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ad3845652ec99fdffd75badceec5143f | 06.32 | 3.10 Tone detection | This clause computes the tone variable needed for the threshold adaptation. Tone is only calculated for the VAD in the downlink. In the uplink VAD tone=0.
To reduce delay, this clause should be calculated after the processing of the current speech encoder frame. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.10.1 Windowing | This clause applies a Hanning window to the input frame sof[0..159] to form the output frame sofh[0..159]. The input frame is the current offset compensated signal frame calculated in the RPE‑LTP codec. The array of constants hann[i] is defined in table 3.2.
Multiply signal frame by Hanning window:
|== FOR i = 0 to 79:
| sofh[i] = mult_r( sof[i], hann[i] );
| sofh[159‑i] = mult_r( sof[159‑i], hann[i] );
|== NEXT i; |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.10.2 Auto‑correlation | This clause computes the auto‑correlation vector L_acfh[0..5] from the windowed input frame sofh[0..159]. The input frame must be scaled in order to avoid an overflow situation. This clause is identical to the one used in the RPE‑LTP algorithm, with the exception that only five auto‑correlation values are calculated.
Dynamic scaling of the array sofh[0..159]:
Search for the maximum:
smax = 0;
|== FOR k = 0 to 159:
| temp = abs( sofh[k] );
| IF ( temp > smax ) THEN smax = temp;
|== NEXT k;
Computation of the scaling factor:
IF ( smax == 0 ) THEN scalauto = 0;
ELSE scalauto = sub( 4, norm( smax << 16));
Scaling of the array sofh[0..159]:
IF ( scalauto > 0 ) THEN
| temp = 16384 >> sub( scalauto,1);
|== FOR k = 0 to 159:
| sofh[k] = mult_r( sofh[k], temp);
|== NEXT k:
Compute the L_ACF[..]:
|== FOR k=0 to 4:
| L_acfh[k] = 0;
|==== FOR i=k to 159:
| L_temp = L_mult( sofh[i], sofh[i‑k] );
| L_acfh[k] = L_add( L_acfh[k], L_temp );
|==== NEXT i:
|== NEXT k: |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.10.3 Computation of the reflection coefficients | This clause calculates the reflection coefficients rc[1..4] from the input array L_acfh[0..4]. This procedure is identical to the one in clause 3.3.1 and the RPE‑LTP codec, with the exception that only four reflection coefficients are calculated.
Schur recursion with 16 bits arithmetic:
IF( L_acfh[0] == 0 ) THEN
|== FOR i = 1 to 4:
| rc[i] = 0;
|== NEXT i:
| EXIT; /continue with clause 3.10.4/
temp = norm( L_acfh[0] );
|== FOR k=0 to 4:
| sacf[k] = ( L_acfh[k] << temp ) >> 16;
|== NEXT k:
Initialize array P[..] and K[..] for the recursion:
|== FOR i=1 to 3:
| K[5‑i] = sacf[i];
|== NEXT i:
|== FOR i=0 to 4:
| P[i] = sacf[i];
|== NEXT i:
Compute reflection coefficients:
|== FOR n=1 to 4:
| IF( P[0] < abs( P[1] ) ) THEN
| |== FOR i = n to 4:
| | rc[i] = 0;
| |== NEXT i:
| | EXIT; /continue with clause 3.10.4/
| rc[n] = div( abs( P[1] ), P[0] );
| IF ( P[1] > 0 ) THEN rc[n] = sub( 0, rc[n] );
| IF ( n == 4 ) THEN EXIT; /continue with clause 3.10.4/
|
Schur recursion:
| P[0] = add( P[0], mult_r( P[1], rc[n] ) );
|==== FOR m=1 to 4‑n:
| P[m] = add( P[m+1], mult_r( K[5‑m], rc[n] ) );
| K[5‑m] = add( K[5‑m], mult_r( P[m+1], rc[n] ) );
|==== NEXT m:
|
|== NEXT n: |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.10.4 Filter coefficient calculation | This clause calculates the direct form filter coefficients a[1..2] from the reflection coefficients rc[1..4].
Step‑up procedure to obtain the a[1..2]:
temp = rc[1] >> 2;
a[1] = add( temp, mult_r( rc[2], temp ) );
a[2] = rc[2] >> 2; |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.10.5 Pole Frequency Test | This clause uses the direct form filter coefficients a[1..2] to determine the pole frequency of the second order LPC analysis. If the pole frequency is less than 385 Hz tone is set to 0 and clause 3 terminates.
L_den = L_mult ( a[1], a[1] );
L_temp = a[2] << 16;
L_num = L_sub ( L_temp, L_den );
If pole is not complex then exit:
IF ( L_num <= 0 ) THEN
| tone = 0;
| EXIT; /clause 3 complete/
If pole frequency is less than 385 Hz then exit:
IF ( a[1] < 0) THEN
| temp = L_den >> 16;
| L_den = L_mult ( temp, 3189 );
| L_temp = L_sub ( L_num, L_den );
| IF ( L_temp < 0 ) THEN
| tone = 0;
| EXIT; /clause 3 complete/ |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.10.6 Prediction gain test | This clause uses the reflection coefficients rc[1..4] to calculate the prediction gain. If the prediction gain is greater than 13,5 dB then tone is set to 1 otherwise tone is set to 0.
Calculate normalized prediction error:
prederr = 32767;
|== FOR i=1 to 4
| temp = mult ( rc[i], rc[i] );
| temp = sub ( 32767, temp);
| prederr = mult( prederr, temp );
|== NEXT i;
Test if prediction error is smaller than threshold:
temp = sub ( prederr, 1464 );
IF ( temp < 0 ) THEN tone = 1;
ELSE tone = 0;
Table 3.2: Values of the Hanning window array hann[i]
i
hann
i
hann
i
hann
i
hann
0
0
20
4856
40
16545
60
28139
1
12
21
5325
41
17192
61
28581
2
51
22
5811
42
17838
62
29003
3
114
23
6314
43
18482
63
29406
4
204
24
6832
44
19122
64
29789
5
318
25
7365
45
19758
65
30151
6
458
26
7913
46
20389
66
30491
7
622
27
8473
47
21014
67
30809
8
811
28
9046
48
21631
68
31105
9
1025
29
9631
49
22240
69
31377
10
1262
30
10226
50
22840
70
31626
11
1523
31
10831
51
23430
71
31852
12
1807
32
11444
52
24009
72
32053
13
2114
33
12065
53
24575
73
32230
14
2444
34
12693
54
25130
74
32382
15
2795
35
13326
55
25670
75
32509
16
3167
36
13964
56
26196
76
32611
17
3560
37
14607
57
26707
77
32688
18
3972
38
15251
58
27201
78
32739
19
4405
39
15898
59
27679
79
32764 |
ad3845652ec99fdffd75badceec5143f | 06.32 | 4 Digital test sequences | This clause provides information on the digital test sequences that have been designed to help the verification of implementations of the Voice Activity Detector. Copies of these sequences are available (see clause A.2). |
ad3845652ec99fdffd75badceec5143f | 06.32 | 4.1 Test configuration | The VAD must be tested in conjunction with the speech encoder defined in GSM 06.10. The test configuration is shown in figure 4.1. The input signal to the speech encoder is the sop[...] signal as defined in GSM 06.10 table 5.1. The relevant parameters produced by the speech encoder are input to the VAD algorithm to produce the VAD output. This output has to be checked against some reference files.
The file format of the encoder output parameters given in GSM 06.10 table 5.1 is extended to carry the VAD information.
The VAD information is placed in the unused bit 15 (MSB) of the first encoded parameter:
LAR(1): bit 15 = 1 if VAD on
bit 15 = 0 if VAD 0ff
Furthermore, in order to facilitate approval testing over the air interface, the SP flag generated by the TX DTX handler (see GSM 06.31) on the basis of the VAD flag is placed in the MSB position of the second encoded parameter:
LAR(2): bit 15 = 1 if SP on
bit 15 = 0 if SP off
The output file will also contain the SID codeword and the comfort noise parameters as described in GSM 06.12 and GSM 06.31.
Figure 4.1: VAD test configuration |
ad3845652ec99fdffd75badceec5143f | 06.32 | 4.2 Test sequences | The test sequences are described in detail in clause A.2.
Annex A (informative):
A.1 Simplified block filtering operation
Consider an 8th order transversal filter with filter coefficients a0..a8, through which a signal is being passed, the output of the filter being:
(1)
If we apply block filtering over 20 ms segments, then this equation becomes:
(2)
If the energy of the filtered signal is then obtained for every 20 ms segment, the equation for this is:
(3)
We know that (see GSM 06.10, clause 3.1.4):
(4)
If equation (3) is expanded and acf0..acf8 are substituted for sn then we arrive at the equations:
(5)
Where:
(6)
A.2 Description of digital test sequences
A.2.1 Test sequences
The VAD algorithm uses results from the full rate speech encoder defined in GSM 06.10. In the testing of the VAD, it is assumed that the relevant speech encoder functions have been verified by the test sequences defined in GSM 06.10.
The five types of input sequences are briefly described below.
Spectral comparison
The two kinds of statements of the spectral comparison algorithm (clause 3.4), arithmetic statements and control statements, are tested by separate test sequences.
Arithmetic statements:
spec_a1.*
spec_a2.*
Control statements
spec_c1.*
spec_c2.*
spec_c3.*
spec_c4.*
Threshold adaptation
There are two types of tests to verify the threshold adaptation described in clause 3.6:
adapt_i1.*
adapt_i2.*
The initial test sequences test the acf0 and VAD decision. A fault in the VAD decision will cause all the other sequences to fail, so it is recommended that this test is run before all other tests.
adapt_m1.*
adapt_m2.*
The main test sequences will check the basic threshold adaptation mechanism.
Periodicity detection
pitch1.*
pitch2.*
These sequences check the periodicity detection algorithm described in clause 3.5.
Tone detection
The tone detector test sequences are only required for downlink VAD implementations. There are three types of test to verify the tone detection algorithm described in clause 3.10. The first test sequence tests the operation of the tone detector by means of a frequency sweep:
freq_sw.*
The following test sequences test the prediction gain calculation within the tone detector:
pred1.*
pred2.*
The following sequences test the second order pole frequency calculation within the tone detector:
pole1.*
pole2.*
"Safety" and initialization
safety.*
This sequence checks that safety tests have been implemented to prevent zero values being passed to the norm function. It checks the functions described in the Adaptive Filtering and Energy Computation clause (clause 3.1), and the Predictor Values Computation (clause 3.3). This sequence also checks the initialization of thvad and the rvad array.
Real speech
good_sp.*
bad_sp.*
Because the test sequences cannot be guaranteed to find every possible error, there is a small possibility that an implementation of the correct output for test sequences, but fail with real speech. Because of this, an extra set of sequences are included that consist of barely detectable speech and very clean speech.
There are 3 different file extensions:
*.inp: speech encoder input sequences, binary files
*.vad: output flag of the VAD algorithm, ASCII files
*.cod: TX DTX handler output sequences, binary files for comparison with VAD/DTX handler output.
The *.cod files contain speech coder output information in the format described in clause 4.
It should be noted that there is no requirement in GSM 06.12 for a bit exact implementation of the averaging procedure to calculate the "LAR" and "xmax" parameters in the SID frames. Different implementations are allowed.
The algorithms used for the calculation of the LAR and xmax parameters of the SID frames are therefore reproduced below:
LAR averaging:
| FOR i = 1 to 8:
| L_Temp = 2; /* const. for rounding*/
| | FOR n = 1 to 4:
| | L_Temp1 = LAR[j‑n](i); /*conversion 16 ‑‑> 32 bit*/
| | L_Temp = L_Add( L_Temp , L_Temp1 );
| | NEXT n
| L_Temp = L_temp >> 2;
| mean (LAR(i)) = L_Temp; /*conversion 32 ‑‑> 16 bit*/
| NEXT i;
xmax averaging
L_Temp = 8; /* const. for rounding*/
| FOR n = 1 to 4:
| | FOR i = 1 to 4:
| | L_Temp1 = xmax[j‑n](i); /*conversion 16 ‑‑> 32 bit*/
| | L_Temp = L_Add( L_Temp , L_Temp1 );
| | NEXT i
| NEXT n
L_Temp = L_Temp >> 4;
mean (xmax) = L_Temp; /*conversion 32 ‑‑> 16 bit*/
A.2.2 File format description
All the *.inp and *.cod files are written in binary using 16 bit words, while all *.vad files are written in ASCII format. The sizes of the files are shown in table A.2.1, A.2.2 and A.2.3. The detailed format of the *.inp and *.cod files is in accordance with the descriptions given in GSM 06.10 clause 5.
