-user detection are&nbsp; "decorrelating detection"&nbsp; and&nbsp; "Interference Cancellation".
 +
 
 +
 
 +
==Near–far problem==   
 
<br>
 
<br>
Der Near–Far–Effekt ist ausschließlich ein Problem des Uplinks, also der Übertragung von mobilen Teilnehmern zu einer Basisstation. Wir betrachten ein Szenario mit zwei unterschiedlich weit von der Basisstation entfernten Nutzern entsprechend folgender Grafik. Diese kann man wie folgt interpretieren:
+
The&nbsp; "near-far problem"&nbsp; is exclusively an uplink problem,&nbsp; i.e.,&nbsp; the transmission from mobile subscribers to a base station.&nbsp; We consider a scenario with two users at different distances from the base station according to the following graph.&nbsp; This can be interpreted as follows:
  
[[File:EN_Mob_T_3_2_S3.png|right|frame|Scenarios for the Near-Far effect|class=fit]]
+
[[File:EN_Mob_T_3_2_S3.png|right|frame|Scenarios for the near-far problem|class=fit]]
  
 +
#If both mobile stations transmit with the same power,&nbsp; the received power of the red user&nbsp; $\rm A$&nbsp; at the base station is significantly smaller than that of the blue user&nbsp; $\rm B$&nbsp; $($left scenario$)$ due to path loss.
 +
#In large macrocells,&nbsp; the difference can be as much as&nbsp; $100$&nbsp; dB.&nbsp; As a result,&nbsp; the red signal is largely obscured by the blue.
 +
#You can largely avoid the near-far problem if user&nbsp; $\rm A$&nbsp; transmits with higher power than user&nbsp; $\rm B$,&nbsp; as indicated in the right scenario.
 +
#Then,&nbsp; at the base station,&nbsp; the received power of both mobile stations is then&nbsp; $($almost$)$&nbsp; equal.
  
*Senden beide Mobilstationen mit gleicher Leistung, so ist die Empfangsleistung des roten Nutzers&nbsp; $\rm A$&nbsp; an der Basisstation aufgrund des Pfadverlustes deutlich kleiner als die des blauen Nutzers&nbsp; $\rm B$ (linkes Szenario). In großen Makrozellen kann der Unterschied bis zu&nbsp; $100 \ \rm dB$&nbsp; ausmachen. Dadurch wird das rote Signal weitgehend durch das blaue verdeckt.
 
  
 +
<u>Note:</u> &nbsp; In an idealized system&nbsp; $($one-way channel,&nbsp; ideal A/D converters,&nbsp; fully linear amplifiers$)$&nbsp; the transmitted data of the users are orthogonal to each other and one could detect the users individually even with very different received powers.&nbsp; This statement is true
 +
*for UMTS&nbsp; $($multiple access:&nbsp; CDMA$)$&nbsp; as well as
  
*Man kann den Near–Far–Effekt weitgehend vermeiden, wenn der weiter entfernte Nutzer&nbsp; $\rm A$&nbsp; mit höherer Leistung sendet als Nutzer&nbsp; $\rm B$, wie im rechten Szenario angedeutet. An der Basisstation ist dann die Empfangsleistung beider Mobilstationen (nahezu) gleich.
+
*for the 2G system GSM&nbsp; $($FDMA/TDMA$)$,&nbsp; and
<br clear=all>
+
 
''Anmerkung'': &nbsp; Bei einem idealisierten System&nbsp; (Einwegekanal, ideale A/D–Wandler, vollständig lineare Verstärker&nbsp;) sind die übertragenen Daten der Nutzer orthogonal zueinander und man könnte die Nutzer auch bei sehr unterschiedlichen Empfangsleistungen einzeln detektieren. Diese Aussage gilt für UMTS&nbsp; (Mehrfachzugriffsverfahren:&nbsp; CDMA)&nbsp; ebenso wie für für das 2G–System GSM&nbsp; (FDMA/TDMA)&nbsp; und für das 4G–System LTE&nbsp; (TDMA/OFDMA).
+
*for the 4G system LTE&nbsp; $($TDMA/OFDMA$)$.
 +
 
 +
 
 +
In reality,&nbsp; however,&nbsp; orthogonality is not always given due to the following reasons:
 +
#Different receive paths &nbsp; &nbsp; multipath channel,
 +
#non-ideal characteristics of the spreading and scrambling codes in CDMA,
 +
#asynchrony of users in the time domain&nbsp; $($basic propagation delay of paths$)$,
 +
#asynchrony of users in the frequency domain&nbsp; $($non-ideal oscillators and Doppler shift due to mobility of users$)$.
  
In der Realität ist jedoch die Orthogonalität aufgrund folgender Ursachen nicht immer gegeben:
 
*verschiedene Empfangspfade &nbsp; ⇒  &nbsp; Mehrwegekanal,
 
*nicht ideale Eigenschaften der Spreiz– und Scramblingcodes bei CDMA,
 
*Asynchronität der Nutzer im Zeitbereich&nbsp; (Grundlaufzeit der Pfade),
 
* Asynchronität der Nutzer im Frequenzbereich&nbsp; (nicht ideale Oszillatoren und Dopplerverschiebung aufgrund der Mobilität der Nutzer).
 
  
 +
Consequently,&nbsp; the users are no longer orthogonal to each other and the signal-to-noise ratio of the user to be detected with respect to the other users is not arbitrarily high:
 +
*For&nbsp; [[Examples_of_Communication_Systems/General_Description_of_GSM|"GSM"]]&nbsp; and&nbsp; [[Mobile_Communications/General_Information_on_the_LTE_Mobile_Communications_Standard|"LTE"]]&nbsp; one can assume signal-to-noise ratios of&nbsp; $25 $&nbsp; dB&nbsp; and more,
  
Folglich sind die Nutzer nicht mehr orthogonal zueinander und der Störabstand des zu detektierenden Nutzers gegenüber den anderen Teilnehmern ist nicht beliebig hoch. Bei&nbsp; [[Examples_of_Communication_Systems/Allgemeine_Beschreibung_von_GSM|GSM]]&nbsp; und&nbsp; [[Mobile_Communications/Allgemeines_zum_Mobilfunkstandard_LTE|LTE]]&nbsp; kann man von Störabständen von&nbsp; $25 \ \rm dB$&nbsp; und mehr ausgehen, bei UMTS (CDMA) jedoch nur von ca.&nbsp; $15 \ \rm dB$, bei hochratiger Datenübertragung eher noch von etwas weniger.
+
*but for UMTS&nbsp; $($CDMA$)$&nbsp; only approx.&nbsp; $15$&nbsp; dB,&nbsp; with high-rate data transmission rather less.
  
 
   
 
   
==Träger–zu–Interferenz–Leistungsverhältnis (CIR)==
+
==Carrier-to-interference power ratio==
 +
 
 +
The term&nbsp; "capacity"&nbsp; is generally understood to mean the number of available transmission channels per cell.&nbsp; However,&nbsp; since the number of subscribers is not strictly limited in UMTS unlike in GSM,&nbsp; no fixed capacity can be specified here.
 +
*In perfect codes,&nbsp; the subscribers do not interfere with each other.&nbsp; Thus,&nbsp; the maximum number of users is determined solely by the spreading factor&nbsp; $J$&nbsp; and the available number of mutually orthogonal codes,&nbsp; which,&nbsp; however,&nbsp; is also limited.
 +
 
 +
*More practical are non-perfect,&nbsp; only quasi-orthogonal codes.&nbsp; Here,&nbsp; the&nbsp; "capacity"&nbsp; of a radio cell is predominantly determined by the resulting interference or the&nbsp; "carrier-to-interference power ratio"&nbsp; $\rm (CIR)$.
 +
 
  
Unter&nbsp; '''Kapazität'''&nbsp; wird allgemein die Anzahl der verfügbaren Übertragungskanäle pro Zelle verstanden werden. Da aber bei UMTS die Teilnehmerzahl im Gegensatz zum GSM nicht streng begrenzt ist, lässt sich hier keine feste Kapazität angeben.
+
[[File:EN_Bei_T_4_3_S8_v2.png|right|frame|Carrier-to-interference ratio depending on the number of subscribers]]
*Bei perfekten Codes stören sich die Teilnehmer gegenseitig nicht. Dadurch wird die maximale Nutzerzahl allein durch den Spreizfaktor&nbsp; $J$&nbsp; und die verfügbare Anzahl der zueinander orthogonalen Codes bestimmt, die aber ebenfalls limitiert ist.
 
