Manual - SQuad Voice Test Result Description

WHENQUALITYMATTERS
SQuad Voice Measurement Description

Manual
November 2010
SwissQual License AG
Allmendweg 8 CH-4528 Zuchwil Switzerland
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www.swissqual.com

Part Number: 16-100-200047-3 Rev 2.20

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SQuad Voice Measurement Description Manual
2000 - 2010 SwissQual AG
Contents |
CONFIDENTIAL MATERIALS
ii
Contents
1 Introduction .......................................................................................................................................... 1
2 SQuad Listening Quality ..................................................................................................................... 2
Introduction ............................................................................................................................................ 2
Speech Quality Definition ...................................................................................................................... 2
SQuad Method ....................................................................................................................................... 2
MOS Rating ........................................................................................................................................... 3
Speech and Noise Level Received Signal ......................................................................................... 4
Channel Gain ......................................................................................................................................... 4
Clipping .................................................................................................................................................. 5
DC-Offset ............................................................................................................................................... 5
Frequency-Shift ..................................................................................................................................... 5
Delay Spread (Voice Jitter) .................................................................................................................... 6
Speech Threshold .................................................................................................................................. 6
Degradations ......................................................................................................................................... 7
AGC Problems ................................................................................................................................. 7
Speech Enhancer / Noise Suppressors ........................................................................................... 8
Impulsive Noise ................................................................................................................................ 8
Background Noise ............................................................................................................................ 8
Interruptions...................................................................................................................................... 9
VAD resp. Silence Suppression Problems ..................................................................................... 10
Variable Delay (Voice Jitter) ........................................................................................................... 10
Delays Deviation ............................................................................................................................ 10
Frequency Shifts ............................................................................................................................ 11
Quality Code ................................................................................................................................... 12
Option: P.862 'PESQ' .......................................................................................................................... 13
3 SQuad Noise Suppression ............................................................................................................... 14
Introduction .......................................................................................................................................... 14
Listening Quality .................................................................................................................................. 14
NS-Speech Power Classes ................................................................................................................. 15
SNRI, Signal-to-Noise Ratio Improvement .......................................................................................... 17
NPLR, Noise Power Level Reduction .................................................................................................. 17
SPLR, Signal Power Level Reduction ................................................................................................. 17
Overall NS Quality ............................................................................................................................... 19
Quality Index ........................................................................................................................................ 20
Convergence Time .............................................................................................................................. 21
Noise Reduction/Suppression Test ..................................................................................................... 23
Contents |
iii
Examples ............................................................................................................................................. 24
Evaluation of the transmitted signal ..................................................................................................... 24
Evaluation of the transmitted signal ..................................................................................................... 25
4 DTMF Tests ........................................................................................................................................ 26
Introduction .......................................................................................................................................... 26
DTMF-Test Overview........................................................................................................................... 26
Criterions ............................................................................................................................................. 27
Results ................................................................................................................................................. 27
5 SQuad Advanced Echo Check (Passive Test) ................................................................................ 30
Introduction .......................................................................................................................................... 30
Echo Measurement.............................................................................................................................. 30
Measurement Results .......................................................................................................................... 30
6 SQuad Advanced Echo Check (Active Test) .................................................................................. 34
Introduction .......................................................................................................................................... 34
Echo Measurement.............................................................................................................................. 34
Measurement Results Echo Evaluation............................................................................................ 34
Measurement Results Listening Quality ........................................................................................... 36
7 Round Trip .......................................................................................................................................... 37
Introduction .......................................................................................................................................... 37
The Round Trip Method ....................................................................................................................... 37
Results ................................................................................................................................................. 37
References .......................................................................................................................................... 37
A Appendix ............................................................................................................................................ 38
Abbreviations ....................................................................................................................................... 38
Figures
Figure 2-1 Block Diagram of the SQuad ............................................................................................................ 3
Figure 2-2 Main outcomes of SQuad-LQ ........................................................................................................... 3
Figure 2-3 Typical MOS-LQ values for Different Codecs .................................................................................. 4
Figure 2-4 Frequency Shift ................................................................................................................................ 6
Figure 2-5 Histogram of Noised Speech Sample .............................................................................................. 7
Figure 2-6 Level Chart with AGC ....................................................................................................................... 8
Figure 2-7 Similarity Chart with Impulsive Noise ............................................................................................... 8
Figure 2-8 Background Noise ............................................................................................................................ 9
Figure 2-9 Level Chart with Handover ............................................................................................................... 9
Figure 2-10 Time Clipping ............................................................................................................................... 10
Figure 2-11 Variable Delay, Voice Jitter .......................................................................................................... 10
Figure 2-12. Example for Variable Delay, which shows that Block B is delayed for 244 samples to the left
Contents |
iv
(arrives earlier when compared with the same reference block). Block B arrives later by 244 samples. ....... 11
Figure 2-13 Frequency Shift ............................................................................................................................ 12
Figure 2-14 P.862 result representation .......................................................................................................... 13
Figure 2-15 P.862 vs. P.862.1 scale transformation ....................................................................................... 13
Figure 3-1 The Principle of MOS Calculation in Squad-NS. ............................................................................ 15
Figure 3-2 The Five Energy Windows for 16 bit Digital System (90.3 dB dynamics) ...................................... 16
Figure 3-3 Speech Power Class Chart ............................................................................................................ 16
Figure 3-4 SPLR Calculation out of Five Values Calculated in Five Different Energy Windows. ................... 18
Figure 3-5 Speech Power Class Chart ............................................................................................................ 19
Figure 3-6 Overall NS Quality .......................................................................................................................... 19
Figure 3-7 Calculation of Quality Index ........................................................................................................... 20
Figure 3-8 Some Experimental Results for DifferentConfigurationofNR in the Network ................................. 21
Figure 3-9 MOS vs. MOSobj & Quality Index .................................................................................................. 21
Figure 3-10 Example of Convergence Time Evaluation .................................................................................. 21
Figure 3-11 Filtered Difference Envelope is Compared with the Threshold Value ......................................... 22
Figure 3-12 Five Point Analysis of the Difference Envelope during Decision on Noise Reduction State ....... 22
Figure 3-13 Additional Condition before a Final Decision is Calculated. ........................................................ 23
Figure 3-14 Typical Reference Signal with White Noise Added ...................................................................... 23
Figure 3-15 Noise Suppression Applied on Signal .......................................................................................... 24
Figure 3-16 Noise Reduction Applied on Signal .............................................................................................. 24
Figure 3-17 NS Signal Envelope ..................................................................................................................... 24
Figure 3-18 Results presentation within NQDI ................................................................................................ 25
Figure 4-1 Allocation of Frequencies to the Various Digits and Symbols of a Push-button Set ..................... 26
Figure 4-2 Block Diagram of DTMF-Test ......................................................................................................... 27
Figure 5-1 Results of SQuad-AEC Test, shown in NQDI presentation ........................................................... 31
Figure 5-2 EOR (Echo Objection Rate) derived from G.131 ........................................................................... 32
Figure 5-3 Echo Loss during scanning versus echo delay .............................................................................. 33
Figure 6-1 Results presentation SQuad-AEC active ....................................................................................... 35
Figure 6-2 Echo Loss as profile versus echo delay ......................................................................................... 35
Figure 6-3 Result presentation in an echo free/non echo detectable connection ........................................... 36
Figure 7-1 Detail of the NQDI Representation ................................................................................................ 37
Tables
Table 2-1 Example for Variable Delay where five blocks are elayed at different offsets regarding Reference
Speech Sample ............................................................................................................................................... 11
Table 3-1 Energy Windows used in Calculation of SPLR and PLR ................................................................ 16
Table 4-1 DTMF Result Code .......................................................................................................................... 27
Table 7-1 One Way Delay Quality Classes ..................................................................................................... 37

Chapter 1 | Introduction
1
1 Introduction
This document describes the parameters that are measured with the SwissQual QoS Measurement
System. It also describes briefly the used algorithms as well as some background information with regards
to the causes of different kind of quality degradations. The screenshots are made from the SwissQuals
Post Processing System NQDI.

