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ad3845652ec99fdffd75badceec5143f | 06.32 | 0.1 Scope | The present document specifies the Voice Activity Detector (VAD) to be used in the Discontinuous Transmission (DTX) as described in GSM 06.31. It also specifies the test methods to be used to verify that a VAD complies with the technical specification.
The requirements are mandatory on any VAD to be used either in the GSM Mobile Stations (MS)s or Base Station Systems (BSS)s. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 0.2 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 telecommunications system (Phase 2+); Abbreviations and acronyms".
[2] GSM 06.10: "Digital cellular telecommunications system(Phase 2+); Full rate speech; Transcoding".
[3] GSM 06.12: "Digital cellular telecommunications system(Phase 2+); Full rate speech; Comfort noise aspect for full rate speech traffic channels".
[4] GSM 06.31: "Digital cellular telecommunications system(Phase 2+); Full rate speech; Discontinuous Transmission (DTX) for full rate speech traffic channels". |
ad3845652ec99fdffd75badceec5143f | 06.32 | 0.3 Abbreviations | Abbreviations used in the present document are listed in GSM 01.04 [1]. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 1 General | The function of the VAD is to indicate whether each 20 ms frame produced by the speech encoder contains speech or not. The output is a binary flag which is used by the TX DTX handler defined in GSM 06.31 [4].
The ETS is organized as follows.
Clause 2 describes the principles of operation of the VAD.
In clause 3, the computational details necessary for the fixed point implementation of the VAD algorithm are given. This clause uses the same notation as used for computational details in GSM 06.10.
The verification of the VAD is based on the use of digital test sequences. Clause 4 defines the input and output signals and the test configuration, whereas the detailed description of the test sequences is contained in clause A.2.
The performance of the VAD algorithm is characterized by the amount of audible speech clipping it introduces and the percentage activity it indicates. These characteristics for the VAD defined in the present document have been established by extensive testing under a wide range of operating conditions. The results are summarized in clause A.3. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2 Functional description | The purpose of this clause is to give the reader an understanding of the principles of operation of the VAD, whereas the detailed description is given in clause 3. In case of discrepancy between the two descriptions, the detailed description of clause 3 shall prevail.
In the following clauses of clause 2, a Pascal programming type of notation has been used to describe the algorithm. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.1 Overview and principles of operation | The function of the VAD is to distinguish between noise with speech present and noise without speech present. The biggest difficulty for detecting speech in a mobile environment is the very low speech/noise ratios which are often encountered. The accuracy of the VAD is improved by using filtering to increase the speech/noise ratio before the decision is made.
For a mobile environment, the worst speech/noise ratios are encountered in moving vehicles. It has been found that the noise is relatively stationary for quite long periods in a mobile environment. It is therefore possible to use an adaptive filter with coefficients obtained during noise, to remove much of the vehicle noise.
The VAD is basically an energy detector. The energy of the filtered signal is compared with a threshold; speech is indicated whenever the threshold is exceeded.
The noise encountered in mobile environments may be constantly changing in level. The spectrum of the noise can also change, and varies greatly over different vehicles. Because of these changes the VAD threshold and adaptive filter coefficients must be constantly adapted. To give reliable detection the threshold must be sufficiently above the noise level to avoid noise being identified as speech but not so far above it that low level parts of speech are identified as noise. The threshold and the adaptive filter coefficients are only updated when speech is not present. It is, of course, potentially dangerous for a VAD to update these values on the basis of its own decision. This adaptation therefore only occurs when the signal seems stationary in the frequency domain but does not have the pitch component inherent in voiced speech. A tone detector is also used to prevent adaptation during information tones.
A further mechanism is used to ensure that low level noise (which is often not stationary over long periods) is not detected as speech. Here, an additional fixed threshold is used.
A VAD hangover period is used to eliminate mid‑burst clipping of low level speech. Hangover is only added to speech‑bursts which exceed a certain duration to avoid extending noise spikes. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2 Algorithm description | The block diagram of the VAD algorithm is shown in figure 2.1. The individual blocks are described in the following clauses. ACF, N and sof are calculated in the speech encoder.
