IEEE P269/D23D25 Draft Standard Methods for Measuring

IEEE P269/D25 October 2004
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IEEE P269/D23D25
(Revision of IEEE Std 269-2002)
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Draft Standard Methods for Measuring Transmission Performance
of Analog and Digital Telephone Sets, Handsets, and Headsets
Prepared by the Subcommittee on Telephone Instrument Testing of the Transmission, Access and Optical Systems
Committee of the IEEE Communications Society (formerly the IEEE Communication Technology Group).
Copyright © 2002 by the Institute of Electrical and Electronics Engineers, Inc.
Three Park Avenue
New York, New York 10016-5997, USA
All rights reserved. This document is an unapproved draft of a proposed IEEE Standard. As such, this document is
subject to change. USE AT YOUR OWN RISK! Because this is an unapproved draft, this document must not be
utilized for any conformance/compliance purposes. Permission is hereby granted for IEEE Standards Committee
participants to reproduce this document for purposes of IEEE standardization activities only. Prior to submitting this
document to another standards development organization for standardization activities, permission must first be
obtained from the Manager, Standards Licensing and Contracts, IEEE Standards Activities Department. Other
entities seeking permission to reproduce this document, in whole or in part, must obtain permission from the
Manager, Standards Licensing and Contracts, IEEE Standards Activities Department.
IEEE Standards Activities Department
Standards Licensing and Contracts
445 Hoes Lane, P.O. Box 1331
Piscataway, NJ 08855-1331, USA
Abstract:
Practical methods for making laboratory measurements of electroacoustic characteristics of analog
and digital telephones, handsets and headsets. The methods may also be applicable to a wide variety of other
communications equipment, including cordless, wireless and mobile communications devices. Measurement results
may be used to evaluate these devices on a standardized basis. Application is in the frequency range from 100 to
8,500 Hz.
Keywords: analog telephones, digital telephones, handsets, headsets, electroacoustic measurements on
telephones, telephony, voice transmission performance.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
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Introduction
(This introduction is not a part of IEEE P269, IEEE Draft Standard Methods for Measuring Transmission
Performance of Analog and Digital Telephone Sets, Handsets, and Headsets.)
This standard has been prepared in response to a widely expressed need by the telecommunications industry for a
standard, comprehensive method for testing the transmission performance of telephone sets, handset, and headsets.
The present standard is a revision of IEEE Std 269-1992. This revision adds coverage for a wide range of ear
simulators and test signals, and incorporates and updates the contents of IEEE Std 1206-1994.
The IEEE will maintain this standard current with the state of the technology. Comments on this standard and
suggestions for the additional material that should be included are invited. Comments should be addressed to:
Secretary, IEEE Standards Board, The Institute of Electrical and Electronics Engineers, Inc., 345 East 47th Street,
New York, NY 10017.
This revision, begun in 1999, was prepared by the Subcommittee on Telephone Instrument Testing of the
Transmission, Access and Optical Systems Committee of the IEEE Communications Society (formerly the IEEE
Communication Technology Group).
At the time this standard was approved the members of the subcommittee were as follows:
John Bareham, Chair
Glenn Hess, Vice Chair
Steve Graham, Secretary
Roger Britt
Chandru Butani
Rodolfo Ceruti
Cliff Chamney
Paul Coverdale
Fred Dekalb
Gijs Dirks
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Dan Foley
H. W. Gierlich
Deborah Gruenhagen
Roger Gutzwiller
Joe Helms
Soren Jonsson
Frederick M. Kruger
Ron Magnuson
Henry Mar
Christopher J. Struck
Steve Temme
Stephen Whitesell
Allen Woo
Robert Young
The following members of the balloting committee voted on this standard:
John Bareham
Chandru Butani
Cliff Chamney
Fred Dekalb
Dan Foley
Steve Graham
Deborah Gruenhagen
Roger Gutzwiller
Glenn Hess
Frederick M. Kruger
Ron Magnuson
Henry Mar
Christopher J. Struck
Stephen Whitesell
Allen Woo
Robert Young
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Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
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Contents
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Overview .............................................................................................................................................................. 9
1.1 Scope .............................................................................................................................................................. 9
1.2 Purpose ........................................................................................................................................................... 9
1.3 Contents of standard ....................................................................................................................................... 9
References........................................................................................................................................................... 11
Definitions .......................................................................................................................................................... 13
Abbreviations, Acronyms and Symbols.............................................................................................................. 16
4.1 Abbreviations and acronyms ........................................................................................................................ 16
4.2 Symbols ........................................................................................................................................................ 16
Test Equipment and Setup .................................................................................................................................. 18
5.1 Ear simulators............................................................................................................................................... 18
5.1.1
Selection.............................................................................................................................................. 18
5.1.2
Headset measurements made on ear simulators compared to real ears............................................... 18
5.1.3
Translation from DRP to ERP ............................................................................................................ 19
5.2 Mouth simulators.......................................................................................................................................... 19
5.3 Test Fixtures ................................................................................................................................................. 19
5.3.1
Selection.............................................................................................................................................. 19
5.3.2
Handset positioning ............................................................................................................................ 20
5.3.3
Headset positioning......................................................................................................................242423
5.4 Measurement microphones........................................................................................................................... 27
5.5 Test environment .......................................................................................................................................... 27
5.5.1
Background noise level................................................................................................................282827
5.5.2
Reflection-free conditions............................................................................................................282827
5.5.3
Diffuse field conditions....................................................................................................................... 28
5.6 Acoustic impairments................................................................................................................................... 29
5.6.1
Reference corner ................................................................................................................................. 29
5.6.2
Hoth room noise...........................................................................................................................303029
Calibration ...................................................................................................................................................313130
6.1 General ..................................................................................................................................................313130
6.2 Electrical measurement instruments ......................................................................................................313130
6.3 Measurement microphones....................................................................................................................313130
6.4 Ear simulator .........................................................................................................................................313130
6.5 Measurement bandwidth and resolution................................................................................................313130
6.6 Electrical test signals .............................................................................................................................313130
6.6.1
Electrical test spectrum ................................................................................................................323231
6.6.2
Electrical test level.......................................................................................................................323231
6.7 Mouth simulator ....................................................................................................................................323231
6.7.1
Acoustic test spectrum .................................................................................................................323231
6.7.2
Acoustic test level ........................................................................................................................333332
6.7.3
Mouth simulator calibration procedure........................................................................................333332
Test Procedure for Analog Sets ...................................................................................................................343433
7.1 General ..................................................................................................................................................343433
7.1.1
Choice of test signals and levels ..................................................................................................343433
7.1.2
Measurement bandwidth and resolution ......................................................................................343433
7.1.3
Choice of ear and mouth simulators and test position..................................................................353534
7.1.4
Tone control setting .....................................................................................................................353534
7.1.5
Reference receive volume control setting ....................................................................................353534
7.1.6
Reference send gain control setting .............................................................................................353534
7.2 Analog DC Feed circuits .......................................................................................................................353534
7.3 Analog telephone network impairments................................................................................................373736
7.3.1
Loop current.................................................................................................................................373736
7.3.2
Network noise ..............................................................................................................................373736
7.3.3
Termination impedance ...............................................................................................................373736
7.3.4
Test loops.....................................................................................................................................383837
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7.3.5
Parallel sets ..................................................................................................................................383837
7.3.6
Cordless range..............................................................................................................................403938
7.4 Receive ..................................................................................................................................................403938
7.4.1
Receive frequency response.........................................................................................................403938
7.4.2
Receive noise ...............................................................................................................................404039
7.4.3
Receive narrow-band noise..........................................................................................................414039
7.4.4
Receive linearity ..........................................................................................................................414039
7.4.5
Receive distortion ........................................................................................................................414140
7.4.6
Receive mute leakage ..................................................................................................................414140
7.5 Send .......................................................................................................................................................424140
7.5.1
Send frequency response..............................................................................................................424140
7.5.2
Send noise ...................................................................................................................................424241
7.5.3
Send narrow-band noise...............................................................................................................424241
7.5.4
Send linearity ...............................................................................................................................434241
7.5.5
Send distortion .............................................................................................................................434342
7.5.6
Send mute leakage .......................................................................................................................434342
7.5.7
Send frequency response in a diffuse field...................................................................................444342
7.5.8
Send signal-to-noise ratio.............................................................................................................444443
7.6 Sidetone .................................................................................................................................................444443
7.6.1
Talker sidetone frequency response .............................................................................................444443
7.6.2
Listener sidetone frequency response ..........................................................................................454443
7.6.3
Alternate method for listener sidetone .........................................................................................454544
7.6.4
Sidetone linearity .........................................................................................................................464544
7.6.5
Sidetone distortion .......................................................................................................................464645
7.6.6
Sidetone delay..............................................................................................................................464645
7.6.7
Sidetone echo response ................................................................................................................464645
7.7 Overall ...................................................................................................................................................474645
7.7.1
Overall frequency response..........................................................................................................474645
7.7.2
Overall linearity ...........................................................................................................................474645
7.7.3
Overall distortion .........................................................................................................................474746
7.8 Telephone set impedance ......................................................................................................................474746
7.8.1
AC impedance..............................................................................................................................484746
7.8.2
Return loss ...................................................................................................................................484746
7.9 Howling .................................................................................................................................................484847
7.10
Maximum acoustic output ................................................................................................................494847
7.10.1
Maximum acoustic pressure (long duration)................................................................................494948
7.10.2
Peak acoustic pressure (short duration)........................................................................................494948
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Test Procedures for Digital and 4-wire Systems..........................................................................................504948
8.1 General ..................................................................................................................................................504948
8.1.1
Choice of test signals and levels ..................................................................................................505049
8.1.2
Measurement bandwidth and resolution ......................................................................................505049
8.1.3
Choice of ear and mouth simulators and test position..................................................................515049
8.1.4
Tone control setting .....................................................................................................................515049
8.1.5
Reference receive volume control................................................................................................515049
8.1.6
Reference send gain control setting .............................................................................................515150
8.2 Digital test circuits.................................................................................................................................515150
8.2.1
Digital telephone interface ...........................................................................................................515150
8.2.2
Reference codec ...........................................................................................................................535251
8.2.3
Wideband reference codec ...........................................................................................................535352
8.3 Digital telephone network impairments.................................................................................................545352
8.3.1
Network Delay .............................................................................................................................545453
8.3.2
Jitter .............................................................................................................................................545453
8.3.3
Network Packet Loss ...................................................................................................................555453
8.3.4
Network Echo Canceller ..............................................................................................................555453
8.3.5
Discontinuous Speech Transmission............................................................................................555554
8.4 Receive ..................................................................................................................................................555554
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8.4.1
Receive frequency response.........................................................................................................555554
8.4.2
Receive noise ..............................................................................................................................565554
8.4.3
Receive narrow-band noise..........................................................................................................565554
8.4.4
Receive linearity ..........................................................................................................................565655
8.4.5
Receive distortion ........................................................................................................................565655
8.4.6
Receive mute leakage ..................................................................................................................575655
8.4.7
Receive delay ...............................................................................................................................575756
8.4.8
Receive out-of-band signals.........................................................................................................575756
8.5 Send .......................................................................................................................................................585756
8.5.1
Send frequency response..............................................................................................................585756
8.5.2
Send noise ....................................................................................................................................585857
8.5.3
Send narrow-band noise...............................................................................................................585857
8.5.4
Send linearity ...............................................................................................................................595857
8.5.5
Send distortion .............................................................................................................................595857
8.5.6
Send mute leakage .......................................................................................................................595958
8.5.7
Send delay....................................................................................................................................595958
8.5.8
Send out-of-band susceptibility ...................................................................................................605958
8.5.9
Send frequency response in a diffuse field...................................................................................606059
8.5.10
Send signal-to-noise ratio.............................................................................................................606059
8.6 Sidetone .................................................................................................................................................606059
8.6.1
Talker sidetone frequency response .............................................................................................616059
8.6.2
Listener sidetone frequency response ..........................................................................................616160
8.6.3
Alternate method for listener sidetone .........................................................................................626160
8.6.4
Sidetone linearity .........................................................................................................................626261
8.6.5
Sidetone distortion .......................................................................................................................626261
8.6.6
Sidetone delay..............................................................................................................................636261
8.6.7
Sidetone echo response ................................................................................................................636261
8.7 Overall ...................................................................................................................................................636261
8.7.1
Overall frequency response..........................................................................................................636261
8.7.2
Overall linearity ...........................................................................................................................636362
8.7.3
Overall distortion .........................................................................................................................646362
8.8 Echo frequency response .......................................................................................................................646362
8.9 Temporally weighted terminal coupling loss ........................................................................................656463
8.10
Stability loss .....................................................................................................................................656564
8.11
Convergence time .............................................................................................................................666564
8.12
Discontinuous speech transmission ..................................................................................................666665
8.12.1
General.........................................................................................................................................666665
8.12.2
Measurement method...................................................................................................................666665
8.13
Maximum acoustic output ................................................................................................................676766
8.13.1
Maximum acoustic pressure (long duration)................................................................................676766
8.13.2
Peak acoustic pressure (short duration)........................................................................................686766
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Test Procedures for Analog 4-wire Handsets and Headsets ........................................................................706968
9.1 General ..................................................................................................................................................706968
9.1.1
Choice of test signals and levels ..................................................................................................706968
9.1.2
Measurement bandwidth and resolution ......................................................................................706968
9.1.3
Choice of ear and mouth simulators and test position..................................................................717069
9.1.4
Tone control setting .....................................................................................................................717069
9.1.5
Default receive volume control and send gain adjustment...........................................................717069
9.2 Handset and headset test circuits ...........................................................................................................717069
9.3 Receive ..................................................................................................................................................737271
9.3.1
General.........................................................................................................................................737271
9.3.2
Receive volume control adjustment .............................................................................................737271
9.3.3
Receive frequency response............................................................................................................ 7472
9.3.4
Receive noise ..............................................................................................................................747372
9.3.5
Receive narrow-band noise..........................................................................................................747372
9.3.6
Receive linearity ..........................................................................................................................747372
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9.3.7
Receive distortion ........................................................................................................................757473
9.3.8
AC impedance..............................................................................................................................757473
9.3.9
DC resistance ...............................................................................................................................757473
9.4 Send .......................................................................................................................................................757473
9.4.1
Send gain control adjustment .......................................................................................................757473
9.4.2
Send frequency response................................................................................................................. 7674
9.4.3
Send noise ....................................................................................................................................767574
9.4.4
Send narrow-band noise...............................................................................................................767574
9.4.5
Send linearity ...............................................................................................................................767574
9.4.6
Send distortion .............................................................................................................................777675
9.4.7
Send frequency response in a diffuse field...................................................................................777675
9.4.8
Send signal-to-noise ratio................................................................................................................ 7876
9.4.9
AC impedance..............................................................................................................................787776
9.4.10
DC resistance ...............................................................................................................................787776
9.5 Echo frequency response .......................................................................................................................787776
9.6 Maximum acoustic output .....................................................................................................................797877
9.6.1
Maximum acoustic pressure (long duration)................................................................................797877
9.6.2
Peak acoustic pressure (short duration)........................................................................................797877
Annex A (normative) Ear Simulators with Flexible Pinnas and Positioning Devices......................................807978
A.1
General characteristics of the ear simulators ....................................................................................807978
A.2
Differences between the two ear simulators .....................................................................................807978
A.3
Handset Positioning devices .............................................................................................................818079
Annex B (normative) Alternative Ear Simulators, Mouth Simulator and Test Fixture....................................828180
B.1
Alternative Ear Simulators................................................................................................................828180
B.2
Alternative Mouth Simulator ............................................................................................................848382
B.2.1
General.........................................................................................................................................848382
B.2.2
Calibration of Alternative Mouth Simulator ...............................................................................848382
B.3
Alternative Test Fixture ....................................................................................................................858483
Annex C (normative) DRP TO ERP and Related Translations........................................................................868584
Annex D (normative) Conditioning for Carbon Transmitters ..........................................................................908988
Annex E (normative) Hoth Room Noise..........................................................................................................919089
Annex F (normative) Test Signals ..................................................................................................................939291
F.1 General ..................................................................................................................................................939291
F.2 Classifications .......................................................................................................................................939291
F.3 Modulation types ...................................................................................................................................939291
F.3.1
Square wave modulation..............................................................................................................949392
F.3.2
Sine wave modulation..................................................................................................................949392
F.3.3
Pseudo-random modulation .........................................................................................................949392
F.4 Deterministic signals .............................................................................................................................949392
F.4.1
Sine wave.....................................................................................................................................949392
F.4.2
Pseudo-random ............................................................................................................................949392
F.5 Random signals .....................................................................................................................................959493
F.5.1
White noise ..................................................................................................................................959493
F.5.2
Pink noise.....................................................................................................................................959493
F.5.3
P.50 noise.....................................................................................................................................959493
F.6 Speech-like signals ................................................................................................................................959493
F.6.1
Simulated speech .........................................................................................................................959493
F.6.2
Synthesized speech ......................................................................................................................969594
F.6.3
Real speech ..................................................................................................................................969594
F.7 Compound signals .................................................................................................................................979695
F.7.1
Sequential presentation ................................................................................................................979695
F.7.2
Simultaneous presentation ...........................................................................................................979695
F.8 Test signal bandwidth............................................................................................................................999897
F.9 Signal parameter summary ..................................................................................................................1009998
F.10
Test signals published on CD-ROM ...............................................................................................1009998
F.11
Signal and test method comparative summary .............................................................................10110099
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Annex G (normative) Analysis Methods...................................................................................................102101100
G.1
General .......................................................................................................................................102101100
G.2
Fast Fourier transform (FFT) and cross spectrum analysis.........................................................103102101
G.2.1
Dual-channel FFT ..................................................................................................................103102101
G.2.2
Single-channel FFT................................................................................................................103102101
G.2.3
Maximum length sequence (MLS) analysis...........................................................................103102101
G.3
Real-time filter analysis (RTA) ..................................................................................................103102101
G.3.1
Dual-channel real-time filter analysis ....................................................................................104103102
G.3.2
Single-channel real-time filter analysis..................................................................................104103102
G.4
Sine-based analysis.....................................................................................................................104103102
G.4.1
Discrete tone (stepped sine) ...................................................................................................104103102
G.4.2
Swept sine ..............................................................................................................................104103102
G.4.3
Time delay spectrometry (TDS) ............................................................................................104103102
G.5
Simulated free field techniques...................................................................................................105104103
G.6
Measurement bandwidth.............................................................................................................105104103
G.7
Measurement resolution..............................................................................................................106105104
Annex H (normative) Loudness Rating Calculations .......................................................................................110108
Annex I (normative) Linearity ........................................................................................................................112110
Annex J (normative) Distortion ......................................................................................................................117115
J.1 Overview ...............................................................................................................................................117115
J.2 Signal suitability test .............................................................................................................................117115
J.3 Signal-to-distortion-and-noise ratio (SDN) ...........................................................................................117115
J.4 Sinusoidal Methods ...............................................................................................................................118116
J.4.1
Total harmonic distortion (THD) and harmonic analysis ............................................................119116
J.4.2
Total Harmonic Distortion (THD) and noise ...............................................................................119117
J.4.3
Difference-frequency distortion (DF Distortion) .........................................................................122118
J.4.4
Intermodulation distortion (IM Distortion) ..................................................................................122119
J.4.5
Alternatives to sinewave stimulus signals....................................................................................122119
J.4.6
Test frequencies ...........................................................................................................................122119
J.5 Coherence methods (N/C Ratio)............................................................................................................123119
Annex K (normative) Send Signal-to-Noise Ratio ...........................................................................................125121
K.1
Send signal-to-noise ratio .................................................................................................................125121
K.2
Weighted send signal-to-noise ratio .................................................................................................125121
Annex L (normative) Delay .............................................................................................................................127123
L.1 General ..................................................................................................................................................127123
L.2 Captured pulse method ..........................................................................................................................127123
L.3 Two-channel analyzer methods .............................................................................................................127123
L.3.1
Impulse response method.............................................................................................................127123
L.3.2
Cross-correlation method.............................................................................................................127123
L.4 Time Delay Spectrometry Method ........................................................................................................127123
L.5 MLS Method .........................................................................................................................................128124
Annex M (normative) Sidetone Echo ...............................................................................................................129125
Annex N (informative) Maximum Acoustic Pressure Limits...........................................................................130126
N.1
Abstract.............................................................................................................................................130126
N.2
Introduction ......................................................................................................................................130126
N.3
Proposal ............................................................................................................................................133129
Annex O (normative) Temporally weighted terminal coupling loss measurement method .............................135131
O.1
General .............................................................................................................................................135131
O.2
Initial signal processing ....................................................................................................................135131
O.3
Modeling echo audibility ..................................................................................................................135131
O.3.1
Frequency weighting....................................................................................................................136132
O.3.2
Temporal combination .................................................................................................................136132
O.3.3
Temporal weighting .....................................................................................................................136132
O.4
Expressing TCL Results ...................................................................................................................137133
Annex P (normative) Temporally weighted terminal coupling loss algorithm................................................139135
P.1 General ..................................................................................................................................................139135
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P.2 TCLT Algorithm ....................................................................................................................................139135
P.2.1
Step 1: measure EPD ...................................................................................................................139135
P.2.2
Step 2: align signals .....................................................................................................................140136
P.2.3
Step 3: apply A-weighting ...........................................................................................................140136
P.2.4
Step 4: subtract noise (conditional)..............................................................................................140136
P.2.5
Step 5: 4ms frames.......................................................................................................................140136
P.2.6
Step 6: initialization .....................................................................................................................140136
P.2.7
Step 7: calculations ......................................................................................................................141137
P.2.8
Step 8: calculate parameters.........................................................................................................142138
P.2.9
Step 9: output statistics ................................................................................................................142138
Annex Q (normative) Simulated Speech Generator (SSG) ..............................................................................143139
Annex R (normative) TDS Sweep with P.50 Noise Bursts..............................................................................145141
Annex S (informative) Use of the Free Field as the Telephonometric Reference Point .................................146142
Annex T (informative) Useful Conversion Procedures...................................................................................148144
T.1 Conversions for dBV to dBm, and for 600 and 900...........................................................................148144
T.2 Conversions for dBmp to dBrnC for electrical noise measurements.....................................................149145
T.3 Loudness rating conversions .................................................................................................................149145
T.4 Acoustic sound pressure conventions....................................................................................................149145
Annex U (informative) Loudness Balance Subjective Test Procedure ...........................................................150146
U.1
Introduction ......................................................................................................................................150146
U.2
Loudness balance test procedure ......................................................................................................150146
U.3
Example test circuit ..........................................................................................................................151147
U.4
Estimate of test headset receive characteristics ................................................................................152148
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Draft Standard Methods for Measuring
Transmission Performance of Analog and
Digital Telephone Sets, Handsets, and Headsets
380
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1
Overview
382
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Objective or subjective methods can be used to measure telephone transmission performance. This standard
discusses objective procedures utilizing mouth simulators, ear simulators, laboratory microphones and test
instruments to characterize transmission performance. Subjective procedures are particularly applicable for rating
overall communication connections involving the real voice and real ear of human subjects. Telephones, handsets,
and headsets can be evaluated by purely objective methods provided the results generally agree with the desirable
performance characteristics of subjective testing.
399
1.1
400
401
402
403
404
405
406
407
408
409
410
411
412
This standard provides the techniques for objective measurement of electroacoustic characteristics of analog and
digital telephones, handsets and headsets. Application is in the frequency range from 100 to 8,500 Hz.
413
1.2
414
415
416
417
The purpose of this standard is to provide practical methods for making laboratory measurements of the
transmission characteristics of analog and digital telephones, handsets and headsets so that their performance may be
evaluated on a standardized basis.
418
1.3
419
420
421
This is a brief summary of the clauses contained in the standard. The primary measurement procedures appear in
Clause 7 through Clause 9 of the document.
The relationships that are established between subjective and objective measurements will vary with the physical
constants of the telephone design, such as the size and shape of the handset or headset, the sound leakage between
the receiver and the ear of the user, and the signal processing in the speech path. Therefore, the correlation between