Chapter 2 | SQuad Listening Quality
2
2 SQuad Listening Quality
Introduction
For network operators or equipment manufacturers, it is important to know where and why the speech
quality may be degraded. Since speech quality is a major factor determining customer satisfaction, encoding
techniques must be designed for optimal speech quality. In order to assess the quality of speech encoding
techniques, large-scale auditory tests are commonly employed. However, it is practically impossible to
reproduce results obtained in such way. Furthermore, such results are depending on the level of motivation
of the individual test candidates. Therefore, it is a big advantage to have an instrumental method capable of
physically measuring speech quality parameters and producing results, which correlates as closely as
possible with subjectively acquired results. The perfect transmission of speech via a telecommunications
channel with a bandwidth of 0.3 - 3.4 kHz results in a sentence intelligibility of approx. 98%. The speech
coders introduced for handsets used in digital mobile radio networks also further impair intelligibility. Speech
quality is a vague term compared with bit rate, echo or loudness. Since customer satisfaction can be
measured directly by the quality of the transmitted speech, encoding techniques must be selected and
optimized based on their speech quality.
Speech Quality Definition
Speech Quality is defined as a measure of a listeners satisfaction based on his experience and expectation
regarding voice communication. It is generally expressed as a Mean Opinion Score (MOS). This
measurement denotes the average of many individual opinions on speech quality, which are obtained from a
representative number of listeners. Speech quality is a complex psycho-acoustic phenomenon within the
process of human perception. As such, it is necessarily subjective. Most objective algorithms are based on a
comparison between a reference sample and a coded version of the reference.
SQuad Method
SQuad consists of three main parts. First, a pre-processing unit adjusts reference and coded sample.
Then, an auditory model is used to reduce both samples to their perceptually relevant features. Finally, an
assessment unit evaluates the perceptual difference between reference and coded sample and outputs the
result as a MOS value.
A speech sample is transmitted over a line with generally unknown combination of speech coders. This
speech sample is available in digital form. The sampling frequency is 8 kHz and the digital quantization is 16
bits. As an initial step, the source speech signal is read into the vector x(i) and the coded speech signal into
the vector y(i). These speech signals are synchronized with respect to both time and level. The DC offset
must be removed from every sample. In addition, the signals are normalized to a common RMS (Root Mean
Square) level, to ensure that the constant amplification factor is not taken into account.
The signals are split into processing units of 32 ms duration, also called Frames. The unit overlap is 50%.
During the first processing step, the frame is multiplied by a hamming window. The source signal x(t) in the
time domain is now transformed to the frequency domain using a discrete Fourier transform, followed by
computation of the squared magnitude FFT spectrum. Both signals are filtered using a filter equivalent to the
receiving curve of the corresponding telephone handset. A rough approximation of the time masking is
already achieved through the frame overlapping during the signal pre-processing. The comparison method of
SQuad is based on the following principle; Signal parts with high energy are more important for the perceived
speech quality. A similarity coefficient for reference and impaired signal is computed for 4 different energy
thresholds. Only the parts of the signal exceeding the respective threshold are considered. This can be
viewed as a multi-resolution analysis with respect to signal energy. The overall similarity is then computed
using the coefficients from all thresholds. A polynomial is used to transform the comparison result to the ITU
MOS scale. The length of the speech sample varies between 4 and 30 seconds.
3

Figure 2-1 Block Diagram of the SQuad
MOS Rating
Speech Quality is defined as a measure of a listeners satisfaction and is generally expressed as a Mean
Opinion Score (MOS). SQuad delivers MOS rating as one number, ranging 1 to 4.5, fully in accordance to
the Listening Scale defined in ITUs P.800 recommendation. This is not exactly the same scope as MOS
which is defined with 1-5. This is allowed since based on subjective tests used for the validation of Squad-LQ
the values above 4.5 have almost never appeared.
As described in ITUs P.800 recommendation Annex B.4.5, various five-point category-judgment scales may
be used for different purposes. The Listening Only quality scale is the most frequently used for ITU-T
applications:
Quality of the speech Score
Excellent 5
Good 4
Fair 3
Poor 2
Bad 1
The following picture gives an overview about the obtained results in the main section of NQDI:

Figure 2-2 Main outcomes of SQuad-LQ
- echo
- call setup quality
Time &
Level
alignment
Referenc
e
signal
Degraded
signal
Frequency
equalization
Psychoacoustic
s
modelling
IRS-filtering &
BG Noise
detection
Psychoacoustic
s
modelling
Listening only
Quality
estimation
Network
Informati
on

Other
measured
data
Overall
Audio
Quality
- round-trip delay
- jitter
Q
N U T

Listening only
Quality
Estimation
LQ
Q
4
3.4 3.4 AMR 4.75
3.7 3.8 AMR 7.4
3.9 4.0 AMR 12.2
3.3 3.4 GSM-HR
3.9 4.0 GSM-EFR
3.5 3.5
G.723.1
(6.3)
3.7 3.8 G.729
4.4 4.3 G.711
Typical
SQuadLQ
Typical MOS
Value Codec
3.4 3.4 AMR 4.75
3.7 3.8 AMR 7.4
3.9 4.0 AMR 12.2
3.3 3.4 GSM-HR
3.9 4.0 GSM-EFR
3.5 3.5
G.723.1
(6.3)
3.7 3.8 G.729
4.4 4.3 G.711
Typical
SQuadLQ
Typical MOS
Value Codec

Figure 2-3 Typical MOS-LQ values for Different Codecs
Speech and Noise Level Received Signal
Within the SQuad-LQ algorithm itself also the Active Speech Level (acc. to ITU-T P.56) of the received
signal is calculated. This value describes the r.m.s level of active speech parts only. Speech pauses will not
influence that value. The result is given in dBov. The level should be in a range from -20 -38dBov.
Related to a sending level of -26dBov it corresponds to a gain/attenuation of +6 / -12 dB.
The Noise Level describes the noise floor of the received signal in speech pauses in dBov too. In normal
noise-free connections a Noise Level of below -55dBov can be obtained. Please note that is in an un-
weighted level, a common A-weigthing is not applied.
Both results are used to calculate a basic signal-to-noise ratio, which describes the distance between the
speech level and the noise floor.
Channel Gain
This is a value in dBr, which shows the power level of the received signal relatively to the reference (input)
signal. Because, SQuad-LQ is applied to the electrical interfaces of the connection, the terminal depending
Send Loudness Rating (SLR) and the Receive Loudness Rating (RLR) as well are modelled in SQuad-LQ
itself. In Principle, SQuad-LQ is connected to the so-called 0dbr-point of the networks input. At this 0dBr
point a nominal level of -26dBov (corresponds to -20dBm at a four-wire 600 Ohms interface) will be inserted.
The Channel Gain reflects only gains or attenuation caused by network (exception: attenuating PSTN
subscriber loops). It is close to the so-called JLR (Junction Loudness Rating) but does not apply any spectral
weighting.
In a transparent ISDN connection the Channel Gain should be around 0 dB. In principle also in a Mobile-to-
ISDN or Mobile-to-Mobile connection this value should be around 0dB too. Caused by individual signal
amplifications of cellular network providers this value might differ. Mainly they amplify the signals, so a gain
in the positive range can be observed. If a overall gain of 6dB is exceeded, amplitude clipping may occur.
This will lead like in a real call to quality impacts and result in a lower SQuad-LQ score.
On the other way around, an attenuating PSTN subscriber loop may lead to negative Channel Gains
because it is part of the evaluated transmission chain. Like a PSTN phone, which is more sensitive, also
SQuad-LQ gain internally such attenuated signals to a nominal level of -26dBov (corresponds to 79dB sound
pressure level at the subscribers ear).
To inform the user of SQuad-LQ, within NQDI Channel Gains outside of the expected range are highlighted.
The expected range is here +6-9dB and in an extended range down to -15dB.
The Channel Gain is available as a single overall value in dBr (total Gain) but also as a range of values in
5
the time domain (every 16ms) like a an attenuation profile. Based on this attenuation profile values a chart
can be created providing information on:
AGC (Adaptive Gain Control) Elements that are not working correctly
Level Jumps (for example after a handover)
Level Interruptions (for example interruptions in the audio path or during handovers)
Clipping
Temporal Speech Clipping (also called front-end clipping) is the loss of speech frames. It may occur
when voice activity detection is used, when Digital Circuit Multiplication Equipment (DCME) is used or
during uncontrolled slips. Time clipping is presented as clipped frames in a function of time.
Clipping is an annoying phenomenon that cuts off a bit of speech in the instant it takes for the transmitter to
detect presence of speech. It is almost impossible to eliminate clipping in a traditional circuit-switched voice
conversation. Using circuit switching, the transmitter is not turned on until sound is detected, and by then, a
piece of the speech has been clipped off. SQuad detects this clipping and generates the results as a
distribution of time. The resolution of the clipping measurement is 8 milliseconds. First, the mean energy per
8 milliseconds is calculated. The energy values are then saved for each frame (both reference and coded).
After the whole speech sample has been processed, the post processing of time clipping data is done. There
are some simple rules during this post-processing:
Time clipping can only occur during transitions pause-speech.
Minimum pause length must be reached. In our case, it is 64 milliseconds.
The difference Energy (ref) Energy (cod) must be at least 10 dB.
Clipped frames are succeeding frames.
The clipping measurement values are indicated as an average % value per sample (number of active speech
frames / number of clipped speech frames) and as a time domain distribution. Time Clipping in SQuad-LQ is
calculated each 8 ms, but only an average value of two succeeding frames is reported in output file.
DC-Offset
This number shows the DC-Offset of the coded signal in percentage. This is an important piece of
information if the measured speech quality is lower than expected. Various interface problems (impedance,
coding technique, HW) can produce DC-offset discrepancies.
DC Offset is calculated as
100 * average_audio_voltage / Max_audio_voltage
Max_audio_voltage for 16 bit digital resolution is equal 2^
15
(32768).
For example: average_audio_voltage=300 results in DC_Offset=100*300/32768=0.91%
Frequency-Shift
A low bit rate encoder can move the formants (spectral peaks) of the speech. This degradation can be
described as frequency shift of one or more components of the source signal. This drift is measured as
a percentage of moved frequency components in the speech active phases. The result is a number of pos-
and neg -shifted frames in %, reflected in a compressed frequency (bark). Figure 2-2 shows a typical
situation for one processing buffer of voice signal (32ms).
6