Figure 2.1: Functional block diagram of the VAD
The global variables shown in the block diagram are described as follows:
‑ ACF are auto‑correlation coefficients which are calculated in the speech encoder defined in GSM 06.10 (clause 3.1.4, see also clause A.1). The inputs to the speech encoder are 16 bit 2's complement numbers, as described in GSM 06.10, clause 4.2.0;
‑ av0 and av1 are averaged ACF vectors;
‑ rav1 are autocorrelated predictor values obtained from av1;
‑ rvad are the autocorrelated predictor values of the adaptive filter;
‑ N is the long term predictor lag value which is obtained every sub‑segment in the speech coder defined in GSM 06.10;
‑ ptch indicates whether the signal has a steady periodic component;
‑ sof is the offset compensated signal frame obtained in the speech coder defined in GSM 06.10;
‑ pvad is the energy in the current frame of the input signal after filtering;
‑ thvad is an adaptive threshold;
‑ stat indicates spectral stationarity;
‑ vvad indicates the VAD decision before hangover is added;
‑ vad is the final VAD decision with hangover included. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.1 Adaptive filtering and energy computation | Pvad is computed as follows:
This corresponds to performing an 8th order block filtering on the input samples to the speech encoder, after zero offset compensation and pre‑emphasis. This is explained in clause A.1. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.2 ACF averaging | Spectral characteristics of the input signal have to be obtained using blocks that are larger than one 20 ms frame. This is done by averaging the auto‑correlation values for several consecutive frames. This averaging is given by the following equations:
Where n represents the current frame, n‑1 represents the previous frame etc. The values of constants are given in table 2.1.
Table 2.1: Constants and variables for ACF averaging
Constant
Value
Variable
Initial value
frames
4
previous ACF's
av0 & av1
All set to 0 |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.3 Predictor values computation | The filter predictor values aav1 are obtained from the auto‑correlation values av1 according to the equation:
where:
‑ ‑
R = | av1[0], av1[1], av1[2], av1[3], av1[4], av1[5], av1[6], av1[7] |
| av1[1], av1[0], av1[1], av1[2], av1[3], av1[4], av1[5], av1[6] |
| av1[2], av1[1], av1[0], av1[1], av1[2], av1[3], av1[4], av1[5] |
| av1[3], av1[2], av1[1], av1[0], av1[1], av1[2], av1[3], av1[4] |
| av1[4], av1[3], av1[2], av1[1], av1[0], av1[1], av1[2], av1[3] |
| av1[5], av1[4], av1[3], av1[2], av1[1], av1[0], av1[1], av1[2] |
| av1[6], av1[5], av1[4], av1[3], av1[2], av1[1], av1[0], av1[1] |
| av1[7], av1[6], av1[5], av1[4], av1[3], av1[2], av1[1], av1[0] |
‑ ‑
and:
‑ ‑ ‑ ‑
p = |av1[1]| a = |aav1[1]|
|av1[2]| |aav1[2]|
|av1[3]| |aav1[3]|
|av1[4]| |aav1[4]|
|av1[5]| |aav1[5]|
|av1[6]| |aav1[6]|
|av1[7]| |aav1[7]|
|av1[8]| |aav1[8]|
‑ ‑ ‑ ‑
aav1[0] = ‑1
av1 is used in preference to av0 as av0 may contain speech.
The autocorrelated predictor values rav1 are then obtained: |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.4 Spectral comparison | The spectra represented by the autocorrelated predictor values rav1 and the averaged auto‑correlation values av0 are compared using the distortion measure dm defined below. This measure is used to produce a Boolean value stat every 20 ms, as given by these equations:
difference = |dm ‑ lastdm|
lastdm = dm
stat = difference < thresh
The values of constants and initial values are given in table 2.2.
Table 2.2: Constants and variables for spectral comparison
Constant
Value
Variable
Initial value
thresh
0.05
lastdm
0 |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.5 Periodicity detection | The frequency spectrum of mobile noise is relatively stationary over quite long periods. The Inverse Filter Autocorrelated Predictor coefficients of the adaptive filter rvad are only updated when this stationarity is detected. Vowel sounds however, also have this stationarity, but can be excluded by detecting the periodicity of these sounds using the long term predictor lag values (Nj) which are obtained every sub‑segment from the speech codec defined in GSM 06.10. Consecutive lag values are compared. Cases in which one lag value is a factor of the other are catered for, however cases in which both lag values have a common factor, are not. This case is not important for speech input but this method of periodicity detection may fail for some sine waves. The Boolean variable ptch is updated every 20 ms and is true when periodicity is detected. It is calculated according to the following equation:
ptch = oldlagcount + veryoldlagcount >= nthresh
The following operations are done after the VAD decision and when the current LTP lag values (N0 .. N3) are available, this reduces the delay of the VAD decision. (N{‑1} = N3 of previous segment.)
lagcount = 0
for j = 0 to 3 do
begin
smallag = maximum(Nj,N{j‑1}) mod minimum(Nj,N{j‑1})
if minimum(smallag,minimum(Nj,N{j‑1})‑smallag) < lthresh
then increment(lagcount)
end
veryoldlagcount = oldlagcount
oldlagcount = lagcount
The values of constants and initial values are given in table 2..
