SIST ETS 300 807 E1:2003
(Main)Integrated Services Digital Network (ISDN); Audio characteristics of terminals designed to support conference services in the ISDN
Integrated Services Digital Network (ISDN); Audio characteristics of terminals designed to support conference services in the ISDN
To standardize the video and audio characteristics for terminals designed to support conference services in the ISDN. The standard will include practical installation guides for systems to be installed in permanent conference rooms.
Digitalno omrežje z integriranimi storitvami (ISDN) – Zvokovne lastnosti priključkov, zasnovanih za podporo konferenčnim storitvam v omrežjih ISDN
General Information
Standards Content (Sample)
SLOVENSKI STANDARD
01-december-2003
'LJLWDOQRRPUHåMH]LQWHJULUDQLPLVWRULWYDPL ,6'1 ±=YRNRYQHODVWQRVWL
SULNOMXþNRY]DVQRYDQLK]DSRGSRURNRQIHUHQþQLPVWRULWYDPYRPUHåMLK,6'1
Integrated Services Digital Network (ISDN); Audio characteristics of terminals designed
to support conference services in the ISDN
Ta slovenski standard je istoveten z: ETS 300 807 Edition 1
ICS:
33.080 Digitalno omrežje z Integrated Services Digital
integriranimi storitvami Network (ISDN)
(ISDN)
35.180 Terminalska in druga IT Terminal and other
periferna oprema IT peripheral equipment
2003-01.Slovenski inštitut za standardizacijo. Razmnoževanje celote ali delov tega standarda ni dovoljeno.
EUROPEAN ETS 300 807
TELECOMMUNICATION November 1997
STANDARD
Source: MTA Reference: DE/MTA-004043
Formerly: DE/TE-04043
ICS: 33.020
Key words: Audio, ISDN, conf, teleservice, terminal
Integrated Services Digital Network (ISDN);
Audio characteristics of terminals designed to support
conference services in the ISDN
ETSI
European Telecommunications Standards Institute
ETSI Secretariat
Postal address: F-06921 Sophia Antipolis CEDEX - FRANCE
Office address: 650 Route des Lucioles - Sophia Antipolis - Valbonne - FRANCE
X.400: c=fr, a=atlas, p=etsi, s=secretariat - Internet: secretariat@etsi.fr
Tel.: +33 4 92 94 42 00 - Fax: +33 4 93 65 47 16
Copyright Notification: No part may be reproduced except as authorized by written permission. The copyright and the
foregoing restriction extend to reproduction in all media.
© European Telecommunications Standards Institute 1997. All rights reserved.
Page 2
ETS 300 807: November 1997
Whilst every care has been taken in the preparation and publication of this document, errors in content,
typographical or otherwise, may occur. If you have comments concerning its accuracy, please write to
"ETSI Editing and Committee Support Dept." at the address shown on the title page.
Page 3
ETS 300 807: November 1997
Contents
Foreword .7
1 Scope .9
2 Normative References .9
3 Definitions and abbreviations .11
3.1 Definitions .11
3.2 Abbreviations .11
4 System description.12
4.1 Audio facilities.13
4.2 Audio encoding .13
4.2.1 CCITT Recommendation G.711 encoding.13
4.2.1.1 A-law.13
4.2.1.2 μ-law .13
4.2.2 CCITT Recommendation G.722 encoding.14
4.2.3 CCITT Recommendation G.728 encoding.14
4.3 Audio decoding .14
4.3.1 CCITT Recommendation G.711 decoding.14
4.3.1.1 A-law.14
4.3.1.2 μ-law .14
4.3.2 CCITT Recommendation G.722 decoding.14
4.3.3 CCITT Recommendation G.728 decoding.14
4.4 Relative level.15
5 Speech transmission characteristics.15
5.1 Handset mode .15
5.2 Headset mode .15
5.3 Hands-free mode (single user) .15
5.4 Hands-free mode (multiple users) .15
5.4.1 Receiving volume control and sensitivity adjustments .16