Figure 2-4 Frequency Shift
For the detection of the frequency shift, the peaks above the loudness threshold in both reference- and
degraded-signals are analyzed. The threshold for compressed loudness is set to 10. The position of each
peak in the reference is compared with the position of the peak in the coded signal (within +/- 1Bark).
Frequency shift is found if the location of the two peaks is not at the same. The amplitude of the coded and
reference loudness must not be equal but above the threshold value. This is allowed because the level- and
frequency-alignment is done previously in a separate module.
Typical Network Elements that are responsible for frequency shift are:
Very low bit rate vocoders
Speech enhancer (Noise suppressors)
Non linear filter elements
Delay Spread (Voice Jitter)
The first stage in SQuad-LQ is the time alignment. This stage is able to deal with variable delays, which can
occur in packet networks, normally indicated with big jitter/delay or packet loss values. It collects information
about shifted frames by comparison with reference speech sample. The result of this alignment is a delay
distribution of the coded signal. A histogram will be presented which shows the number of speech frames
reflected in arrival time (delay) in milliseconds. The results are generated for each 32ms frame.
Speech Threshold
This is a value in dBov, which shows a level of the speech in a coded signal. The measurement is based on
building of r.m.s. histograms for both coded and reference signals. dBov means decibel relative to a digital
over-load point. The range for this value is 90 to 0 dBov. For signals containing background noise, this
value is between 55 to 40 dBov.
A histogram evaluates an individual frequency for a set of data bins. The result is a number of occurrences of
a value in a data set. A histogram table presents the energy-grade boundaries and the number of scores
between the lowest bound and the current bound.
7
Energy Histogram for noisy signal
0
5
10
15
20
25
-
5
3
.
0
-
5
1
.
5
-
4
9
.
9
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4
8
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3
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7
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5
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1
.
9
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.
4
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2
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0
RMS of the coded Signal (dB)
C
o
u
n
t
Noise position
Bound position
Speech level

Figure 2-5 Histogram of Noised Speech Sample
In Figure 2-4, is shown an example of a histogram for noisy speech signal of 10 seconds duration. In SQuad-
LQ, internally, the histogram is presented with 50 bins between minimum r.m.s and maximum r.m.s. values.
There are two maxima, one for speech-pauses and one for speech active intervals. In our example, the first
maximum is found at 45.1 dB, which is level of silent intervals. Second peak is at about 26 dB which is
speech active level. Speech threshold measured in SQuad-LQ is defined as a boundary between these
two peaks (Bound position).
Degradations
The below list present some possible degradation reasons for the Listening Quality Value using a clean
reference sample:
AGC (Adaptive Gain Control) Elements
Speech Enhancer / Noise Suppressors
Impulsive Noise
Background Noise
Interruptions
VAD (Voice Activity Detectors)
Variable Delay or Jitter in Packet Networks
AGC Problems
Indications: LQ less than expected, Level Chart indicates an abnormal level trend.
Example of an AGC of a mobile handset that attenuates too strong toward the end of a sample:
8

Figure 2-6 Level Chart with AGC
Speech Enhancer / Noise Suppressors
Indications: LQ less than expected. Similarity Chart shows a bigger degradation over the complete speech
signal. This must be checked with the SQuad Noise Suppression Test.
Impulsive Noise
Indications: LQ less than expected. Similarity Chart shows a lot of quite big degradation peaks.

Figure 2-7 Similarity Chart with Impulsive Noise
Background Noise
Indications: LQ less than expected. Signal Envelope Chart shows some additional energy during the
speech pause.

9
Example:

Figure 2-8 Background Noise
Interruptions
Indications: LQ less than expected. Similarity Chart shows blue bars and Signal Envelope indicates a
peak drop.
Example of an Interruption due to a Handover (interruption is indicated in blue):

Figure 2-9 Level Chart with Handover
Interruption measurement is based on processing frames of 32ms duration.
Such frames are divided into 16 sub-frames (2ms) in order to achieve better resolution. For each sub-frame,
the signal level for both reference and degraded is calculated. Interruption flag for sub-frame is set to TRUE
if the level in the reference signal is higher then 35 dBov (r.m.s. = 400) and the level in the degraded signal
is lower then 61 dBov (r.m.s. = 20).
The result is the ratio of sum of sub-frames with signal interruption and total nr. of sub-frames (16).
16
_ _
_
IntFrames Of Nr
result on Interrupti =
Interruption result is in the range 0 and 1 with step=1/16. If only one sub-frame is lost (interrupted) in the
signal, then is Interruption=1/16=0.0625. When the signal in all sub-frames is deleted (lost) then is
Interruption=1.
10
VAD resp. Silence Suppression Problems
Indications: LQ less than expected. Average clipping values are high; Clipping Chart shows some
significant clipping.
Example: Clipping at the beginning of the sentence. At the top is shown the signal envelope chart, at the
bottom the clipping chart.

Figure 2-10 Time Clipping
Variable Delay (Voice Jitter)
Indications: LQ less than expected. Variable Delay Chart shows some delay values. This typically happens
if there was a packet network (or backbone) used and if there were Jitter buffers used.