subjective and objective measurements should be established separately for each telephone, headset or handset
design before measurements obtained using the techniques covered herein can be interpreted to reflect performance
under conditions of actual use.
Execution of this standard provides a means of determining the operational characteristics of a telephone over the
range of conditions encountered during normal operation.
Scope
Although not specifically within the scope of this standard, the methods described are generally applicable to a wide
variety of other communications equipment, including cordless, wireless and mobile communications devices.
Telephones with handsfree or loudspeaking features are covered by IEEE Standard 1329-1999, Method for
Measuring Transmission Performance of Handsfree Telephone Sets.
Due to the various characteristics of these devices and the environments in which they operate, not all of the test
procedures in this standard are applicable to all types of telephones, handsets or headsets. Application of the test
procedures to atypical telephones should be determined on an individual basis.
Purpose
Contents of standard
Copyright © 2004 IEEE. All rights reserved.
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Clauses 2, 3 and 4 provide references, definitions, abbreviations, acronyms, and symbols which will be useful in
executing the tests of this standard. These clauses provide a background in the terminology used for telephone,
handset, and headset testing.
Clause 5 specifies the test equipment, test environment and acoustic impairments. The test equipment portion
includes ear and mouth simulators, test fixtures, and measurement microphones, as well as procedures for
positioning the telephone handset or headset for testing. The test environment includes both the acoustical and
physical characteristics of the test space. Impairments include the acoustic conditions.
Clause 6 describes the calibration procedures needed to ensure that the equipment is in a known state. Calibration of
the acoustic transducers and electrical interfaces is explained.
Clauses 7, 8 and 9 contain the transmission test procedures, such as send and receive, for analog telephones, digital
telephones, and analog 4-wire handsets and headsets, respectively.
Attached annexes contain additional information or details of procedures referred to from within the relevant clause.
Normative annexes contain information which is considered to be an official part of the standard. Informative
annexes contain information which may be useful, or of general interest, but are not part of the standard.
Copyright © 2004 IEEE. All rights reserved.
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2
References
This standard shall be used in conjunction with the following publications. When the following standards are
superseded by an approved standard, the revision shall apply but the impact on results should be determined.
ANSI S1.1-1994 (Reaff. 1999), American National Standard Acoustical Terminology.
ANSI S1.4-1983 (Reaff. 2001), American National Standard Specification for Sound Level Meters.
ANSI S1.6-1984 (Reaff. 2001), American National Standard Preferred Frequencies, Frequency Levels, and Band
Numbers for Acoustical Measurements.
ANSI S1.11-1986 (Reaff. 1998), American National Standard Specifications for Octave Band and Fractionaloctave-band Analog and Digital Filters.
ANSI S1.12-1967 (Reaff. 1997), American National Standard Specifications for Laboratory Standard Microphones.
ANSI/TIA/EIA-810-A-2000, Transmission Requirements for Narrowband Voice over IP and Voice over PCM
Digital Wireline Telephones.
IEC 61000-4-5 (2001), Electromagnetic compatibility (EMC) –Part 4-5: Testing and measurement techniques –
Surge immunity test.
I
EEE10
0™,The Authoritative Dictionary of IEEE Standards Terms, Seventh Edition.
IEEE Std 661-1979 (Reaff. 1998), IEEE Standard Method for Determining Objective Loudness Ratings of
Telephone Connections.
IEEE Std 743-1995, IEEE Standard Equipment Requirements and Measurement Techniques for Analog
Transmission Parameters for telecommunications
IEEE Std 1329-1999, IEEE Standard Method for Measuring Transmission Performance of Handsfree Telephone
Sets.
ISO 3 (1973) Preferred Numbers-Series of preferred Numbers.
ITU-T Recommendation G.122 (1993), Influence of National Systems on Stability and Talker Echo in International
Connections.
ITU-T Recommendation G.131 (1996), Control of Talker Echo.
ITU-T Recommendation G.711 (1988), Pulse Code Modulation (PCM) of Voice Frequencies.
ITU-T Recommendation G.714 (1988), Separate Performance Characteristics for the Encoding and Decoding Sides
of PCM Channels Applicable to 4-wire Voice-frequency Interfaces.
ITU-T Recommendation G.722 (1988), 7 kHz Audio-coding Within 64 kbit/s.
ITU-T Recommendation G.723 (1988), Extensions of Recommendation G.721 Adaptive Differential Pulse Code
Modulation to 24 and 40 kbit/s for Digital Circuit Multiplication Equipment Application.
ITU-T Recommendation G.726 (1990), 40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code Modulation
(ADPCM).
Copyright © 2004 IEEE. All rights reserved.
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ITU-T Recommendation G.729 (1996), Coding of Speech at 8 kbit/s Using Conjugate-structure Algebraic-codeexcited Linear-prediction (CS-ACELP).
ITU-T Recommendation O.41 (1994), Psophometer for Use on Telephone-type Circuits.
ITU-T Recommendation O.133 (1993), Equipment for Measuring the Performance of PCM Encoders and Decoders.
ITU-T Recommendation P.50 (1999), Artificial Voices.
ITU-T Recommendation P.50, Appendix 1 (1998), Test Signals.
ITU-T Recommendation P.51 (1996), Artificial Mouths.
ITU-T Recommendation P.56 (1993), Objective Measurement of Active Speech Level.
ITU-T Recommendation P.57 (1996), Artificial Ears.
ITU-T Recommendation P.58 (1996), Head and Torso Simulator for Telephonometry.
ITU-T Recommendation P.59 (1993), Artificial Conversational Speech.
ITU-T Recommendation P.64 (1999), Determination of Sensitivity/Frequency Characteristics of Local Telephone
Systems.
ITU-T Recommendation P.79 (1999), Calculation of Loudness Ratings for Telephone Sets.
ITU-T Recommendation P.501 (2000), Test Signals for Use in Telephonometry.
ITU-T Recommendation P.862 (2001), Perceptual Evaluation of Speech Quality (PESQ), an Objective Method for
End-to-end Speech Quality Assessment of Narrowband Telephone Networks and Speech Codecs.
Copyright © 2004 IEEE. All rights reserved.
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3
Definitions
These definitions apply specifically to measurements of the transmission performance of telephone sets, handsets,
and headsets and may not be applicable to other disciplines. For definitions not covered, see IEEE 100 and ANSI
S1.1-1994.
3.1
A-weighted. A measurement made using the A frequency weighting specified in ANSI S1.4-1983. Aweighted sound pressure level is expressed as dBA, and the reference level is always 20 micropascals.
3.2
acoustic echo path. The acoustic coupling from the handset or headset receiver to the handset or headset
microphone.
3.3
acoustic input. The free-field sound pressure level developed by a mouth simulator at the mouth reference
point. See sound pressure level.
3.4
acoustic output. The sound pressure level developed in an ear simulator. See sound pressure level.
3.5
analog telephone set. A telephone set in which the two-way voice communication interface to the network
is in an analog format.
3.6
boom microphone position (BMP). The default point at which to place a boom microphone for testing on
a mouth simulator. It is specified as measurement point #21 in ITU-T Recommendation P.58. With
respect to the intersection of the mouth reference axis with the lip plane, it is located 6mm back towards the
mouth, 42mm to the right (or left), and 9mm downward.
3.7
codec. A combination of an analog-to-digital encoder and a digital-to-analog decoder operating in opposite
directions of transmission within the same equipment.
3.8
dBA. Sound pressure level in decibels, relative to 20 micropascals, A-weighted (3.1).
3.9
dBm. Power level in decibels, relative to a power of 1 mW (milliwatt).
3.10
dBm0. Power level in dBm, relative to a reference point called the zero transmission level point, or 0 TLP.
(See 3.52). A signal level of X dBm at the 0 TLP is designated X dBm0. In a codec, the 0 TLP is specified
in relationship to the full-scale digital level or saturation. However, digital saturation is generally not 0
dBm0. For -law codecs 0 dBm0 is 3.17 dB below digital full scale. For A-law codecs 0 dBm0 is 3.14 dB
below digital full scale.
3.11
dBm(A). Power level in decibels, relative to a power of 1 mW (milliwatt), A-weighted (3.1).
3.12
dBmp. Power level in decibels, relative to a power of 1mW (milliwatt), measured with psophometric
weighting defined in ITU-T Recommendation O.41.
3.13
dBPa. Sound pressure level in decibels, relative to a sound pressure of 1 Pa (pascal).
3.14
dBSPL. Sound pressure level in decibels, relative to a sound pressure of 20 micropascals.
3.15
dBV. Voltage level in decibels, relative to 1 volt rms.
3.16
dBV(A). Voltage level in decibels, relative to 1 volt rms, measured with A-weighting (3.1).
3.17
dBV(p). Voltage level in decibels, relative to 1 volt rms, measured with psophometric weighting defined
in ITU-T Recommendation O.41.
Copyright © 2004 IEEE. All rights reserved.
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3.18
digital telephone set. A telephone set in which the two-way voice communication interface to the network
is in a digital format.
3.19
ear cap reference point. The intersection of the external ear-cap reference plane with a normal axis
through the effective acoustic center of the sound outlet ports. Generally, the acoustic center of the sound
outlet ports is at the center of their distribution.
3.20
ear reference point.. A virtual point for geometric and acoustic reference located outside the entrance to
the ear canal. The exact location is specified for each type of ear simulator.
3.21
feed circuit. An electrical circuit for supplying dc power to a telephone set and an ac path between the
telephone set and a terminating circuit.
3.22
four-wire transmission. A transmission method, circuit or system which provides separate paths (one pair
each) for signals in the send and receive directions.
3.23
frequency response. Electrical, acoustic, or electroacoustic sensitivity (output/input), or gain, as a
function of frequency.
3.24
head and torso simulator (HATS) for telephonometry. A manikin extending downward from the top of
the head to the waist, designed to simulate the sound pick-up characteristics and acoustic diffraction
produced by a median human adult and to reproduce the acoustic field generated by the human mouth. See
ITU-T Recommendation P.58 (1996).
3.25
howling. Audible squealing sound in a telephone, handset or headset. Acoustic feedback, or oscillation,
typically caused by too much acoustic coupling between the receiver and the microphone.
3.26
ideal codec. A codec that has theoretically optimum characteristics.
3.27
listener sidetone. The signal present at the receiver due to sound in the environment where the telephone is
used.
3.28
loudness rating guard-ring position (LRGP). The test position a handset assumes when it is placed on an
artificial test head as described in Annex A to ITU-T Recommendation P.76 [8].
3.29
microphone. An electroacoustic transducer that converts sound to an electrical signal.
3.30
mouth reference point (MRP). A point on the axis of the mouth simulator, 25 mm in front of the center of
the external plane.
3.31
overall. The direction of speech transmission from the mouth of one person to the ear of another person.
Also called end-to-end.
3.32
overall loudness rating (OLR). A single-number value which corresponds to the perceived loudness loss
of an overall connection, as specified in ITU-T Recommendation P.79 (1999).
3.33
pinna. The flexible part of the outer ear at the side of the head.
3.34
receive. The direction of speech transmission from the network to the ear of the telephone user.
3.35
receiver. An electroacoustic transducer that converts an electrical signal to sound and delivers it directly to
the ear, sealed or unsealed.
3.36
reference codec. A codec that approaches the performance of an ideal codec and has superior, well-defined
characteristics used for testing digital telephone sets.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
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3.37
receive loudness rating (RLR). A single-number value which corresponds to the perceived loudness loss
of a receive connection, as specified in ITU-T Recommendation P.79 (1999).
3.38
reference receive volume control setting. The receive volume control setting of a telephone which results
in the receive loudness rating (RLR) closest to the specified target value.
3.39
reference send gain control setting. The send gain control setting of a telephone which results in the send
loudness rating (SLR) closest to the specified target value.
3.40
send. The direction of speech transmission from the mouth of the telephone user to the network.
3.41
send loudness rating (SLR). A single-number value which corresponds to the perceived loudness loss of
a send connection, as specified in ITU-T Recommendation P.79 (1999).
3.42
sidetone. The direction of speech transmission from the microphone to the receiver of the handset or
headset. There are two types of sidetone to be considered: listener sidetone and talker sidetone.
3.43
single frequency interference (SFI). An audible impairment that can be perceived as a tone relative to the
overall noise level.
.
3.44
sidetone masking rating (STMR). A single-number value which corresponds to the perceived loudness
loss of the talker sidetone connection, as specified in ITU-T Recommendation P.79 (1999).
3.45
sound pressure level. The sound pressure level, in decibels, of a sound is 20 times the logarithm to the
base 10 of the ratio of the pressure of the sound to the reference pressure. For this standard, the reference
pressure is normally 1 pascal (Pa), and sound pressure levels are expressed in dB re 1 Pa (dBPa). When a
reference pressure of 20 uPa is used, the sound pressure level will be expressed as dBSPL. Unless
otherwise indicated, rms values of pressure are used. Most telephony acoustic measurements are
referenced to 1 Pa (1 newton per square meter). However, measurements such as receive noise and room
noise are generally referenced to 20 uPa. Note: 0 dB Pa 94 dBSPL, 0 dBSPL 20 micropascals, 1 Pa 1
N/m2. A-weighted sound pressure level in dB (dBSPL, A-weighted) is often abbreviated as dBA. (see
ANSI S1.4-1983 (R 1997) )
3.46
speaker (also loudspeaker). An electroacoustic transducer that converts an electrical signal to sound and
delivers it to the ear from a distance of several centimeters or greater.
3.47
spectrum. A distribution of amplitude (or phase, or some other quantity) as a function of frequency. It is
often expressed in bands. Bands may be of constant percentage width, such as 1/3 or 1/12th octave bands
(~23% and ~6% of the center frequency, respectively). Bands may also be of fixed width, regardless of
center frequency (e.g. 50 Hz). Instead of bands, a spectrum may also be expressed as spectrum density,
which is equivalent to 1 Hz bands.
3.48
talker sidetone. The direction of speech transmission from mouth to ear of the telephone user.
3.49
telephone set. A device that, when connected to a telephone network, allows two-way voice
communication.
3.50
test head. A fixture containing a mouth simulator and an ear simulator located in a specified relationship
with each other. See loudness rating guard-ring position (3.28).
3.51
two-wire transmission. A transmission method, circuit, or system which provides common paths (one
pair) for signals in the send and receive directions.
3.52
zero transmission level point (0 TLP). An arbitrarily established point relative to which transmission
levels at all other points are specified.
695
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15
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4
Abbreviations, Acronyms and Symbols
696
4.1
Abbreviations and acronyms
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
AGC
ANTHD
DFTP
DRP
DRTP
DSP
DTX
ECRP
EPD
ERP
ERUP
FFT
HATS
LRGP
MRP
OLR
RETP
RLR
RTP
SETP
SDN
SFI
SLR
SNR
STMR
STP
TDS
THD
VAD
727
4.2
728
729
730
731
732
733
734
735
736
737
738
739
740
741
Th
el
e
t
t
e
r“
G”i
su
s
e
df
ors
pe
c
t
r
a
. Th
i
sc
or
r
e
s
pon
dst
oc
ommonus
a
g
e
,e
s
pe
c
i
ally in two-channel FFT analysis
literature. The analysis bandwidth shall be specified:
GDFTP(f) = rms spectrum at Diffuse Field Test Point, in dBPa
GERP(f) = rms spectrum at Ear Reference Point, in dBPa
GMRP(f) = rms spectrum at Mouth Reference Point, in dBPa
G(MRP)(ERP)(f) = cross-spectrum between MRP and ERP, in dB (Pa/Pa)
G(MRP)(SETP)(f) = cross-spectrum between MRP and SETP, in dB (V/Pa)
GRETP (f) = rms spectrum at Receive Electrical Test Point, in dBV
G(RETP)(ERP)(f) = cross-spectrum between RETP and ERP, in dB (Pa/V)
G(RETP)(SETP)(f) = cross-spectrum RETP and SETP, in dB (V/V)
GSETP(f) = rms spectrum at Send Electrical Test Point, in dBV
GSETP(S+N)(f) = rms spectrum at SETP with both the mouth simulator and noise sources active, in dBV
GSETP(N)(f) = rms spectrum at SETP with only the noise source active, in dBV
automatic gain control
amplitude normalized total harmonic distortion
diffuse field test point
drum reference point
derived recommended test position
digital signal processor
discontinuous speech transmission
ear cap reference point
echo path delay
ear reference point
estimated real use position
fast Fourier transform
head and torso simulator
loudness rating guard-ring position
mouth reference point
overall loudness rating
receive electrical test point
receive loudness rating
recommended test position
send electrical test point
signal-to-distortion-and-noise ratio
single frequency interference
send loudness rating
signal-to-noise ratio
sidetone masking rating
standard test position
time delay spectrometry
total harmonic distortion
voice activity detector
Symbols
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
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743
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745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
Th
el
e
t
t
e
r“
H”i
su
s
e
df
orf
r
e
qu
e
n
c
yr
e
s
pon
s
e
:
H(f) = frequency response, in dB
H’
(
f
)= response at the new preferred ISO R10 frequency
HEP(f) = echo path frequency response, in dB (V/V)
HLS(f) = listener sidetone frequency response, in dB (Pa/Pa)
HO(f) = overall frequency response, in dB (Pa/Pa)
HR(f) = receive frequency response, in dB (Pa/V)
HS(f) = send frequency response, in dB (V/Pa)
HSD(f) = diffuse field send frequency response, in dB (V/Pa)
HTS(f) = talker sidetone frequency response, in dB (Pa/Pa)
Th
el
e
t
t
e
r“
L”i
su
s
e
df
orr
msl
e
v
e
l
sme
a
s
u
r
e
dov
e
rawi
deba
n
d,wi
t
ht
h
eba
n
dwi
dt
ht
obes
pe
c
i
f
i
e
d. Th
i
s
corresponds to common usage in sound level measurements, as specified in ANSI S1.1-1994:
LERP = level at Ear Reference Point, in dBPa
LMID = level of stimulus to be determined for headset tests (see 9.3.2)
LMRP = level at Mouth Reference Point, in dBPa
LRETP = level at Receive Electrical Test Point, in dBV or dBm
LROOM = level of background noise in measurement environment, in dBSPL
LSETP = level at Send Electrical Test Point, in dBV or dBm
Th
el
e
t
t
e
r“
S”i
su
s
e
df
ors
pe
c
i
a
l
l
yc
a
l
c
u
l
a
t
e
ds
e
ns
i
t
i
v
i
t
i
e
s
:
SDE = translation from HATS Drum Reference Point to Ear Reference Point
Th
el
e
t
t
e
r“
T”i
su
s
e
df
ort
i
meme
a
s
u
r
e
me
n
t
s
:
T = length of time window in simulated free-field techniques, in sec
TC = convergence time of acoustic echo cancellers (AEC) algorithm, in sec
“
TCL”i
sus
e
df
orTe
r
mi
n
a
lCou
pl
i
ngLos
s
:
ATCLT = active temporally weighted terminal coupling loss, in dB
ALTCLT = active long-term temporally weighted terminal coupling loss, in dB
LTCLT = long-term temporally weighted terminal coupling loss, in dB
TCL = terminal coupling loss, in dB
TCLT = temporally weighted terminal coupling loss, in dB
TCLW = frequency weighted terminal coupling loss, in dB
Other symbols:
SendSNR = send signal-to-noise ratio, in dB
SendSNRW = weighted send signal-to-noise ratio, in dB
780
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
17
IEEE P269/D25 October 2004
780
781
782
783
784
785
786
787
788
789
5
Test Equipment and Setup
Test equipment generally required to test all the devices covered by this standard is covered in this clause. The
specific test equipment required to produce test signals and analyze the resulting output is determined by the test
signal and analysis method chosen. Test circuits, interfaces and impairments for analog and digital telephones, as
well as 4-wire devices such as handsets and headsets, are described in Clauses 7, 8, and 9 respectively.
All equipment should be calibrated in accordance with the recommendations of the manufacturer before performing
the system calibration procedures in Clause 6.
790
5.1
Ear simulators
791
792
793
794
795
The fundamental purpose of ear simulators is to test a receiver under conditions that most closely approximate actual
use by real persons. The recommendations that follow are based upon the manner in which the receivers are
intended to be used. Modifications to an ear simulator or test procedure shall not be made. For example, flexible
sealing material, such as putty, shall not be used.
796
5.1.1
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
An ear simulator with a flexible pinna shall be used for all measurements, unless the applicable performance
specification requires or allows an alternative. In this case the requirements of Annex B shall be met. The Type 3.3
ear simulator is recommended for all devices. The Type 3.4 ear simulator is recommended for all devices except
supra-concha headsets, supra-aural headsets and intra-concha headsets with acoustic outlets that do not face the ear
canal.
821
5.1.2
822
823
824
825
826
827
Type 3.3 and Type 3.4 ear simulators both simulate the acoustical and mechanical characteristics of real ears. They
are likely to give results comparable to the typical listening experience of real persons for the widest possible variety
of headsets. However, the correlation between measurements on ear simulators and on real ears is better for some
headset types than others. For headsets that are in close proximity to, or in the ear canal, the correlation is not as
good as for most other types.
Selection
The ear simulators shall comply with the specifications given in ITU-T Recommendation P.57-2002, except for
Type 3.3. Type 3.3. shall have a hardness of 35 ±3 degrees Shore-OO, as measured according to ASTM 2240. (ITUT Recommendation P.57-2002 specifies a hardness of 55 ±10 degrees Shore-OO for Type 3.3.)
Type 3.3 and Type 3.4 ear simulators both simulate the acoustical and mechanical characteristics of real ears. They
are likely to give results comparable to the typical listening experience of real persons for the widest possible variety
of handsets or headsets and applications, including non-traditional designer handsets and headsets. Both types
simulate typical leakage and how it changes with position and/or applied force. There are, however, some
differences between the two types, as well as the positioning devices available for use with them. For more
information, please see Annex A.
(Type 3.3 was formerly called the soft HATS pinna. It has a hardness of 55, ±10 degrees Shore-OO, as measured
according to ASTM 2240. It is an anatomically-shaped pinna which is structurally identical to the pinna formerly
described as Type 3.3 in ITU-T Recommendation P.57.)
The same ear simulator shall be used for all measurements on a particular device. The choice of ear simulator and
positioning method shall be clearly stated in all test reports.
Headset measurements made on ear simulators compared to real ears
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
18
IEEE P269/D25 October 2004
828
829
830
831
For insert headsets, both Type 3.3 and Type 3.4 ear simulators are likely to provide a greater seal than on many
human subjects, resulting in an overestimation of output at low frequencies. Nonetheless, both ear simulators are
recommended for this application.
832
5.1.3
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
Type 3.3 and Type 3.4 ear simulators both measure at the eardrum reference point (DRP). Measurements collected
at the DRP shall be translated to the ERP. This is done because receive and sidetone specifications are referenced to
the ERP. It also permits comparison of measurements made on the various type ear simulators.
850
5.2
851
852
853
854
855
856
857
858
859
860
861
862
863
The fundamental purpose of mouth simulators is to test a microphone under conditions that most closely
approximate actual use by real persons. The mouth simulator shall comply with the specifications given in ITU-T
Recommendation P.58, unless the applicable performance specification requires or allows an alternative. See Annex
B. This mouth is generally installed in a HATS.
864
5.3
865
866
867
The fundamental purpose of test fixtures is to test a device equipped with a handset or headset under conditions that
most closely approximate actual use by real persons.
868
5.3.1
869
870
871
872
873
874
875
876
The test fixture shall be a HATS which complies with ITU-T Recommendation P.58. When using the Type 3.3 ear
simulator, the HATS shall also comply with ITU-T Recommendation P.64, Annex E. When using the Type 3.4 ear
simulator, the HATS shall also comply with ITU-T Recommendation P.64, Annex D.
Translation from DRP to ERP
For all measurements, the translation from DRP to ERP may be fulfilled by using a filter as specified in Annex C. A
filter shall be used for measurements of peak acoustic pressure, and is recommended for measurements of distortion.
For measurements made with any kind of spectrum analysis, the translation from DRP to ERP may be performed by
using one of the tables in Annex C. Measurement examples include frequency response, noise, linearity and
distortion. Tables may also be used for frequency response measurements made with sine waves, if only the
fundamental or total response is included.
For measurements of distortion using a sine or narrowband stimulus, a translation table may be constructed based on
one of the tables in Annex C. Separate tables are required for each harmonic or difference-frequency distortion
product, taking into account the frequency offset between the stimulus frequency and the frequency of the distortion
product.
Mouth simulators
ITU-T Recommendation P.58 does not define a sound field behind the lip plane of the mouth simulator. However,
experience has shown that at least one implementation of the mouth has a sound field distribution which closely
approximates the sound field behind the lip plane of a real human head, up to at least 4 kHz. The investigated region
extends from behind the lip plane to the base of the rubber ear and equal to or greater than 5 mm above the surface
of the HATS cheek. This makes HATS suitable for testing headsets, cordless and cellular phones, handsfree phones,
and traditional corded handsets. The sound field approximation may extend in frequency range as well as to other
regions around HATS, but these have not yet been verified.
Test Fixtures
Selection
The LRGP position was specified in previous editions of this standard. Send frequency response measurements
made on ordinary telephones from 300-3400 Hz are expected to give practically identical results, whether obtained
with LRGP or the HATS position. Systematic differences of about 1-2 dB in send frequency response measurements
on pressure gradient microphones have to be expected from the upwards tilted speaking direction of about 19
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
19
IEEE P269/D25 October 2004
877
878
879
880
881
882
degrees using the LRGP position. See ITU-T Recommendation P.64 (1999), Annex F. (Similar effects might be
observed in sidetone and overall measurements.)
883
5.3.2
884
885
886
887
888
The two alternatives for handset positioning are the standard test position (STP) and the manufacturer’
s
recommended test position (RTP). STP is defined in clause 5.3.2.1 and RTP is defined in clause 5.3.2.2. STP shall
be used unless an RTP is defined by the manufacturer.
889
5.3.2.1
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
The handset receiver must be nominally placed in the HATS position as specified by Annex D or E of ITU-T
Recommendation P.64. To do this, the ear-cap reference point (ECRP) must lie on the axis of motion of the
positioning device. This axis is defined by a line that passes through the ERP of the left and right ears. The ECRP
may be inside of or outside of ERP depending on the applied force and the shape of the receiver.
For information about an alternative test fixture, and the ear simulators and mouth simulator with which it can be
used, see Annex B.
Handset positioning
Standard test position (STP)
For STP, the handset receiver must be nominally placed in the HATS position as specified by Annex D or E of ITUT Recommendation P.64. To do this, the ear-cap reference point (ECRP) must lie on the axis of motion of the
positioning device. This axis (X axis) is defined by a line that passes through the ERP of the left and right ears. The
ECRP may be inside of or outside of ERP depending on the applied force and the shape of the receiver.
For the purposes of this standard, the ECRP is the intersection of the external ear-cap reference plane with a normal
axis through the effective acoustic center of the sound outlet ports. Generally, the acoustic center of the sound outlet
ports is at the center of their distribution.
Unless otherwise specified by the manufacturer, the ECRP is the intersection of the external ear-cap reference plane
(ECRP Plane) with a normal axis (X axis as defined by ITU-T P.64, Annex E) through the effective acoustic center
of the sound outlet ports. Generally, the acoustic center of the sound outlet ports is at the center of their distribution.
For many handsets, the ECRP plane is parallel to the reference plane of the positioning device.
For many handsets, the ear-cap reference plane is parallel to the reference plane of the positioning device.
For some handsets, the above positioning may not apply, and the position that best represents intended use shall be
utilized.
The receiver shall contact the pinna with a force of 6 newtons. This is the default force for all measurements.
In general, it is desirable that receive frequency response should not change too much as application force changes.
For this reason, the device should also be tested at 2N and 10N, which represent minimum and maximum forces
likely to be used by real persons on a long-term basis. These results are for information, but do not have to be
included in the test report.
In general, it is desirable to know how the receive frequency response and loudness rating change as application
force changes. For this reason, the device shall be tested at a high leak position and a low leak position, which
represent minimumtypical and maximum forces likely to be used by real persons on a long-term basis.
All tests shall be performed at the high leak position as defined below:
1. For the Type 3.3 artificial ear the receiver shall be located at the ERP.
2. For the Type 3.4 artificial ear the receiver shall be applied with a force of 6 N.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
20
IEEE P269/D25 October 2004
930
931
932
The tests in Table ?? shall also be performed at the low leak position.
Test type
Receive
Send
Sidetone
Echo
Max Acoustic Output
933
934
935
936
937
938
939
940
941
942
Analog
7.4.1
7.5.1
7.6.1, 7.6.2
7.10
Digital
8.4.1
8.5.1
8.6.1, 8.6.2
8.8, 8.9
8.13
4-Wire Handsets
9.3.3
9.4.2
9.5
9.6
The low leak position is defined below:
1. For the Type 3.3 artificial ear the receiver shall contact the pinna with a force of 18 N
2. For the Type 3.4 artificial ear the receiver shall contact the pinna with a force of 15 N.
Maximum acoustic output (clauses 7.10, 8.13 and 9.6) shall be tested at both high leak and low leak position. The
f
i
n
a
lr
e
s
ul
ts
h
a
l
lbea
n“
u
ppe
re
n
v
e
l
ope
”c
u
r
v
ec
on
s
i
s
t
i
n
goft
h
ema
xi
mum ou
t
pu
tofe
a
c
hme
a
s
u
r
e
me
n
ta
te
a
c
h
frequency. See Figure 3Figure 3Figure 3 and Figure 4Figure 4Figure 4.
130
L ERP(f) dB
120
110
100
90
100
943
944
945
946
947
1000
10000
Frequency (H z)
Figure 1 Maximum Acoustic Output, LERP(f), 2 measurements on one handset
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
21
IEEE P269/D25 October 2004
130
L ERP(f) dB
120
110
100
90
100
1000
10000
Frequency (H z)
948
949
950
951
952
953
954
955
Figure 2 Maximum Acoustic Output, LERP(f).
Upper envelope (heavy line) and 2 individual measurements on one handset (light lines).
956
5.3.2.2
Recommended Test Position (RTP)
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
A manufacturer may specify a recommended test position (RTP) on either the Type 3.3 or Type 3.4 or both ear
simulators. The force applied shall not exceed 18N for Type 3.3, and 15N for Type 3.4.
The definition of the RTP, including evidence of its authorization by the device manufacturer, shall be included in
the test report.
RTP is defined in following steps:
1.
Find ECRP on the handset. Unless otherwise specified by the manufacturer, the ECRP is the intersection
of the external ear-cap reference plane (ECRP Plane) with a normal axis (X axis as defined by ITU-T P.64,
Annex E) through the effective acoustic center of the sound outlet ports. Generally, the acoustic center of
the sound outlet ports is at the center of their distribution. For many handsets, the ECRP plane is parallel to
the reference plane of the positioning device.
2.
Line up ECRP at Ear simulator ERP on the positioning device. The ECRP plane shall be identical to the
plane of the ERP as defined by the positioning device.
3.
Move ECRP in ECRP plane (Ear cap reference plane) relative to ERP. This can be defined as (Y, Z)
coordinates (ITU-T P.64, Annex E) in the ECRP plane. If none given, leave the ECRP centered on ERP,
equivalent to (0, 0) coordinates. The Y axis is defined along the length of the phone with positive Y being
i
nadi
r
e
c
t
i
ont
owa
r
dst
h
emi
c
r
oph
on
ef
r
omECRP(
mov
i
ngt
h
eph
on
e“
down
”ont
h
epos
i
t
i
on
e
r
)
.Th
eZ
axis intersects at ECRP, and is perpendicular to the Y axis, with positive Z being towards the right as the
ph
on
ei
sobs
e
r
v
e
df
r
omt
h
ef
r
on
t(
mov
i
ngt
h
eph
on
et
ot
h
e“
r
i
g
h
t
”onpos
i
t
i
on
e
r
)(
de
f
i
n
e
don
l
yf
orr
i
g
h
t
ear).
4.
Adjust the three angles as defined for device under test (angles A, B, C, which are rotated about three
mutually perpendicular axis originating at ERP for a given positioner type). If none given, use angles
consistent with HATS position which shall be defined by the handset positioner equipment manufacturer.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
22
IEEE P269/D25 October 2004
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
1001
1002
1003
1004
1005
1006
1007
1008
1009
1010
1011
5.
Adjust pressure or distance along ECRP axis (vector in X direction) to given force setting or X coordinate.
State the force or X coordinate used. If none given, use 6N. Regardless of force specified for RTP, the
device should also be tested at 6N.
6.
RTP can then be defined as the combination (Y, Z) coordinates, three angles and force or X coordinate.
The manufacturer of the device under test is responsible for providing this data.
All tests shall be performed at the RTP.
Maximum acoustic output (clauses 7.10, 8.13 and 9.6) shall be tested at 6 N as well as the force or X coordinate
defined by RTP. (One could experience a force of approximately 10N by placing a weight of 1 Kg on top of a
lightweight receiver placed on his or her pinna, with the ear cap reference plane horizontal.)
A manufacturer may specify a recommended test position (RTP) on either the Type 3.3 or Type 3.4 ear simulator.
The RTP may specify position with respect to ERP, a specific force, or other aspects of the test position intended to
simulate actual use. The force applied shall not exceed the range of 2-10N. If the phone is tested at the RTP, the
definition of the RTP, including evidence of its authorization by the device manufacturer, shall be included in the
test report.
If the RTP is used, the device should also be tested at 2N, 6N and 10N on the same ear simulator. These results are
for information, but do not have to be included in the test report.
For maximum acoustic output measurements, the device shall be tested at either 6N or the RTP, and also at 13N.
Th
ef
i
n
a
lr
e
s
u
l
ts
h
a
l
lbea
n“
uppe
re
n
v
e
l
ope
”c
u
r
v
ec
on
s
i
s
t
i
n
goft
h
ema
x
i
mum ou
t
pu
tofe
a
c
hme
a
s
u
r
e
me
n
ta
te
a
c
h
frequency. See Figure 3Figure 3Figure 3 and Figure 4Figure 4Figure 4.
130
L ERP(f) dB
120
110
100
90
100
1012
1013
1014
1015
1016
1000
10000
Frequency (H z)
Figure 3 Maximum Acoustic Output, LERP(f), 2 measurements on one handset
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
23
IEEE P269/D25 October 2004
130
L ERP(f) dB
120
110
100
90
100
1000
10000
Frequency (H z)
1017
1018
1019
1020
1021
1022
1023
1024
1025
1026
1027
Except for maximum acoustic output, the same positioning shall be used for all measurements of any particular
device. The positioning method shall be clearly stated in the test report.
1028
5.3.3
1029
1030
1031
1032
1033
1034
1035
1036
1037
1038
1039
1040
1041
1042
1043
1044
1045
1046
1047
1048
1049
1050
1051
1052
1053
1054
For a given headset placement, the same position shall be used to test all the electro-acoustic parameters.
Figure 4 Maximum Acoustic Output, LERP(f).
Upper envelope (heavy line) and 2 individual measurements on one handset (light lines).
Handsets with carbon microphones require conditioning procedures before positioning for measurement. See Annex
D.
Headset positioning
a)
RTP: If the manufacturer specifies a recommended test position (RTP), the headset shall be tested in that
position. The RTP shall reflect the way the headset is intended by the manufacturer to be used in a real
situation. The manufacturer should provide suitable pictures to illustrate the RTP.
b) DRTP: I
ft
h
ema
nu
f
a
c
t
u
r
e
r
’
sRTPi
sn
otpr
ov
i
de
d,bu
tt
h
e
r
ei
sar
e
c
omme
n
de
dwe
a
r
i
ngpos
i
t
i
on(
RWP)
pr
ov
i
de
di
nt
h
eus
e
r
’
sgu
i
de
,t
h
e
na
nRTPs
h
a
l
lbede
r
i
v
e
df
r
om t
h
eRWPa
n
du
s
e
d.I
ts
ha
l
lbec
a
l
l
e
dt
h
e
derived recommended test position (DRTP).
c) ERUP: I
ft
h
ema
nuf
a
c
t
u
r
e
r
’
sRTPa
n
dRWPa
r
en
otpr
o
v
i
de
d,t
h
e
nt
h
eh
e
a
ds
e
ts
h
a
l
lbet
e
s
t
e
di
nt
h
e
estimated real use position (ERUP). The test lab shall define an ERUP that closely approximates real use.
Natural headband pressure, or other positioning techniques normally used by a real person, shall be used.
The test position shall define how the receiving part of the headset is to be placed against or inside the ear simulator.
After providing receiver placement instructions, the test position shall then describe the subsequent positioning and
orientation of the microphone.
The exact positioning of the microphone shall be specified using geometric coordinates relative to centre of lips
using three axes (See ITU-T Rec. P.64):
a) The Xm axis coincides with the mouth reference axis and has positive direction into the mouth.
b) The Ym axis is horizontal, perpendicular to the Xm axis with positive direction towards the right side of the
head.
c) The Zm axis is perpendicular to the Xm axis and Ym axis with positive direction upwards.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
24
IEEE P269/D25 October 2004
1055
1056
1057
1058
1059
1060
1061
1062
1063
1064
1065
1066
1067
1068
1069
1070
1071
1072
1073
1074
1075
1076
1077
1078
1079
1080
1081
1082
1083
1084
1085
1086
1087
1088
1089
1090
1091
The closer the microphone is to the mouth, the more sensitive the results will be to any inaccuracy of the
geometrical positioning. Pressure gradient microphones (cardioid, noise canceling, etc.) are especially sensitive to
both position and orientation. The recommended orientation of the microphone towards the mouth shall be stated.
For headsets that are fixed at the ear, and have a short microphone boom configuration, then the receiver coupling
provides the main positioning element, with the boom pointed at MRP.
1092
5.3.3.15.3.4
1093
1094
1095
1096
1097
1098
1099
1100
1101
1102
1103
1104
1105
1106
1107
1108
When a headset is placed on the Type 3.3 or Type 3.4 ear simulatorHATS, the receiver test results may vary from
trial to trial due to slight variations in positioning. Relatively accurate and repeatable results can be obtained by
making several measurements and averaging the results. The procedures in this clause shall be followed for send,
receive, sidetone and overall measurements.
When positioning a headset on a HATS, it is generally possible to approximate real use in an obvious way. In any
case where the headset does not fit on the HATS and its ear simulator quite in the way intended for real persons,
adjustments may be made so the receiver and microphone are as close as possible to positions corresponding to real
use. Particular caution should be excercised when positioning receivers (such as earbuds) near or in the ear canal, as
they are subject to more variation than other receiver types. The body of the receiver, the headband or any other
non-acoustical component may be positioned as necessary.
Jigs may be used to improve repeatability of postioning provided that they do not cause any acoustic impairment to
the measurement.
The test operator should become acquainted with the specific headset by running some preliminary learning tests.
The final positioning used should be documented with pictures. Closeups of both send and receive positioning are
strongly recommended.
Headsets should be tested in a position that most closely approximates real use. Natural headband pressure, or other
positioning techniques normally used by a real person, shall be used for testing.
If the manufacturer specifies a recommended test position (RTP), the headset shall be tested in that position. The
purpose of the RTP shall be to clarify how to position the headset in a way that corresponds to real use. The RTP
shall be defined geometrically with respect to the MRP, center of the lip plane, ERP, ear entrance point (EEP) and/or
the HATS reference point (HRP). See ITU-T Recommendation P.58. Facial features, such as the corner of the
mouth, shall not be used as reference points. If no RTP is specified, the test position can be determined by observing
the actual design of the headset and by following any guidelines for positioning provided by the manufacturer.
When positioning a headset on a HATS, it is generally possible to approximate real use in an obvious way. In any
case where the headset does not fit on the HATS and its ear simulator quite in the way intended for real persons,
adjustments may be made so the receiver and microphone are as close as possible to positions corresponding to real
use. The body of the receiver, the headband or any other non-acoustical component may be positioned as necessary.
Headset receiver positioningmeasurement repeatability and averaging
A minimum of 5 measurements of frequency response and loudness rating shall be made on each individual unit
tested. The headset shall be completely removed from the ear simulator and re-mounted for each trial. The mean
and standard deviation of the loudness rating and each point of the frequency response shall be computed for this
group of measurement trials.
The accuracy of the final results shall be considered acceptable if the standard deviation of the measurements meet
the following criteria: For receive, the standard deviation of loudness rating isshall be 1 dB or less, and if the
standard deviation of the frequency response shall beis 2 dB or less from 200 to 14000 Hz, and 1 dB or less from
1000 through 4000 Hz. For send, the standard deviation of loudness rating shall be 1 dB or less, and the standard
deviation of the frequency response shall be 1 dB or less from 200 to 4000 Hz. If the results of the first 5 trials do
not meet this criteriathese criteria, report the results, but label them as reduced accuracy.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
25
IEEE P269/D25 October 2004
1109
1110
1111
1112
1113
1114
1115
1116
1117
1118
1119
1120
1121
1122
1123
1124
1125
1126
1127
1128
For sidetone, the standard deviation will likely be wider than either send or receive. The standard deviation of the
loudness ratings and frequency response measurements shall be reported.
The reported results shall include the mean loudness rating and standard deviation, the mean frequency response and
standard deviation, a description of the test position, and the number of trials. Additional measurements may be
made in order to meet the mean and standard deviation criteria.
Similarly, a minimum of 5 measurements of distortion shall be made, and the mean shall be reported. A minimum of
5 measurements of all other parameters should also be made, and the mean reported. The mean results shall be
reported.
For maximum acoustic output measurements made using the Type 3.3 or Type 3.4 ear simulator, at least 5
me
a
s
u
r
e
me
n
t
ss
h
a
l
lbema
de
.Th
ef
i
n
a
lr
e
s
u
l
ts
h
a
l
lbea
n“
uppe
re
n
v
e
l
ope
”c
u
r
v
ec
on
s
i
s
t
i
n
goft
h
ema
x
i
mum output
of each measurement at each frequency. All curves shall be reported, for a total of at least 5 individual
measurements plus the upper envelope curve. See the example in Figure 5Figure 5Figure 5 and Figure 6Figure
6Figure 6.
130
L ERP(f) dB
120
110
100
90
100
1129
1130
1131
1000
10000
Frequency (H z)
Figure 5 Maximum Acoustic Output, LERP(f), 5 measurements on one receiver
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26
IEEE P269/D25 October 2004
130
L ERP(f) dB
120
110
100
90
100
1132
1133
1134
1135
1136
1137
1138
1139
1000
10000
Frequency (Hz)
Figure 6 Maximum Acoustic Output, LERP(f).
Upper envelope (heavy line) and 5 individual measurements on one receiver (light lines).
For headsets with large hard-cap receivers which are similar to receivers in handsets, Clause 5.3.2 may apply.
1140
5.3.3.2Headset microphone positioning
1141
1142
1143
1144
1145
1146
1147
1148
1149
1150
1151
1152
1153
1154
1155
1156
Three test positions are defined for the location of the headset microphone sound port, in order of preference.
1157
5.4
1158
1159
1160
1161
The sizes and types of measurement microphones are specified in the clauses where their use is required. All
microphones used to implement this standard shall comply with the relevant specifications in ANSI S1.12-1967
(Reaff. 1997).
1162
5.5
1163
1164
Electroacoustic measurements should be conducted in a test environment that will not affect the results beyond the
intended influence of the test fixture and measurement transducers. The test environment should have a low
a)Recommended test position (RTP).
b)Boom microphone position (BMP). If an RTP is not provided, the BMP may be used if it corresponds to
the intended usage of the microphone. The BMP is defined in clause 3.6 and in ITU-T
Recommendation P.58.
c)Estimated real use position.
These positions may fall behind the lip plane of the mouth. Please see clause B.2 for more information.
Pressure gradient microphones (cardioid, noise canceling etc.) are sensitive to both position and orientation. It is
important that the correct orientation be used to obtain results representing actual use performance.
For microphones that are not measured at RTP or BMP, or are not on a fixed boom, state the exact geometric test
position following the guidelines in clause 5.3.3.
Measurement microphones
Test environment
Copyright © 2004 IEEE. All rights reserved.
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27
IEEE P269/D25 October 2004
1165
1166
1167
1168
1169
1170
background noise level, and the test fixtures and device under test should be isolated from reflections and
mechanical disturbances that could cause significant error.
1171
5.5.1
1172
1173
1174
1175
1176
1177
1178
1179
The background noise level in the test environment shall not exceed the limits shown in Table 1Table 1Table 1. The
overall level shall not exceed 29 dBA. However, these limits may be relaxed if it can be shown that the accuracy of
the measurement is not impaired.
Be sure to record the test environmental conditions of temperature, humidity, and barometric pressure, in addition to
the background noise. Overall A-weighted noise level and octave band sound levels are defined below.
Background noise level
Background noise measurements shall be made using a 12.5mm pressure microphone with a microphone system
noise level not exceeding 20 dBA. The individual factory-calibrated frequency response of the microphone, if
available, shall be taken into account.
Octave Band
Center Frequency (Hz)
63
125
250
500
1000
2000
4000
8000
1180
1181
1182
Octave Band
Level
(dBSPL)
49
34
29
29
29
29
29
29
Table 1 Test room noise levels
1183
5.5.2
Reflection-free conditions
1184
1185
1186
1187
The test environment should be sufficiently free of reflections. There should be no large objects within 1m of the
MRP. Small objects such as tripods that are used for positioning may be acceptable. Errors due to the influence of
reflections shall not exceed ± 1.5 dB below 800 Hz. Errors above 800 Hz shall not exceed ± 1.0 dB.
1188
5.5.3
1189
1190
1191
1192
1193
1194
1195
1196
1197
1198
1199
1200
1201
1202
1203
1204
A uniform diffuse sound field shall exist in a volume of radius 0.15 m. The diffuse field test point (DFTP) is at the
center of this spherical volume. There shall be no obstacles, including the loudspeakers, within 0.5m of the DFTP.
Diffuse field conditions
The classical method of creating a diffuse field is to construct a reverberation chamber. If one is available, it is
generally the best method. (Construction and verification of a reverberation room is outside the scope of this
standard.)
For the purposes of this standard, a diffuse field may be approximated by using several loudspeakers and
uncorrelated noise sources. Experience has shown that 4-8 speakers and uncorrelated sources in an ordinary room
may be sufficient for measurements in 1/3 octaves. However, more may be required, especially if measurements are
to be made in 1/12 octave resolution.
Diffuse field measurement should be made using a 6.25 mm pressure microphone, but may also be made using a
12.5 mm random pressure microphone. The individual factory-calibrated frequency response of the microphone, if
available, shall be taken into account.
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28
IEEE P269/D25 October 2004
1205
1206
1207
1208
Diffuse field conditions shall be verified by the following two tests, performed in the same resolution and bandwidth
used for measurements. The test is performed with reference to the DFTP, with no mouth simulator or other objects
present.
1209
5.5.3.1
1210
1211
1212
1213
1214
1215
1216
1217
1218
Diffuse field conditions at the DFTP shall be verified by measurements with a cardioid or bi-directional microphone
with a free field rejection of at least 10 dB front-to-back (cardioid) or front-to-side (bi-directional) in each frequency
band. The sound field shall be considered to approximate a diffuse (random-incidence) field if the variation is
within the tolerance in Table 2Table 2Table 2. The microphone is rotated about the DFTP through 360 degrees in
each of three perpendicular planes. Measurement must be made in 15 degree increments or less. Since microphone