Table A.2.1: File sizes for *.inp extension files
File:
Frames:
Size in bytes:
spec_a1.inp
22
7 040
spec_a2.inp
22
7 040
spec_c1.inp
48
15 360
spec_c2.inp
48
15 360
spec_c3.inp
48
15 360
spec_c4.inp
48
15 360
adapt_i1.inp
67
21 440
adapt_i2.inp
48
15 360
adapt_m1.inp
403
128 960
adapt_m2.inp
376
120 320
pitch1.inp
35
11 200
pitch2.inp
35
11 200
freq_sw.inp
560
179 200
pred1.inp
126
40 320
pred2.inp
126
40 320
pole1.inp
97
31 040
pole2.inp
42
13 440
safety.inp
5
16 00
good_sp.inp
312
99 840
bad_sp.inp
312
99 840
Table A.2.2: File sizes for *.cod extension files
File:
Frames:
Size in bytes:
spec_a1.cod
22
3 344
spec_a2.cod
22
3 344
spec_c1.cod
48
7 296
spec_c2.cod
48
7 296
spec_c3.cod
48
7 296
spec_c4.cod
48
7 296
adapt_i1.cod
67
10 184
adapt_i2.cod
48
7 296
adapt_m1.cod
403
61 256
adapt_m2.cod
376
57 152
pitch1.cod
35
5 320
pitch2.cod
35
5 320
freq_sw.cod
560
85 120
pred1.cod
126
19 152
pred2.cod
126
19 152
pole1.cod
97
14 744
pole2.cod
42
6 384
safety.cod
5
760
good_sp.cod
312
47 424
bad_sp.cod
312
47 424
Table A.2.3: File sizes for *.vad extension files
File:
Frames:
Size in bytes:
spec_a1.vad
22
88
spec_a2.vad
22
88
spec_c1.vad
48
192
spec_c2.vad
48
192
spec_c3.vad
48
192
spec_c4.vad
48
192
adapt_i1.vad
67
268
adapt_i2.vad
48
192
adapt_m1.vad
403
1 612
adapt_m2.vad
376
1504
pitch1.vad
35
140
pitch2.vad
35
140
freq_sw.inp
560
2 240
pred1.vad
126
504
pred2.vad
126
504
pole1.vad
97
388
pole2.vad
42
168
safety.vad
5
20
good_sp.vad
312
1 248
bad_sp.vad
312
1 248
A.3 VAD performance
In optimizing a VAD a difficult trade‑off has to be made between speech clipping which reduces the subjective performance of the system, and the average activity factor. The benefit of DTX is increased as the average activity factor is reduced. However, in general, a reduction of the activity will be associated with a greater risk for audible speech clipping.
In the optimization process, great emphasis has been placed on avoiding unnecessary speech clipping. However, it has been found that a VAD with virtually no audible clipping would result in a very high activity and very little DTX advantage.
The VAD specified in this technical specification introduces audible and possibly objectionable clipping in certain cases, mainly with low input levels. However, a comprehensive evaluation programme consisting of about 600 individual conversations conducted in a wide range of realistic conditions, it was found that about 90% of the conversations were free from objectionable clipping.
The voice activity performance of the VAD is summarized in table A.3.1. The activity figures are averages of a large number of conversations covering factors like different talkers, noise characteristics and locations. It should be noted that the actual activity of a particular talker in a specific conversation may vary considerably relative to the averages given. This is due both to the variation in talker behaviour as well as to the level dependency of the VAD (the channel activity has been found to decrease by about 0,5 points of percentage per dB level reduction). However, as mentioned above, a decreased speech input level increases the risk of objectionable speech clipping.
All the values given are activity figures, i.e. the % of time the radio channel has to be on.
Table A.3.1: Summary of channel activity
Telephone
instrument
Situation
Typical channel
activity factor:
Handset
Quiet location
55%
Handset
Moderate office
noise with
voice interference
60%
Handset
Strong voice
interference (e.g.
airport/railway station)
65‑70%
Handsfree/
handset
Variable vehicle
noise
60%
A.4 Pole frequency calculation
This annex describes the algorithm used to determine whether the pole frequency for a second order analysis of the signal frame is less than 385 Hz.
The filter coefficients for a second order synthesis filter are calculated from the first two unquantized reflection coefficients rc[1..2] obtained from the speech encoder. This is done using the routine described in clause 3.10.4. If the filter coefficients a[0..2] are defined such that the synthesis filter response is given by:
H(z) = 1 / (a[0] + a[1]z‑1 + a[2]z‑2 ) (1)
Then the positions of the poles in the Z‑plane are given by the solutions to the following quadratic:
a[0]z2 + a[1]z + a[2] = 0, a[0] = 1 (2)
The positions of the poles, z, are therefore:
z = re j*sqrt(im), j2 = ‑1 (3)
where:
re = ‑ a[1] / 2 (4)
im = (4*a[2] ‑ a[1]2 ) / 4 (5)
If im is negative then the poles lie on the real axis of the Z‑plane and the signal is not a tone and the algorithm terminates. If re is negative then the poles lie in the left hand side of the Z‑plane and the frequency is greater than 2 000 Hz and the prediction error test can be performed.
If im is positive and re is positive then the poles are complex and lie in the right hand side of the Z‑plane and the frequency in Hz is related to re and im by the expression:
freq = arctan (sqrt(im)/re ) * 4 000 / (6)
Having ensured that both im and re are positive, the test for a dominant frequency less than 385 Hz can be derived by substituting Equations 4 and 5 into Equation 6 and re‑arranging:
(4*a[2] ‑ a[1]2 ) / a[1]2 < (tan(*385/4 000))2 (7)
or
(4*a[2] ‑ a[1]2 ) / a[1]2 < 0.0973 (8)
If this test is true then the signal is not a tone and the algorithm terminates, otherwise the prediction error test is performed.
Annex B (normative):
Test sequences
The test vectors are described in the present document are supplied in archive en_300965v080001p0.zip which accompanies the present document. The files contained in this archive are listed in clause A.2.
The full rate test vectors apply to both GSM Phase 1 and Phase 2. However, the files pole1.* pole2.* pred1.* pred2.* and freq_sw.* are not required for Phase 1 (uplink and downlink) and Phase 2 uplink implementations.
Annex C (informative):
Change Request History
Change history
SMG No.
TDoc. No.
CR. No.
Section affected
New version
Subject/Comments
SMG#09
4.0.5
ETSI Publication
SMG#17
4.2.1
ETSI Publication
SMG#23
4.3.1
ETSI Publication
SMG#23
5.0.3
Release 1996 version
SMG#27
6.0.0
Release 1997 version
SMG#29
7.0.0
Release 1998 version
7.0.1
Version update to 7.0.1 for Publication
SMG#31
8.0.0
Release 1999 version
8.0.1
Update to Version 8.0.1 for Publication
History
Document history
V8.0.0
July 2000
One-step Approval Procedure OAP 20001103: 2000-07-05 to 2000-11-03
V8.0.1
November 2000
Publication |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 1 Scope | The present document specifies the digital test sequences for the GSM enhanced full rate speech codec. These sequences test for a bit exact implementation of the enhanced full rate speech transcoder (GSM 06.60 [2]), Voice Activity Detection (GSM 06.82 [6]), comfort noise (GSM 06.62 [4]) and the discontinuous transmission (GSM 06.81 [5]). |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 2 References | The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
• References are either specific (identified by date of publication, edition number, version number, etc.) or non‑specific.
• For a specific reference, subsequent revisions do not apply.
• For a non-specific reference, the latest version applies.
• A non-specific reference to an ETS shall also be taken to refer to later versions published as an EN with the same number.
• For this Release 1999 document, references to GSM documents are for Release 1999 versions (version 8.x.y).
[1] GSM 01.04: "Digital cellular telecommunication system (Phase 2+); Abbreviations and acronyms".
[2] GSM 06.60: "Digital cellular telecommunications system (Phase 2+); Enhanced Full Rate (EFR) speech transcoding".
[3] GSM 06.61: "Digital cellular telecommunications system (Phase 2+); Substitution and muting of lost frames for Enhanced Full Rate (EFR) speech traffic channels".
[4] GSM 06.62: "Digital cellular telecommunications system (Phase 2+); Comfort noise aspects for Enhanced Full Rate (EFR) speech traffic channels".
[5] GSM 06.81: "Digital cellular telecommunications system (Phase 2+); Discontinuous Transmission (DTX) for Enhanced Full Rate (EFR) speech traffic channels".
[6] GSM 06.82: "Digital cellular telecommunications system (Phase 2+); Voice Activity Detection (VAD) for Enhanced Full Rate (EFR) speech traffic channels".
[7] GSM 06.53: "Digital cellular telecommunications system (Phase 2+); ANSI-C code for the GSM Enhanced Full Rate (EFR) speech codec".
[8] GSM 06.51: "Digital cellular telecommunications system (Phase 2+); Enhanced Full Rate (EFR) speech coding functions; General description". |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 3 Definitions and abbreviations | |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 3.1 Definitions | Definition of terms used in the present document can be found in GSM 06.60 [2], GSM 06.61 [3], GSM 06.62 [4], GSM 06.81 [5] and GSM 06.82 [6]. |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 3.2 Abbreviations | For the purposes of the present document, the following abbreviations apply:
ETS European Telecommunication Standard
GSM Global System for Mobile communications
For abbreviations not given in this subclause see GSM 01.04 [1]. |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 4 General | Digital test sequences are necessary to test for a bit exact implementation of the enhanced full rate speech transcoder (GSM 06.60 [2]), Digital test Voice Activity Detection (GSM 06.82 [6]), comfort noise (GSM 06.62 [4]) and the discontinuous transmission (GSM 06.81 [5]).
The test sequences may also be used to verify installations of the ANSI C code in GSM 06.53 [7].
Clause 5 describes the format of the files which contain the digital test sequences. Clause 6 describes the test sequences for the speech transcoder. Clause 7 describes the test sequences for the VAD, comfort noise and discontinuous transmission.
Clause 8 describes the method by which synchronisation is obtained between the test sequences and the speech codec under test.
Clause 9 describes the alternative acceptance testing of the speech encoder and decoder in the TRAU by means of 8 bit A- or -law compressed test sequences on the A-Interface.
Test sequences for an alternative and fully interoperable implementation using as a basis the 12.2 kbit/s mode of the Adaptive Multi Rate speech coder are described in section 10.
Electronic copies of the digital test sequences are provided as clause 10, these digital test sequences are contained in the archive ts_100725v080100p0.zip which accompanies the present document. |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 5 Test sequence format | This clause provides information on the format of the digital test sequences for the GSM enhanced full rate speech transcoder (GSM 06.60 [2]), Voice Activity Detection (GSM 06.82 [6]), comfort noise (GSM 06.62 [4]) and the discontinuous transmission (GSM 06.81 [5]). |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 5.1 File format | The test sequence files are provided in archive ts_100725v080100p0.zip which accompanies the present document.
Following decompression, four types of file are provided:
- Files for input to the GSM enhanced full rate speech encoder: *.INP
- Files for comparison with the encoder output: *.COD
- Files for input to the GSM enhanced full rate speech decoder: *.DEC
- Files for comparison with the decoder output: *.OUT
The *.DEC files are generated from the corresponding *.COD files.
Tables 1, 2, 3 and 4 define the formats of the four types of file.
Each speech parameter within the speech frame of 244 bits/20 ms is contained in a serial string of 16 bit words, where each word contains the value of one bit of the parameter. In each string of n 16 bit words containing the n bits of a parameter, the most significant bit of the parameter is written first, and the least significant bit is written last. The bit value contained in a single 16 bit word is either 0x0000 or 0x0001 (right justified) for the binary values of “0” and “1”, respectively. See table 6 of GSM 06.60 [2] for the order of occurrence and bit allocation of speech parameters within the speech frame of 244 bits/20 ms.