*Praxisnäher sind nichtperfekte, nur quasi–orthogonale Codes. Hier wird die „Kapazität” einer Funkzelle vorwiegend durch die entstehenden Interferenzen bzw. das&nbsp; ''Träger–zu–Interferenz–Leistungsverhältnis''&nbsp; (englisch:&nbsp; ''Carrier–to–Interference Ratio'', CIR) bestimmt.
 
  
 +
As can be seen from this graph,&nbsp; CIR depends directly on the number of active participants.&nbsp; The more active subscribers there are,&nbsp; the more interference power is generated and the smaller the CIR becomes.
  
[[File:EN_Bei_T_4_3_S8_v2.png|right|frame|''Carrier–to–Interference Ratio''&nbsp; ("CIR") abhängig von der Teilnehmerzahl]]
+
Furthermore,&nbsp; this  decisive criterion for UMTS also depends on the following variables:
<br><br><br><br>Wie aus dieser Grafik zu ersehen ist, hängt das "CIR" direkt von der Anzahl aktiver Teilnehmer ab. Je mehr Teilnehmer aktiv sind, desto mehr Interferenzleistung entsteht und desto kleiner wird das "CIR".  
+
#The topology and user behavior&nbsp; $($number of services called up$)$,
 +
#the spreading factor&nbsp; $J$&nbsp; and the orthogonality of the used spreading code.
  
Desweiteren hängt dieses für UMTS entscheidende Kriterium auch von folgenden Größen ab:
+
 
*der Topologie und dem Nutzerverhalten (aufgerufene Dienste),
+
In order to limit the disturbing influence of the interference power on the transmission quality,&nbsp; there are two possible criteria:
*dem Spreizfaktor&nbsp; $J$&nbsp; und der Orthogonalität des verwendeten Spreizcodes.
+
 
<br clear=all>
+
&raquo;'''Cell breathing'''&laquo;: &nbsp; If the number of active subscribers increases significantly with UMTS,&nbsp; the cell radius is reduced and&nbsp; $($because of the now fewer subscribers in the cell$)$&nbsp; also the current interference power is lower.&nbsp; A less loaded neighboring cell then steps in to supply the subscribers at the edge of the reduced cell.
Um den störenden Einfluss der Interferenzleistung auf die Übertragungsqualität zu begrenzen, gibt es zwei Möglichkriten:
+
 
*'''Zellatmung''':&nbsp; Nimmt bei UMTS die Anzahl der aktiven Teilnehmer signifikant zu, so wird der Zellenradius verkleinert und&nbsp; (wegen der nun weniger Teilnehmer in der Zelle)&nbsp; auch die aktuelle Interferenzleistung geringer. Für die Versorgung der Teilnehmer am Rande der verkleinerten Zelle springt dann eine weniger belastete Nachbarzelle ein.
+
&raquo;'''Power control'''&laquo;: &nbsp; If the total interference power within a radio cell exceeds a specified limit,&nbsp; the transmission power of all subscribers is reduced accordingly and/or the data rate is reduced,&nbsp; resulting in poorer transmission quality for all.&nbsp; More about this in the next section.
*'''Leistungsregelung''':&nbsp; Überschreitet die Gesamtinterferenzleistung innerhalb einer Funkzelle einen vorgegebenen Grenzwert, so wird die Sendeleistung aller Teilnehmer entsprechend herabgesetzt und/oder die Datenrate reduziert, was eine schlechtere Übertragungsqualität für alle zur Folge hat. Hierzu mehr auf der nächsten Seite.
 
  
 
   
 
   
 
==Power and power control in UMTS== 
 
==Power and power control in UMTS== 
 
<br>
 
<br>
Als Regelgröße bei der Leistungsregelung in UMTS wird das Verhältnis zwischen der Signalleistung und der Interferenzleistung verwendet. Dabei gibt es Unterschiede zwischen dem FDD– und TDD–Modus.
+
The ratio between the signal power and the interference power is used as the controlled variable for power control in UMTS.&nbsp; There are differences between the&nbsp; "frequency division duplex"&nbsp; $\rm (FDD)$&nbsp; and the&nbsp; "time division duplex"&nbsp; $\rm (TDD)$&nbsp; modes.
  
[[File:EN_Bei_T_4_3_S10a.png|right|frame|Leistungsregelung im FDD–Modus]]
+
[[File:EN_Bei_T_4_3_S10a_neu.png|right|frame|Power control in the FDD mode]]
Wir betrachten die&nbsp; ''Leistungsregelung im FDD–Modus''&nbsp; genauer. In der Grafik erkennt man zwei verschiedene Regelkreise:
+
We take a closer look at the FDD power control.&nbsp; In the diagram you can see two different control loops:
*Der&nbsp; '''innere Regelkreis'''&nbsp; steuert die Sendeleistung auf der Basis von Zeitschlitzen, wobei in jedem Zeitschlitz ein Leistungskommando übertragen wird. Die Leistung des Senders wird mit Hilfe der CIR–Schätzungen im Empfänger und den Vorgaben des&nbsp; ''Radio Network Controllers''&nbsp; (RNC) aus dem äußeren Regelkreis bestimmt und verändert.
+
*The&nbsp; &raquo;'''inner control loop'''&laquo;&nbsp; controls the transmitter power based on time slots,&nbsp; where one power command is transmitted in each time slot.&nbsp;  
*Der&nbsp; '''äußere Regelkreis'''&nbsp; regelt auf Basis von Rahmen mit $10$ Millisekunden Dauer. Er wird im RNC realisiert und ist dafür zuständig, den Soll–Wert für den inneren Regelkreis zu bestimmen.
 
  
 +
::The power of the transmitter is determined and changed using the CIR estimates in the receiver and the specifications of the&nbsp; "radio network controller"&nbsp; $\rm (RNC)$&nbsp; from the outer control loop.
  
Der Ablauf der FDD–Leistungsregelung sieht folgendermaßen aus:
+
*The&nbsp; '''outer loop'''&nbsp; controls based on $10$ millisecond duration frames. It is implemented in the RNC and is responsible for determining the set point for the inner loop.
*Der RNC gibt einen Sollwert für das Träger–zu–Interferenz–Verhältnis (CIR–Sollwert) vor.
 
*Der Empfänger schätzt den CIR–Istwert und generiert Steuerkommandos für den Sender.
 
*Der Sender ändert entsprechend dieser Steuerkommandos die Sendeleistung.
 
  
  
Das Prinzip der&nbsp; ''Leistungsregelung im TDD–Modus''&nbsp; ähnelt der oben vorgestellten Regelung für den FDD–Modus, in der Abwärtsrichtung sind sie sogar praktisch identisch.
+
The FDD power control sequence is as follows:
 +
#The RNC provides a carrier-to-interference ratio&nbsp; $\rm (CIR)$&nbsp; setpoint.
 +
#The receiver estimates the actual CIR value and generates control commands for the transmitter.
 +
#The transmitter changes the transmitted power according to these control commands.
 +
 
 +
 
 +
The principle of&nbsp; "power control in TDD mode"&nbsp; is similar to the control presented here for the FDD mode.&nbsp; In fact in the downlink direction they are practically identical.
  