Figure 2-11 Variable Delay, Voice Jitter
Delays Deviation
DelaysDeviation is placed in the section !SQuad_LQ_AVG (in Squad result file) and is defined as an
absolute value of the standard deviation of block delays (D), divided by an average of block delays [in
samples]. The duration of one sample at 8000 Hz, sampling frequency is 125 s. DelaysDeviation shows
the smoothness of an array of delays. Small DelaysDeviation value means there is a uniform delay
distribution, where a large value indicates a big delay-jitter like in IP networks. For only one single delay, this
value is equal zero.
11
(
=
) (
) (
D average
D stdev
fabs ation DelaysDevi
Example: Coded file has fixed offset of 1024 samples to the reference file. Six blocks with different variable
delays are found with Squad-LQ:
Table 2-1 Example for Variable Delay where five blocks are elayed at different offsets regarding Reference Speech
Sample
Block Delays (D) in samples D
All 1024 0
A 780 -244
B 1024 244
C 1536 512
D 1024 -512
E 1304 280
Stdev(D)=
( )
N
D D N

2
2
=241.53 average(D)=

D
N
1
=1115.3
DelaysDeviation=241.53/1115.3=0.217
Delay Spread is also another important parameter, which describes the maximum delay amplitude
calculated over all single group delays. Based on the example above, we can calculate new Delay Values
(D), which are scaled D values by subtracting a fix delay from all other values.
delay fix D D
i
i _
"
=
For example, fix_delay=1024 samples.

Figure 2-12. Example for Variable Delay, which shows that Block B is delayed for 244 samples to the left (arrives earlier
when compared with the same reference block). Block B arrives later by 244 samples.
Delay Spread is calculated as a distance between the minimum and the maximum block delay. In our
example, minimum value is 512 samples and maximum is +512 samples. So the distance between max
and min equals 1024 samples. This is then converted to time in ms.
duration smp smp Delay d DelaySprea _ _ =
Fs
duration smp
1
_ =
Hz Fs 8000 =
In the calculation for our example, we get the value for DelaySpread=1024/8000=128 ms.
Frequency Shifts
The distribution of the frequency shifts is shown in the histogram below, with the number of frames in which a
shift at a certain frequency occurred. The diagram covers the whole range of frequencies in steps of 31.25
12
Hz.

Figure 2-13 Frequency Shift
Quality Code
The thresholds for each degradation descriptor is, as follows:
MOS-drops
Quality distribution is unsteady such as during handovers or interruptions.
Received signal level out of recommended range
The level difference to the reference level exceeds +9dB or falls below -12dB.
Signal interruptions
Temporal clipping for more then 8 ms.
High DC-Offset
Malfunction of terminal or interface card. DC-Offset > 0.2%.
Variable delay
Indicates possible packet-switched transmission.
Variable delay during speech
Same as Variable delay but occurring during speech active intervals.
Background noise
High level of circuit noise. Higher then 50 dBov.
Impulse noise
Relay/switching problems detected. More then 1 pulse / second.
Low bitrate coding / coding artefacts
Low bit rate coding scheme has been used (e.g. Less then 8 kbit/s) or residual errors from decoding are
introduced (e.g. by frame loss concealment).
Not Specified
signalizes that the speech quality is degraded but no outstanding reason for that degradation could be
classified.
OK
shows that the speech quality is nearly non-degraded
Furthermore, special problems in the audio-path will be reported:
Silence/Audio Level Too Low
There is no signal activity in the audio path or the signal level is below -45dBov. SQuad-LQ will not
calculated since it will lead to misleading results.
Corrupted Signal/Wrong Reference
Here the received audio signal is heavily corrupted (e.g. only partly transmitted or the audio stream was lost
completely). Such a behaviour can observed e.g. during a call drops. Normally, SQuad will score those
13
signals with close to 1.0. For statistical reasons, NQDI allows the exclusion of such results from the
reporting.
This indicator will also signalize if a wrong reference signal was used for SQuad.
Option: P.862 'PESQ'
Optionally, the SQuad-LQ framework can also include ITU-T P.862 'PESQ' an additional model for objective
speech quality prediction. The principle function of P.862 as a psycho-acoustic driven comparison method is
very close to SQuad-LQ and so the received signals can be evaluated also by P.862.
The ITU-T Recommendation P.862 was finalized 1999 and approved in February 2000. It was trained over
huge amount of databases mainly from codec standardization activities in ITU-T.
If the P.862 option is used in SQuad-LQ, the SQuad-framework will report two additional quality results:
P.862 score: raw outcome of the ITU-T algorithm
Listening Quality (P.862.1) : transformed result according to P.862.1 into a MOS-scale from 15

Figure 2-14 P.862 result representation
That both results are basing on the same algorithm. The transformation according to P.862.1 describes only
a scale mapping.

Figure 2-15 P.862 vs. P.862.1 scale transformation
Note: The P.862 results are a bit lower in tendency compared to SQuad-LQ especially in the range from
3.0 4.0. It is mainly caused by a high sensitivity of P.862 regarding clipping and time-variant filtering.
It has taken also into account that P.862 does not rate any linear distortions such as frequency responses.
Those linear distortions will be compensated completely by P.862 itself before the quality prediction starts.
The P.862 option has to be enabled by the test-type 'Speech-P.862' and requires a special software key.

1.0
1.5
2.0
2.5
3.0
3.5
4.0
4.5
5.0
-0.5 0.0 0.5 1.0 1.5 2.0 2.5 3.0 3.5 4.0 4.5 5.0
P.862
P
.
8
6
2
.
1
Scale limit = 4.5
Chapter 3 | SQuad Noise Suppression
14
3 SQuad Noise Suppression
Introduction
The noise suppression is a feature designed to enhance speech quality in a range of environments where
there is significant (acoustic) background noise. The noise suppression function is a pre-processing module
that is used to improve the signal to noise ratio of a speech signal prior to voice coding.
For noise suppressors, there are certain requirements that need to be fulfilled:
- The noise suppression function must not have a statistically significant distorting effect on clean speech in
comparison with the performance of the speech codec without noise suppression applied.
- The noise suppression function must not introduce any degradation of speech and no undesirable effects
in the residual noise when there is (acoustic) background noise in the speech signal.
- DTMF and other signalling tones transmission performance during the application of noise suppression
shall be no worse than when noise suppression is turned off.
The above requirements are all checked with SQUAD Noise Suppression test.
The algorithm measures the Noise Power Level Reduction (NPLR) and Signal-to-Noise Ratio
Improvement (SNRI), similar to the definitions in ETSI STC SMG11 (GSM 06.77) document. A comparison
of the SNRI and NPLR measures are used to acquire an indication of possible speech distortion produced by
the tested NS method.
For the Noise Suppression test, two reference signals are used:
- Clean speech reference
- Clean speech with background noise
The sample with background noise is sent as a test sample.
Listening Quality
Speech Quality is measured according to ITUs P.800 where the coded file and the clean reference are
inputs for SQuad LQ algorithm. The algorithm is elaborated in Section 2.
The Listening Quality evaluation is running twice. The LQ of the noised input signal is estimated and the LQ
of the de-noise output signal as well. From both results the change of the speech quality is derived.
15

Figure 3-1 The Principle of MOS Calculation in Squad-NS.
First, the internal reference (MOS_ref) is calculated. Degraded signal is assessed by comparing it with the
clean reference. Result is presented on CCR scale by subtracting MOS_ref from the measured MOS.
Speech quality measurement in noisy environment is done by sending the noisy reference through the
network under test. The noisy reference is made by adding a noise signal to the clean reference. Comparing
the clean reference with the coded signal would not produce stable results, since the SNR of the noisy
reference will impact the results. To make this measurement independent from the reference properties,
MOS_ref is calculated first. MOS_ref defines the reference speech quality, which will be measured in
degraded signal if there would not be any degradations or improvements in the network. This value is mostly
lower then 4.5 (excellent quality) because of noise influence. The range of MOS generated by Squad-NS is
3.5 to +3.5, which is slightly different from the ITU definition.
Comparison Category Rating (CCR)
The range of the Comparison Category Scale (CCR) as defined in source ITU P.800:
3: Much Better
2: Better
1: Slightly Better
0: About the Same
1: Slightly Worse
2: Worse
3: Much Worse
The CCR methods are particularly useful for assessing the performance of telecommunications systems
when the input has been corrupted by background noise. An advantage of the CCR method over the other
scales is the possibility to assess speech processing that either degrades or improves the quality of the
speech.
NS-Speech Power Classes
SNRI and SPLR are calculated once as overall average values and once per speech power class. There are
6 different power classes:
16
Definition:
ETSI
Performance objective:
Speech level = -26 dBov, determined according ITU-T P.56
Table 3-1 Energy Windows used in Calculation of SPLR and PLR
Range Description Level class
3 high power frames > speech level 1 dB h
2 medium power frames > speech level 10 dB m
1 low power frames > speech level 16 dB l
0 noise only frames < speech level 19 dB
> speech level 34 dB
noise
-1 pause frames < speech level 34 dB p
-2 not used for calculation < speech level 16 dB
> speech level 19 dB
unused