Table 2.3: Constants and variables for periodicity detection
Constant
Value
Variable
Initial value
lthresh
nthresh
2
4
oldlagcount
veryoldlagcount
N3
0
0
40 |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.6 Information tone detection | The tone flag is only evaluated in the downlink VAD. In the uplink VAD, tone detection is not performed and tone = false.
Computation of the tone flag is complex. It is therefore evaluated after the processing of the current speech encoder frame. In this way transmission of the speech or SID frame is not delayed.
Information tones and environmental noise can be classified by inspecting the short term prediction gain, information tones resulting in higher prediction gains than environmental noise. Tones can therefore be detected by comparing the prediction gain to a fixed threshold. By limiting the prediction gain calculation to a fourth order analysis, information signals consisting of one or two tones can be detected whilst minimizing the prediction gain for environmental noise.
The prediction gain decision is implemented by comparing the normalized prediction error with a threshold. This measure is used to evaluate the Boolean variable tone every 20 ms. The signal is classified as a tone if the prediction error is smaller than the threshold predth. This is equivalent to a prediction gain threshold of 13,5 dB.
Mobile noise can contain very strong resonances at low frequencies, resulting in a high prediction gain. A further test is therefore made to determine the pole frequency of a second order analysis of the signal frame. The signal is classified as noise if the frequency of the pole is less than 385 Hz. The pole frequency calculation is described in clause A.4.
The algorithm for detecting information tones is as follows:
tone = false
den = a[1]*a[1]
num = 4*a[2] ‑ a[1]*a[1]
if ( num <= 0 )
return
if (( a[1] < 0 ) AND ( num / den < freqth ))
return
4
prederr = MULT (1 ‑ RC[i]*RC[i])
i=1
if (prederr < predth)
tone = true
return
The values of the constants are given in table 2.4. The coefficients a[1..2] are transversal filter coefficients calculated from rc[1..2]. The calculation of the reflection coefficients rc[1..4] is described below.
The offset compensated signal frame sof[0..159] is multiplied by the Hanning window to give the windowed frame sofh[0..159]:
where
The auto‑correlation acfh[0..4] of the windowed signal frame is then calculated:
rc[1..4] are then calculated from acfh[0..4] using the Schur recursion described in the RPE‑LTP codec.
Table 2.4: Constants for information tone detection
Constant
Value
freqth
predth
0,0973
0,0158
NOTE: Reflection coefficients are available in the RPE‑LTP codec. However, they are calculated after pre‑emphasis using a rectangular window and do not give good tone detection results. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.7 Threshold adaptation | A check is made every 20 ms to determine whether the VAD decision threshold (thvad) should be changed. This adaptation is carried out according to the flowchart shown in figure 2.2. The constants used are given in table 2.5.
Adaptation takes place in two different situations: firstly whenever ACF[0] is very low and secondly whenever there is a very high probability that speech and information tones are not present.
In the first case, the threshold is adapted if the energy of the input signal is less than pth. The threshold is set to plev without carrying out any further tests because at these very low levels the effect of the signal quantization makes it impossible to obtain reliable results from these tests.
In the second case, the decision threshold (thvad) and the adaptive filter coefficients (rvad) are only updated with the rav1 values when there is a very high probability that speech and information tones are not present. Adaptation occurs if the following conditions are met over a number (adp) of signal frames:
‑ stationarity is detected in the frequency domain;
‑ the signal does not contain a periodic component;
‑ information tones are not present.
The step‑size by which the threshold is adapted is not constant but a proportion of the current value (determined by constants dec and inc). The adaptation begins by experimentally multiplying the threshold by a factor of (1‑1/dec). If the new threshold is now higher than or equal to Pvad times fac then the threshold needed to be decreased and it is left at this new lower level. If, on the other hand, the new threshold level is less than Pvad times fac then the threshold either needed to be increased or kept constant. In this case it is set to Pvad times fac unless this would mean multiplying it by more than a factor of (1+1/inc) (in which case it is multiplied by a factor of (1+1/inc)). The threshold is never allowed to be greater than Pvad+margin.