rejection may not be the same in all bands, the tolerance for each band shall be determined by the microphone
rejection in that particular band.
Test for diffuse field
Microphone rejection
at least
25 dB or greater
20 dB
15 dB
10 dB
Less than 10 dB
1219
1220
1221
1222
Allowable variation
over 360 degrees
6 dB
5 dB
4 dB
3 dB
Microphone not suitable
Table 2 Diffuse field variation allowable vs microphone rejection
1223
5.5.3.2
Test for spectrum uniformity
1224
1225
1226
1227
1228
1229
GDFTP (f) (the spectrum at DFTP) shall be measured at the DFTP and 6 additional points +/- 15cm from DFTP on
each of three mutually perpendicular axes passing through the DFTP. The spectrum at these 6 points shall not vary
from GDFTP (f) by more than +/- 3 dB in each band.
1230
5.6
1231
1232
1233
1234
1235
In order to test a telephone, handset or headset realistically, it may be useful to test it in environments similar to
those in which it is expected to operate. Such environments can be considered acoustic impairments, which may
cause the telephone to work differently than in a quiet test space. Two such impairments are described in this
clause, but others may also be relevant for some applications.
1236
5.6.1
1237
1238
1239
1240
1241
1242
1243
The reference corner is one physical setup used for echo, howling and stability tests. The reference corner consists
of three perpendicular plane, smooth, hard surfaces 0.5 m square, as shown in Figure 7Figure 7Figure 7. A handset
shall be placed along the diagonal from the apex of the reference corner to the outside corner, with the earcap end of
the handset 250 mm from the apex. A headset is placed on the surface as if it was put down briefly by a user, with
the receiver 250 mm from the apex.
Calibration of the diffuse field shall be according to 6.7.1 and 6.7.2, except performed at the DFTP.
Acoustic impairments
Reference corner
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29
IEEE P269/D25 October 2004
250 mm
500 mm
1244
1245
1246
1247
Figure 7 –Reference corner for echo, howling and stability tests
1248
5.6.2
Hoth room noise
1249
1250
Hoth noise is random acoustic noise which has a spectrum designed to simulate typical ambient room noise. See
Annex E for details. The noise level shall be specified in dBA.
1251
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30
IEEE P269/D25 October 2004
1251
6
Calibration
1252
6.1
1253
1254
1255
1256
All equipment shall be calibrated according to the manuf
a
c
t
u
r
e
r
’
sr
e
c
omme
n
da
t
i
on
sa
ndi
na
c
c
or
da
n
c
ewi
t
hg
ood
laboratory practice. Be sure that all equipment has been powered up long enough to be stable before calibration,
typically 30 minutes.
1257
6.2
1258
1259
Analyzers and level meters shall be calibrated to an accuracy of at least 0.5 dB.
1260
6.3
1261
1262
1263
1264
1265
The sensitivity of measurement microphones shall be calibrated prior to each use, to an accuracy of at least 0.5 dB.
An acoustical calibrator with an accuracy of at least 0.2 dB shall be used.
1266
6.4
1267
1268
1269
1270
1271
The sensitivity of the ear simulator should be calibrated each day the system is in use. For best accuracy, it should
be calibrated prior to measurements on each new device. An acoustical calibrator with an accuracy of at least 0.2
dB shall be used, along with the factory-supplied adapters for the ear simulator to be calibrated. Be sure to apply
any correction factor required for any particular combination of calibrator, adapter and ear simulator.
1272
6.5
1273
1274
1275
1276
1277
1278
1279
1280
1281
The calibration procedures shall be performed using the same format as will be used for measurements. Format
examples are 1/N octave bandwidth analysis, constant bandwidth analysis and R-series preferred frequencies.
Bandwidth shall be the same as or greater than that which will be used for measurements. Resolution shall be the
same as or finer than that which will be used for measurements. Amplitude accuracy shall be the same as or better
than that which will be used for measurements. The actual format, bandwidth, resolution and amplitude accuracy
shall be stated. See Annex F for additional details. Also, review clauses F.9, F.11 and G.6 and G.7 for a summary of
the test signal parameters, comparison of test methods and signals, and details about measurement bandwidth and
resolution.
1282
6.6
1283
1284
1285
1286
1287
1288
1289
1290
1291
1292
Electrical test signals should be calibrated each day the system is in use. For best accuracy, they should be
calibrated prior to measurements of each new device. The analyzer or level meters used shall be calibrated first
(6.2).
General
Electrical measurement instruments
Measurement microphones
The individual factory-calibrated frequency response, if available, shall be taken into account.
Ear simulator
Measurement bandwidth and resolution
Electrical test signals
Electrical test signals shall be applied from a 900 ohm resistive source impedance for most analog telephones, and a
600 ohm resistive source impedance for digital telephones. These calibrations are performed across matched
calibrated resistive loads. As a result, receive and echo test signals are specified under nominally loaded conditions.
This is equivalent to one-half the open-circuit voltage. Following a calibration, the resistive load is removed and the
source is connected to RETP without further adjustment.
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31
IEEE P269/D25 October 2004
1293
1294
1295
A similar calibration is required for testing handset and headset 4-wire devices. See clause 9.2 for the source
impedance requirements.
1296
6.6.1
1297
1298
1299
1300
1301
1302
1303
1304
1305
The electrical test spectrum is measured across a calibrated resistive load. For sinusoidal test signals, the spectrum
shall be flat within ± 0.5 dB over the actual measurement bandwidth. Equalization may be used to meet this
requirement.
1306
6.6.2
1307
1308
1309
1310
1311
1312
1313
1314
1315
1316
1317
1318
1319
1320
1321
1322
1323
The standard electrical test level, nominal LRETP, is -16 dBV rms, 0.5 dB, for analog telephones. This test level is
recommended for measurements at minimum and reference volume control settings, and -30 dBV is recommended
at maximum volume. Total harmonic distortion shall be less than 1% for these test conditions.
1324
6.7
1325
1326
1327
1328
The mouth simulator should be calibrated each day the system is in use. For best accuracy, it should be calibrated
prior to measurements on each new device. The measurement microphone used to calibrate the mouth simulator
shall be calibrated first (6.3).
1329
6.7.1
1330
1331
1332
1333
1334
1335
1336
1337
1338
The acoustic test spectrum is measured at the Mouth Reference Point (MRP). For sinusoidal test signals, the
spectrum shall be flat within 0.5 dB over the actual measurement bandwidth. Equalization may be used to meet
this requirement.
Electrical test spectrum
For all other test signals, the electrical spectrum shall meet the target spectrum and spectrum tolerance for the type
of signal used, as defined in Annex F. If no tolerance is specified in the signal definition, the default tolerance is ± 3
dB from 175 - 4500 Hz (or the 1/3 octave bands from 200 - 4000 Hz), and +3/-5 dB elsewhere. Equalization may be
used to meet this requirement.
Electrical test level
The standard test level for digital telephones, nominal LRETP, is –18.2 dBV, 0.5 dB, for a 600 ohm interface. This
corresponds to –
16 dBm0. This test level is recommended for measurements at minimum and reference volume
control settings. For measurements at maximum volume control settings, LRETP is -32.2 dBV. This corresponds to –
30 dBm0. Total harmonic distortion of the test signal shall be less than 1% for these test conditions.
The standard test level for handsets and headsets tested as 4-wire devices is determined by the procedure for setting
the default receive volume control adjustment in clause 9.3.2.
For sinusoidal test signals, the level shall be held constant at all test frequencies.
For continuous spectrum test signals, the level shall be measured over the entire spectrum. Out-of-band signals from
40 to 20,000 Hz shall add no more than 0.5 dB to this level.
Mouth simulator
Acoustic test spectrum
For all other test signals, the acoustic spectrum shall meet the target spectrum and spectrum tolerance for the type of
signal used, as defined in Annex F. If no tolerance is specified in the signal definition, the default tolerance is 3 dB
from 175 - 4500 Hz (or the 1/3 octave bands from 200 - 4000 Hz), and +3/-5 dB elsewhere. Equalization may be
used to meet this requirement.
Copyright © 2004 IEEE. All rights reserved.
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32
IEEE P269/D25 October 2004
1339
6.7.2
Acoustic test level
1340
1341
1342
1343
1344
1345
1346
1347
The standard acoustic test level for send, LMRP, is -4.7 dBPa rms, 0.5 dB, at the MRP. Total harmonic distortion of
the mouth simulator shall be less than 2% for this test condition.
1348
6.7.3
1349
1350
1351
1352
A 6.25mm pressure or free-field microphone shall be used to calibrate the HATS mouth simulator. The microphone
axis shall be oriented 90 degrees to the mouth axis with the center of the protection grid at the MRP (see Figure
8Figure 8Figure 8). (The HATS manufacturer generally supplies a jig for this purpose.)
For sinusoidal test signals, the level at MRP shall be held constant at all test frequencies.
For continuous spectrum test signals, the level shall be measured over the entire spectrum. Out-of-band signals from
40 to 20,000 Hz shall add no more than 0.5 dB to this level.
Mouth simulator calibration procedure
Equivalent
Lip-Plane
Mouth Axis
Microphone
25mm
HATS
1353
1354
1355
1356
1357
1358
1359
1360
1361
1362
1363
1364
1365
1366
1367
1368
1369
1370
1371
1372
Figure 8 HATS Mouth Calibration
If a pressure microphone is used, the results may be used directly. The individual factory-calibrated frequency
response of the microphone, if available, shall be taken into account.
If a free-field microphone is used, the free-field correction curve for 90 degrees shall be taken into account, and the
individual factory-calibrated frequency response of the microphone, if available, shall be taken into account.
To calibrate the mouth, measure GMRP(f), the spectrum at the MRP. Adjust the mouth equalization to meet the target
spectrum for the signal being used at a total sound pressure of -4.7 dBPa. This spectrum is used to calculate the
send, sidetone and overall frequency responses.
NOTE - In principle, a very small ideal microphone should be used to calibrate a mouth simulator, so that the
physical size of the microphone does not influence the measurement. In practice, a 6.25mm laboratory measurement
microphone with flat frequency response in a pressure field may be used to calibrate a HATS mouth simulator to the
required accuracy. Some manufacturers recommend a free-field microphone instead, which typically has less
sensitivity to mechanical vibration and results in a better calibration. The free-field microphone can be compensated
to give the same frequency response as a pressure microphone by using free-field correction curves for the angle of
sound incidence. The compensation is on the order of 1 dB at 8 kHz.
1373
Copyright © 2004 IEEE. All rights reserved.
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33
IEEE P269/D25 October 2004
7
Test Procedure for Analog Sets
1375
7.1
General
1376
1377
1378
1379
1380
1381
1382
1383
1384
1385
1386
1387
1388
1389
1390
1391
1392
1393
1394
Procedures are given in the following clauses for measurement of receive, send, sidetone, and overall performance
characteristics of handset and headset telephones. Parameters include frequency response, noise, input-output
linearity, distortion, and mute.,. In addition, procedures are given for measuring telephone set impedance, howling,
and maximum acoustic output.
1395
7.1.1
1396
1397
1398
1399
1400
1401
1402
1403
1404
1405
1406
1407
1408
1409
1410
1411
1412
In general, multiple test signals and stimulus levels should be used to ensure the telephone is characterized in
realistic, stable, and well-defined states. This is especially the case for telephones with non-linear processes such as
compression or voice activated switching (VOX) circuitry, etc. See Annex F and Annex G for further information
on test signals and analysis methods.
1413
7.1.2
1414
1415
1416
1417
1418
1419
1420
The measurement shall be performed using the same format as was used for calibration. Format examples are 1/N
octave bandwidth analysis, constant bandwidth analysis and R-series preferred frequencies. Measurement
bandwidth shall be the same as or less than that which was used for calibration. Measurement resolution shall be the
same as or coarser than that which was used for calibration. The actual bandwidth used shall be stated.
1373
1374
The telephone should be connected to the test circuit(s) described in clause 7.2. Other test circuits may be used for
specific applications. Because telephone set characteristics are affected by loop impedances, terminations, loop
currents, and operating levels, the measurements should be made using test loops and other conditions representative
of those conditions the telephone is expected to encounter in use. Records should be kept of the measurement
conditions.
The measured frequency responses shall be presented as decibels relative to one pascal per volt [dB (Pa/V)] for
receive, decibels relative to one volt per pascal [dB (V/Pa)] for send, decibels relative to one pascal per pascal [dB
(Pa/Pa)] for sidetone and overall, and decibels relative to one volt per volt [dB (V/V)] for echo. The stimulus level
and signal type shall be reported for each test.
The calibration procedures described in clause 6 shall be carried out before making any measurements. The
acoustical test environment shall meet the specifications given in clause 5.5.
Choice of test signals and levels
The standard test signal for all telephones consists of artificial voices defined in ITU-T Recommendation P.50. See
(F.6.1.1) for details.
Sinusoidal test signals (F.4.1) may be used for testing telephones, handsets or headsets if it can be shown that they
do not have adaptive, nonlinear or dynamic signal processing (e.g. compressors, AGC, voice activity detection,
adaptive echo cancellers, etc.). Such evidence must be given in the test report if sinusoidal test signals are used.
Other test signals may be used when it can be shown that they produce results consistent with actual use. They also
may be necessary for some specific purposes as discussed in relevant places within this standard.
The measurements in this clause shall be performed at the standard test levels specified in clauses 6.7.2 and 6.6.2.
Measurement bandwidth and resolution
In general, the test signals and analysis methods in this standard cover a frequency range from approximately 100 to
8500 Hz. The exact range depends on the analysis method, and the test signal (see G.6 and G.7)
Copyright © 2004 IEEE. All rights reserved.
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34
IEEE P269/D25 October 2004
1421
7.1.3
Choice of ear and mouth simulators and test position
1422
1423
1424
1425
Choose the ear simulator, mouth simulator and test position according to clauses 5.1, 5.2 and 5.3. This equipment
shall be used for all tests described in clause 7, unless otherwise specified. The ear simulator, mouth simulator, and
test position used shall be stated.
1426
7.1.4
1427
1428
1429
1430
1431
1432
1433
1434
If the telephone is equipped wi
t
hat
on
ec
on
t
r
ol
,t
h
et
on
ec
on
t
r
ols
h
a
l
lbes
e
tt
ot
h
ema
n
uf
a
c
t
u
r
e
r
’
sde
f
a
u
l
ts
e
t
t
i
ng
.
This is the default tone control adjustment that shall be used for all measurements.
1435
7.1.5
1436
1437
1438
All measurements shall be done at the reference receive volume control setting (3.38). A range of volume control
settings may also be used where appropriate, such as minimum and maximum volume.
1439
7.1.6
1440
1441
1442
All measurements shall be done at the reference send volume control setting (3.39). A range of volume control
settings may be used where appropriate, such as minimum and maximum volume.
1443
7.2
1444
1445
1446
1447
1448
1449
1450
1451
1452
1453
1454
1455
1456
1457
1458
A general-purpose DC feed circuit is shown in Figure 9Figure 9Figure 9. Since the parameters of the feed circuit
affect transmission performance, they should be recorded as part of the test setup. If available, parameters should
be obtained from the applicable performance specification. If not, the following values should be used:
Tone control setting
If no default setting is defined by the manufacturer, the tone control shall be set so that the frequency response is as
close as possible to the center of the required frequency response template. The tone control shall be set before
setting the volume control. If the tone and volume controls interact, an iterative process for setting these controls
may be necessary.
Reference receive volume control setting
Reference send gain control setting
Analog DC Feed circuits
C 50 microfarads
L 5 henries (each)
R = 400 ohms, including resistance of inductors
V = 50 volts
A = ammeter used to measure current drawn by the telephone under test. Alternatively, the current can be
fixed by a current source, regardless of the R value.
In some cases, ground loops may occur when connecting test equipment to RETP or SETP. The insertion of a high
quality 1:1 audio transformer can usually prevent this. If used, this transformer shall be included during calibration
and when determining the loss of the feed circuit.
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35
IEEE P269/D25 October 2004
receive electrical test point
(RETP) from 900 Ohm source
or
C
C
send electrical test point
(SETP) to 900 Ohm load
L
L
analog
telephone
R
-
+
A
V
1459
1460
1461
1462
1463
1464
1465
1466
1467
1468
1469
1470
1471
1472
1473
1474
1475
1476
1477
1478
1479
1480
Figure 9 - General purpose DC feed circuit for 2-wire analog telephone
The send electrical test point (SETP) is for measuring send output signals. It shall be connected to a 900 ohm load.
The receive electrical test point (RETP) is for applying receive input signals. It shall be connected to a 900 ohm
source. (Other terminations may be substituted as defined by applicable performance specifications.)
The loss of the feed circuit used should be measured. The loss should not be greater than 0.1 dB over the range of
100 Hz to 8,500 Hz. The loss from 20 Hz to 100 Hz should not exceed 1 dB. The circuit of Figure 9Figure 9Figure
9, using ideal components, should just meet this specification.
The following procedure may be used to determine the loss of the feed circuit:
a)
Connect a signal generator or similar device with a 900 ohm source impedance directly to a 900 ohm
resistive termination and measure the voltage across the termination.
b) Insert the feed circuit between the generator and the 900 ohm resistive termination and again measure the
voltage across the termination.
c)
The loss of the feed circuit in decibels is:
Voltage across 900 ohm resistor
Voltage across feed bridge
1481
Feed circuit loss  20 log
1482
1483
1484
1485
1486
This procedure should be followed for every value of direct current and frequency of interest. In practice, it is
usually sufficient to measure a few conditions covering the range of values likely to be encountered and, if the effect
of the feed circuit is relatively small and constant for the measured conditions, assume the characterization is
sufficient.
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36
IEEE P269/D25 October 2004
1487
1488
1489
1490
1491
1492
1493
The noise level of the feeding bridge should be low enough not to influence measurement results.
Feed circuit for overall measurements using two phones is shown in Figure 10Figure 10Figure 10.
C
C
L
Near-end
Analog
telephone
C
Term ination
&
Interconnection
L
RN
C
L
L
RF
AN
V
1494
1495
1496
1497
Far-end
Analog
telephone
AF
V
Figure 10 - General purpose DC feed circuit for 2 analog telephones for overall measurements.
1498
7.3
Analog telephone network impairments
1499
1500
1501
1502
1503
1504
Telephone performance can be influenced by various conditions in the network to which a telephone is connected.
The specific impairments described in clauses 7.3.1 through 7.3.6 should be investigated where applicable. Other
impairments, such as ADSL signals from a high-speed modem, may be relevant for specific situations. The general
method is to make a standard measurement as specified in Clauses 7.4 through 7.7, but with the impairment
introduced.
1505
7.3.1
1506
1507
1508
Loop current may be varied to determine if there are any detrimental effects. This is especially important if the
telephone is powered from the line rather than from a local power supply.
1509
7.3.2
1510
1511
Network noise can affect non-linear processes within an analog telephone. Network noise shall be approximated
using white noise, with levels measured in dBmp. Noise shall be inserted at the RETP.
1512
7.3.3
1513
1514
1515
1516
1517
1518
1519
Network termination impedance will affect analog telephone transmission performance. A 900 ohm termination is
recommended for telephones connected to a central office network. A 600 ohm termination is recommended for
telephones connected to a PBX network.
Loop current
Network noise
Termination impedance
Other terminations may be used for specific applications. For example, a complex termination more typical of North
American loops, may be useful for sidetone measurements.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
37
IEEE P269/D25 October 2004
1520
7.3.4
Test loops
1521
1522
1523
1524
1525
1526
1527
1528
A wireline analog telephone should be tested with various lengths of cable or simulated cable. Recommended loop
lengths for testing North American telephones are 0, 2.7, and 4.6 km (0, 9, and 15 kft) of 26 AWG non-loaded cable
The recommended loop simulator circuit is shown in Figure 11Figure 11Figure 11 and components for various
lengths are shown in Table 3Table 3Table 3.
For some measurements, particularly sidetone and howling, real cable may give results more representative of actual
performance compared to the loop simulator of Figure 11Figure 11Figure 11 and Table 3Table 3Table 3.
R1
L1
C2
R2
C1
R4
1529
1530
1531
1532
1533
1534
1535
1536
1537
1538
1539
1540
R3
C4
C3
L2
Figure 11 Loop simulator circuit
Component
0.305 kma
0.914 kmb
1.83 kmc
R1, R4
41.7 
124 
249 
R2, R3
109 
174 
312 
C1, C4
3.77 nF
0.0113 F
0.0226 F
C2, C3
4.02 nF
0.0122 F
0.0255 F
L1, L2
0.336 mH
0.983 mH
96.0 H
Notes:
(1) All values are 1%
(2) 2.7 km and 4.6 km can be made up of cascaded sections of the aboved
Table 3 Component values for 26 AWG cable
a
0.305 km = 1 kft
0.914 km = 3 kft
c
1.83 km = 6 kft
d
2.7 km = 9 kft. 4.6 km = 15 kft
b
1541
7.3.5
Parallel sets
1542
1543
1544
1545
1546
1547
1548
1549
1550
The telephone should be tested with a parallel telephone set simulator with suitable DC and AC characteristics. In
general, measurements made with the parallel set simulator should be compared to the same measurements made
without the simulator. The minimum recommended measurements are send and receive frequency response,
loudness ratings, and distortion, each measured with standard loop lengths.
The parallel telephone set simulator circuit shall have the VI curve as shown in Figure 12Figure 12Figure 12, 0.3
V over the current range of 0 to 100 mA. The return loss shall be greater than 10 dB with respect to 600 ohms from
200 to 4000 Hz. Component values may be adjusted to meet these tolerances. One possible implementation of this
is shown in Figure 13Figure 13Figure 13.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
38
IEEE P269/D25 October 2004
1551
1552
Test Circuit Voltage (Volts)
15
(106.25, 10)
10
(50, 7)
5
(20, 4)
0
0
20
40
60
80
100
Test Circuit Current (mA)
1553
1554
1555
1556
1557
Figure 12 Parallel set test circuit VI curve
T(+)
L1
1H
600
R1
1W
47 
100
1559
1560
20
F
20 
F
NOTES:
1. Circuit is polarity sensitive.
2 Use IN4004, or similar, for diode strings.
3. L1 may be > 1 H
4. Resistor values shall be ± 1 %.
5. R1 + L1 coil resistance should be 53 ohms.
6. The performance of this circuit is dependent
on the characteristics of the diodes used and
temperature. Performance of the circuit
must be verified and component values may
need to be adjusted to meet the
requirements.
R(–
)
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
39
IEEE P269/D25 October 2004
T (+)
R1, 1 W
2
4
1
6
3
8
47 
600 
100 
5
R (–)
7
T1
1561
1562
1563
1564
1565
Figure 13 Parallel telephone set simulator
1566
7.3.6
Cordless range
1567
1568
1569
A cordless telephone should be tested across the range of expected usage. This should include the minimum and
maximum specified distance the telephone is expected to operate between the base unit and mobile unit.
1570
7.4
Receive
1571
7.4.1
Receive frequency response
1572
1573
1574
1575
1576
1577
1578
Receive frequency response is the ratio of sound pressure measured in the ear simulator, referred to the Ear
Reference Point (ERP), to the voltage input at the Receive Electrical Test Point (RETP), which is expressed in
decibels. The receive frequency response in dB, HR(f), is given by Equation 7.1Equation 7.1Equation 7.1. HR(f)
may be used to calculate the receive loudness rating (RLR) according to ITU-T Recommendation P.79-1999. Please
see Annex H.
H R ( f )  20 log
1579
1580
1581
1582
1583
1584
1585
1586
1587
1588
1589
G ERP ( f )
G RETP ( f )
in dBPa / V
Equation 7.11
where:
GERP(f) is the rms spectrum at ERP
GRETP(f) is the rms spectrum at RETP
In some cases, frequency response calculation may be performed with cross-spectrum or related techniques.
Justification for such techniques shall be given in the test report. See clause G.1 for more information.
1590
7.4.2
Receive noise
1591
1592
1593
1594
1595
1596
Receive noise is internally generated audio frequency noise present at the receiver when no stimulus is applied. The
receiver shall be coupled to the ear simulatorwi
t
ht
h
eRETPt
e
r
mi
n
a
t
e
da
n
dwi
t
hn
os
i
g
n
a
li
n
pu
t
.Th
et
e
l
e
ph
on
e
’
s
microphone should be isolated from sound input and mechanical disturbances that would cause significant error.
Measure the acoustic output signal, referred to the ERP, from 25100 to 8,500 Hz, averaging over a minimum period
of5s
e
c
on
ds
.Re
c
e
i
v
en
oi
s
es
h
ou
l
dbeme
a
s
u
r
e
dwi
t
ht
h
es
e
n
dmu
t
ef
e
a
t
u
r
ebot
h“
on
”a
n
d“
of
f
.
”
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
40
IEEE P269/D25 October 2004
1597
1598
1599
1600
The overall receive noise level is measured with A-weighting in dBA. The measurement may be implemented
directly using an A-weighting filter, or by using single-channel FFT with Hann windowing or real-time spectrum
analysis, followed by an A-weighted power summation.
1601
7.4.3
1602
1603
1604
1605
1606
1607
1608
1609
1610
1611
Receive narrow-band noise, including single frequency interference (SFI), is an impairment that can be perceived as
a tone relative to the overall weighted noise level. This test measures the weighted noise level characteristics in
narrow bands of not more than 31 Hz maximum from 25100 to 8,500 Hz. These levels can then be compared to the
receive noise (7.4.2).
1612
7.4.4
1613
1614
1615
1616
1617
1618
1619
1620
1621
1622
1623
1624
1625
1626
1627
Receive linearity is a measure of how the frequency response changes with input level.
1628
7.4.5
1629
1630
1631
1632
1633
1634
1635
1636
1637
1638
1639
The preferred distortion measurement method is receive signal-to-distortion-and-noise ratio (SDN), measured using
narrow-band pseudo-random noise as the stimulus. See A.1.1J.3 for details of the method.
1640
7.4.6
1641
1642
1643
1644
1645
Re
c
e
i
v
emut
ei
ss
ome
t
i
me
sc
a
l
l
e
d“
DTMFmu
t
e
”or“
a
u
t
odi
a
lmu
t
e
”
.Re
c
e
i
v
emu
t
i
ngi
sus
u
a
l
ly automatic, but may
be manually controlled, and would normally be activated by touch-t
on
edi
a
l
i
ng
,l
i
n
e“
h
ol
d”ope
r
a
t
i
on
,a
c
t
i
v
a
t
i
n
gt
h
e
hold button, or other means. Mute leakage is the amount of signal measured at the ERP when an electrical stimulus
is applied to the RETP.
Receive narrow-band noise
The receiver shall be coupled to the ear simulator with the RETP terminated and with no signal input. Measure the
A-weighted receive noise level, referred to the ERP, using a selective voltmeter or spectrum analyzer with an
effective bandwidth of not more than 31 Hz, over the frequency range of 25100 to 8,500 Hz, averaging over a
minimum period of 5 seconds.I
fFFTa
n
a
l
y
s
i
si
su
s
e
d,t
h
e
na“
Fl
a
tTop”wi
n
dowi
n
gs
h
a
l
lbee
mpl
oy
e
d.
Receive linearity
The test consists of measuring the receive frequency response as specified in Clause 7.4.1 and applying the
procedures described in Annex I. Linearity shall be measured using the same test method and stimulus type used to
measure frequency response.
If artificial voices or another wideband stimulus are used, the test shall be performed at 7 levels, from –46 to –16
dBV, in 5 dB intervals, measured in 1/3 octave bands. Smaller intervals and/or a wider range of levels may also be
used. The reference stimulus level is –16 dBV. These levels take into account the high crest factor of artificial
voices, which approaches 23 dB.
If sine wave signals are used, they shall be applied at the R10 frequencies from 200 through 5000 Hz, at 7 levels,
from –36 to –5 dBV, in 5 dB intervals. Smaller intervals and/or a wider range of levels may also be used. The
reference stimulus level is –21 dBV.
Receive distortion
Receive distortion is measured at ERP using the standard input level of –16.0 dBV. Other input levels should be
tested covering a range from –30 to 0 dBV. Measurements should also be made over a range of frequencies within
the telephone band, such as the ISO R10 preferred frequencies. For higher input levels above 0 dBV, verify that
distortion of the test system is less than 1% THD.
For information about THD and other distortion measurement methods and test signals, and the conditions under
which they may be used, see Annex J. Different distortion measurement methods are likely to give different results.
Receive mute leakage
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
41
IEEE P269/D25 October 2004
1646
1647
1648
1649
1650
1651
1652
1653
1654
1655
1656
1657
To measure mute leakage, engage the mute, apply the test signal, and measure the receive noise according to clause
7.4.2. The test signal shall be the same as that used for receive frequency response (7.4.1), at 0 dBV. An additional
measurement shall be made using the DTMF tones of the telephone set being tested. In the case of the DTMF tones
oft
h
et
e
l
e
ph
on
es
e
t
,t
h
e
r
ewon
’
tbea
nyc
ont
r
olov
e
rt
h
el
e
ve
l
.Ea
c
hr
e
s
u
l
ti
se
x
pressed in dBA, the weighted noise
level which should be compared to muted receive noise measured according to 7.4.2 (with no stimulus applied).
1658
7.5
Send
1659
7.5.1
Send frequency response
1660
1661
1662
1663
1664
Send frequency response is the ratio of voltage output at the Send Electrical Test Point (SETP) to the sound pressure
at the Mouth Reference Point (MRP), which is expressed in decibels. The send frequency response in dB, HS(f), is
given by Equation 7.3Equation 7.3Equation 7.2 The send frequency response, HS(f) may be used to calculate the
send loudness rating (SLR) according to ITU-T Recommendation P.79-1999. Please see Annex H.
Note - If a sinusoidal stimulus was used to measure receive frequency response in 7.4.1, the same sinusoidal
frequency pattern shall be used for the mute measurement, but only over the range of 200-4000 Hz, at 0 dBV. The
absolute level at each frequency is measured, not the frequency response. A-weighting should be applied to the
result, expressed in dBA as a function of frequency. The weighting permits more relevant comparison with results
obtained with artificial voices.
H S ( f )  20 log
1665
1666
1667
1668
1669
1670
1671
1672
1673
1674
G SETP ( f )
G MRP ( f )
in dBV / Pa
Equation 7.3322
where:
GSETP(f) is the rms spectrum at SETP
GMRP(f) is the rms spectrum at MRP
In some cases, frequency response calculation may be performed with cross-spectrum or related techniques.
Justification for such techniques shall be given in the test report. See clause G.1 for more information.
1675
7.5.2
Send noise
1676
1677
1678
1679
1680
1681
1682
1683
1684
1685
1686
1687
1688
1689
Send noise is internally generated audio frequency noise present at the SETP. Measure the electrical output signal at
SETP, averaging over a mi
n
i
mum pe
r
i
odof5s
e
c
on
ds
.Th
et
e
l
e
ph
on
e
’
smi
c
r
oph
on
es
h
ou
l
dbei
s
ol
a
t
e
df
r
om s
ou
n
d
input and mechanical disturbances that would cause significant error. Send noise should be measured with the mute
f
e
a
t
u
r
ebot
h“
on
”a
n
d“
of
f
.
”
1690
7.5.3
1691
1692
Send narrow-band noise, including single frequency interference (SFI), is an impairment that can be perceived as a
tone relative to the overall weighted noise level. This test measures the weighted noise level characteristics in
Send overall noise shall be measured and reported in units of dBmp. It shall also be measured with A-weighting
(defined in ANSI S1.4), reported in units of dBm(A). Measurements in dBmp and dBm(A) are generally not the
same, and they may not be correlated.
Psophometric measurements are made from 25100-6000 Hz, while A-weighted measurements are made from
25100-8,500 Hz. These measurements can be made directly using a psophometrically weighted or A-weighted noise
meter with the correct terminating impedance. The measurement may also be implemented using a single-channel
FFT with Hann windowing, or a real-time spectrum analysis, followed by a weighted power summation.
Send narrow-band noise
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
42
IEEE P269/D25 October 2004
1693
1694
1695
1696
1697
1698
1699
1700
1701
1702
1703
1704
narrow bands of not more than 31 Hz maximum from 25100 –60500 Hz. These levels can then be compared to the
send noise (7.5.2).
1705
7.5.4
1706
1707
1708
1709
1710
1711
1712
1713
1714
1715
1716
1717
1718
1719
1720
1721
1722
Send linearity is a measure of how the frequency response changes with input level.
1723
7.5.5
1724
1725
1726
1727
1728
1729
1730
1731
1732
1733
1734
The preferred distortion measurement method is send signal-to-distortion-and-noise ratio (SDN), measured using
narrow-band pseudo-random noise as the stimulus. See A.1.1J.3 for details of the method.
1735
7.5.6
1736
1737
1738
1739
1740
1741
1742
1743
The mute function is for voice privacy during line hold and mute. Send muting is often manually controlled, but may
be automatically controlled. Mute leakage is the amount of signal measured at the SETP when an acoustic stimulus
is applied to the handset or headset microphone.
The handset or headset should be isolated from sound input and mechanical disturbances that would cause
significant error. Measure the psophometrically-weighted noise level at the SETP with a selective voltmeter or
spectrum analyzer with an effective bandwidth of not more than 31 Hz, over the frequency range of 25100 to 60500
Hz, averaging over a minimum period of 5 seconds.I
fFFTa
n
a
l
y
s
i
si
sus
e
d,t
h
e
na“
Fl
a
tTop”wi
n
dowi
ngs
h
a
l
lbe
employed.
The procedure shall be repeated using A-weighting instead of psophometric weighting, and the frequency range
shall be changed to 25100 –8500 Hz.
Send linearity
The test consists of measuring the send frequency response as specified in Clause 7.5.1 and applying the procedures
described in Annex I. Linearity shall be measured using the same test method and stimulus type used to measure
frequency response.
If artificial voices or another wideband test signal are used, the test shall be performed at 7 levels from –34.7 dBPA
to –4.7dBPa, in 5 dB intervals, measured in 1/3 octave bands. Smaller intervals and/or a wider range of levels may
also be used. The reference stimulus level is –4.7 dBPa. These levels take into account the high crest factor of
artificial voices, which approaches 23 dB.
If sine wave signals are used, they shall be applied at the R10 frequencies from 200 through 5000 Hz, at 7 levels,
from –24.7 to +5.3 dBPa, in 5 dB intervals. Smaller intervals and/or a wider range of levels may also be used. The
reference stimulus level is –9.7 dBPa.
Send distortion
Send distortion is measured at SETP using the standard input level of –4.7 dBPa. Other input levels should be tested
covering a range from –30 to +10 dBPa. Measurements should also be made over a range of frequencies within the
telephone band, such as the ISO R10 preferred frequencies. For higher input levels, verify that distortion of the test
system is less than 2% THD.
For information about THD and other distortion measurement methods and test signals, and the conditions under
which they may be used, see Annex J. Different distortion measurement methods are likely to give different results.
Send mute leakage
To measure mute leakage, engage the mute, apply the test signal, and measure the send noise according to clause
7.5.2. The test signal shall be the same as that used for send frequency response (7.5.1), at +5 dBPa. The result is
expressed in dBmp, a weighted noise level which should be compared to muted broad-band noise measured
according to 7.5.2 (with no stimulus).
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
43
IEEE P269/D25 October 2004
1744
1745
1746
1747
1748
1749
1750
Note - If a sinusoidal stimulus was used to measure send frequency response in 7.5.1, the same sinusoidal frequency
pattern shall be used for the mute measurement, but only over the range of 200-4000 Hz, at +5 dBPa. The absolute
level at each frequency is measured, not the frequency response. Psophometric weighting should be applied to the
result, expressed in dBmp as a function of frequency. The weighting permits more relevant comparison with results
obtained with artificial voices.
1751
7.5.7
1752
1753
1754
1755
1756
1757
1758
1759
1760
1761
1762
1763
1764
Send frequency response in a diffuse field is a measure of how much of the noise in the room where a telephone is
being used is transmitted to the network. It is the ratio of voltage output at the Send Electrical Test Point (SETP) to
the sound pressure at the Diffuse Field Test Point (DFTP, see 5.5.3), which is expressed in decibels. The diffuse
field send frequency response in dB, HSD(f), is given by equation Equation 7.5Equation 7.5Equation 7.3.
Send frequency response in a diffuse field
The diffuse field send frequency response may be sensitive to both the level and type of signal used. This
measurement may be performed in 1/3 octave resolution.
During the measurement, the mouth simulator is present but not active, with the MRP is located at the DFTP. The
mouth simulator is not present during calibration.
H SD ( f )  20 log
1765
1766
1767
1768
1769
1770
1771
1772
1773
1774
G SETP ( f )
G DFTP ( f )
in dBV / Pa
Equation 7.5533
where:
GSETP(f) is the rms spectrum at SETP
GDFTP(f) is the rms spectrum at DFTP
The cross-spectrum method is not recommended.
1775
7.5.8
Send signal-to-noise ratio
1776
1777
1778
Send signal-to-noise ratio is a measure of the desired speech transmission relative to unwanted noise in the room
whe
r
et
h
et
a
l
k
e
r
’
sph
on
ei
sus
e
d.Se
eAnnex K.
1779
7.6
1780
1781
Sidetone should be measured at minimum, reference, and maximum volume settings.
1782
7.6.1
1783
1784
1785
1786
1787
1788
1789
1790
Talker sidetone frequency response is the ratio of the sound pressure measured in the ear simulator, referred to the
Ear Reference Point (ERP), to the sound pressure at the Mouth Reference Point (MRP), which is expressed in
decibels. The talker sidetone frequency response in dB, HTS(f), is given by Equation 7.7Equation 7.7Equation 7.4.
Talker sidetone frequency response may be used to calculate the sidetone masking rating (STMR) according to ITUT Recommendation P.79-1999. Please see Annex H.
Sidetone
Talker sidetone frequency response
The STMR measured on an open-ear HATS is approximately 24 dB. This represents the effective floor of STMR
measurements on actual telephones.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
44
IEEE P269/D25 October 2004
1791
1792
H TS ( f )  20 log
1793
1794
1795
1796
1797
1798
1799
1800
1801
1802
G ERP ( f )
G MRP ( f )
in dBPa / Pa
Equation 7.7744
where:
GERP(f) is the rms spectrum at ERP
GMRP(f) is the rms spectrum at MRP
In some cases, frequency response calculation may be performed with cross-spectrum or related techniques.
Justification for such techniques shall be given in the test report. See clause G.1 for more information.
1803
7.6.2
1804
1805
1806
1807
1808
1809
1810
1811
Listener sidetone is a measure of the signal present at the receiver due to sound in the room where the telephone is
used. The measurement is similar to talker sidetone, except that the stimulus signal is generated in the entire test
room, and not presented from a mouth simulator.
Listener sidetone frequency response is the ratio of the sound pressure measured in the ear simulator, referred to the
Ear Reference Point (ERP), to the sound pressure from a diffused sound field at the DFTP (5.5.3), which is
expressed in decibels. The listener sidetone frequency response in dB, HLS(f), is given by Equation 7.9Equation
7.9Equation 7.5.
H LS ( f )  20 log
1812
1813
1814
1815
1816
1817
1818
1819
1820
1821
1822
1823
1824
1825
1826
1827
1828
1829
Listener sidetone frequency response
G ERP ( f )
G DFTP ( f )
in dBPa / Pa
Equation 7.9955
where:
GERP(f) is the rms spectrum at ERP
GDFTP(f) is the rms spectrum of the diffuse sound field in the room
The cross-spectrum method is not recommended for listener sidetone frequency response calculation.
This measurement is conducted using a uniform diffuse sound field as specified in clause 5.5.3. This measurement
may be performed in 1/3 octave resolution.
The level of the test signal should be in the range of 40–65 dBA. The level and spectrum used should be reported.
For measurement of listener sidetone, the handset or headset is mounted on an appropriate test fixture. The mouth
simulator is present, but not active, with the MRP at the DFTP.
1830
7.6.3
Alternate method for listener sidetone
1831
1832
1833
1834
1835
1836
1837
1838
For the alternate method, listener sidetone response HLS(f) can be approximated by Equation 7.11Equation
7.11Equation 7.6. It is the talker sidetone response HTS(f) minus the difference in send frequency responses from
the standard near field method and a similar method using a diffuse noise signal.
To use this alternate method, measure the talker sidetone per 7.6.1, measure the send frequency response per 7.5.1,
then measure the send frequency response in a diffuse field per 7.5.7 and apply Equation 7.11Equation 7.11Equation
7.6.
Copyright © 2004 IEEE. All rights reserved.
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45
IEEE P269/D25 October 2004
H LS ( f )  H TS ( f )  [ H S ( f ) H SD ( f ) ] in dBPa / Pa
1839
1840
1841
1842
1843
1844
1845
1846
1847
1848
1849
Equation 7.111166
where
HTS(f) = Talker sidetone response
HS(f) = Send frequency response, standard method
HSD(f) = Send frequency response in a diffuse field
CAUTION: This method may not be valid when the send, receive or sidetone path has nonlinear characteristics.
1850
7.6.4
Sidetone linearity
1851
1852
1853
1854
1855
1856
1857
1858
1859
1860
1861
1862
1863
1864
1865
Sidetone linearity is a measure of how the frequency response changes with input level.
1866
7.6.5
1867
1868
1869
1870
1871
1872
1873
1874
1875
1876
1877
The preferred distortion measurement method is sidetone signal-to-distortion-and-noise ratio (SDN), measured using
narrow-band pseudo-random noise as the stimulus. See A.1.1J.3 for details of the method.
1878
7.6.6
1879
1880
1881
Sidetone delay is measured between the mouth simulator and the ear simulator, using one of the methods described
in Annex L
1882
7.6.7
1883
1884
If round trip sidetone delay is more than 5 ms, sidetone echo response should be measured. See Annex M.
The test consists of measuring the talker sidetone frequency response as specified in Clause 7.6.1 and applying the
procedures described in Annex I. Linearity shall be measured using the same test method and stimulus type used to
measure frequency response.
If artificial voices or another wideband test signal are used, the test shall be performed at 7 levels from –34.7 dBPA
to –4.7 dBPa, in 5 dB intervals, measured in 1/3 octave bands. Smaller intervals and/or a wider range of levels may
also be used. The reference stimulus level is –4.7 dBPa. These levels take into account the high crest factor of
artificial voices, which approaches 23 dB.
If sine wave signals are used, they shall be applied at the R10 frequencies from 200 through 5000 Hz, at 7 levels,
from –24.7 to +5.3 dBPa, in 5 dB intervals. Smaller intervals and/or a wider range of levels may also be used. The
reference stimulus level is –9.7 dBPa.
Sidetone distortion
Sidetone distortion is measured at ERP using the standard input level of –4.7 dBPa. Other input levels should be
tested covering a range from –30 to +10 dBPa. Measurements should also be made over a range of frequencies
within the telephone band, such as the ISO R10 preferred frequencies. For higher input levels, verify that distortion
of the test system is less than 2% THD.
For information about THD and other distortion measurement methods and test signals, and the conditions under
which they may be used, see Annex J. Different distortion measurement methods are likely to give different results.
Sidetone delay
Sidetone echo response
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46
IEEE P269/D25 October 2004
1885
7.7
Overall
1886
7.7.1
Overall frequency response
1887
1888
1889
1890
1891
1892
1893
1894
1895
1896
1897
Overall frequency response is measured on two telephones connected as shown in Figure 10Figure 10Figure 10.
This is a simulated end-to-end setup requiring two test fixtures acoustically isolated from each other. The test
conditions should generally be the same as those used for send and receive measurements on the same telephone(s).
Overall frequency response is the ratio of the sound pressure measured in the ear simulator, referred to the Ear
Reference Point (ERP), on the far-end telephone, to the sound pressure at the Mouth Reference Point (MRP) for the
near-end telephone, which is expressed in decibels. The overall frequency response in dB, HO(f), is given by
Equation 7.13Equation 7.13Equation 7.7. It may be used to calculate the overall loudness rating (OLR) according to
ITU-T Recommendation P.79-1999. Please see Annex H.
H O ( f )  20 log
1898
1899
1900
1901
1902
1903
1904
1905
1906
1907
G ERP ( f )
G MRP ( f )
in dBPa / Pa
Equation 7.131377
where:
GERP(f) is the rms spectrum at ERP
GMRP(f) is the rms spectrum at MRP
In some cases, frequency response calculation may be performed with cross-spectrum or related techniques.
Justification for such techniques shall be given in the test report. See clause G.1 for more information.
1908
7.7.2
Overall linearity
1909
1910
1911
1912
1913
1914
1915
1916
1917
1918
1919
1920
1921
1922
1923
Overall linearity is a measure of how the frequency response changes with input level.
1924
7.7.3
1925
1926
1927
Overall distortion is measured in a similar manner to sidetone distortion. However, this measurement is between two
telephone sets connected across a network connection.
1928
7.8
1929
1930
These measurements should be made at the standard test level of –16 dBV.
The test consists of measuring the overall frequency response as specified in Clause 7.7.1 and applying the
procedures described in Annex I. Linearity shall be measured using the same test method and stimulus type used to
measure frequency response.
If artificial voices or another wideband test signal are used, the test shall be performed at 7 levels from –34.7 dBPA
to –4.7 dBPa, in 5 dB intervals, measured in 1/3 octave bands. Smaller intervals and/or a wider range of levels may
also be used. The reference stimulus level is –4.7 dBPa. These levels take into account the high crest factor of
artificial voices, which approaches 23 dB.
If sine wave signals are used, they shall be applied at the R10 frequencies from 200 through 5000 Hz, at 7 levels,
from –24.7 to +5.3 dBPa, in 5 dB intervals. Smaller intervals and/or a wider range of levels may also be used. The
reference stimulus level is –9.7 dBPa.
Overall distortion
Telephone set impedance
Copyright © 2004 IEEE. All rights reserved.
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47
IEEE P269/D25 October 2004
1931
7.8.1
AC impedance
1932
1933
1934
The impedance, measured at the line terminals of the telephone set, should be determined over the frequency range
100 Hz to 8500 Hz.
1935
7.8.2
1936
1937
1938
1939
Return loss is defined by Equation 7.15Equation 7.15Equation 7.8, and measured with the circuit in Figure 14Figure
14Figure 14:
Return loss
RL ( f )  20 log
1940
1941
1942
1943
1944
1945
1946
1947
VA
VB
Equation 7.151588
Where
RL(f) = return loss
VA = voltage applied to test circuit
VB = voltage measured at test point
ZR
G
R1
VA
VB
R1
1948
1949
1950
1951
1952
1953
1954
1955
1956
1957
1958
DC Feed Circuit
CR
RR
Telephone
Figure 14 Return loss test circuit
ZR = reference impedance consisting of RR and CR (typical)
R1 = 600 ohms (typical). Resistors R1 shall match within 0.5% or better
G = signal generator
Echo return loss can be calculated from return loss according to ITU-T G.122 (1993) Annex B, Section B.4
(trapezoidal rule).
1959
7.9
Howling
1960
1961
1962
1963
1964
1965
1966
1967
Telephones can experience instability such as howling, acoustic feedback, or oscillation , when subjected to various
loop circuits, receive volume control settings, and physical positioning of the handset or headset. Instability can be
evaluated using the feed circuit described in Clause 7. The instability should be checked over a range of loop
lengths. The position of the handset or headset can have a major effect on the acoustic stability of the telephone, as
nearby acoustic reflecting surfaces can add to the feedback of the receiver into the microphone.
For each test setup, the telephone shall be evaluated with the receive volume control in the reference receive volume
control and reference send gain control setting, the lowest volume setting, and the highest volume setting. Instability
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48
IEEE P269/D25 October 2004
1968
1969
1970
1971
1972
1973
can be perceived as an audible howling or whistling from the telephone receiver, or repetitive fluctuation of the
telephone set line current.
1974
7.10
1975
1976
1977
1978
1979
1980
1981
1982
1983
1984
1985
1986
1987
1988
In each test loop or receive volume control setting, the handset or headset should be placed in a minimum of four
physical positions: Face up, face down and lying sideways on a hard flat surface, and in the reference corner shown
in Figure 7Figure 7Figure 7 of clause 5.6.1.
Maximum acoustic output
The testing methods provided in this clause only cover the application of in-band signals, but the same sound
pressure limits may apply if ringing signals appear in the handset or headset receiver while the telephone set is offhook. See Annex N for a discussion of maximum pressure limits.
Maximum acoustic output measurements shall be made on the same ear simulator and with the same positioning and
force as used for receive frequency response measurements. For handsets measured on HATS, an additional
measurement with a force of 13N is required. See 5.3.2 for handsets, and 5.3.3 for headsets. Telephone sets with
adjustable receive volume controls shall be adjusted to the maximum setting.