Figure 3-2 The Five Energy Windows for 16 bit Digital System (90.3 dB dynamics)
Example of the Signal Level Group Chart:

Figure 3-3 Speech Power Class Chart
17
SNRI, Signal-to-Noise Ratio Improvement
Definition:
ETSI
6 dB or higher
Formula:
SNRI
x
= per speech power class,
SNRI = 1 / (N
h
+ N
m
+ N
l
) * (N
h
* SNRI
h
+ N
m
* SNRI
m
+ SNRI
l
* N
l
)
SNRI
x
= 10 * ( log(SNR
cod_x
) log(SNR
ref_x
) )
SNR
y_x
= cod / nse
Range Description Quality
0 SNR from coded and reference signal are equal l no improvement
< 0 SNR
ref
> SNR
cod
, lower SNR in coded signal worse
> 0 SNR
ref
< SNR
cod
, higher SNR in coded signal better
NPLR, Noise Power Level Reduction
Definition:
ETSI
-7 dB or lower
Formula:
NPLR = 10 * ( log(PL
cod_nse
) log(PL
ref_nse
) )
Range Description Quality
0 Noise levels from coded and reference signal are equal no noise reduction
< 0 PL
ref
> PL
cod
, lower noise level in coded signal good
> 0 PL
ref
< PL
cod
, higher noise level in coded signal bad
SPLR, Signal Power Level Reduction
Both SNRI and NPLR are defined in ETSIs document TS 101 512, V8.0.0. Signal Power Level Reduction
(SPLR) is a SwissQuals improvement of NPLR measurement, where NPLR is a subset of SPLR. SPLR is
the difference between coded and reference energy, calculated separately for each energy window.
Note: Noise reduction should reduce only noise parts in a signal.
The definition of Windows is given in Table 3. The aim of this measurement is to detect the influence of noise
reduction circuits on speech parts of the signal.
Five SPLR values are calculated:
h
SPLR ,
m
SPLR ,
l
SPLR ,
n
SPLR and
p
SPLR .
n
SPLR is equal to NPLR
value. Good noise reduction would generate
h
SPLR closed to zero and
p
SPLR below 10 dB. The trend
curve down through these five values shows the quality and ability of noise reduction circuit to reduce only
18
noisy frames and to keep unchanged the speech active frames. In other words, the first coefficient (a) of the
trend curve y=ax+b must be negative (see example in Figure 16). The SPLR measure in SquadNS
algorithm is equal to this coefficient (a) of the trend curve.

Figure 3-4 SPLR Calculation out of Five Values Calculated in Five Different Energy Windows.
The bottom picture shows good noise reduction, whereas on the right is shown poor noise reduction.
SPLR is then mapped to a new range 1 4.5 (like MOS scale). This mapping from SPLR to
m
SPLR is
shown in Figure 16.
m
SPLR > 2.5 should be achieved for good noise reduction.
19

Figure 3-5 Speech Power Class Chart
Overall NS Quality

Figure 3-6 Overall NS Quality
20
Quality Index

Figure 3-7 Calculation of Quality Index
The calculation of Quality Index is done by using of four input parameters: NPLR, SNRI, SPLR and
MOS_acr
Quality Index was introduced first in SW Release 2.2. Four values: SMOS, SNRI, NPLR and SPLR are
combined into one objective number. SMOS is measured with SQuad-LQ, where the clean reference is
compared with the coded signal. The range for Quality Index is 1 to 4.5 (like for MOS). Rating 1 is standing
for bad quality and 4.5 for excellent one. The following equation shows the calculation of quality index based
on four input parameters previously scaled into range 1-4.5.
MOS SPLR SNRI NPLR Q
MOS m SPLR m SNRI m NPLR idx
+ + + = o o o o
=1
i
o
Note: The Quality Index describes the performance of the noise reduction system in combination with
the network and not the Listening Quality of the de-noised signal.
The following table shows some measurement examples for different network conditions including noise
reduction effects:
21

Figure 3-8 Some Experimental Results for DifferentConfigurationofNR in the Network

Figure 3-9 MOS vs. MOSobj & Quality Index
The Quality Index correlates much better with MOS_CCR than MOSobj based only on speech quality
evaluation.
Convergence Time
For the measurement of the Convergence Time in a noisy signal, the algorithm examines the first two
seconds of the given signal. For the calculations it uses the filtered difference between the coded and the
reference signal (red color, see Figure 3-6).
Figure 3-10 Example of Convergence Time Evaluation
First it checks whether the signal belongs to the noise or pause group and then it compares data with the set
threshold. The threshold is calculated as NPLR + 25 (default, use PERCENT to change) percent of the
difference between the maximum value of the filtered signal in these first 2 seconds and noise level
afterwards (NPLR). If the filtered data is lower than the threshold the first condition for the convergence is
fulfilled (see Figure 3-7).
22
Filtered difference
-25
-20
-15
-10
-5
0
5
1 11 21 31 41 51 61 71 81 91 101 111 121 131 141 151 161 171 181 191

Figure 3-11 Filtered Difference Envelope is Compared with the Threshold Value
The second condition is that the signal has a falling tendency. To verify that we check 5 (default, use
CT_NR_POINTS to change) equally spaced points over the tested convergence time. In case of the falling
signal the difference in values between every two consecutive points has to be less than zero. In Fig. 21, we
see that the difference between signal values in third and forth points is bigger that 0, which signifies raising
tendency of the signal. Here we perform additional check to clarify what is actually going on.
-40
-30
-20
-10
0
10
20
30
1 8
1
5
2
2
2
9
3
6
4
3
5
0
5
7
6
4
7
1
7
8
8
5
9
2
9
9
1
0
6
1
1
3
1
2
0
1
2
7
1
3
4
1
4
1
1
4
8
1
5
5
1
6
2
1
6
9
1
7
6
1
8
3
1
9
0

Figure 3-12 Five Point Analysis of the Difference Envelope during Decision on Noise Reduction State
This test is based on the average level of the signal before and after the first convergence criterion is met. If
the average level of the signal after falling below the threshold is less than that threshold, and the average
level of the signal before that point, is higher than the same threshold, we say that the signal has converged.
If not, the algorithm continues searching for convergence until the end of 2 seconds buffer.
Threshold
Convergence
time
Threshold
23
-40
-30
-20
-10
0
10
20
30
1
1
1
2
1
3
1
4
1
5
1
6
1
7
1
8
1
9
1
1
0
1
1
1
1
1
2
1
1
3
1
1
4
1
1
5
1
1
6
1
1
7
1
1
8
1
1
9
1

Figure 3-13 Additional Condition before a Final Decision is Calculated.
The levels before and after place of discontinuity are calculated.
Noise Reduction/Suppression Test
This measurement gives us general information about the type of noise treatment applied in communication
channel. Based on measurement of the differences between signal power level reductions in speech power
classes, the algorithm decides whether noise suppression or noise reduction was applied. Experimental
results have shown that it is necessary to define one more situation. In a case that signal power level
reduction in pause class is higher than SPLR in high class + offset of 3 dB, we say that the noise in
communication channel wasnt treated in either way. Reference suitability is a fourth possible result coming
from this measurement.