  
 
{{BlaueBox|TEXT=
 
{{BlaueBox|TEXT=
$\text{Fazit:}$&nbsp;  
+
$\text{Conclusion:}$&nbsp;  
Die&nbsp; '''TDD–Leistungsregelung'''&nbsp; ist viel langsamer und dadurch auch unpräziser als bei&nbsp; '''FDD'''. Eine schnelle Leistungsregelung ist in diesem Fall aber auch gar nicht möglich, da jeder Teilnehmer jeweils nur einen Bruchteil des Zeitrahmens zur Verfügung hat.}}
+
The&nbsp; &raquo;'''TDD power control'''&laquo;&nbsp; is much slower and thus less precise than in&nbsp; FDD.  
 +
 
 +
*However,&nbsp; fast power control is not even possible in this case,&nbsp; since each participant has only a fraction of the time frame available for him.}}
  
 
   
 
   
==Link–Budget ==
+
==Link budget ==
 
<br>
 
<br>
Bei der Planung von UMTS–Netzen ist die Berechnung des Link-Budgets ein wichtiger Schritt. Die Kenntnis des Link–Budgets ist sowohl bei der Dimensionierung der Versorgungsgebiete als auch für die Bestimmung der Kapazität und der Dienstgüte–Anforderungen erforderlich.  
+
When planning UMTS networks,&nbsp; calculating the link budget is an important step.&nbsp; Knowledge of the link budget is required both for dimensioning the coverage areas and for determining the capacity and quality of service requirements.  
  
 
{{BlaueBox|TEXT=
 
{{BlaueBox|TEXT=
'''Ziel des Link–Budgets'''&nbsp; ist die Berechnung der&nbsp; '''maximalen Zellgröße'''&nbsp; unter Berücksichtigung folgender Kriterien:
+
The&nbsp; &raquo;'''objective of the link budget'''&laquo;&nbsp; is to calculate the&nbsp; '''maximum cell size'''&nbsp; considering the following criteria:
*Art und Datenrate der Services,
+
#Type and data rate of the services,
*Topologie der Umgebung,
+
#topology of the environment,
*Systemkonfiguration (Lage und Leistung der Basisstationen, Handover–Gewinn),
+
#system configuration&nbsp; $($location and power of base stations,&nbsp; handover gain$)$,
*Service–Anforderungen (Verfügbarkeit),
+
#service requirements&nbsp; $($availability$)$,
*Art der Mobilstation (Geschwindigkeit, Leistung),
+
#type of mobile station&nbsp; $($speed,&nbsp; power$)$,
*finanzielle und wirtschaftliche Aspekte.}}
+
#financial and economic aspects}}
  
  
[[File:EN_Bei_T_4_3_S9.png|right|frame|Budget für einen Sprachübertragungskanal]]
 
 
{{GraueBox|TEXT=
 
{{GraueBox|TEXT=
$\text{Beispiel 7:}$&nbsp;   
+
[[File:EN_Bei_T_4_3_S9_v222.png|right|frame|Budget for a voice transmission channel]]
Die Berechnung des Link–Budgets wird am Beispiel eines Sprachübertragungskanals im UMTS–Downlink dargestellt. Zu den beispielhaften Zahlenwerten ist zu bemerken:
+
$\text{Example 7:}$&nbsp;   
*Die Sendeleistung  betrage&nbsp; $P_{\rm S} =19 \ \rm dBm$, was ca.&nbsp; $79 \ \rm mW$&nbsp; entspricht. Hierbei ist der Antennenverlust mit&nbsp; $2\ \rm  dB$&nbsp; berücksichtigt.
+
The calculation of the link budget is illustrated using the example of a voice transmission channel in the UMTS downlink.&nbsp; Regarding the exemplary numerical values,&nbsp; it should be noted:
*Die Rauschleistung&nbsp; $P_{\rm R} = 5 · 10^{-11} \ \rm mW$&nbsp; ist das Produkt aus UMTS–Bandbreite und Rauschleistungsdichte &nbsp; <br>&nbsp; &rArr; &nbsp; $P_{\rm R} = -103 \ \rm dBm $.
+
#The transmitted power is&nbsp; $P_{\rm S} =19$&nbsp; dBm,&nbsp; which corresponds to approx.&nbsp; $79$&nbsp; mW.&nbsp; <br>Here,&nbsp; the antenna loss is considered to be&nbsp; $2$&nbsp; dB.
*Die Interferenzleistung ist&nbsp; $P_{\rm I} = –99\ \rm  dBm$&nbsp; entsprechend&nbsp; $1.25 · 10^{-10} \ \rm mW$.  
+
#The noise power&nbsp; $P_{\rm N} = 5 \cdot 10^{-11}$&nbsp; mW &nbsp; &rArr; &nbsp; $P_{\rm N} = -103$&nbsp;  dBm &nbsp; <br>product of UMTS bandwidth and noise power density.
*Damit ergibt sich die gesamte Störleistung zu&nbsp; $P_{\rm R+I} = P_{\rm R} + P_{\rm I} = 1.25 · 10^{-10} \ \rm mW$&nbsp; <br>&nbsp; &rArr; &nbsp; $P_{\rm R+I} =- 97.5\ \rm dBm$.
+
#The interference power is&nbsp; $P_{\rm I} = -99$&nbsp; dBm&nbsp; corresponding to&nbsp; $1.25 \cdot 10^{-10}$&nbsp;  mW.  
*Die Antennenempfindlichkeit ergibt sich zu&nbsp; $-97.5 - 27 + 5 - 17 + 3.5 = - 133 \ \rm dBm$. Ein großer negativer Wert ist hierbei „gut”.
+
#This gives the total interference power&nbsp; $P_{\rm N+I} = P_{\rm N} + P_{\rm I} = 1.25 \cdot 10^{-10}$&nbsp; mW &nbsp; &rArr; &nbsp; $P_{\rm N+I} =- 97.5$&nbsp; dBm.
*Der maximal zulässige Pfadverlust soll  möglichst groß sein. Man erhält im Beispiel&nbsp; $19 - (-133) = 152 \ \rm  dB$.
+
#The antenna sensitivity results in $-97.5 - 27 + 5 - 17 + 3.5 = - 133$&nbsp; dBm.&nbsp; <br>A large negative value here is&nbsp; "good".
*Das&nbsp; '''Link–Budget'''&nbsp; beinhaltet den Margin für Fading und den Handover–Gewinn und beträgt im Beispiel&nbsp; $140 \ \rm  dB$.
+
#Maximum allowable path loss should be as large as possible.&nbsp; Here&nbsp; $19 - (-133) = 152$&nbsp; dB.
*Der '''maximale Zellradius''' lässt sich aus dem Link–Budget mit einer [https://en.wikipedia.org/wiki/Path_loss empirischen Formel] von Okumura–Hata bestimmen. Es gilt:
+
#The&nbsp; '''link budget'''&nbsp; includes the margin for fading and the handover gain,&nbsp; in the example:&nbsp;  $140$&nbsp; dB.
:$$ {r}\ [{\rm km}] = 10^{({\rm LinkBudget}- 137)/35}= 10^{0.0857}\approx 1.22 . $$
+
#The&nbsp; '''maximum cell radius'''&nbsp; can be determined from the link budget using an [https://en.wikipedia.org/wiki/Path_loss &raquo;empirical formula&laquo;] of Okumura-Hata.&nbsp; It holds:&nbsp; $ {r}\ [{\rm km}] = 10^{({\rm LinkBudget}- 137)/35}= 10^{0.0857}\approx 1.22 . $
 +
<br><br><br><br><br>
 +
 
 +
<u>Notes:</u> &nbsp;
 +
*"$\rm dB$"&nbsp; denotes a logarithmic power specification,&nbsp; referenced to&nbsp; $1 \rm W$.
  
''Anmerkungen'': &nbsp;
+
*In contrast&nbsp; "$\rm dBm$"&nbsp; refers to the power&nbsp; $1 \rm mW$.}}
*Die Angabe&nbsp; $\rm dB$&nbsp; kennzeichnet eine logarithmische Leistungsangabe, bezogen auf&nbsp; $1 \ \rm W$.
 