Acoustic output can be referenced to the ERP, DRP, free field (0 degrees elevation and azimuth) or to a diffuse field,
as required by the appropriate safety standard. This may require measurements made at one reference point be
translated to the required reference point. A filter may be required. See Annex C.
1989
7.10.1
Maximum acoustic pressure (long duration)
1990
1991
1992
1993
1994
1995
1996
1997
1998
1999
2000
2001
2002
The maximum acoustic pressure is the maximum steady state sound pressure emitted from a receiver. The stimulus
for this test is a slow logarithmic sine sweep applied at RETP from 100 to 8500 Hz. The measurement shall be made
with real-time filter analysis (RTA) in 1/12 octave bands, described in G.3. The detector shall be set to rms fast,
which is a 250ms effective averaging time (equivalent to a 125ms time constant). The detector shall be set to hold
the maximum level achieved in each band during the entire sweep.
2003
7.10.2
2004
2005
2006
2007
2008
2009
2010
2011
2012
2013
2014
The peak acoustic pressure is the maximum unweighted peak sound pressure emitted from a telephone receiver.
The stimulus for this test is a surge applied at RETP. The measurement shall be made at the ear simulator with an
u
nwe
i
gh
t
e
d“
pe
a
kh
ol
d”l
e
v
e
lde
t
e
c
t
orwi
t
har
i
s
et
i
mee
qu
a
lt
o,orl
e
s
st
h
a
n
,50µs
.
The sweep time shall be at least 90 seconds. A sweep time should be selected that provides consistent results with
no underestimation. That is, the result should be within 0.5 dB at all frequencies for a test period ± 30 seconds.
Additional consideration should be given to the acoustic pressure caused by tones, other audio signals, or long
duration, high amplitude electrical signals applied to power, network, or auxiliary leads of the telephone.
Peak acoustic pressure (short duration)
The 10/700 s surge generator specified in clause 6.2 of IEC 61000-4-5 shall be used. The open circuit voltage shall
be 1000 volts, and the short circuit current shall be 25 amps.
Measure the peak pressure in the ear simulator while operating the surge generator. An oscilloscope or a sound level
me
t
e
r
,h
a
v
i
nga
nu
nwe
i
gh
t
e
d“
pe
a
kh
ol
d”s
e
t
t
i
ngi
sus
e
dt
oma
k
et
h
eme
a
s
u
r
e
me
n
t
.Re
v
e
r
s
et
h
et
e
l
e
ph
on
es
e
t
connections and repeat.
Copyright © 2004 IEEE. All rights reserved.
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49
IEEE P269/D25 October 2004
2015
8
Test Procedures for Digital and 4-wire Systems
2016
8.1
General
2017
2018
2019
2020
2021
2022
2023
2024
2025
2026
2027
2028
2029
2030
2031
2032
2033
2034
2035
2036
The test procedures for digital telephone sets generally follow those for analog telephone sets when using a
reference codec as the digital interface. The procedures in this clause assume a telephone set equipped with a
handset or headset.
2037
8.1.1
2038
2039
2040
2041
2042
2043
2044
2045
2046
2047
2048
2049
2050
2051
2052
2053
2054
In general, multiple test signals and stimulus levels should be used to ensure the telephone is characterized in
realistic, stable and well-defined states. This is especially the case for telephones with non-linear processes such as
compression or voice activated switching (VOX) circuitry, etc. See Annex F & Annex G for further information on
test signals and analysis methods.
2055
8.1.2
2056
2057
2058
2059
2060
2061
2062
2063
The measurement shall be performed using the same format as was used for calibration. Format examples are 1/N
octave bandwidth analysis, constant bandwidth analysis and R-series preferred frequencies. Measurement
bandwidth shall be the same as or less than that which was used for calibration. Measurement resolution shall be the
same as or coarser than that which was used for calibration. The actual bandwidth used shall be stated.
Procedures are given in the following clauses for measurement of parameters affecting the receive, send, sidetone,
and overall performance characteristics of digital telephone sets. These parameters include frequency response,
noise, linearity, distortion, delay, and out-of-band signals. In addition, procedures are given for measuring echo,
stability loss, convergence time, discontinuous speech transmission and maximum acoustic output.
The telephone should be connected to the test circuit(s) described in clause 8.2. Other test circuits may be used for
specific applications. Records should be kept of the measurement setup and conditions.
The measured frequency responses shall be presented as decibels relative to one pascal per volt [dB (Pa/V)] for
receive, decibels relative to one volt per pascal [dB (V/Pa)] for send, decibels relative to one pascal per pascal [dB
(Pa/Pa)] for sidetone and overall, and decibels relative to one volt per volt [dB (V/V)] for echo. The stimulus level
and signal type shall be reported for each test.
The calibration procedures described in clause 6 shall be carried out before making any measurements. The
acoustical test environment shall meet the specifications given in clause 5.5.
Choice of test signals and levels
The standard test signal for all telephones consists of artificial voices defined in ITU-T Recommendation P.50. See
(F.6.1.1) for detail
Sinusoidal test signals (F.4.1) may be used for testing telephones, handsets or headsets if it can be shown that they
do not have adaptive, nonlinear or dynamic signal processing (e.g. compressors, AGC, voice activity detection,
adaptive echo cancellers, etc.). Such evidence must be given in the test report if sinusoidal test signals are used.
Other test signals may be used when it can be shown that they produce results consistent with actual use. They also
may be necessary for some specific purposes as discussed in relevant places within this standard.
The measurements in this clause shall be performed at the standard test levels specified in 6.7.2 and 6.6.2.
Measurement bandwidth and resolution
In general, the test signals and analysis methods in this standard cover a frequency range from approximately 100 to
8500 Hz. The exact range depends on the codec, analysis method, and the test signal (see G.6 and G.7).
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50
IEEE P269/D25 October 2004
2064
8.1.3
Choice of ear and mouth simulators and test position
2065
2066
2067
2068
2069
2070
Choose the ear simulator, mouth simulator and test position according to clauses 5.1, 5.2, & 5.3. This equipment
shall be used for all tests described in clause 8, unless otherwise specified. The ear simulator, mouth simulator, and
test position used shall be stated.
2071
8.1.4
2072
2073
2074
2075
2076
2077
2078
2079
If the telephone is equipped with a tone control, the tone control shall be set to the manuf
a
c
t
u
r
e
r
’
sde
f
a
u
l
ts
e
t
t
i
ng
.
This is the default tone control adjustment that shall be used for all measurements.
2080
8.1.5
2081
2082
2083
All measurements shall be done at the reference receive volume control setting (3.38) A range of volume control
settings may also be used where appropriate, such as minimum and maximum volume.
2084
8.1.6
2085
2086
2087
All measurements shall be done at the reference send volume control setting (3.39). A range of volume control
settings may be used where appropriate, such as minimum and maximum volume.
2088
8.2
Digital test circuits
2089
8.2.1
Digital telephone interface
2090
2091
2092
2093
2094
2095
2096
2097
2098
2099
2100
2101
2102
2103
2104
2105
2106
2107
2108
If analog test equipment is used, the digital telephone under test shall be connected to the reference codec through an
interface as shown in Figure 15Figure 15Figure 15. The interface shall provide all the signaling and supervisory
sequences necessary for the telephone set to work in all test modes. The interface shall also be capable of converting
a digital stream to or from the telephone set under test to a format compatible with the reference codec.
For wideband applications, the Type 1 ear simulator shall not be used, since it is intended for use only to 4,000 Hz.
Tone control setting
If no default setting is defined by the manufacturer, the tone control shall be set so that the frequency response is as
close as possible to the center of the required frequency response template. The tone control shall be set before
setting the volume control. If the tone and volume controls interact, an iterative process for setting these controls
may be necessary.
Reference receive volume control
Reference send gain control setting
The send electrical test point (SETP) is for measuring send output signals. It shall be connected to a 600 ohm load.
The receive electrical test point (RETP) is for applying receive input signals. It shall be connected to a 600 ohm
source.
If digital test equipment is used, the digital telephone under test shall be connected using a direct digital interface as
shown in Figure 16Figure 16Figure 16. In this case, a reference codec is not required, as the measurements are done
in the digital domain. SETP and RETP would then be located at the digital translation interface. Digital signals must
be referenced to the analog equivalent as defined in 8.2.2.
For wireless telephones, the interface is the same, except that a radio link is also included in the interface.
The interfacing for overall response consists of two telephone sets connected back-to-back through the appropriate
digital telephone interface, with or without the ideal codec as necessary. (Figure 17Figure 17Figure 17)
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51
IEEE P269/D25 October 2004
2109
Reference Codec
Linear PCM to
Analog
(600 Ohms)
RETP
(600 Ohm Source)
SETP
(600 Ohm Load)
2110
2111
2112
2113
2114
2115
2116
Figure 15 –Analog interface to a digital set
Linear PCM
Digital Format
2117
2118
2119
2120
2121
2122
2123
2124
2125
Digital Translation
Interface
(as needed)
RETP
SETP
Digital Translation
Interface
(as needed)
Figure 16 –Digital interface to a digital set
Digital Translation
Interface
(as needed)
2126
2127
2128
2129
2130
2131
2132
Digital Translation
Interface
(as needed)
Figure 17 –digital interfacing for overall measurements
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52
IEEE P269/D25 October 2004
2133
8.2.2
Reference codec
2134
8.2.2.1
General
2135
2136
2137
2138
A reference codec is used for testing a digital telephone with analog test equipment. The standard for encoding voice
frequency signals in North America is the µ-law, which is defined in ITU-T Recommendation G.711. The codec
defined in this clause is based on that standard. For other coding schemes, an appropriate codec should be used.
2139
8.2.2.2
2140
2141
2142
2143
2144
2145
2146
2147
2148
2149
2150
2151
2152
2153
2154
The analog input and output impedance of the reference codec shall be 600 ohms. In the previous version of this
standard, a termination impedance of 900 ohms was used to be consistent with analog telephone set measurements.
In this version, a 600 ohm termination is used for international harmonization.
2155
8.2.2.3
2156
2157
2158
2159
2160
2161
2162
2163
In addition, reference codec characteristics, such as attenuation versus frequency distortion, idle channel noise, and
quantizing distortion should meet or exceed characteristics specified in ITU-T Recommendation G.714 [6].
2164
8.2.3
Wideband reference codec
2165
8.2.3.1
General
2166
2167
2168
2169
There are a number of wideband codecs being used including ITU-T G.722, and low bit rate vocoders, such as
G.722.1, G.723.1, and G.729. However, the codec defined in this clause is based on 16 bit, 16 kHz linear PCM
coding or 256 kbit/s. For other coding schemes, an appropriate codec should be used.
2170
8.2.3.2
2171
2172
2173
2174
2175
2176
2177
2178
The analog input and output impedance of the reference codec shall be 600 ohms. In the previous version of this
standard, a termination impedance of 900 ohms was used to be consistent with analog telephone set measurements.
In this version, a 600 ohm termination is used for international harmonization.
Conversion Relationships
For the digital-to-analog (D/A) converter, a digital test sequence (DTS) representing the pulse-code modulation
(PCM) equivalent of an analog sinusoidal signal whose rms value is 3.17 dB below the maximum full load capacity
of the codec shall generate 0 dBm in a 600 ohm load.
For the analog-to-digital (A/D) converter, a 0 dBm signal from a 600 ohm source shall give the DTS representing
the PCM equivalent of an analog sinusoidal signal whose rms value is 3.17 dB below the maximum full-load
capacity of the codec.
Note that a 0 dBm signal is not the maximum digital code. For µ-law codecs 0 dBm is 3.17 dB below digital full
scale. For A-law codecs 0 dBm is 3.14 dB below digital full scale.
Other Parameters
The idle channel noise should be less than -84 dBmp when receiving one of the quiet codes or when the A/D digital
output is connected to the D/A digital input. The quantizing distortion of the reference codec should approach
theoretical limits specified in Annex A of ITU-T Recommendation O.133. The intrinsic error of µ-law PCM
encoding limits the signal-to-distortion ratio to about 38 dB.
Conversion Relationships
For the digital-to-analog (D/A) converter, a digital test sequence (DTS) representing the pulse-code modulation
(PCM) equivalent of an analog sinusoidal signal whose rms value is 3.17 dB below the maximum full load capacity
of the codec shall generate 0 dBm in a 600 ohm load. This is the same as prescribed for G.711.
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53
IEEE P269/D25 October 2004
2179
2180
2181
2182
For the analog-to-digital (A/D) converter, a 0 dBm signal from a 600 ohm source shall give the DTS representing
the PCM equivalent of an analog sinusoidal signal whose rms value is 3.17 dB below the maximum full-load
capacity of the codec. Here again, the conversion is the same as G.711.
2183
8.2.3.3
2184
2185
2186
2187
2188
2189
2190
In addition, wideband reference codec characteristics, such as frequency bandwidth and idle channel noise should
meet or exceed the characteristics specified below.
2191
8.3
2192
2193
2194
2195
2196
2197
2198
2199
2200
2201
2202
2203
The most common digital impairments include delay, bit errors, frame or packet loss, and network echo cancellers
which the phone might encounter. There are many commercial units available to induce these impairments, and are
usually specific to the type of digital transmission system being tested.
2204
8.3.1
2205
2206
2207
2208
2209
2210
2211
2212
2213
2214
Network delay, or latency, is the most important impairment for packet voice network devices. If the device features
non-linear processes (echo cancellation, or voice activity detection) to enhance voice quality, these processes can be
sensitive to network delay. In this case, echo canceller performance (see clauses 8.11 and 8.12) should be checked
with simulated network delay of 50, 150 and 300ms one way to ensure that performance is not degraded.
2215
8.3.2
2216
2217
2218
2219
2220
2221
2222
2223
2224
2225
Jitter is a variation in network or device delay due to the late or early arrival of packets in packet based systems.
Jitter can cause problems with some of the test methods. If possible, jitter should be removed during testing. This
can be done by increasing the size of the jitter buffer, resulting in a longer, but stable delay. This stable delay can
then be measured, and the result used to offset the source and measurement signal where temporal correlation of
these are important to the test.
Other Parameters
The nominal 3 dB bandwidth shall be 50 Hz to 7,000 Hz with anti-aliasing filter ripple less than ± 0.5 dB. The idle
channel noise should be less than -89 dBm unweighted across this same bandwidth when receiving the quiet code or
when the A/D digital output is connected to the D/A digital input.
Digital telephone network impairments
Network impairments can vary between types of voice networks. With the introduction of packet voice transmission,
such as Voice over IP (VoIP), new types of impairments have been introduced. Impairments in ISDN and similar
systems are typically limited to speech compression transcoding (A-law to u-law conversions etc.), speech path
compression (G.726 ADPCM compression) and delay.
Some tests are sensitive to these impairments, and it is important to understand the performance of the telephone and
the suitability of the test method in the presence of these impairments. For each impairment, the affected tests are
described.
Network Delay
Delay in the device or test setup may affect some tests and the methodology used. It is important to understand how
much delay can be tolerated by each particular test performed. Delay should be measured, and the result used to
offset the source and measurement signals where temporal correlation of these are important to the test. The crossspectrum method for frequency response is one example where system delay must be taken into account.
Jitter
If the jitter can not be sufficiently controlled, then all tests must be performed with caution. In this case, the crossspectrum method for frequency response shall not be used.
Copyright © 2004 IEEE. All rights reserved.
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54
IEEE P269/D25 October 2004
2226
8.3.3
2227
2228
2229
2230
2231
2232
2233
2234
2235
2236
Packet networks can suffer from congestion, causing jitter, buffer under-runs/over-runs, or packets arriving out of
order. This typically results in lost packets. Some devices may feature packet loss protection algorithms. It is
beyond the scope of this standard to detail a test method for packet loss protection performance. It is recommended
that a perceptually based test of packet loss protection be used. A suitable example is PESQ (ITU-T
Recommendation P.832) with 1%, 5% and 10% packet loss, normally distributed with respect to time. Many
networks may experience bursty packet loss; however, it is outside the scope of this standard to define a bursty
distribution.
2237
8.3.4
2238
2239
2240
2241
2242
2243
2244
2245
2246
2247
2248
Network echo cancellers are typically deployed when network delay exceeds 25ms, one way, and may also be
present in the test interface. Echo cancellers can affect non-linear speech path quality enhancing processes due to
filters and echo suppression algorithms.
2249
8.3.5
2250
2251
2252
2253
2254
2255
2256
2257
2258
Discontinuous speech transmission (DTX) is implemented by a voice/speech activity detector (VAD/SAD) in both
the phone and network. It detects when the speech path is idle in a particular transmission direction. The system
will then disable the speech path, allowing additional bandwidth for other network traffic. DTX may cause noise
pumping, and both front end speech clipping and trailing speech clipping.
2259
8.4
Receive
2260
8.4.1
Receive frequency response
2261
2262
2263
2264
2265
2266
Receive frequency response is the ratio of sound pressure measured in the ear simulator, referred to the Ear
Reference Point (ERP), to the voltage input at the Receive Electrical Test Point (RETP), which is expressed in
decibels. The receive frequency response in dB, HR(f), is given by Equation 8.1Equation 8.1Equation 8.1. The
receive response, HR(f), may be used to calculate the receive loudness rating (RLR), according to ITU-T
Recommendation P.79-1999. Please see Annex H.
All other measurements in Clause 8 shall be performed with packet loss set to zero.
Network Echo Canceller
When a network echo canceller is inserted into the system under test, it is recommended that the send, receive and
overall frequency responses and respective loudness rating, as well as linearity, distortion and noise be tested with
network echo cancellers both enabled and disabled.
All other measurements should operate transparently in the presence of a network echo canceller and should not
need to be investigated.
Discontinuous Speech Transmission
If the phone and test system support DTX, frequency responses, loudness ratings, linearity, and distortion should be
measured with the feature both enabled and disabled. Speech like test signals shall be used for frequency response
and loudness rating measurements since the DTX algorithm may interpret steady state test signals as noise.
H R ( f )  20 log
2267
2268
2269
2270
2271
2272
Network Packet Loss
G ERP ( f )
G RETP ( f )
in dBPa / V
Equation 8.19
where:
GERP(f) is the rms spectrum at ERP
GRETP(f) is the rms spectrum at RETP
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
55
IEEE P269/D25 October 2004
2273
2274
2275
2276
In some cases, frequency response calculation may be performed with cross-spectrum or related techniques.
Justification for such techniques shall be given in the test report. See clause G.1 for more information.
2277
8.4.2
2278
2279
2280
2281
2282
2283
2284
2285
2286
2287
2288
2289
Receive noise is internally generated audio frequency noise present at the receiver when no stimulus is applied.
Connect the telephone set to the reference codec, and transmit idle code or silence to RETP. The receiver shall be
c
ou
pl
e
dt
ot
h
ee
a
rs
i
mul
a
t
or
. Th
et
e
l
e
ph
on
e
’
smi
c
r
oph
ones
h
oul
dbei
s
ol
a
t
e
df
r
om s
ou
n
di
n
pu
ta
n
dme
c
h
a
n
i
c
a
l
disturbances that would cause significant error. Measure the acoustic output signal, referred to the ERP, from 25100
to
8,500 Hz, averaging over a minimum period of 5 seconds. Receive noise should be measured with the telephone
mut
ef
e
a
t
u
r
ebot
h“
on
”a
n
d“
of
f
.
”
2290
8.4.3
2291
2292
2293
2294
2295
2296
2297
2298
2299
2300
2301
2302
Narrow-band noise, including single frequency interference (SFI), is an impairment that can be perceived as a tone
depending on its level relative to the overall weighted noise level. This test measures the weighted noise level
characteristics in narrow bands of not more than 31 Hz, from 25100 to 8,500 Hz. These levels can then be compared
to the receive noise level (8.4.2).
2303
8.4.4
2304
2305
2306
2307
2308
2309
2310
2311
2312
2313
2314
2315
2316
2317
2318
Receive linearity is a measure of how the frequency response changes with input level.
2319
8.4.5
2320
2321
The preferred distortion measurement method is receive signal-to-distortion-and-noise ratio (SDN), measured using
narrow-band pseudo-random noise as the stimulus. See A.1.1J.3 for details of the method. For the narrow band
Receive noise
The receive noise level is measured with A-weighting in dBA. The measurement may be implemented directly
using an A-weighting filter, or by using single-channel FFT with Hann windowing or real-time spectrum analysis,
followed by an A-weighted power summation.
Receive narrow-band noise
The receiver shall be coupled to the ear simulator with idle code or silence at RETP. Measure the A-weighted
receive noise level, referred to the ERP, using a selective voltmeter or spectrum analyzer with an effective
bandwidth of not more than 31 Hz, over the frequency range of 25100 to 8500 Hz, averaging over a minimum
period of 5 seconds.I
fFFTa
n
a
l
y
s
i
si
sus
e
d,t
h
e
n“
Fl
a
tTop
”wi
n
dowi
ngs
h
a
l
lbee
mpl
oyed.
The same procedure applies for wide-band telephony applications.
Receive linearity
The test consists of measuring the receive frequency response as specified in Clause 8.4.1 and applying the
procedures described in Annex I. Linearity shall be measured using the same test method and stimulus used to
measure frequency response, except that the analysis bandwidth is different. For the narrow band codec, the
analysis bandwidth is 100 to 3400 Hz. For the wideband codec, the analysis bandwidth is typically 100 to 6800 Hz.
If artificial voices or another wideband stimulus are used, the test shall be performed at 7 levels, from–48.2 to 18.2
dBV, in 5 dB intervals, measured in 1/3 octave bands. Smaller intervals and/or a wider range of levels may also be
used. The reference stimulus level is –18.2 dBV. These levels take into account the high crest factor of artificial
voices, which approaches 23 dB.
If sine wave signals are used, they shall be applied at the R10 frequencies, at 7 levels, from –38.2 to –8.2 dBV, in 5
dB intervals. Smaller intervals and/or a wider range of levels may also be used. The reference stimulus level is –23.2
dBV.
Receive distortion
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
56
IEEE P269/D25 October 2004
2322
2323
2324
2325
2326
2327
2328
2329
2330
2331
2332
codec, the stimulus bandwidth is 100 to 3400 Hz. For the wideband codec, the stimulus bandwidth is typically 100
to 6800 Hz. In all cases the analysis bandwidth is 100 Hz to 8500 Hz.
2333
8.4.6
2334
2335
2336
2337
2338
2339
2340
2341
2342
2343
2344
2345
2346
2347
2348
2349
2350
Re
c
e
i
v
emut
ei
ss
ome
t
i
me
sc
a
l
l
e
d“
DTMFmu
t
e
”or“
a
u
t
odi
a
lmu
t
e
”
.Re
c
e
i
v
emu
t
i
ngi
sus
u
a
l
l
ya
u
t
oma
t
i
c
,bu
tma
y
be manually controlled, and would normally be activated by touch-t
on
edi
a
l
i
ng
,l
i
n
e“
h
ol
d”ope
r
a
t
i
on
,a
c
t
i
v
a
t
i
n
gt
h
e
hold button, or other means. Mute leakage is the amount of signal measured at the ERP when an electrical stimulus
is applied to the RETP.
2351
8.4.7
2352
2353
2354
2355
2356
2357
2358
2359
Delay is an important factor for digital telephones and network edge devices. It is a measure of the time taken for an
excitation signal to traverse a given speech path for the device. Some devices may have delay in excess of 50 ms, as
well as a variable delay or jitter.
2360
8.4.8
2361
2362
2363
2364
2365
2366
2367
2368
2369
2370
2371
2372
Receive out-of-band signals are signals that appear outside the specified frequency range for any input that is inside
the specified frequency range. This test is designed to ensure that speech processing, coding, or compression is
properly implemented.
Receive distortion is measured at ERP using the standard input level of –18.2 dBV. Other input levels should be
tested covering a range from –30 to 0 dBV. Measurements also should be made over a range of frequencies within
the telephone band, such as the ISO R10 preferred frequencies. For higher input levels, verify that distortion of the
test system is less than 1% THD.
For information about THD and other distortion measurement methods and test signals, and the conditions under
which they may be used, see Annex J. Different distortion measurement methods are likely to give different results.
Receive mute leakage
To measure mute leakage, engage the mute, apply the test signal, and measure the receive noise according to clause
8.4.2. The test signal shall be the same as that used for receive frequency response (8.4.1), at 0 dBV. An additional
measurement shall be made using the DTMF tones of the telephone set being tested. In the case of the DTMF tones
oft
h
et
e
l
e
ph
on
es
e
t
,t
h
e
r
ewon
’
tbea
nyc
on
t
r
olov
e
rt
h
el
e
v
e
l
.Ea
c
hr
e
s
u
l
ti
se
x
pr
e
s
s
e
di
ndBA,awe
i
gh
t
e
dn
oi
s
e
level which should be compared to muted receive noise measured according to 8.4.2 (with no stimulus applied).
Note - If a sinusoidal stimulus was used to measure receive frequency response in 8.4.1, the same sinusoidal
frequency pattern shall be used for the mute measurement, but only over the range of 200-4000 Hz, at 0 dBV. The
absolute level at each frequency is measured, not the frequency response. A-weighting should be applied to the
result, expressed in dBA as a function of frequency. The weighting permits more relevant comparison with results
obtained with artificial voices.
Receive delay
Receive delay is measured between RETP and the ear simulator.
See Annex L for appropriate measurement methods.
Receive out-of-band signals
Apply a sinewave signal at RETP at a level of –18.2 dBV, in the frequency range 300 to 3400 Hz. Measure the
signal level at the ear simulator of any spurious tones that may appear between 4.6 kHz and 8.0 kHz. No weighting
is applied to the result.
For wideband applications, apply a sinewave signal at RETP in the frequency range of 150 to 6.7 kHz. At the ear
simulator measure the level of any spurious tones that may appear from 7.2 kHz to approximately 8.5 kHz (seeF.8).
The out-of-band signals shall be compared to the 1 kHz signal level at the ear simulator.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
57
IEEE P269/D25 October 2004
2373
2374
8.5
Send
2375
8.5.1
Send frequency response.
2376
2377
2378
2379
2380
Send frequency response is the ratio of voltage output at the send electrical test point (SETP) to the sound pressure
at the Mouth Reference Point (MRP), which is expressed in decibels. The send frequency response in dB. HS(f), is
given by Equation 8.3Equation 8.3Equation 8.2. The send frequency response, HS(f) may be used to calculate the
send loudness rating (SLR) according to ITU-T Recommendation P.79-1999. Please see Annex H
H S ( f )  20 log
2381
2382
2383
2384
2385
2386
2387
2388
2389
2390
G SETP ( f )
G MRP ( f )
in dBV / Pa
Equation 8.33210
where:
GSETP(f) is the rms spectrum at SETP
GMRP(f) is the rms spectrum at MRP.
In some cases, frequency response calculation may be performed with cross-spectrum or related techniques.
Justification for such techniques shall be given in the test report. See clause G.1 for more information.
2391
8.5.2
Send noise
2392
2393
2394
2395
2396
2397
2398
2399
2400
2401
2402
2403
2404
2405
2406
2407
Send noise is internally generated audio frequency noise present at the SETP. Connect the telephone set to the
reference codec and place it in an active state with no acoustic input. Measure the electrical output signal at SETP,
averaging over a minimum period of 5 seconds. The telephone microphone should be isolated from sound input and
mechanical disturbances that would cause significant error. Send noise should be measured with the mute feature
bot
h“
on
”a
n
d“
of
f
.
”
Psophometric measurements are made from 100 Hz to 3400 Hz for narrow band codecs, while A-weighted
measurements are made from 100 Hz to 6800 Hz for wide band codecs. These measurements can be made directly
using a psophometrically weighted or A-weighted noise meter with the correct terminating impedance. The
measurement may also be implemented using a single-channel FFT with Hann windowing, or a real-time spectrum
analysis, followed by a weighted power summation.
2408
8.5.3
2409
2410
2411
2412
2413
2414
2415
2416
2417
2418
2419
Narrow-Band noise, including single frequency interference (SFI), is an impairment that can be perceived as a tone
depending on its level relative to the overall weighted noise level. This test measures the weighted noise level
characteristics in narrow bands of not more than 31 Hz., These levels can then be compared to the send noise (8.5.2).
The handset or headset should be isolated from sound input and mechanical disturbances that would cause
significant error. Measure the psophometrically-weighted noise level at the SETP, using a selective voltmeter or
spectrum analyzer, with an effective bandwidth of not more than 31 Hz, averaging over a minimum period of 5
seconds, over the frequency range of 100 to 3400 Hz for narrow band codecs. For wide band codecs, use Aweighting instead of psophometric weighting, over the frequency range of 100 to 6800 Hz.
For narrow band codecs, send overall noise shall be measured and reported in units of dBmp. For wide band codecs,
send overall noise shall be measured with A-weighting (defined in ANSI S1.4), reported in units of dBm(A).
Measurements in dBmp and dBm(A) are generally not the same, and they may not be correlated.
Send narrow-band noise
I
fFFTa
n
a
l
y
s
i
si
sus
e
d,t
h
e
na“
Fl
a
tTop”wi
n
dows
h
a
l
lbee
mpl
oy
e
d.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
58
IEEE P269/D25 October 2004
2420
8.5.4
Send linearity
2421
2422
2423
2424
2425
2426
2427
2428
2429
2430
2431
2432
2433
2434
2435
2436
Send linearity is a measure of how the frequency response changes with input level.
2437
8.5.5
2438
2439
2440
2441
2442
2443
2444
2445
2446
2447
2448
2449
2450
The preferred distortion measurement method is send signal-to-distortion-and-noise ratio (SDN), measured using
narrow-band pseudo-random noise as the stimulus. See A.1.1J.3 for details of the method. For the narrow band
codec, the stimulus bandwidth is 100 to 3400 Hz. For the wideband codec, the stimulus bandwidth is typically 100
to 6800 Hz. In all cases the analysis bandwidth is 100 Hz to 8500 Hz.
2451
8.5.6
2452
2453
2454
2455
2456
2457
2458
2459
2460
2461
2462
2463
2464
2465
2466
The mute function is for voice privacy during line hold and mute. Send muting is often manually controlled, but may
be automatically controlled. Mute leakage is the amount of signal measured at the SETP when an acoustic stimulus
is applied to the handset or headset microphone.
2467
8.5.7
2468
2469
2470
Delay is an important factor for digital telephones and network edge devices. It is a measure of the time taken for an
excitation signal to traverse a given speech path for the device. Some devices may have delay in excess of 50 ms, as
well as a variable delay or jitter.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
59
The test consists of measuring the send frequency response as specified in Clause 8.5.1 and applying the procedures
described in Annex I. Linearity shall be measured using the same test method and stimulus used to measure
frequency response, except that the analysis bandwidth is different. For the narrow band codec, the analysis
bandwidth is 100 to 3400 Hz. For the wideband codec, the analysis bandwidth is typically 100 to 6800 Hz.
If artificial voices or another wideband test signal are used, the test shall be performed at 7 levels from –34.7 dBPA
to –4.7 dBPa, in 5 dB intervals, measured in 1/3 octave bands. Smaller intervals and/or a wider range of levels may
also be used. The reference stimulus level is –4.7 dBPa. These levels take into account the high crest factor of
artificial voices, which approaches 23 dB.
If sine wave signals are used, they shall be applied at the R10 frequencies, at 7 levels, from –24.7 to +5.3 dBPa, in 5
dB intervals. Smaller intervals and/or a wider range of levels may also be used. The reference stimulus level is –9.7
dBPa.
Send distortion
Send distortion is measured at SETP using the standard input level of –4.7 dBPa. Other input levels should be tested
covering a range from –30 to +10 dBPa. Measurements should also be made over a range of frequencies within the
telephone band, such as the ISO R10 preferred frequencies. For higher input levels, verify that distortion of the test
system is less than 2% THD.
For information about THD and other distortion measurement methods and test signals, and the conditions under
which they may be used, see Annex J. Different distortion measurement methods are likely to give different results.
Send mute leakage
To measure mute leakage, engage the mute, apply the test signal, and measure the send noise according to clause
8.5.2. The test signal shall be the same as that used for send frequency response (8.5.1), at +5 dBPa. The result is
expressed in dBmp, a weighted noise level which should be compared to muted broad-band noise measured
according to 8.5.2 (with no stimulus).
Note - If a sinusoidal stimulus was used to measure send frequency response in 8.5.1, the same sinusoidal frequency
pattern shall be used for the mute measurement, but only over the range of 200-4000 Hz, at +5 dBPa. The absolute
level at each frequency is measured, not the frequency response. Psophometric weighting should be applied to the
result, expressed in dBmp as a function of frequency. The weighting permits more relevant comparison with results
obtained with artificial voices.
Send delay
IEEE P269/D25 October 2004
2471
2472
2473
2474
2475
2476
Send delay is measured between the MRP and SETP. The electro-acoustic delay between the electrical input to the
MRP and the microphone of the device should be considered unimportant.
See Annex L for appropriate measurement methods.
2477
8.5.8
Send out-of-band susceptibility
2478
2479
2480
2481
2482
2483
2484
2485
2486
2487
2488
Send out-of-band susceptibility is a measure of signals that appear inside the specified frequency range for any input
that is outside the specified frequency range. This test is designed to ensure that speech processing, coding, or
compression, is properly implemented.
2489
8.5.9
2490
2491
2492
2493
2494
2495
2496
2497
2498
2499
2500
2501
Send frequency response in a diffuse field is a measure of how much of the noise in the room where a telephone is
being used is transmitted to the network. It is the ratio of voltage output at the Send Electrical Test Point (SETP) to
the sound pressure at the Diffuse Field Test Point (DFTP, see 5.5.3), which is expressed in decibels. The diffuse
field send frequency response in dB, HSD(f), is given by equation Equation 8.5Equation 8.5Equation 8.3.
Apply a sinewave signal at the MRP at a level of –4.7 dBPa, in the frequency range 4.5 kHz and 8.5 kHz. Measure
the signal level at the SETP of any spurious tones that may appear between 300 to 3400 Hz. No weighting is
applied to the result.
For wideband applications, apply a sinewave signal at SETP in the frequency range of 7.1 kHz to 8.5kHz. At the
SETP measure the level of any spurious tones that may appear from 100 to 6.8 kHz.
Send frequency response in a diffuse field
The diffuse field send frequency response may be sensitive to both the level and type of signal used. This
measurement may be performed in 1/3 octave resolution.
During the measurement, the mouth simulator is present but not active, with the MRP is located at the DFTP. The
mouth simulator is not present during calibration.
H SD ( f )  20 log
2502
2503
2504
2505
2506
2507
2508
2509
2510
G SETP ( f )
G DFTP ( f )
in dBV / Pa
Equation 8.55311
where:
GSETP(f) is the rms spectrum at SETP
GDFTP(f) is the rms spectrum at DFTP
The cross-spectrum method is not recommended.
2511
8.5.10
Send signal-to-noise ratio
2512
2513
2514
Send signal-to-noise ratio is a measure of the desired speech transmission relative to unwanted noise in the room
whe
r
et
h
et
a
l
k
e
r
’
sph
on
ei
sus
e
d.Se
eAnnex K.
2515
8.6
2516
2517
Sidetone should be measured at minimum, reference, and maximum volume settings.
Sidetone
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
60
IEEE P269/D25 October 2004
2518
8.6.1
2519
2520
2521
2522
2523
2524
2525
2526
2527
2528
Talker sidetone frequency response is the ratio of the sound pressure measured in the ear simulator, referred to the
Ear Reference Point (ERP), to the sound pressure at the Mouth Reference Point (MRP), which is expressed in
decibels. The talker sidetone frequency response in dB, HTS(f), is given by Equation 8.7Equation 8.7Equation 8.4.
Talker sidetone frequency response may be used to calculate the sidetone masking rating (STMR) according to ITUT Recommendation P.79-1999. Please see Annex H.
The STMR measured on an open-ear HATS is approximately 24 dB. This represents the effective floor of STMR
measurements on actual telephones.
H TS ( f )  20 log
2529
2530
2531
2532
2533
2534
2535
2536
2537
2538
Talker sidetone frequency response
G ERP ( f )
G MRP ( f )
in dBPa / Pa
Equation 8.77412
where:
GERP(f) is the rms spectrum at ERP
GMRP(f) is the rms spectrum at MRP
In some cases, frequency response calculation may be performed with cross-spectrum or related techniques.
Justification for such techniques shall be given in the test report. See clause G.1 for more information.
2539
8.6.2
2540
2541
2542
2543
2544
2545
2546
2547
2548
Listener sidetone is a measure of the signal present at the receiver due to sound in the room in which the receiver is
used. The measurement is similar to talker sidetone, except that the stimulus signal is generated in the entire test
room, and not presented from a mouth simulator.
Listener sidetone frequency response is the ratio of the sound pressure measured in the ear simulator, referred to the
Ear Reference Point (ERP), to the sound pressure from a diffused sound field at the DFTP (5.5.3), which is
expressed in decibels. The listener sidetone frequency response in dB, HLS(f), is given by Equation 8.9Equation
8.9Equation 8.5.
H LS ( f )  20 log
2549
2550
2551
2552
2553
2554
2555
2556
2557
2558
2559
2560
2561
2562
2563
2564
2565
2566
Listener sidetone frequency response
G ERP ( f )
GDFTP ( f )
in dBPa / Pa
Equation 8.99513
where:
GERP(f) is the rms spectrum at ERP
GDFTP(f) is the rms spectrum of the diffuse sound field in the room at the DFTP
The cross-spectrum method is not recommended.
This measurement is conducted using a uniform diffuse sound field as specified in clause 5.5.3. This measurement
may be performed in 1/3 octave resolution.
The level of the test signal should be in the range of 40–65 dBA. The level and spectrum used should be reported.
For measurement of listener sidetone, the handset or headset is mounted on an appropriate test fixture. The mouth
simulator is present, but not active, with the MRP at the DFTP.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
61
IEEE P269/D25 October 2004
2567
8.6.3
Alternate method for listener sidetone
2568
2569
2570
2571
2572
2573
2574
2575
2576
For the alternate method, listener sidetone response HLS(f) is given approximately by Equation 8.11Equation
8.11Equation 8.6. It is the talker sidetone response HTS(f) minus the difference in send frequency responses from the
standard near field method and a similar method using a diffuse noise signal.
2577
2578
2579
2580
2581
2582
2583
2584
2585
2586
Equation 8.1111614
To use this alternate method, measure the talker sidetone per 8.6.1, measure the send frequency response per 8.5.1,
then measure the send frequency response in a diffuse field per 8.5.9, and apply Equation 8.11Equation
8.11Equation 8.6.
H LS ( f )  H TS ( f )  [ H S ( f ) H SD ( f ) ] in dBPa / Pa
where
HTS(f) = Talker sidetone response
HS(f) = Send frequency response, standard method
HSD(f) = Send frequency response in a diffuse field
CAUTION: This method may not be valid when the send, receive or sidetone path has nonlinear characteristics.
2587
8.6.4
Sidetone linearity
2588
2589
2590
2591
2592
2593
2594
2595
2596
2597
2598
2599
2600
2601
2602
2603
Sidetone linearity is a measure of how the frequency response changes with input level.
2604
8.6.5
2605
2606
2607
2608
2609
2610
2611
2612
2613
2614
2615
The preferred distortion measurement method is sidetone signal-to-distortion-and-noise ratio (SDN), measured
using narrow-band pseudo-random noise as the stimulus. See A.1.1J.3 for details of the method.
The test consists of measuring the talker sidetone frequency response as specified in Clause 8.6.1 and applying the
procedures described in Annex I. Linearity shall be measured using the same test method and stimulus used to
measure frequency response, except that the analysis bandwidth is different. For the narrow band codec, the
analysis bandwidth is 100 to 3400 Hz. For the wideband codec, the analysis bandwidth is typically 100 to 6800 Hz.
If artificial voices or another wideband test signal are used, the test shall be performed at 7 levels from –34.7 dBPA
to –4.7 dBPa, in 5 dB intervals, measured in 1/3 octave bands. Smaller intervals and/or a wider range of levels may
also be used. The reference stimulus level is –4.7 dBPa. These levels take into account the high crest factor of
artificial voices, which approaches 23 dB.
If sine wave signals are used, they shall be applied at the R10 frequencies, at 7 levels, from –24.7 to +5.3 dBPa, in 5
dB intervals. Smaller intervals and/or a wider range of levels may also be used. The reference stimulus level is –9.7
dBPa.
Sidetone distortion
Sidetone distortion is measured at ERP using the standard input level of –4.7 dBPa. Other input levels should be
tested covering a range from –30 to +10 dBPa. Measurements also should be made over a range of frequencies
within the telephone band, such as the ISO R10 preferred frequencies. For higher input levels, verify that distortion
of the test system is less than 2% THD.
For information about THD and other distortion measurement methods and test signals, and the conditions under
which they may be used, see Annex J. Different distortion measurement methods are likely to give different results.
Copyright © 2004 IEEE. All rights reserved.
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62
IEEE P269/D25 October 2004
2616
8.6.6
Sidetone delay
2617
2618
2619
Sidetone delay is measured between the mouth simulator and the ear simulator, using one of the methods described
in Annex L.
2620
8.6.7
2621
2622
If round trip sidetone delay is more than 5 ms, sidetone echo response should be measured. See Annex M.
2623
8.7
Overall
2624
8.7.1
Overall frequency response
2625
2626
2627
2628
2629
2630
2631
2632
2633
2634
2635
2636
2637
2638
2639
The overall response should be measured using two telephone sets connected back-to-back, through the appropriate
digital telephone interface, with or without the reference codec as necessary.
Sidetone echo response
Overall frequency response is measured on two telephones connected back-to-back, using the interface shown in
Figure 17Figure 17Figure 17 of 8.2.1. The test conditions should be chosen according to 8.1, except that two test
fixtures are used. In general, the test conditions should be the same as those used for send and receive
measurements on the same telephone(s).
Overall frequency response is the ratio of the sound pressure measured in the ear simulator, referred to the Ear
Reference Point (ERP), on the far-end telephone, to the sound pressure at the Mouth Reference Point (MRP) for the
near-end telephone, which is expressed in decibels. The overall frequency response in dB, HO(f), is given by
Equation 8.13Equation 8.13Equation 8.7. It may be used to calculate the overall loudness rating (OLR) according to
ITU-T Recommendation P.79-1999. Please see Annex H.
H O ( f )  20 log
2640
2641
2642
2643
2644
2645
2646
2647
2648
2649
G ERP ( f )
G MRP ( f )
in dBPa / Pa
Equation 8.1313715
where:
GERP(f) is the rms spectrum at ERP
GMRP(f) is the rms spectrum at MRP
In some cases, frequency response calculation may be performed with cross-spectrum or related techniques.
Justification for such techniques shall be given in the test report. See clause G.1 for more information.
2650
8.7.2
Overall linearity
2651
2652
2653
2654
2655
2656
2657
2658
2659
2660
2661
2662
Overall linearity is a measure of how the frequency response changes with input level.
The test consists of measuring the overall frequency response as specified in Clause 8.7.1 and applying the
procedures described in Annex I. Linearity shall be measured using the same test method and stimulus used to
measure frequency response, except that the analysis bandwidth is different. For the narrow band codec, the
analysis bandwidth is 100 to 3400 Hz. For the wideband codec, the analysis bandwidth is typically 100 to 6800 Hz.
If artificial voices or another wideband test signal are used, the test shall be performed at 7 levels from –34.7 dBPA
to –4.7 dBPa, in 5 dB intervals, measured in 1/3 octave bands. Smaller intervals and/or a wider range of levels may
also be used. The reference stimulus level is –4.7 dBPa. These levels take into account the high crest factor of
artificial voices, which approaches 23 dB.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
63
IEEE P269/D25 October 2004
2663
2664
2665
2666
If sine wave signals are used, they shall be applied at the R10 frequencies, at 7 levels, from –24.7 to +5.3 dBPa, in 5
dB intervals. Smaller intervals and/or a wider range of levels may also be used. The reference stimulus level is –9.7
dBPa.
2667
8.7.3
2668
2669
2670
2671
2672
Overall distortion can be measured in a manner similar to sidetone distortion (8.6.5), except that the measurement is
made from one set to another connected in a back to back configuration. For the narrow band codec, the stimulus
bandwidth is 100 to 3400 Hz. For the wideband codec, the stimulus bandwidth is typically 100 to 6800 Hz. In all
cases the analysis bandwidth is 100 Hz to 8500 Hz.
2673
8.8
2674
2675
2676
2677
2678
2679
2680
2681
2682
2683
Echo frequency response and TCLW are traditional means of evaluating echo in telephones (and also networks). For
an alternative measure which may overcome certain shortcomings in this method, please see clause 8.9.
Echo frequency response
Echo frequency response is the ratio of the voltage output at the send electrical test point (SETP) to the voltage input
at the receive electrical test point (RETP), expressed in dB. Echo response in dB, HE(f), is given by Equation
8.15Equation 8.15Equation 8.8. The inverse of this response is echo path loss, which may be used to calculate
TCLW, the weighted terminal coupling loss, according to ITU-T Recommendation G.122 (1993) Annex B, Clause
B.4 (trapezoidal rule).
H E ( f )  20 log
2684
2685
2686
2687
2688
2689
2690
2691
2692
2693
2694
2695
2696
2697
2698
2699
2700
2701
2702
2703
2704
2705
2706
2707
2708
2709
2710
2711
2712
Overall distortion
G SETP ( f )
G RETP ( f )
in dBV / V
Equation 8.1515816
where:
GSETP(f) is the rms spectrum at SETP
GRETP(f) is the rms spectrum at RETP
In some cases, frequency response calculation may be performed with cross-spectrum or related techniques.
Justification for such techniques shall be given in the test report. See clause G.1 for more information.
Echo shall be measured under the following two conditions:
a) Receiver placed on the same ear simulator used for receive frequency response measurements (8.4.1).
b) Handset or headset is suspended in an anechoic chamber at least 500 mm from any reflecting objects.
Echo may also be measured under the following two conditions:
c)
Receiver and microphone facing a hard, smooth surface free of any object for 500 mm. A headset is placed
on the surface as if it was put down briefly by a user.
d) In the reference corner of Figure 7Figure 7Figure 7 (5.6.1), with the receiver placed 250 mm from the
corner.
Telephone sets with adjustable receive volume controls shall be tested at the reference receive volume control
setting.
The recommended test signal for this test is the composite source signal, with a white spectrum for the noise part
(CSS, see F.7.1). The recommended test signal level is –12.2 dBV (-10 dBm0). This level results in a relatively good
signal to noise ratio for the measurement. The crest factor of CSS can be less than 10 dB, allowing more headroom
than artificial voices. For devices that incorporate non-linear processes, additional measurements using signal levels
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
64
IEEE P269/D25 October 2004
2713
2714
2715
2716
2717
2718
of –8.2 dBV (–6 dBm0) and –18.2 dBV (–16 dBm0) may be performed. The measurement and calibration shall be
determined duri
n
gt
h
e“
On
”por
t
i
on
soft
h
es
i
g
n
a
l
.
2719
8.9
2720
2721
2722
2723
2724
2725
2726
2727
2728
2729
2730
2731
2732
2733
2734
2735
2736
2737
2738
2739
2740
2741
2742
2743
2744
2745
2746
2747
2748
2749
2750
2751
2752
2753
2754
2755
2756
2757
2758
Temporally weighted terminal coupling loss (TCLT) is an alternative measure of echo which may overcome some
shortcomings of echo frequency response and TCLW , the traditional means of evaluating echo in telephones (and
also networks). There can be two problems with echo frequency response and TCLW:
2759
8.10
2760
2761
2762
2763
2764
The stability measurement is the same as echo (8.8) except the test signal is a sinewave at an input level greater than
or equal to –12.2 dBV (-10 dBm0) and less than or equal to –2.2 dBV (0 dBm0), at one-twelfth octave intervals (or
R40) for frequencies from 200 Hz to 4 kHz. Stability loss is the maximum value of the inverse of echo (Equation
8.15Equation 8.15Equation 8.8). The measurement is performed under all four physical configurations specified in
8.8.
For the narrow band codec, the stimulus and analysis bandwidth is 100 to 3400 Hz. For the wideband codec, the
stimulus and analysis bandwidth is typically 100 to 6800 Hz.
Temporally weighted terminal coupling loss