Figure 3-14 Typical Reference Signal with White Noise Added
The noise suppression and the noise reduction are both used to enhance speech quality in a range of
environments where there is significant (audible) background noise (see Fig. 24). The noise suppression
reduces the noise in pause and noise power classes, and has very little, or not at all, influence on higher
power classes (speech active intervals). To draw a distinction, the noise reduction reduces the noise equally
in all power classes. Therefore we have based our algorithm on measurement of the difference between
signal power level reduction in high and medium speech power class. If the difference between two levels is
less than a calculated threshold we say that the noise suppression was applied.
Threshold
Mean signal level
before threshold
Mean signal level
after threshold
24

Figure 3-15 Noise Suppression Applied on Signal

Figure 3-16 Noise Reduction Applied on Signal
An important role has the level of noise in the reference signal. If the signal to noise ratio of the reference
signal is higher than 30 dB, we say that reference signal is not good for conducting the measurement, due to
too low noise level. The same reference SNR is used for calculating the threshold offset between SPLRs of
high and medium power classes.
Examples
The Signal Envelope shows that the noise is really reduced and the speech part is more or less the same
as for the reference signal.

Figure 3-17 NS Signal Envelope
Evaluation of the transmitted signal
In addition to ratings of the noise reduction systems described above, the results of the listening quality
evaluation of the transmitted and de-noised signal is given as well in separate section.
Here the SQuad-LQ is applied on the transmitted signal.
25

Figure 3-18 Results presentation within NQDI
Besides of the pure LQ also the speech and noise levels are shown. Furthermore, the clipping value can be
used for evaluation non-linear processings by the NS device.
Evaluation of the transmitted signal
ETSI TS 101 512, V8.0.0, (GSM 06.77 version 8.0.0 Release 1999), Digital cellular telecommunication
system (Phase 2+), Minimum Performance Requirements for Noise Suppresser, Application to the AMR
Speech Encoder
ETSI TS 101 745, V8.0.0, (GSM 02.76 version 8.0.0 Release 1999), Noise Suppression for the AMR Codec,
Service Description, Stage 1
ETSI TS 101 831, V8.0.0, (GSM 06.78 version 8.0.0 Release 1999), Digital cellular telecommunication
system (Phase 2+), Results of the AMR Noise Suppression Selection Phase, Application to the AMR Speech
Encoder

Chapter 4 | DTMF Tests
26
4 DTMF Tests
Introduction
In telecommunications today, the most used signalling system is DTMF signalling. DTMF stands for Dual
Tone Multi-Frequency. As the name suggests, the DTMF signal consists of two superimposed sinusoidal
waveforms with frequencies chosen from a set of eight standardized frequencies.
When a DTMF signal is sent over a network it can be degraded, especially when it is encoded. For an
operator of a network, it is of interest to know if the receiver of the DTMF signals can convert the DTMF
signal back into a digit or a symbol. The objective is to measure the percentage of detected and undetected
DTMF digits.
In the first part of SwissQual's algorithm for DTMF test, the algorithm scans through a given signal and
detects the locations of DTMF signals. Once a DTMF signal is found, the algorithm calculates the
characteristics and decides if the signal is valid. If the tone is invalid, the DTMF-Test describes which
condition that was not accomplished. The algorithm collects all characteristics and saves them in a file.
The DTMF signal used for tests, which consist of two frequencies. According to the CCITT Recommendation
Q.23 [5] and Q.24, there are two frequency groups, each with four frequencies:

The figure below shows how the frequencies are allocated to the various digits and symbols of a push-button
set. Every digit and symbol consists of a frequency from the low and the high group.

Figure 4-1 Allocation of Frequencies to the Various Digits and Symbols of a Push-button Set
DTMF-Test Overview
One or more DTMF signals are sent over a network. The coded signal will be used for the DTMF-Test. This
signal is available in digital form; the data format is PCM (without compression). The sampling frequency is 8
kHz or 16 kHz. The digital quantization of the signal can be 8 bit (unsigned or signed) or 16 bit (big or little
endian). Inside, the algorithm works with 16 bit resolution. The figure below illustrates the basic algorithm of
DTMF-Test. DTMF-Test saves its result in a comma delimitated text file.
27

Figure 4-2 Block Diagram of DTMF-Test
Criterions
The objective of the SwissQual model for DTMF testing is to measure the percentage of undetected DTMF
digits processed through the network. The DTMF signals are generated at the frequencies specified in the
ITU-T Rec. Q.23.
The algorithm follows the ETSI guidelines defined in "TS 101 235-1" "Technical Specification of Dual Tone
Multi-Frequency (DTMF)".
The received DTMF signal shall be detected as valid when:
Only two of the signalling frequencies are present, one from the high group and one from the low group,
fulfilling the conditions as described above
Each of these signalling frequencies are within +/-(1,5 % +2 Hz) of the nominal value
The level of each of these two signalling frequencies is within the range -27 dBV to -5 dBV
The difference in level of these two signalling frequencies is not more than 6 dB.
Results
Table 4-1 DTMF Result Code
Code Description
Tone Length Length of a DTMF Tone
Pause Length Pause Length between two DTMF tones
Measured Level Average Level of a DTMF Tone
Level Deviation Level Deviation of the two frequencies of a DTMF Tone
28
Code Description
Freq. Low Low Frequency value in Hertz
Freq. High High Frequency value in Hertz
DevFreqLow [Hz] Deviation of the low frequency from the standard in Hertz
DevFreqHigh [Hz] Deviation of the high frequency from the standard in Hertz
DevFreqLow [%] Deviation of the low frequency from the standard in percent
DevFreqHigh [%] Deviation of the high frequency from the standard in percent
Twist [dB] Level difference between the high and the low frequency
Signal Valid
Signal valid code:
Cause: Valid (0)
If the received tone matches all conditions, then the signal is valid and the field
'SignalValid' is set to '0'
Cause: TooShort (-1)
If the received tone is too short (<40ms) then the signal is invalid and the field
'SignalValid' is set to the code '-1'
Cause: NoDigit (-2)
If the received tone does not contain the two frequencies as specified, the signal
is invalid and the field 'SignalValid' is set to '-2'
Cause: LowLevel (-3)
If the Noise Level is more than 10% of the Signal Level of a tone, then the signal
is invalid and the field 'SignalValid' is set to '-3'
Cause: FreqDeviation (-4)
If one of the two frequencies is out of the specified range (+/- 1.5%), then the
signal is invalid and the field 'SignalValid' is set to '-4'
Cause: LevelDiff (-5)
If the level difference of the two frequencies for one tone is more than 10dB, then
the signal is invalid and the field 'SignalValid' is set to '-5'
Cause: Unknown (-6)
If there is no tone at all, then the signal is invalid as well and the field
'SignalValid' is set to '-6'
Cause: TooLong (-7)
If the received tone is too long (>90ms), then the signal is invalid and the field
'SignalValid' is set to the code '-7
29
Code Description
Signal Match Signal match code:
Cause: NotRegular (A)
If a tone in the coded file is too short, or wrong in any other aspect, it is not
matched with the reference tone the field 'SignalMatch' is set to the code 'A'
Cause: AdditionalNotRegular (B)
If there are one or more irregular tones with no reference, the field 'SignalMatch'
is set to the code 'B'
Cause: MissingTone (C)
If a reference tone has no match but there are two or more irregular tones, the
field 'SignalMatch' is set to the code 'C'
Cause: MultipleMissingTones (D)
If there are two or more reference tones with no match but three or more
irregular tones, the field 'SignalMatch' is set to the code 'D'
Cause: MultipleMissingTones (E)
If there are more reference tones with no match than irregular tones, the field
'SignalMatch' is set to the code 'E'
Cause: MultipleTone (F)
If a reference tone has two or more matching tones in the coded file, the field
'SignalMatch' is set to the code 'F'
Cause: AdditionalTone (G)
If there is no reference for a tone in the coded file, the field 'SignalMatch' is set to
the code 'G'
Cause: MissingTone (H)
If for a reference tone there is no tone in the coded file, the field 'SignalMatch' is
set to the code 'H' Cause: Disparity (I)
If the number and order for a string of tones in reference and coded file cannot
be matched, the field 'SignalMatch' is set to the code 'I'