The samples in the encoder input signal and in the decoder output signal are left justified. |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 5.2 Codec homing | Each *.INP file includes two homing frames at the start of the test sequence. The function of these frames is to reset the speech encoder state variables to their initial value. In the case of a correct installation of the ANSI-C simulation (GSM 06.53 [7]), all speech encoder output frames shall be identical to the corresponding frame in the *.COD file. In the case of a correct hardware implementation undergoing testing, the first speech encoder output frame is undefined and need not be identical to the first frame in the *.COD file, but all remaining speech encoder output frames shall be identical to the corresponding frames in the *.COD file.
Each *.DEC file includes two homing frames at the start of the test sequence. The function of these frames is to reset the speech decoder state variables to their initial value. In the case of a correct installation of the ANSI-C simulation (GSM 06.53 [7]), all speech decoder output frames shall be identical to the corresponding frame in the *.OUT file. In the case of a correct hardware implementation undergoing testing, the first speech decoder output frame is undefined and need not be identical to first frame in the *.OUT file, but all remaining speech decoder output frames shall be identical to the corresponding frames in the *.OUT file.
Table 1: Encoder input sequence (*.INP) format
Name
Description
No. of bits
Justification
s(n)
Encoder input signal
13
Left
Table 2: Encoder output sequence (*.COD) format
Name
Description
No. of bits
Justification
Speech parameters
SPEECH
Serial stream of speech parameter bits to the channel encoder
244
Right
Additional information
VAD
SP
Voice activity detection flag
SP flag
1
1
Right
Right
Table 3: Decoder input sequence (*.DEC) format
Name
Description
No. of bits
Justification
Additional information
BFI
Bad Frame Indicator flag
1
Right
Speech parameters
SPEECH
Serial stream of speech parameter bits to the channel encoder
244
Right
Additional information
SID
TAF
Silence Descriptor flag
Time Alignment Flag
1
1
Right
Right
Table 4: Decoder output sequence (*.OUT) format
Name
Description
No. of bits
Justification
s'(n)
Decoder output signal
13
Left |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 6 Speech codec test sequences | This clause describes the test sequences designed to exercise the GSM enhanced full rate speech transcoder (GSM 06.60 [2]). |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 6.1 Codec configuration | The speech encoder shall be configured to operate in the non-DTX mode. The VAD and SP flags shall be set to 1 at the speech encoder output. |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 6.2 Speech codec test sequences | Table 5 lists the location and size of the speech codec test sequences. |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 6.2.1 Speech encoder test sequences | Twenty-one encoder input sequences are provided. Note that for the input sequences TEST0.INP to TEST3.INP, the amplitude figures are given in 13-bit precision. The active speech levels are given in dBov.
- TEST0.INP - Synthetic harmonic signal. The pitch delay varies slowly from 18 to 143.5 samples. The minimum and maximum amplitudes are -997 and +971.
- TEST1.INP - Synthetic harmonic signal. The pitch delay varies slowly from 144 down to 18.5 samples. Amplitudes at saturation point -4096 and +4095.
- TEST2.INP - Sinusoidal sweep varying from 150 Hz to 3400 Hz. Amplitudes 1250.
- TEST3.INP - Sinusoidal sweep varying from 150 Hz to 3400 Hz. Amplitudes 4000.
- TEST4.INP - Female speech, active speech level: -19.4 dBov, flat frequency response.
- TEST5.INP - Male speech, active speech level: -18.7 dBov, flat frequency response.
- TEST6.INP - Female speech, ambient noise, active speech level: -35.0 dBov, flat frequency response.
- TEST7.INP - Female speech, ambient noise, active speech level: -25.0 dBov, flat frequency response.
- TEST8.INP - Female speech, ambient noise, active speech level: -15.6 dBov, flat frequency response.
- TEST9.INP - Female speech, car noise, active speech level: -35.5 dBov, flat frequency response.
- TEST10.INP - Female speech, car noise, active speech level: -26.1 dBov, flat frequency response.
- TEST11.INP - Female speech, car noise, active speech level: -15.8 dBov, flat frequency response.
- TEST12.INP - Male speech, ambient noise, active speech level: -34.9 dBov, flat frequency response.
- TEST13.INP - Male speech, ambient noise, active speech level: -24.8 dBov, flat frequency response.
- TEST14.INP - Male speech, ambient noise, active speech level: -15.0 dBov, flat frequency response.
- TEST15.INP - Male speech, babble noise, active speech level: -34.1 dBov, flat frequency response.
- TEST16.INP - Male speech, babble noise, active speech level: -24.3 dBov, flat frequency response.
- TEST17.INP - Male speech, babble noise, active speech level: -14.4 dBov, flat frequency response.
- TEST18.INP - Female speech, ambient noise, active speech level: -26.0 dBov, modified IRS frequency response, with many zero frames.
- TEST19.INP - Male speech, ambient noise, active speech level: -36.0 dBov, modified IRS frequency response, with many zero frames.
- TEST20.INP - Sequence for exercising the LPC vector quantisation codebooks and ROM tables of the codec.
The TEST0.INP and TEST1.INP sequences were designed to test the pitch lag of the GSM enhanced full rate speech encoder. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the TEST0.COD and TEST1.COD sequences, respectively.
The TEST2.INP and TEST3.INP sequences are particularly suited for testing the LPC analysis, as well as for finding saturation problems. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the TEST2.COD and TEST3.COD sequences, respectively.
The TEST4.INP and TEST5.INP sequences contain a lot of low-frequency components. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the TEST4.COD and TEST5.COD sequences, respectively.
The TEST18.INP and TEST19.INP sequences contain some “all zeros” frames (silence) in between segments of speech. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the TEST18.COD and TEST19.COD sequences, respectively.
The TEST20.INP sequence was designed to force the encoder to select each of the LPC code indices and each but one of the the ROM table indices of the codec.
The remaining sequences (TEST6.INP to TEST17.INP) were selected on the basis of bringing various input characteristics (background noise) and levels to the test sequence set. In a correct implementation, the resulting speech encoder output parameters shall be identical to those specified in the TEST6.COD to TEST17.COD sequences, respectively. |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 6.2.2 Speech decoder test sequences | Twenty-one speech decoder input sequences TESTXX.DEC (XX = 0..20) are provided. These are derived from the corresponding TESTXX.INP sequences. In a correct implementation, the resulting speech decoder output shall be identical to the corresponding TESTXX.OUT sequences. |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 6.2.3 Codec homing sequence | In addition to the test sequences described above, two homing sequences are provided to assist in codec testing. TEST21.INP contains one encoder-homing-frame. TEST21.DEC contains one decoder-homing-frame. The use of these sequences is described in GSM 06.51 [8].
Table 5: Location and size of speech codec test sequences
Disk No.
File Name
No. of frames
Size (bytes)
1/8
1/8
1/8
1/8
TEST0.INP
TEST0.COD
TEST0.DEC
TEST0.OUT
285
91 200
140 220
140 790
91 200
1/8
1/8
1/8
1/8
TEST1.INP
TEST1.COD
TEST1.DEC
TEST1.OUT
285
91 200
140 220
140 790
91 200
1/8
1/8
1/8
1/8
TEST2.INP
TEST2.COD
TEST2.DEC
TEST2.OUT
402
128 640
197 784
198 588
128 640
1/8
1/8
1/8
1/8
TEST3.INP
TEST3.COD
TEST3.DEC
TEST3.OUT
402
128 640
197 784
198 588
128 640
1/8
1/8
1/8
1/8
TEST4.INP
TEST4.COD
TEST4.DEC
TEST4.OUT
301
96 320
148 092
148 694
96 320
1/8
1/8
1/8
1/8
TEST5.INP
TEST5.COD
TEST5.DEC
TEST5.OUT
224
71 680
110 208
110 656
71 680
1/8
1/8
1/8
1/8
TEST6.INP
TEST6.COD
TEST6.DEC
TEST6.OUT
335
107 200
164 820
165 490
107 200
1/8
1/8
1/8
1/8
TEST7.INP
TEST7.COD
TEST7.DEC
TEST7.OUT
363
116 160
178 596
179 322
116 160
1/8
1/8
1/8
1/8
TEST8.INP
TEST8.COD
TEST8.DEC
TEST8.OUT
340
108 800
167 280
167 960
108 800
2/8
2/8
2/8
2/8
TEST9.INP
TEST9.COD
TEST9.DEC
TEST9.OUT
407
130 240
200 244
201 058
130 240
2/8
2/8
2/8
2/8
TEST10.INP
TEST10.COD
TEST10.DEC
TEST10.OUT
383
122 560
188 436
189 202
122 560
2/8
2/8
2/8
2/8
TEST11.INP
TEST11.COD
TEST11.DEC
TEST11.OUT
367
117 440
180 564
181 298
117 440
2/8
2/8
2/8
2/8
TEST12.INP
TEST12.COD
TEST12.DEC
TEST12.OUT
298
95 360
146 616
147 212
95 360
2/8
2/8
2/8
2/8
TEST13.INP
TEST13.COD
TEST13.DEC
TEST13.OUT
338
108 160
166 296
166 972
108 160
2/8
2/8
2/8
2/8
TEST14.INP
TEST14.COD
TEST14.DEC
TEST14.OUT
318
101 760
156 456
157 092
101 760
(continued)
Table 5 (concluded): Location and size of speech codec test sequences
Disk No.
File Name
No. of frames
Size (bytes)
2/8
2/8
2/8
2/8
TEST15.INP
TEST15.COD
TEST15.DEC
TEST15.OUT
328
104 960
161 376
162 032
104 960
2/8
2/8
2/8
2/8
TEST16.INP
TEST16.COD
TEST16.DEC
TEST16.OUT
354
113 280
174 168
174 876
113 280
3/8
3/8
3/8
3/8
TEST17.INP
TEST17.COD
TEST17.DEC
TEST17.OUT
316
101 120
155 472
156 104
101 120
3/8
3/8
3/8
3/8
TEST18.INP
TEST18.COD
TEST18.DEC
TEST18.OUT
402
128 640
197 784
198 588
128 640
3/8
3/8
3/8
3/8
TEST19.INP
TEST19.COD
TEST19.DEC
TEST19.OUT
402
128 640
197 784
198 588
128 640
3/8
3/8
3/8
3/8
TEST20.INP
TEST20.COD
TEST20.DEC
TEST20.OUT
631
201 920
310 452
311 714
201 920
3/8
3/8
TEST21.INP
TEST21.DEC
1
320
494 |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 7 DTX test sequences | This subclause describes the test sequences designed to exercise the VAD algorithm (GSM 06.82 [6]), comfort noise (GSM 06.62 [4]) and discontinuous transmission (GSM 06.81 [5]). |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 7.1 Codec configuration | The VAD, comfort noise and discontinuous transmission shall be tested in conjunction with the speech encoder (GSM 06.60 [2]). The speech encoder shall be configured to operate in the DTX mode defined in GSM 06.62 [4]. |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 7.2 DTX test sequences | Each DTX test sequence consists of four files:
- Files for input to the GSM enhanced full rate speech encoder: *.INP
- Files for comparison with the encoder output: *.COD
- Files for input to the GSM enhanced full rate speech decoder: *.DEC
- Files for comparison with the decoder output: *.OUT
The *.DEC files are generated from the corresponding *.COD files.
In a correct implementation, the speech encoder parameters generated by the *.INP file shall be identical to those specified in the *.COD file; and the speech decoder output generated by the *.DEC file shall be identical to that specified in the *.OUT file.