a)
If the echo is low enough in level, the signal-to-noise ratio of the measurement can be poor or even
negative. In such a case, it may not be clear if a measured result is echo or noise.
b) The results obtained may not correlate well with perception, particularly if the echo comes in bursts.
The temporally weighted terminal coupling loss method is described in Annex O, and an algorithm is given in
Annex P. Several results are available from this method, but long-term temporally weighted terminal coupling loss,
single talk (LTCLT ) is recommended for single-number description of telephone echo. LTCLT is comparable to
TCLW, except that it incorporates psychoacoustic factors and separates echo from noise.
The test signal is real speech (F.6.3), with pauses edited so they are less than 20ms. Synthesized real speech (F.6.2)
may also be suitable, but has not yet been validated. The test level is –18.2 dBV, measured during the active
portions of the speech signal.
The test signal is applied at the RETP for 30 seconds so that the telephone reaches its steady state. No signal other
than the acoustic return from the receiver is applied to the microphone.
Record the electrical signals at RETP and SETP for the next 1 minute. Align the RETP and SETP signals in time by
adding delay equal to EPD (echo path delay) to the RETP signal. LTCLTis the difference (in dB) between the signal
level at RETP and SETP calculated using the algorithm in Annex P.
LTCLT shall be measured under the following two conditions:
a) Receiver placed on the same ear simulator used for receive frequency response measurements (8.4.1).
b) Handset or headset is suspended in an anechoic chamber, at least 500 mm from any reflecting objects.
LTCLT may also be measured under the following two conditions:
c)
Receiver and microphone facing a hard, smooth surface free of any object for 500 mm. A headset is
placed on the surface as if it was put down briefly by a user.
d) In the reference corner of Figure 7Figure 7Figure 7 (5.6.1), with the receiver placed 250 mm from the
corner.
Telephone sets with adjustable receive volume controls shall be tested at the reference receive volume control
setting.
Stability loss
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
65
IEEE P269/D25 October 2004
2765
2766
2767
2768
2769
2770
2771
For the narrow band codec, the stimulus and analysis bandwidth is 100 to 3400 Hz. For the wideband codec, the
stimulus and analysis bandwidth is typically 100 to 6800 Hz.
During the measurements, the operator should monitor the telephone for any sign of howling, whistling or other
signs of instability.
2772
8.11
Convergence time
2773
2774
2775
2776
2777
2778
2779
2780
2781
2782
2783
2784
2785
Some devices may have a nonlinear process, such as an echo canceller, to improve TCLw.. The convergence time
is a measure of how fast the full attenuation of the echo signal is achieved.
2786
8.12
2787
2788
2789
2790
2791
2792
2793
2794
2795
2796
2797
2798
2799
2800
2801
2802
2803
2804
2805
Discontinuous speech transmission (DTX) is often featured as a voice/speech activity detector (VAD/SAD) that
detects when the speech path is idle in a particular direction. The system will then mute the speech path, allowing
additional bandwidth for data traffic.
2806
8.12.1
2807
2808
2809
2810
The receive comfort noise of a digital telephone is the short-term average background noise level measured at the
output of the telephone receiver, with the digital telephone receiving either a silence indication packet from the
transmitting telephone, or no packets from the transmitting telephone, for some non-transient period of time.
2811
8.12.2
2812
2813
Telephone sets with adjustable receive levels shall be adjusted as close as possible to the reference receive volume
control setting. Use the same ear simulator and positioning which was used for receive measurements (8.4).
To measure convergence time, reset the device to a nominal state by initiating a new call in a quiet environment of
less than 30 dBA. Trigger a time capture with the onset of the input signal at RETP for a duration of 1 second.
Capture the trigger signal at RETP and the return signal at SETP. The convergence time is taken from the onset of
the trigger signal at RETP to where 90% of full echo path loss is achieved at SETP. If the canceller does not appear
to converge inside of 1 second, a longer time capture may be needed.
CSS at –12.2 dBV is the preferred test signal for this measurement. (F.7.1)
For the narrow band codec, the stimulus and analysis bandwidth is 100 to 3400 Hz. For the wideband codec, the
stimulus and analysis bandwidth is typically 100 to 6800 Hz.
Discontinuous speech transmission
DTX may cause noise pumping, and both front end speech and trailing speech clipping, especially if the device has
its own VAD feature working in tandem with a network DTX.
If the device has its own VAD feature, this can be characterized by measuring comfort noise matching, speech
detection switching time, and hangover switching time. Techniques to characterize switching times, can be found in
IEEE Std. 1329, clause 10.
Comfort noise matching is a comparison of background or network noise levels heard during active speech
transmission, and inserted replacement noise once the speech path is discontinued.
The comfort noise level introduced to replace the actual background noise should roughly match the loudness as
perceived by the user of the original background noise. This level matching is subjectively asymmetric, in that there
is more likely to be annoyance in the comfort noise loudness being greater than the original noise than in being less
than the original.
General
Measurement method
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
66
IEEE P269/D25 October 2004
2814
2815
2816
2817
2818
2819
2820
2821
2822
2823
2824
2825
2826
2827
2828
2829
2830
2831
2832
2833
2834
2835
2836
2837
For the narrow band codec, the stimulus and analysis bandwidth is 100 to 3400 Hz. For the wideband codec, the
stimulus and analysis bandwidth is typically 100 to 6800 Hz.
With both VAD disabled at the transmitting source and comfort noise generation on the receiving unit under test
turned off, a band-limited white noise test signal should be sent from the transmitting end such that the receive noise
l
e
v
e
lme
a
s
u
r
e
da
tt
h
er
e
c
e
i
v
i
ngt
e
l
e
ph
on
ei
s48dBA.Th
i
st
e
s
ts
i
g
n
a
l
,a
tt
h
i
sl
e
v
e
l
,wi
l
lbea
s
s
i
gn
e
dt
h
el
e
v
e
lof‘
N
dB’a
sac
a
l
i
br
a
t
e
dpoi
n
tf
ort
h
epu
r
pos
eoft
h
ec
omf
or
tn
oi
s
et
e
s
t
,s
i
n
c
ei
tma
y be generated either as an acoustic
s
i
gn
a
la
ta‘
g
ol
de
n’t
r
a
ns
mi
t
t
i
ngt
e
l
e
ph
on
e(
a
n
dme
a
s
u
r
e
di
ndBA)
,ori
n
j
e
c
t
e
ddi
g
i
t
a
l
l
y(
a
ndme
a
s
u
r
e
di
ndBm0p)
.
If it is not possible to disable the VAD, then a band-limited white noise signal at –62.2 dBV (–60 dBm0) is input at
RETP with a –12.2 dBV (-10 dBm0), 1 kHz tone. The injected noise level is measured at the receiver, with the 1
kHz tone filtered out. Remove the 1k Hz tone, and, once the device has discontinued the speech path, measure the
generated comfort noise.
Th
ef
ol
l
owi
ngt
e
s
ts
e
qu
e
n
c
emus
tbef
ol
l
owe
df
ora
l
lc
a
l
i
br
a
t
e
dt
e
s
tn
oi
s
el
e
v
e
l
sof‘
M dB’whi
c
hwi
l
lr
a
n
g
ef
r
omN10 to N+10 dB.
a) The echo canceller at both ends should be disabled
b) 10 seconds of silence (or idle code) is inserted at the transmitting point
c) Band-limited white noise of level M dB is inserted at the transmitting point for 130 seconds
d) During the final 10 seconds of level M noise insertion, the acoustic noise level at the receive will be
measured
e) Steps b)-d) are repeated for varying M in 1 dB gradations
2838
8.13
Maximum acoustic output
2839
2840
2841
2842
2843
2844
2845
2846
2847
2848
2849
2850
2851
The testing methods provided in this clause only cover the application of in-band signals, but the same sound
pressure limits may apply if ringing signals appear in the handset or headset receiver with the telephone set in offhook conditions. See Annex N for a discussion of maximum pressure limits.
2852
8.13.1
2853
2854
2855
2856
2857
2858
2859
2860
2861
2862
2863
2864
2865
2866
The maximum acoustic pressure is the maximum steady state sound pressure emitted from a receiver. The
measurement shall be made with real-time filter analysis (RTA) in 1/12 octave bands, described in G.3. The
detector shall be set to rms fast, which is a 250ms effective averaging time (equivalent to a 125ms time constant).
The detector shall be set to hold the maximum level achieved in each band during the entire sweep.
Maximum acoustic output measurements shall be made on the same ear simulator and with the same positioning and
force as used for receive frequency response measurements. For handsets measured on HATS, an additional
measurement with a force of 13N is required. See 5.3.2 for handsets, 5.3.3 for headsets. Telephone sets with
adjustable receive volume controls shall be adjusted to the maximum setting.
Acoustic output can be referenced to the ERP, DRP, free field (0 degrees elevation and azimuth), or to a diffuse
field, as required by the appropriate safety standard. This may require that measurements made at one reference
point be translated to the required reference point. A filter may be required. See Annex C.
Maximum acoustic pressure (long duration)
Additional consideration should be given to the acoustic pressure caused by tones, other audio signals or long
duration, high amplitude electrical signals applied to power, network, or auxiliary leads of the digital telephone.
For digital telephones, the long duration acoustic pressure shall be determined by applying digital codes to the
receive input. This may be performed by using an analogue test set to drive a reference codec or by use of a digital
code generator. If a set other than a G.711 type set is to be tested, then an analogue codec should be used. The
analog level shall be set to switch between the maximum positive and the maximum negative values for the
reference codec. The switching rate shall sweep through the range of 100 Hz to 3400 Hz for narrowband and 100 Hz
to 6800 Hz for wideband.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
67
IEEE P269/D25 October 2004
2867
2868
2869
2870
2871
2872
2873
2874
2875
If a G.711 type of set is to be tested, a digital generator may be used. In this case, the codes shall be switched
between the maximum positive and the maximum negative values, defined in ITU-T Recommendation G.711 (viz.
+3.17 dBm0 for mu-law coding and +3.14 dBm0 for A-law coding). The switching rate shall sweep through a range
of 100 Hz to 3400 Hz for narrowband and 100 Hz to 6800 Hz for wideband.
The sweep time shall be at least 90 seconds. A sweep time should be selected that provides consistent results with
no underestimation. That is, the result should be within 0.5 dB at all frequencies for a test period ± 30 seconds.
2876
8.13.2
Peak acoustic pressure (short duration)
2877
2878
2879
2880
2881
2882
2883
2884
2885
2886
2887
2888
2889
2890
2891
2892
2893
2894
2895
2896
2897
2898
2899
2900
2901
2902
2903
2904
2905
2906
The peak acoustic pressure is the maximum unweighted peak sound pressure emitted from a telephone receiver.
The stimulus for this test is a series of very short sweeps applied at RETP. The short sweeps are to avoid activating
any long-term non-linear processes, such as AGC, that may be operating in the device. The measurement shall be
ma
dea
tt
h
ee
a
rs
i
mu
l
a
t
orwi
t
ha
nunwe
i
gh
t
e
d“
pe
a
kh
ol
d”l
e
v
e
lde
t
e
c
t
or having a rise time equal to or less than 50
µs.
Additional consideration should be given to the peak acoustic pressure caused by tones or short duration, high
amplitude electrical pulses applied to power, network, or auxiliary leads of the digital telephone.
For digital telephones, the short duration acoustic pressure shall be determined by applying digital codes to the
receive input. This may be performed by using an analog test set to drive a reference codec, or by use of a digital
code generator. If a set other than a G.711 type set is to be tested, then an analog codec should be used. The analog
level shall be set to switch between the maximum positive and the maximum negative values for the reference
codec. The switching rate shall sweep through the range of 100 Hz to 3400 Hz for narrowband and 100 Hz to 6800
Hz for wideband.
If a G.711 type of set is to be tested, a digital generator may be used. In this case the codes shall be switched
between the maximum positive and the maximum negative values, defined in ITU-T Recommendation G.711 (viz.
+3.17 dBm0 for mu-law coding and +3.14 dBm0 for A-law coding). The switching rate shall sweep through a range
of 100 Hz to 3400 Hz for narrowband and 100 Hz to 6800 Hz for wideband.
The duration of the ON codes shall be a number of complete cycles approximating but not exceeding 500 ms. The
ON codes must be followed by a quiet interval of at least 500 ms before repeating the codes, as shown in
Figure 18
Figure 18
Figure 18.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
68
IEEE P269/D25 October 2004
250 on 500 ms
500 ms
maximum positive
digital word
maximum negative
digital word
2907
2908
2909
2910
2911
2912
2913
Figure 18 - On/Off Time for Short Duration Peak Acoustic Pressure
NOTE –It is advisable to repeat some tests more than one time, to ensure that the protection system is not damaged.
2914
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
69
IEEE P269/D25 October 2004
2914
9
Test Procedures for Analog 4-wire Handsets and Headsets
2915
9.1
General
2916
2917
2918
2919
2920
2921
2922
2923
2924
2925
2926
2927
2928
2929
2930
2931
2932
2933
2934
2935
2936
2937
Procedures are given in this clause for measurement of send and receive performance characteristics of handsets and
headsets tested as 4-wire devices, which are not connected to a complete telephone. Parameters include frequency
response, noise, input-output linearity, distortion, ac impedance, and dc resistance. In addition, procedures are given
for measuring echo frequency response and maximum acoustic output.
2938
9.1.1
2939
2940
2941
2942
2943
2944
2945
2946
2947
2948
2949
2950
2951
2952
2953
2954
2955
2956
2957
In general, multiple test signals and stimulus levels should be used to ensure the handset or headset is characterized
in realistic, stable and well-defined states. This is especially the case for devices with non-linear processes such as
compression or voice activated switching (VOX) circuitry, etc. See Annex F & Annex G for further information on
test signals and analysis methods.
2958
9.1.2
2959
2960
2961
2962
The measurement shall be performed using the same format as was used for calibration. Format examples are 1/N
octave bandwidth analysis, constant bandwidth analysis and R-series preferred frequencies. Measurement
bandwidth shall be the same as or less than that which was used for calibration. Measurement resolution shall be the
same as or coarser than that which was used for calibration. The actual bandwidth used shall be stated.
Loudness ratings (RLR and SLR) should not be used for 4-wire handsets and headsets as they are only defined for
complete telephone systems. It is possible to calculate loudness ratings for handsets and headsets, but the results can
only be used to compare similar devices since they are not generally meaningful. In this case the numbers shall be
r
e
f
e
r
r
e
dt
oa
s“
r
e
l
a
t
i
v
eRLR”or“
r
e
l
a
t
i
v
eSLR”
.
The handset or headset shall be connected to the appropriate test circuit(s) described in clause 9.2. Other test
circuits may be used for specific applications. Because 4-wire devices are affected by changes in voltage, current
and impedance, the measurements should be made over the conditions that are expected in actual use. Records
should be kept of the measurement conditions.
The measured frequency responses shall be presented as decibels relative to one pascal per volt [dB (Pa/V)] for
receive, decibels relative to one volt per pascal [dB (V/Pa)] for send, and decibels relative to one volt per volt [dB
(V/V)] for echo. The stimulus level and signal type shall be reported for each test.
The calibration procedures described in clause 6 shall be carried out before making any measurements. The
acoustical test environment shall meet the specifications given in clause 5.5.
Choice of test signals and levels
The standard test signal for all handsets and headsets consists of artificial voices defined in ITU-T Recommendation
P.50. See (F.6.1.1) for details.
Sinusoidal test signals may be used for testing handsets or headsets if it can be shown that they do not have
adaptive, nonlinear or dynamic signal processing (e.g. compressors, AGC, voice activity detection, adaptive echo
cancellers, etc.). Such evidence must be given in the test report if sinusoidal test signals are used.
Other test signals may be used when it can be shown that they produce results consistent with actual use. They also
may be necessary for some specific purposes as discussed in relevant places within this standard.
The measurements in this clause shall be performed at the standard test level for send specified in 6.7.2, and at the
receive stimulus level determined by the procedure in 9.3.2.
Measurement bandwidth and resolution
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
70
IEEE P269/D25 October 2004
2963
2964
2965
2966
In general, the test signals and analysis methods in this standard cover a frequency range from approximately 100 to
8500 Hz. The exact range depends on the analysis method, and the test signal (see G.6 and G.7)
2967
9.1.3
2968
2969
2970
2971
Choose the ear simulator, mouth simulator and test position according to clauses 5.1, 5.2 and 5.3. This equipment
shall be used for all tests described in clause 9, unless otherwise specified. The ear simulator, mouth simulator, and
test position used shall be stated.
2972
9.1.4
2973
2974
2975
2976
2977
2978
2979
2980
I
ft
h
eh
a
n
ds
e
torh
e
a
ds
e
ti
se
q
u
i
ppe
dwi
t
hat
on
ec
on
t
r
ol
,t
het
on
ec
on
t
r
ols
h
a
l
lbes
e
tt
ot
h
ema
n
uf
a
c
t
u
r
e
r
’
sde
f
a
u
l
t
setting. This is the default tone control adjustment that shall be used for all measurements.
2981
9.1.5
2982
2983
2984
2985
2986
All measurements shall be done at the default receive volume control setting (9.3.2) and default send gain
adjustment (9.4.1). These default settings for handsets and headsets are defined differently than the reference
receive volume control setting for complete telephones. A range of control settings may also be used where
appropriate, such as minimum and maximum.
2987
9.2
2988
2989
2990
2991
2992
The test circuits are terminated into an load greater than 100 k. This termination is the send electrical test point
(SETP) for measuring send output signals. This same termination is also the receive electrical test point (RETP) for
applying receive input signals. Note, other terminating loads may be substituted as defined by applicable
performance specifications.
Choice of ear and mouth simulators and test position
Tone control setting
If no default setting is defined by the manufacturer, the tone control shall be set so that the frequency response is as
close as possible to the center of the required frequency response template. The tone control shall be set before
setting the volume control. If the tone and volume controls interact, an iterative process for setting these controls
may be necessary.
Default receive volume control and send gain adjustment
Handset and headset test circuits
R1
Microphone
C
R2
+
V
2993
2994
2995
2996
2997
Figure 19 Electret Microphone Test Circuit
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
71
Send
Electrical
Test Point
(SETP)
IEEE P269/D25 October 2004
Microphone
2998
2999
3000
3001
Send
Electrical
Test Point
(SETP)
R2
Figure 20 Dynamic Microphone Test Circuit
C
R1
+
V
Microphone
R2
D
3002
3003
3004
3005
3006
3007
3008
3009
3010
3011
3012
3013
3014
3015
3016
3017
3018
3019
3020
Send
Electrical
Test Point
(SETP)
L
Figure 21 Carbon Microphone Test Circuit
In the microphone test circuits, Figure 19Figure 19Figure 19, Figure 20Figure 20Figure 20 and Figure 21Figure
21Figure 21, the values for voltage V, capacitance C, resistances R1 and R2, inductance L and diode D should
simulate the range of operating parameters of the headset or handset interface. These values are intended to provide
support for both DC and AC characteristics.
The effective load impedance provided by these test circuits shall be equal to the range of operating impedances of
the headset or handset interface. For electret microphones, the effective load impedance ZL = (R1 x R2) / (R1 + R2).
This assumes C is large enough so that its impedance is small compared to R1 and R2 at the lowest frequency tested.
In many cases, R2 is infinite, so the effective load impedance ZL = R1.
In the case of a completely self-powered microphone system, or a microphone system powered by its intended host,
the circuit of Figure 20Figure 20Figure 20 may be used. The microphone should be connected to its intended host
or suitable simulation.
ZS
(Z TERM )
3021
3022
3023
3024
3025
3026
3027
3028
Receive
Electrical
Test Point
(RETP)
Z S Ohm s
Receiver or
Receiver System
Figure 22 Receiver Test Circuit (ZTERM 100 kohm, used for calibration only)
The effective impedance ZS in the receiver test circuit of
Figure 22
Figure 22
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
72
IEEE P269/D25 October 2004
3029
3030
3031
3032
3033
3034
3035
3036
Figure 22 shall be equal to the specified nominal impedance of the receiver under test. ZS should take into account
both the output impedance of the signal generator and any other added impedances.
In case of a receiver system which needs to be powered, the headset or handset should be connected to its intended
host or suitable simulation.
The circuit of Figure 23Figure 23Figure 23 may be used for measurement of DC characteristics.
A
Microphone or
Receiver
3037
3038
3039
3040
3041
DC
Am m eter
R
DC
Voltm eter
V
+
V
Figure 23 DC Characteristics Test Circuit
3042
9.3
Receive
3043
9.3.1
General
3044
3045
3046
3047
3048
3049
Receive characteristics of handsets and headsets are measured with the receiver sound port terminated in the
appropriate ear simulator, as defined in Clause 5.
3050
9.3.2
3051
3052
3053
3054
3055
3056
3057
3058
3059
3060
3061
3062
3063
3064
3065
3066
3067
3068
I
fah
e
a
ds
e
torh
a
n
ds
e
ti
se
qu
i
ppe
dwi
t
har
e
c
e
i
v
ev
ol
umec
on
t
r
ol
,i
ts
h
a
l
lbes
e
tt
ot
h
ema
nu
f
a
c
t
u
r
e
r
’
sde
f
a
u
l
ts
e
t
t
i
ng
.
For frequency response measurements, LRETP shall be adjusted so that LERP = –14 dBPa.
Receive measurements should be taken with the handset or headset driven from a source equivalent to the interface
circuitry as specified in Clause 9.2.
Receive volume control adjustment
If no default setting is defined by the manufacturer, the following procedure shall be followed to determine the
default receive volume control setting, using the test signal chosen for subsequent receive measurements. If a sine
wave signal is used, the frequency shall be 1 kHz:
a)
Set the volume control to maximum. Adjust LRETP so that LERP = –14 dBPa. Record this level and call
it LMAX.
b) Set the volume control to minimum. Adjust LRETP so that LERP = –14 dBPa. If this is not possible,
move the control up slightly. Record this level and call it LMIN.
c)
Calculate the halfway point in dB between LMIN and LMAX, and call it LMID. Set LRETP to LMID. This
value of LRETP shall be used for frequency response measurements. Adjust the volume control so that
LERP = –14 dBPa. This is the default receive volume control setting which shall be used for all
measurements unless otherwise specified.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
73
IEEE P269/D25 October 2004
3069
9.3.3
Receive frequency response
3070
3071
3072
3073
Receive frequency response is the ratio of sound pressure measured in the ear simulator, referred to the Ear
Reference Point (ERP), to voltage input at the receive electrical test point (RETP), which is expressed in decibels.
The receive frequency response in dB, HR(f), is given by Equation 9.1Equation 9.1Equation 9.1.
H R ( f )  20 log
3074
3075
3076
3077
3078
3079
3080
3081
3082
3083
G ERP ( f )
G RETP ( f )
in dBPa / V
Equation 9.117
where:
GERP(f) is the rms spectrum at ERP
GRETP(f) is the rms spectrum at RETP
In some cases, frequency response calculation may be performed with cross-spectrum or related techniques.
Justification for such techniques shall be given in the test report. See clause G.1 for more information.
3084
9.3.4
Receive noise
3085
3086
3087
3088
3089
3090
3091
3092
3093
3094
3095
Receive noise is internally generated audio frequency noise present at the handset or headset receiver when no
stimulus is applied. The receiver shall be coupled to the ear simulator with the RETP terminated and with no signal
input. The handset or headset microphone should be isolated from sound input and mechanical disturbances that
would cause significant error. Measure the acoustic output signal, referred to the ERP, from 25100 to 8,500 Hz,
averaging over a minimum period of 5 seconds. Receive noise should be measured with the send mute feature both
“
on
”a
n
d“
of
f
.
”
3096
9.3.5
3097
3098
3099
3100
3101
3102
3103
3104
3105
3106
Receive narrow-band noise, including single frequency interference (SFI), is an impairment that can be perceived as
a tone relative to the overall weighted noise level. This test measures the weighted noise level characteristics in
narrow bands of not more than 31 Hz maximum, from 100 25 to 8,500 Hz. These levels can then be compared to the
receive noise (9.3.4).
3107
9.3.6
3108
3109
3110
3111
3112
3113
3114
3115
3116
Receive linearity is a measure of how the frequency response changes with input level.
The receive noise level is measured with A-weighting in dBA. The measurement may be implemented directly using
an A-weighting filter, or by using single-channel FFT with Hann windowing or real-time spectrum analysis,
followed by an A-weighted power summation.
Receive narrow-band noise
The receiver shall be coupled to the ear simulator with the RETP terminated and with no signal input. Measure the
A-weighted receive noise level, referred to the ERP, using a selective voltmeter, or a spectrum analyzer with an
effective bandwidth of not more than 31 Hz, over the frequency range of 100 25 to 8,500 Hz, averaging over a
minimum period of 5 seconds. If FFT analysis is use
d,t
h
e
n“
Fl
a
tTop”wi
n
dowi
ngs
h
a
l
lbee
mpl
oy
e
d.
Receive linearity
The test consists of measuring the receive frequency response as specified in Clause 9.3.3 and applying the
procedures described in Annex I. Linearity shall be measured using the same test method and stimulus type used to
measure frequency response.
The default receive volume and tone control setting shall be used. If artificial voices or another wideband test signal
are used, the test shall be performed in 1/3 octave bands. If sine wave signals are used, they shall be applied at the
R10 frequencies from 200 through 5000 Hz.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
74
IEEE P269/D25 October 2004
3117
3118
3119
3120
3121
For any test signal, the reference stimulus level is LMID, as determined according to the procedure in 9.3.2. The test
shall be performed at 7 levels, from LMID –15 dB to LMID + 15 dB, in 5 dB intervals. Smaller intervals and/or a
wider range of levels may also be used.
3122
9.3.7
3123
3124
3125
3126
3127
3128
3129
3130
3131
3132
3133
3134
The preferred distortion measurement method is receive signal-to-distortion-and-noise ratio (SDN), measured using
narrow-band pseudo-random noise as the stimulus. See A.1.1J.3 for details of the method.
3135
9.3.8
3136
3137
3138
Mount the receiver to the appropriate ear simulator. Connect an impedance bridge to the receive circuitry described
in clause 9.2. Measure the impedance at each frequency of interest.
3139
9.3.9
3140
3141
3142
3143
The resistance of the receive circuit should be obtained by the current-voltage method shown in Figure 23Figure
23Figure 23. This measurement may be taken for various dc supply voltages, but use caution to avoid damaging the
receive circuitry.
3144
9.4
3145
9.4.1
3146
3147
3148
3149
3150
3151
3152
3153
3154
3155
3156
3157
3158
3159
3160
3161
3162
3163
I
fah
e
a
ds
e
torh
a
n
ds
e
ti
se
qui
ppe
dwi
t
has
e
n
dg
a
i
na
dj
u
s
t
me
nt
,t
h
eg
a
i
nc
on
t
r
ols
h
a
l
lbes
e
tt
ot
h
ema
nuf
a
c
t
u
r
e
r
’
s
default setting. This is the default send gain control adjustment that shall be used for send frequency response
measurements.
Receive distortion
Receive distortion is measured at ERP using an input level of LMID, as determined according to the procedure in
9.3.2. Other input levels should be tested covering a range of at least from –30 to +5 dBV and above, if necessary,
until obvious clipping or limiting occurs. Measurements should also be made over a range of frequencies within the
telephone band, such as the ISO R10 preferred frequencies. For higher input levels, verify that distortion of the test
system is less than 1% THD.
For information about THD and other distortion measurement methods and test signals, and the conditions under
which they may be used, see Annex J. Different distortion measurement methods are likely to give different results.
AC impedance
DC resistance
Send
Send gain control adjustment
If no default setting is defined by the manufacturer, the following procedure shall be followed to determine the
default send gain control setting, using the test signal chosen for subsequent send measurements. If a sinewave
signal is used, the frequency shall be 1 kHz:
a)
Set the gain adjustment to maximum. Set LMRP to –4.7 dBPa, then measure LSETP. Record this level
and call it LMAX.
b) Set the gain adjustment to minimum. Set LMRP to –4.7 dBPa, then measure LSETP. Record this level
and call it LMIN. If this is not possible, move the control up slightly, then repeat the procedure.
c)
Calculate the halfway point in dB between LMAX and LMIN, and call it LMID. Set LMRP to –4.7 dBPa,
then measure LSETP Adjust the send gain control so that LSETP = LMID. This is the default send gain
control adjustment that shall be used for all measurements unless otherwise specified.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
75
IEEE P269/D25 October 2004
3164
9.4.2
Send frequency response
3165
3166
3167
3168
Send frequency response is the ratio of voltage output at the Send Electrical Test Point (SETP) to the sound pressure
at the Mouth Reference Point (MRP) , which is expressed in decibels. The send frequency response in dB, HS(f), is
given by Equation 9.3Equation 9.3Equation 9.2.
H S ( f )  20 log
3169
3170
3171
3172
3173
3174
3175
3176
3177
3178
3179
3180
G SETP ( f )
G MRP ( f )
in dBV / Pa
Equation 9.33218
where:
GSETP(f) is the rms spectrum at SETP
GMRP(f) is the rms spectrum at MRP.
In some cases, frequency response calculation may be performed with cross-spectrum or related techniques.
Justification for such techniques shall be given in the test report. See clause G.1 for more information.
3181
9.4.3
Send noise
3182
3183
3184
3185
3186
3187
3188
3189
3190
3191
3192
3193
3194
3195
3196
Send noise is internally generated audio frequency noise present at the microphone terminals or circuitry with no
stimulus applied. Measure the electrical output signal at SETP, averaging over a minimum period of 5 seconds. The
handset or headset microphone should be isolated from sound input and mechanical disturbances that would cause
s
i
gn
i
f
i
c
a
n
te
r
r
or
.Se
n
dn
oi
s
es
h
ou
l
dbeme
a
s
u
r
e
dwi
t
ht
h
emu
t
ef
e
a
t
u
r
ebot
h“
on
”a
n
d“
of
f
.
”
3197
9.4.4
3198
3199
3200
3201
3202
3203
3204
3205
3206
3207
Send narrow-band noise, including single frequency interference (SFI), is an impairment that can be perceived as a
tone relative to the overall weighted noise level. This test measures the A-weighted noise level characteristics in
narrow bands, of not more than 31 Hz maximum, from 100 25 –8500 Hz.
3208
9.4.5
3209
3210
Send linearity is a measure of how the frequency response changes with input level.
Send overall noise shall be measured with psophometric weighting, and reported in units of dBV(p). It shall also be
measured with A-weighting and reported in units of dBV(A). Measurements in dBV(p) and dBV(A) are generally
not the same, and they may not be correlated. Units of dBmp or dBm(A) can then be calculated based on the method
described in Annex T.
Psophometric measurements are made from 10025-6000 Hz, while A-weighted measurements are made from
10025-8,500 Hz.. These measurements can be made directly using a psophometrically weighted or A-weighted noise
meter with the correct terminating impedance. The measurement may also be implemented using a single-channel
FFT with Hann windowing, or a real-time spectrum analysis, followed by a weighted power summation.
Send narrow-band noise
The handset or headset should be isolated from sound input and mechanical disturbances that would cause
significant error. Measure the A-weighted noise level across R2 with a selective voltmeter, or a spectrum analyzer
with an effective bandwidth of not more than 31 Hz, over the frequency range of 100 25 to 8500 Hz, averaging over
a minimum period of 5 seconds.I
fFFTa
n
a
l
y
s
i
si
sus
e
d,t
h
e
n“
Fl
a
tTop”wi
n
dowi
ngs
h
a
l
lbee
mpl
oy
e
d.
The same procedure may be applied, but with psophometric weighting, if specified by a performance standard.
Send linearity
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
76
IEEE P269/D25 October 2004
3211
3212
3213
3214
3215
3216
3217
3218
3219
3220
3221
3222
3223
The test consists of measuring the send frequency response as specified in Clause 9.4.2 and applying the procedures
described in Annex I. Linearity shall be measured using the same test method and stimulus type used to measure
frequency response.
3224
9.4.6
3225
3226
3227
3228
3229
3230
3231
3232
3233
3234
3235
The preferred distortion measurement method is send signal-to-distortion-and-noise ratio (SDN), measured using
narrow-band pseudo-random noise as the stimulus. See A.1.1J.3 for details of the method.
3236
9.4.7
3237
3238
3239
3240
3241
3242
3243
3244
3245
3246
3247
3248
Send frequency response in a diffuse field is a measure of how much of the noise in the room where a telephone is
being used is transmitted to the network. It is the ratio of voltage output at the Send Electrical Test Point (SETP) to
the sound pressure at the Diffuse Field Test Point (DFTP, see 5.5.3), which is expressed in decibels. The diffuse
field send frequency response in dB, HSD(f), is given by equation Equation 9.5Equation 9.5Equation 9.3.
If artificial voices or another wideband stimulus are used, the test shall be performed at 7 levels, from –34.7 dBPA
to –4.7 dBPa, in 5 dB intervals, measured in 1/3 octave bands. Smaller intervals and/or a wider range of levels may
also be used. The reference stimulus level is –4.7 dBPa. These levels take into account the high crest factor of
artificial voices, which approaches 23 dB.
If sine wave signals are used, they shall be applied at the R10 frequencies from 200 through 5000 Hz for 7 levels,
from –24.7 to +5.3 dBPa, in 5 dB intervals. Smaller intervals and/or a wider range of levels may also be used. The
reference stimulus level is –9.7 dBPa.
Send distortion is measured using the standard input level of –4.7 dBPa. Other input levels should be tested covering
a range from –30 to +10 dBPa. Measurements should also be made over a range of frequencies within the telephone
band, such as the ISO R10 preferred frequencies. For higher input levels, verify that distortion of the test system is
less than 2% THD.
For information about THD and other distortion measurement methods and test signals, and the conditions under
which they may be used, see Annex J. Different distortion measurement methods are likely to give different results.
Send frequency response in a diffuse field
The diffuse field send frequency response may be sensitive to both the level and type of signal used. This
measurement may be performed in 1/3 octave resolution.
During the measurement, the mouth simulator is present but not active, with the MRP is located at the DFTP. The
mouth simulator is not present during calibration.
H SD ( f )  20 log
3249
3250
3251
3252
3253
3254
3255
3256
3257
Send distortion
G SETP ( f )
G DFTP ( f )
in dBV / Pa
Equation 9.55319
where:
GSETP(f) is the rms spectrum at SETP
GDFTP(f) is the rms spectrum at DFTP
The cross-spectrum method is not recommended.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
77
IEEE P269/D25 October 2004
3258
9.4.8
Send signal-to-noise ratio
3259
3260
3261
Send signal-to-noise ratio is a measure of the desired speech transmission relative to unwanted noise in the room
whe
r
et
h
et
a
l
k
e
r
’
sph
on
ei
sus
e
d.Se
eAnnex K.
3262
9.4.9
3263
3264
3265
3266
3267
3268
Connect the headset or handset receiver according to Figure 19Figure 19Figure 19, Figure 20Figure 20Figure 20 or
Figure 21Figure 21Figure 21. Temporarily disconnect R2 and measure the electrical output for an input level
representing the magnitude of typical voice signals. Connect R2 and adjust its resistance to cause a 6 dB drop in
output voltage level. This resistance value is the magnitude of the impedance of the microphone circuit, which may
include R1, at each frequency of interest.
3269
9.4.10
3270
3271
3272
3273
3274
The resistance of a dynamic type microphone can be measured directly. The resistance of electret and carbon type
microphones should be obtained from the current-voltage characteristics. This measurement may be taken for
various dc supply voltages, but use caution to avoid damaging the microphone circuitry. The microphone should be
isolated from sound input and mechanical disturbances for these measurements.
3275
9.5
3276
3277
3278
3279
3280
3281
3282
3283
3284
3285
3286
3287
3288
3289
Echo frequency response is the ratio of the voltage output at the send electrical test point (SETP) to the voltage input
at the receive electrical test point (RETP), expressed in dB. Echo response in dB, HE(f), is given by Equation
9.7Equation 9.7Equation 9.4. The inverse of this response is echo path loss.
AC impedance
DC resistance
Echo frequency response
Echo path loss may be used to calculate TCLW, the weighted terminal coupling loss, according to ITU-T
Recommendation G.122 (1993) Annex B, Clause B.4 (trapezoidal rule). For handsets and headsets this calculation
s
h
a
l
lbel
a
be
l
e
da
s“
r
e
l
a
t
i
v
eTCLW,
”s
i
n
c
et
r
u
eTCLW is defined only for complete telephones.
TCLW may be normalized to nominal RLR and SLR target specifications, corrected from relative SLR and relative
RLR.I
ts
h
a
l
lt
h
e
nbel
a
be
l
e
d“
n
or
ma
l
i
z
e
dTCLW,
”a
n
dt
h
eme
t
h
odofn
or
ma
l
i
z
a
t
i
ons
hall be stated.
H E ( f )  20 log
3290
3291
3292
3293
3294
3295
3296
3297
3298
3299
3300
3301
3302
3303
3304
3305
G SETP ( f )
G RETP ( f )
in dBV / V
Equation 9.77420
where:
GSETP(f) is the rms spectrum at SETP
GRETP(f) is the rms spectrum at RETP
In some cases, frequency response calculation may be performed with cross-spectrum or related techniques.
Justification for such techniques shall be given in the test report. See clause G.1 for more information.
Echo frequency response shall be measured under the following two conditions:
a.
b.
Receiver placed on the same ear simulator used for receive measurements (9.3).
Handset or headset is suspended in the anechoic chamber at least 500 mm from any reflecting objects.
Echo frequency response may be measured under the following two conditions:
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
78
IEEE P269/D25 October 2004
3306
3307
3308
3309
3310
3311
3312
3313
3314
3315
3316
3317
c.
d.
Receiver and microphone facing a hard, smooth surface free of any object for 500 mm. Handset
receiver and microphone facing down. Headset is placed on the surface as if it was put down briefly
by a user.
In the reference corner of Figure 7Figure 7Figure 7 in clause 5.6.1, with the receiver placed 250 mm
from the corner.
The recommended test signal for this test is the composite source signal, with a white spectrum for the noise part
(CSS, see F.7.1). The recommended test signal level is LMID + 6 dB. (This level is intended to result in a test roughly
comparable to an echo test with the same handset or headset installed in a complete telephone. It also results in an
improved signal to noise ratio for the measurement.)
3318
9.6
Maximum acoustic output
3319
3320
3321
3322
3323
3324
3325
3326
3327
3328
3329
3330
3331
The testing methods provided in this clause only cover the application of in-band signals, but the same sound
pressure limits may apply if ringing signals appear in the handset or headset receiver while the telephone set is offhook. See Annex N for a discussion of maximum pressure limits.
3332
9.6.1
3333
3334
3335
3336
3337
3338
3339
3340
3341
3342
3343
3344
3345
3346
The maximum acoustic pressure is the maximum steady state sound pressure emitted from a receiver. The stimulus
for this test is a slow logarithmic sine sweep applied at RETP from 100 to 8500 Hz. The measurement shall be made
with real-time filter analysis (RTA) in 1/12 octave bands, described in G.3. The detector shall be set to rms fast,
which is a 250ms effective averaging time (equivalent to a 125ms time constant). The detector shall be set to hold
the maximum level achieved in each band during the entire sweep.
3347
9.6.2
3348
3349
3350
3351
3352
3353
3354
3355
3356
The peak acoustic pressure is the maximum unweighted peak sound pressure emitted from a receiver. The stimulus
for this test is a surge applied to the receive terminals of the handset or headset. The measurement shall be made at
t
h
ee
a
rs
i
mu
l
a
t
orwi
t
ha
nunwe
i
g
ht
e
d“
pe
a
kh
ol
d”l
e
v
e
lde
t
e
c
t
or
,wi
t
har
i
s
et
i
mee
qu
a
lt
oorl
e
s
st
h
a
n50µs
.
3357
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
79
maximum acoustic output measurements shall be made on the same ear simulator and with the same positioning and
force as used for receive frequency response measurements. For handsets measured on HATS, an additional
measurement with a force of 13N is required. See 5.1, as well as 5.3.2 for handsets, and 5.3.3 for headsets. Handset
and headsets with adjustable receive volume controls shall be adjusted to the maximum setting.
Acoustic output can be referenced to the ERP, DRP, free field (0 degrees elevation and azimuth), or to a diffuse
field, as required by the appropriate safety standard. This may require measurements made at one reference point be
translated to the required reference point. A filter may be required. See Annex C.
Maximum acoustic pressure (long duration)
The test shall be performed under the two following conditions:
a) 10 dBV with a source impedance less than 10
b) 15 dBV with a source impedance of 150
The sweep time shall be at least 90 seconds. A sweep time should be selected that provides consistent results with
no underestimation. That is, the result should be within 0.5 dB at all frequencies for a test period ± 30 seconds.
Peak acoustic pressure (short duration)
Connect the positive terminal of the surge generator (Error! Reference source not found.) to the positive terminal
of the receive circuitry. Measure the peak pressure in the ear simulator while operating the surge generator. An
os
c
i
l
l
os
c
opeoras
ou
n
dl
e
v
e
lme
t
e
r
,h
a
v
i
n
ga
nunwe
i
g
h
t
e
d“
pe
a
kh
ol
d”s
e
t
t
i
ngi
sus
e
dt
oma
k
et
h
eme
a
s
u
r
e
me
n
t
.
Reverse the connection and repeat.
IEEE P269/D25 October 2004
3357
Annex A
3358
3359
(normative)
3360
3361
Ear Simulators with Flexible Pinnas and Positioning Devices
3362
Note to committee: Update with respect to new positioning methods.
3363
A.1 General characteristics of the ear simulators
3364
3365
3366
3367
3368
3369
3370
3371
3372
3373
3374
3375
3376
Type 3.3 and Type 3.4 ear simulators have a soft pinna which deforms when the receiver is pressed against it. The
resulting leak depends on force or position, as well as the exact shape of the receiver. The relationship between
position and force will vary depending on the shape of the receiver.
3377
A.2 Differences between the two ear simulators
3378
3379
3380
3381
3382
3383
3384
3385
3386
3387
3388
3389
3390
3391
3392
3393
3394
3395
3396
3397
3398
3399
3400
3401
The Type 3.3 ear simulator is shaped like a real human ear, while Type 3.4 ear simulator has a simplified shape.
The change in force or position of the receiver against the ear simulator will cause the acoustic leak to vary. The
leak will generally introduce variations in the frequency response, especially at the lower frequency range of the
receiver, just as it does on a human ear.
The variation of leak with force or position is often not linear, especially at very low forces (2N or less) or very high
forces (13N or more).
Both ears have acoustical characteristics similar to the average human adult ear.
The measured results obtained by the Type 3.3 and Type 3.4 ear simulators may differ:
a)
The receiver position, or force applied, may result in leaks that are slightly different. In order to
achieve a similar leak on the two different ear simulators with a handset, different forces may have to
be applied.
b) The acoustical input impedance of the two simulators is not identical. In general, the impedance of the
Type 3.3 ear simulator is slightly higher than that of the Type 3.4 ear simulator. For measurements
with similar leakage, the effect is that the receive loudness rating calculated from measurement on a
Type 3.3 ear simulator could be one to two dB lower (louder) than that obtained from the Type 3.4 ear
simulator.
Regardless of these differences, both the Type 3.3 and Type 3.4 ear simulators are generally the most realistic way
to measure handsets and headsets in a way that relates to the actual experience of real listeners. The choice between
the Type 3.3 and Type 3.4 ear simulator is up to the user, as long as the restrictions in clause 5.1.1 are complied
with. However, measurements using the two simulators cannot be expected to be exactly equal.
The recommendations in this standard for using the Type 3.3 and Type 3.4 ear simulators reflect the currently
available equipment. When new or revised simulators become available, their use should be carefully considered in
view of the principles expressed in this standard as well as the information and recommendations provided by the
equipment manufacturer.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
80
IEEE P269/D25 October 2004
3402
A.3 Handset Positioning devices
3403
3404
3405
3406
3407
3408
3409
3410
3411
3412
3413
3414
3415
3416
3417
3418
3419
3420
3421
3422
3423
3424
3425
3426
3427
3428
3429
3430
3431
3432
3433
3434
3435
3436
3437
3438
3439
In principle, positioning of a handset on the Type 3.3 or Type 3.4 ear simulator can be specified either by position
relative to the ERP or by the applied force. The two are related, since greater applied force results in moving the
receiver inward toward the center of the head. However, the relationship between applied force and position may be
nonlinear.
PThe positioning devices currently available for the Type 3.3 ear simulator can hold the receiver by position relative
to the ERP or by force on the pinna. Positioning relative to ERP is typically very repeatable. The recommended
procedure is to begin by placing the receiver in the positioning device without contacting the pinna, then gradually
moving the receiver inward so as to increase the force, stopping at the target force or position.
When using the positioning device currently available for the Type 3.3 ear simulator, placement by force is typically
somewhat less repeatable than placement by position relative to the ERP. In addition, there can be a large difference
in pinna deformation at a given force reading depending on whether the force has been increased from a low value
to arrive at the target, or decreased from a high value. In other words, whether the receiver has been positioned from
outside the ERP and moved in toward the center of the head, or the reverse. The recommended procedure is to begin
by placing the receiver in the positioning device without contacting the pinna, and to gradually move the receiver
inward so as to increase the force, stopping at the target force. 6 newtons is the default target force.
The positioning device currently available for the Type 3.4 ear simulator can hold the receiver by force on the pinna.
The positioning by force is typically very repeatable. There can be a difference in pinna deformation at a given force
reading depending on whether the force has been increased from a low value to arrive at the target, or decreased
from a high value. In other words, whether the receiver has been positioned from outside the ERP and moved in
toward the center of the head, or the reverse. The recommended procedure is to begin by placing the receiver in the
positioning device without contacting the pinna, and to gradually move the receiver inward so as to increase the
force, stopping at the target force. 6 newtons is the default target force.
When using the positioning device currently available for the Type 3.4 ear simulator, it is not possible to hold the
receiver by position relative to the ERP.
The positioning recommendations in this standard for positioning handsets or headsets on the Type 3.3 or Type 3.4
ear simulators reflect the currently available equipment. When new or revised positioning devices become
available, their use should be carefully considered in view of the principles expressed in this standard as well as the
information and recommendations provided by the equipment manufacturer.
3440
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
81
IEEE P269/D25 October 2004
3440
Annex B
3441
3442
(normative)
3443
3444
Alternative Ear Simulators, Mouth Simulator and Test Fixture
3445
3446
Note to committee: Update with respect to new positioning methods.
3447
B.1 Alternative Ear Simulators
3448
3449
3450
3451
3452
3453
3454
3455
3456
3457
3458
3459
3460
3461
3462
3463
3464
3465
3466
3467
3468
3469
The following specialized ear simulators may be used as alternates if the applicable performance specification
requires or allows it, and if the following application requirements are met:
a)
The Type 1 ear simulator may be used for large, supra-aural or supra-concha, hard-cap, conically
symmetrical receivers, which naturally seal to the simulator rim, in the band of 100-4,000 Hz. (The
frequency range may be extended to 8500 Hz, but only if the performance specification requires it.
However, the accuracy or relevance of the results in this extended range are not assured.) These
receivers should also be tested in a realistic unsealed condition using the Type 3.3 or Type 3.4 ear
simulator as specified in this sub-clause.
b) The Type 2 ear simulator may be used for sealing or non-sealing receivers that are inserted into the ear
canal.
c)
The Type 3.1 ear simulator may be used for intra-concha receivers designed for sitting on the bottom
of the concha cavity.
d) The Type 3.2 ear simulator with a high- or low-grade leak may be used for large, supra-aural or supraconcha, hard-cap, receivers, which naturally seal to the simulator rim, in the band of 100-8,000 Hz.
The low leak is intended for receivers that are pressed firmly to the ear, while the high leak is intended
for loosely coupled receivers.
Ear simulator type numbers are defined in ITU-T Recommendation P.57.
3470
3471
3472
3473
3474
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
82
IEEE P269/D25 October 2004
3474
3475
3476
3477
3478
3479
Ear simulator recommendations are summarized in
Table B. 1
Table B. 1
Table B. 1:
Ear Simulator Type
(ITU-T Recommendation P.57)
Receiver Type
Type 1