5.4.1.1 Sending sensitivity adjustment .16
5.4.1.2 Receiving sensitivity adjustment.16
5.4.1.3 Receiving volume control.16
5.4.1.4 Adaptive gain control (optional) .16
5.4.2 Sensitivity-frequency response.16
5.4.2.1 Sending.16
5.4.2.2 Receiving.17
5.4.3 Loudness rating.18
5.4.3.1 Sending.18
5.4.3.2 Receiving.18
5.4.4 Terminal coupling loss.19
5.4.4.1 TCL .19
w
5.4.4.2 Stability loss.19
5.4.5 Distortion .19
5.4.5.1 Sending.19
5.4.5.2 Receiving.20
5.4.6 Out-of-band signals.20
5.4.6.1 Discrimination against out-of-band input signals (sending).20
5.4.6.2 Spurious out-of-band (receiving) .20
5.4.7 Noise .21
5.4.7.1 Sending.21
5.4.7.2 Receiving.21
5.4.8 Delay .21
Annex A (normative): Test methods.22
Page 4
ETS 300 807: November 1997
A.1 General conditions for testing. 22
A.1.1 Testing environment. 22
A.1.1.1 Test room . 22
A.1.1.2 Testing arrangements. 22
A.1.1.2.1 Sending. 23
A.1.1.2.2 Receiving . 23
A.1.1.2.3 Terminal coupling loss . 24
A.1.2 Test equipment interface.24
A.1.3 Test equipment requirements . 24
A.1.3.1 Electroacoustic equipment. 24
A.1.3.2 Test signals and spectrum measurements. 24
A.1.3.2.1 Standard test signals. 24
A.1.3.2.2 Composite source signal. 25
A.1.3.3 Test signals levels . 25
A.1.3.3.1 Sending. 25
A.1.3.3.2 Receiving . 25
A.1.3.4 Test equipment for the digital interface . 25
A.1.3.4.1 Codec specifications . 25
A.1.3.4.2 Analogue interface . 26
A.1.3.4.3 Definition of 0 dBr point. 26
A.1.4 Accuracy of calibrations . 26
A.2 Testing of transmission requirements . 26
A.2.1 Sensitivity-frequency response . 26
A.2.1.1 Sending. 26
A.2.1.2 Receiving . 27
A.2.2 Loudness rating. 27
A.2.2.1 Sending Loudness Rating. 27
A.2.2.2 Receiving Loudness Rating . 28
A.2.3 Terminal Coupling Loss . 28
A.2.3.1 Weighted Terminal Coupling Loss. 28
A.2.3.2 Stability loss. 28
A.2.4 Distortion . 28
A.2.4.1 Sending. 28
A.2.4.2 Receiving . 29
A.2.5 Out-of-band signals. 29
A.2.5.1 Discrimination against out-of-band input signals (sending) . 29
A.2.5.2 Spurious out-of-band (receiving) . 29
A.2.6 Noise . 29
A.2.6.1 Sending. 29
A.2.6.2 Receiving . 29
A.2.7 CCITT Recommendation G.725 encoding. 29
Annex B (informative): Test Methods for Delay measurement. 30
B.1 Introduction. 30
B.2 Cross-correlation method . 30
B.2.1 Sending . 30
B.2.2 Receiving. 31
B.2.3 Total delay. 31
B.3 Method based on group delay . 31
B.3.1 Sending . 31
B.3.2 Receiving. 32
B.3.3 Total delay. 32
Annex C (informative): Audio alignment and practical installation guides . 33
C.1 Introduction. 33
C.2 Audio alignment and sensitivity-frequency characteristics . 33
C.2.1 Sending . 33
Page 5
ETS 300 807: November 1997
C.2.1.1 Sensitivity adjustment(s) .33
C.2.1.2 In-site frequency response.34
C.2.2 Receiving .34
C.2.2.1 Sensitivity adjustment.34
C.2.2.2 In-site frequency response.34
C.3 Practical installation criteria.34
C.3.1 Maximum talker to microphone distance.34
C.3.1.1 Background noise level constraints.34
C.3.1.2 Reverberation constraints .35
C.3.1.3 Preferred maximum talker to microphone distance .36
C.3.2 Sound insulation .36
Annex D (informative): Bibliography.37
History.38
Page 6
ETS 300 807: November 1997
Blank page
Page 7
ETS 300 807: November 1997
Foreword
This European Telecommunication Standard (ETS) has been produced by the Terminal Equipment (TE)
Technical Committee and later the Multimedia Terminals and Applications (MTA) Project of the European
Telecommunications Standards Institute (ETSI).