Table 2.5: Constants and variables for threshold adaptation
Constant
Value
Variable
Initial value
pth
plev
fac
adp
inc
dec
margin
300 000
800 000
3.0
8
16
32
80 000 000
adaptcount
thvad
rvad[0]
rvad[1]
rvad[2]
rvad[3] to
rvad[8]
0
1 000 000
6
‑4
1
All 0
Figure 2.2: Flow diagram for threshold adaptation |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.8 VAD decision | Prior to hangover the VAD decision condition is:
vvad = pvad > thvad |
ad3845652ec99fdffd75badceec5143f | 06.32 | 2.2.9 VAD hangover addition | VAD hangover is only added to bursts of speech greater than or equal to burstconst blocks. The Boolean variable vad indicates the decision of the VAD with hangover included. The values of the constants are given in table 2.6. The hangover algorithm is as follows:
if vvad then increment(burstcount) else burstcount = 0
if burstcount >= burstconst then
begin
hangcount = hangconst;
burstcount = burstconst
end
vad = vvad or (hangcount >= 0)
if hangcount >= 0 then decrement(hangcount)
Table 2.6: Constants and variables for VAD hangover addition
Constant
Value
Variable
Initial value
burstconst
hangconst
3
5
burstcount
hangcount
0
‑1 |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3 Computational details | In the next paragraphs, the detailed description of the VAD algorithm follows the preceding high level description. This detailed description is divided in ten clauses related to the blocks of figure 2.1 (except periodicity updating) in the high level description of the VAD algorithm.
Those clauses are:
1) adaptive filtering and energy computation;
2) ACF averaging;
3) predictor values computation;
4) spectral comparison;
5) periodicity detection;
6) threshold adaptation;
7) VAD decision;
8) VAD hangover addition;
9) periodicity updating;
10) information tone detection.
The VAD algorithm takes as input the following variables of the RPE‑LTP encoder (see the detailed description of the RPE‑LTP encoder GSM 06.10):
‑ L_ACF[0..8], auto‑correlation function (GSM 06.10/4.2.4);
‑ scalauto, scaling factor to compute the L_ACF[0..8] (GSM 06.10/4.2.4);
‑ Nc, LTP lag (one for each sub‑segment, GSM 06.10/4.2.11);
‑ sof, offset compensated signal frame (GSM 06.10/4.2.2).
So four Nc values are needed for the VAD algorithm.
The VAD computation can start as soon as the L_ACF[0..8] and scalauto variables are known. This means that the VAD computation can take place after part 4.2.4 of GSM 06.10 (Auto‑correlation) of the LPC analysis clause of the RPE‑LTP encoder. This scheme will reduce the delay to yield the VAD information. The periodicity updating (included in clause 2.2.5) and information tone detection, are done after the processing of the current speech encoder frame.
All the arithmetic operations and names of the variables follow the RPE‑LTP detailed description. To increase the precision within the fixed point implementation, a pseudo‑floating point representation of some variables is used. This stands for the following variables (and related constants) of the VAD algorithm:
pvad: Energy of filtered signal;
thvad: Threshold of the VAD decision;
acf0: Energy of input signal.
For the representation of these variables, two integers (16 bits) are needed:
‑ one for the exponent (e_pvad, e_thvad, e_acf0);
‑ one for the mantissa (m_pvad, m_thvad, m_acf0).
The value e_pvad represents the lowest power of 2 just greater or equal to the actual value of pvad and the m_pvad value represents a integer which is always greater or equal to 16384 (normalized mantissa). It means that the pvad value is equal to:
This scheme guarantees a large dynamic range for the pvad value and always keeps a precision of 16 bits. All the comparisons are easy to make by comparing the exponents of two variables and the VAD algorithm needs only one pseudo‑floating point addition. All the computations related to the pseudo‑floating point variables require very simple 16 or 32 bits arithmetic operations defined in the detailed description of the RPE‑LTP encoder. This pseudo‑floating point arithmetic is only used in clauses 3.1 and 3.6.
Table 3.1 gives a list of all the variables of the VAD algorithm that must be initialized in the reset procedure and kept in memory for processing the subsequent frame of the RPE‑ LTP encoder. The types (16 or 32 bits) and initial values of all these variables are clearly indicated and their related clause is also mentioned. The bit exact implementation uses other temporary variables that are introduced in the detailed description whenever it is needed.