 Supra-aural, hard cap

 Supra-concha, hard cap
Type 2

 Insert, sealed & unsealed
Type 3.1

 Intra-concha
Concha bottom simulator
Type 3.2
Simplified pinna simulator,
Low- or High- grade leak
Type 3.3
Anatomically-shaped soft pinna
(Recommended choice)
Type 3.4
Simplified soft pinna

 Supra-aural, hard cap

 Supra-concha, hard cap








Intra-concha
Supra-aural
Supra-concha
Circum-aural








Intra-concha
Supra-aural
Supra-concha
Circum-aural
Application Notes












5-10 N force (handset only)
Must naturally seal to rim
No sealing putty allowed
100-4000 Hz bandwidth
100-8,500 Hz* bandwidth
Headsets only










5-10 N force
Must naturally seal to rim
No sealing putty allowed
100-8,500 Hz* bandwidth
6 N force (handset only)

 100-8,500 Hz* bandwidth

 6 N force (handset only)

 100-8,500 Hz* bandwidth
(Recommended choice)
3480
3481
3482
3483
3484
3485
3486
3487
3488
3489
3490
3491
3492
3493
3494
*8,500 Hz. is the nominal upper frequency. See clause G.6 for details.
Table B. 1 Ear simulator usage
The Type 1 ear simulator measures at the ear reference point (ERP), while all the other ear simulators measure at the
eardrum reference point (DRP). Measurements collected at the DRP shall be translated to the ERP. This is done
because receive and sidetone specifications are referenced to the ERP. It also permits comparison of measurements
made on the various type ear simulators.
For Types 2, 3.1, 3.3 and 3.4 ear simulators, DRP to ERP translation shall be performed according to Annex C.
For measurements of receive or talker sidetone using the Type 1 ear simulator, a leakage correction is often applied
to the loudness rating calculation. Follow the applicable performance standard for the correction and how to apply
it. The leakage correction is not applied to the frequency response.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
83
IEEE P269/D25 October 2004
3495
B.2 Alternative Mouth Simulator
3496
B.2.1
3497
3498
3499
3500
3501
3502
3503
3504
3505
3506
3507
3508
3509
3510
3511
3512
3513
3514
3515
When an alternative ear simulator described in clause B.1 is used, an alternative mouth simulator may be used.
When measurements are being made exclusively in the send direction, an alternative mouth simulator may also be
used. The mouth simulator recommended in Clause 5.2 is usually installed in a HATS, but the alternative ear
simulators generally cannot be mounted to a HATS. Alternative mouth simulators and ear simulators are generally
installed on a test head. The alternative mouth shall comply with the specification given in ITU-T Recommendation
P.51, whereas the mouth recommended in Clause 5.2 must comply with ITU-T Recommendation P.58. There are
minor differences between these specifications, so there may be small differences between the simulators.
3516
B.2.2
3517
3518
3519
3520
3521
3522
3523
3524
3525
3526
3527
3528
3529
3530
In principle, a 6.25 mm free-field microphone should be used to calibrate the mouth simulator.
General
The alternative mouth is suitable for measurements at or in front of the lip plane only. Traditionally, it has been used
for measuring corded telephone handsets.
Neither ITU-T Recommendation P.51 nor ITU-T Recommendation P.58 defines a sound field behind the lip plane.
However, practical experience has shown that the sound field distribution in the region between the HATS mouth
and ear closely approximates the sound field around a real human head up to at least 4 kHz. The region extends from
beyond the lip plane to the base of the rubber ear and equal to or greater than 5 mm above the surface of the HATS
cheek. This makes HATS suitable for testing headsets, cordless and cellular phones, handsfree phones, and
traditional corded handsets. The sound field approximation may extend in frequency range as well as to other
regions around HATS, but these have not yet been verified.
Calibration of Alternative Mouth Simulator
In practice, the mouth simulator may be calibrated at the MRP using a 12.5 mm free-field microphone oriented at 0
degrees to the mouth axis with the center of the protection grid at the MRP (Figure B. 1Figure B. 1Figure B. 1). The
calibrated frequency response of the microphone should be taken into account. Subtract 0.6 dB from the
measurement to give the actual sound pressure at the MRP. This compensates for the acoustic center of the
microphone being slightly in front of the protection grid. The method is valid over the entire frequency range
covered in this standard.
An alternate method is to calibrate at the MRP using a 12.5 mm pressure microphone oriented at 90 degrees to the
mouth axis with the center of the protection grid at the MRP. The calibrated frequency response of the microphone
should be taken into account. This method is valid only to 5 kHz.
25 mm
Free Field
Microphone
MRP
3531
3532
3533
3534
Lip Ring of
Mouth Simulator
Figure B. 1 On-Axis Calibration of mouth simulator
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
84
IEEE P269/D25 October 2004
3535
3536
3537
3538
3539
To calibrate the mouth, measure GMRP(f), the spectrum at the MRP. Adjust the mouth equalization to meet the target
spectrum for the signal being used at a total sound pressure of -4.7 dBPa. This spectrum is used to calculate the
send, sidetone and overall frequency responses.
3540
B.3 Alternative Test Fixture
3541
3542
3543
3544
3545
3546
3547
3548
3549
The test fixture shall implement the HATS position defined in ITU-T Recommendation P.64, Annex E. The HATS
position may be implemented on a standard test head.
The LRGP position was specified in previous editions of this standard. Send frequency response measurements
made on ordinary telephones from 300-3400 Hz are expected to give practically identical results, whether obtained
with LRGP or the HATS position. Systematic differences of about 1-2 dB in send frequency response
measurements on pressure gradient microphones have to be expected from the upwards tilted speaking direction of
about 19 degrees using the LRGP position. See ITU-T Recommendation P.64 (1999), Annex F.
3550
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
85
IEEE P269/D25 October 2004
3550
Annex C
3551
3552
(normative)
3553
3554
3555
3556
3557
3558
3559
3560
3561
3562
3563
3564
3565
3566
3567
3568
3569
3570
3571
3572
3573
3574
3575
3576
3577
3578
3579
3580
3581
3582
3583
3584
3585
3586
3587
3588
3589
DRP TO ERP and Related Translations
All ear simulators except Type 1 are made with the measurement microphone in a position corresponding to the
eardrum, so measurements are made at the drum reference point (DRP). For telephony measurements, the ear
reference point (ERP) is used for loudness rating calculations and to maintain comparability to historical
measurements. The measurements collected at the DRP are therefore generally translated to the ERP, or to another
suitable acoustical terminal, depending on the application.
For all measurements, the translation may be implemented by using a minimum-phase filter based on one of the
tables or other transfer functions referred to in this annex. The magnitude of the filter response shall match the
transfer function within a tolerance of 2 dB. A tolerance of 1 dB is preferred.
A filter shall be used for measurements of peak acoustic pressure. (Peak measurements should be made on the actual
waveform at the desired acoustic terminal. Both the magnitude and phase of the transfer function is necessary to best
preserve the waveshape for a proper measurement of its peak value.)
For measurements made with any kind of spectrum analysis, the translation may be performed with post-processing
using one of the tables or other transfer functions referred to in this annex Measurement examples include
frequency response, noise, linearity and distortion. These tables may also be used for frequency response
measurements made with sine waves, if only the fundamental or total response is included.
A filter is recommended for measurements of distortion. However, for measurements of distortion using a sine or
narrowband stimulus, translation tables for post-processing may be constructed based on one of the tables or other
transfer functions referred to in this annex. Separate tables are required for each harmonic or difference-frequency
distortion product, taking into account the frequency offset between the stimulus frequency and the frequency of the
distortion product.
The translations given or referred to in this annex may be interpolated to match the frequency format of the
measurement to which they are applied.
The DRP to ERP translation, SDE, must be added to the data measured at the DRP in order to translate to the ERP.
The effect is to remove a broad frequency response peak of about 10 dB in the region of 3000 Hz. The DRP to ERP
translation in this annex is from ITU-T Recommendation P.57 (1996) as shown in table C.1. For this standard, the
DRP to ERP correction curve is modified to extend to 25Hz. For table C.1, the correction below 92 Hz is zero.
Table C.2 is derived from table C.1, in ISO R40 format and the correction below 100 Hz for table C.2 is zero.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
86
IEEE P269/D25 October 2004
DRP to ERP Translation (SDE )
5
Amplitude (dB)
0
-5
-10
-15
-20
-25
100
1000
Frequency (Hz)
3590
3591
3592
3593
3594
3595
Figure C. 1 SDE at 1/12 Octave Filter Center Frequencies
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
87
10000
IEEE P269/D25 October 2004
3595
Frequency
(Hz)
92
97
103
109
115
122
130
137
145
154
163
173
183
193
205
218
230
244
259
274
3596
3597
3598
3599
3600
SDE
(dB)
0.1
0.0
0.0
0.0
0.0
0.0
0.0
0.0
0.0
0.0
0.0
-0.1
-0.1
0.0
0.1
0.0
-0.1
-0.2
-0.3
-0.3
Frequency
(Hz)
290
307
325
345
365
387
410
434
460
487
516
546
579
613
649
688
729
772
818
866
SDE
(dB)
-0.3
-0.2
-0.2
-0.2
-0.4
-0.5
-0.4
-0.6
-0.3
-0.7
-0.6
-0.6
-0.6
-0.6
-0.8
-0.8
-1.0
-1.1
-1.1
-1.2
Frequency
(Hz)
917
972
1029
1090
1155
1223
1296
1372
1454
1540
1631
1728
1830
1939
2054
2175
2304
2441
2585
2738
SDE
(dB)
-1.3
-1.4
-1.8
-2.0
-2.3
-2.4
-2.6
-3.1
-3.3
-3.9
-4.4
-4.8
-5.3
-6.0
-6.9
-7.5
-8.1
-9.1
-9.5
-10.4
Frequency
(Hz)
2901
3073
3255
3447
3652
3868
4097
4340
4597
4870
5158
5464
5788
6131
6494
6879
7286
7718
8175
8659
Table C. 1 SDE at 1/12 Octave Filter Center Frequencies
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
88
SDE
(dB)
-11.0
-10.5
-10.2
-9.1
-8.0
-6.9
-5.8
-5.0
-4.2
-3.3
-2.7
-2.4
-2.4
-2.5
-3.3
-4.5
-5.9
-9.0
-14.2
-20.7
IEEE P269/D25 October 2004
3600
3601
3602
3603
3604
3605
3606
3607
3608
3609
3610
Frequency
(Hz)
100
106
112
118
125
132
140
150
160
170
180
190
200
212
224
236
250
265
280
SDE
(dB)
0.0
0.0
0.0
0.0
0.0
0.0
0.0
0.0
0.0
-0.1
-0.1
0.0
0.1
0.0
-0.1
-0.1
-0.2
-0.3
-0.3
Frequency
(Hz)
335
355
375
400
425
450
475
500
530
560
600
630
670
710
750
800
850
900
950
SDE
(dB)
-0.2
-0.3
-0.4
-0.4
-0.5
-0.4
-0.5
-0.7
-0.6
-0.6
-0.6
-0.7
-0.8
-0.9
-1.1
-1.1
-1.2
-1.3
-1.4
Frequency
(Hz)
1120
1180
1250
1320
1400
1500
1600
1700
1800
1900
2000
2120
2240
2360
2500
2650
2800
3000
3150
SDE
(dB)
-2.1
-2.3
-2.5
-2.8
-3.2
-3.6
-4.2
-4.7
-5.2
-5.8
-6.5
-7.2
-7.8
-8.5
-9.3
-9.9
-10.6
-10.7
-10.4
300
-0.2
1000
-1.6
3350
-9.6
315
-0.2
1060
-1.9
3550
-8.5
Frequency
(Hz)
3750
4000
4250
4500
4750
5000
5300
5600
6000
6300
6700
7100
7500
8000
8500
9000
9500
10000
SDE
(dB)
-7.5
-6.3
-5.3
-4.5
-3.7
-3.0
-2.6
-2.4
-2.5
-2.9
-4.0
-5.3
-7.5
-12.2
-18.6
*
*
*
Table C. 2 SDE at ISO R40 Preferred Frequencies
Translation from DRP to free field at 0 degrees azimuth and 0 degrees elevation, or diffuse field, or any other
similar acoustical terminal shall be made using the transfer function supplied by the manufacturer of the ear
simulator, if available. Alternatively, the transfer functions specified in ITU-T Recommendation P.58 may be used.
Transfer functions with resolution of at least 1/12 octave or R40 format shall be used if available. Report the transfer
function used.
3611
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
89
IEEE P269/D25 October 2004
3611
Annex D
3612
3613
(normative)
3614
3615
3616
3617
3618
3619
3620
3621
3622
3623
3624
3625
3626
3627
3628
3629
3630
3631
3632
3633
3634
3635
3636
3637
3638
3639
3640
3641
3642
3643
Conditioning for Carbon Transmitters
The orientation of a carbon transmitter during test, and the treatment it receives immediately prior to a test, can have
a significant influence on test results. Conditioning should be applied before making any measurement, and the
measurement should start within 10 s after conditioning. Because of the wide possible variation in handset
geometries and test fixtures, general guidelines are given for conditioning, rather than detailed specifications. For
tests between different locations, it is recommended that identical procedures, as nearly as possible, be used to
reduce differences and to make results comparable.
For best reproducibility, automatic mechanical conditioning should be used. Connect the telephone set terminals as
required to the feed circuit and the appropriate terminating load. Turn the feed current on. After 5 s, condition the
microphone by rotating it smoothly. Rotation is made so that the plane of the granule bed moves through an arc of at
least 180° and back. The rotation procedure is repeated twice with the handset coming to rest in the test position
without jarring the carbon granules. The time of each rotation cycle should lie within the range of 2–12 s.
The final handset position should be 45 degrees face-up for all transmission testing, i.e. send, receive, sidetone,
overall.
NOTE: The axis of rotation for conditioning may be arbitrarily located with respect to the transmitter axis. In
practice, one orientation that provides the proper motion for many existing telephone sets is to have the axis of
rotation coaxial with the axis of the mouth simulator.
The performances of existing types of handset receivers are independent of the position (vertical, horizontal face-up,
or down) of the handset, but carbon transmitter resistance may affect receiving output. In this case, the conditioning
procedure in this annex should be followed.
3644
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
90
IEEE P269/D25 October 2004
3644
Annex E
3645
3646
(normative)
3647
3648
3649
3650
3651
3652
3653
3654
3655
3656
3657
3658
3659
3660
3661
3662
3663
3664
3665
3666
Hoth Room Noise
Hoth noise can be described as acoustic random noise that has a power density spectrum corresponding to that
specified in Table E. 1Table E. 1Table E. 1. The spectrum of Hoth noise is designed to simulate typical ambient
room noise over time.
Table E. 1Table E. 1Table E. 1 below gives the spectrum density adjusted in level to produce a reading of 50 dBA.
Figure E. 1Figure E. 1Figure E. 1 shows a plot of this spectrum. The spectrum below is independent of level and
shifts equally for each 1/3 octave band.
Frequency (Hz)
Spectrum Density Bandwidth
(dB SPL/Hz)
_ƒ(
d
B)
100
125
160
200
250
315
400
500
630
800
1000
1250
1600
2000
2500
3150
4000
5000
6300
8000
32.4
30.9
29.1
27.6
26.0
24.4
22.7
21.1
19.5
17.8
16.2
14.6
12.9
11.3
9.6
7.8
5.4
2.6
-1.3
-6.6
13.5
14.7
15.7
16.5
17.6
18.7
19.7
20.6
21.7
22.7
23.5
24.7
25.7
26.5
27.6
28.7
29.7
30.6
31.7
32.7
10 log Total power in each
1/3 Octave Band
(dBSPL)
45.9
45.5
44.9
44.1
43.6
43.1
42.3
41.7
41.2
40.4
39.7
39.3
38.7
37.8
37.2
36.5
34.8
33.2
30.4
26.0
Tolerance (dB)
±3
±3
±3
±3
±3
±3
±3
±3
±3
±3
±3
±3
±3
±3
±3
±3
±3
±3
±3
±3
Table E. 1 Hoth noise parameters
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
91
IEEE P269/D25 October 2004
Spectrum Density (dBSPL/Hz)
Hoth Noise Spectrum Density Vs Frequency
35
30
25
20
15
10
5
0
-5
-10
100
3667
3668
3669
3670
3671
3672
3673
3674
3675
3676
3677
3678
3679
3680
3681
3682
1000
Frequency (Hz)
10000
Figure E. 1 Hoth noise spectrum
Typical Hoth noise levels range from 35 dBA to 65 dBA.
At low frequencies, sound levels are somewhat difficult to control due to both the size of the test chamber, and the
introduction of external noise (air-conditioning/heating etc.). The test chamber should be designed to minimize
undesirable low frequency sound levels.
For optimum ambient noise simulation in the test chamber, it is best to have a diffuse source for Hoth noise. This
can best be achieved by having somewhat reflective walls, and multiple sound sources. A compromise can be made
with either the room, or the number of sound sources.
3683
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
92
IEEE P269/D25 October 2004
3683
Annex F
3684
3685
(normative)
3686
3687
Test Signals
3688
3689
F.1 General
3690
3691
3692
3693
3694
3695
3696
3697
3698
3699
3700
3701
3702
3703
3704
3705
3706
3707
3708
3709
3710
3711
3712
3713
3714
3715
The test signal should place the telephone in a well-defined, reproducible state for the period of the measurement. It
should insure that the transfer function of the unit remains stable during the measurement period, and yet provide a
suitable signal for the specific measurement. The choice of the signal will be a balance between one that correctly
stimulates the processing algorithms in the telephone, and one that is suitable for the specific measurement.
3716
F.2 Classifications
3717
3718
3719
3720
3721
3722
3723
3724
3725
The various types of signals are divided into several groups, as discussed below. The classical measurement signals
can be separated into deterministic signals and continuous random signals. More complex random signals include
modulated random signals and speech-like signals that characterize human speech. Finally, there are compound
signals composed of two sources: one for biasing the unit into a stable state, and the other being the actual test signal
itself.
3726
F.3 Modulation types
3727
3728
3729
Several types of modulation may be applied to deterministic or random signals.
approximate the syllabic rhythm of real speech.
Unless otherwise stated, test signal levels are specified as long-term rms levels during at least one complete period
of the active part of the signal.
The long-term rms level of artificial voices (F.6.1.1), speech-like signals with pauses less than 20ms, and signals
with sinusoidal or pseudo-random modulation (F.3.2 & F.3.3 & F.6.1.3) shall be measured during at least one
complete cycle of the modulation pattern.
The level of signals with square-wave modulation (F.3.1) may be measured during the entire signal and then
c
or
r
e
c
t
e
dt
oa
c
c
ou
n
tf
ort
h
edut
yc
y
c
l
e
.Fore
x
a
mpl
e
,a250ms“
on
”pe
r
i
odf
ol
l
owe
dbya150ms“
of
f
”pe
r
i
odwou
l
d
have a duty cycle of 5/8, which corresponds to –2.04 dB. A long-term rms measurement of such a signal, including
the pauses, would underestimate the active level by 2.04 dB.
The level of random signals (F.5) shall be measured for long enough (large enough time-bandwidth product) to
insure that the measurement error (+/- one standard deviation) is 0.5 dB or less.
The level of speech–like signals with pauses (F.6.1.2, F.6.2, F.6.3) shall be measured with a speech voltmeter or
other algorithm which meets the specifications of ITU-T Recommendation P.56, Method B.
The level of compound signals (F.7) shall be measured according to the principles of this clause. The details vary
according to the exact signal. Clause F.7 offers some additional guidelines.
In addition to the signals described in Annex F, signals described in ITU-T Recommendation P.501 are also
recommended when they are appropriate.
This is done in order to
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
93
IEEE P269/D25 October 2004
3730
3731
3732
3733
3734
3735
Test signals may be modulated in various ways to correctly stimulate a telephone, depending on the signal
processing actually used in the phone. For example, a modulated noise signal is often an appropriate stimulus for a
send circuit with a noise-guard feature. In the presence of a continuous signal over a few hundred milliseconds in
duration, the noise-guard process reduces gain substantially. On the other hand, a continuous noise signal is often an
appropriate stimulus for a receive circuit with automatic gain control (AGC).
3736
F.3.1
3737
3738
3739
3740
3741
3742
3743
3744
3745
3746
Square wave modulation is an on-off pattern. The recommended pattern is 250 ms ON and 150 ms OFF, 10ms.
This pattern is common in many telephone testing methods. It is close to the modulation rate of real speech. Other
timing patterns may also be used.
3747
F.3.2
3748
3749
3750
Sine wave modulation may be used to produce a simple and smooth speech amplitude envelope. The recommended
rate is 4 Hz. Modulation depth should be at least 50%, but not so great as to introduce distortion.
3751
F.3.3
3752
3753
3754
3755
Pseudo-random modulation may be used to produce a relatively speech-like amplitude envelope. The modulation
spectrum should cover from approximately 1 to 10 Hz, with the center at approximately 4 Hz. The extremes of the
modulation spectrum should be rolled off gradually.
3756
F.4 Deterministic signals
3757
3758
3759
3760
Deterministic (periodic) signals can always be used to measure the frequency response of linear, time invariant
telephones. When modulated, they can be used to measure the response of telephones with many, but not all,
speech-processing features.
3761
F.4.1
3762
3763
3764
3765
3766
3767
In addition to use in measuring the frequency response of linear, time invariant telephones, sine waves are useful for
measurements of harmonic and difference-frequency distortion. This signal can be modulated by square wave, sine
wave, and pseudo-random signals.
3768
F.4.2
3769
3770
3771
3772
3773
3774
3775
A pseudo-random signal has a periodic structure in the time domain. In the frequency domain, almost any
magnitude and phase spectrum is possible. When used with FFT types of analysis, the period of the pseudo-random
signal is generally matched in length and synchronously triggered at the start of the analysis period. When used
with an MLS analyzer, the period of the MLS signal must be matched to the analysis period. This signal can be
modulated by square wave, sine wave and pseudo-r
a
n
doms
i
g
n
a
l
s
.I
fs
qu
a
r
ewa
v
emodul
a
t
i
oni
su
s
e
d,t
h
e“
on
”t
i
me
must correspond to one or more complete period(s) of the pseudo-random signal.
Square wave modulation
In some cases, a periodic pulse pattern of this type will not correctly activate the telephone circuit. In such cases, a
r
a
n
doml
yv
a
r
i
e
dpu
l
s
epa
t
t
e
r
nma
ybeus
e
d.Th
ea
v
e
r
a
g
e“
on
”a
n
d“
of
f
”t
i
me
ss
h
ou
l
da
ppr
ox
i
ma
t
e250msa
n
d150
ms respectively.
With this type of modulation, all measurements are t
obep
e
r
f
or
me
ddu
r
i
ngt
h
e“
on
”pa
r
toft
h
epa
t
t
e
r
n
.Forot
h
e
r
types of modulation, the signal is to be measured during the entire presentation time.
Sine wave modulation
Pseudo-random modulation
Sine wave
The target spectrum for sine wave signals is flat, which means equal amplitude at all frequencies.
Pseudo-random
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
94
IEEE P269/D25 October 2004
3776
The target spectrum for pseudo-random signals can be white (F.5.1), pink (F.5.2) or P.50 (F.5.3).
3777
F.5 Random signals
3778
3779
3780
3781
3782
3783
3784
3785
3786
Random signals can be described by their statistical characteristics, such as the long-term power spectral density and
probability density functions. These signals are not periodic, but are stationary as far as these statistical
characteristics are concerned. When measuring such signals, a sufficient number of averages should be taken to
obtain a given accuracy in estimating the long-term spectrum.
3787
F.5.1
3788
3789
3790
3791
3792
3793
White noise has a constant spectral density per hertz. The amplitude distribution is typically truncated Gaussian,
with a crest factor of 12 dB, 2 dB. This signal can be modulated by square wave, sine wave and pseudo-random
signals.
3794
F.5.2
3795
3796
3797
3798
3799
3800
3801
Pink noise has a power spectral density that decreases 3 dB per octave. The amplitude distribution is typically
truncated Gaussian, with a crest factor of 12 dB, 2 dB. This signal can be modulated by square wave, sine wave
and pseudo-random signals.
3802
F.5.3
3803
3804
3805
3806
3807
3808
3809
The spectrum of this signal is the same as artificial voices (F.6.1.1). The amplitude distribution is typically truncated
Gaussian, with a crest factor of 12 dB, 2 dB. This signal can be modulated by square wave, sine wave and pseudorandom signals.
3810
F.6 Speech-like signals
3811
3812
3813
3814
3815
3816
Speech-like signals include ITU-T Recommendation P.50 (1999) artificial voices, ITU-T Recommendation P.59
(1993) artificial conversational speech, simulated speech generator (SSG), as well as synthesized and real speech
signals. When long term averaging is used, these signals place the telephone in a well-defined reproducible state,
ensure that the transfer function of the unit remains stable, and provide a suitable signal for the specific
measurement.
3817
F.6.1
3818
3819
3820
Typical parameters of simulated speech include long-term average spectrum, short-term spectrum, instantaneous
amplitude distribution, speech waveform structure, and the syllabic envelope.
In practice, many practical noise generators produce pseudo-random signals, typically with a very long period. If
the period of such signals is very long compared to the analysis period, and if the analysis period is not correlated to
the generator period, then these signals can be considered random.
White noise
The target spectrum for white noise is flat, when analyzed in fixed bandwidths. When analyzed in constant
percentage bandwidths, this is equivalent to a spectrum with band levels rising at 3 dB per octave.
Pink noise
The target spectrum for pink noise is flat, when analyzed in constant percentage bandwidths. When analyzed in
fixed bandwidths, this is equivalent to a spectrum with band levels falling at 3 dB per octave.
P.50 noise
Th
et
a
r
g
e
ts
pe
c
t
r
um f
orP.
5
0n
oi
s
ei
st
h
ec
ol
umn“
Soun
dpr
e
s
s
u
r
el
e
v
e
l(
t
h
i
r
doc
t
a
v
e
)
”i
nTa
bl
e1ofI
TU-T
Recommendation P.50. The table can be used directly for the acoustic test spectrum at an overall level of–4.7 dBPa.
A constant can be added in all frequency bands to give other overall levels.
Simulated speech
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
95
IEEE P269/D25 October 2004
3821
3822
3823
3824
The target spectrum for s
i
mu
l
a
t
e
ds
pe
e
c
hi
st
h
ec
ol
umn“
Sou
n
dpr
e
s
s
u
r
el
e
v
e
l(
t
h
i
r
doc
t
a
v
e
)
”i
nTa
bl
e1ofI
TU-T
Recommendation P.50. The table can be used directly for the acoustic test spectrum at an overall level of–4.7 dBPa.
A constant can be added in all frequency bands to give other overall levels.
3825
F.6.1.1
3826
3827
3828
3829
3830
3831
3832
3833
3834
3835
3836
3837
3838
P.50 Artificial Voices
ITU-T Recommendation P.50 defines the temporal and spectral parameters for test signals which emulate the
characteristics of male and female speech. These artificial voices are continuous speech signals with a frequency
range of 89.1 Hz to 8919 Hz. See Clause F.10,“
Test signals published on CD-ROM,
”f
oron
es
ou
r
c
eoft
h
e
s
ea
n
d
other test signals.
Note: The P.50 signals published on the CD-ROM will have to be equalized to meet the target spectrum.
At least one complete segment of both male and female artificial voices should be used as the standard test signal.
The male and female artificial voices are each approximately 10.5 sec. long. The combined test signal should consist
of the male followed by the female artificial voices, resulting in a signal length of approximately 21 sec. No gap in
the combined test signal should exceed 100 ms.
3839
F.6.1.2
P.59 Artificial Conversational Speech.
3840
3841
3842
3843
3844
Artificial conversational speech is a test signal generated by inserting pauses in the continuous artificial voices
described by ITU-T Recommendation P.50 (1999). The on-off temporal characteristics of conversational speech are
defined in ITU-T Recommendation P.59 (1993). This test signal is useful for evaluating devices that are sensitive to
the on-off nature of conversational speech, in both single and double-talk modes.
3845
F.6.1.3
3846
3847
3848
3849
3850
3851
3852
To generate a signal approximating the amplitude distribution of speech, a main signal having a Gaussian
distribution is modulated by a specially tailored modulating signal, and the resultant signal is shaped to approximate
the long-term frequency spectrum of speech. See Annex Q for details of this signal. See Clause F.10,“
Test signals
published on CD-ROM,
”f
oron
es
ou
r
c
eoft
h
i
sa
n
dot
h
e
rt
e
s
ts
i
gn
a
l
s
.
3853
F.6.2
3854
3855
3856
3857
3858
3859
3860
Speech-like signals may be produced using a digital processing technique rather than applying one of the signal
sources described above. Conversational speech can be sampled, digitized, processed, and reproduced as
synthesized speech. It also may be created from complex multiple tones that simulate the talk-spurts, pauses, and
activity factors associated with speech characteristics.
3861
F.6.3
3862
3863
3864
3865
3866
3867
3868
Speech-like signals are not limited to signal sources or synthesized digital processing, but also may include real
speech signals. This is often done by recording conversational speech, preferably in a digital format, to avoid signal
degradation with use. These real speech recordings are then reproduced using a playback device as the signal
source. See Clause F.10,“
Test signals published on CD-ROM,
”f
oron
es
ou
r
c
eoft
h
e
s
ea
ndot
h
er test signals.
Simulated Speech Generator (SSG).
Note: The SSG signal published on the CD-ROM will have to be equalized to meet the target spectrum.
Synthesized speech
The target spectrum for synthesized speech is the original spectrum produced by the synthesis procedure.
Real speech
The target spectrum for real speech is the original spectrum of the recorded speech.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
96
IEEE P269/D25 October 2004
3869
F.7 Compound signals
3870
3871
3872
3873
3874
3875
3876
3877
3878
3879
3880
3881
3882
3883
3884
3885
The signals described above rely on one signal source to place the telephone in a well-defined reproducible state,
insure that the transfer function of the unit remains stable, and provide a suitable signal for the specific
me
a
s
u
r
e
me
n
t
.Bya
ppl
y
i
ngt
wos
i
gn
a
ls
ou
r
c
e
s
,on
ec
a
nbeu
s
e
ds
pe
c
i
f
i
c
a
l
l
yf
or“
bi
a
s
i
n
g
”t
h
eu
n
i
ti
nt
oas
t
a
bl
e
,
reproducible state, while the other is the actual test signal required for measurements. These compound signals
include those where the two sources are applied in sequence, and those where both sources are applied
simultaneously.
Compound test signals can provide extra test flexibility and solve problems which are difficult or impossible using
simple test signals. The bias signal can be a signal that, by itself, is unsuitable or very inconvenient for the actual
measurement. The measurement signal can be a signal that, by itself, is unsuitable as a bias signal.
If desired, the measurement signal can be presented so as not to have a substantial effect on the action of the bias
signal. This can be done by adjusting the temporal and/or level relationships between the two signals. The bias
signal can be changed to put the telephone in different states with minor or even no change in the measurement
signal.
3886
F.7.1
Sequential presentation
3887
3888
3889
3890
3891
3892
3893
3894
3895
3896
3897
3898
3899
3900
3901
3902
3903
3904
This class of test signals is characterized by the separation of the bias and analysis signals in time. The bias signal is
presented until the telephone is in a stable state. Once a stable state is reached, the appropriate analysis signal is
applied and a measurement is performed. The analysis should be completed while the telephone is still in its stable
state. The CSS is one example of this type of signal.
3905
F.7.2
3906
3907
3908
3909
3910
3911
3912
3913
3914
3915
This class of test signals is characterized by presentation of the bias and analysis signals at the same time. Some
conditioning of the telephone may be required before beginning the analysis. The bias and analysis signals must be
separable by the analysis method. A synchronous analysis method is usually required. The P.50 Burst with Sine
Sweep is one example of this type of signal.
3916
F.7.2.1
3917
3918
3919
This compound signal has two components, which are presented at the same time, but not synchronized with each
other. The bias signal is P.50 noise presented in bursts (see F.5.3 and F.3.1) The bias is intended to ensure that the
telephone is in a stable, well-defined operating state. The measurement signal (TDS sweep) is intended to ensure a
The Composite Source Signal (CSS) is a compound signal using a voiced signal to simulate the voice properties,
followed by a noise-like signal for measuring the transfer functions, and an inserted pause to provide amplitude
modulation. The noise-like signal has either a flat or speech shaped power density spectrum. It has the advantage of
short measurement periods and duplex operation where, using an uncorrelated double-talk signal, the test signals can
be applied from the talking and listening directions at the same time. See ITU-T Recommendation P.501 (1996) for
the definition of this signal. See Clause F.10,“
Test signals published on CD-ROM,
”f
oron
es
ou
r
c
eoft
h
i
sa
n
dother
test signals.
I
ft
h
es
i
gn
a
li
n
c
l
u
de
spa
u
s
e
s
,c
a
l
i
br
a
t
i
ona
n
dme
a
s
u
r
e
me
nt
sa
r
et
obepe
r
f
or
me
ddu
r
i
n
gt
h
e“
on
”pa
r
toft
h
epa
t
t
e
r
n
.
The target spectrum of the voiced part of this signal is defined in ITU-T Recommendation P.501. The noise part
may have various target spectra, according to the application. The target spectrum may be white noise (F.5.1), pink
noise (F.5.2) or P.50 noise (F.5.3).
Simultaneous presentation
The target spectrum of the total signal (bias plus analysis) is the same as the target spectrum of the bias signal.
The analysis part of the signal (for example, the TDS sweep) may have to be shaped to fulfill this requirement, but
there is no other requirement on the spectrum of the analysis part of the signal.
TDS Sweep with P.50 Noise Bursts.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
97
IEEE P269/D25 October 2004
3920
3921
3922
3923
3924
3925
3926
well-defined, reproducible measurement, which is especially well adapted to simulated free-field techniques. An
anechoic room is not necessary when using this signal. See Annex R for a detailed description of this signal.
3927
F.7.2.2
3928
3929
3930
3931
3932
3933
3934
Similar to F.7.2.1, except the bias is the continuous artificial voices signal defined in ITU-T Recommendation P.50
(1999). (See F.6.1.1).
3935
F.7.2.3
3936
3937
3938
3939
Similar to F.7.2.1, except the bias is real speech (F.6.3).
3940
F.7.2.4
3941
3942
3943
3944
Similar to F.7.2.1, except the bias is synthesized speech (F.6.2).
3945
F.7.2.5
3946
3947
3948
3949
3950
Similar to F.7.2.1, except the bias is white or pink random noise (F.5). Pseudorandom noise (F.4.2) with white or
pink spectrum is considered equivalent if the pseudorandom period is not correlated with the bias.
3951
F.7.2.6
3952
3953
3954
3955
3956
3957
3958
3959
3960
3961
3962
This compound signal has two components, which are presented at the same time, but not synchronized with each
other. The bias signal is P.50 noise presented in bursts (see F.5.3 and F.3.1) The bias is intended to insure that the
telephone is in a stable, well-defined operating state. The measurement signal (pseudorandom noise) is intended to
ensure a well-defined, reproducible measurement, which is especially well adapted to simulated free-field
techniques. An anechoic room is not necessary when using this signal.
3963
F.7.2.7
3964
3965
3966
Similar to F.7.2.6, except the bias is the continuous artificial voices signal defined in ITU-T Recommendation P.50
(1999). (See F.6.1.1).
Th
et
a
r
g
e
ts
pe
c
t
r
umi
st
h
ec
ol
umn“
Sou
n
dpr
e
s
s
u
r
el
e
v
e
l(
t
hi
r
doc
t
a
v
e
)
”i
nTa
bl
e1ofI
TU-T Recommendation P.50.
The table can be used directly for the acoustic test spectrum at an overall level of–4.7 dBPa. A constant can be
added in all frequency bands to give other overall levels.
TDS Sweep with P.50 Artificial Voices.
Th
et
a
r
g
e
ts
pe
c
t
r
umi
st
h
ec
ol
umn“
Sou
n
dpr
e
s
s
u
r
el
e
v
e
l(
t
hi
r
doc
t
a
v
e
)
”i
nTa
bl
e1ofI
TU-T Recommendation P.50.
The table can be used directly for the acoustic test spectrum at an overall level of–4.7 dBPa. A constant can be
added in all frequency bands to give other overall levels.
TDS Sweep with Real Speech
The target spectrum is the original spectrum of the recorded speech.
TDS Sweep with Synthesized Speech
The target spectrum is the original spectrum produced by the synthesis procedure.
TDS Sweep with Random or Pseudorandom Noise
The target spectrum is white or pink.
Pseudorandom Noise with P.50 Noise Bursts.
Th
et
a
r
g
e
ts
pe
c
t
r
umi
st
h
ec
ol
umn“
Sou
n
dpr
e
s
s
u
r
el
e
v
e
l(
t
hi
r
doc
t
a
v
e
)
”i
nTa
bl
e1ofI
TU-T Recommendation P.50.
The table can be used directly for the acoustic test spectrum at an overall level of–4.7 dBPa. A constant can be
added in all frequency bands to give other overall levels.
Pseudorandom Noise with P.50 Artificial Voices.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
98
IEEE P269/D25 October 2004
3967
3968
3969
3970
Th
et
a
r
g
e
ts
pe
c
t
r
umi
st
h
ec
ol
umn“
Sou
n
dpr
e
s
s
u
r
el
e
v
e
l(
t
hi
r
doc
t
a
v
e
)
”i
nTa
bl
e1ofI
TU-T Recommendation P.50.
The table can be used directly for the acoustic test spectrum at an overall level of–4.7 dBPa. A constant can be
added in all frequency bands to give other overall levels.
3971
F.7.2.8
3972
3973
3974
3975
Similar to F.7.2.6, except the bias is real speech (F.6.3).
3976
F.7.2.9
3977
3978
3979
3980
Similar to F.7.2.6, except the bias is synthesized speech (F.6.2).
3981
F.7.2.10 Pseudorandom Noise with Random or Pseudorandom Noise
3982
3983
3984
3985
3986
Similar to F.7.2.6, except the bias is white or pink random noise (F.5). Pseudorandom noise (F.4.2) with white or
pink spectrum is considered equivalent if the pseudorandom period is not correlated with the bias.
3987
F.7.2.11 Sine Wave with Notched Real Speech.
3988
3989
3990
3991
3992
A sine wave is the measurement signal and real speech is the bias signal. A notch filter removes a band of the
speech signal at the sine wave frequency.
3993
F.8 Test signal bandwidth
3994
3995
3996
3997
3998
3999
4000
4001
In general, the test signals and analysis methods in this standard cover a frequency range from approximately 100 to
8500 Hz. The exact range depends on the analysis method, and perhaps also the test signal (see G.6). The lower
limit is the practical lower limit of the mouth simulator, while the upper limit is determined by the range of the
DRP-to-ERP translation curve (Annex C). For digital phones, the exact range may also be determined by the codec.
Pseudorandom Noise with Real Speech
The target spectrum is the original spectrum of the recorded speech.
Pseudorandom Noise with Synthesized Speech
The target spectrum is the original spectrum produced by the synthesis procedure.
The target spectrum is white or pink.
The target spectrum is the original spectrum of the recorded speech.
Some signals, such as SSG (F.6.1.3), are defined only for a smaller bandwidth, and cannot be used outside their
defined range.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
99
IEEE P269/D25 October 2004
4001
F.9 Signal parameter summary
4002
4003
4004
4005
4006
Table F. 1Table F. 1Table F. 1 defines the bandwidth and maximum analysis interval for the various test signals
identified in Annex F. Signals may be analyzed with finer resolution if desired. These parameters shall be applied
to both the calibration and test procedures.
Document
Test Signal
Ref.
Maximum
Analysis
Interval
Alternative
Analysis Format
F.4.1
Sine Wave*
Flat
ISO R40 steps
1/12 Oct. steps
F.4.2
Pseudo-Random*
White, pink
or P.50
25 Hz bands
1/12 Oct. bands
F.5.1
White Noise*
White
25 Hz bands
1/12 Oct. bands
F.5.2
Pink Noise*
Pink
1/12 Oct. bands
25 Hz bands
F.6.1.1
P.50 Artificial Voices
P.50
1/12 Oct. bands
25 Hz bands
P.50
1/12 Oct. bands
25 Hz bands
P.50
1/12 Oct. bands
25 Hz bands
F.6.1.2
F.6.1.3
P.59 Artificial
Conversational Speech
Simulated Speech
Generator
F.6.2
Synthesized Speech
As synthesized
1/12 Oct. bands
25 Hz bands
F.6.3
Real Speech
As recorded
1/12 Oct. bands
25 Hz bands
F.7.1
Composite Source
Signal
See F.7.1
25 Hz bands
1/12 Oct. bands
TDS Sweep with Bias
Same as bias
50 Hz Bands
1/12 Oct. bands
Pseudorandom noise
with Bias
Same as bias
50 Hz Bands
1/12 Oct. bands
F.7.2.1F.7.2.5
F.7.2.6F.7.2.10
4007
4008
4009
4010
4011
4012
Target
Spectrum
* Modulation may be required depending on the application. See F.3.
Table F. 1Test signal parameters
4013
F.10 Test signals published on CD-ROM
4014
4015
4016
4017
The artificial voices according to ITU-T Recommendation P.50 , as well as a large speech database, is included on a
CD-ROM published as ITU-T Recommendation P.50, Appendix 1: Test signals. Other specialized signals,
including the composite source signal (CSS) are published on a CD-ROM included with ITU-T Recommendation
P.501
.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
100
IEEE P269/D25 October 2004
4018
4019
F.11 Signal and test method comparative summary
4020
4021
4022
Table F. 2Table F. 2Table F. 2 identifies the various test signals described previously. The corresponding test
methods and conditions are shown for each signal. The various method classifications are described in Annex G.
Signal Type
Pink Noise
Simulated Speech
Synthesized Speech
Real Speech
Sequential
Simultaneous
Random
Signal
White Noise
Deterministic
Signal
Pseudorandom
Test Method
Anechoic
Chamber
Compound
Needed?
Speech-Like Signal
Signal
Sine Wave
Sec.
Ref.
5.2.1
5.2.2
5.2.3
FFT/Cross Spectrum
Dual-Channel FFT
Single-Channel FFT
Max. Length Seq.
Y
Y
N
Y
Y
R
Y
Y
N
Y
Y
N
Y
Y
N
Y
Y
N
Y
Y
N
Y
Y
Y
Y
Y
Y
Y*
Y
Y*
5.3.1
5.3.2
Real-Time Filter
Dual-Channel RTA
Single-Channel RTA
Y
Y
Y
Y
Y
Y
Y
Y
Y
Y
Y
Y
Y
Y
Y
Y
N
N
Y
Y
R
N
N
N
N
N
N
N
Y
Y
5.4.2 Swept Sine
R
N
N
N
N
N
N
5.4.3 Time Delay Spectr.
R
N
N
N
N
N
N
Y Test method is appropriate for this signal.
N Should not be used.
R Required signal with this test method.
* Anechoic chamber is required unless simulated free field methods are used.
Y
Y
Y
Y
Y
Y*
Sine-Based
5.4.1
4023
4024
4025
Discrete Tone
Table F. 2 Signal compatibility with test method
4026
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
101
IEEE P269/D25 October 2004
4026
Annex G
4027
4028
(normative)
4029
4030
Analysis Methods
4031
4032
4033
G.1 General
4034
4035
4036
4037
4038
4039
4040
4041
4042
4043
4044
4045
4046
4047
4048
4049
4050
4051
4052
4053
4054
4055
4056
4057
4058
Various analysis techniques are available for electroacoustic measurements. Each technique has inherent
advantages and limitations. A particular method can be better suited for use with certain stimulus signals. Certain
methods, in fact, rely upon the use of a synchronized or otherwise unique stimulus signal. This clause describes the
most common techniques and their application to measurements of analog and digital telephones using handsets and
headsets.
The recommended method for calculating frequency response is based on dividing one rms spectrum by another.
See Equation 7.1Equation 7.1Equation 7.1 as an example in the case of receive frequency response. This method is
satisfactory and accurate in the great majority of cases. It applies to methods that measure an rms spectrum such as
single-channel FFT(G.2.2) and real-time filter analysis (G.3). It also applies to stepped or swept sine methods
(G.4.1 & G.4.2), if an rms detector insensitive to jitter is used.
In some cases it may be useful or desirable to use an alternative measurement method which calculates frequency
response by use of a cross-spectrum or similar process. These methods include dual-channel FFT (G.2.1), MLS
(G.2.3), TDS (G.4.3) and stepped or swept sine methods (G.4.1 & G.4.2) in which a quadrature or similar detector is
used. Such methods can sometimes reduce measurement time, reduce the influence of noise, or offer other benefits.
Cross-spectrum and related methods are usually sensitive to jitter or other unstable phase or time relationships that
can exist between input and output of a telephone with some kinds of digital processing. The test report must
include sufficient justification for use of cross-spectrum methods, such as demonstrating that the delay and phase
response are repeatable.
If cross-spectrum or related methods are used, the receive frequency response in dB, HR(f), is given by Equation G.
1Equation G. 1Equation G. 1:
H R ( f )  20 log
4059
4060
4061
4062
4063
4064
4065
4066
4067
4068
4069
4070
4071
4072
4073
G( RETP )( ERP ) ( f )
G RETP ( f )
in dBPa / V
Equation G. 1
where:
G(RETP)(ERP) (f) is the cross spectrum between RETP and ERP.
GRETP(f) is the rms spectrum at RETP
Similar formulas apply to other measurement paths.
Some systems may introduce significant delay such that the output is not time aligned with the stimulus signal. Care
should be taken that this does not introduce errors. The measurement system shall compensate for this delay as long
as the delay is stable. If the delay is not stable, cross-spectrum methods shall not be used, and the test signal shall be
at least 10 times longer than the variation in delay. See Annex L for measurements in delay.
Copyright © 2004 IEEE. All rights reserved.
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102
IEEE P269/D25 October 2004
4074
G.2 Fast Fourier transform (FFT) and cross spectrum analysis
4075
4076
4077
4078
4079
4080
4081
4082
4083
4084
4085
4086
4087
4088
4089
4090
4091
The Fourier Transform is a mathematical operation that decomposes a time signal into its complex frequency
components. The Inverse Fourier Transform reverses the process, reconstructing the time signal from its Fourier
components. By applying the FFT algorithm to a sampled time signal, a spectrum can be computed. This is a
parallel analysis resulting in a narrow band (constant bandwidth) frequency spectrum. Low frequency resolution can
be limited. Here, blocks of time data are analyzed.
4092
G.2.1
4093
4094
4095
4096
4097
4098
4099
4100
A dual-channel FFT analyzer performs simultaneous measurements of the telephone input and output. This type of
measurement is optimized for system analysis. Most FFT analyzers calculate the frequency response from the cross
spectrum and either the input or output autospectrum. In this way, different response estimators can be used to
minimize noise at the system input or output. This also enables computation of other functions such as coherence,
phase, group delay, coherent power and non-coherent power. Extensive data processing is normally available in both
the time and frequency domains. It is possible to improve measurement S/N by averaging and delay compensation.
Special care is needed when applying this method to telephones that are time variant or employ non-linear signal
processing.
4101
G.2.2
4102
4103
4104
4105
4106
4107
4108
Without cross spectrum capabilities, the system input and output are measured separately. These response
measurements require control of the excitation spectrum and/or a two-pass analysis. Therefore, measurement S/N
due to noise at the system input or output is not improved. Any post-processing features available will apply only to
the directly measured spectra, not to the response function. Special care is needed when applying this method to
telephones that are time variant or employ non-linear signal processing. This method requires the stimulus to be
stable between measurement of the system input (or calibration) and measurement of the system output.
4109
G.2.3
4110
4111
4112
4113
4114
4115
4116
The MLS technique employs a large (typically 16K) well-defined pseudo-random pulse excitation. The length of
the excitation signal is equal to the correlation length, eliminating leakage. The MLS excitation and analysis are
inherently synchronized. The received response signal is cross-correlated with the MLS signal, typically using a fast
Hadamard transform, to obtain the time response. An FFT is then used to obtain the frequency response. This also
enables computation of coherence, phase, group delay, coherent power and non-coherent power. Some non-linear
analysis capabilities and post-processing are available. This method can improve measurement S/N.
4117
G.3 Real-time filter analysis (RTA)
4118
4119
4120
4121
4122
Real-time analysis is essentially a parallel filter bank, usually implemented digitally. This results in a constant
percentage (logarithmic) frequency resolution. The analysis is carried out in parallel and the signal is processed
continuously. The filters shall be 1/12 or 1/24 octave, which comply with the ANSI S1.11 standard. The statistical
accuracy of real-time measurements is usually determined by the averaging time or the confidence level. This type
of analysis is optimized for single-port acoustical measurements (i.e., no control of the system input).
Care should be taken in the proper windowing of the data (i.e., Hanning, flat-top, etc.), overlap processing, and the
number of averages, to ensure an accurate analysis. The record length and window type determine the frequency
resolution. The frequency range and time resolution are inversely related. Because the data is discrete, the highest
frequency that can be measured is determined by the sampling frequency. Some degree of data processing is usually
available in both the time domain and in the frequency domain. An FFT analyzer can also have a zoom capability,
for increased frequency resolution across a restricted bandwidth.
When analyzing a periodic signal such as pseudo-random noise or a segment of real or artificial speech or artificial
voices, the averaging time shall be at least one full period of the signal. Averaging time shall be stated for all
measurements.
Dual-channel FFT
Single-channel FFT
Maximum length sequence (MLS) analysis
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
103
IEEE P269/D25 October 2004
4123
4124
4125
4126
4127
When analyzing a periodic signal such as pseudo-random noise or a segment of real or artificial speech or artificial
voices, the averaging time shall be at least one full period of the signal. Averaging time shall be stated for all
measurements.
4128
G.3.1
4129
4130
4131
4132
Two channels enable simultaneous measurement of the system input and output, for direct computation of the
frequency response (output/input). This method does provide limited harmonic distortion measurement capability,
and some direct post-processing of the data.
4133
G.3.2
4134
4135
4136
4137
4138
A single-channel real-time analyzer requires separate measurements of the system input and output. Response
measurements will require control of the excitation spectrum and/or a two-pass analysis. This method requires the
stimulus to be stable between measurement of the system input (or calibration) and measurement of the system
output. This method does provide limited harmonic distortion measurement capability, and some direct postprocessing of the data.
4139
G.4 Sine-based analysis
4140
4141
4142
4143
4144
4145
4146
4147
Sinusoidal excitation provides a high measurement S/N ratio and high degree of frequency selectivity. The analysis
is performed serially using either a quadrature or rms detector. This often includes a tracking filter for noise
suppression and selective measurements of distortion components. The quadrature detector multiplies the response
signal by a synchronized (and appropriately delayed) sine and cosine signal. This enables measurement of the
complex, steady-state frequency response (i.e., magnitude and phase, real and imaginary parts). Complex averaging
algorithms can be employed to improve the measurement S/N ratio. The use of an rms detector requires a separate
phase meter to obtain phase information.
4148
G.4.1
4149
4150
4151
4152
4153
4154
4155
4156
Discrete tone testing allows a measurement to be performed at precisely defined frequencies. These frequencies can
be at the ANSI/ISO preferred numbers or in other user-defined formats. See ISO 3 and ANSI S1.6 for preferred
number series. The actual frequency interval (not resolution) used in the measurement shall be stated. In addition to
frequency response measurements, intermodulation and difference frequency distortion testing are often carried out
using this method. Additionally, phase and group delay information is provided. These tests normally require an
anechoic room, although tone-burst techniques can be used with gating to obtain simulated free field results.
Measurement S/N can be improved using complex averaging.
4157
G.4.2
4158
4159
4160
4161
4162
This technique is similar to discrete tone testing, but instead employs a continuous linear or logarithmic sine sweep
excitation. The measurement is typically slow due to sweep rate limitations. This method is well suited for
frequency response and harmonic distortion measurements. An anechoic room is generally required, although toneburst techniques can be used with gating to obtain simulated free field results.
4163
G.4.3
4164
4165
4166
4167
4168
TDS, as classically implemented, utilizes a linearly swept sine excitation signal that is synchronized to the
measuring instrument. With this signal, a one-to-one relationship is established between time and frequency and
simulated free field measurements can be performed. The measured response signal is multiplied with an
appropriately delayed version of the excitation. This, in turn, is fed to a selectable constant bandwidth tracking filter
and a detector.
Dual-channel real-time filter analysis
Single-channel real-time filter analysis
Discrete tone (stepped sine)
Swept sine
Time delay spectrometry (TDS)
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
104
IEEE P269/D25 October 2004
4169
4170
4171
4172
4173
4174
4175
4176
4177
4178
4179
4180
4181
4182
4183
In practice, TDS can be implemented by many modern techniques. For example, post-processing algorithms can
substitute for the tracking filter. TDS can also be implemented using a logarithmic sweep followed by convolution.
Like other simulated free field techniques, the effective time window determines frequency resolution and the
lowest valid frequency. The time window is determined by the time between the arrival of the direct sound and the
arrival of the first reflection.
The TDS method also is well suited for harmonic distortion, and provides phase, group delay, and time response
information. This method may be implemented using an analog or digital process. In the later case, refinements and
corrections for deterministic errors in the measurement process may be incorporated. It is possible to improve
measurement S/N through complex averaging or delay compensation. This method allows post-processing of the
data. Special care is needed when applying this method to telephones that are time variant or employ non-linear
signal processing.
4184
G.5 Simulated free field techniques
4185
4186
4187
4188
4189
4190
4191
4192
4193
Simulated free field techniques employ some method of time windowing the measured response. Time windowing
enables the direct sound in a measurement to be separated from its reflections, producing a simulated free field
condition. In this case, the frequency resolution is the reciprocal of the applied time window. Both gating and postprocess windowing can be used on measurements in ordinary rooms.
4194
G.6 Measurement bandwidth
4195
4196
4197
4198
4199
4200
4201
4202
4203
4204
4205
4206
4207
4208
4209
4210
4211
4212
4213
4214
4215
4216
4217
4218
4219
4220
In general, the test signals and analysis methods in this standard cover a frequency range from approximately 100 to
8500 Hz. The lower limit is determined by the mouth simulator, whose practical lower limit is approximately 100
Hz for general use. The upper limit is determined by the range of the DRP-to-ERP translation curve (Annex C).
These limits may be somewhat modified when using standardized test signals which specify a particular bandwidth.
The exact range also depends on the analysis method. For measurements of frequency response, the analysis should
cover the same bandwidth as the test signal.
As discussed previously, MLS and TDS are inherently simulated free field techniques. Dual-channel FFT analysis
can also be used. The time windowing may be performed as a part of the data collection or as a post-processing
window operation.
For example, if artificial voices (F.6.1.1) are analyzed in 1/12th octave bands, the range should include the bands
centered from 91.7 through 7286 Hz. In ITU-T Recommendation P.50, the test signal is defined for the 1/3 octave
bands from 100 through 8000 Hz. The corresponding 1/12th octave bands extend from 91.7 through 8660 Hz.
However, the version of artificial voices currently published in ITU-T Recommendation P.50, Appendix 1 is
sampled at 16 kHz, thereby limiting the useful upper band to 7286 Hz. Other implementations of the artificial voices
would have to be evaluated on a case-by-case basis with respect to sampling rate and other characteristics.
If signals such as CSS (F.7.1) are analyzed in linear format, the range includes the lowest band at approximately 100
Hz, through approximately 8500 Hz.
The frequency range for sinusoidal signals is from 100 through 8500 Hz.
Digital telephones with a sampling rate of 8 kHz have an upper cutoff frequency just below 4 kHz. When testing
telephones known to be of this type, the high frequency limit of test signals should be reconsidered. When using
artificial voices or any signal with a speech-like spectrum, the full bandwidth should be used (up to approximately
8500 Hz.). Artificial voices and other speech-like signals have little long-term power above 4 kHz, so only a few
hundredths of a dB total stimulus power is lost due to a cutoff slightly below 4 kHz. However, when using test
signals with a relatively flat or pink spectrum (F.4 or F.5), the test signal should only extend to approximately 4
kHz.
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For noise measurements, the measurement bandwidth is nominally 25 Hz to 8500 Hz. The actual low frequency cut
off is 22.4 Hz (the lower edge of the 25 Hz 1/3rd octave band)
4224
G.7 Measurement resolution
4225
4226
4227
4228
4229
4230
4231
4232
4233
4234
4235
4236
4237
4238
4239
4240
The standard frequency pattern for sinusoidal test signals is the R40 sequence. (See Table G. 1Table G. 1Table G. 1,
Table G. 2Table G. 2Table G. 2 as well as ISO 3 and ANSI S1.6.) However, when testing digital devices, or devices
which have any internal digital processing, some of these frequencies should be adjusted up to +/- 1% so they do not
coincide with the sampling frequency, typically 8000 Hz, or submultiples thereof. An example would be to use
1004 Hz instead of 1000 Hz as a test tone.
4241
4242
4243
4244
4245
4246
4247
4248
4249
4250
4251
4252
4253
4254
4255
4256
4257
4258
4259
4260
4261
4262
4263
4264
The R10 frequency pattern is used for calculating loudness ratings. (See Annex H)
Constant-percentage bandwidth filters with 1/3 or 1/12 octave bandwidth have center frequencies and passband
upper & lower limit frequencies which are calculated by specific equations. See Table G. 1Table G. 1Table G. 1
and Table G. 2Table G. 2Table G. 2 for a complete list of 1/3 and 1/12 octave band frequencies within the scope of
this standard.
Exact center frequencies of 1/3 octave filters can be calculated according to Equation G. 2Equation G. 2Equation G.
2. The frequencies are actually based on 10 bands per decade.
n / 10 
f 10 
Equation G. 2
Where: n is the band number.
f is the frequency
The 1/3 octave passband upper & lower limit frequencies can be calculated according to Equation G. 3Equation G.
3Equation G. 3.
f 10n / 10 0.05
Equation G. 3
Example: For the 100 Hz band, the 1/3 octave band number = 20. The exact center frequency is 100 Hz, the lower
limit is 89.13 Hz, and the upper limit is 112.20 Hz.
For the 125 Hz band, the band number = 21. The exact center frequency is 125.89 Hz, the lower limit is 112.20 Hz,
and the upper limit is 141.25 Hz.
For 1/12 octave bands, the formulas are similar, except the centers are shifted one-half a band. This is done so that
four 1/12 octave bands will cover the exactly same range as a 1/3 octave band encompassing them. The frequencies
are actually based on 40 bands per decade, according to Equation G. 4Equation G. 4Equation G. 4.
f 10 n 0.540
Equation G. 4
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Example: 1/12 octave band number 80 has a center frequency of 102.92 Hz.
The 1/12 octave passband upper & lower limit frequencies can be calculated according to Equation G. 5Equation G.
5Equation G. 5.
f 10 n 0.5400.0125
Equation G. 5
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4274
R40 Preferred
Frequencies, Hz.
90
95
100
106
112
118
125
132
140
150
160
170
180
190
200
212
224
236
250
265
280
300
315
335
355
375
400
425
450
475
500
530
560
600
630
670
710
750
800
850
900
950
1000
1060
1120
4275
4276
4277
4278
1/12 Oct. Band
Center Freq, Hz.
91.73
97.16
102.92
109.02
115.48
122.32
129.57
137.25
145.38
153.99
163.12
172.78
183.02
193.87
205.35
217.52
230.41
244.06
258.52
273.84
290.07
307.26
325.46
344.75
365.17
386.81
409.73
434.01
459.73
486.97
515.82
546.39
578.76
613.06
649.38
687.86
728.62
771.79
817.52
865.96
917.28
971.63
1029.20
1090.18
1/3 Oct. Band
Center Freq, Hz.
R10 Preferred
Frequencies, Hz.
100.00
100
125.89
125
158.49
160
199.53
200
251.19
250
316.23
315
398.11
400
501.19
500
630.96
630
794.33
800
1000.00
1000
Table G. 1 Frequency formats, first decade
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R40 Preferred
Frequencies, Hz.
1/12 Oct. Band
Center Freq, Hz.
900
950
1000
1060
1120
1180
1250
1320
1400
1500
1600
1700
1800
1900
2000
2120
2240
2360
2500
2650
2800
3000
3150
3350
3550
3750
4000
4250
4500
4750
5000
5300
5600
6000
6300
6700
7100
7500
8000
8500
9000
9500
10000
10600
11200
917.28
971.63
1029.20
1090.18
1154.78
1223.21
1295.69
1372.46
1453.78
1539.93
1631.17
1727.83
1830.21
1938.65
2053.53
2175.20
2304.09
2440.62
2585.23
2738.42
2900.68
3072.56
3254.62
3447.47
3651.74
3868.12
4097.32
4340.10
4597.27
4869.68
5158.22
5463.87
5787.62
6130.56
6493.82
6878.60
7286.18
7717.92
8175.23
8659.64
9172.76
9716.28
10292.01
10901.84
1/3 Oct. Band
Center Freq, Hz
R10 Preferred
Frequencies, Hz.
1000.00
1000
1258.93
1250
1584.89
1600
1995.26
2000
2511.89
2500
3162.28
3150
3981.07
4000
5011.87
5000
6309.57
6300
7943.28
8000
10000.00
10000
Table G. 2 Frequency formats, second decade
4282
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Annex H
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4284
(normative)
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4287
4288
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4290
4291
4292
4293
4294
4295
4296
4297
4298
4299
4300
4301
4302
4303
4304
4305
4306
4307
4308
4309
Loudness Rating Calculations
ISO R10 format data is required for calculating loudness ratings according to ITU-T Recommendation P.79.
Measured frequency responses (receive, send, sidetone, etc.) should be directly converted to R10 format for this
purpose.
Although it has been common practice to remeasure at the R10 frequencies only for the purpose of calculating
loudness ratings, this practice is neither necessary or desirable. The conversion procedure in this annex makes
remeasurement unnessary. Measurement at the R10 points is not always desirable, since undersampling can occur.
While this is not likely to introduce much error when the frequency response is smooth, when the frequency
response is irregular the undersampling error can be larger. Irregular frequency response it not generally desirable,
but it may be more likely in devices with digital signal processes running than in some types of simple analog
systems.
Leakage correction is not used for Type 2 and Type 3 ear simulators. Historically, a leakage correction was used to
calculate loudness ratings on a Type 1 ear simulator.
Measurements may be performed in various frequency formats, depending upon the analysis method employed.
Response measurements can contain numerous peaks and dips. This conversion, therefore, should be performed
u
s
i
ng“
ba
n
d-a
v
e
r
a
g
i
n
g
”
.Th
eme
a
s
u
r
e
dpoi
n
t
swi
t
h
i
napa
r
t
i
c
u
l
a
r1/
3oc
t
a
v
eba
n
da
r
e“
powe
ra
v
e
r
a
g
e
d”a
c
c
or
di
ngt
o
Equation H. 1Equation H. 1Equation H. 1, and assigned to the R10 frequency at the band center.
At each ISO R10 preferred frequency
1
H
( f ) 10 log 10 
N
4310
4311
4312
4313
4314
4315
4316
4317
4318
4319
4320
4321
4322
4323
4324
4325
4326
4327
4328
4329
4330
Hi