Chapter 5 | SQuad Advanced Echo Check (Passive Test)
30
5 SQuad Advanced Echo Check (Passive Test)
Introduction
The measurement application of the Acoustic Echo Check (also called: Acoustic Echo Check) can be applied
to a SwissQual measurement probe at the far end side as well as to any number on that an automatic hook-
up device is connected. SQuad-AEC does not require any artificial test signals but it is optimized to detect
echoes by using human speech as measuring signal. So it works for all technologies that serve voice
communications and make the algorithm ready for in-service live monitoring.
The SQuad-AEC measurement will detect echoes in that active connection by sending a speech signal to the
far end side and observing the receiving direction for any reflections. If no signal is inserted at the far end
side the procedure is measuring during that single talk situation only. Because commonly used Non-Linear-
Processors like VAD's suppress low power send signals also echoes will not occur in such connections.
Therefore the SQuad-AEC algorithm is also designed to detect echoes during double talk situations. Such a
double talk situation may simulate by an active playing answering station or by using a real phone at the far-
end side and talking in during the measurement. This double talk at the far end side will switch through the
sending path and also the echo can be transmitted.
The SQuad-AEC Test is especially designed to detect electrical as well as acoustical echoes and is able to
detect 'dry' and 'hallow' acoustical echoes as well as hybrid echoes and is more robust against double talk. In
case of a 4-wire connected the far-end station the echoes introduced by the network will be found. By using
real (echo-producing) terminals the insertion of echoes caused by network AND the terminal can be
calculated.
Echo Measurement
This Advanced Echo Check Passive Test (AEC passive) does not simulate anything on B-Side. The A-
Side starts a call and after B-Side has answered the call; the collecting of the down-link (B->A) audio stream
is started. When the recording of the stream has finished, the search of echo signal is started by comparing
the registered signal with the reference signal.
On the B-Side, we can use any (self-answering) voice terminal or a SwissQual Diversity measurement
probe. The AEC algorithm is able to detect echo in presence of background noise and double talk.
The algorithm is running in two steps:
- Observing a wide range of echo delay for possible echoes (scan procedure)
- Analysing accepted echo regions in detail for calculating the echo loss and the other results
Measurement Results
The AEC algorithm generates the following results:
Signal type
Echo Delay in milliseconds
Echo Loss during Single Talk acc. ITU-T G.122
Echo Loss for the complete signal (incl. Double Talk)
Echo Objection Rate acc. ITU-T G.131 in %
Distance to 1% Echo Objection Rate acc. ITU-T G.131
ECHO status
GSM3.50 test
Double Talk Ratio
31
Level of Received Signal in dBov
Signal type can be Echo, SideTone, Double Talk, Silence or combinations of them. As Sidetone
received signal parts will be rated, which are correlated to the send signal and were received with less than
20ms delay. Double Talk will be signalized if in more than 40% of the signal duration the receiving signal is
exceeding the defined double talk threshold.
A Signal type NoEcho informs that the connection is echo-free or the echo will be not perceptible.
Additionally, a signal type NoEchoFound signalizes that the echo detection was hardly disturbed be
unexpected distortions in the signal. It might be echo-free but also can include a masked echo.
An additional parameter shows the Double Talk Ratio in %. The Double Talk Ratio shows in which ratio the
found echo is superset by Double Talk. Only signal parts are defined as doubletalk, where the Near-End
sending signal exceeds -36dBov (approx. -30dBm) and the Far-End signal is -48dBov (approx. -42dBm) in
minimum. In case of strong echoes this double talk threshold will be increased to minimize the classification
of strong echoes as double talk.
1

Echo Delay in milliseconds is a time offset between the reference- and the returned echo- signal. A range
of values is between 0 and 1000 ms. If no echo was detected the Echo Delay is 0.0 ms.
Echo Loss is the weighted echo signal level measured relative to the reference signal level. The calculation
is in accordance to ITU-T G.122. If no echo was detected the Echo Loss is 99dB, which is no echo in fact.
Under the assumption of an SLR + RLR = 10dB a TELR = Echo Loss + 10dB can be estimated by using the
Echo Loss result. SLR stands for Send Loudness Rating and RLR for Receive Loudness Rating.

Figure 5-1 Results of SQuad-AEC Test, shown in NQDI presentation
Echo status shows the grade of annoyance of the echo signal. There are three possible values: GOOD
stands for good echo performance, FAIR for an acceptable echo performance and POOR for annoying
echoes. Using the Echo Objection curves given in ITU-T G.131 the Echo Status is derived. Therefore the
TELR is estimated by adding 10dB to the Echo Loss and the corresponding cross-point between the TELR
and half of the Echo Delay (= one way transmission time of the echo).

1
Please remark that in case of pure single talk situations a powerful echo region might classified as Double
Talk and a Double Talk Ratio of some percentage will shown if the adaptive Double Talk Threshold is
exceeded. This might be observed especially for time varying echo paths. Furthermore, a huge mount of
noise in the receiving part may be classified as double talk too.
32

Figure 5-2 EOR (Echo Objection Rate) derived from G.131
The GSM 03.50 test is defined in ETSI GSM 03.50 (Section 3.4) and derives the required Terminal Coupling
Loss (TCL) from the G.131 TELR (talker echo loudness rating) chart. Under the assumption of a no-loss 4-
wire connection from the measuring point to the terminal, the ERL can be interpreted directly as the TCL,
because the terminal itself is the only existing source of echoes. Thus, SQuad-AEC measures the TCL value
directly in this case. Thus, we can set:
TCL = TELR - (SLR + RLR) dB, where typically SLR + RLR = 10 dB
TCL = TELR - 10dB
TCL should be ideally 40 dB to 46 dB. 46dB is derived from 1% EOR curve of G.131, with maximum delay
(about 400 ms). If a TCL of higher than 46 dB is reached, the 1% EOR curve will never pass even for high
delays (if no other echo sources besides the terminal exist).
If the measured TELR = TCL + 10 dB is higher than the 1% EOR, the GSM3.50 test shows a passed
value. In the case of a lower TCL, this value is considered to be in the failed range.
EOR (Echo Objection Rate) in % is an estimate of the percentage of the listeners who has perceived a
talker echo when listening to a given telephone setup. ITU-T G.131 shows two different curves one for 1%
EOR and one for 10% EOR. It is assumed that a set of equally shaped curves will describe each EOR
between 0100%. Based on the described crossing point of estimated TELR and the half of the Echo Delay
a (theoretical) corresponding curve can be derived and the assumed EOR can be taken.
If this EOR is less than 1%, the Echo Status is also GOOD, if this EOR is above 10% it is POOR. Between
both values the echo will be rated as FAIR.
The Distance to 1% EOR is also calculated directly from the chart given in ITU-T G.131. This value gives
the distance to the 1% EOR curve for the calculated echo delay. All negative values are in the 'green region,'
values above 10dB are in the 'red region.'
Echo Loss profile: The Echo Loss is shown graphically versus the delay time. This figure is for detailed
information and should visualize the echo region found. This Echo Loss profile is the one result of the scan
process and represents only situation during single talk. This profile is used for pin-pointing echoes only. The
detailed echo analysis itself will be done in a separated step and therefore the echo loss can not derived
directly from this curve.
33

Figure 5-3 Echo Loss during scanning versus echo delay
The Level of Received Signal in dBov gives only information about the r.m.s. level of the received signal at
all. It covers echoes, double talk sequences and noises.