* Dagegen bezieht sich&nbsp; $\rm dBm$&nbsp; auf die Leistung&nbsp; $1 \ \rm mW$.}}
 
  
==UMTS–Funkressourcenverwaltung ==
+
==UMTS radio resource management ==
 
<br>
 
<br>
Zentrale Aufgabe der&nbsp; '''Funkressourcenverwaltung'''&nbsp; (englisch:&nbsp; ''Radio Resource Management'', RRM) ist die dynamische Anpassung der Funkübertragungsparameter an die aktuelle Situation&nbsp; (Fading, Bewegung der Mobilstation, Auslastung, usw.)&nbsp; mit dem Ziel,
+
The central task of&nbsp; &raquo;'''radio resource management'''&laquo;&nbsp; $\rm (RRM)$&nbsp; is the dynamic adaptation of radio transmission parameters to the current situation&nbsp; $($fading,&nbsp; mobile station movement,&nbsp; load,&nbsp; etc.$)$&nbsp; with the aim to
 
[[File:EN_Bei_T_4_3_S11.png|right|frame|Radio Resource Management in UMTS]]
 
[[File:EN_Bei_T_4_3_S11.png|right|frame|Radio Resource Management in UMTS]]
*die Übertragungs– und Teilnehmerkapazitäten zu steigern,
+
#increase the transmission and subscriber capacity,
*die individuelle Übertragungsqualität zu verbessern und
+
#improve the individual transmission quality,&nbsp; and
*die vorhandenen Funkressourcen ökonomisch zu nutzen.
+
#use existing radio resources economically.
  
  
Nachfolgend werden die im Schaubild zusammengestellten wichtigsten RRM–Mechanismen erläutert.
+
The main RRM mechanisms summarized in the diagram are explained below.
  
'''Sendeleistungsregelung'''
+
'''Transmit power control'''<br>The&nbsp; radio resource management&nbsp; attempts to keep the received power and thus the carrier-to-interference ratio&nbsp; $\rm (CIR)$&nbsp; at the receiver constant,&nbsp; or at least to prevent it from falling below a specified limit.  
<br>Das&nbsp; ''Radio Resource Management''&nbsp; versucht, die Empfangsleistung und damit das Träger–zu–Interferenz–Verhältnis (CIR) am Empfänger konstant zu halten oder zumindest zu vermeiden, dass ein vorgegebener Grenzwert unterschritten wird.  
 
  
Ein Beispiel für die Notwendigkeit der Leistungsregelung ist der&nbsp; [[Examples_of_Communication_Systems/Nachrichtentechnische_Aspekte_von_UMTS#Near.E2.80.93Far.E2.80.93Effekt|Near–Far–Effekt]], der bekanntlich zu einem Verbindungsabbruch führen kann.
+
An example of the need for power control is the&nbsp; [[Examples_of_Communication_Systems/Telecommunications_Aspects_of_UMTS#Near.E2.80.93far_problem|"near-far problem"]],&nbsp; which is known to cause a disconnect.
  
Die Schrittweite der Leistungsregelung beträgt&nbsp; $1 \ \rm dB$&nbsp; oder&nbsp; $2 \ \rm dB$, die Frequenz der Regelungskommandos ist&nbsp; $1500$&nbsp; Kommandos pro Sekunde.
+
The step size of the power control is&nbsp; $1 \ \rm dB$&nbsp; or&nbsp; $2 \ \rm dB$,&nbsp; and the frequency of the control commands is&nbsp; $1500$&nbsp; commands per second.
  
'''Regelung der Datenrate'''  
+
'''Regulation of data rate'''  
<br>Bei UMTS ist ein Austausch zwischen Datenrate und Übertragungsqualität möglich, die sich über die Wahl des Spreizfaktors realisieren lässt. Eine Verdopplung des Spreizfaktors entspricht hierbei einer Halbierung der Datenrate und erhöht die Qualität um&nbsp; $3\ \rm dB$&nbsp; (Spreizgewinn).
+
<br>UMTS allows an exchange between data rate and transmission quality,&nbsp; which can be realized by selecting the spreading factor.&nbsp; Doubling the spreading factor corresponds to halving the data rate and increases the quality by&nbsp; $3\ \rm dB$&nbsp; $($spreading gain$)$.
  
'''Zugangskontrolle'''  
+
'''Access control'''  
<br>Um Überlastsituationen im Gesamtnetz zu vermeiden, wird vor dem Aufbau einer neuen Verbindung überprüft, ob die notwendigen Ressourcen vorhanden sind. Andernfalls wird die neue Verbindung abgewiesen. Diese Überprüfung wird durch Abschätzung der Sendeleistungsverteilung nach der Aufnahme der neuen Verbindung realisiert.
+
<br>To avoid overload situations in the overall network,&nbsp; a check is made before a new connection is established to see whether the necessary resources are available.&nbsp; If not,&nbsp; the new connection is rejected.&nbsp; This check is realized by estimating the transmission power distribution after the new connection is established.
  
'''Lastregelung'''  
+
'''Load control'''  
<br>Diese wird aktiv, wenn trotz Zugangskontrolle eine Überlast auftritt. In diesem Fall wird ein Handover zu einem anderen "Node B" initiiert und – falls dies nicht möglich ist – werden die Datenraten bestimmter Teilnehmer gesenkt.
+
<br>This becomes active if an overload occurs despite access control.&nbsp; In this case,&nbsp; a handover to another&nbsp; base station is initiated and&nbsp; &ndash; if this is not possible &ndash;&nbsp; the data rates of certain nodes are lowered.
  
 
'''Handover'''  
 
'''Handover'''  
<br>Die Funkressourcenverwaltung ist schließlich auch für das Handover verantwortlich, um unterbrechungsfreie Verbindungen zu gewährleisten. Die Zuordnung der Mobilstationen zu den einzelnen Funkzellen erfolgt auf Grundlage von CIR–Messungen.
+
<br>Finally,&nbsp; radio resource management is also responsible for handover to ensure uninterrupted connections.&nbsp; Mobile stations are assigned to the individual radio cells on the basis of CIR measurements.
  
 
   
 
   
==Aufgaben zum Kapitel ==  
+
==Exercises for the chapter ==  
 
<br>
 
<br>
[[Aufgaben:Aufgabe_4.5:_Pseudo_Noise-Modulation|Aufgabe 4.5: Pseudo Noise-Modulation]]
+
[[Aufgaben:Exercise_4.5:_Pseudo_Noise_Modulation|Exercise 4.5: Pseudo Noise Modulation]]
  
[[Aufgaben:Aufgabe_4.5Z:_Zur_Bandspreizung_bei_UMTS|Aufgabe 4.5Z: Zur Bandspreizung bei UMTS]]
+
[[Aufgaben:Exercise_4.5Z:_About_Band_Spreading_with_UMTS|Exercise 4.5Z: About Spread Spectrum with UMTS]]
  
[[Aufgaben:Aufgabe_4.6:_OVSF-Codes|Aufgabe 4.6: OVSF-Codes]]
+
[[Aufgaben:Exercise_4.6:_OVSF_Codes|Exercise 4.6: OVSF Codes]]
  
[[Aufgaben:Aufgabe_4.7:_Zum_Rake-Empfänger|Aufgabe 4.7: Zum Rake-Empfänger]]
+
[[Aufgaben:Exercise_4.7:_To_the_Rake_Receiver|Exercise 4.7: About the Rake Receiver]]
  
==Quellenverzeichnis==
+
==Sources==
 
<references />
 
<references />
  
  
 
{{Display}}
 
{{Display}}

Latest revision as of 17:15, 20 April 2023

Improvements regarding speech coding


In the chapter  "Global System for Mobile Communications"  $\rm (GSM)$  of this book,  several speech codecs have already been described in detail.

$\text{Reminder:}$  A speech codec is used to reduce the data rate of a digitized speech or music signal.

  1. In the process,  redundancy and irrelevance are removed from the original signal.
  2. The artificial word  "codec"  indicates that the same functional unit is used for both,  encoding and decoding.