Transposition dates
Date of adoption: 24 October 1997
Date of latest announcement of this ETS (doa): 28 February 1998
Date of latest publication of new National Standard
or endorsement of this ETS (dop/e): 31 August 1998
Date of withdrawal of any conflicting National Standard (dow): 31 August 1998
Page 8
ETS 300 807: November 1997
Blank page
Page 9
ETS 300 807: November 1997
1 Scope
This European Telecommunication Standard (ETS) specifies the audio characteristics of terminals
designed to support the audiographic conference teleservice as specified in ETS 300 675 [1]. The same
audio requirements of this ETS are also applicable to terminals supporting the Integrated Services Digital
Network (ISDN) videoconference teleservice.
This ETS does not specify the terminal procedures, both with respect to in-band signalling and to ISDN
signalling on the D channel. Also the procedures and protocols for data exchange and conference control
are outside the scope of this ETS.
2 Normative References
This ETS incorporates by dated and undated reference, provisions from other publications. These
normative references are cited at the appropriate places in the text and the publications are listed
hereafter. For dated references, subsequent amendments to or revisions of any of these publications
apply to this ETS only when incorporated in it by amendment or revision. For undated references the latest
edition of the publication referred to applies.
[1] ETS 300 675: "Integrated Services Digital Network (ISDN); Audiographic
conference teleservice; Service description".
[2] I-ETS 300 245-2 (1996): "Integrated Services Digital Network (ISDN); Technical
Characteristics of Telephony Terminals, Part 2: PCM A-law, handset telephony".
[3] I-ETS 300 245-3 (1995): "Integrated Services Digital Network (ISDN); Technical
Characteristics of Telephony Terminals, Part 3: Pulse Code Modulation (PCM)
A-law, Loudspeaking and Handsfree telephony".
[4] I-ETS 300 245-5 (1996): "Integrated Services Digital Network (ISDN); Technical
Characteristics of Telephony Terminals, Part 5: Wideband (7 kHz) Handset
Telephony".
[5] I-ETS 300 245-6 (1996): "Integrated Services Digital Network (ISDN); Technical
Characteristics of Telephony Terminals, Part 6: Wideband (7 kHz)
Loudspeaking and Handsfree telephony".
[6] I-ETS 300 245-8 (1996): "Integrated Services Digital Network (ISDN); Technical
Characteristics of Telephony Terminals, Part 8: Speech transmission
characteristics when using Low Delay Code-Excited Linear Prediction (LD-
CELP) coding at 16 kbit/s".
[7] I-ETS 300 302-1 (1996): "Integrated Services Digital Network (ISDN);
Videotelephony teleservice; Part 1: Electroacoustic characteristics for 3,1 kHz
bandwidth handset terminals".
[8] I-ETS 300 302-2 (1995): "Integrated Services Digital Network (ISDN)
Videotelephony teleservice; Part 1: Electroacoustic characteristics for 3,1 kHz
bandwidth loudspeaking and handsfree terminals".
[9] I-ETS 300 302-3 (1996): "Integrated Services Digital Network (ISDN);
Videotelephony teleservice; Part 3 Audio aspects-Wideband and Handset".
[10] I-ETS 300 144 (1996): "Integrated Services Digital Network (ISDN); Audiovisual
services, Frame structure for a 64 to 1 920 kbit/s channel and associated syntax
for in-band signalling".
[11] I-ETS 300 143 (1994): "Integrated Services Digital Network (ISDN); Audiovisual
services, In-band signalling procedures for audiovisual terminals using digital
channels up to 2 048 kbit/s".
Page 10
ETS 300 807: November 1997
[12] CCITT Recommendation G.711 (1988): "Pulse code modulation (PCM) of voice
frequencies".
[13] CCITT Recommendation G.722 (1988): "7 kHz audio coding within 64 kbit/s".
[14] CCITT Recommendation G.728 (1992): "Coding of speech at 16 kbit/s using
low-delay code-excited linear prediction".
[15] CCITT Recommendation G.725 (1988): "System aspects for the use of the
7 kHz audio codec within 64 kbit/s".
[16] ITU-T Recommendation P.57 (1996): "Artificial Ears".
[17] ITU-T Recommendation P.10 (1993): "Vocabulary of terms on telephone
transmission quality and telephone sets".
[18] ITU-T Recommendation G.701 (1993): "Vocabulary of digital transmission and
multiplexing and pulse code modulation (PCM) terms".
[19] ITU-T Recommendation P.51 (1996): "Artificial Mouths".
[20] ITU-T Recommendation P.79 (1993): "Calculation of loudness ratings for
telephone sets".