Chapter 6 | SQuad Advanced Echo Check (Active Test)
34
6 SQuad Advanced Echo Check (Active Test)
Introduction
The active named measurement application of the Acoustic Echo Check can be applied only to an
SwissQual measurement probe at the far end caused by active actions has to be done. It is generating an
echo at he far end side.
Echo Measurement
The SQUAD AEC active measurement is using the same echo detection approach as the passive
measurement described above. Compared to the passive measurement, where the far-end side is silent in
the active mode the far-end side will create an echo actively. The SQUAD AEC active measurement includes
an inband synchronization between both sides. In a first communication the incoming signal will be recorded
at the far-end side. Based on that signal an echo is generated by applying selectable echo path responses
on that. If required, the generated echo can be interlaced with double talk. During receiving the signal
second time, the pre-processed echo will be played back to the sending side for evaluation.
This measurement is especially designed to detect and rate echo cancellers or suppressors in the network.
The generated echo will challenge these echo cancellers and possible integrated level-switching devices will
be forced by an inserted double talk signal.
The echo-detection is more confident if the remaining echo has linear components. Especially during double
talk, only linear dependent echoes can be detected. In connections including low bit-rate codecs and/or non-
linear processors the residual or low echoes might be non-detectable by the measurement. For more
confidence chose higher echo levels to increase their differentiation from doubletalk and other non-linear
components.

Measurement Results Echo Evaluation
The SQuad AEC active measurement generates the same results as described in Chapter 5.
Signal type
Echo Delay in milliseconds
Echo Loss during Single Talk acc. ITU-T G.122
Echo Loss for the complete signal (incl. Double Talk)
Echo Objection Rate acc. ITU-T G.131 in %
Distance to 1% Echo Objection Rate acc. ITU-T G.131
ECHO status
35
GSM3.50 test
Double Talk Ratio
Level of Received Signal in dBov
Furthermore, SwissQuals database interface NQDI displays the settings used at the far end together with
the measurement results:

Figure 6-1 Results presentation SQuad-AEC active
The results in the Figure presented should be used here for discussion of the results as well. The
measurement was done in a Mobile to PSTN connection. The PSTN-side was the echo generating loop. At
this side the incoming signal was convoluted by the echo path response M1 from ITU-T G.168 (G168_M1).
Afterwards it was attenuated by 20dB and interlaced by a double talk signal containing 50% active speech
(dt_50_08kHz.wav). An additional delay was not chosen at PSTN side.
The results show that this echo was detected at the mobile side. The echo path delay of 224ms is typical for
a Mobile to PSTN connection. The echo loss over the complete signal is 21dB, which reflects pretty well the
range of the defined echo at the PSTN side. The echo loss during single talk is a bit lower, which signalizes
that there is an active component reducing the echo in speech pauses at least a bit.
2

Using this results the corresponding Echo Objection Rate is calculated (here: 54%) and the distance to the
G.131 1% curve (12dB) as well. That means a increasing of the echo loss by 12dB would be necessarily to
reach the 1% curve and therefore the echo status good.
Consequently, by the reached results the echo status is rated as poor.
Additionally, the Double Talk Ratio is 47%, which is caused by the defined signal at the far-end side.

Figure 6-2 Echo Loss as profile versus echo delay
Also for SQuad-AEC in the active mode the echo loss profile is displayed.
If no echo is found or if it was not detectable, only the status messages and the level of the received signal
will be displayed:

2
Please note that the channel gain will also influence the measured echo loss. Basically, the channel
attenuation in both directions has to be added to the defined echo loss at far-end side. The measuring signal
is attenuated due to the transmission from A to B, is there attenuated again (during the defined loss value)
and will be attenuated again due to the transmission from B to A again. The echo loss reflects the level of the
received echo compared to the original measuring signal.
36
Figure 6-3 Result presentation in an echo free/non echo detectable connection
Measurement Results Listening Quality
Within the SQuad AEC active measurement an additionally evaluation of the Listening Quality is
integrated. Here the SQuad-LQ is applied on the signal received by the echo-generating far-end side.

So a simple Listening Quality measurement can be done in parallel. The interesting point is here: How the
Listening Quality is affected by double talk/echo in the other direction. By comparison of both Listening
Quality values the double talk capability can be evaluated. If a network is fully duplex both Listening Quality
values should be the same even a double talk signal is chosen.

The right value gives the Listening Quality for the first transmission where no echo or double talk is played
back. The right value gives the LQ during the echo / double talk is sent at the same time. In addition the
channel gain and the clipping of the received signal is also given.
Please note that a strong side-tone at that B-side may affect the SQuad-LQ measurement, because it
interleaves with the received and evaluated signal.

Chapter 7 | Round Trip
37
7 Round Trip
Introduction
The Round Trip Time is the time a signal needs to travel from the near end side to the far end side and
back. The Round Trip Time is mostly close to the delay of the latest possible echo. The time speech needed
to travel from one talker to the other (One Way Signal Delay) is an important indicator of the conversational
quality of a call. A travel time that is too high leads to the annoying effect that the talkers interrupt each other
unintentionally.
The Round Trip Method
The RTT inband measurement measures the Round Trip Time of a connection by using short voice-like
sequences. This guarantees the transmission over the complete link and avoids suppressions how it may
happen in case of artificial signals like sweeps or impulses. The RTT inband measurement is a point-to-
point measurement, i.e. an A-Side user calls a B-Side user. After a successful call establishment, the A-side
sends the RTT synchronisation signal (RTTvoiceA) three times one after another but separated by a
silence gap of 5.4s to the B-Side.
After receiving this sequence the B-Side sends back the RTTvoiceB sequence. In comparison to a pure
reflecting at B-Side, the usage of different sequences at A- and B-Side avoids the suppression of the
reflected signal by an echo-compensation system in the network. In minimum two of three samples has to be
detected at the
A-Side again.
Results
The measurable Round Trip Time is limited from 4ms in minimum to 3000ms in maximum; the maximal
delay jitter between the three repetitions within one measurement has to be below 500ms. The results of
the measurement are presented in the following table. In addition, the lowest of the one way and round trip
time of the measurements in milliseconds is shown as final results.

Figure 7-1 Detail of the NQDI Representation
The quality classes according to ETSI:
Table 7-1 One Way Delay Quality Classes
4 (BEST) 3 (HIGH) 2 (MEDIUM) 1 (BEST
EFFORT)
One Way Delay < 100 ms < 100 ms < 150 ms < 400 ms
References
ETSI TS 101 329-2 V1.1.1 (2000-07), Part 2: Definition of Quality of Service (QoS) Classes

Appendix A | Appendix
38
A Appendix
Abbreviations
Abbreviation Description
ACR Absolute Category Rating
CELP Code Excited Linear Prediction
DCR Degradation Category Rating
DMOS Degradation Mean Opinion Score
MOS Mean Opinion Score
dBov dB relative to the overload point of a digital system
ADPCM Adaptive Differential Pulse Code Modulation
BFI Bad Frame Indication
CCITT Comit Consultatif International Tlgraphique et Tlphonique (The International
Telegraph and Telephone Consultative Committee)
CDMA Code-Division Multiple Access
CRC Cyclic Redundancy Check (3 bit)
DAC Digital to Analogue Converter
DMR Digital Mobile Radio
DTMF Dual Tone Multi-Frequency (signalling)
DTX Discontinuous Transmission (mechanism)
EPROM Erasable Programmable Read Only Memory
ETR ETSI Technical Report
ETS European Telecommunication Standard
ETSI European Telecommunications Standards Institute
FER Frame Erasure Ratio
FR Full Rate
GMSK Gaussian Minimum Shift Keying (modulation)
GSM Global System for Mobile communications
GSM MS GSM Mobile Station
HANDO Handover
HDLC High level Data Link Control
HR Half Rate
IEC International Electro-technical Commission
ISDN Integrated Services Digital Network
ISO International Organization for Standardization
ITU International Telecommunication Union
LAN Local Area Network
Appendix A | Appendix
39
Abbreviation Description
MSC Mobile-services Switching Center, Mobile Switching Center
OSI Open System Interconnection
PABX Private Automatic Branch eXchange
PDN Public Data Networks
PSPDN Packet Switched Public Data Network
PSTN Public Switched Telephone Network
QOS Quality Of Service
RXLEV Received signal level
RXQUAL Received Signal Quality
S/W Software
SIM Subscriber Identity Module
SS7 Signaling System No. 7
TDMA Time Division Multiple Access
TE Terminal Equipment
VAD Voice Activity Detection

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