10
 10 
i 1

N
Equation H. 1
where
H’
(
f
)
f
N
i
Hi
= response at the new preferred ISO R10 frequency
= preferred ISO R10 frequency
= number of response values within the 1/3 octave band centered at f
= index for each response value within the 1/3 octave band
= measured response value (in dB)
For the lowest frequency within the band, i = 1. For the highest included frequency, i = N. The 1/3 octave passband
limit frequencies can be calculated according to Equation H. 2Equation H. 2Equation H. 2:
f 10( n /10) 0.05
Equation H. 2
where
n is the band number.
Example: For the 100 Hz band, the band number = 20; For the 125 Hz band, the band number = 21, etc. See also
Annex G.7.
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4337
For measured data at frequencies coinciding with a band-edge frequency (i = 1 and/or i = N), reduce the value by 3
dB, and use that data point in both the upper and lower frequency band calculations.
For constant percentage bandwidth measurements, there will always be the same number of points for each
converted band (4 or 8, for 1/12 or 1/24 octave bands, respectively). For constant bandwidth data (e.g., FFT) on a
log frequency axis, the measurement data will appear under sampled at low frequencies and over sampled at higher
frequencies.
4338
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111
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4338
Annex I
4339
4340
(normative)
4341
4342
4343
4344
4345
4346
4347
4348
4349
4350
4351
4352
4353
4354
4355
4356
4357
4358
4359
4360
4361
4362
4363
4364
4365
4366
4367
4368
4369
4370
4371
4372
4373
4374
4375
4376
4377
4378
4379
4380
4381
4382
4383
4384
4385
4386
4387
4388
Linearity
Linearity is a measure of how frequency response changes with input level. The test consists of measuring the
relevant frequency response, but performing the measurement at several different stimulus levels. If the telephone is
linear, the frequency response should be the same regardless of the stimulus level. Frequency responses are to be
measured according to Clauses 7-9 (for example, Clause 7.4.1).
The purpose of this method is to give a complete overview of the linearity of a device over a wide frequency and
dynamic range, all in one graph. The method is a particular combination of measurements, post-processing and
display procedures.
The stimulus intervals and frequency patterns for linearity measurements have been specified in the body of this
standard (for example, Clause 7.4.4). These parameters have been selected to reveal typical nonlinearities over the
basic frequency and dynamic range of typical devices, without taking too much measurement time. For additional
investigation of specific behaviors, these parameters may be altered. For example, it may be useful to use a much
smaller stimulus interval, say 1 dB, for a more detailed look at the dynamic behavior of a device. If sharp
resonances are to be investigated, a more dense frequency pattern, such as 1/12th octaves or R40, might be useful.
Linearity shall be measured using the same stimulus type used to measure frequency response (send, receive,
sidetone, or overall). When using artificial voices, the linearity measurement includes the effects of anything
nonlinear, whatever the cause. Nonlinearities could be intentional or unintentional compression or expansion,
distortion of various kinds, or other nonlinear processes. The linearity measurement shows if nonlinearity occurs, as
well as the level and frequency range where it occurs. To analyze the cause, further investigation is required.
However, the common patterns are shown in the figures in this annex.
The linearity test shall be performed at 7 levels, in 5 dB intervals. Smaller intervals and/or a wider range of levels
may also be used. The reference stimulus level shall be specified.
Each individual measurement is processed according to Equation I. 1Equation I. 1Equation I. 1:
C
 G x Gr ( x r )
Equation I. 1
where
C = displayed linearity curve
Gx = frequency response in dB at stimulus level x
Gr = frequency response in dB at reference stimulus r
x = the stimulus level in dB
r = the reference stimulus level in dB
For a linear phone measured with artificial voices, the result is 7 parallel lines at levels from 0 to –30 dB relative to
the reference stimulus (see Figure I. 1Figure I. 1Figure I. 1). If the measurement is made with sine waves, the result
Copyright © 2004 IEEE. All rights reserved.
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4393
is 7 parallel lines at levels from –15 to +15 dB relative to the reference stimulus (see Figure I. 2Figure I. 2Figure I.
2). Nonlinearities are displayed as variations from the parallel lines (see Figure I. 4Figure I. 4Figure I. 4 through
Figure I. 7Figure I. 7Figure I. 7).
Reference
Stimulus
-4.7dBPa
0
-5
dB
-10
-10
-15
-20
-20
-25
-30
-30
100
4394
4395
4396
4397
4398
4399
200
500
1k
2k
5k
10k
Frequency (Hz)
Figure I. 1 Linear phone measured with artificial voices
20
+15
+10
10
+5
dB
Reference
Stimulus
-11.7dBPa
0
-5
-10
-10
-15
-20
100
4400
4401
4402
4403
4404
4405
4406
4407
4408
4409
200
500
1k
2k
5k
10k
Frequency (Hz)
Figure I. 2 Linear phone measured with sine wave
Each displayed curve is a relative frequency response which shows any deviations from linearity. Each curve is
displaced vertically by the amount the stimulus level differs from the reference stimulus. The linearity information
for the entire frequency and dynamic range is shown in one graph.
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4410
15
Output
dB
0
-15
-15
4411
4412
4413
4414
4415
4416
4417
4418
4419
4420
4421
4422
4423
4424
4425
-10
-5
0
+5
+10
+15
Input, dB
Figure I. 3 Linearity displayed as ordinary input-output curve at one frequency (for information only)
If an imaginary vertical line were drawn through all the curves of Figure I. 1Figure I. 1Figure I. 1 or Figure I.
2Figure I. 2Figure I. 2 at a particular frequency, it would intersect the points typically displayed in a one-frequency
input/output curve. In that case, the intersected points would be the y-values, and the stimulus levels would be the x
values, as in Figure I. 3Figure I. 3Figure I. 3. In Figure I. 1Figure I. 1Figure I. 1 and Figure I. 2Figure I. 2Figure I.
2, the same information is displayed at all frequencies in one graph.
Figure I. 4Figure I. 4Figure I. 4 through Figure I. 7Figure I. 7Figure I. 7 show examples of nonlinearities measured
according to this method.
20
+15
+10
10
+5
dB
Reference
Stimulus
-21dBV
0
-5
-10
-10
-15
-20
100
4426
4427
4428
4429
4430
4431
200
500
1k
2k
5k
10k
Frequency (Hz)
Figure I. 4 Receiver with compression at low-frequency resonance, measured with sine waves
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114
IEEE P269/D25 October 2004
Reference
Stimulus
-16dBV
0
-5
dB
-10
-10
-15
-20
-25
-20
-30
100
4432
4433
4434
4435
4436
4437
200
500
1k
2k
5k
10k
Frequency (Hz)
Figure I. 5 Wideband compressor with 1.5 to 1 ratio, measured with artificial voices
10
+15
+10
Reference
Stimulus
LMID
+5
dB
0
-5
-10
-10
-15
-20
100
4438
4439
4440
4441
200
500
1k
2k
5k
10k
Frequency (Hz)
Figure I. 6 Headset with limiter, measured with artificial voices, 15 dB re LMID
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115
IEEE P269/D25 October 2004
20
+15
+10
10
+5
dB
Reference
Stimulus
-21dBV
0
-5
-10
-10
-15
-20
100
4442
4443
4444
4445
4446
4447
200
500
1k
2k
5k
10k
Frequency (Hz)
Figure I. 7 High frequency limiting due to overload in pre-emphasis circuit, measured with sine waves
4448
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116
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4448
Annex J
4449
4450
(normative)
4451
4452
Distortion
4453
4454
4455
[need to finish integrating this section and its relationship with annex F and all other references in doc about
preferred method (formally SDN). Decide whether SDN is it a distortion percentage or described in dB (make
words of formula agree)]
4456
J.1
4457
4458
4459
4460
4461
4462
4463
4464
4465
4466
4467
4468
4469
4470
4471
4472
4473
4474
4475
4476
Distortion is a measure of unwanted signals which appear at the output of a device at frequencies not present in the
input. Distortion is a function of input level, frequency, and the type of signal. Because of this, different methods
cannot necessarily be expected to correlate with each other.
4477
J.2
4478
4479
4480
4481
4482
4483
4484
4485
To test the suitability of a proposed distortion test signal, the signal should be applied at each distortion test
frequency using the standard level. The frequency response should then be measured at those test frequencies. If the
result is within ±2 dB of the comparable values previously obtained in the complete frequency response
measurement, then the proposed distortion test signal is suitable.
4486
J.3Signal-to-distortion-and-noise ratio (SDN)
4487
4488
4489
4490
4491
4492
4493
4494
The recommended distortion test method for this standard is signal-to-distortion -and-noise ratio. This method uses a
narrow-band pseudo-random noise as the stimulus, and analyzes THD + noise with a weighted notch filter. See
Equation J. 1Equation J. 1 and Equation J. 2Equation J. 2.
Overview
The recommended method for all telephones is signal-to-distortion-and-noise ratio (SDN), defined in Clause
A.1.1J.3. It uses a narrow-band pseudo-random noise as the stimulus, and analysis of THD + noise with a weighted
notch filter.
Distortion test methods using sinewave stimulus may be suitable for use on many handsets and headsets and on
some telephones. Sine methods and extensions of sine methods are described in clause J.3J.4.
Continuous spectrum distortion methods may be a suitable alternative under some conditions where artificial voices
or other continuous-spectrum test signals are used, and cross-spectrum methods are valid. See clause J.4J.5.
Subjective predictors, such as algorithms which estimate mean opinion scores (MOS), may also be useful in
identifying distortions and degradations peculiar to digital processing. These algorithms have generally been
developed primarily for measurements of distortions found in networks, and may not be completely applicable to
telephones or headsets. The results may not correlate directly with other measures. However, their use is
encouraged as a supplemental investigation. One example is PESQ (ITU-T Recommendation P.832)
Signal suitability test
Distortion does not have to be measured using the same test signal as is used for measuring frequency response, but
the suitability test shall be fulfilled.
The narrow-band pseudo-random noise should have an effective bandwidth of 25 to 50 Hz. Out-of-band signals
should add no more than 0.5 dB to the overall level of the test signal. The periodic nature of this signal will provide
some modulation effect, depending on how the signal is constructed. The period should be at least 250 ms, with
frequency components no more than 4 Hz apart. The crest factor should be 9±3 dB.
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4496
4497
4498
4499
4500
4501
4502
4503
4504
4505
4506
The output fundamental is measured with a bandpass filter or equivalent algorithm. Measurement is made using an
A-weighting filter according to ANSI S1.4-1983 (R1997), but with a notch added to eliminate the test signal. (Send
distortion may be measured using the psophometric weighting if required by the relevant performance standard.)
Output from the notched filter includes harmonics and nonharmonic products, as well as both continuous noise and
modulation noise. The notch filter output is divided by the fundamental and expressed in percent, using Equation J.
1Equation J. 1 and Equation J. 2Equation J. 2. The result is the A-weighted signal-to-distortion-and-noise ratio.
The notch shall attenuate the test signal by at least 50 dB. This will result in a distortion floor of 0.3%, permitting
measurements of distortion from 1% and above with 6% or better accuracy.
% SDN
4507
 100
output from weighted notch filter
narrow band noise stimulus
4508
4509
4510
4511
4512
Equation J. 1
% SDN
4513
4514
4515
4516
4517
4518
4519
4520
4521
4522
4523
4524
4525
4526
 100
W ( f ) { ( A2 ) 2 ( A3 ) 2 . . . ( An ) 2 ( Anoise ) 2 }
A1
Equation J. 2
Where
An = amplitude of nth product
Anoise = amplitude of wideband noise and nonharmonic products
W(f) = amplitude weighting function (A-weighting with a notch)
Measurements should be made over a range of frequencies within the telephone band, such as the ISO R10 preferred
frequencies from 315 Hz to 3150 Hz. Test frequencies over one half the upper frequency limit of the telephone may
not be useful for evaluation of harmonic distortion. For high acoustic test levels, verify that the distortion of the test
system is less than 2%.
4527
J.4J.3 Sinusoidal Methods
4528
4529
4530
4531
4532
4533
4534
4535
4536
4537
4538
4539
4540
4541
4542
4543
Note to committee: add ANTHD, no restriction on harmonics? (digital phones bandwidth vs harmonic #) (see
linearity). Add to definitions, ANTHD.The sinusoidal methods may be used with a sine, modulated sine or narrowband pseudo-random noise as the stimulus. The choice of stimulus is discussed in greater detail in annex J.1 and J.2
[fix up stimulus references…]
For discussions on modulation, see annex F.3.
Narrow-band pseudo-random noise should have an effective bandwidth of 25 to 50 Hz. Out-of-band signals should
add no more than 0.5 dB to the overall level of the test signal. The periodic nature of this signal will provide some
modulation effect, depending on how the signal is constructed. The period should be at least 250 ms, with frequency
components no more than 4 Hz apart. The crest factor should be 9±3 dB.
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118
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4544
J.4.1J.3.1
Total harmonic distortion (THD) and harmonic analysis
4545
4546
4547
4548
4549
4550
4551
4552
4553
4554
4555
Total harmonic distortion is the ratio of the power sum of all the harmonics to the fundamental. It is usually
expressed as a percentage, according to Equation J. 3Equation J. 3Equation J. 3 - Equation J. 6Equation J. 6Equation
J. 6.
Harmonics may also be expressed separately to give diagnostic information in addition to THD.
Harmonic analysis may be done using bandpass filters, or an equivalent algorithm. Equations J.3 and J.4 are
preferred.
% THD
4556
 100
power sum of included harmonics
fundamental
4557
4558
4559
4560
4561
Equation J. 3
% THD
4562
4563
4564
4565
4566
4567
4568
 100
( A2 ) 2 ( A3 ) 2  . . . ( An ) 2
A1
Equation J. 4
Equations J.5 and J.6 may be used as an aAlternatively. For low values of distortion the results are similar to that
obtained by using equations J.3 and J.4. For higher values of distortion the values will differ.,
% THD
4569
 100
power sum of included harmonics
power sum of fundamental and included harmonics
4570
4571
4572
4573
Equation J. 5
% THD  100
4574
( A2 ) 2 ( A3 ) 2  . . . ( An ) 2
( A1 ) 2 ( A2 ) 2 ( A3 ) 2  . . . ( An ) 2
4575
4576
4577
4578
4579
Equation J. 6
Where
4580
J.4.2J.3.2
4581
4582
4583
4584
4585
4586
THD + Noise is the ratio of the rms amplitude of the residual harmonics and noise to the rms amplitude of the
fundamental, harmonics and noise combined. (Equation J. 7Equation J. 7Equation J. 7 and
Equation J. 8
Equation J. 8
Equation J. 8.) It is usually expressed as a percent.
An = amplitude of nth product
Total Harmonic Distortion (THD) and noise
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
119
IEEE P269/D25 October 2004
4587
4588
4589
4590
4591
4592
4593
4594
Total harmonic distortion and noise is measured by use of a notch (bandstop) filter to eliminate the fundamental.
This measurement will be equivalent to total harmonic distortion, with an error of less than 5%, if the magnitude of
the distortion does not exceed 30%, and if there is no significant noise component.
The notch shall attenuate the test signal by at least 50 dB. This will result in a distortion floor of 0.3%, permitting
measurements of distortion from 1% and above with 6% or better accuracy.
% THD Noise  100
4595
output from notch filter
unfiltered total output
4596
4597
4598
4599
Equation J. 7
% THD Noise  100
4600
4601
4602
4603
4604
4605
4606
4607
4608
( A2 ) 2 ( A3 ) 2  . . . ( An ) 2 ( Anoise ) 2
( A1 ) 2 ( A2 ) 2 ( A3 ) 2  . . . ( An ) 2 ( Anoise ) 2
Equation J. 8
Where
An = amplitude of nth product
Anoise = amplitude of wideband noise and nonharmonic products
4609
J.3.3
4610
4611
4612
4613
4614
4615
4616
4617
4618
4619
4620
4621
4622
4623
4624
4625
4626
4627
4628
The signal-to-distortion-and-noise ratio method uses a sine, modulated sine or narrow-band pseudo-random noise as
the stimulus, and analyzes THD + noise with a weighted notch filter. See and .
4629
Signal-to-distortion-and-noise ratio (SDN)
The narrow-band pseudo-random noise should have an effective bandwidth of 25 to 50 Hz. Out-of-band signals
should add no more than 0.5 dB to the overall level of the test signal. The periodic nature of this signal will provide
some modulation effect, depending on how the signal is constructed. The period should be at least 250 ms, with
frequency components no more than 4 Hz apart. The crest factor should be 9±3 dB.
The output fundamental is measured with a bandpass filter or equivalent algorithm. Measurement is made using an
A-weighting filter according to ANSI S1.4-1983 (R1997), but with a notch added to eliminate the test signal. (Send
distortion may be measured using the psophometric weighting if required by the relevant performance standard.)
Output from the notched filter includes harmonics and nonharmonic products, as well as both continuous noise and
modulation noise. The notch filter output is divided by the fundamental and expressed in percent, using and . The
result is the A-weighted signal-to-distortion-and-noise ratio.
The notch shall attenuate the test signal by at least 50 dB. This will result in a distortion floor of 0.3%, permitting
measurements of distortion from 1% and above with 6% or better accuracy.
% SDN
 100
output from weighted notch filter
narrow band noise stimulus
4630
4631
4632
4633
Equation J. 91
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
120
IEEE P269/D25 October 2004
4634
% SDN
4635
4636
4637
4638
4639
4640
4641
4642
4643
4644
4645
4646
4647
4648
4649
4650
4651
4652
4653
4654
4655
4656
4657
4658
4659
4660
4661
 100
W ( f ) { ( A2 ) 2 ( A3 ) 2 . . . ( An ) 2 ( Anoise ) 2 }
A1
Equation J. 102
Where
An = amplitude of nth product
Anoise = amplitude of wideband noise and nonharmonic products
W(f) = amplitude weighting function (A-weighting with a notch)
Measurements should be made over a range of frequencies within the telephone band, such as the ISO R10 preferred
frequencies from 315 Hz to 3150 Hz. Test frequencies over one half the upper frequency limit of the telephone may
not be useful for evaluation of harmonic distortion. For high acoustic test levels, verify that the distortion of the test
system is less than 2%.
J.3.4
Amplitude normalized total harmonic distortion (ANTHD)
ANTHD is an alternative method which makes it possible to separate the nonlinear transducer distortions from the
linear distortions of the system under test. Linear distortions include the frequency response of the speaker or
receiver and how it is coupled to the ear. ANTHD is generally not preferred for most telecom measurements since it
may mask poor distortion performance in devices that have poor low frequency performance.
Amplitude Normalized Total Harmonic Distortion (ANTHD): The ratio of the square root of the sum of all of the
squared second, third, and higher harmonic amplitudes, at their harmonic frequencies, (normalized to the amplitude
of the fundamental at the same frequencies) to the amplitude of the fundamental. For this document ANTHD is
calculated as a percentage using the second and third harmonics.
2
4662
4663
4664
4665
4666
4667
4668
4669
4670
4671
4672
4673
4674
4675
4676
4677
4678
4679
2
2
A ( f )  A3 ( f ) 
A ( f ) 
% ANTHD 100  2

...  n



A1 ( 2 f )  A1 (3 f ) 
A1 ( nf ) 
Equation J. 119
Where:
Total Distortion (TD) is the power sum of all the harmonics (Error! Not a valid link.), which may or may not be
included in the calculation of % ANTHD.
Total Distortion  ( A2 ) 2 ( A3 ) 2 ... ( An ) 2
Equation J. 1210
And:
n is the harmonic number
A1(nf) is the amplitude of the fundamental at frequency n•
f.
An (f) is the amplitude of the nth harmonic at excitation frequency f.
Copyright © 2004 IEEE. All rights reserved.
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121
IEEE P269/D25 October 2004
4680
J.4.3J.3.5
Difference-frequency distortion (DF Distortion)
4681
4682
4683
4684
4685
4686
4687
4688
Difference-frequency distortion is measured by using two stimulus signals, typically spaced from 20 to 200 Hz
apart. A complex group of distortion products results, consisting of odd and even order products. (Equation J.
13Equation J. 11Equation J. 11 and Equation J. 14Equation J. 12Equation J. 12) It is essentially the same as the
production sidebands in a mixer or modulator.
Difference-frequency distortion tests may be the best way to evaluate a telephone above 1000 Hz, where the
harmonics of a single tone (or narrow-band pseudo-random noise signal) l
i
ea
bov
et
h
es
e
t
’
sc
u
t
of
ff
r
e
qu
e
n
c
y
.
4689
% Total DF Distortion
 100
power sum of included products
power sum of both stimulus signals
4690
4691
4692
4693
Equation J. 1311
% Total DF Distortion  100
4694
4695
4696
4697
4698
4699
4700
( A2 ) 2 ( A3 ) 2 ( A3 ) 2 ( A4 ) 2 ( A5 ) 2 ( A5 ) 2  . . .
( A f 1 ) 2 ( A f 2 ) 2
Equation J. 1412
Where
An = amplitude of nth product
Afn = amplitude of nth stimulus signal products
4701
J.4.4J.3.6
Intermodulation distortion (IM Distortion)
4702
4703
4704
4705
4706
Intermodulation distortion measurement typically uses one test tone at a fixed low frequency, such as 60 Hz,
together with a second tone stepped or swept through the band of the device. Intermodulation distortion
measurement is not recommended for use with telephone products operating in the normal speech band. It may be
usable in wideband telephony, but that has not been studied for use in this standard.
4707
J.4.5J.3.7
4708
4709
4710
4711
4712
4713
4714
4715
4716
4717
4718
4719
4720
4721
4722
Harmonic and difference-frequency distortion measurement methods can be extended for more appropriate
application to telephone and headset testing, where a sinusoidal stimulus is not always suitable.
4723
J.4.6J.3.8
4724
4725
Measurements should be made over a range of frequencies within the telephone band, such as the ISO R10 preferred
frequencies from 315 Hz to 3150 Hz. Test frequencies over one half the upper frequency limit of the telephone may
Alternatives to sinewave stimulus signals
One alternative is to use modulated sine waves as the stimulus. A square wave, sine wave, or a pseudo-random
modulation can be used to modulate the sine wave signals. Refer to Clause F.3 for details. One modulated sinewave
is used for harmonic distortion, while two are used for difference-frequency distortion.
Another alternative is to use a narrow-band pseudo-random noise signal as the stimulus. One narrow-band noise
signal is used for harmonic distortion measurements, while two narrow-band noise signals are used for differencefrequency distortion. The total stimulus level is calculated on a power basis. See clause A.1.1J.3 for details about
the narrow-band noise stimulus.
When using these alternative test signals, analysis is with rms detectors and bandpass filters, or equivalent
algorithm.
Test frequencies
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
122
IEEE P269/D25 October 2004
4726
4727
4728
not be useful for evaluation of harmonic distortion. For high acoustic test levels, verify that the distortion of the test
system is less than 2%.
4729
J.5J.4 Coherence methods (N/C Ratio)
4730
4731
4732
4733
4734
4735
Conventional techniques for measuring harmonic and intermodulation distortion are not usable in continuous
spectrum methods. An alternative is to use the ratio of noncoherent to coherent power (N/C), each summed over the
most important part of the telephone bandwidth of 300 –3300 Hz. (Equation J. 15Equation J. 13Equation J. 13).
This method is suitable if, and only if, the telephone or headset under test has a stable coherent frequency response.
(Magnitude and phase are stable.)
% Continuous Spectrum Distortion (N/C) 
4736
4737
4738
4739
4740
4741
4742
4743
4744
4745
4746
4747
4748
4749
4750
4751
4752
4753
4754
4755
4756
4757
4758
4759
4760
4761
4762
4763
4764
4765
noncoheren t power (300 3300Hz)
coherent power (300 3300Hz)
Equation J. 1513
Coherent power is the power in the output spectrum that is linearly related to the input. Noncoherent power is the
remainder. The following can cause this nonlinear remainder:
1.
2.
3.
4.
5.
Nonlinearity in the telephone under test.
Noise in the telephone or measurement system.
Analysis leakage due to an inappropriate time window or insufficient measurement resolution.
Multiple inputs or multiple outputs.
Uncompensated delay between input and output.
An analyzer cannot distinguish among these factors, so care is needed in setting up the measurement and in
interpreting the results. Factors 3, 4, and 5 can be largely eliminated by proper measurement setup.
A separate measurement of noise in the device under test, summed over the telephone bandwidth of 300 –3300 Hz,
should be made with the continuous spectrum test signal deactivated. If this noise is significantly less than the
noncoherent power, then the noncoherent power is due to nonlinearity in the device and Equation J. 16Equation J.
14Equation J. 14 is valid.
Another method for interpreting the N/C ratio is to perform the measurement at different levels and compare the
results. For example at moderate levels, the N/C ratio will usually be at its lowest, indicating relatively low noise as
well as relatively low nonlinearity. At low levels, the N/C ratio typically increases due to noise. At high levels, the
N/C ratio normally increases due to nonlinearity.
Further equations and definitions:
noncoherent power
coherent power
(1 2 ) Gbb

2 Gbb
(1 2 )

2

Gaa Gbb  Gab
4766
4767
4768
Gab
2
2
Equation J. 1614
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
123
IEEE P269/D25 October 2004
4769
4770
where
2
Coherence 

Gab
2
Gaa Gbb
4771
4772
Gaa
 input autospectrum
Gbb
 output autospectrum
Gab
 cross spectrum
4773
4774
4775
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
124
IEEE P269/D25 October 2004
4775
Annex K
4776
4777
(normative)
4778
4779
Send Signal-to-Noise Ratio
4780
4781
K.1 Send signal-to-noise ratio
4782
4783
4784
4785
4786
4787
4788
4789
4790
4791
4792
4793
4794
4795
4796
4797
4798
Send signal-to-noise ratio, SendSNR(f), is a measure of the desired speech transmission relative to unwanted noise
i
nt
h
er
oom whe
r
et
h
et
a
l
k
e
r
’
sph
on
e
,h
a
n
ds
e
torh
e
a
ds
e
ti
su
s
e
d.Th
eme
a
s
u
r
e
me
n
ti
si
n
t
e
n
de
dt
oa
ppl
yt
obot
h
passive and active systems. SendSNR(f) is given by equation Equation K. 1Equation K. 1Equation K. 1.
Two test signals are used for this measurement. The first is the desired speech signal, presented from the mouth
simulator. The signal and positioning should be the same as used to determine send frequency response (7.5.1). The
second is a noise signal presented in a diffuse field (5.5.3). This noise signal may be Hoth noise (Annex E) or any
other noise signal representative of actual working conditions. The DFTP and the MRP shall coincide.
The desired speech signal is presented together with the diffuse noise signal to obtain GSETP(S+N)(f). The diffuse
noise signal is presented alone, to obtain GSETP(N)(f).
The results are sensitive to the relative levels of the both signals, and may be sensitive to the absolute levels and
types of signals used. The results may also be sensitive to the spectrum of the test signals. The recommended noise
spectrum is Hoth noise, at –4.7 dBPa. The recommended speech signal test level is also –4.7 dBPa.
SendSNR ( f )
4799
4800
4801
4802
4803
4804
4805
4806
4807
4808
G SETP ( S N ) ( f ) G SETP ( N ) ( f )  
 


10
 10 log 
10
1 in dB




Equation K. 1
for:
GSETP(S+N)(f) > GSETP(N)(f)
where:
inactive.
GSETP(S+N)(f) is the rms spectrum at SETP with both the mouth simulator and noise sources active
GSETP(N)(f) is the rms spectrum at SETP with only the noise source active. The mouth simulator present, but
4809
K.2 Weighted send signal-to-noise ratio
4810
4811
4812
4813
Weighted send signal-to-noise ratio, SendSNRw, is a single number which results from applying an intelligibility
weighting WSNR (Table K. 1Table K. 1Table K. 1) to the SendSNR (Equation K. 2Equation K. 2Equation K. 2).
4814
SendSNRW