[21] ITU-T Recommendation P.64 (1993): "Determination of sensitivity/frequency
characteristics of local telephone systems".
[22] CCITT Recommendation P.76 (1988): "Determination of loudness rating;
fundamental principles".
[23] TBR 3 (1995): "Integrated Services Digital Network (ISDN); Attachment
requirements for terminal equipment to connect to an ISDN using ISDN basic
access".
[24] ISO 3 (1973): "Preferred numbers-Series of preferred numbers".
[25] ITU-T Recommendation P.340 (1996): "Transmission characteristics of hands
free telephones".
[26] ITU-T Recommendation G.122 (1993): "Influence of national systems on
stability talker echo in international connections".
[27] IEC Publication 651 (1979): "Sound level meters".
[28] ITU-T Recommendation P.310 (1996): "Transmission characteristics for
telephone band (300-3400 Hz) digital telephones".
[29] ITU-T Recommendation P.50 (1993): "Artificial Voices".
[30] ITU-T Recommendation G.101 (1996): "The transmission plan".
Page 11
ETS 300 807: November 1997
3 Definitions and abbreviations
3.1 Definitions
For the purposes of this ETS, the definitions provided in the referenced standards and the following
definitions apply:
Acoustic Reference Level (ARL): Acoustic level which gives -10 dBm0 at the digital interface.
audiographic terminal: Terminal supporting the audiographic teleconference service.
Hands Free Reference Point (HFRP): A point located on the axis of the Artificial Mouth, at 50 cm from
the lip ring, where the level calibration is made in free field. It corresponds to the measurement point n.11,
as defined in ITU-T Recommendation P.51 [19].
reference sphere: Sphere of radius 1 metre where the anechoic conditions of the acoustic testing
environment are verified.
lip synchronization delay: The delay introduced in the sending and receiving audio paths in order to
align the audio signals with the moving pictures respectively transmitted and received by the terminal.
digital interface: For the purposes of this ETS, the digital interface refers to the B channels available at
the coincident S and T reference points at an ISDN basic access.
3.2 Abbreviations
For the purposes of this ETS, the abbreviations used in ITU-T Recommendations G.701 [18], P.10 [17],
P.51 [19], P.57 [16], P.64 [21], P.76 [22] and P.79 [20] and the following abbreviations apply:
ARL Acoustic Reference Level
ERP Ear Reference Point
F Correction factor for receiving measurements (annex A, subclause A.1.1.2.2)
r
F Correction factor for sending measurements (annex A, subclause A.1.1.2.1)
s
F Correction factor for terminal coupling loss measurements (annex A,
tcl
subclause A.1.1.2.3)
HFRP HandsFree Reference Point
ISDN Integrated Services Digital Network
LD-CELP Low Delay-Code Excited Linear Prediction
MRP Mouth Reference Point
PCM Pulse Code Modulation
PSTN Public Switched Telephone Network
RLR Receiving Loudness Rating
rms root mean square
S/D Signal to Distortion
SB-ADPCM Sub Band-Adaptive Differential Pulse Code Modulation
SLR Sending Loudness Rating
TCL Terminal Coupling Loss
TCLw Weighted Terminal Coupling Loss
TEUT Telephone Equipment Under Test
Page 12
ETS 300 807: November 1997
4 System description
For the purposes of this ETS a teleconference terminal can be represented by the functional block
diagram shown in figure 1.
Video I/O
Video Codec(s)
Audio Codec(s)
Audio I/O MUX
(incl. delay)
Network
&
Interface
Data I/O for
DEMUX
telematic aids
ISDN
End-to-end
signalling
SYSTEM
CONTROL
D-channel
signalling
Figure 1: Functional block diagram of a teleconference terminal
The following units compose a teleconference terminal:
- video I/O-includes camera(s), monitor(s) and video processing units providing functions like split
screen, picture superposition (e.g. for combining the pictures generated by telewriting and still
pictures) and text generation (e.g. for supporting conference management and text processing
features);
- audio I/O-inclusive of handset/headset and handsfree facilities, both for single users and for
multiple users;
- data I/O-used for interfacing telematic aids to the teleconference terminal (e.g. external PCs, fax
machines, telewriting facilities, electronic blackboards, etc.);
- system control-carrying out the following functions: network access through user-to-network
signalling, end-to-end in band signalling, control of audio and video codecs, control of data I/O
facilities, human interface with the user;
NOTE 1: End-to-end signalling is defined in I-ETS 300 143 [11] and I-ETS 300 144 [10].