Table 6 lists the DTX test sequences and their size in frames. |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 7.2.1 Predictor values computation | The computation of the predictor values described in GSM 06.82 [6] is not tested explicitly, since the results from the computation are tested many times via the spectral comparison and threshold adaptation tests. |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 7.2.2 Spectral comparison | The spectral comparison algorithm described in GSM 06.82 [6] is tested by the following test sequence:
- DTX01. * |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 7.2.3 Threshold adaptation | The threshold adaptation algorithm described in GSM 06.82 [6] is tested by the following test sequence:
- DTX02. * |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 7.2.4 Periodicity detection | The periodicity detection algorithm described in GSM 06.82 [6] is tested by the following test sequence:
- DTX03. * |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 7.2.5 Tone detection | The tone detection algorithm described in GSM 06.82 [6] is tested by the following test sequence:
- DTX04. * |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 7.2.6 Safety and initialisation | This sequence checks the safety paths used to prevent zero values being passed to the norm function. It checks the functions described in the adaptive filtering and energy computation, and the prediction values computation given in GSM 06.82 [6]. This sequence also checks the initialisation of thvad and the rvad array:
- DTX05. * |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 7.2.7 Comfort noise test sequence | The test sequences described in sub-subclauses 7.2.2 to 7.2.6 are designed to exercise the VAD described in GSM 06.82 [6] and the discontinuous transmission described in GSM 06.81 [5]. The following test sequence is defined to exercise the comfort noise algorithm described in GSM 06.62 [4]:
- DTX06.* |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 7.2.8 Real speech and tones | The test sequences cannot be guaranteed to find every possible error. There is therefore a small possibility that an incorrect implementation produces the correct output for the test sequences, but fails with real signals. Consequently, an extra sequence is included, which consists of very clean speech, barely detectable speech and a swept frequency tone:
- DTX07. *
NOTE: Some of the DTX test sequences contain homing frames. The DTX test sequences are therefore only suitable for testing a single transcoding.
Table 6: Location and size of DTX test sequences
size
(bytes)
Disk No.
File Name
No. of Frames
*.INP
*.COD
*.DEC
*.OUT
4/8
DTX01
710
227 200
349 320
350 740
227 200
4/8
DTX02
933
298 560
459 036
460 902
298 560
4/8
DTX03
156
49 920
76 752
77 064
49 920
4/8
DTX04
245
78 400
120 540
121 030
78 400
4/8
DTX05
56
17 920
27 552
27 664
17 920
4/8
DTX06
771
246 720
379 332
380 874
246 720
4/8
DTX07
1188
380 160
584 496
586 872
380 160 |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 8 Sequences for finding the 20 ms framing of the GSM enhanced full rate speech encoder | When testing the decoder, alignment of the test sequences used to the decoder framing is achieved by the air interface (testing of MS) or can be reached easily on the Abis-interface (testing on network side).
When testing the encoder, usually there is no information available about where the encoder starts its 20 ms segments of speech input to the encoder.
In the following, a procedure is described to find the 20 ms framing of the encoder using special synchronisation sequences. This procedure can be used for MS as well as for network side.
Synchronisation can be achieved in two steps. First, bit synchronisation has to be found. In a second step, frame synchronisation can be determined. This procedure takes advantage of the codec homing feature of the enhanced full rate codec, which puts the codec in a defined home state after the reception of the first homing frame. On the reception of further homing frames, the output of the codec is predefined and can be triggered to. |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 8.1 Bit synchronisation | The input to the speech encoder is a series of 13 bit long words (104 kbits/s, 13 bit linear PCM). When starting to test the speech encoder, no knowledge is available on bit synchronisation, i.e., where the encoder expects its least significant bits, and where it expects the most significant bits.
The encoder homing frame consists of 160 samples, all set to zero with the exception of the least significant bit, which is set to one (0 0000 0000 0001 binary, or 0x0008 hex if written into 16 bit words left justified). If two such encoder homing frames are input to the encoder consecutively, the decoder homing frame is expected at the output as a reaction of the second encoder homing frame.
Since there are only 13 possibilities for bit synchronisation, after a maximum of 13 trials bit synchronisation can be reached. In each trial three consecutive encoder homing frames are input to the encoder. If the decoder homing frame is not detected at the output, the relative bit position of the three input frames is shifted by one and another trial is performed. As soon as the decoder homing frame is detected at the output, bit synchronisation is found, and the first step can be terminated.
The reason why three consecutive encoder homing frames are needed is that frame synchronisation is not known at this stage. To be sure that the encoder reads two complete homing frames, three frames have to be input. Wherever the encoder has its 20 ms segmentation, it will always read at least two complete encoder homing frames.
An example of the 13 different frame triplets is given in sequence BITSYNC.INP (see table 7). |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 8.2 Frame synchronisation | Once bit synchronisation is found, frame synchronisation can be found by inputting one special frame that delivers 160 different output frames, depending on the 160 different positions that this frame can possibly have with respect to the encoder framing.
This special synchronisation frame was found by taking one input frame and shifting it through the positions 0 to 159. The corresponding 160 encoded speech frames were calculated and it was verified that all 160 output frames were different. When shifting the input synchronisation frame, the samples at the beginning were set to 0x0008 hex, which corresponds to the samples of the encoder homing frame.
Before inputting this special synchronisation frame to the encoder, again the encoder has to be reset by one encoder homing frame. A second encoder homing frame is needed to provoke a decoder homing frame at the output that can be triggered to. And since the framing of the encoder is not known at that stage, three encoder homing frames have to precede the special synchronisation frame to ensure that the encoder reads at least two homing frames, and at least one decoder homing frame is produced at the output, serving as a trigger for recording.
The special synchronisation frame preceded by the three encoder homing frames are given in SEQSYNC.INP. The corresponding 160 different output frames are given in SYNC000.COD through SYNC159.COD. The three digit number in the filename indicates the number of samples by which the input was retarded with respect to the encoder framing. By a corresponding shift in the opposite direction, alignment with the encoder framing can be reached. |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 8.3 Formats and sizes of the synchronisation sequences | BIT SYNC.INP:
This sequence consists of 13 frame triplets. It has the format of the speech encoder input test sequences (13 bit left justified with the three least significant bits set to zero).
The size of it is therefore:
SIZE (BITSYNC.INP) = 13 * 3 * 160 * 2 bytes = 12480 bytes
SEQSYNC.INP:
This sequence consist of 3 encoder reset frames and the special synchronisation frame. It has the format of the speech encoder input test sequences (13 bit left justified with the three least significant bits set to zero).
The size of it is therefore:
SIZE (SEQSYNC.INP) = 4 * 160 * 2 bytes = 1280 bytes
SYNCXXX.COD:
These sequences consists of 1 encoder output frame each. They have the format of the speech encoder output test sequences (16 bit words right justified). The values of the VAD and SP flags are set to one in these files.
The size of them is therefore:
SIZE (SYNCXXX.COD) = (244 + 2) * 2 bytes = 492 bytes
Table 7 summarises this information.
Table 7: Location, size and justification of synchronisation sequences
Disk No.
Purpose of Sequence
Name of Sequence
No. of Frames
Size in Bytes
Justification
3/8
Bit Synchronisation
BITSYNC.INP
39
1 2480
Left
3/8
Frame Synchronisation (input)
SEQSYNC.INP
4
1 280
Left
3/8
3/8
3/8
"
"
"
3/8
Frame Synchronisation (output)
SYNC000.COD
SYNC001.COD
SYNC002.COD
"
"
"
SYNC159.COD
1
1
1
"
"
"
1
492
492
492
"
"
"
492
Right
Right
Right
"
"
"
Right |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 9 Trau Testing with 8 Bit A- and µ-law PCM Test Sequences | In the previous clauses, tests for the transcoder in the TRAU are described, using 13 bit linear test sequences. However, these 13 bit test sequences require a special interface in the TRAU and do not allow testing in the field. In most cases the TRAU has to be set in special mode before testing.
As an alternative, the speech codec tests in the TRAU can be performed using A- or -law compressed 8 bit PCM test sequences on the A interface. For this purpose modified input test sequences (*-X.INP) are generated from the original sequences by A or law compression. As an input to the encoder they result in modified encoder output sequences (*-X.COD). The same *.dec decoder input sequences as in subclause 6.2.2. are then used to produce the output sequences *-X.OUT, which are A- or µ compressed.
The A- and µ-law compression and decompression does not change the homing frames at the encoder input. The format of all A- and -law PCM files *-X.INP and *-X.OUT is one sample (8 bit) per byte. The format of all other files is as described in clause 5.
All files are contained in archive ts_100725v080100p0.zip which accompanies the present document. The ‘X’ in the tables below with the filenames stands for A (A-law) and U (-law), respectively. The decoder input files *.dec are the same as in table 5 and are not described in this clause.
Table 8: Location and size of compressed 8 bit PCM speech codec test sequences
Disk No.
File Name
No. of frames
Size (bytes)
5-8/8
5-8/8
5-8/8
TEST0-X.INP
TEST0-X.COD
TEST0-X.OUT
285
45 600
140 220
45 600
5-8/8
5-8/8
5-8/8
TEST1-X.INP
TEST1-X.COD
TEST1-X.OUT
285
45 600
140 220
45 600
5-8/8
5-8/8
5-8/8
TEST2-X.INP
TEST2-X.COD
TEST2-X.OUT
402
64 320
197 784
64 320
5-8/8
5-8/8
5-8/8
TEST3-X.INP
TEST3-X.COD
TEST3-X.OUT
402
64 320
197 784
64 320
5-8/8
5-8/8
5-8/8
TEST4-X.INP
TEST4-X.COD
TEST4-X.OUT
301
48 160
148 092
48 160
5-8/8
5-8/8
5-8/8
TEST5-X.INP
TEST5-X.COD
TEST5-X.OUT
224
35 840
110 208
35 840
5-8/8
5-8/8
5-8/8
TEST6-X.INP
TEST6-X.COD
TEST6-X.OUT
335
53 600
164 820
53 600
5-8/8
5-8/8
5-8/8
TEST7-X.INP
TEST7-X.COD
TEST7-X.OUT
363
58 080
178 596
58 080
5-8/8
5-8/8
5-8/8
TEST8-X.INP
TEST8-X.COD
TEST8-X.OUT
340
54 400
167 280
54 400
5-8/8
5-8/8
5-8/8
TEST9-X.INP
TEST9-X.COD
TEST9-X.OUT
407
65 120
200 244
65 120
5-8/8
5-8/8
5-8/8
TEST10-X.INP
TEST10-X.COD
TEST10-X.OUT
383
61 280
188 436
61 280
5-8/8
5-8/8
5-8/8
TEST11-X.INP
TEST11-X.COD
TEST11-X.OUT
367
58 720
180 564
58 720
5-8/8
5-8/8
5-8/8
TEST12-X.INP
TEST12-X.COD
TEST12-X.OUT
298
47 680
146 616
47 680
5-8/8
5-8/8
5-8/8
TEST13-X.INP
TEST13-X.COD
TEST13-X.OUT
338
54 080
166 296
54 080
5-8/8
5-8/8
5-8/8
TEST14-X.INP
TEST14-X.COD
TEST14-X.OUT
318
50 880
156 456
50 880
5-8/8
5-8/8
5-8/8
TEST15-X.INP
TEST15-X.COD
TEST15-X.OUT
328
52 480
161 376
52 480
5-8/8
5-8/8
5-8/8
TEST16-X.INP
TEST16-X.COD
TEST16-X.OUT
354
56 640
174 168
56 640
5-8/8
5-8/8
5-8/8
TEST17-X.INP
TEST17-X.COD
TEST17-X.OUT
316
50 560
155 472
50 560
5-8/8
5-8/8
5-8/8
TEST18-X.INP
TEST18-X.COD
TEST18-X.OUT
402
64 320
197 784
64 320
5-8/8
5-8/8
5-8/8
TEST19-X.INP
TEST19-X.COD
TEST19-X.OUT
402
64 320
197 784
64 320
5-8/8
5-8/8
5-8/8
TEST20-X.INP
TEST20-X.COD
TEST20-X.OUT
631
100 960
310 452
100 960
5-8/8
TEST21-X.INP
1
160
Table 9: Location and size of compressed 8 bit PCM DTX test sequences
size
bytes
Disk No.