f 5000
SendSNR( f ) WSNR
f 200
4815
4816
4817
4818
Equation K. 2
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
125
IEEE P269/D25 October 2004
4818
1/3 octave band
center frequency
200
250
315
400
500
630
800
1000
1250
1600
2000
2500
3150
4000
5000
4819
4820
4821
4822
4823
4824
Weighting
WSNR
.012
.030
.030
.042
.042
.060
.060
.072
.090
.112
.114
.102
.102
.072
.060
Table K. 1 Intelligibility weightings
4825
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
126
IEEE P269/D25 October 2004
4825
Annex L
4826
4827
(normative)
4828
4829
Delay
4830
4831
L.1 General
4832
4833
4834
4835
4836
Delay can be measured in several ways, many of which are described in this clause. Electroacoustical delays in the
test equipment, such as the mouth simulator, can generally be ignored. The range of delay that can be measured by
the test equipment must exceed the expected delay in the device under test, or time domain aliasing may occur. The
method used should be stated with the measurement.
4837
L.2 Captured pulse method
4838
4839
4840
4841
4842
4843
Delay can be measured using a captured pulse. The pulse can be a swept sine or a gated sine. The recommended
timing for a pulse is 30 to 50ms on and 500 to 800ms off. This timing allows measuring equipment, such as a digital
storage oscilloscope, to acquire sufficient data for a clean measurement. The pulse is delivered to the input test point
and triggers the time capture. Record the difference in time between the start of the input pulse and the start of the
measured pulse at the output test point.
4844
L.3 Two-channel analyzer methods
4845
L.3.1
4846
4847
4848
4849
4850
4851
4852
Measure the impulse response. Delay between channels is the time at which the magnitude of the impulse response
is at its maximum. The delay between two events is the time difference between the maxima of the two impulse
responses.
4853
L.3.2
4854
4855
4856
4857
Measure the cross-correlation. Delay between channels is the time at which the cross-correlation coefficient is at its
maximum. The delay between two events is the time difference between the maxima of the two impulse responses.
If available on the analyzer, the magnitude of the cross correlation should be used rather than the real part.
4858
L.4 Time Delay Spectrometry Method
4859
4860
4861
4862
4863
4864
4865
Measure the impulse response. Delay between channels is the time at which the magnitude of the impulse response
is at its maximum. The delay between two events is the time difference between the maxima of the two impulse
responses.
Impulse response method
The magnitude of the impulse response is calculated as the square root of the sum of the squares of the impulse
response (real part) and the Hilbert transform of the impulse response (imaginary part).
Cross-correlation method
The magnitude of the impulse response is calculated as the square root of the sum of the squares of the impulse
response (real part) and the Hilbert transform of the impulse response (imaginary part).
Copyright © 2004 IEEE. All rights reserved.
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127
IEEE P269/D25 October 2004
4866
L.5 MLS Method
4867
4868
4869
4870
4871
4872
4873
Measure the impulse response. Delay between channels is the time at which the magnitude of the impulse response
is at its maximum. The delay between two events is the time difference between the maxima of the two impulse
responses.
The magnitude of the impulse response is calculated as the square root of the sum of the squares of the impulse
response (real part) and the Hilbert transform of the impulse response (imaginary part).
4874
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
128
IEEE P269/D25 October 2004
4874
Annex M
4875
4876
(normative)
4877
4878
4879
4880
4881
4882
4883
4884
4885
4886
4887
4888
4889
4890
4891
4892
4893
4894
4895
4896
4897
4898
4899
4900
4901
4902
Sidetone Echo
In some phones there may be an audible delay in the sidetone path. This delay may be heard as an unnatural quality
and/or as an echo. The perceived quality can depend on the amount of delay, the amplitude and spectrum of the
delayed sidetone, and the amplitude and spectrum of the local (acoustic) sidetone.
Sidetone delay is measured between the mouth simulator and the ear simulator, using one of the methods described
in Annex L. If the delay is 5ms or less, talker sidetone may be measured in the standard way (7.6.1 or 8.6.1).
If the delay exceeds 5ms, the local (undelayed) sidetone and the sidetone echo should be measured separately, using
one of the simulated free field techniques described in Annex G.5. In this application the time window is used to
separate the local sidetone from the sidetone echo, not necessarily to simulate a free field.
To measure local sidetone, the window should begin at approximately 0ms, depending on the exact shape of the
time window. The window should be as long as possible without including the sidetone echo.
To measure sidetone echo, the window should begin just before the onset of the echo, depending on the exact shape
of the time window. The window should be as long as possible without including the sidetone echo
The true frequency resolution of a simulated free field measurement will be determined by the time window chosen.
The effective time window should be at least 5.7 ms, which corresponds to a frequency resolution (lowest
measurable frequency) of 175 Hz.
Both local sidetone frequency response and sidetone echo frequency response are defined similarly to Equation
7.7Equation 7.7Equation 7.4 in 7.6.1. The exact formula depends on the method chosen.
4903
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
129
IEEE P269/D25 October 2004
4903
Annex N
4904
4905
(informative)
4906
4907
Maximum Acoustic Pressure Limits
4908
4909
N.1 Abstract
4910
4911
4912
4913
4914
4915
4916
4917
4918
4919
4920
4921
4922
4923
4924
4925
4926
4927
Both North American and European acoustic pressure limits for telephone headsets are under review. Two new
limits at ERP (Ear Reference Point) and DRP (Eardrum Reference Point) are proposed. The new limits are based on
the generally accepted 85 dBA 8-hour TWA (Time-Weighted-Average) free-field exposure limit. The TWA allows
the exposure limit to increase 3 dB for each time the exposure duration is cut in half, e.g. 88 dBA for 4 hours, 91
dBA for 2 hours and so on and so forth. With a 2 second duration (as specified in ITU-T Recommendation P.360)
the allowable free-field exposure level is 127 dBA. Subtract 4 dB from 127 dBA to compensate for narrower
telephony bandwidth compared to the free-field broad frequency bandwidth. The maximum allowable exposure
level for telephone for a 2 second duration is 123 dBA. The new proposed acoustic pressure limits for headset at
ERP and DRP are then obtained by applying the ERP and DRP transfer functions to the 123 dBA free-field limit
across the frequency bandwidth. The proposed limits also suggest adding the A-weighting coefficients to simplify
actual tests.
4928
N.2 Introduction
4929
4930
4931
4932
4933
4934
4935
4936
4937
4938
4939
4940
4941
4942
The two most common telephone headset acoustic pressure limits are the North American frequency dependent
curves at ERP and DRP and the European 118 dBA flat (independent of frequency) at ERP.
The proposal contained in this Annex is a procedure for deriving new telephone headset acoustic pressure limits that
combine the best aspects of both current North American and European limits. The specific numbers and
coefficients, such as the selection of the transfer functions and the damage risk factor should be further examined
and discussed. Hopefully, this proposal will help in resolving years of differences over the proper telephone headset
acoustic pressure limit on both sides of Atlantic Ocean.
The North American limit curves were based on United States OSHA (Occupational Safety and Health
Administration) 90 dBA 8-hour TWA free-field noise exposure limit. OSHA allows the exposure limit to increase 5
dB for each time the exposure duration is cut in half, e.g. 95 dBA for 4 hours, 100 dBA for 2 hours and so on and so
forth. With a 15-minute duration the allowable free-field exposure level is 115 dBA. OSHA regulates the maximum
free-field exposure limit at 115 dBA.
In 1980, Bell Labs published two telephone headset acoustic pressure limits for ERP and DRP in its PUB 48006.
The limits were obtained by transferring the OSHA 115 dBA free-field limit to ERP and DRP and adding Aweighting coefficients across the frequency bandwidth. Presently, the Bell Labs limits are known as the North
American telephone headset acoustic pressure limits. These limits are shown in Figure N. 1Figure N. 1Figure N. 1
and Figure N. 2Figure N. 2Figure N. 2.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
130
IEEE P269/D25 October 2004
140
135
130
dBS PL
125
120
115
110
105
100
100
1000
10000
Frequency (Hz)
4943
4944
4945
4946
4947
Figure N. 1 Bell Labs telephone headset acoustic pressure limit at ERP
140
135
130
dBS PL
125
120
115
110
105
100
100
4948
4949
4950
4951
4952
4953
4954
1000
Frequency (Hz)
Figure N. 2 Bell Labs telephone headset acoustic pressure limit at DRP
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
131
10000
IEEE P269/D25 October 2004
4955
4956
4957
4958
4959
4960
4961
4962
4963
4964
4965
4966
4967
4968
4969
4970
4971
4972
4973
ITU-T Recommendation P.360 explains the derivation of the European 118 dBA limit. This limit was based on an
85 dBA 8-hour TWA free-field noise exposure limit. (Th
i
si
s5dB l
owe
rt
h
a
nOSHA’
s90dBA l
i
mi
t
.
)Th
e
allowable limit increases 3 dB for every halving of exposure duration. (OSHA uses a 5 dB increment.) The
following additional assumptions have been made in ITU-T Recommendation P.360 to adapt these limits to
telephone usage:
a)
For the 2-second duration, the allowable limit is 127 dBA.
b) 10dBi
ss
u
bt
r
a
c
t
e
df
r
omt
h
e127d
BAl
i
mi
tbe
c
a
u
s
eof“
n
on-oc
c
u
pa
t
i
on
a
le
x
pos
u
r
e
”
.
c)
Another 4 dB is subtracted to compensate for narrower telephony bandwidth compared to the free-field
broad frequency bandwidth.
d) 5dBi
sa
dde
dt
ot
h
el
i
mi
tt
oa
c
c
ou
n
tf
or“
s
oun
df
i
e
l
d”di
f
f
e
r
e
n
c
e(
t
h
edi
f
f
e
r
e
n
c
ebe
t
we
e
nERPa
n
df
r
e
e
field).
Thus, 127 –10 –4 + 5 = 118 dBA. The limit is shown in Figure N. 3Figure N. 3Figure N. 3.
140
135
130
dB S PL
125
120
115
110
105
100
100
1000
10000
F requency (Hz)
4974
4975
4976
4977
4978
4979
4980
4981
4982
4983
4984
4985
Figure N. 3 Current European telephone headset acoustic pressure limit at ERP
Bot
ht
h
eNor
t
hAme
r
i
c
a
na
n
dEu
r
ope
a
nl
i
mi
t
sh
a
v
et
h
e
i
rs
t
r
e
n
g
t
h
sa
n
ds
h
or
t
c
omi
ng
s
.TheNor
t
hAme
r
i
c
a
n
’
sERP
and DRP limits were based on OSHA’
soc
c
u
pa
t
i
on
a
ln
oi
s
ee
x
pos
u
r
el
i
mi
t
s
.Th
e90dBA 8-hour TWA free-field
limit has been called too high. Neither the 5 dB increment for every halving of duration, nor the absolute maximum
of115dBA,a
r
ewi
de
l
ya
c
c
e
pt
e
d.Ne
v
e
r
t
h
e
l
e
s
s
,Ame
r
i
c
a
n
’
sme
t
h
odofu
tilizing frequency dependant transfer
functions to transfer the free-field limit to ERP and DRP limits is correct.
The European limit is based on 85 dBA 8-hour TWA free-field limit that is generally accepted. Its increment of 3dB
for every halving of dura
t
i
oni
smor
ea
c
c
e
pt
e
dt
h
a
nOSHA’
s5dBi
n
c
r
e
me
n
t
.Howe
v
e
r
,t
r
a
n
s
f
e
r
r
i
ngf
r
e
e
-field limit
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
132
IEEE P269/D25 October 2004
4986
4987
4988
to ERP by simply adding 5 dB without frequency dependency is hardly justifiable. Subtracting 10 dB from the limit
f
or“
n
on-oc
c
u
pa
t
i
on
a
le
x
pos
u
r
e
”i
sa
l
s
oqu
e
s
t
i
on
a
bl
e
.
4989
N.3 Proposal
4990
4991
4992
4993
4994
4995
4996
4997
4998
4999
5000
5001
5002
5003
5004
The North American and European limits can be combined.
Since 85 dBA 8-hour TWA free-field exposure limit is more accepted globally, it should be adopted for the new
limits. The 3dB increment for every halving of duration is also more generally accepted and should also be adopted.
Applying these assumptions at a 2 second duration as specified in ITU-T Recommendation P.360 gives a limit of
127 dBA in free-field. Subtracting 4 dB from the 127 dBA to compensate the narrower telephony bandwidth, as
done in ITU-T Recommendation P.360, reduces the limit to 123 dBA. Applying the ERP and DRP transfer functions
and adding the A-weighting coefficients to this 123 dBA free-field limit across the frequency bandwidth, as done in
the North American method, produces the new proposed ERP and DRP telephone headset acoustic pressure limits,
shown in Figure N. 4Figure N. 4Figure N. 4 and Figure N. 5Figure N. 5Figure N. 5.
Th
e10d
B“
da
ma
g
er
i
s
k
”r
e
du
c
t
i
ons
pe
c
i
f
i
e
di
nI
TU-T Recommendation P.360 needs to be re-considered. A
reduction to 1/2 of the energy is 3 dB, reduction to 1/4 of the energy is 6 dB. Figure N. 4Figure N. 4Figure N. 4 and
Figure N. 5Figure N. 5Figure N. 5 show the preliminary proposed limits at ERP and DRP and the limits with a 3 dB
and 6 dB safety margin.
P reliminary P roposed H eadset
Acoustic P ressure Limits at E R P
140
135
dBS P L
130
125
120
115
110
105
100
100
1000
10000
Fre que ncy (Hz )
w/o S afety M argin
5005
5006
5007
5008
5009
5010
w/ 3dB S afety M argin
w/ 6dB S afety M argin
Figure N. 4
Copyright © 2004 IEEE. All rights reserved.
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133
IEEE P269/D25 October 2004
P reliminary Proposed H eadset
Acoustic Pressure Limits at D R P
140
135
dBS P L
130
125
120
115
110
105
100
100
1000
10000
Fre que ncy (Hz)
w/o S afety M argin
5011
5012
5013
5014
5015
5016
5017
5018
5019
w/ 3dB S afety M argin
w/ 6dB S afety M argin
Figure N. 5
This proposal offers a procedure for deriving new telephone headset acoustic pressure limits that combine the best
aspects of both current North American and European limits. The specific numbers and coefficients, such as the
selection of the transfer functions and safety margin, should be further examined and discussed.
5020
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
134
IEEE P269/D25 October 2004
5020
Annex O
5021
5022
(normative)
5023
5024
Temporally weighted terminal coupling loss measurement method
5025
5026
5027
5028
5029
5030
5031
5032
5033
5034
5035
5036
5037
5038
5039
5040
5041
5042
5043
5044
5045
5046
5047
5048
5049
5050
5051
5052
5053
5054
5055
5056
5057
5058
5059
5060
5061
5062
5063
O.1 General
The temporally weighted terminal coupling loss (TCLT) measurement method is described for single-talk
application. This method requires that the echo and the source signal be recorded over the duration of the
measurement, and post processing be used. Real-time measurement techniques are possible, but are not described in
this Standard.
Freezing the canceller is not recommended for TCL tests. Test results with non-stationary signals have shown that
convergence times and subsequent converged TCL when "thawed" depend upon the point in time at which the
canceller was frozen.
TCLT, is intended to:
a)
Provide a measure of time dependent echo return loss with peaky behavior, psycho-acoustically weighted
b) Provide an estimate of the number of potentially objectionable echo bursts, and the psychoacoustically
weighted echo return loss during the bursts
c)
Provide several other useful parameters describing echo, including long-term temporally weighted terminal
coupling loss, single talk (LTCLT )
O.2 Initial signal processing
An example test algorithm in pseudo code is detailed in Annex P. The rest of this clause defines the method and
gives some background information.
The echo signal is first filtered to model the frequency sensitivity of human hearing at low levels. The echo and
stimulus files are synchronized. Noise subtraction may then be applied, if it can be assumed that the noise is
stationary and not correlated to the echo. Echo and source are converted into 4 ms power averaged frames allowing
adequate resolution and immunity to synchronization errors.
If the stimulus is inactive, the algorithm simply skips that frame, and moves on to the next echo and stimulus frames.
If the stimulus is declared active, the echo frame is compared with a threshold to determine if an echo event occurs.
The period of echo activity between inactive echo states is termed an echo "event". These events are then weighted
using psycho-acoustic modeling.
By using a threshold of –67.2 dBV (-65 dBm) (5 dB above law noise floor), TCLT can be determined.
5064
O.3 Modeling echo audibility
5065
5066
In modeling echo audibility, the algorithm accounts for 3 fundamental aspects of human hearing behavior:
frequency weighting, temporal combination and temporal weighting.
Copyright © 2004 IEEE. All rights reserved.
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135
IEEE P269/D25 October 2004
5067
5068
5069
O.3.1
Frequency weighting
5070
5071
5072
5073
5074
5075
5076
The frequency sensitivity of human hearing at a loudness level of 30 Phons is approximated. (30 Phons is equivalent
to 30 dB at 1 kHz).
5077
O.3.2
5078
5079
5080
5081
5082
5083
5084
5085
5086
5087
5088
5089
Temporal combination is the ear's tendency to combine the loudness of sequential signals even though they may be
discrete in time. This typically occurs when the two signals are separated by less than about 20 ms. The exact time
is a complex function of many variables, but 20 ms is a suitable value in this application. This is sometimes referred
to as the Haas effect.
30 Phons was chosen as it represents echo levels that result from terminals that just fail handset terminals coupling
loss specifications (determined using loss planning analysis). Variance from 20 to 50 Phons provide essentially the
same weighting within the telephony band. An A-weighting filter is used. See ANSI S1.4-1983 (R 1997).
Temporal combination
If two bursts of echo are separated by a period of inactivity less than 20 ms, they are considered as one longer echo
event as far as loudness is concerned. This continues until the gap between events is at least 20 ms, at which time
the echo event is declared over. This can be thought of as a 20 ms hangover for the current echo event. During this
hangover period, echo and stimulus powers are not included as part of the event. An example of temporal
combination follows in Figure O. 1Figure O. 1Figure O. 1:
New Echo
Duration of Echo = 100 ms
Echo
Amplitude
(Power)
Temporal
Combination
20
Activity
Threshold
40
60
80
100
120
Time (20 ms/div)
5090
5091
Figure O. 1 Temporal combination
5092
5093
5094
O.3.3
Temporal weighting
5095
5096
5097
5098
5099
Te
mpor
a
lwe
i
gh
t
i
ngmode
l
st
h
el
i
s
t
e
n
e
r
’
sr
e
du
c
e
ds
e
ns
i
t
i
v
i
t
yt
os
oun
dsa
st
h
e
i
rdu
r
a
t
i
onde
c
r
e
a
s
e
sbe
l
ow 750ms
.
The exact time is a complex function of many variables, but 750 ms is a suitable value in this application.
The duration of the total echo event after temporal combination is measured. The total duration includes any gap(s)
between events that are captured by temporal combination, but not the final 20 ms hangover. If the total duration is
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
136
IEEE P269/D25 October 2004
less than 750 ms, the level of the event is reduced to account for the temporal integration behavior of human
hearing. If the duration is longer than 750 ms, the level of the total event is left unweighted. Test results have shown
echo bursts less than 750 ms to be common occurrences from cancellers.
A simplified equation (Equation O.1) describing the relationship was derived based upon audition studies with
noise. (Tones result in a slightly different relationship, but it was felt that noise was a much closer approximation to
the true nature of the echo than a sine.)
Temporal integration weighting = -23 + 8log(t), in dB
Equation O.1
Where:
t = total duration of echo event (ms), t 750 ms
A graphical representation of temporal weighting is shown in Figure O. 2Figure O. 2Figure O. 2.
Relative Loudness Level (dB)
5100
5101
5102
5103
5104
5105
5106
5107
5108
5109
5110
5111
5112
5113
5114
5115
5116
5117
0
-10
-20
Broadband Noise
10
5118
5119
5120
5121
5122
5123
5124
100
Duration (ms)
1000
Figure O. 2 Temporal weighting
5125
O.4
Expressing TCL Results
5126
5127
5128
5129
5130
5131
5132
5133
Traditional TCL methods refer the echo power during the duration of measurement to the source power during the
duration of measurement to arrive at the terminal coupling loss. In the TCLT method, the final weighted power of
echo during each event is referred to the power of the source signal during the same event, to arrive at the "Active
TCLT", or ATCLT, of each event. The echo is referred to the source signal during the event only, as this is the way
in which our ear would compare the echo. This parameter can be statistically analyzed to give information about
echo events during the entire test sequence.
Copyright © 2004 IEEE. All rights reserved.
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137
IEEE P269/D25 October 2004
5134
5135
5136
5137
5138
5139
5140
5141
5142
5143
5144
5145
5146
5147
5148
5149
5150
5151
5152
5153
5154
A long term average of the weighted active echo return loss is found by summing the power of all weighted echo
during active events, and comparing to the power of the source as seen during all events only. The result is the
"Active Long Term TCLT," or ALTCLT.
For comparison with traditional TCL methods, the power of all weighted echo during events is summed, then
referred to the total source power as measured for the entire duration of the measurement. The result is the "Long
Term TCLT," or LTCLT.
The terminology for TCLT results was chosen to be consistent with the nomenclature of ITU-T recommendation
P.56 (1993).
Other statistics compiled by the algorithm in Annex P include minimum, maximum, mean and standard deviation of
ATCLT, the total number of echo events, the number of echo events per minute, the percentage of echo event free
speech, number of events < 750 ms, and the average length of an event and the duration of source inactivity.
5155
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
138
IEEE P269/D25 October 2004
5155
Annex P
5156
5157
(normative)
5158
5159
Temporally weighted terminal coupling loss algorithm
5160
5161
5162
5163
5164
5165
5166
5167
5168
5169
5170
5171
5172
5173
5174
5175
5176
5177
5178
5179
5180
5181
5182
5183
5184
5185
5186
5187
5188
5189
5190
5191
5192
5193
5194
P.1 General
This algorithm is an example of how to implement the measurement of TCLT as defined in Annex O. The algorithm
is provided as an assistance to the test developer, but it is not the definition of TCLT. Modifications to this
algorithm may be made, and may be necessary, to completely fulfill the intent of Annex O.
TCLT is a newly proposed method for evaluating the echo return loss of a terminal using psychoacoustic modeling
and for predicting the occurrences of potentially objectionable echoes. The principles are defined in Annex O. It
incorporates 3 fundamental aspects of human audition:
a) frequency sensitivity of human hearing for low-level sounds
b) temporal addition of level for events within 20 ms of each other
c) temporal integration for stimuli below 750 ms
Speech based stimulus signals are recommended as their results are most representative of real world usage. The
measured output from the telephone is always some echo or noise making its way through the system uncancelled.
It may be useful to record the stimulus and echo in digital format. Echo and stimulus frames shall be calibrated
according to the principles of this standard. It is possible to apply this method to 2-wire analog sets by use of a test
hybrid. (See IEEE Std 1329-1999)
The stimulus and the echo files will be processed as power values averaged over 4 ms frames. The successive
stimulus file frames will be termed xi, the echo frames will be denoted yi, where i = 1, 2, 3.... is the actual frame
index. Intermediate frames conforming to an "echo event" will be noted as xk, and yk, where k = 1, 2, 3... is the
echo event index, and is reset when the event ends a new one commences.
Statistics compiled during the TCLT measurement include the Active Long Term TCLT (ALTCLT), Long Term
TCLT (LTCLT), minimum and maximum Active TCLT (MINTCLT, MAXTCLT), its standard deviation (sigma) and
mean, the total number of echo events (NEVENTS), the number of echo events per minute (NEVMIN), the
percentage of echo event free speech (PER), number of events < 750 ms (N750), the average length of an event
(AVGEVENT), and the duration stimulus was inactive (DUR). The terminology for TCLT results was chosen to be
consistent with the nomenclature of ITU-T recommendation P.56 (1993). The duration of stimulus inactivity is not
included in the time-based results.
5195
P.2 TCLT Algorithm
5196
5197
Stimulus and measured results shall be calibrated according to the requirements in Clauses 6-9.
5198
P.2.1
5199
5200
Calculate the correlation of stimulus and echo file to fine tune EPD (echo path delay). See Annex L for methods.
Use the criteria that the present correlation peak occurs at EPD unless a following correlation peak has a magnitude
Step 1: measure EPD
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
139
IEEE P269/D25 October 2004
5201
5202
5203
at least 10 dB greater. This approximate guideline is based upon subjective studies on delay detection with multiple
impulses.
5204
P.2.2
5205
5206
Align the echo and stimulus files in time by removing delay equal to EPD from the echo file.
5207
P.2.3
5208
5209
The individual echo samples are A-weighted filter. (See ANSI S1.4-1983 (R 1997).
5210
P.2.4
5211
5212
5213
5214
5215
If it can be assumed that the noise in the echo path is stationary and uncorrelated with the echo, the noise is
measured for 2 seconds after the stop of source and echo activity. The noise is then subtracted, on a power basis,
from the echo plus noise to arrive at a better estimate of the echo alone. This procedure shall be performed only if
the echo plus noise is at least 3 dB greater than the noise alone.
5216
P.2.5
5217
5218
5219
5220
Samples are converted to absolute numbers using the calibration data. The stimulus samples are combined into 4 ms
power averaged frames denoted as xi. The weighted, noise filtered echo samples are combined into 4 ms power
averaged frames denoted as yi.
5221
P.2.6
5222
5223
5224
5225
5226
5227
5228
5229
5230
5231
5232
5233
5234
5235
5236
5237
5238
5239
5240
5241
5242
5243
5244
Initialize variables:
i = 0 (frame counter)
j = 0 (frame counter for inactive signal duration)
nk=0 = 0 (number of frames in current echo event)
NSAMPS = 0 (accumulated number of frames for all events)
HAAS = 0 (counter up to 20 ms)
ei=0 = 0 (running summation of all echo power for all events after weighting, as seen at frame counter i)
pi=0 = 0 (running summation of all stimulus power during the measurement, as seen at frame counter i)
ek=0 = 0 (running summation of echo power during the particular echo event after weighting, as seen at event
frame counter k)
sk=0 = 0 (running summation of stimulus power during the particular echo event after weighting, as seen at
event frame counter k)
WEIGHT = 0 (temporal based weight of most recent event)
LEVENT = 0 (echo return loss level of most recent event, after weighting)
NEVENT = 0 (total number of echo events)
N750 = 0 (total number of echo events < 750 ms)
MINTCL = 75 (minimum echo return loss level of all events)
MAXTCL = 0 (maximum echo return loss level of all events)
EVENT[NEVENT] = 0 (initialize array for all event loss levels (in dB) to zero; used to calculate sigma)
TEMPSK = 0 (running sum of stimulus power during all events)
SUM = 0 (used in calculating sigma)
SQ = 0 (used in calculating sigma)
Step 2: align signals
Step 3: apply A-weighting
Step 4: subtract noise (conditional)
Step 5: 4ms frames
Step 6: initialization
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
140
IEEE P269/D25 October 2004
5245
P.2.7
Step 7: calculations
5246
5247
5248
5249
5250
5251
5252
5253
5254
5255
5256
5257
5258
5259
5260
5261
5262
5263
5264
5265
5266
5267
5268
5269
5270
5271
5272
5273
5274
5275
5276
5277
5278
5279
5280
5281
5282
5283
5284
5285
5286
5287
5288
5289
5290
5291
5292
5293
5294
5295
5296
5297
5298
Increment frame counter and read in 4 ms averaged echo power yi, and 4 ms averaged stimulus power, xi; if there
are no more valid inputs and either measurement file is complete, go to Step 8: calculate parameters.
1 i = i +1 (unless last i, then go to Step 8: calculate parameters)
Sum stimulus powers
pi = pi + xi
Is stimulus loud enough for a valid echo loss calculation? If not, disregard present frame and move to next frame.
4 If xi < (long term stimulus rms level - 25 dB)
j=j+1
i=i+1
Go to 4
Else
Test echo against threshold
If yi -67.2 dBV (-65 dBm) _ (5 dB above law noise floor)
Increment frame event counter
k = k +1
Increment frame event length including any gaps < 20 ms
nk = nk +1 + HAAS
Reset "Haas kicker"
HAAS = 0
Accumulate echo power of event
ek = ek + yi
Accumulate stimulus power during event
sk = sk + xi
Go to 1
Else
Has there been no event within last 20 ms?
If k=0
HAAS = 0
Go to 1
Else
There has been an event within the last 20 ms
HAAS = HAAS + 1
Has 20 ms without an event elapsed after a recent event?
If HAAS*4 < 20
Go to 1
Else
An event is over, add an event to the event counter
NEVENT = NEVENT + 1
Increment the total events duration counter by adding the duration in frames of the most recent event
NSAMPS = NSAMPS + nk
Was the most recent event duration < 750 ms?
If nk*4 < 750
Calculate temporal integration weighting for most recent echo event
WEIGHT = 8*log10(nk*4) - 23
Increment the counter for the number of events that were temporally weighted
N750 = N750 +1
Else
Calculate weighted echo return loss of the most recent event in dB
LEVENT = 10*log10(sk/ek) - WEIGHT
Store the minimum and maximum echo return losses in dB
IF LEVENT < MINTCL; MINTCL = LEVENT
IF LEVENT > MAXTCL; MAXTCL = LEVENT
Store the echo return loss of the most recent event in dB for future sigma calculation
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
141
IEEE P269/D25 October 2004
5299
5300
5301
5302
5303
5304
5305
5306
5307
5308
5309
5310
5311
5312
5313
5314
EVENT(NEVENT) = LEVENT
Reconvert the echo return loss of the most recent event into linear; recalculate weighted linear echo power
ek = sk/(10**(LEVENT/10))
Accumulate all the echo event powers for future use in calculating ALTCLt and LTCLt
ei = ei + ek
Accumulate all the stimulus powers during events for future use in calculating ALTCLt
TEMPSK = TEMPSK + sk
Reset echo event variables
k=0
nk = 0
WEIGHT = 0
HAAS = 0
ek = 0
sk = 0
Go to 1
5315
P.2.8
5316
5317
5318
5319
5320
5321
5322
5323
5324
5325
5326
5327
5328
5329
5330
5331
5332
5333
5334
5335
5336
5337
5338
5339
5340
Calculate Active Long Term TCLt (ALTCLt), Long Term TCLt (LTCLt), the number of echo events per minute
(NEVMIN), the percentage of echo event free speech (PER), the average length of an event (AVGEVENT) and
duration during which speech was inactive (DUR).
5341
P.2.9
5342
5343
5344
Step 8: calculate parameters
Note: Zero check ei before computing; if ei = 0, set ALTCLt and LTCLt to 100 dB.
ALTCLt = 10*log10(TEMPSK/ei)
LTCLt = 10*log10(pi/ei)
NEVMIN = 60*NEVENT/((i-j)*0.004) {number of events per minute}
PER = 100*((i-j) - NSAMPS)/(i-j)
{percentage of echo free speech)
AVGEVENT = NSAMPS*4/NEVENT {average length of an event in milliseconds}
DUR = j**0.004
Calculate sigma by analyzing the EVENT array which contains the echo return loss of each event; each event,
regardless of duration, is given equal weighting in the sigma calculation; the suggestion is that it is the transition
between discreet events and not their duration that is most objectionable.
Loop j from 1 to NEVENT
SUM=SUM+EVENT(j)
SQ=SQ+EVENT(j)**2
ENDLOOP
SIGMA = SQRT(SQ/NEVENT - [SUM/NEVENT]**2)
Calculate mean of the events
MEAN = SUM/NEVENT
Step 9: output statistics
Print ALTCLt, LTCLt, MINTCL, MAXTCL, NEVENT, NEVMIN, PER, N750, AVGEVENT, DUR, SIGMA,
MEAN
5345
Copyright © 2004 IEEE. All rights reserved.
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5345
Annex Q
5346
5347
(normative)
5348
5349
5350
5351
5352
5353
5354
5355
5356
5357
5358
5359
5360
5361
5362
5363
5364
5365
5366
5367
5368
5369
5370
5371
5372
5373
5374
5375
5376
5377
Simulated Speech Generator (SSG)
Main Signal The main signal consists of eight 1024-point pseudo-random noise segments. Each segment has the same magnitude
spectrum but a different phase spectrum with the phase randomized within and between the segments uniformly
from 0 to 360 degrees, in order to randomize the interaction between the intermodulation products of the
harmonically related spectral components. The duration of each segment is 80 ms. They are merged with each other
through a raised cosine window, with an additional 80 ms. merging segment between them. The simultaneous fadeout of the previous segment and the fade-in of the following segment eliminate the transients, which would occur at
the segment boundaries. The complete main signal thus consists of eight pseudo-random segments interleaved with
eight merging segments, each of 80 ms. Duration, having a total length of 1.28 seconds. A simple filter at the output
provides the desired frequency shaping to approximate an average speech spectrum.
Modulating Signal Measurements show that a Gamma distribution with parameter m = 0.545 provides a good approximation to the
instantaneous amplitude distribution of continuous speech. The syllabic characteristics can be represented by a low
pass response that is practically flat up to about 4 Hz (the -3 dB point) followed by -6 dB per octave roll-off.
The final wave shape of the modulating signal was derived empirically from the Gamma distribution. Varying the
period of this pulse in a pseudo-random manner and adjusting its rise and fall time ratio results in a satisfactory
approximation to the spectrum of the modulation envelope of real speech.
Combined Signal In order to extend the repetition time of the final signal and to spread more evenly the maxima of the modulating
signal over the repeated sequence of the Gaussian signal, the ratio between the sampling clock frequencies of both
signals was chosen to be 4/255. Thus the clocking frequency of the main signal is 12,800 Hz, and the clock
frequency for the modulating signal is about 200.8 Hz. The repetition times are: .28 seconds for the Gaussian signal,
10.2 seconds for the modulating signal and 326.4 seconds for the final modulated signal.
Main
Signal Source
(Gaussian)
Shaping
Filter
Output
Modulating
Signal Source
(Gamma)
5378
5379
5380
5381
5382
5383
5384
Figure Q. 1 Block diagram of simulated speech generator
Gaussian Signal Generator -
Copyright © 2004 IEEE. All rights reserved.
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143
IEEE P269/D25 October 2004
5385
5386
5387
5388
5389
5390
5391
5392
The Gaussian signal is made up of sixteen segments. The odd number segments are generated by filling a 2 by n
array with zeros and then filling in the desired real and imaginary spectrum components using equations one and
two. The first entry is zero i.e. no DC component and there are no components above 5500 Hz.
5393
5394
5395
5396
5397
5398
5399
5400
5401
Equation Q. 2
5402
5403
5404
5405
5406
5407
5408
5409
5410
5411
5412
5413
5414
5415
5416
5417
5418
5419
5420
5421
5422
5423
5424
5425
5426
5427
5428
5429
5430
5431
5432
X r 
 cos
2
Equation Q. 1
X i 
 sin
2
where:
is a random number with uniform dostribution 0 1
The inverse FFT is then taken to transfer the signal to the time domain.
x 
n  X r 
 X i 

Equation Q. 3
The even number segments S(n) are:
Si(n) =Si(n-1)*0.5(1+cos(((i-0.5)/1024) + Si(n+1)*0.5(1-cos(((i-0.5)/1024)
i = 1 to 1024
n = 2, 4...,16 for n+1>16 use n+1-16
Gamma Function:
For the Gamma function the 2048 samples are divided into 21 random length pulse periods (number of samples).
The periods are 167, 43, 63, 119, 48, 57, 78, 88, 93, 107, 51, 71, 259, 60,67, 207, 143, 54, 130, 45, 98. Each period
is divided into rise time of one third and a fall time of two thirds. That is, rise and fall times are in 1:2 ratio.
The cubic interpolating spline function is used to model the rising and falling section of each segment.
First calculate the coefficients B(I), C(I), D(I) for I =1 to 60 for a cubic interpolating spline (G.E. Forsythe, M.A.
Malcolm, and C.B. Moler [L3]). The number of points (knots) is 60. The abscissas of the knots, in increasing order,
range in value from 0.05648176 to 0.983219. Y is the ordinate of the knots. Y (I) =I-0.5.
where:
n = number of samples in the rising (or falling) section
s(i) is the value of the ith data point in the period
For the rising time period:
s(i) = spline value at abscissa (-0.5/n)+(1/n*i)
For the falling time:
s(i) = spline value at abscissa (-0.5/n)+(1/n*(n+1-i)
5433
Copyright © 2004 IEEE. All rights reserved.
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144
IEEE P269/D25 October 2004
5433
Annex R
5434
5435
(normative)
5436
5437
5438
5439
5440
5441
5442
5443
5444
5445
5446
5447
5448
5449
5450
5451
5452
5453
5454
5455
5456
5457
5458
5459
5460
5461
5462
5463
TDS Sweep with P.50 Noise Bursts
The bias signal consists of P.50 noise (F.5.3). For send measurements, it is presented in bursts at a 4 Hz rate and
50% duty cycle (125 ms "ON", 125 ms "OFF"). The bias is presented at the standard test level during the "ON"
bursts.
For receive measurements, the bias may be presented either continuously or in the burst pattern. Continuous
presentation may be the most appropriate bias of a telephone with a simple AGC function, but burst presentation
may be better for telephones with more complex functions. Ideally, both ways should be measured to determine
which gives the most typical results. The telephone will be measured in its average state during the entire
measurement.
The measurement signal is a series of sine sweeps from 100 to 8500 Hz, at any rate suitable for Time Delay
Spectrometry (TDS) measurements. The sweeps are not synchronized with the bias pulses. The sweep spectrum may
approximate the P.50 spectrum. At 315 Hz, the level of the measurement signal is 15 dB below the overall level of
the bias signal.
The measurement is performed by TDS (G.4.3). The sweep length and number of averages are adjusted to obtain a
satisfactory signal-to-noise ratio in the measurement. Typically, a measurement time (sweep length times number of
averages) in the range of 16 to 128 seconds gives good results.
The true frequency resolution of the TDS measurement will be determined by the time window chosen, not by the
frequency interval in the analyzer. The minimum effective time window is 5.7 ms, which corresponds to a frequency
resolution (lowest measurable frequency) of 175 Hz.
In principle, this method can be used with any desired bias signal, including any of the speech-like signals (see F.6).
5464
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
145
IEEE P269/D25 October 2004
5464
Annex S
5465
5466
(informative)
5467
5468
5469
5470
5471
5472
5473
5474
5475
5476
5477
5478
5479
5480
5481
Use of the Free Field as the Telephonometric Reference Point
Current performance requirements are based on measurements referred to the ERP. Future requirements may be
based on measurements referred to the free field. This annex provides background information on this concept.
One goal of a telephonic experience is to simulate a conversation where two people are one meter apart, talking to
each other. Now insert a complete telephone system between our two talkers. In a perfect world, the quality of the
conversation would be the same with a telephone system and in free space. This is called the orthotelephonic
reference.
Consider a loudspeaker with a perfectly flat free field frequency response through the audio band (Figure S. 1Figure
S. 1Figure S. 1):
Speaker
Microphone
Free Field Response (simplified)
100Hz
5482
5483
5484
5485
5486
5487
10K
Figure S. 1
Play the same speaker into a HATS ear simulator, and the result is a 17 dB peak at 2.8 kHz. (For more complete
data, see ITU-T Recommendations P.57 and P.58). See Figure S. 2Figure S. 2Figure S. 2.
Speaker
HATS
Eardrum (DRP) to Free Field Transfer function (simplified)
100Hz
5488
5489
5490
5491
1K
1K
Figure S. 2
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
146
10K
IEEE P269/D25 October 2004
5492
5493
5494
5495
5496
5497
The HATS ear simulator replicates the resonances which occur in a typical human pinna and ear canal system, and
measures at the (ear) Drum Reference Point or DRP. It is because of the pinna and the resonances in the ear canal
that a loudspeaker with a flat free field response will not measure flat into a HATS, at the DRP. Therefore, if a
telephone receiver or headset is to sound the same as a hypothetical flat speaker in the free field, the frequency
response at the DRP should follow the freefield curve(s) referenced in ITU-T Recommendation P.57/58. (Figure S.
3Figure S. 3Figure S. 3)
Phone on HATS
Eardrum (DRP) to Free Field Transfer Function (simplified)
100Hz
5498
5499
5500
5501
5502
5503
5504
5505
5506
10K
Figure S. 3
Most telephone companies are more familiar with the Type 1 ear simulator. This type of simulator uses the Ear
Reference Point (ERP) rather than the DRP, which results in a different frequency curve shape. Using the above
hypothetical receiver tested into a Type 1 ear simulator yields a frequency response which looks like Figure S.
4Figure S. 4Figure S. 4:
Phone on Type 1 Ear Simulator
5507
5508
5509
5510
5511
5512
5513
5514
5515
5516
5517
5518
5519
1K
ERP to Free Field Transfer Function (simplified)
100Hz
1K
10K
Figure S. 4
The important thing to remember is that the above curves, using either Type 1 simulators or HATS, all can be
referenced to a free field response. Another way of looking at it is that if you want your handset or headset to sound
like a flat loudspeaker in the free field, e.g. simulating the orthotelephonic reference, the frequency response should
look like either the ERP or DRP to free field transfer function curves above.
The complete orthotelephonic response is due to the combination of frequency responses in the send, network, line,
and receive paths of an overall (end-to-end) connection. The exact distribution of frequency response shaping in
these paths is outside the scope of this Annex.
5520
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
147
IEEE P269/D25 October 2004
5520
Annex T
5521
5522
(informative)
5523
5524
Useful Conversion Procedures
5525
5526
5527
5528
5529
5530
5531
5532
5533
5534
5535
5536
5537
5538
5539
5540
5541
5542
5543
5544
5545
5546
5547
5548
5549
5550
5551
5552
5553
5554
5555
5556
5557
5558
5559
5560
5561
5562
5563
5564
5565
5566
5567
T.1 Conversions for dBV to dBm, and for 600 and 900
0 dBm is accepted as 1 mW, typically using a circuit impedance of 600 ohms or 900 ohms.
0 dBm = 10 log 1(mW)
dBV = 10 log V2
= 20 log V
For R = 600 ohms:
P = V2/R, therefore
dBm = 10 log ((V2/R) * 1000)
= 10 log ((V2/600) * 1000)
= 10 log V2/0.600
So, V = 774.6 mV or 0 dBm -2.22 dBV
For R = 900 ohms:
P = V2/R, therefore
dBm = 10 log ((V2/R) * 1000)
= 10 log ((V2/900) * 1000)
= 10 log V2/0.900
So, V = 948.7 mV or 0 dBm -0.46 dBV
To change from 600 ohms to 900 ohms or vice versa, for a constant voltage:
Correction (dB) = -10 log (0.600/0.900) = 10 log (0.900/0.600) = 1.76 dB
Correction (dB) = 10 log (|Z1| / |Z2|), i.e., the log of the ratio of the magnitude of the impedances, when converting
from impedance Z1 to Z2.
If converting from " Z1 = 600 ohms" to " Z2 = 900 ohms", the correction factor is -1.76 dB, thus we subtract 1.76 dB
from the measurement.
Depending on the impedance being used, conversion factors can be applied dB for dB to the measured or calculated
result.
Example 1: To convert a 600 ohm -20 dBm signal to dBV, simply subtract 2.22 to get -22.2 dBV.
Copyright © 2004 IEEE. All rights reserved.
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148
IEEE P269/D25 October 2004
5568
5569
5570
Example 2: -20 dBm is measured across 600 ohms. To find the level across 900 ohms, add a correction of -1.76 dB
to get -21.76 dBm (since the larger load dissipates less power).
5571
T.2 Conversions for dBmp to dBrnC for electrical noise measurements
5572
5573
5574
5575
5576
5577
5578
5579
5580
5581
5582
Two weighted noise measurement units have typically been used in telephony, dBmp and dBrnC. The main
differences between these two measurement units are the shape of the weighting filter and the reference unit. The
weighting filter for dBrnC is described in IEEE Standard 743-1995.
The differences in the weighting functions are extremely slight, as to be insignificant; thus the conversion between
the two units can be expressed as:
dBrnC = dBmp + 90
5583
T.3 Loudness rating conversions
5584
5585
5586
5587
5588
5589
5590
5591
5592
5593
5594
5595
5596
Conversion from loudness ratings defined in IEEE Standard 661-1979 to those defined in ITU-T Recommendation
P.79 (1993), as specified by ANSI/TIA/EIA-810-A-2000, is as follows:
5597
T.4 Acoustic sound pressure conventions
5598
5599
5600
5601
5602
5603
5604
5605
5606
dBPa (dB pascals)
dBSPL (dB Sound Pressure Level)
SLR (P.79) = TOLR (IEEE 661) + 57
RLR (P.79) = ROLR (IEEE 661) - 51
STMR (P.79) = SOLR (IEEE 661) + 9
The above conversions should be used as an approximation only. These conversions are based upon approximated
frequency response curves as specified in ANSI/TIA/EIA-810-A-2000. Proper conversion may depend upon actual
measurements being made with each measurement standard where frequency responses deviate significantly from
the norm.
Where,
0 dBPa 94 dBSPL, and 0 dBSPL 20 micropascals, 1 Pa = 1 N/m2
An A-we
i
g
h
t
e
ds
ou
n
dpr
e
s
s
u
r
el
e
v
e
li
ndB(
r
e20mi
c
r
opa
s
c
a
l
s
)i
sof
t
e
na
bbr
e
v
i
a
t
e
da
s“
dBA”
.
5607
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
149
IEEE P269/D25 October 2004
5607
Annex U
5608
5609
(informative)
5610
5611
Loudness Balance Subjective Test Procedure
5612
5613
U.1 Introduction
5614
5615
5616
5617
5618
5619
5620
5621
5622
5623
5624
5625
5626
The results of a subjective loudness balance test procedure may be used to estimate the receive loudness in those
cases where objective measurements do not correlate well with real use performance. This loudness balance
subjective test procedure differs in specific test details but is similar to the CCM laboratory "Contra-Lateral
Balance" procedure. The procedure has been used to obtain loudness differences between a reference headset
receiver and four test headset receivers. All of the headsets had on-ear type receivers. The standard deviations of the
loudness balances obtained from 10 subjects ranged from 1.8-4.9 dB, and averaged 2.7 dB over 23 trials. (A trial
consisted of four loudness balances for each of 10 subjects for one sound source and one test headset.) The accuracy
of the average loudness differences obtained in the tests for the four test headsets was represented by 95%
confidence intervals about the average of 2.1 dB.
5627
U.2 Loudness balance test procedure
5628
5629
5630
5631
5632
5633
5634
5635
5636
5637
5638
5639
5640
5641
5642
5643
5644
5645
5646
5647
5648
5649
5650
5651
5652
5653
5654
5655
The loudness balance procedure is used to obtain loudness differences between a test and reference headset. The
receiver in the reference headset shall have objectively measured performance which correlates well with its real-use
receive performance. With the type of artificial ears currently available, this requires a tight acoustic seal between
the receiver and artificial ear during objective measurements, and between the receiver and human ear in real use.
The methods described in this clause have been successfully used with headsets. In principle, similar methods can
be used with handsets.
The loudness balance tests should be performed in a quiet room with background noise no greater than 40 dBA. A
loudness balance between the reference and test headsets is obtained by allowing the subject to adjust the signal
level to the test receiver until loudness of the sound from the test receiver is judged equal to the loudness of the
sound from the reference receiver. During this determination, the test receiver is on one ear and the reference
receiver is on the other ear. After the loudness balance is determined, the loudness difference between the test and
reference receivers is represented by the difference in signal levels to the two receivers. To counteract the effects of
hearing acuity differences between the subject's left and right ears, the tests should be repeated with the test and
reference headset receivers reversed on the subjects ears. The results of the two trials are averaged to determine the
loudness difference. To obtain reasonably reliable data, a minimum of 10 subjects should be used in the tests. These
t
e
s
ts
u
bj
e
c
t
ss
h
ou
l
dh
a
v
e“
c
l
i
n
i
c
a
l
l
yn
or
ma
l
”h
e
a
r
i
n
g
.Th
a
ti
s
,t
h
ema
g
n
i
t
u
deofme
a
s
u
r
e
dh
e
a
r
i
ngl
os
sa
ta
nyt
e
s
t
frequency shall be less than 30 dB. If possible, each test subject should have approximately equal hearing in both
ears.
The loudness differences should be determined for six different signal sources consisting of one-third octave band
noise centered at the following frequencies: 315 Hz, 500 Hz, 800 Hz, 1250 Hz, 2000 Hz, and 3150 Hz. The use of a
narrow band of noise is preferred over pure tones since sounds that are normally heard are more complex than pure
tones. Furthermore, subjects may adapt to pure tones after a short period of listening. This could result in inaccurate
measurements. This adaptation is less likely when narrow band noise is used.
Two loudness balances are made for each of the six signals for each ear. The signal sources are presented in a
random order to the subject. The subject determines a loudness balance by adjusting an attenuator that controls
signal level to the test headset receiver, while alternating the signal between the test and reference receivers with a
switch. With each new signal, the starting signal level in the test receiver should always be below that of the
Copyright © 2004 IEEE. All rights reserved.
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150
IEEE P269/D25 October 2004
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5657
5658
5659
5660
5661
5662
5663
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5665
5666
5667
5668
5669
5670
5671
5672
5673
5674
5675
5676
5677
5678
reference receiver. That is, the subject should always initially need to increase the test receiver level to arrive at a
loudness balance. After completing the loudness balances for the first receiver - ear placement, the receivers are then
reversed on the subject's ears and the tests repeated.
5679
U.3 Example test circuit
5680
5681
5682
5683
5684
5685
5686
5687
5688
A block diagram of an example test circuit for implementing the loudness balance tests is given in Figure U. 1Figure
U. 1Figure U. 1. Amplifiers 1, 2, and 4 provide an impedance transformation function as well as providing gain.
Amplifier 1 converts from the signal source impedance to the 600 ohm circuit impedance while Amplifier 2 and 4
convert from the circuit impedance to the headset receiver impedance, which is 300 ohm for this example. Switch 1
is a hand-held push button switch, which enables the subject to alternate the signal between the test and reference
headsets. Attenuator 1 is adjusted by the subject to attain preferred listening levels in the reference headset receiver.
Attenuator 2 is adjusted by experimenter to randomly shift the balance point. Attenuator 3 is adjusted by the subject
to attain a loudness balance between the test and reference receivers.
Loudness differences between some test headsets and the reference headset may be similar for the six test sounds.
The subject may thus learn during the test to set the balance attenuator at a specific location to achieve a loudness
balance. However, the subject's final decision may be influenced more by what he or she thinks is the correct
position to produce a balance than by the actual balance itself. To prevent any such biasing of the results, a means
should be incorporated in the test design to randomly shift the loudness balance point.
Before the tests begin, the subject should be given ample time to adjust the headsets to his or her ears. The
importance of proper receiver-to-ear coupling should be stressed to the subject and directions given not to change
the positioning of the receivers once the tests begin. Each test subject should adjust the signal level to the reference
receiver for his or her preferred listening level. After the receivers have been properly positioned, the 1250 Hz sound
source should be directed to the reference headset and the subject instructed to adjust an attenuator until the sound is
at his or her preferred level. This level for the reference headset should then remain constant for all sound sources
for that subject. (When the test and reference receivers are reversed on the subject's ears, the subject is again asked
to adjust the attenuator for preferred listening level.) In those cases where the test headset incorporates receive
compression, it is necessary to determine, in pre- tests with the test and reference receivers, the signal level for the
reference receiver. This level, which will probably be below the preferred level of most subjects, should be such that
it prevents the acoustic output of the test receiver from being limited for at least 10 dB or so above the expected
balance point for the six signal sources.
Am plifier 2
TP 1
Attenuator 1
Source
Signal
Am plifier 1
Reference
Headset
---Reference
Switch 1
---Test
Attenuator 2
TP 2
Attenuator 3
Am plifier 3
5689
5690
5691
5692
5693
5694
5695
5696
Am plifier 4
Test
Headset
Figure U. 1 Loudness Balance Test Circuit
For circuit line-up, the test and reference headset receive jacks are terminated at 300 ohm (in the example). Using a
1000 Hz tone, the gain of the amplifiers is adjusted, such that when the gain of Amplifier 3 is numerically equal to
the sum of the losses of Attenuators 2 and 3, the voltage levels at TP1 and TP2 are equal.
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
151
IEEE P269/D25 October 2004
5697
5698
5699
5700
5701
5702
5703
5704
5705
After a loudness balance has been attained by the subject, the loudness difference between the test and reference
headset receivers is represented by the difference in the voltage levels at TPI and TP2. The loudness difference is
also represented by the difference between the sum of the dB losses of Attenuator 2 and 3 and the dB gain of
Amplifier 3. For example, assume an amplifier gain of 15 dB and a total loss of 16 dB for Attenuator 2 and 3. The
loudness difference would be 16 –15 = 1 dB, the test receiver is 1 dB louder than the reference receiver. The gain of
Amplifier 3 should be determined in pre-tests with the test and reference headsets so that the combination of gain in
Amplifier 3 and loss in Attenuator 2 and 3 provide a-maximum range of adjustment on either side of the estimated
loudness balance point.
5706
U.4 Estimate of test headset receive characteristics
5707
5708
5709
5710
5711
5712
5713
5714
5715
5716
5717
5718
5719
5720
5721
5722
5723
To estimate the receive characteristics of the test headset, the receive characteristics of the reference headset shall
first be objectively measured. The measurement bands are the same as specified for the loudness balance procedure:
315 Hz, 500 Hz, 800 Hz, 1250 Hz, 2000 Hz, and 3150 Hz. The desired results from the objective measurements are
the receiver output pressures, in dB SPL, at the six test frequencies.
The loudness difference between the test and reference receivers, at each of the test frequencies, is calculated by
averaging the 40 loudness differences (2 repetitions/2 ears/10 subjects) obtained at each test frequency. The
estimated output pressure for the test headset at each test frequency (assuming the same input voltage as had been
used to objectively measure the reference receiver) is calculated by:
TREP = RROP + LD
where
TREP
RROP
LD
is the test receiver estimated pressure in dB Pa
is the reference receiver objective pressure in dB Pa
is the loudness difference between test and reference receivers in dB
Copyright © 2004 IEEE. All rights reserved.
This is an unapproved IEEE Standards Draft, subject to change.
152

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