- audio codec(s)-performing CCITT Recommendations G.711 [12], G.722 [13] and G.728 [14] coding
according to in band and D-channel signalling information;
- video codec(s)-performing moving and still picture coding according to the terminal characteristics;
- mux/demux unit-multiplexes transmitted video, audio, data and control signals into a single bit
stream and demultiplexes the received bit stream, as defined in I-ETS 300 144 [10];
- network interface-makes the necessary adaptation between the network and the terminal according
to the user-network interface requirements.
Page 13
ETS 300 807: November 1997
NOTE 2: Teleconference terminals using up to 2 B channels can use the ISDN Basic Access,
while terminals designed for using more then 2 B channels can either be equipped with
more Basic Access interfaces or with a Primary Rate Interface.
The teleconference terminals can be connected through the ISDN both to other terminals, either of the
same type or of different types, or to Multipoint Conference Units. The behaviour of the terminals for both
type of connections shall be the same.
4.1 Audio facilities
The teleconference terminals are intended in principle for supporting communications between groups of
users. However, single-user terminals compliant with this ETS can also be designed for allowing a
single-user access to audiographic teleconferencing in order to permit, for instance, single users to enter
multipoint conferences or to connect end-to-end to multi-user terminals.
The following range of audio facilities can be provided by teleconference terminals:
- handset (single user);
- headset (single user);
- handsfree (single user);
- handsfree (multiple users).
Some implementations can provide audio facilities to groups of users by supplying each user with a
personal hands-free unit. The same requirements applicable to single user handsfree terminals apply to
this type of multiple user terminal.
4.2 Audio encoding
Audiographic terminals shall be able to operate both narrow band, CCITT Recommendations G.711 [12]
Pulse Code Modulation (PCM) and G.728 [14], and wide band CCITT Recommendation G.722 [13]
coding. The default mode for narrow band PCM coding is A-law, however μ-law coding shall also be
implemented.
4.2.1 CCITT Recommendation G.711 encoding
4.2.1.1 A-law
At the beginning of a call operation, mode 0F (CCITT Recommendation G.711 [12]) shall be used.
The default encoding shall be A-law.
When in mode 0U the requirements of I-ETS 300 245-2 [2] and I-ETS 300 245-3 [3] shall be met for
handset and handsfree operations respectively.
4.2.1.2 μ-law
If information is available to the terminal, either by configuration or by user input, as to whether the
destination is within a μ-law region, then this encoding law shall be used after the reception of the
ALERTING message or, if the ALERTING message is not received, the CONNECT message, or inband
signalling, as described in I-ETS 300 245-5 [4] shall be initiated. The information shall be encoded using
the μ-law at 64 kbit/s as defined in CCITT Recommendation G.711 [12].
It is the responsibility of the calling terminal to ensure that the correct encoding law is used. If no indication
on the coding law has been received during the D-channel signalling sequence or during the in-band
signalling sequence, the calling terminal shall use the default coding law while monitoring the statistics of
the incoming signal. In order to determine whether the incoming signal was encoded by A-law or μ-law
PCM, the algorithm described in appendix 1 to CCITT Recommendation G.725 [15] shall be used.
Compliance with the CCITT Recommendation G.725 [15] algorithm implementation shall be checked by
the test described in subclause A.2.7.
For terminals also supporting handset operations, conformance to the μ-law coding requirements shall be
checked in the handset mode, as specified in I-ETS 300 245-2 [2].
Page 14
ETS 300 807: November 1997
For terminals not supporting the handset mode the coding requirements shall be checked by using the
test methods specified in I-ETS 300 245-3 [3], with the following amendments:
- the test signal generator and analyser shall use μ-law encoding/decoding;
- only sensitivity/frequency, loudness rating, harmonic distortion and noise requirements shall be
verified;
- the sending noise shall meet the requirement described in ITU-T Recommendation P.310 [28],
section 4.
4.2.2 CCITT Recommendation G.722 encoding
When operating in modes 1, 2 or 3 the signal shall be encoded as specified in CCITT Recommendation
G.722 [13].
The lower sub-band shall be encoded using 6 bits independent of operational mode. When operating in
modes 2 or 3, the least significant bit or the two least significant bits shall be used for the auxiliary data
channel, see CCITT Recommendation G.722 [13], section 1.3.