File Name
No. of Frames
*.INP
*.COD
*.OUT
5-8/8
DTX01-X
710
113 600
349 320
113 600
5-8/8
DTX02-X
933
149 280
459 036
149 280
5-8/8
DTX03-X
156
24 960
76 752
24 960
5-8/8
DTX04-X
245
39 200
120 540
39 200
5-8/8
DTX05-X
56
8 960
27 552
8 960
5-8/8
DTX06-X
771
123 360
379 332
123 360
5-8/8
DTX07-X
1188
190 080
584 496
190 080
In addition to the test sequences above, special input (seqsyncX.inp) and output (syncxxxX.cod) sequences for frame synchronization are provided. The X again stands for A and law compressed PCM. The synchronization procedure is described in clause 8.
Table 10: Location, size and justification of compressed 8 bit PCM test sequences
Disk No.
Purpose of Sequence
Name of Sequence
No. of Frames
Size in Bytes
Justification
5-8/8
Frame Synchronisation (input)
SEQSYNCX.INP
4
640
-
5-8/8
5-8/8
5-8/8
"
"
"
5-8/8
Frame Synchronisation (output)
SYNC000X.COD
SYNC001X.COD
SYNC002X.COD
"
"
"
SYNC159X.COD
1
1
1
"
"
"
1
492
492
492
"
"
"
492
Right
Right
Right
"
"
"
Right
5-8/8
5-8/8
5-8/8
"
"
"
5-8/8
Frame Synchronisation (output)
SYNC000X.COD
SYNC001X.COD
SYNC002X.COD
"
"
"
SYNC159X.COD
1
1
1
"
"
"
1
492
492
492
"
"
"
492
Right
Right
Right
"
"
"
Right |
2d561ed8a3b06dda8ca0195c2d72b489 | 06.54 | 10 Alternative Enhanced Full Rate implementation using the Adaptive Multi Rate 12.2 kbit/s mode | The 12.2 kbit/s mode of the Adaptive Multi Rate speech coder described in TS 26.071 is functionally equivalent to the GSM Enhanced Full Rate speech coder. An alternative implementation of the Enhanced Full Rate speech service based on the 12.2 kbit/s mode of the Adaptive Multi Rate coder is allowed. Alternative implementations shall implement the functionality specified in TS 26.071 for the 12.2 kbit/s mode, with the difference that the DTX transmission format from GSM 06.81, the comfort noise generation from GSM 06.62 and the decoder homing frame from GSM 06.60 shall be used.
The test sequences are derived from the corresponding AMR test sequences. The modifications that were made and the use of the respective sequences are described below. The input sequences are identical to the AMR test input sequences *.inp.
Speech codec test sequences
• with DTX disabled
t00.inp ... t22.inp (encoder input, from TS 26.074)
t00_efr.cod ... t22_efr.cod (encoder output)
t00_efr.dec ... t22_efr.dec (decoder input)
t00_efr.out ... t22_efr.out (decoder output)
• with DTX enabled, VAD option 1
Dtx1.inp ... Dtx4.inp (encoder input, from TS 26.074)
Dtx1_efr.cod ... Dtx4_efr.cod (encoder output)
Dtx1_efr.dec ... Dtx4_efr.dec (decoder input)
Dtx1_efr.out ... Dtx4_efr.out (decoder output)
• with DTX enabled, VAD option 2
Dt21.inp .... Dt24.inp (encoder input, from TS 26.074)
Dt21_efr.cod ... Dt24_efr.cod (encoder output)
Dt21_efr.dec ... Dt24_efr.dec (decoder input)
Dt21_efr.out ... Dt24_efr.out (decoder output)
The format of the *.cod files is identical to the GSM_EFR *.cod file format (244 Data Bits, VadFlag, SpFlag equaling 246 Words per 20ms frame). The format of the *.dec files is identical to the GSM_EFR *.dec file format, that is (Bfi, 244 Data Bits, Sid, Taf equaling 247 Words per frame (20ms).
In summary, the differences to the AMR Mode MR122 test sequences are:
• DTX handling (VadFlag and SpFlag instead of TxType; different SID frames)
• Decoder homing frame (Decoder homing frame for GSM_EFR is different than for AMR MR122).
Annex A (informative):
Change Request History
Change history
SMG No.
TDoc. No.
CR. No.
Section affected
New version
Subject/Comments
SMG#23
4.0.1
ETSI Publication
SMG#23
5.1.0
Release 1996 version
SMG#27
6.0.0
Release 1997 version
SMG#29
7.0.0
Release 1998 version
7.0.1
Version update to 7.0.1 for Publication
SMG#31
8.0.0
Release 1999 version
SMG#32
P-00-274
A006
4 and new 10
8.1.0
Alternative EFR implementation using the AMR 12.2 mode
Change history
Date
TSG SA#
TSG Doc.
CR
Rev
Subject/Comment
Old
New
12-2000
10
SP-000573
A011
Correction to the test vectors of the alternative EFR version
8.1.0
8.2.0 |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 1 Scope | The present document defines rate adaptation functions to be used in GSM PLMN Base Station Systems (BSS) transcoders and IWF for adapting radio interface data rates to the 64 kbit/s used at the A-interface in accordance with 3GPP TS 03.10.
The number of Base Station System - Mobile-services Switching Centre (BSS - MSC) traffic channels supporting data rate adaptation may be limited. In this case some channels may not support data rate adaptation. Those that do, shall conform to this specification.
NOTE: This specification should be considered together with 3GPP TS 04.21 to give a complete description of PLMN rate adaptation. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 2 References, abbreviations and definitions | |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 2.1 References | The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
• References are either specific (identified by date of publication, edition number, version number, etc.) or non‑specific.
• For a specific reference, subsequent revisions do not apply.
• For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document in the same Release as the present document.
[1] 3GPP TS 01.04: "Digital cellular telecommunications system (Phase 2+); Abbreviations and acronyms".
[2] 3GPP TS 02.34: “Digital cellular telecommunications system (Phase2+): High Speed Circuit Switched Data (HSCSD) - Stage1"
[3] 3GPP TS 03.10: "Digital cellular telecommunications system (Phase 2+); GSM Public Land Mobile Network (PLMN) connection types".
[4] 3GPP TS 03.34: “Digital cellular telecommunications system (Phase 2+): High Speed Circuit Switched Data (HSCSD) - Stage2".
[5] 3GPP TS 04.21: "Digital cellular telecommunications system (Phase 2+); Rate adaption on the Mobile Station ‑ Base Station System (MS ‑ BSS) interface".
[6] 3GPP TS 24.022: "3rd Generation Partnership Project; Technical Specification Group Core Network; Radio Link Protocol (RLP) for Circuit Switched Bearer and Teleservices".
[7] 3GPP TS 05.03: "Digital cellular telecommunications system (Phase 2+); Channel coding".
[8] 3GPP TS 27.001: "3rd Generation Partnership Project; Technical Specification Group Core Network; General on Terminal Adaptation Functions (TAF) for Mobile Stations (MS)".
[9] 3GPP TS 08.08: "Digital cellular telecommunications system (Phase 2+); Mobile Switching Centre ‑ Base Station System (MSC ‑ BSS) interface; Layer 3 specification".
[10] 3GPP TS 29.007: "3rd Generation Partnership Project; Technical Specification Group Core Network; General requirements on interworking between the Public Land Mobile Network (PLMN) and the Integrated Services Digital Network (ISDN) or Public Switched Telephone Network (PSTN)".
[11] ITU-T Recommendation V.110: "Support of data terminal equipment’s (DTEs) with V-Series interfaces by an integrated services digital network".
[12] ITU-T Recommendation I.460:-Multiplexing, rate adaption and support of existing interfaces. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 2.2 Abbreviations | For the purposes of the present document, the following abbreviations apply:
FPS Frame Pattern Substitution
FSI Frame Start Identifier
ZSP Zero Sequence Position |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 2.3 Definitions | For the purposes of the present document, the following terms and definitions apply.
Substream: Stream of data with explicit or implicit numbering between splitter and combine functions.
Channel: A physical full rate channel on the radio interface (TCH/F) independent of the contents.
A interface circuit: The 8 bits that constitute one 64 kbps circuit on the A interface.
A interface subcircuit: One specific bit position or one specific pair of bit positions within the A interface circuit.
EDGE channel: A general term referring to channels based on 8PSK modulation; i.e. TCH/F28.8, TCH/F32.0, and TCH/F43.2. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 3 General approach | 3GPP TS 03.10 (clause 6) defines the PLMN connection types necessary to support the GSM PLMN data and telematic services.
Within the BSS , transcoder and IWF, there are several data rate adaptation functions which are combined as shown in 3GPP TS 03.10 as part of a connection type.
These functions are RA0, RA1, RA1/RA1' , RA1’’ , RAA", RA1'/RAA', RAA' and RA2. The RA2 function is equivalent to that described in ITU-T Recommendation V.110. In addition, splitting/combining, padding and inband numbering functions as defined in 3GPP TS 04.21 and multiplexing as defined herein are used in cases where more than one channel is allowed.
The RA1/RA1' and RA1'/RAA' are relay functions used as indicated in 3GPP TS 03.10.
The BSS uses the information contained in the ASSIGNMENT REQUEST message on the A-interface (see 3GPP TS 08.08) to set the "E bits" and to map the "D bits" as shown below, as well as to choose the correct channel coding. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 4 The RA0 Function | The RA0 function is specified in 3GPP TS 04.21 |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 5 The RA1 Function | For connections where only one channel is allowed used on the radio interface, the specification in 3GPP TS 04.21 for adaptation of synchronous data rates up to and including 9,6 kbit/s to intermediate rates 8 or 16 kbit/s shall apply.
For connection where more than one channel are used on the radio interface, rate adaptation shall apply on the corresponding substreams as specified in 3GPP TS 04.21 for AIUR of 4,8 kbit/s or 9,6 kbit/s. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 6 The RA1’’ Function | The RA1’’ function is specified in 3GPP TS 04.21. The RA1’’ function is only applicable in BSS for AIUR higher than 38,4 kbit/s. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 7 Split/Combine and Padding Functions | The Split/Combine-function in the IWF shall be used in cases when up to and including 4substreams are used.
The Split/Combine-function in the BSS shall be used only when more than four substreams are used. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 7.1 Data Frame distribution into the channels by the Split/Combine function | Described in 3GPP TS 04.21 |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 7.2 Substream numbering | Described in 3GPP TS 04.21 |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 7.3 Initial Substream Synchronisation for Transparent Services | Described in 3GPP TS 04.21 |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 7.4 Frame Synchronisation and Action on loss of Synchronisation | When in the IWF, the Split/Combine function is responsible for controlling the initial frame synchronisation procedure and re-synchronisation procedure as described in 3GPP TS 29.007. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 7.5 Network Independent Clocking | NIC is specified in 3GPP TS 04.21 |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 7.6 Padding | Padding is specified in 3GPP TS 04.21 |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 8 The EDGE Multiplexing Function | In EDGE configurations where the number of radio interface channels and number of channels or substreams used between BTS and MSC do not match, a multiplexing function described below shall be used at BTS to perform data multiplexing/demultiplexing between the radio interface and network channel configurations. A similar functionshall be used also at MS as described in 04.21.
The EDGE multiplexing function is located between the radio interface and RA1’/RAA’ function.
8.1 Transparent services
TCH/F28.8;
Uplink direction
Refer to the description of corresponding downlink procedures in 3GPP TS 04.21. Two TCH/F14.4 substreams are forwarded towards the MSC as in a 2TCH/F14.4 multislot connection.
Downlink direction
The multiplexing function combines the data received through the two TCH/F14.4 substreams into the 29.0 kbit/s radio interface channel. Refer to the description of corresponding uplink procedures in 3GPP TS 04.21.
TCH/F32.0
Uplink direction
The multiplexing function maps the data received from the radio interface into one 64 kbit/s channel so that data carried by timeslot a (0a6) precedes data carried by timeslot a+n (1a+n7) the timeslots belonging to one TDMA-frame.
Downlink direction
The multiplexing function distributes the data received from the 64 kbit/s channel into two 32.0 kbit/s radio interface channels so that 640-bit data blocks are allocated to timeslots a (0a6) and a+n (1a+n7). In the datastream, data carried by timeslot a precedes data carried by timeslot a+n of the same TDMA-frame.