Table 3.1: Initial values for variables to be stored in memory
Names of variables:
type (# of bits):
Initialization:
Subclause:
Adaptive filter coefficients:
rvad[0]
16
24 576
3.1, 3.6
rvad[1]
16
‑16 384
3.1, 3.6
rvad[2]
16
4 096
3.1, 3.6
rvad[3..8]
16
0
3.1, 3.6
Scaling factor of ravd[0..8]:
normrvad
16
7
3.1, 3.6
Delay line of the auto‑correlation coefficients:
L_sacf[0..26]
32
0
3.2
L_sav0[0..35]
32
0
3.2
Pointers on the delay lines:
pt_sacf
16
0
3.2
pt_sav0
16
0
3.2
Distance measure:
L_lastdm
32
0
3.4
Periodicity counters:
oldlagcount
16
0
3.5, 3.9
veryoldlagcount
16
0
3.5, 3.9
Adaptive threshold:
e_thvad (exponent)
16
20
3.6
m_thvad (mantissa)
16
31 250
3.6
Counter for adaptation:
adaptcount
16
0
3.6
Hangover flags:
burstcount
16
0
3.8
hangcount
16
‑1
3.8
LTP lag memory:
oldlag
16
40
3.9
Tone Detection
tone
16
0
3.10 |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.1 Adaptive filtering and energy computation | This clause computes the e_pvad and m_pvad variables which represent the pvad value. It needs the L_ACF[0..8] and scalauto variables of the RPE‑LTP algorithm and the rvad[0..8] and normrvad variables produced by clause 3.6 of the VAD algorithm. It also computes a floating point representation of L_ACF[0] (e_acf0 and m_acf0) used in clause 3.6.
Test if L_ACF[0] is equal to 0:
IF ( scalauto < 0 ) THEN scalvad = 0;
ELSE scalvad = scalauto; / keep scalvad for use in clause 3.2 /
IF ( L_ACF[0] == 0 ) THEN
| e_pvad = ‑32768;
| m_pvad = 0;
| e_acf0 = ‑32768;
| m_acf0 = 0;
| EXIT /continue with clause 3.2/
Re‑normalization of the L_ACF[0..8]:
normacf = norm( L_ACF[0] );
| FOR i = 0 to 8:
| sacf[i] = ( L_ACF[i] << normacf ) >> 19;
| NEXT i:
Computation of e_acf0 and m_acf0:
e_acf0 = add( 32, (scalvad << 1 ) );
e_acf0 = sub( e_acf0, normacf);
m_acf0 = sacf[0] << 3;
Computation of e_pvad and m_pvad:
e_pvad = add( e_acf0, 14 );
e_pvad = sub( e_pvad, normrvad );
L_temp = 0;
| FOR i = 1 to 8:
| L_temp = L_add( L_temp, L_mult( sacf[i], rvad[i] ) );
| NEXT i:
L_temp = L_add( L_temp, L_mult( sacf[0], rvad[0] ) >> 1 );
IF ( L_temp <= 0 ) THEN L_temp = 1;
normprod = norm( L_temp );
e_pvad = sub( e_pvad, normprod );
m_pvad = ( L_temp << normprod ) >> 16; |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.2 ACF averaging | This clause uses the L_ACF[0..8] and the scalvad variables to compute the array L_av0[0..8] and L_av1[0..8] used in clause 3.3 and 3.4.
Computation of the scaling factor:
scal = sub( 10, (scalvad << 1) );
Computation of the arrays L_av0[0..8] and L_av1[0..8]:
| FOR i = 0 to 8:
| L_temp = L_ACF[i] >> scal;
| L_av0[i] = L_add( L_sacf[i], L_temp );
| L_av0[i] = L_add( L_sacf[i+9], L_av0[i] );
| L_av0[i] = L_add( L_sacf[i+18], L_av0[i] );
| L_sacf[ pt_sacf + i ] = L_temp;
| L_av1[i] = L_sav0[ pt_sav0 + i ];
| L_sav0[ pt_sav0 + i] = L_av0[i];
| NEXT i:
Update of the array pointers:
IF ( pt_sacf == 18 ) THEN pt_sacf = 0;
ELSE pt_sacf = add( pt_sacf, 9);
IF ( pt_sav0 == 27 ) THEN pt_sav0 = 0;
ELSE pt_sav0 = add( pt_sav0, 9); |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.3 Predictor values computation | This clause computes the array rav1[0..8] needed for the spectral comparison and the threshold adaptation. It uses the L_av1[0..8] computed in clause 3.2, and is divided in the three following clauses:
‑ Schur recursion to compute reflection coefficients.
‑ Step up procedure to obtain the aav1[0..8].