4.2.3 CCITT Recommendation G.728 encoding
When audio coding at 16 kbit/s, the Low Delay-Code Excited Linear Prediction (LD-CELP) speech coding
algorithm as defined in CCITT Recommendation G.728 [14] shall be used.
4.3 Audio decoding
4.3.1 CCITT Recommendation G.711 decoding
4.3.1.1 A-law
At the beginning of a call, the operating mode 0F (CCITT Recommendation G.711 [12]) shall be used.
The default decoding law shall be A-law.
When in mode 0U the requirements of I-ETS 300 245-2 [2] and I-ETS 300 245-3 [3] shall be met for
handset and handsfree operations respectively.
4.3.1.2 μ-law
When a terminal has detected, by D-channel signalling, in-band signalling or by other means, that the
incoming bit stream is μ-law encoded, the decoding algorithm shall fulfil the characteristics described in
CCITT Recommendation G.711 [12].
For terminals also supporting handset operations, the conformance to the μ-law decoding requirements
shall be checked in the handset mode as specified in I-ETS 300 245-2 [2]. For terminals not supporting
the handset mode the coding requirements shall be checked by using the test methods specified in
I-ETS 300 245-3 [3], with the first two amendments specified in subclause 4.2.1.2.
4.3.2 CCITT Recommendation G.722 decoding
The same requirements specified in I-ETS 300 245-5 [4] for wide band handset terminals with respect to
the operating mode selection and to fallback procedures to 3,1 kHz telephony and to Public Switched
Telephone Network (PSTN) are applicable.
4.3.3 CCITT Recommendation G.728 decoding
When audio coding at 16 kbit/s, the LD-CELP speech decoding algorithm as defined in CCITT
Recommendation G.728 [14] shall be used.
NOTE: The quality provided by CCITT Recommendation G.728 [14] speech coding is not
recommended for high quality hands free operations.
Page 15
ETS 300 807: November 1997
4.4 Relative level
The digital interface is defined as a 0 dBr point according to ITU-T Recommendation G.101 [30].
5 Speech transmission characteristics
The audiographic terminals shall be able to interwork with PSTN and ISDN telephones, as well as with
ISDN videophones and wide band telephones. To this purpose, the audio standards applicable to these
devices are also applicable to audiographic terminals as appropriate and specified below.
5.1 Handset mode
The requirements and measurement methods of I-ETS 300 245-2 [2] and I-ETS 300 245-5 [4] apply,
respectively for narrow band and wide band operations at an audio bit rate of 64 kbit/s.
For narrow band operations at 16 kbit/s I-ETS 300 245-8 [6] applies.
If lip-synchronization delay is introduced, the delay requirement and measurement method specified by
I-ETS 300 302-1 [7] applies.
5.2 Headset mode
In principle the requirements for headset operations shall be based on the requirements for handset
operations, with the following allowances:
- the suitable positioning of the microphone with respect to the Mouth Reference Point (MRP)
position shall be defined;
- an appropriate Artificial Ear shall be used and the measurement results shall be referred to the Ear
Reference Point (ERP) position as specified in ITU-T Recommendation P.57 [16].
NOTE: The subject is for further study.
5.3 Hands-free mode (single user)
The requirements and measurement methods of I-ETS 300 245-3 [3] and I-ETS 300 245-6 [5] apply,
respectively for narrow band and wide band operations at an audio bit rate of 64 kbit/s.
If lip-synchronization delay is introduced, the delay requirement and measurement method specified by
I-ETS 300 302-1 [7] applies.
For narrow band PCM coding at 56 kbit/s and LD-CELP coding at 16 kbit/s, I-ETS 300 302-2 [8] applies.
For wide band operations at 56 kbit/s and 48 kbit/s, I-ETS 300 302-3 [9] applies.
5.4 Hands-free mode (multiple users)
Multiple users audiographic terminals can be designed for guaranteed optimum audio performance when
installed in "typical" conference rooms (i.e. in "live" environments with acoustic reverberation effects and
background noise). To this purpose, suitable signal processing techniques can be used for eliminating
reverberation in the transduced signals and/or suppressing room noise. Performance under real operating
conditions can then only be evaluated in the actual installation environments, whose characteristics can
not be specified in a general way.