8.2 Non-Transparent services
TCH/F28.8;
Uplink direction
The multiplexing function demultiplexes the data received through the 29.0 kbit/s radio interface channel into two TCH/F14.4 substreams. Two 290-bit blocks carrying the two halves of one RLP frame belong to the same substream. Refer to the corresponding downlink procedures in 3GPP TS 04.21.
Downlink direction
The multiplexing function multiplexes the 290-bit blocks received through two TCH/F14.4 substreams into the 29.0 kbit/s radio interface channel. Refer to the corresponding uplink procedures in 3GPP TS 04.21.
TCH/F43.2;
Uplink direction
The multiplexing function demultiplexes the data received through the 43.5 kbit/s radio interface channel into three TCH/F14.4 substreams. Two 290-bit blocks carrying the two halves of one RLP frame belong to the same substream. Refer to the corresponding downlink procedures in 3GPP TS 04.21.
Downlink direction
The multiplexing function multiplexes the 290-bit blocks received through three TCH/F14.4 substreams into the 43.5 kbit/s radio interface channel. Refer to the corresponding uplink procedures in 3GPP TS 04.21. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 9 The RA1/RA1' Function | For AIURs less than or equal to 38,4 kbit/s, the RA1/RA1’ function in the BSS shall be applied on each of the n substreams and there are no significant differences between the single slot case and the multislot case. For AIURs less than or equal to 38,4 kbit/s RA1/RA1’ is as specified in 3GPP TS 04.21 for the single slot case. The table below gives a relation between the AIUR, channel coding and number of substreams. As an example from table 1: The wanted AIUR is 28,8 kbit/s, the number of substreams needed to support this rate is 3. Each individual substream shall be rate adapted as in the single slot case.
For AIURs of 48 kbit/s, 56 kbit/s and 64 kbit/s, RA1/RA1’’ shall be as specified in 3GPP TS 04.21 for these rates.
Table 1: Relationship between AIUR, channel coding and number of channels
Multislot intermediaterate 8 kbps
Multislot intermediate rate of 16 kbps
AIUR
Transparent
Non-transparent
Transparent
Non-transparent
2,4 kbit/s
1
N/A
N/A
N/A
4,8 kbit/s
1
1
N/A
N/A
9,6 kbit/s
2
2
1
1
14,4 kbit/s
3
3
2
N/A
19,2 kbit/s
4
4
2
2
28,8 kbit/s
N/A
N/A
3
3
38,4 kbit/s
N/A
N/A
4
4
48 kbit/s
N/A
N/A
5
N/A
56 kbit/s
N/A
N/A
5
N/A
64 kbit/s
N/A
N/A
6
N/A |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 9.1 Radio Interface rate of 12 kbit/s | Described in 3GPP TS 04.21. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 9.2 Radio Interface rate of 6 kbit/s | Described in 3GPP TS 04.21. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 9.3 Radio Interface rate of 3.6 kbit/s | Described in 3GPP TS 04.21. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 9.4 Synchronisation | Refer to 3GPP TS 04.21. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 9.5 Idle frames | Refer to 3GPP TS 04.21 |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 10 THE RA1'/RAA' FUNCTION | The RA1'/RAA' shall be applied only when TCH/F14.4, TCH/F28.8, or TCH/F43.2 channel coding is used. The RA1/RAA' converts 290-bit blocks from the channel coder or EDGE multiplexing function into E-TRAU frames and vice versa. The format of E-TRAU frame is specified in 3GPP TS 08.60.
The RA1'/RAA' function in the BSS shall be applied on each of the n substreams and there are no significant differences between the single slot case and the multislot case. The table below gives a relation between the AIUR, channel coding and number of substreams. As an example from table 2 : The wanted AIUR is 28,8 kbit/s, the number of substreams needed to support this rate is 2. Each individual substream shall be rate adapted as in the single slot case.
Table 2 Relationship between AIUR, channel coding and number of channels.
AIUR
Transparent
Non-transparent
14,4 kbit/s
1
1
28,8 kbit/s
2
2
38,4 kbit/s
3
N/A
43,2 kbit/s
N/A
3
48 kbit/s
4
N/A
56 kbit/s
4
N/A
57,6 kbit/s
N/A
4
64 kbit/s
5
N/A |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 10.1 Radio Interface rate of 14,5 kbit/s | See 3GPP TS 08.60. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 10.2 Synchronisation | See 3GPP TS 08.60. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 10.3 Idle frames | See 3GPP TS 08.60. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 11 THE RAA' FUNCTION | The RAA' function shall be applied only when TCH/F14.4, TCH/F28.8, or TCH/F43.2 channels are used.
The RAA' converts E-TRAU frame into A-TRAU frame and vice versa.
The format of the E-TRAU frame is specified in 3GPP TS 08.60. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 11.1 Coding of A-TRAU frame | The format of the A-TRAU frame is given in Figure 5.
An A-TRAU frame carries eight 36 bit-data frames.
C Bits
Table 3
C1
C2
C3
C4
Date Rate
0
1
1
1
14,4 kbit/s
0
1
1
0
14.4 kbit/s idle (IWF to BSS only)
Table 4
C5
BSS to IWF
Frame Type
note 1
IWF to BSS
UFE (Uplink Frame Error)
1
idle
framing error
0
data
no framing error
NOTE 1: Bit C5 corresponds to bit C6 of the E-TRAU frame as defined in 3GPP TS 08.60.
M Bits
Transparent data
M1 and M2 are as defined in 3GPP TS 04.21.
Non transparent data
See subclause 15.2 of this GSM TS.
Z bits
Bits Zi are used for Framing Pattern Substitution.
See subclause 11.2. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 11.2 Framing Pattern Substitution in A-TRAU frame | The Framing Pattern Substitution is used in each of the eight 36 bit data fields of the A-TRAU frame (see Figure 5) to avoid transmitting a sequence of eight zeroes (called Z sequence in the following).
The purposes of FPS is to avoid erroneous synchronisation to the A-TRAU due to sixteen zeroes occurring accidentally in the data bits and to avoid erroneous synchronisation to V.110. The synchronisation pattern of two consecutive V.110 frames cannot be found within a stream of A TRAU frames. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 11.2.1 FPS encoding | A Zero Sequence Position (ZSP) field is used to account for the occurrence of eight zeroes in the 36 bit data field.
NOTE: A sequence of eight zeroes is considered as a block (e.g. a stream of eleven consecutive zeroes produces only one ZSP and not four ZSPs).
The ZSP field is defined as follows:
Table 5
1
2
3
4
5
6
7
8
1
C
A0
A1
A2
A3
A4
1
The meaning of the different bits of the ZSP field is :
C : Continuation bit. '0' means that there is another ZSP in the data field. '1' means that there is no other ZSP.
A0-A4 :address of the next Z sequence (eight zeroes) to be inserted. The address ‘00001’ corresponds to the bit D1, the value ‘11101’ to the bit D29, (A0 is the msb, A4 is the lsb).
NOTE: a Z sequence substitution cannot occur at bit D30..D36 (as it is 8 bit long)
1 : locking bit prevent the false occurrence of a Z sequence.
The Framing Pattern Substitution is applied in each of the eight 36 bit data field (see Figure 5).
Bit Zi indicates whether FPS is used in the ith 36 bit data field (i=1 to 8). The coding of the Zi bit is the following:
Table 6
Zi (i=1..8)
meaning
1
no substitution
0
at least one substitution
If Zi bit indicates no substitution, the output data bits of FPS are equal to the input data bits.
If Zi indicates at least one substitution, the bits D1-D8 contain the first ZSP.
The following description indicates the general operating procedures for FPS. It is not meant to indicate a required implementation of the encoding procedure.
Figure 1
Step 1 :
The input 36 bit sub frame is considered as a bit stream in which the bits are numbered from 1 to 36.
This bit stream contains 0, 1 or several Z sequences, (Zseq1 to Zseq3 on the figure)
The Z sequence is a sequence of 8 consecutive zeroes : '0000 0000'
Step 2 :
Starting from this bit stream, two lists are built up :
2-a : the 'a' list which contains the address of the first bit of each Z sequences.
2-d : the 'd' list which contains all the data blocks which do not have the Z sequence.
Step 3 :
The 'a' list is transformed so as to build the ZSP list. Each ZSP element is used to indicate:
at which address is the next Z sequence of the message
if yet another ZSP element is found at this address (link element)
Step 4 :
The output 37 bit sub frame is built from:
the Zi field which indicates whether the original message has been transformed or not with this technique. In the example given in Figure 1, Zi shall be set to '0' to indicate that at least one FPS has occurred.
the ZSP and D elements interleaved.
As the ZSP elements have exactly the same length as the Z sequence, the sub frame length is only increased by one (the Zi bit), whatever the number of frame pattern substitutions may be.
For special cases, refer to annex A. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 11.3 A-TRAU Synchronisation Pattern | The frame synchronisation is obtained by means of the first two octets in each frame, with all bits coded binary "0" and the first bit in octet no 2 coded binary "1". The following 17 bit alignment pattern is used to achieve frame synchronisation :
00000000 00000000 1XXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX
XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX
XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX
XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX
XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 12 THE RAA'' FUNCTION | On the IWF side of the A interface, the RAA" function shall convert between the A-TRAU format and a synchronous stream. FPS shall be performed by this function as well, see subclause 11.2. In transparent operation, the RAA" function shall handle the M1 and M2 bits as specified for the RA1' function in 3GPP TS 04.21.
In non-transparent operation, the RAA" function shall map between the A-TRAU format and 290 bit blocks consisting of M1, M2 and 288 bits making up half of an RLP frame, see subclause 15.2 of this GSM TS. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 13 The RA2 Function | Described in 3GPP TS 04.21. The RA2 function shall be applied only for single slot operations. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 14 The A-interface Multiplexing Function | The multiplexing function shall be applied only for AIUR up to and including 57.6 kbit/s for multislot operations.
The multiplexing function is based on the ITU-T I.460. The multiplexing function is used to combine n (n=2 to 4) substreams of multislot intermediate rate of 8 kbit/s or n substreams of multislot intermediate rate of 16 kbit/s on one 64 kbit/s stream by using subcircuits in each octet to each substream such that:
i) An 8 kbit/s substream is allowed to occupy subcircuits with positions 1,3,5 or 7 of each octet of the 64 kbit/s stream; a 16 kbit/s stream occupies bit positions (1,2) or (3,4) or (5,6) or (7,8).
ii) The order of the bits at each substream is identical before and after multiplexing.
iii) All unused bit positions shall be set to binary “1".
iv) For transparent multislot configurations the lowest allowed subcircuits are always used.
v) For non-transparent multislot configurations, the lowest allowed subcircuits shall be used at call set up and after change of channel configuration except at downgrading. At downgrading any of the used subcircuits may be released in uplink direction. Always, the released subcircuit(s) in downlink direction shall be the same as the released subcircuit(s) in uplink direction. At a possible subsequent upgrading, the lowest available bit positions shall be used for the added substreams.
NOTE: The rules given here are almost identical to those of I.460, Section ‘Fixed format multiplexing’, except for the rule i) is stricter in that 8 kbit/s substreams cannot occupy any positions, iv) and v) are added. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15 Support of non-transparent bearer services | |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.1 TCH/F9.6 and TCH/F4.8 kbit/s channel codings | In the case of non-transparent services the RA1/RA1' function shall perform the same mapping as that described for transparent services, using 12 and 6 kbit/s radio interface data rates, with the following modification.
The E2 and E3 bits in the modified ITU-T V.110 80 bit frames shown in Figure 3 (derived from the standard ITU-T V.110 frame shown in Figure 2) are used to indicate each consecutive sequence of ITU-T V.110 80 bit frames corresponding to the four modified ITU-T V.110 60 bit frames (Figure 4) received/transmitted in one radio interface frame. This allows 240 bit Radio Link Protocol frames to/from the MSC to be aligned with the 4x60 bit frames encoded by the radio subsystem channel coder as a single unit (see 3GPP TS 05.03). The 8 bits consisting of the E2 and E3 bits in one of the above sequences is referred to as the Frame Start Identifier. The FSI value is 00 01 10 11. This value is assigned to the E2 and E3 bits as shown in Table7.