‑ Computation of the rav1[0..8]. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.3.1 Schur recursion to compute reflection coefficients | This clause is identical to the one used in the RPE‑LTP algorithm. The array vpar[1..8] is computed with the array L_av1[0..8] as an input.
Schur recursion with 16 bits arithmetic:
IF( L_av1[0] == 0 ) THEN
|== FOR i = 1 to 8:
| vpar[i] = 0;
|== NEXT i:
| EXIT; /continue with clause 3.3.2/
temp = norm( L_av1[0] );
|== FOR k=0 to 8:
| sacf[k] = ( L_av1[k] << temp ) >> 16;
|== NEXT k:
Initialize array P[..] and K[..] for the recursion:
|== FOR i=1 to 7:
| K[9‑i] = sacf[i];
|== NEXT i:
|== FOR i=0 to 8:
| P[i] = sacf[i];
|== NEXT i:
Compute reflection coefficients:
|== FOR n=1 to 8:
| IF( P[0] < abs( P[1] ) ) THEN
| |== FOR i = n to 8:
| | vpar[i] = 0;
| |== NEXT i:
| | EXIT; /continue with
| | clause 3.3.2/
| vpar[n] = div( abs( P[1] ), P[0] );
| IF ( P[1] > 0 ) THEN vpar[n] = sub( 0, vpar[n] );
| IF ( n == 8 ) THEN EXIT; /continue with clause 3.3.2/
|
| Schur recursion:
|
| P[0] = add( P[0], mult_r( P[1], vpar[n] ) );
|==== FOR m=1 to 8‑n:
| P[m] = add( P[m+1], mult_r( K[9‑m], vpar[n] ) );
| K[9‑m] = add( K[9‑m], mult_r( P[m+1], vpar[n] ) );
|==== NEXT m:
|
|== NEXT n: |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.3.2 Step‑up procedure to obtain the aav1[0..8] | Initialization of the step‑up recursion:
L_coef[0] = 16384 << 15;
L_coef[1] = vpar[1] << 14;
Loop on the LPC analysis order:
|= FOR m = 2 to 8:
|== FOR i = 1 to m‑1:
|== temp = L_coef[m‑i] >> 16; / takes the msb /
|== L_work[i] = L_add( L_coef[i], L_mult( vpar[m], temp ) );
|== NEXT i
|=
|== FOR i = 1 to m‑1:
|== L_coef[i] = L_work[i];
|== NEXT i
|=
|= L_coef[m] = vpar[m] << 14;
|= NEXT m:
Keep the aav1[0..8] on 13 bits for next clause:
| FOR i = 0 to 8:
| aav1[i] = L_coef[i] >> 19;
| NEXT i: |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.3.3 Computation of the rav1[0..8] | |= FOR i= 0 to 8:
|= L_work[i] = 0;
|== FOR k = 0 to 8‑i:
|== L_work[i] = L_add( L_work[i], L_mult( aav1[k], aav1[k+i] ) );
|== NEXT k:
|= NEXT i:
IF ( L_work[0] == 0 ) THEN normrav1 =0;
ELSE normrav1 = norm( L_work[0] );
|= FOR i= 0 to 8:
|= rav1[i] = ( L_work[i] << normrav1 ) >> 16;
|= NEXT i:
Keep the normrav1 for use in clause 3.4 and 3.6. |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.4 Spectral comparison | This clause computes the variable stat needed for the threshold adaptation. It uses the array L_av0[0..8] computed in clause 3.2 and the array rav1[0..8] computed in clause 3.3.3.