This ETS is not intended to cover all the installation conditions of multiple user audiographic terminals, but
specifies the audio performances under reference echo-free conditions. In case the specified testing
environment is not compatible with the technical characteristics of specific terminals, the manufacturers
are given the option to state alternative, more suitable acoustic testing environments.
A testing procedure intended to guarantee the correct audio alignment of actually installed terminals is
provided in annex C.
Page 16
ETS 300 807: November 1997
5.4.1 Receiving volume control and sensitivity adjustments
Due to the wide range of operating conditions, both the sending and receiving sensitivity of multiple-user
handsfree terminals can generally be adjusted at the installation, based on the room characteristics and
operational distance of the transducers. These adjustment controls shall not be accessible to normal
users. An additional user-accessible receiving volume control (referred to in the following as "receiving
volume control") can also be provided in order to allow the user to establish the optimum listening
conditions on the basis of room noise, number of participants and other variable factors.
5.4.1.1 Sending sensitivity adjustment
The sending sensitivity adjustment is intended to allow for varying the positioning of the microphone(s)
with respect to the user(s), according to the installation environment constraints. The manufacturer shall
) for which the
state the regulation range of this control and the microphone-speaker operating distance (d
s
terminal submitted to the tests has been regulated.
5.4.1.2 Receiving sensitivity adjustment
The receiving sensitivity adjustment is intended to allow for varying the positioning of the loudspeaker(s)
with respect to the user(s) according to the installation environment constraints. The manufacturer shall
state the regulation range of this control and the loudspeaker-listener operating distance (d ) for which the
r
terminal actually submitted to the tests has been regulated.
5.4.1.3 Receiving volume control
Unless stated otherwise, the compliance requirements refer to the maximum position (maximum
sensitivity) of the receiving volume control (when manually operated).
The minimum range of the receiving volume control (when manually operated) shall be 15 dB.
The operation of the receiving volume control shall not affect the sending sensitivity. The only
user-accessible control allowed for sending, is the "muting" function of the hands-free microphone.
5.4.1.4 Adaptive gain control (optional)
An adaptive gain control, depending on the level of environmental noise, may be implemented into the set.
The gain variation in the set corresponds to a gain in the receiving path and to a symmetrical attenuation
in the sending path for increased ambient noise level. Table 2 of I-ETS 300 245-3 [3] presents, for
guidance and illustration only, three examples of gain variation characteristics.
5.4.2 Sensitivity-frequency response
For narrow band operations I-ETS 300 245-3 [3] and I-ETS 300 302-2 [8] apply. For wide band operations
the following applies.
5.4.2.1 Sending
The sending sensitivity-frequency response (from MRP to digital interface) shall fall within the mask
specified in table 1.
The mask ordinates are in relative dB units; compliance shall be checked by vertically shifting the mask
with respect to the sending characteristic of the terminal under test.
Page 17
ETS 300 807: November 1997
Table 1: Sending sensitivity/frequency mask
Centre Upper limit Lower limit
Frequency
[Hz] [dB] [dB]
100 4
-�
125 4 -7
160 4 -5,5
200 4 -4
250 4 -4
315 4 -4
400 4 -4
500 4 -4
630 4 -4
800 4 -4
1 000 4 -4
1 250 4,6 -4
1 600 5,2 -4
2 000 5,9 -4
2 500 6,5 -4
3 150 7,1 -4
4 000 7,7 -4
5 000 8,4 -4
6 300 9 -7
8 000 9
-�
NOTE: Under ideal conditions, the speech quality can be improved by transmitting frequencies
lower than 125 Hz. However, under normal operating conditions, extending the
frequency range to lower frequencies can cause excessive transmission of unwanted
noise. Furthermore, a rising sensitivity-frequency response is recommended.
Compliance shall be checked by the tests described in subclause A.2.1.1.
5.4.2.2 Receiving
The receiving sensitivity-frequency response (from the digital interface to the measurement point) shall fall
within the mask specified in table 2.
The mask ordinates are in relative dB units; compliance shall be checked by vertically shifting the mask
with respect to the receiving characteristic of the terminal under test.
Page 18
ETS 300 807: November 1997
Table 2: Receiving sensitivity/frequency mask
Centre Upper limit Lower limit
Frequency
[Hz] [dB] [dB]
100 8
-∞
125 8
-∞
160 8 -15
200 8 -9
250 8 -6
315 7 -6
400 6 -6
500 6 -6
630 6 -6
800 6 -6
1 000 6 -6
1 250 6 -6
1 600 6 -6
2 000 6 -6
2 500 6 -6
3 150 6 -6
4 000 6 -6
5 000 6 -6
6 300 6 -9
8 000 6
-∝
Compliance shall be checked by the tests described in subclause A.2.1.2.