Among others,  the  "Adaptive Multi-Rate Codec"  $\rm (AMR)$  based on  $\rm ACELP$  $($"Algebraic Code Excited Linear Prediction"$)$  was introduced, 

  • which in the frequency range from  $\text{300 Hz}$  to  $\text{3400 Hz}$ 
  • dynamically switches between eight different modes  $($single codecs$)$ 
  • of different data rate in the range of  $\text{4. 75 kbit/s}$  to  $\text{12.2 kbit/s}$.


These codecs are also supported in UMTS Release 99 and Release 4.  Compared to the earlier speech codecs  $($Full Rate,  Half Rate, Enhanced Full Rate Vocoder$)$,  they allow

  1. independence from channel conditions and network load,
  2. the ability to adapt data rates to conditions,
  3. improved flexible error protection in the event of more severe radio interference, and
  4. thereby providing better overall voice quality.


Composition of wideband AMR modes

In 2001,  the  "3rd Generation Partnership Project"  $\text{(3gpp)}$  and the  "International Telecommuncation Union"    $\text{(ITU)}$  specified the new voice codec  »Wideband AMR«  for UMTS Release 5.  This is a further development of AMR and offers

  • an extended bandwidth from  $\text{50 Hz}$  to  $\text{7 kHz}$ 
    $($sampling frequency  $\text{16 kHz})$,
  • a total of nine modes between  $\text{6.6 kbit/s}$  and  $\text{23.85 kbit/s}$ 
    $($of which only five modes are used$)$,  and
  • improved voice quality and better (more natural) sound.



$\text{Some features of wideband AMR}$ 

  1. Speech data is delivered to the codec as PCM encoded speech with  $16\hspace{0.05cm}000$  samples per second. 
  2. The speech coding is done in blocks of  $\text{20 ms}$  and the data rate is adjusted every  $\text{20 ms}$.
  3. The band  $\text{(50 Hz}$  to  $\text{7000 Hz})$  is divided into two sub-bands,  which are encoded differently to allocate more bits to the subjectively important frequencies.
  4. The upper band  $\text{(6400 Hz}$  to  $\text{7000 Hz})$  is transmitted only in the highest mode $($with  $\text{23.85 kbit/s)}$ .
  5. In all other modes,  only frequencies  $\text{50 Hz}$  to  $\text{6400 Hz}$  are considered in encoding.
  6. Wideband AMR supports  "discontinuous transmission"'  $\rm (DTX)$.  This feature means that transmission is paused during voice pauses,  reducing both mobile station power consumption and overall interference at the air interface.  This process is also known as  "Source-Controlled Rate"  $\rm (SCR)$.
  7. The  "Voice Activity Detection"  $\rm (VAD)$  determines whether speech is in progress or not and inserts a  "silence descriptor frame"  during speech pauses.
  8. The subscriber is suggested the feeling of a continuous connection by the decoder inserting synthetically generated  "comfort noise"  during speech pauses.


Application of the CDMA method to UMTS


UMTS uses the multiple access method  "Direct Sequence Code Division Multiple Access"  $\rm (DS-CDMA)$,  which has already been discussed in the  "PN modulation"  chapter  of the book "Modulation Methods".

Here follows a brief summary of this method according to the diagram describing such a system in the equivalent low-pass range and highly simplified:

CDMA transmission system for two subscribers
  • The two data signals  $q_1(t)$  and  $q_2(t)$  are supposed to use the same channel without interfering with each other. The bit duration of each is  $T_{\rm B}$.
  • Each of the data signals is multiplied by an associated spreading code   $c_1(t)$  resp.  $c_2(t)$.
  • The sum signal  $s(t) = q_1(t) · c_1(t) + q_2(t) · c_2(t)$  is formed and transmitted.
  • At the receiver,  the same spreading codes  $c_1(t)$  resp.  $c_2(t)$  are added,  thus separating the signals again.
  • Assuming orthogonal spreading codes and a small AWGN noise,  the two reconstructed signals at the receiver output are:
$$v_1(t) = q_1(t) \ \text{and} \ v_2(t) = q_2(t).$$
  • For AWGN noise signal  $n(t)$  and orthogonal spreading codes,  this does not change the error probability due to other participants.


$\text{Example1:}$  The upper graph shows three data bits   $(+1, -1, +1)$   of the rectangular signal  $q_1(t)$  from subscriber 1,  each with symbol duration  $T_{\rm B}$.

Signals at  "Direct–Sequence Spread Spectrum"
  • Here,  the symbol duration  $T_{\rm C}$  of the spreading code  $c_1(t)$   ⇒   also called  "chip duration"  is smaller by a factor  $4$.
  • The multiplication  $s_1(t) = q_1(t) · c_1(t)$  results in a chip sequence of length  $12 · T_{\rm C}$.


One recognizes from this sketch that the signal  $s_1(t)$  is of higher frequency than  $q_1(t)$.


  • This is why this modulation method is often also called  "spread spectrum".
  • The CDMA receiver reverses this  "spreading".  We refer to this  "receiver-side spreading"  as  "despreading".


$\text{Summarizing:}$   By applying  "Direct Sequence Code Division Multiple Access"  $\rm (DS-CDMA)$  to a data bit sequence  $q(t)$ 

  • increases the bandwidth of  $s(t) = q(t) \cdot c(t)$  by the  »spreading factor«   $J = T_{\rm B}/T_{\rm C}$  –  this is equal to the number of  "chips per bit";
  • the chip rate  $R_{\rm C}$  is greater than the bit rate  $R_{\rm B}$ by a factor  $J$;
  • the bandwidth of the entire CDMA signal is greater than the bandwidth of each user by a factor  $J$.


That is:    $\text{In UMTS, the entire bandwidth is available to each subscriber for the entire transmission duration}$.

Recall:  In GSM,  both  "Frequency Division Multiple Access"  and  "Time Division Multiple Access"  are used as multiple access methods.

  • Here,  each subscriber has only a limited frequency band  $\rm (FDMA)$,  and
  • he only has access to the channel within time slots  $\rm (TDMA)$.


Spreading codes and scrambling with UMTS


The spreading codes for UMTS should

  • be orthogonal to each other to avoid mutual interference between subscribers,
  • allow a flexible realization of different spreading factors  $J$.


⇒   The issue presented here is also illustrated by the German-language SWF applet  "OVSF codes".

$\text{Example 2:}$  An example of this is the   »orthogonal variable spreading factor«   $\rm (OVSF)$,  which provide codes of lengths from  $J = 4$  to  $J = 512$.

Chart on the OVSF code family

These can be created using a code tree, as shown in the diagram.  Here,  at each branch,  two new codes are created from one code  $C$:

  1.   $(+C \ +\hspace{-0.1cm}C)$,  and
  2.   $(+C \ -\hspace{-0.1cm}C)$.

Note that no predecessor and successor of a code may be used.

  • So in the drawn example,  eight spreading codes with spreading factor  $J = 8$  could be used.
  • Also possible are the four codes with yellow background
  • once with  $J = 2$,
  • once with  $J = 4$,  and
  • twice with  $J = 8$.


But the lower four codes with spreading factor  $J = 8$  cannot be used,  because they all start with  "$+1 \ -\hspace{-0.1cm}1$",  which is already occupied by the OVSF code with spreading factor  $J = 2$.


Additional scrambling after spreading

To obtain more spreading codes and thus be able to supply more participants,  after the band spreading with  $c(t)$ 

  • the sequence is chip-wise scrambled again with  $w(t)$, 
  • without any further spreading.


The  »scrambling code« $w(t)$  has same length and rate as the spreading code  $c(t)$.

Typical spreading and scrambling codes for UMTS

⇒   Scrambling causes the codes to lose their complete orthogonality; they are called  "quasi-orthogonal".