Table 7
E2
E3
First Modified ITU-T V.110 80 bit frame
0
0
Second
0
1
Third
1
0
Fourth
1
1
As each RLP frame is transported between the BSS and MSC in four modified ITU-T V.110 80 bit frames, it is necessary following a transmission break and at start up, to determine which modified ITU-T V.110 80 bit frame of the stream is the first for a particular RLP frame. This is needed so that correct alignment with the radio subsystem can be achieved.
Modified V.110 80 bit frames can slip in time during re-routing, and whilst sync exists within the modified ITU-T V.110 80 bit frame to determine the modified ITU-T V.110 80 bit frame boundaries, the FSI is required to determine which quarter of an RLP frame each modified ITU-T V.110 80 bit frame contains.
Table 8 : Relationship between FNUR, AIUR, substream rate, number of substreams and intermediate rate
FNUR
AIUR
Number of Channels x Substream Rate
Channel Coding
Multislot Intermediate Rate
2,4 kbit/s
2,4 kbit/s
2-8 times duplication of each bit to reach 2,4 kbit/s
TCH/F4.8
8 kbit/s
4,8 kbit/s
4,8 kbit/s
4,8 kbit/s
TCH/F4.8
8 kbit/s
4,8 kbit/s
9,6 kbit/s
9,6 kbit/s
TCH/F9.6
16 kbit/s
9,6 kbit/s
9,6 kbit/s
2x4,8 kbit/s
2XTCH/F4.8
8 kbit/s
9,6 kbit/s
9,6 kbit/s
9,6 kbit/s
TCH/F9.6
16 kbit/s
14,4 kbit/s
14,4 kbit/s
3X4,8 kbit/s
3XTCH/F4.8
8 kbit/s
14,4 kbit/s
19,2 kbit/s
2X9,6 kbit/s
2XTCH/F9.6
16 kbit/s
19,2 kbit/s
19,2 kbit/s
4X4,8 kbit/s
4XTCH/F4.8
8 kbit/s
19,2 kbit/s
19,2 kbit/s
2X9,6 kbit/s
2XTCH/F9.6
16 kbit/s
28,8 kbit/s
28,8 kbit/s
3X9,6 kbit/s
3XTCH/F9.6
16 kbit/s
38,4
38,4 kbit/s
4X9,6 kbit/s
4XTCH/F9.6
16 kbit/s
NOTE: The table gives the relation between the FNUR, AIUR, Substream Rate, Channel Coding and Intermediate Rate. As an example: the wanted FNUR is 14,4 kbit/s and the selected channel coding is TCH/F9.6. The data stream is split into two substreams of 9,6 kbit/s yielding an AIUR of 19,2 kbit/s. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.1.1 Alignment | An alignment window spanning four modified ITU-T V.110 80 bit frames shall be used to search for the pattern of 8 bits described above in order to identify alignment with an RLP frame.
In the event of failure to detect the 8 bit pattern, the alignment window is shifted one complete modified V.110 80 bit frame, discarding the contents of the most historical frame and then checking the new 8 bit pattern. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.1.2 Support of Discontinuous Transmission (DTX) | The E1 bit in the modified ITU-T V.110 80 bit frame shown in Figure 3 shall be used in the direction MSC-BSS to indicate that DTX may be invoked (see 3GPP TS 24.022). The E1 bit in all of the four consecutive frames relating to the RLP frame to which DTX may be applied shall be set to 1. If DTX is not to be applied, the E1 bit shall be set to 0.
In the direction BSS-MSC the E1 bit shall always be set to 0. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.1.3 Order of Transmission | The first bit of each quarter of an RLP frame to be transmitted shall correspond to bit D1 of a modified V.110 frame (figures 3 and 4). The remaining 59 bits of each quarter of an RLP frame shall correspond to the D and D' bits , D2 - D'12, in order left to right and top to bottom as shown in figures 3 and 4.
The first quarter of an RLP frame to be transmitted shall contain the E2 and E3 bit code 00 as shown in Table 1. The second quarter contains the code 01, etc. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.2 TCH/F14.4, TCH/F28.8, and TCH/F43.2 channel codings | In case of non-transparent service, a 576 bit RLP frame shall be mapped over two consecutive A-TRAU frames.
Because of that mapping, it is required, following a transmission break and at start up, to determine which A-TRAU frame of the stream is the first for a particular RLP frame. This is needed so that correct alignment with the radio subsystem can be achieved.
The two consecutive M1 bits are referred to as the Frame Start Identifier. The FSI value is 01. This value is assigned to the M1 bits as shown in Table 9.
Table 9
M1 bit
First A-TRAU frame
0
Second A-TRAU frame
1
A-TRAU frames can slip in time during re-routing, and whilst A-TRAU frame synchronisation exists, the FSI is required to determine which half of an RLP frame each A-TRAU frame contains.
Table 10 : Relationship between AIUR, substream rate, number of substreams and intermediate rate
AIUR
Number of substreams x
AIUR per substream
Channel Coding
Multislot intermediate Rate
14,4 kbit/s
14,4 kbit/s
TCH/F14.4
16 kbit/s
28,8 kbit/s
2X14,4 kbit/s
2XTCH/F14.4
1XTCH/F28,8
16 kbit/s
43,2 kbit/s
3X14,4 kbit/s
3XTCH/F14.4
1XTCH/F43,2
16 kbit/s
57,6 kbit/s
4X14,4 kbit/s
4XTCH/F14.4
16 kbit/s
57,6 kbit/s
4X14,4 kbit/s
4XTCH/F14.4
2XTCH/F28,8
16 kbit/s
NOTE: The table gives the relation between AIUR, Substream Rate, Channel Coding and Intermediate Rate. As an example: the AIUR is 28,8 kbit/s and the selected channel coding is 14,5 kbit/s. The data stream is split into two substreams of 14,5 kbit/s yielding an AIUR of 28,8 kbit/s
The same number of substreams is used in each direction, even if the AIURs in each direction differ. Superfluous substreams are filled with idle frames. These are inserted at the BTS or IWF and are discarded at the IWFor BTS respectively. At the IWF, the down link AIUR is determined by the out of band signalling (Assignment Complete, Handover Performed), whereas the up link AIUR is determined inband by examining the possible substream positions on the A interface.
. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.2.1 Alignment | An alignment window spanning two 290 bit blocks in case of TCH/F14.4 channel shall be used to search for the pattern of 2 bits '01' described in subclause 15.2, in order to identify alignment with an RLP frame.
In the event of failure to detect the 2 bits pattern the alignment window is shifted one 290 bit block, discarding the contents of the most historical frame and then checking the new 2 bits pattern. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 15.2.2 Support of Discontinuous Transmission (DTX) | The M2 bit in the A-TRAU frame shown in Figure 5 shall be used in the direction MSC to BSS to indicate that DTX may be invoked (see 3GPP TS 24.022). The M2 bit in all of the two consecutive A-TRAU frames relating to the RLP frame to which DTX may be applied shall be set to 1. If DTX is not to be applied, the M2 bit shall be set to 0.
In the direction BSS to MSC the M2 bit shall always be set to 0. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16 Support of transparent bearer services | |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.1 TCH/F9.6 and TCH/F4.8 channel codings | |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.1.1 User rate adaptation on the A interface, AIUR less than or equal to 38,4 kbit/s | The ITU-T V.110 80 bit frame shall be used for transparent data on the A interface. These frames are transmitted on up to four substreams multiplexed into one stream sent over the A interface. The split/combine function is applied on the substreams as specified in clause 5 of this GSM TS. The relation between the AIUR and the number of channels is specified in table11.
The 64 kbit/s consists of octets, bits 1 through 8, with bit 1 transmitted first.
For a 9 600 bit/s radio interface user rate the V.110 frame is carried with a 16 kbits/s stream which occupies bit positions (1,2).
For radio interface user rates of either 4 800 bit/s, 2 400 bit/s, 1 200 bit/s or 300 bit/s the V.110 frame is carried with a 8 kbits/s stream which occupies bit position (1). For user rates < 1 200bit/s asynchronous characters are padded with additional stop elements by the RA0 function (in the MSC/IWF) to fit into 600 bit/s synchronous RA1 rate prior to rate adaptation to 64 kbits/s.
No use of 4 kbit/s stream is foreseen.
In a given V.110 frame on the A interface:
- for 9 600 bit/s there is no repetition of bits D within the 16 kbit/s stream ;
- for 4 800 bit/s there is no repetition of bits D within the 8 kbit/s stream ;
- for 2 400 bit/s each bit D is repeated twice within the 8 kbit/s stream (D1 D1 D2 D2 etc) ;
- for 1 200 bit/s each bit D is repeated four times within the 8 kbit/s stream (D1 D1 D1 D1 D2 D2 D2 D2 etc) ;
- for 600 bit/s each bit D is repeated eight times within the 8kbit/s stream (D1 D1 D1 D1 D1 D1 D1 D1 D2 D2 D2 D2 D2 D2 D2 D2 etc); |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.1.2 User rate Adaptation on the A-interface, AIUR greater than 38,4 kbit/s | For AIUR of 48 kbit/s, 56 kbit/s and 64 kbit/s one stream consisting of ITU-T V.110 32 bit frames or 64 bit frames, as specified in 3GPP TS 04.21 shall be transmitted over the A-interface. Splitting/Combining which occurs in the BSS, is as specified in 3GPP TS 04.21.
Table 11 gives the relation between the User Rate, Substream Rate Channel Coding and the Intermediate Rate. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.1.3 Relation between AIUR and the number of channels | Table11: Relationship between the AIUR, substream rate, channel coding, intermediate rate and number of channels
AIUR
Number of channels x Substream Rate
Channel Coding
(Multislot) intermediate Rate (Note1)
2,4 kbit/s
2-8 times duplication of each bit to reach 4,8 kbit/s
TCH/F4.8
8 kbit/s
4,8 kbit/s
4,8 kbit/s
TCH/F4.8
8 kbit/s
9,6 kbit/s
2X4,8 kbit/s
2XTCH/F4.8
8 kbit/s
9,6 kbit/s
9,6 kbit/s
TCH/F9.6
16 kbit/s
14,4 kbit/s
3X4,8 kbit/s
3XTCH/F4.8
8 kbit/s
14,4 kbit/s
2X9,6 kbit/s w/ padding
2XTCH/F9.6
16 kbit/s
19,2 kbit/s
4X4,8 kbit/s
4XTCH/F4.8
8 kbit/s
19,2 kbit/s
2X9,6 kbit/s
2XTCH/F9.6
16 kbit/s
28,8 kbit/s
3x9,6 kbit/s
3XTCH/F9.6
16 kbit/s
38,4 kbit/s
4X9,6 kbit/s
4XTCH/F9.6
16 kbit/s
48 kbit/s
5X9,6 kbit/s
5XTCH/F9.6
64 kbit/s
56 kbit/s
5X11,2 kbit/s
5XTCH/F9.6
64 kbit/s
64 kbit/s
66x11,2 kbit/s w/padd.
6XTCH/F9.6
64 kbit/s
NOTE: For AIURs 38,4 kbit/s this column indicates the multislot intermediate rate: for higher AIURs it indicates the intermediate rate. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.1.4 Handling of status bits X, SA, SB | In the single slot case, status bit SA shall be coded repeatedly as S1, S3, S6, S8, and SB is coded repeatedly as S4 and S9 in Figure 2. In the multislot case, status bit SA is coded repeatedly as S6, S8 and SB is coded as S9 in figures 2, 5 and 6.
The handling of the status bits shall comply with the synchronisation procedures for transparent services which are as described in 3GPP TS 29.007 (MSC), 3GPP TS 04.21 (BSS), 3GPP TS 27.001 (MS). |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.1.5 Handling of bits E1 to E7 | Bits E1 to E3 shall be used according to 04.21.
Bits E4 to E7 may be used for network independent clocking as indicated in 3GPP TS 04.21. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.2 TCH/F14.4, TCH/F28.8, and TCH/F32.0 channel codings | |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.2.1 User rate adaptation on the A interface, AIUR less than or equal to 56 kbit/s | The A-TRAU frame shall be used for transparent user data rates other than 32 kbit/s on the A interface. The A-TRAU frames are transmitted on up to four substreams multiplexed into one stream sent over the A interface. The split/combine function is applied on the substreams as specified in clause 7 of this TS. The relation between the AIUR and the number of channels is specified in table 12.