5.4.3 Loudness rating
5.4.3.1 Sending
The nominal value of Sending Loudness Rating (SLR) shall be:
SLR = (+12 - F ) dB
s
where F is as defined in subclause A.1.1.2.1.
s
A manufacturing tolerance of ±3 dB is allowed.
Compliance shall be checked by measurement of the SLR as described in subclause A.2.2.1.
5.4.3.2 Receiving
The nominal value of Receiving Loudness Rating (RLR) shall be:
RLR = (+6 - F ) dB
r
where F is as defined in subclause A.1.1.2.2.
r
A manufacturing tolerance of ±3 dB is allowed.
The RLR measured with the manual volume control at the maximum position shall be:
min
RLR = (-4 - F ) dB
min r
A manufacturing tolerance of ±3 dB is allowed.
The RLR measured with the volume control at its minimum position shall be by 15 dB to 30 dB quieter
(higher) than RLR .
min
Page 19
ETS 300 807: November 1997
The nominal value of RLR shall be met (within its tolerance) for at least one setting of the volume control
(when manually operated).
For sets only equipped with automatic (receiving) gain control, the RLR measured with an input signal of
-15 dBm0 shall be higher by 10 dB to 15 dB than the RLR measured with an input signal of -30 dBm0. The
nominal RLR shall be included in the measured range. The RLR measured with an input signal of
-30 dBm0 shall be:
RLR (-30 dBm0) = (-4 - F ) dB.
r
A manufacturing tolerance of ±3 dB is allowed.
Compliance shall be checked by measurement of the RLR as described in subclause A.2.2.2 with the
volume control set as specified.
5.4.4 Terminal coupling loss
For narrow band operations I-ETS 300 245-3 [3] and I-ETS 300 302-2 [8] apply. For wide band operations
the following applies.
5.4.4.1 TCL
w
The Weighted Terminal Coupling Loss (TCL ), measured from the digital input to the digital output shall
w
be at least 35 dB.
NOTE: The TCL requirement specified here refers to stationary single-talk conditions.
w
Reduced performances may occur during the transient switching conditions of the
echo control devices and under double-talk conditions. Further information on this
subject is available in table 3 of I-ETS 300 245-3 [3].
Compliance shall be checked by the tests described in subclause A.2.3.1.
5.4.4.2 Stability loss
The attenuation from the digital input to the digital output shall be at least 6 dB at all frequencies in the
range from 100 Hz to 8 kHz.
Compliance shall be checked by the test described in subclause A.2.3.2.
5.4.5 Distortion
For narrow band operations I-ETS 300 245-3 [3] and I-ETS 300 302-2 [8] apply. For wide band operations
the following applies.
The distortion requirement is specified in terms of the total distortion evaluated in the frequency band from
100 Hz to 7 kHz (harmonic and quantizing) measured with input signals of 300 Hz, 1 kHz and 6 kHz.
According to CCITT Recommendation G.722 [13], where the nominal 1 kHz frequency is indicated, the
actual measurement frequency shall be 1 020 Hz (+2 Hz / -7 Hz).
5.4.5.1 Sending
The sending Signal to Distortion (S/D) ratio is the ratio of the signal power of the measurement tone to the
distortion power at the digital output. The S/D ratio shall be above the limits given in table 3.
Limits for the intermediate levels of the 1 kHz measurement are found by drawing straight lines between
the breaking points in the table on a linear (dB signal level)-linear (dB ratio) scale.
Measurements at 300 Hz and 6 kHz shall only be carried out at -3 dB rel. Acoustic Reference Level (ARL).
Page 20
ETS 300 807: November 1997
Table 3: Limits for signal to total distortion ratio
Tone input level 300 Hz 1 kHz 6 kHz
[dB rel.ARL] [dB] [dB] [dB]
10 24,5
3 35,0
-3 29,0 35,0 29,0
-11 35,0
-18 35,0
-40 15,0
Complian
...
Questions, Comments and Discussion
Ask us and Technical Secretary will try to provide an answer. You can facilitate discussion about the standard in here.
Loading comments...