  • But they are characterized by a pronounced  "auto-correlation function"  $\rm (ACF)$  around zero,  which facilitates detection at the receiver.
  • Using quasi-orthogonal codes makes sense because the set of orthogonal codes is limited and scrambling allows also different users to use the same spreading codes.


The table summarizes some data of spreading and scrambling codes.

Generator for creating Gold codes

$\text{Example 3:}$  In UMTS, so-called  »Gold codes«  are used for scrambling. The graphic from  [3gpp][1]  shows the block diagram for the circuitry generation of such codes.

  • Two different pseudonoise sequences of equal length  $($here:  $N = 18)$  are first generated in parallel using shift registers and added bitwise using  "exclusive-or"  gates.
  • In the uplink,  each mobile station has its own scrambling code and the separation of each channel is done using the same code.
  • In contrast,  in the downlink, each coverage area of a  "Node B"  has a common scrambling code.


Channel coding for UMTS


With UMTS,  the EFR- and AMR-encoded voice data pass through a two-stage error protection  $($similar to GSM$)$,  consisting of

Insertion of CRC bits and tailbits in UMTS
  1. formation of  "cyclic redundancy check bits"  $\rm (CRC)$,
  2. subsequent convolutional encoding.


However,  these methods differ from those used for GSM in that they are more flexible,  since for UMTS they have to take different data rates into account.

⇒   For  »error detection«,  eight, twelve, sixteen or  $24$  CRC bits are formed depending on the size of the transport block  $\text{(10 ms}$  or  $\text{20 ms})$,  and appended to it.

  • Eight tail bits are also inserted at the end of each frame for synchronization purposes.
  • The diagram shows a transport block of the DCH channel with  $164$  user data bits,  to which  $16$  CRC bits and eight tail bits are appended.


⇒   For  »error correction«,  UMTS uses two different methods,  depending on the data rate:

  • For low data rates,  "convolutional codes"  with code rates  $R = 1/2$   or   $R = 1/3$  are used as with GSM.  These are generated with eight memory elements of a feedback shift register  $(256$  states$)$.  The coding gain is approximately  $4.5$  to  $6$  dB with code rate  $R = 1/3$  and at low error rates.
  • For higher data rates,  one uses  "turbo codes"  of rate   $R = 1/3$.  The shift register consists here of three memory cells,  which can assume a total of eight states.  The gain of turbo codes is larger by  $2$  to  $3$  dB than by convolutional codes and depends on the number of iterations.  You need a processor with high processing power for this and there may be relatively large delays.


After channel coding,  the data is fed to an  "interleaver"  as in GSM,  in order to be able to resolve bundle errors caused by fading on the receiving side.  Finally, for  "rate matching"  of the resulting data to the physical channel,  individual bits are removed  $($"puncturing"$)$  or repeated  $($"repetition"$)$  according to a predetermined algorithm.

Error correction mechanisms in UMTS

$\text{Example 4:}$  The graph first shows the increase in bits due to a convolutional or turbo code of rate  $R =1/3$, where  the  $188$  bit time frame  $($after the CRC checksum and tail bits$)$  becomes a  $564$  bit frame.

  • Followed by a first  $($external$)$  nesting and then a second  $($internal$)$  nesting.
  • After this,  the time frame is divided into four subframes of  $141$  bits each,  and these are then matched to the physical channel by rate matching.


Frequency responses and pulse shaping for UMTS


Block diagram of the optimal Nyquist equalizer at ideal channel

In this section,  we assume the following block diagram of a binary system with ideal channel   ⇒   $H_{\rm K}(f) = 1$.

In particular,  let hold:

  • The  "transmitter pulse filter"  converts the binary  $\{0, \ 1\}$ data into physical signals.  The filter is described by the frequency response  $H_{\rm S}(f)$,  which is identical in shape to the spectrum of a single transmitted pulse.


  • In UMTS,  the receiver filter  $H_{\rm E}f) = H_{\rm S}(f)$  is matched to the transmitter  $($"matched filter"$)$  and the overall frequency response  $H(f) = H_{\rm S}(f) \cdot H_{\rm E}(f)$  satisfies the  "first Nyquist criterion":
$$ H(f) = H_{\rm CRO}(f) = \left\{ \begin{array}{c} 1 \\ 0 \\ \cos^2 \left( \frac {\pi \cdot (|f| - f_1)}{2 \cdot (f_2 - f_1)} \right)\end{array} \right.\quad \begin{array}{*{1}c} {\rm{for}} \\ {\rm{for}}\\ {\rm else }\hspace{0.05cm}. \end{array} \begin{array}{*{20}c} |f| \le f_1, \\ |f| \ge f_2.\\ \\\end{array}$$

This means:   Consecutive pulses in time do not interfere with each other   ⇒   no  "intersymbol interference"  $\rm (ISI)$  occur.  The associated time function is:

$$h(t) = h_{\rm CRO}(t) ={\rm sinc}(t/ T_{\rm C}) \cdot \frac{\cos(r \cdot \pi t/T_{\rm C})}{1- (2r \cdot t/T_{\rm C})^2}\hspace{0.4cm} \text{with } \hspace{0.4cm} r = \frac{f_2 - f_1}{f_2 + f_1}. $$
  • The sum  $f_1 + f_2$  is equal to the inverse of the chip duration  $T_{\rm C} = 260 \ \rm ns$,  so it is equal to  $3.84 \ \rm MHz$.
  • The  "rolloff factor"  has been determined to  $r = 0.22$  for UMTS.  The two  "corner frequencies"  are thus
$$f_1 = {1}/(2 T_{\rm C}) \cdot (1-r) \approx 1.5\,{\rm MHz},$$
$$f_2 ={1}/(2 T_{\rm C}) \cdot (1+r) \approx 2.35\,{\rm MHz}.$$
  • The required bandwidth is  $B = 2 \cdot f_2 = 4.7 \ \rm MHz$.  Thus,  there is sufficient bandwidth available for each UMTS channel with  $5 \ \rm MHz$.


$\text{Conclusion:}$  The graph shows.

Raised cosine spectrum and impulse response
  • on the left,  the  $($normalized$)$  Nyquist spectrum  $H(f)$,
  • on the right,  the corresponding Nyquist pulse  $h(t)$,  whose zero crossings are equidistant with distance  $T_{\rm C}$.


$\text{It should be noted:}$

  1. The transmission filter  $H_{\rm S}(f)$  and the matched filter  $H_{\rm E}(f)$  are each  "root raised cosine".
  2. Only the product  $H(f) = H_{\rm S}(f) \cdot H_{\rm E}(f)$  leads to the raised cosine.  This also means:
  3. The impulse responses  $h_{\rm S}(t)$  and  $h_{\rm E}(t)$  by themselves do not satisfy the first Nyquist condition.
  4. Only the combination of the two  $($in the time domain the convolution$)$  leads to the desired equidistant zeros.

Modulation methods for UMTS


The modulation techniques used in UMTS can be summarized as follows:

  1. In the downlink:  "Quaternary Phase Shift Keying"  is used for modulation  both in  "frequency division duplex"  $\rm (FDD)$  and in  "time division duplex""  $\rm (TDD)$.
  2. Here,  user data  $($DPDCH channel$)$  and control data  $($DPCCH channel$)$  are multiplexed in time.
  3. With TDD,  the signal is modulated in the uplink also by means of QPSK,  but not with  FDD. 
  4. Here,  a  "dual channel binary phase shift keying"  is used   ⇒   different channels are transmitted in  "in-phase"  and  "quadrature components".
  5. Thus,  two chips are transmitted per modulation step.  The gross chip rate is therefore twice the modulation rate of  $3.84$ Mchip per second.


$\text{Example 5:}$  The graph shows in the equivalent low-pass domain this  "I/Q multiplexing method",  as it is also called:

Modulation and pulse shaping for UMTS
  1. The spread useful data of the DPDCH channel is modulated onto the inphase component.
  2. The spread control data of the DPCCH channel is modulated onto the quadrature component.
  3. After modulation,  the quadrature component is weighted by the root of the power ratio  $G$  between the two channels to minimize the influence of power differences between  $I$  and  $Q$.
  4. Finally,  the complex sum signal  $(I +{\rm j} \cdot Q)$  is multiplied by a scrambling code that is also complex.