Re‑normalize L_av0[0..8]:
IF ( L_av0[0] == 0 ) THEN
| FOR i = 0 to 8:
| sav0[i] = 4095;
| NEXT i:
ELSE
| shift = norm( L_av0[0] );
|= FOR i = 0 to 8:
|= sav0[i] = ( L_av0[i] << shift‑3 ) >> 16;
|= NEXT i:
Compute partial of dm:
L_ p = 0;
|= FOR i = 1 to 8:
|= L_ p = L_add( L_ p, L_mult( rav1[i], sav0[i] ) );
|= NEXT i:
Compute the division of partial by sav0[0]:
IF ( L_ p < 0 ) THEN L_temp = L_sub( 0, L_ p );
ELSE L_temp = L_ p;
IF ( L_temp == 0 ) THEN
| L_dm = 0;
| shift = 0;
ELSE
| sav0[0] = sav0[0] << 3;
| shift = norm( L_temp );
| temp = ( L_temp << shift ) >> 16;
| IF ( sav0[0] >= temp ) THEN
| | divshift = 0;
| | temp = div( temp, sav0[0] );
| ELSE
| | divshift = 1;
| | temp = sub( temp, sav0[0] );
| | temp = div( temp, sav0[0] );
|
| IF( divshift == 1 ) THEN L_dm = 32768;
| ELSE L_dm = 0;
|
| L_dm = L_add( L_dm, temp) << 1;
| IF( L_ p < 0 ) THEN L_dm = L_sub( 0, L_dm);
Re‑normalization and final computation of L_dm:
L_dm = ( L_dm << 14 );
L_dm = L_dm >> shift;
L_dm = L_add( L_dm, ( rav1[0] << 11 ) );
L_dm = L_dm >> normrav1;
Compute the difference and save L_dm:
L_temp = L_sub( L_dm, L_lastdm );
L_lastdm = L_dm;
IF ( L_temp < 0 ) THEN L_temp = L_sub( 0, L_temp );
L_temp = L_sub( L_temp, 3277 );
Evaluation of the stat flag:
IF ( L_temp < 0 ) THEN stat = 1;
ELSE stat = 0; |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.5 Periodicity detection | This clause just sets the ptch flag needed for the threshold adaptation.
temp = add( oldlagcount, veryoldlagcount );
IF ( temp >= 4 ) THEN ptch = 1;
ELSE ptch = 0; |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.6 Threshold adaptation | This clause uses the variables e_pvad, m_pvad, e_acf0 and m_acf0 computed in clause 3.1. It also uses the flags stat (see clause 3.4) and ptch (see clause 3.5). It follows the flowchart represented on figure 2.2.
Some constants, represented by a floating point format, are needed and a symbolic name (in capital letter) for their exponent and mantissa is used; table 3.2 lists all these constants with the symbolic names associated and their numerical constant values.
Table 3.2: List of constants
Constant
Exponent
Mantissa
pth
margin
plev
E_PTH = 19
E_MARGIN = 27
E_PLEV = 20
M_PTH = 18 750
M_MARGIN = 19 531
M_PLEV = 25 000
NOTE: Floating point representation of constants used in clause 3.6:
pth = 2(E_PTH)x(M_PTH/32768).
margin = 2(E_MARGIN)x(M_MARGIN/32768).
plev = 2(E_PLEV)x(M_PLEV/32768).
Test if acf0 < pth; if yes set thvad to plev:
comp = 0;
IF ( e_acf0 < E_PTH ) THEN comp = 1;
IF ( e_acf0 == E_PTH ) THEN IF ( m_acf0 < M_PTH ) THEN comp =1;
IF ( comp == 1 ) THEN
| e_thvad = E_PLEV;
| m_thvad = M_PLEV;
| EXIT; /continue with clause 3.7/
Test if an adaptation is needed:
comp = 0;
IF ( ptch == 1 ) THEN comp = 1;
IF ( stat == 0 ) THEN comp = 1;
IF ( tone == 1 ) THEN comp = 1;
IF ( comp == 1 ) THEN
| adaptcount = 0;
| EXIT; /continue with clause 3.7/
Incrementation of adaptcount:
adaptcount = add( adaptcount, 1 );
IF ( adaptcount <= 8 ) THEN EXIT; /continue with clause 3.7/
Computation of thvad‑(thvad/dec):
m_thvad = sub( m_thvad, (m_thvad >> 5 ) );
IF ( m_thvad < 16384) THEN
| m_thvad = m_thvad << 1;
| e_thvad = sub( e_thvad, 1 );
Computation of pvad*fac:
L_temp = L_add( m_pvad, m_pvad );
L_temp = L_add( L_temp, m_pvad );
L_temp = L_temp >> 1;
e_temp = add( e_pvad, 1 );
IF ( L_temp > 32767 ) THEN
| L_temp = L_temp >> 1;
| e_temp = add( e_temp, 1 );
m_temp = L_temp;
Test if thvad < pvad*fac:
comp = 0;
IF ( e_thvad < e_temp) THEN comp = 1;
IF (e_thvad == e_temp) THEN IF (m_thvad < m_temp) THEN comp =1;
Computation of minimum (thvad+(thvad/inc), pvad*fac) if comp = 1:
IF ( comp == 1 ) THEN
| Compute thvad +(thvad/inc).