In a given A-TRAU frame on the A interface:
- for 14 400 bit/s there is no repetition of bits D within the 16 kbit/s stream in a given A-TRAU frame on the A interface.
The ITU-T I.460 rate adaptation is used for the transparent 32 kbit/s user rate on the A interface, i.e. four bits of each octet in the 64 kbit/s time slot are used for transporting the 32 kbit/s user data. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.2.2 User Rate Adaptation on the A-interface, AIUR greater than 56 kbit/s | For AIUR of 64 kbit/s one stream consisting of ITU-T V.110 32 bit frames or 64 bit frames, as specified in 3GPP TS 04.21 shall be transmitted over the A-interface. Splitting/Combining which occurs in the BSS, shall be as specified in 3GPP TS 04.21.
Table 12 gives the relation between the User Rate, Substream Rate Channel Coding and the Intermediate Rate. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.2.3 Relation between AIUR and the number of channels | Table 12: Relationship between the AIUR, AIUR per substream, channel coding, intermediate rate and number of substreams
AIUR
Number of substreams x
AIUR per substream
Channel Coding
Multislot intermediate Rate (note 1)
14,4 kbit/s
14,4 kbit/s
TCH/F14.4
16 kbit/s
28,8 kbit/s
2X14,4 kbit/s
TCH/F14.4
TCH/F28.8
16 kbit/s
32 kbit/s
1x32 kbit/s
TCH/F32.0
32 kbit/s
38,4 kbit/s
3X14,4 kbit/s w/padding
TCH/F14.4
16 kbit/s
48 kbit/s
4X14,4 kbit/s w/padding
TCH/F14.4
16 kbit/s
56 kbit/s
4X14,4 kbit/s w/padding
1x64.0 kbit/s (Note 2)
TCH/F14.4
TCH/F32.0
16 kbit/s
64 kbit/s
64kbit/s
5X14,4 kbit/s w/padding
1x64.0 kbit/s (Note 2)
TCH/F14.4
TCH/F32.0
64 kbit/s
NOTE 1: For AIURs 56 kbit/s this column indicates the multislot intermediate rate: for higher AIURs it indicates the intermediate rate.
NOTE 2: One substream over two air interface timeslots. No multislot intermediate rate. |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 16.2.4 Handling of status bits X and SB | The X and SB bits shall be carried over the A interface in a multiframe structure as described in subclause 8.1.1.1 of 3GPP TS 04.21. SA bit is not carried over the A interface.
The handling of the status bits shall comply with the synchronisation procedures for transparent services which are as described in 3GPP TS 29.007 (MSC), 3GPP TS 04.21 (BSS), 3GPP TS 27.001 (MS). |
d584819bc08aaf1044fa432bfa99ac6d | 08.20 | 17 Frame Formats | Octet
Bit number
No.
0
1
2
3
4
5
6
7
0
0
0
0
0
0
0
0
0
1
1
D1
D2
D3
D4
D5
D6
S1
2
1
D7
D8
D9
D10
D11
D12
X
3
1
D13
D14
D15
D16
D17
D18
S3
4
1
D19
D20
D21
D22
D23
D24
S4
5
1
E1
E2
E3
E4
E5
E6
E7
6
1
D25
D26
D27
D28
D29
D30
S6
7
1
D31
D32
D33
D34
D35
D36
X
8
1
D37
D38
D39
D40
D41
D42
S8
9
1
D43
D44
D45
D46
D47
D48
S9
Figure 2: The ITU-T V.110 80 bit frame for Transparent Data
octet
bit number
no.
0
1
2
3
4
5
6
7
0
0
0
0
0
0
0
0
0
1
1
D1
D2
D3
D4
D5
D6
D'1
2
1
D7
D8
D9
D10
D11
D12
D'2
3
1
D13
D14
D15
D16
D17
D18
D'3
4
1
D19
D20
D21
D22
D23
D24
D'4
5
1
E1
E2
E3
D'5
D'6
D'7
D'8
6
1
D25
D26
D27
D28
D29
D30
D'9
7
1
D31
D32
D33
D34
D35
D36
D'10
8
1
D37
D38
D39
D40
D41
D42
D'11
9
1
D43
D44
D45
D46
D47
D48
D'12
Figure 3: The modified ITU-T V.110 80 bit frame for Non-Transparent Data
D1
D2
D3
D4
D5
D6
D'1
D7
D8
D9
D10
D11
D12
D'2
D13
D14
D15
D16
D17
D18
D'3
D19
D20
D21
D22
D23
D24
D'4
D'5
D'6
D'7
D'8
D25
D26
D27
D28
D29
D30
D'9
D31
D32
D33
D34
D35
D36
D'10
D37
D38
D39
D40
D41
D42
D'11
D43
D44
D45
D46
D47
D48
D'12
Figure 4: Modified ITU-T V.110 60 bit frame for Non-Transparent Data
bit number
octet number
0
1
2
3
4
5
6
7
0
0
0
0
0
0
0
0
0
1
0
0
0
0
0
0
0
0
2
1
C1
C2
C3
C4
C5
M1
M2
3
Z1
D1
D2
D3
D4
D5
D6
D7
4
D8
D9
D10
D11
D12
D13
D14
D15
36 bit data field 1
5
D16
D17
D18
D19
D20
D21
D22
D23
6
D24
D25
D26
D27
D28
D29
D30
D31
7
D32
D33
D34
D35
D36
Z2
D1
D2
8
D3
D4
D5
D6
D7
D8
D9
D10
9
D11
D12
D13
D14
D15
D16
D17
D18
36 bit data field 2
10
D19
D20
D21
D22
D23
D24
D25
D26
11
D27
D28
D29
D30
D31
D32
D33
D34
12
D35
D36
Z3
D1
D2
D3
D4
D5
13
D6
D7
D8
D9
D10
D11
D12
D13
14
D14
D15
D16
D17
D18
D19
D20
D21
36 bit data field 3
15
D22
D23
D24
D25
D26
D27
D28
D29
16
D30
D31
D32
D33
D34
D35
D36
Z4
17
D1
D2
D3
D4
D5
D6
D7
D8
18
D9
D10
D11
D12
D13
D14
D15
D16
36 bit data field 4
19
D17
D18
D19
D20
D21
D22
D23
D24
20
D25
D26
D27
D28
D29
D30
D31
D32
21
D33
D34
D35
D36
Z5
D1
D2
D3
22
D4
D5
D6
D7
D8
D9
D10
D11
23
D12
D13
D14
D15
D16
D17
D18
D19
36 bit data field 5
24
D20
D21
D22
D23
D24
D25
D26
D27
25
D28
D29
D30
D31
D32
D33
D34
D35
26
D36
Z6
D1
D2
D3
D4
D5
D6
27
D7
D8
D9
D10
D11
D12
D13
D14
28
D15
D16
D17
D18
D19
D20
D21
D22
36 bit data field 6
29
D23
D24
D25
D26
D27
D28
D29
D30
30
D31
D32
D33
D34
D35
D36
Z7
D1
31
D2
D3
D4
D5
D6
D7
D8
D9
32
D10
D11
D12
D13
D14
D15
D16
D17
33
D18
D19
D20
D21
D22
D23
D24
D25
36 bit data field 7
34
D26
D27
D28
D29
D30
D31
D32
D33
35
D34
D35
D36
Z8
D1
D2
D3
D4
36
D5
D6
D7
D8
D9
D10
D11
D12
37
D13
D14
D15
D16
D17
D18
D19
D20
36 bit data field 8
38
D21
D22
D23
D24
D25
D26
D27
D28
39
D29
D30
D31
D32
D33
D34
D35
D36
Figure 5: A-TRAU 320 bit frame
octet
bit number
no.
0
1
2
3
4
5
6
7
0
0
0
0
0
0
0
0
0
1
1
D1
D2
D3
D4
D5
D6
S1
2
1
D7
D8
D9
D10
D11
D12
X
3
1
D13
D14
D15
D16
D17
D18
S3
4
1
D19
D20
D21
D22
D23
D24
S4
5
1
E1
E2
E3
E4
E5
E6
E7
6
1
1
1
1
1
1
1
S6
7
1
1
1
1
1
1
1
X
8
1
1
1
1
1
1
1
S8
9
1
1
1
1
1
1
1
S9
Figure 6: The modified ITU-T V.110 80 bit frame padded for 4,8 kbit/s transparent data at intermediate rate 16 kbit/s
Annex A (informative):
Frame Pattern Substitution
A.1 Special cases
If the sub frame starts with a Zseq, D1 is empty. With the above example, the resulting input and output sub frames are the following :
In the same case as above but with only one ZSP, the resulting input and output sub frames are the following:
A.2 False Z sequence detection
The Framing Pattern Substitution algorithm presented in subclause 10.2 ensures sure that all the Z sequences found in the original sub frame are removed, but it shall be checked that the transformations performed do not introduce new unwanted Z sequences.
The goal of this subclause is to show that the transformed sub frame does not contain new Z sequences introduced by the algorithm itself.
The coding of the ZSP is the key point to avoid such an emulation. The different cases are considered below.
1 : Sequence ZSP
The worst case is when the address is equal to 1 :
1
C
A0
A1
A2
A3
A4
1
1
0
0
0
0
0
1
1
There is a maximum of 5 zeroes.
2 : Sequence Di / ZSP.
By definition, a data block always ends up with a one (except the last one of the message) and the ZSP always starts with a 1.
3 : Sequence ZSP / Di
ZSP always ends up with a 1 and Di has a maximum of 7 zeroes : it is not possible to find 16 zeroes in a row.
4 : Sequence Di / Dj
Di is not the last data block of the message.
As already mentioned, Di ends up with a one (except the last one) : this is the same case as 3.
5 : Sequence Zi / D or D / Zi
This case only occurs when there is no substitution. In this case, the Zi bit close to the D field is always a one: this does not change the number of zeroes in sequence.
6 : Sequence last Di / new framing pattern
The last D sequence can end up with up to 7 zeroes, followed by the 16 zeroes of the next frame.
There is anyhow no ambiguity, when considering that the framing pattern is made up of 16 zeroes followed by a one.
7 : Sequence last Di / Z bit of the next sub frame
The last D sequence can end up with up to 7 zeroes, followed in the worst case by Z=0 and then a ZSP. As a ZSP starts with a one, this makes a maximum of 8 zeroes in a row.
8 : Sequence ZSP / ZSP (not shown on the figure)
This case arrives when the original message has at least 16 zeroes in a row.
As the ZSP element always starts and ends up with a one, this always induces two consecutive ones.
Annex B (informative):
Change History
Change history
Date
TSG #
TSG Doc.
CR
Rev
Subject/Comment
Old
New
s27
A005
Synchronisation
5.3.0
7.0.0
s29
A006
Introduction of EDGE channel codings into the specifications
7.0.0
8.0.0
s30
A007
Asymmetric channel coding
8.0.0
8.1.0
09-2000
TSG#09
NP-000551
A008
1
32 kbit/s UDI/RDI multimedia in GSM
8.1.0
8.2.0
12-2000
TSG#10
NP-000604
A009
Removal of 1200/75 bit/s data rate and clean-up
8.2.0
8.3.0
03-2001
TSG#11
NP-010040
A013
Correction of downgrading procedure for HSCSD
8.3.0
8.4.0 |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.1 Bearer capabilities | |
12a04a2241e852c6515b51dc33bf8149 | 10.59 | 14.1.1 Bearer capabilities | The EDGE radio interface shall be designed to work in all typical GSM radio environments like rural area (RA), typical urban (TU) and an indoor environment. EDGE shall also work in a Hilly Terrain (HT) environment however the main focus is on channels with lower delay spread than HT, as specified in GSM05.05.
The peak rates mentioned below may not be available in the full cell area. The radio interface should however be optimised to provide as much coverage/availability as possible.
In addition to peak data rates, the average throughput and the area where 384 kbps can be achieved are important measures and should be optimized. |
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