$\text{Conclusion:}$  An advantage of dual channel BPSK modulation is the  possibility of usinglow-power amplifiers.

  • But time division multiplexing of user and control data as in the uplink  is not possible in the downlink.
  • One reason for this is the use of  "Discontinuous Transmission"  $\rm (DTX)$  and the associated time constraints.


Single-user receiver


The task of a CDMA receiver is to separate and reconstruct the transmitted data of the individual subscribers from the sum of the spread data streams.  A distinction is made between  "single-user receivers"  and  "multi-user receivers".

In the UMTS downlink,  it is always used a  »single-user receiver«,  since in the mobile station a joint detection of all subscribers would be too costly

  • due to the large number of active subscribers
  • as well as the length of the scrambling codes and the asynchronous operation.


Such a receiver consists of a bank of independent correlators.

  • Each one of the total  $J$  correlators belongs to a specific spreading sequence.
  • The correlation is usually formed in a so-called  "correlator database"  by software.


Thereby one receives at the correlator output the sum of

Single-user receiver with matched filter
  • the  "auto-correlation function"  of the spreading code and
  • the  "cross-correlation function"  of all other users with their own spreading code.


The graphic shows the simplest realization of such a receiver with matched filter.

  1. The received signal  $r(t)$  is first multiplied by the spreading code  $c(t)$  of the considered subscriber,  which is called  "despreading"  $($yellow background$)$.
  2. Followed by convolution with the matched filter impulse response  $($"Root Raised Cosine"$)$  to maximize SNR,  and sampling in bit clock  $(T_{\rm B})$.
  3. Finally, the threshold decision is made, which provides the sink signal  $v(t)$ and  thus the data bits of the considered subscriber.


$\text{Please note:}$ 

  • For the AWGN channel,  spreading at the transmitter and the matched despreading at the receiver have no effect on the bit error probability because of  $c(t)^2 = 1$.  As shown in  $\text{Exercise 4.5}$,  even with spreading/despreading at the optimal receiver, regardless of spreading factor  $J$:
$$p_{\rm B} = {\rm Q} \left( \hspace{-0.05cm} \sqrt { {2 \cdot E_{\rm B} }/{N_{\rm 0} } } \hspace{0.05cm} \right )\hspace{0.05cm}. $$
  • This result can be justified as follows:  The statistical properties of white noise  $n(t)$  are not changed by multiplication with the  $±1$  signal  $c(t)$.

Rake receiver


Another receiver for single-user detection is the  »rake receiver«,  which leads to significant improvements for a multipath channel.

Structure of the rake receiver  $($shown in the equivalent low-pass domain$)$

The diagram shows its setup for a two-way channel with 

  • a direct path with coefficient  $h_0$  and delay time  $τ_0$,
  • an echo with coefficient  $h_1$  and delay time  $τ_1$.


For simplicity,  the coefficients  $h_0$  and  $h_1$  are assumed to be real.  Due to the representation in the equivalent low-pass domain,  these could also be complex.

⇒   The task of the rake receiver is to concentrate the signal energies of all paths  $($in this example only two$)$  to a single instant.  It works accordingly like a  "rake"  for the garden.
If one applies a Dirac delta impulse at time  $t = 0$  to the channel input,  there will be three Dirac delta impulses at the output of the rake receiver:

$$ s(t) = \delta(t) \hspace{0.3cm}\Rightarrow\hspace{0.3cm} y(t) = h_0 \cdot h_1 \cdot \delta(t - 2\tau_0) + (h_0^2 + h_1^2) \cdot \delta(t - \tau_0 - \tau_1)+ h_0 \cdot h_1 \cdot \delta(t - 2\tau_1) .$$
  • The signal energy is concentrated at the time  $τ_0 + τ_1$.  Of the total four paths,  two contribute  $($middle term$)$.
  • The Dirac delta functions at  $2τ_0$  and  $2τ_1$  do cause momentum interference.  However, their  weights are much smaller than those of the main path.


$\text{Example 6:}$  With channel parameters  $h_0 = 0.8$  and  $h_1 = 0.6$  the main path  $($with weight  $h_0)$  contains  $0.82/(0.82 + 0.62) = 64\%$  of the total signal energy.

  • With rake receiver and the same weights,  the above equation is:
$$ y(t) = 0.48 \cdot \delta(t - 2\tau_0) + 1.0 \cdot \delta(t - \tau_0 - \tau_1)+ 0.48 \cdot \delta(t - 2\tau_1) .$$
  • The share of the main path in the total energy amounts in this simple example to  ${1^2}/{(1^2 + 0.48^2 + 0.48^2)} ≈ 68\%.$


Rake receivers are preferred for implementation in mobile devices,  but have a limited performance when there are many active participants.

  1. In a multipath channel with many  $(M)$  paths,  the Rake has also  $M$  fingers.
  2. The main finger  – also called  "searcher"  – is responsible for identifying and ranking the individual paths of multiple propagation.
  3. It searches for the strongest paths and assigns them to other fingers along with their control information.
  4. In the process,  the time and frequency synchronization of all fingers is continuously compared with the control data of the received signal.

Multi-user receiver


In a single-user receiver,  only the data signal of one subscriber is decided,  while all other subscriber signals are considered as additional noise.  However,  the bit error rate of such a detector will be very large

  • if there is large  "intracell interference"  $($many active subscribers in the considered radio cell$)$
  • or large  "intercell interference"  $($highly interfering subscribers in neighboring cells$)$.


In contrast,  »multi-user receivers«  make a joint decision for all active subscribers.» Their characteristics can be summarized as follows:

  1. Such a multi-user receiver does not consider the interference from other participants as noise,  but also uses the information contained in the interference signals for detection.
  2. The receiver is expensive to implement and the algorithms are extremely computationally intensive.  It contains an extremely large correlator database followed by a common detector.
  3. The multi-user receiver must know the spreading codes of all active users.  This requirement precludes use in the UMTS downlink  $($i.e.,  at the mobile station$)$.  In contrast,  all subscriber-specific spreading codes are known a-priori to the base stations,  so that multi-user detection is only used in the uplink.
  4. Some detection algorithms additionally require knowledge of other signal parameters such as energies and delay times.  The common detector  – the heart of the receiver –  is responsible for applying the appropriate detection algorithm in each case.
  5. Examples of multi-user detection are  "decorrelating detection"  and  "Interference Cancellation".


Near–far problem


The  "near-far problem"  is exclusively an uplink problem,  i.e.,  the transmission from mobile subscribers to a base station.  We consider a scenario with two users at different distances from the base station according to the following graph.  This can be interpreted as follows:

Scenarios for the near-far problem
  1. If both mobile stations transmit with the same power,  the received power of the red user  $\rm A$  at the base station is significantly smaller than that of the blue user  $\rm B$  $($left scenario$)$ due to path loss.
  2. In large macrocells,  the difference can be as much as  $100$  dB.  As a result,  the red signal is largely obscured by the blue.
  3. You can largely avoid the near-far problem if user  $\rm A$  transmits with higher power than user  $\rm B$,  as indicated in the right scenario.
  4. Then,  at the base station,  the received power of both mobile stations is then  $($almost$)$  equal.


Note:   In an idealized system  $($one-way channel,  ideal A/D converters,  fully linear amplifiers$)$  the transmitted data of the users are orthogonal to each other and one could detect the users individually even with very different received powers.  This statement is true

  • for UMTS  $($multiple access:  CDMA$)$  as well as
  • for the 2G system GSM  $($FDMA/TDMA$)$,  and
  • for the 4G system LTE  $($TDMA/OFDMA$)$.