| L_temp = L_add( m_thvad, (m_thvad >> 4 ) );
| IF ( L_temp > 32767 ) THEN
| | m_thvad = L_temp >> 1;
| | e_thvad = add( e_thvad,1 );
| ELSE m_thvad = L_temp;
| comp2 = 0;
| IF ( e_temp < e_thvad) THEN comp2 = 1;
| IF (e_temp == e__hvad) THEN IF (m_temp<m_thvad) THEN comp2 = 1;
| IF ( comp2 == 1 ) THEN
| | e_thvad = e_temp;
| | m_thvad = m_temp;
Computation of pvad + margin:
IF ( e_pvad == E_MARGIN ) THEN
| L_temp = L_add(m_pvad, M_MARGIN);
| m_temp = L_temp >> 1;
| e_temp = add( e_pvad, 1 );
ELSE
| IF ( e_pvad > E_MARGIN ) THEN
| | temp = sub( e_pvad, E_MARGIN );
| | temp = M_MARGIN >> temp;
| | L_temp = L_add( m_pvad, temp );
| | IF ( L_temp > 32767) THEN
| | | e_temp = add( e_pvad, 1 );
| | | m_temp = L_temp >> 1;
| | ELSE
| | | e_temp = e_pvad;
| | | m_temp = L_temp;
| ELSE
| | temp = sub( E_MARGIN, e_pvad );
| | temp = m_pvad >> temp;
| | L_temp = L_add( M_MARGIN, temp );
| | IF (L_temp > 32767) THEN
| | | e_temp = add( E_MARGIN, 1);
| | | m_temp = L_temp >> 1;
| | ELSE
| | | e_temp = E_MARGIN;
| | | m_temp = L_temp;
Test if thvad > pvad + margin:
comp = 0;
IF ( e_thvad > e_temp) THEN comp = 1;
IF (e_thvad == e_temp) THEN IF (m_thvad > m_temp) THEN comp =1;
IF ( comp == 1 ) THEN
| e_thvad = e_temp;
| m_thvad = m_temp;
Initialize new rvad[0..8] in memory:
normrvad = normrav1;
|= FOR i = 0 to 8:
|= rvad[i] = rav1[i];
|= NEXT i:
Set adaptcount to adp + 1:
adaptcount = 9; |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.7 VAD decision | This clause only outputs the result of the comparison between pvad and thvad using the pseudo‑floating point representation of thvad and pvad. The values e_pvad and m_pvad are computed in clause 3.1 and the values e_thvad and m_thvad are computed in clause 3.6.
vvad = 0;
IF (e_pvad > e_thvad) THEN vvad = 1;
IF (e_pvad == e_thvad) THEN IF (m_pvad > m_thvad) THEN vvad =1; |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.8 VAD hangover addition | This clause finally sets the vad decision for the current frame to be processed.
IF ( vvad == 1 ) THEN burstcount = add( burstcount, 1 );
ELSE burstcount = 0;
IF ( burstcount >= 3 ) THEN
| hangcount = 5;
| burstcount = 3;
vad = vvad;
IF ( hangcount >= 0 ) THEN
| vad = 1;
| hangcount = sub( hangcount, 1 ); |
ad3845652ec99fdffd75badceec5143f | 06.32 | 3.9 Periodicity updating | This clause must be delayed until the LTP lags are computed by the RPE‑LTP algorithm. The LTP lags called Nc in the speech encoder are renamed lags[0..3] (index 0 for the first sub‑ segment of the frame, 1 for the second and so on).
Loop on sub‑segments for the frame:
lagcount = 0;
|= FOR i = 0 to 3:
|= Search the maximum and minimum of consecutive lags.
|= IF ( oldlag > lags[i] ) THEN
|= | minlag = lags[i];
|= | maxlag = oldlag;
|= ELSE
|= | minlag = oldlag;
|= | maxlag = lags[i] ;
|=
|= Compute smallag (modulo operation not defined ):
|=
|= smallag = maxlag;
|== | FOR j = 0 to 2:
|== | IF (smallag >= minlag) THEN smallag =sub( smallag, minlag);
|== | NEXT j;
|=
|= Minimum of smallag and minlag ‑ smallag:
|=
|= temp = sub( minlag, smallag );
|= IF ( temp < smallag ) THEN smallag = temp;
|= IF ( smallag < 2 ) THEN lagcount = add( lagcount, 1 );
|= Save the current LTP lag.
|= oldlag = lags[i];
|= NEXT i:
Update the veryoldlagcount and oldlagcount:
veryoldlagcount = oldlagcount;
oldlagcount = lagcount; |
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. |
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