In reality,  however,  orthogonality is not always given due to the following reasons:

  1. Different receive paths   ⇒   multipath channel,
  2. non-ideal characteristics of the spreading and scrambling codes in CDMA,
  3. asynchrony of users in the time domain  $($basic propagation delay of paths$)$,
  4. asynchrony of users in the frequency domain  $($non-ideal oscillators and Doppler shift due to mobility of users$)$.


Consequently,  the users are no longer orthogonal to each other and the signal-to-noise ratio of the user to be detected with respect to the other users is not arbitrarily high:

  • For  "GSM"  and  "LTE"  one can assume signal-to-noise ratios of  $25 $  dB  and more,
  • but for UMTS  $($CDMA$)$  only approx.  $15$  dB,  with high-rate data transmission rather less.


Carrier-to-interference power ratio

The term  "capacity"  is generally understood to mean the number of available transmission channels per cell.  However,  since the number of subscribers is not strictly limited in UMTS unlike in GSM,  no fixed capacity can be specified here.

  • In perfect codes,  the subscribers do not interfere with each other.  Thus,  the maximum number of users is determined solely by the spreading factor  $J$  and the available number of mutually orthogonal codes,  which,  however,  is also limited.
  • More practical are non-perfect,  only quasi-orthogonal codes.  Here,  the  "capacity"  of a radio cell is predominantly determined by the resulting interference or the  "carrier-to-interference power ratio"  $\rm (CIR)$.


Carrier-to-interference ratio depending on the number of subscribers

As can be seen from this graph,  CIR depends directly on the number of active participants.  The more active subscribers there are,  the more interference power is generated and the smaller the CIR becomes.

Furthermore,  this decisive criterion for UMTS also depends on the following variables:

  1. The topology and user behavior  $($number of services called up$)$,
  2. the spreading factor  $J$  and the orthogonality of the used spreading code.


In order to limit the disturbing influence of the interference power on the transmission quality,  there are two possible criteria:

»Cell breathing«:   If the number of active subscribers increases significantly with UMTS,  the cell radius is reduced and  $($because of the now fewer subscribers in the cell$)$  also the current interference power is lower.  A less loaded neighboring cell then steps in to supply the subscribers at the edge of the reduced cell.

»Power control«:   If the total interference power within a radio cell exceeds a specified limit,  the transmission power of all subscribers is reduced accordingly and/or the data rate is reduced,  resulting in poorer transmission quality for all.  More about this in the next section.


Power and power control in UMTS


The ratio between the signal power and the interference power is used as the controlled variable for power control in UMTS.  There are differences between the  "frequency division duplex"  $\rm (FDD)$  and the  "time division duplex"  $\rm (TDD)$  modes.

Power control in the FDD mode

We take a closer look at the FDD power control.  In the diagram you can see two different control loops:

  • The  »inner control loop«  controls the transmitter power based on time slots,  where one power command is transmitted in each time slot. 
The power of the transmitter is determined and changed using the CIR estimates in the receiver and the specifications of the  "radio network controller"  $\rm (RNC)$  from the outer control loop.
  • The  outer loop  controls based on $10$ millisecond duration frames. It is implemented in the RNC and is responsible for determining the set point for the inner loop.


The FDD power control sequence is as follows:

  1. The RNC provides a carrier-to-interference ratio  $\rm (CIR)$  setpoint.
  2. The receiver estimates the actual CIR value and generates control commands for the transmitter.
  3. The transmitter changes the transmitted power according to these control commands.


The principle of  "power control in TDD mode"  is similar to the control presented here for the FDD mode.  In fact in the downlink direction they are practically identical.


$\text{Conclusion:}$  The  »TDD power control«  is much slower and thus less precise than in  FDD.

  • However,  fast power control is not even possible in this case,  since each participant has only a fraction of the time frame available for him.


Link budget


When planning UMTS networks,  calculating the link budget is an important step.  Knowledge of the link budget is required both for dimensioning the coverage areas and for determining the capacity and quality of service requirements.

The  »objective of the link budget«  is to calculate the  maximum cell size  considering the following criteria:

  1. Type and data rate of the services,
  2. topology of the environment,
  3. system configuration  $($location and power of base stations,  handover gain$)$,
  4. service requirements  $($availability$)$,
  5. type of mobile station  $($speed,  power$)$,
  6. financial and economic aspects


Budget for a voice transmission channel

$\text{Example 7:}$  The calculation of the link budget is illustrated using the example of a voice transmission channel in the UMTS downlink.  Regarding the exemplary numerical values,  it should be noted:

  1. The transmitted power is  $P_{\rm S} =19$  dBm,  which corresponds to approx.  $79$  mW. 
    Here,  the antenna loss is considered to be  $2$  dB.
  2. The noise power  $P_{\rm N} = 5 \cdot 10^{-11}$  mW   ⇒   $P_{\rm N} = -103$  dBm  
    product of UMTS bandwidth and noise power density.
  3. The interference power is  $P_{\rm I} = -99$  dBm  corresponding to  $1.25 \cdot 10^{-10}$  mW.
  4. This gives the total interference power  $P_{\rm N+I} = P_{\rm N} + P_{\rm I} = 1.25 \cdot 10^{-10}$  mW   ⇒   $P_{\rm N+I} =- 97.5$  dBm.
  5. The antenna sensitivity results in $-97.5 - 27 + 5 - 17 + 3.5 = - 133$  dBm. 
    A large negative value here is  "good".
  6. Maximum allowable path loss should be as large as possible.  Here  $19 - (-133) = 152$  dB.
  7. The  link budget  includes the margin for fading and the handover gain,  in the example:  $140$  dB.
  8. The  maximum cell radius  can be determined from the link budget using an »empirical formula« of Okumura-Hata.  It holds:  $ {r}\ [{\rm km}] = 10^{({\rm LinkBudget}- 137)/35}= 10^{0.0857}\approx 1.22 . $






Notes:  

  • "$\rm dB$"  denotes a logarithmic power specification,  referenced to  $1 \rm W$.
  • In contrast  "$\rm dBm$"  refers to the power  $1 \rm mW$.

UMTS radio resource management


The central task of  »radio resource management«  $\rm (RRM)$  is the dynamic adaptation of radio transmission parameters to the current situation  $($fading,  mobile station movement,  load,  etc.$)$  with the aim to

Radio Resource Management in UMTS
  1. increase the transmission and subscriber capacity,
  2. improve the individual transmission quality,  and
  3. use existing radio resources economically.


The main RRM mechanisms summarized in the diagram are explained below.

Transmit power control
The  radio resource management  attempts to keep the received power and thus the carrier-to-interference ratio  $\rm (CIR)$  at the receiver constant,  or at least to prevent it from falling below a specified limit.

An example of the need for power control is the  "near-far problem",  which is known to cause a disconnect.

The step size of the power control is  $1 \ \rm dB$  or  $2 \ \rm dB$,  and the frequency of the control commands is  $1500$  commands per second.

Regulation of data rate
UMTS allows an exchange between data rate and transmission quality,  which can be realized by selecting the spreading factor.  Doubling the spreading factor corresponds to halving the data rate and increases the quality by  $3\ \rm dB$  $($spreading gain$)$.

Access control
To avoid overload situations in the overall network,  a check is made before a new connection is established to see whether the necessary resources are available.  If not,  the new connection is rejected.  This check is realized by estimating the transmission power distribution after the new connection is established.

Load control
This becomes active if an overload occurs despite access control.  In this case,  a handover to another  base station is initiated and  – if this is not possible –  the data rates of certain nodes are lowered.

Handover
Finally,  radio resource management is also responsible for handover to ensure uninterrupted connections.  Mobile stations are assigned to the individual radio cells on the basis of CIR measurements.


Exercises for the chapter


Exercise 4.5: Pseudo Noise Modulation

Exercise 4.5Z: About Spread Spectrum with UMTS

Exercise 4.6: OVSF Codes

Exercise 4.7: About the Rake Receiver

Sources

  1. 3gpp Group:  UMTS Release 6 - Technical Specification 25.213 V6.4.0,  Sept. 2005.